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| 00:52.32 | jblack | wwalker: That is a good question. |
| 00:52.55 | jblack | My answer would be 'hp hates its customers almost as much as microsoft does' |
| 00:54.09 | wwalker | Dell hates it's customers more that MS and HP combined. |
| 00:54.26 | jblack | wow. That's almost as much as comcast hates it's customers |
| 00:54.30 | wwalker | HP-UX was "special" as in olympics. |
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| 03:30.48 | Katty | hi |
| 03:31.13 | Katty | Merry Christmas |
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| 03:46.37 | jaytee | hi Katty |
| 03:46.45 | Katty | hugs jaytee |
| 03:46.46 | jaytee | Merry Christmas to you too |
| 03:46.50 | jaytee | hugs Katty |
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| 04:21.19 | Katty | <PROTECTED> |
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| 05:04.12 | jaytee | vewy, vewy qwiet |
| 05:42.06 | ChannelZ | coughs |
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| 06:29.15 | neurosys | burps |
| 06:29.55 | ChannelZ | reboots |
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| 07:59.19 | baldaris | hey guys i am a newbi...I have a did , I need to setup a IVRS , and call forwarding.. |
| 07:59.31 | baldaris | can asterisk help me.. |
| 08:01.20 | baldaris | ? |
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| 08:03.14 | baldaris | Dont mean to sound desperate...but i am willing to pay for services offered.. |
| 08:33.42 | baldaris | ? |
| 08:33.45 | baldaris | any one? |
| 08:33.56 | baldaris | some one who could setup asterix , on my server? |
| 08:34.17 | baldaris | if any expert is intrested , let me know.. |
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| 08:40.32 | TJNII | ~book |
| 08:40.43 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 08:40.51 | TJNII | baldaris: ^^^^ |
| 08:41.26 | baldaris | TJNII, yeah |
| 08:41.28 | baldaris | i am here.. |
| 08:41.41 | baldaris | Ooh i need to read these books.. |
| 08:42.26 | baldaris | ? |
| 08:43.46 | TJNII | It would be very wise to read that book. Or at least skim it, reading chapter 5. |
| 08:45.48 | baldaris | okay |
| 08:46.24 | baldaris | i am basically a web developer , i work on languages like python , php , ajax , xml. |
| 08:46.48 | baldaris | I was just going through the book..thanks for your help.. |
| 08:48.04 | TJNII | The config files and extensions language isn't hard if you can grasp the context of what it is doing. Plus, if you really need, you can program it in other languages through AGI. |
| 08:48.12 | TJNII | But read the book first. :) |
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| 09:02.46 | sawgood | If I wanted an inbound SIP trunk call to ring 4 specific phones when a call arrives, and then if no answer at any of the 4 phones, to ring 2 times, at another phone, then 2 times at another phone, and finally if no answer ... go to a generic VM (is find me / follow me the way this is done)? |
| 09:03.58 | TJNII | That sounds like a job for a dialplan with specific timeouts. |
| 09:06.05 | sawgood | TJNII: thank you |
| 09:07.02 | sawgood | I've found call routing under Asterisk to be different then commerical IP PBX boxes which lay it out so simply in a GUI like database window |
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| 09:36.07 | drmessano | sawgood: The "asterisk" project rightfully doesn't include a GUI... there's plenty of 3rd party options for that |
| 09:38.32 | sawgood | drmessano: I understand ... but working with my extensions.conf file (which is now 18 pages can be daunting) at times |
| 09:38.49 | drmessano | *nix development isn't like Windows development.. you don't start with a GUI and then hope all the knobs, buttons, and switches actually do something |
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| 09:48.14 | sawgood | Thats so weird, I just ran update-source, and when the process was down ... my Asterisk 'level' wen down to 1.4.21.2 instead of up to 1.4.28.x |
| 09:48.16 | sawgood | strange |
| 09:50.15 | sawgood | technically, what are the 'Asterisk Add-Ons" included when when updates Asterisk using update-source |
| 09:50.43 | sawgood | And, why isn't there a way to go from 1.4.x to 1.6.x wihtout have to do it manually" |
| 09:56.00 | drmessano | Are you using PBX In a Flash? |
| 09:58.39 | sawgood | yes |
| 09:59.02 | sawgood | I have several Asterisk boxes running in a LAB and most of them are PIAF |
| 09:59.32 | sawgood | I have one which I proudly build with CentOS 5.4 and added Asterisk 1.6 manually for practice |
| 10:00.09 | drmessano | So you installed a "PBX Distro" which specifically makes it easy to install Asterisk + all the components of FreePBX, are, for some reason, building your dialplan manually, and complaining about the lack of a GUI? |
| 10:00.58 | sawgood | Well, drmessano: its more of an approach of wanting to know more then one way to do things depending on the sub-variables of whos box it is I might be working on |
| 10:02.27 | drmessano | <sawgood> I've found call routing under Asterisk to be different then commerical IP PBX boxes which lay it out so simply in a GUI like database window <-- I dont understand this comment then.. Asterisk is a *Telephony Engine*.. Whether you use it to buid a PBX using flat config files, or if you put a GUI on it, it doesnt change what Asterisk itself is |
| 10:02.50 | drmessano | Even a 3com NBX system has an engine running which is fed by a config sitting in the file system |
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| 10:04.00 | sawgood | right ... but in FreePBX ... (under Extensions) ... there is nothing more then find me / follow me options if you want the call to 'continue on to another extensions |
| 10:04.18 | sawgood | You can use ring groups, but its just not the same in some sub-variables |
| 10:05.13 | drmessano | <sawgood> If I wanted an inbound SIP trunk call to ring 4 specific phones when a call arrives, and then if no answer at any of the 4 phones, to ring 2 times, at another phone, then 2 times at another phone, and finally if no answer ... go to a generic VM (is find me / follow me the way this is done)? <-- I could set this up in 2 mins using FreePBX |
| 10:05.21 | drmessano | But thats not really the point |
| 10:05.58 | sawgood | would you use find me / follow me? |
| 10:06.05 | drmessano | No |
| 10:06.23 | sawgood | can you tell me what you would do to accomplish this goal? |
| 10:07.22 | drmessano | Ring group that fails over to a queue that fails over to a custom extension (which is the "general mailbox") |
| 10:07.48 | sawgood | nice approach ...thanks ... |
| 10:08.12 | sawgood | I have one other situation ... I am facing ... see if you think a misc. destination is the answer |
| 10:09.09 | sawgood | A call comes into the Asterisk box via a SIP trunk call, the PBX answers the call, but immediately sends the call back out the SIP trunk to a PSTN number which is a waiting fax machine |
| 10:09.36 | sawgood | The fax machine is not on a ATA, it has its own PSTN copper line which is out of the scope of the IP PBX |
| 10:09.52 | drmessano | custom extension and use SIP/trunkname/number as the dial string.. done |
| 10:10.19 | sawgood | I did this one correctly then, thank you! |
| 10:10.23 | sawgood | whew! |
| 10:10.24 | drmessano | Of course its not going to work that well |
| 10:10.49 | sawgood | I know .. its a fax machine which is working about 85% of the tim |
| 10:10.51 | sawgood | time |
| 10:12.59 | sawgood | I will probably have to put an ATA on the fax machine to make it an extension of the Asterisk box, and then deliver the calls to the ATA (which I've done for faxing with the SIP trunk provider and it works 95-98%) of the time |
| 10:13.12 | drmessano | Questions like these are better asked in the #freePBX channel.. if you really have a need to try to accomplish something in there, some of us try to engineer solutions that keep everything in the GUI, which is really the point of using it in the first place. If you're going to rewrite half the dialplan in the _custom or _override files, your better off with a non-GUI config |
| 10:23.47 | sawgood | right on ... I'm running 1.4.28 with the latest add ons ... this update-source process is nice |
| 10:25.38 | drmessano | I think the Nxtvox site has been compromised |
| 10:26.12 | drmessano | I just got a nice spam message from their support box, and I see the 4 port and 8 port clone cards are linked to, but not showing up on their product pages |
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| 10:32.40 | m0e | Question.. I am getting really high "core show translation" values.. in the thousands.. though this machine is a fresh install running centos 5.3 + asterisk 1.6. The machine is a c2d 2.6ghz 4gb ram... before when I installed asterisknow on the same machine.. i was getting translation times like "9" etc. Are there some tweaks that I need? |
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| 11:22.32 | m0e | anyone? |
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| 12:47.20 | mazpe | is there any way to debug a conference is using the options from the databse? for MeetMe RealTime |
| 12:47.52 | mazpe | or where can i find the lastest docs for MeetMe realtime |
| 13:08.40 | mazpe | I'm getting "res_config_mysql.c:384 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info." |
| 13:08.50 | mazpe | what debug info should i be checking for? |
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| 13:25.53 | mazpe | i've enabled console: ...,debug and sip set debug on and i still do not see any relevant to the "Failed to query database" |
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| 13:31.27 | m0e | with regards to speex.. which files do I need to add to /usr/lib/asterisk/modules I dont want to do a "make install" |
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| 15:52.18 | ArtemMakhutov | Hello, is zaptel timing not required for IAX trunking (trunk=yes) with asterisk 1.6.2? |
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| 16:31.36 | mazpe | for music on hold is recommended to use .gsm vs mp3? |
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| 16:39.19 | [TK]D-Fender | mazpe: its recommended to use files in whatever codec your CALL is in |
| 16:40.49 | m0e | anyone on speex install? |
| 16:41.47 | m0e | I have followed a couple of tutorials but i end up with "-" in core show translation (asterisk 1.6.0) |
| 16:45.30 | mazpe | [TK]D-Fender: makes sense |
| 16:45.52 | mazpe | no conversion |
| 16:47.56 | mazpe | when i change moh files, do i need to restart asterisk? seems that with reload still looking for old files |
| 16:48.28 | fenrus | you probably can reload the module |
| 16:49.23 | m0e | if i build asterisk from source.. what does the XXX mean in make menuconfig? |
| 16:49.42 | Nivex | m0e: the feature is unavailable, usually because a build dependency is not present |
| 16:49.55 | m0e | aah.. thanks |
| 16:50.05 | *** join/#asterisk lmns972 (n=lmns972@cac94-8-82-245-26-103.fbx.proxad.net) |
| 16:51.10 | mazpe | module reload res_musiconhold.so |
| 16:51.16 | mazpe | still looking for old files. |
| 16:51.20 | mazpe | restart i guess |
| 16:51.37 | mazpe | just doesnt sound sane |
| 16:55.06 | [TK]D-Fender | m0e: means "look down its already telling you what you're missing" |
| 16:55.26 | *** join/#asterisk lost_soul (n=noymfb@cpe-74-71-234-100.twcny.res.rr.com) |
| 16:56.49 | m0e | [TK]D-Fender, sorry.. look down where? |
| 16:57.14 | [TK]D-Fender | m0e: DEPENDS ON |
| 16:57.44 | m0e | oh |
| 16:57.51 | m0e | my bad.. thank you :) |
| 16:58.18 | *** part/#asterisk inkubus08 (n=inkubus0@c-71-207-46-164.hsd1.pa.comcast.net) |
| 17:02.51 | *** part/#asterisk lmns972 (n=lmns972@cac94-8-82-245-26-103.fbx.proxad.net) |
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| 17:07.52 | *** part/#asterisk neurosys (n=neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 17:15.43 | *** join/#asterisk |R (n=bob@dsl-67-212-17-46.acanac.net) |
| 17:17.41 | |R | I've been running 1.6.x on debian up to the 1.6.2.0 final now, and since then, my voicemail is pretty fubar... if i send it to a mailbox normally, i hear the first word of vm-intro then nothing and if i disabled ("s") vm-intro, i get a beep, and then it creates an empty message. |
| 17:18.03 | |R | Has anyone experienced something similar? I have a pretty plain setup for home use... |
| 17:19.28 | *** join/#asterisk d00gster (n=doughant@94.96.110.232) |
| 17:22.13 | ChannelZ | pastebin your voicemail.conf |
| 17:25.42 | *** join/#asterisk coppice (n=chatzill@25.176.64.202.dyn.pacific.net.hk) |
| 17:27.49 | ArtemMakhutov | must zaptel.conf contain anything for ztdummy? |
| 17:28.06 | *** join/#asterisk Tim_Toady (n=moi@77.49.123.166.dsl.dyn.forthnet.gr) |
| 17:29.24 | ChannelZ | ztdummy is only needed if you need MeetMe and otherwise have no telephony hardware |
| 17:30.26 | ChannelZ | but there is nothing to configure for it there |
| 17:31.00 | ArtemMakhutov | I want to use IAX with trunk=yes on asterisk 1.4 |
| 17:31.12 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 17:32.08 | ArtemMakhutov | I never did anything with zaptel so I have no clue. |
| 17:32.28 | ArtemMakhutov | Is this ok for ztdummy? |
| 17:32.33 | ArtemMakhutov | zap show status |
| 17:32.35 | ArtemMakhutov | Description Alarms IRQ bpviol CRC4 |
| 17:32.37 | ArtemMakhutov | ZTDUMMY/1 (source: HRtimer) 1 UNCONFIGUR 0 0 0 |
| 17:33.40 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 17:34.21 | ChannelZ | looks about right from what I remember |
| 17:34.48 | ArtemMakhutov | ok, thx |
| 17:42.46 | *** join/#asterisk YaCk (n=Brayan@unaffiliated/yack) |
| 17:46.19 | ArtemMakhutov | Has anyone send faxes with a mobile phone (GSM)? |
| 17:46.37 | [TK]D-Fender | ArtemMakhutov: that is NOT going to happen |
| 17:47.01 | ArtemMakhutov | why not? |
| 17:47.19 | [TK]D-Fender | ArtemMakhutov: because the codec is too lossy for the connection. G.711 or bust |
| 17:47.49 | ArtemMakhutov | yeah, but some mobile phones are supporting fax class 1 over GSM |
| 17:48.32 | ArtemMakhutov | Faxes over GSM are working. There is a special gsm protocol etc for that |
| 17:48.47 | [TK]D-Fender | ArtemMakhutov: then its not "fax" |
| 17:48.50 | coppice | ArtemMakhutov: GSM has always supported FAX, but not all installations do |
| 17:50.16 | ArtemMakhutov | sorry, this is only in german: http://www.dafu.de/rechts/gsm-fax.html |
| 17:50.50 | jblack | anyone here that I like that has steam? |
| 17:51.24 | Akiraa | Would you recommend a particular book to learn asterisk? |
| 17:51.28 | [TK]D-Fender | jblack: what if I'm only tepid? |
| 17:51.44 | jblack | i like you. =) |
| 17:51.52 | jblack | can you msg me your steamid? I have something for you |
| 17:52.26 | [TK]D-Fender | jblack: Sorry, don't actually have it... was just asking :) |
| 17:52.32 | [TK]D-Fender | ~book |
| 17:52.39 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:52.43 | [TK]D-Fender | Akiraa: ^^^ |
| 17:52.44 | *** join/#asterisk icyValk77 (n=icyValk7@host86-161-121-104.range86-161.btcentralplus.com) |
| 17:53.04 | Akiraa | ~buybook |
| 17:53.05 | infobot | [~buybook] You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 17:53.36 | Akiraa | <[TK]D-Fender> thanks |
| 17:54.47 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 17:55.46 | ChannelZ | Steam as in the game thing? |
| 17:58.46 | Akiraa | <[TK]D-Fender> What about a book on IP telephony in general, if you may know |
| 17:59.29 | [TK]D-Fender | Akiraa: http://www.amazon.ca/SIP-Demystified-Gonzalo-Camarillo/dp/0071373403 |
| 18:00.13 | [TK]D-Fender | Akiraa: jsut one I'd heard of anyway |
| 18:00.35 | [TK]D-Fender | Akiraa: TFOT has a decent little primer. Depends what you want to know/do |
| 18:01.44 | *** join/#asterisk lost_sou1 (n=noymfb@cpe-74-71-234-100.twcny.res.rr.com) |
| 18:02.40 | Akiraa | <[TK]D-Fender> TFOT? -- looking for something that begins with little assumptions |
| 18:04.10 | *** part/#asterisk icyValk77 (n=icyValk7@host86-161-121-104.range86-161.btcentralplus.com) |
| 18:04.19 | [TK]D-Fender | Akiraa: Only assumption TFOT makes is that you are likely to use * to access these techs |
| 18:05.14 | ChannelZ | (TFOT == Asterisk: The Future of Telephony book) |
| 18:07.04 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 18:08.16 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
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| 18:19.41 | *** join/#asterisk dkirker-openmobl (n=dkirker@adsl-99-161-109-81.dsl.pltn13.sbcglobal.net) |
| 18:22.37 | *** join/#asterisk xmitter (n=xmitter@c-24-21-212-187.hsd1.or.comcast.net) |
| 18:24.49 | lesouvage | Akiraa: I read the SIP book and it is a great book if you want to learn and know about SIP. |
| 18:26.14 | lesouvage | Switching to Voip from O'Reilly is also a god book. |
| 18:26.21 | lesouvage | god = good |
| 18:26.42 | hardwire | it's both |
| 18:27.13 | *** join/#asterisk thenthenio (n=thenthen@93-36-216-12.ip62.fastwebnet.it) |
| 18:27.43 | ArtemMakhutov | Is there a way to make asterisk send the RTP packets twice to avoid packet loss on a bad link? |
| 18:29.43 | hardwire | ArtemMakhutov: chances are low that it would replay in time |
| 18:30.17 | hardwire | ArtemMakhutov: can I suggest using IAX2 trunking and reduce the trunking time to 10ms if your link is happy with that. |
| 18:30.18 | ArtemMakhutov | not replaying the packets, jut send all packets two times insted of only one time |
| 18:30.49 | hardwire | ArtemMakhutov: it is possible to do that without asterisk, actually |
| 18:31.07 | hardwire | set up a two or more TAP tunnel between asterisk servers. |
| 18:31.13 | hardwire | then enable bonding in broadcast mode. |
| 18:31.36 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 18:31.45 | coppice | ArtemMakhutov: most packets are lost due to congestion on the tributaries. Doubling the number of packets there is probably a bad thing |
| 18:31.51 | hardwire | ArtemMakhutov: but I doubt asterisk has the facilities to discard duplicate packets |
| 18:32.38 | [TK]D-Fender | turns his packet transmitters to 11... thats 1 more... |
| 18:33.04 | ArtemMakhutov | ok, thx, I will try reducing the trunking time first |
| 18:33.19 | hardwire | anyways.. it's possible.. but probably not a good solution :) |
| 18:39.29 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 18:39.46 | ChannelZ | It's a bit like saying we needed SAW II, SAW III, etc. in case someone missed the first one. |
| 18:40.47 | ChannelZ | more is *definately* not always better. |
| 18:40.58 | ChannelZ | (and sometimes even one is too much...) |
| 18:42.05 | *** join/#asterisk thenthenio (n=thenthen@93-36-216-12.ip62.fastwebnet.it) |
| 18:53.11 | vally | can anyone help me on this? http://pastebin.com/m119e2db2 |
| 19:02.45 | *** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr) |
| 19:02.59 | Katty | hi |
| 19:03.19 | jaytee | herro |
| 19:05.02 | *** join/#asterisk thenthenio (n=thenthen@93-36-216-12.ip62.fastwebnet.it) |
| 19:08.24 | *** join/#asterisk coppice (n=chatzill@202.64.176.25) |
| 19:10.42 | [TK]D-Fender | vally: You're using ${EXTEN} with chanspy.. There's a prblem right there. a numbered extension is not how a CHANNEL NAME starts |
| 19:10.53 | [TK]D-Fender | vally: its TECH/PEER/NUMBER, etc |
| 19:11.47 | [TK]D-Fender | vally: Unless you've encoded it elsewhere, but you didn't provide any debug of an actual attempt |
| 19:13.01 | *** join/#asterisk TheDavidFactor-H (n=thedavid@c-68-34-116-180.hsd1.md.comcast.net) |
| 19:15.48 | vally | [TK]D-Fender: well, i took the channel name in the macro and encode it. is there any other way to spy on the dialed channel using my setup? how would i give the reference to chanspy? (more than one call like this can happen) |
| 19:18.26 | *** join/#asterisk SteelDig (i=SteelDig@124.83.27.2) |
| 19:19.06 | *** part/#asterisk SteelDig (i=SteelDig@124.83.27.2) |
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| 19:20.55 | [TK]D-Fender | vally: I'm not seeing any debug.... |
| 19:24.06 | [TK]D-Fender | vally: based on your code I also am have trouble seeing the point of what you're doing |
| 19:26.44 | vally | [TK]D-Fender: i thought i had described it. so basically: originating a call to a phone. streaming the phone voice to an external interface (context mychan) while playing sound TO the phone (context audiosource, no moh!). only one phone in this setup. will provide debug output now. |
| 19:29.19 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:30.47 | vally | [TK]D-Fender: http://pastebin.com/m684d5e23 phone number changed to 999999999999 |
| 19:33.25 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
| 19:51.51 | *** join/#asterisk Akiraa (n=Akiraa@79.112.36.41) |
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| 19:57.52 | *** join/#asterisk jo3sm1th (n=email@c83-250-80-70.bredband.comhem.se) |
| 19:58.50 | jo3sm1th | Is it possible to put something in the extensions.conf file where I can call my DID, then have the DID Dial(a toll free number), enter a PIN# (my phone-card PIN), just a fun experiment so I can bypass entering in my own PIN |
| 19:59.28 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
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| 20:11.41 | ChannelZ | jo3sm1th: yes |
| 20:11.56 | ChannelZ | SendDTMF |
| 20:13.02 | *** join/#asterisk Tech_Travis (n=Administ@cpe-76-87-9-130.socal.res.rr.com) |
| 20:18.52 | telnettech | ok someone help me with a brain-fart |
| 20:19.04 | telnettech | what is the command to look at what processes are running |
| 20:19.56 | jaytee | ps |
| 20:20.27 | jaytee | or ps aux | grep "something" |
| 20:20.30 | telnettech | thanks jaytee |
| 20:26.55 | *** join/#asterisk Akiraa (n=Akiraa@79.112.36.41) |
| 20:32.47 | *** join/#asterisk _heniu (n=Miranda@50-goc-24.acn.waw.pl) |
| 20:34.01 | _heniu | hi all ! |
| 20:34.09 | fenrus | hi there! |
| 20:34.59 | ChannelZ | flashes his moobs |
| 20:37.26 | _heniu | I'm looking for info how woulb be to connect asterisk to pbx (old analog Panasonic) |
| 20:37.45 | _heniu | we are trying to build small (5 users system) for it |
| 20:38.17 | _heniu | it = IT department |
| 20:43.02 | fenrus | what interfaces does the panasonic have? |
| 20:43.09 | fenrus | SIP, E1, T1 ? |
| 20:49.09 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 20:49.42 | *** join/#asterisk TheDavidFactor-H (n=thedavid@c-68-34-116-180.hsd1.md.comcast.net) |
| 20:54.44 | _heniu | its some old one (ive seen it only once) D500 ? it one for building , and contact with person who takes care about it is very hard |
| 20:58.00 | _heniu | ive got only socket in wall ;-) (probably no possibility to reconfigure anything on pbx) |
| 20:58.20 | ChannelZ | well what is it you want to actually do by connection * to an existing PBX? |
| 20:59.59 | wwalker | any recommendations for setup in a high packet loss environment? |
| 21:00.15 | drmessano | wwalker: Get a PRI |
| 21:00.36 | wwalker | my son is travelling in foreign countries. |
| 21:01.15 | wwalker | I've set him up as a sip peer over openvpn. it works, but it's often _very_ choppy |
| 21:01.16 | drmessano | Well, if the packet loss is on the client end, nope.. |
| 21:01.28 | drmessano | gulp |
| 21:01.35 | drmessano | Over OpenVPN? |
| 21:01.35 | fenrus | wwalker, did you try without the openvpn ? |
| 21:01.46 | drmessano | Try without the VPN |
| 21:02.24 | _heniu | there is about 100 users connected to it and we would like to have some kind of IVR and voice mail |
| 21:03.07 | _heniu | [about openvpn]: try to change to UDP and disable LZO compression |
| 21:03.11 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 21:04.09 | drmessano | _heniu: Sounds like you want to put a Small PBX in place of the IT dept "phone" and have VM and a little inner-dept communication. I dont think that endpoint PBX connection will help you here |
| 21:04.32 | _heniu | my question is would it be better to use some kind of PCI card or external fxo/fxs gateway |
| 21:05.00 | drmessano | To connect to that jack? |
| 21:05.21 | _heniu | drmessano: yes exacly |
| 21:05.57 | _heniu | drmessano : ive read a lot about problems with noise cancellation , detecting hook off etc |
| 21:06.01 | drmessano | Doesnt matter if you have an internal or an external.. we've not even established you CAN use what's on that jack, which is highly unlikely |
| 21:07.51 | *** join/#asterisk dkirker-openmobl (n=dkirker@adsl-99-161-109-81.dsl.pltn13.sbcglobal.net) |
| 21:08.54 | fenrus | drmessano, get an external provider instead of some homebrew pbx in the house with someone not being helpful managing it :) |
| 21:10.04 | drmessano | Its not a homebrew PBX.. he said he believes its a Panasonic system |
| 21:10.44 | drmessano | It sounds like he wants to branch out from the IT depts "extension" and put a small IT dept PBX on the end of it |
| 21:10.46 | fenrus | i meant homebrew as in non telecom-company-provided |
| 21:11.25 | fenrus | well, of course the first step is to actually speak to the guy and make him understand your needs, perhaps is theese functions already in the pbx |
| 21:11.39 | fenrus | if he refuses i'd go external instead of being controlled by someone |
| 21:11.41 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:12.05 | drmessano | I dont think they want the functions.. I think they want their own little PBX and just need an interconnect to the main system |
| 21:12.23 | drmessano | Which means either an analog or PRI interface |
| 21:13.32 | *** join/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 21:15.26 | _heniu | its 100 worker company pople are calling to helpdesk (throu internal number) |
| 21:16.17 | _heniu | there are 5 internal numbers for IT |
| 21:17.10 | _heniu | and people are always calling to wrong person so we would like to add some kind of ivr (if ypu need to contact with DB admin press 1 etc) |
| 21:17.12 | drmessano | You need an extra PRI card in the office PBX and a PRI card for the asterisk box if you want those 5 extensions brough to Asterisk |
| 21:20.11 | _heniu | ok thanks for help |
| 21:25.54 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 21:31.59 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 21:42.03 | *** join/#asterisk rdahlin_1 (n=rdahlin_@78-73-19-238-no168.tbcn.telia.com) |
| 21:47.42 | jo3sm1th | Can anyone show me just one line of sampel code that would send a string of numbers perhaps 1234 with send DTMF (after the call has been connected) |
| 21:52.01 | ChannelZ | SendDTMF(1234) |
| 21:52.05 | ChannelZ | Crazy, huh? |
| 21:52.17 | fenrus | thats innovative |
| 21:59.13 | jo3sm1th | Save the commentary for the DVD release |
| 21:59.28 | jo3sm1th | This is a channel for the serious discussion of Asterisk technologies |
| 21:59.47 | ChannelZ | We talk about squirrels too |
| 22:00.49 | drmessano | and nuts |
| 22:01.40 | ChannelZ | and squirrel nuts |
| 22:03.48 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 22:05.26 | *** part/#asterisk thenthenio (n=thenthen@93-36-216-12.ip62.fastwebnet.it) |
| 22:08.27 | jo3sm1th | does SendDTMF have to be in the dialplan like ,2,Dial then on the next line ,3,SendDTMF? |
| 22:08.45 | drmessano | Yep |
| 22:09.59 | Gugge | jo3sm1th, no |
| 22:10.07 | jo3sm1th | it goes BEFORE you Dial ? |
| 22:10.11 | ChannelZ | You probably want a Wait() in between if you're waiting for your phone company to answer and ask to enter your pin |
| 22:10.17 | Gugge | read the dial() docs |
| 22:10.36 | ChannelZ | Gugge: he wants to send additional digits besides just dialing a number |
| 22:11.08 | Gugge | yes, and stuff after the dial() command wont be run unless the dial doesnt go through |
| 22:12.15 | TJNII | I think Gugge is right. IIRC there is some extra hoodoo you need to do to get it to run during an outgoing dial. |
| 22:12.28 | ChannelZ | yeah you need to arg up Dial |
| 22:13.26 | Gugge | "read the dial() docs" :P |
| 22:13.47 | jo3sm1th | dial docs or send DTMF docs |
| 22:17.41 | jo3sm1th | http://www.the-asterisk-book.com/unstable/applikationen-senddtmf.html |
| 22:17.55 | jo3sm1th | It just says a basic description but that doesnt actually send digits while in a call does it? |
| 22:19.53 | Gugge | you need to read the dial docs to find out how to do stuff after dialing somewhere |
| 22:19.56 | ChannelZ | look at the G option of Dial |
| 22:20.44 | ChannelZ | Or you might be able to do it more ghetto, and use the 'w' character in your dialstring.. like Dial(Whatever/1235551212wwww6666) |
| 22:21.07 | ChannelZ | At least I think you can throw multiple w's in there |
| 22:23.17 | jo3sm1th | this doesnt work Dial(Whatever/1235551212wwww6666 |
| 22:23.29 | jo3sm1th | but i would have been certainly willing to do it that way |
| 22:24.06 | *** join/#asterisk TheDavidFactor-H (n=thedavid@c-68-34-116-180.hsd1.md.comcast.net) |
| 22:27.55 | ChannelZ | it actually does work, but it's not going to connect the other side of the channel until it's done dialing |
| 22:28.00 | ChannelZ | so you won't hear it |
| 22:29.00 | ChannelZ | (or the 'local side' I guess I should say) |
| 22:29.18 | ChannelZ | but it probably depends more on the type of channel you're dialing on, you didn't say |
| 22:30.19 | jo3sm1th | so after Dial on the dialplan there is no way to make it generate DTMF tones |
| 22:32.47 | Gugge | sure there is |
| 22:34.05 | Gugge | with the G option, as ChannelZ said |
| 22:35.03 | Gugge | <PROTECTED> |
| 22:36.44 | Gugge | Dial(type/nr,timeout,D(ww123)) |
| 22:36.49 | jo3sm1th | So you put G option in Dialplan such as ,3,G(context^exten^priority) |
| 22:37.13 | Gugge | G is if you wanna do more complex stuff, needing dialplan magic |
| 22:37.26 | TJNII | jo3sm1th: Please go read the dial documentation. Voip-info.org is a good place to start. |
| 22:45.21 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 22:49.25 | *** join/#asterisk pepesmith (n=pepesmit@unaffiliated/pepesmith) |
| 22:49.38 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 22:50.37 | pepesmith | hello,i have an error here.. utils.c:1126 ast_carefulwrite: write() returned error: Connection refused |
| 22:50.49 | pepesmith | but i have already my fastagi running |
| 22:51.19 | pepesmith | how will i know if that fastagi script is working? |
| 22:51.59 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 23:07.28 | *** join/#asterisk Katty (n=User@adsl-71-157-174-124.dsl.stlsmo.sbcglobal.net) |
| 23:07.53 | Katty | well my squirrely tray isn't going anywhere now |
| 23:08.55 | Katty | not unless the squirrels can rip out wood screws. |
| 23:09.10 | ChannelZ | Depends on what they got for Christmas |
| 23:19.00 | Katty | think there will be any vistors to the tray tonight? |
| 23:19.39 | m0e | any guides on how to make the "core show translation" show lower times? |
| 23:20.09 | m0e | i get really low times with asterisknow.. but when I install centos then the 1.6 rpm from the asterisk repo.. i get extremely high times |
| 23:34.12 | *** part/#asterisk teknomega (n=Chris@c-76-117-79-94.hsd1.pa.comcast.net) |
| 23:36.14 | drmessano | 1.6 uses microseconds instead of milliseconds |
| 23:40.55 | dlynes | Katty: you decided to stop getting frustrated with them, and just fix their wagon? :) |
| 23:42.22 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 23:54.54 | ChannelZ | m0e: It shows in microseconds now, not milliseconds |
| 23:55.14 | m0e | ChannelZ, oh.. thanks.. didnt notice that! |
| 23:55.25 | m0e | i was wondering why the number was so high :) |
| 23:55.26 | ChannelZ | yeah it threw me off the first time I looked too |
| 23:55.41 | m0e | thanks a bunch :) |
| 23:55.53 | ChannelZ | oh drmessano told you earlier, I didn't see it.. but I guess neither did you |
| 23:56.11 | m0e | oops.. i was out |
| 23:58.07 | m0e | thanks drmessano :) |
| 23:58.19 | drmessano | Dont mention it |
| 23:58.22 | drmessano | Ever. |
| 23:58.35 | m0e | :) |