| 00:02.13 | ChannelZ | yes |
| 00:03.44 | ChannelZ | just change the value before you define each channel |
| 00:11.36 | nix8n82 | Thank you ChannelZ |
| 00:15.45 | *** join/#asterisk guam (~craig@bach154-110.guamcell.net) |
| 00:21.39 | *** part/#asterisk guam (~craig@bach154-110.guamcell.net) |
| 00:21.54 | *** join/#asterisk guam (~craig@bach154-110.guamcell.net) |
| 00:28.15 | *** join/#asterisk joobie (~joobie@CPE-124-176-179-1.lns3.win.bigpond.net.au) |
| 00:28.39 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 00:33.23 | *** part/#asterisk guam (~craig@bach154-110.guamcell.net) |
| 00:37.44 | *** join/#asterisk guam (~craig@bach154-110.guamcell.net) |
| 00:40.34 | *** part/#asterisk guam (~craig@bach154-110.guamcell.net) |
| 00:50.39 | *** join/#asterisk fnordus (~dnall@S01060023693bfad4.va.shawcable.net) |
| 00:56.18 | *** join/#asterisk n0cturnal (~n0c@2403:dc00:ffff:fffc:226:18ff:fe3a:3bad) |
| 01:11.59 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
| 01:12.00 | *** part/#asterisk irieKEN (~km_spam_a@rrcs-76-79-107-76.west.biz.rr.com) |
| 01:23.26 | *** join/#asterisk harovali (~harovali@r190-134-144-28.dialup.adsl.anteldata.net.uy) |
| 02:24.42 | *** join/#asterisk duwan66 (~duwan66@70.88.111.109) |
| 02:33.37 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 03:02.58 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 03:11.45 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 03:18.11 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 04:17.16 | *** join/#asterisk nsgn (~nsgn@rrcs-24-153-223-124.sw.biz.rr.com) |
| 04:17.29 | nsgn | is there any inherent ability in asterisk's CDR to email a daily report? |
| 04:17.39 | Kobaz | nope |
| 04:17.42 | nsgn | rats :) |
| 04:17.52 | nsgn | had a customer whos former phone company apparently did this and they were spoiled by it |
| 04:18.59 | Kobaz | i would be a pretty easy little shell script to write |
| 04:19.12 | Kobaz | er.. it would |
| 04:20.44 | nsgn | yeah |
| 04:20.46 | nsgn | and yeah |
| 04:21.11 | *** join/#asterisk moltar_net (~Roman@180.183.204.216) |
| 04:29.40 | *** join/#asterisk aruntomar (~arun@61.17.193.163) |
| 04:34.16 | aruntomar | hi. my asterisk server is behind NAT. earlier i had a router with port forwarding enabled and things were fine. recently we deployed a linux machine as a router and iptables as a firewall, the port forwarding is working fine,and we can connect to asterisk but neither the caller or receiver is able to hear each other. plz help. |
| 04:37.08 | *** join/#asterisk roshan08 (~roshan@122.172.43.234) |
| 04:38.14 | roshan08 | Hi. I had an idea. Can I reroute an incoming call to an outside phone using voip on asterisk? Will this rerouting incur additional expenses at my end? |
| 04:46.40 | aruntomar | roshan08, yes, you can |
| 04:47.26 | roshan08 | aruntomar, cool. I am thinking that the incoming call will be rerouted via voip. But what cost will it incur? |
| 04:47.55 | roshan08 | Is there a way to make the incoming caller pay this cost? |
| 04:48.53 | aruntomar | will your outside phone somehow connected to asterisk? |
| 04:49.04 | aruntomar | are you routing the call locally or internationally |
| 04:49.40 | roshan08 | aruntomar, no, the outside phone will not be connected to anything. I want to connect to some random phone number using voip. |
| 04:50.20 | roshan08 | aruntomar, it might be international as well. |
| 04:51.04 | aruntomar | roshan08, ok. so there are 2 things, if you are rerouting locally in india, then legally you can't terminate a ip call to pstn line |
| 04:51.28 | aruntomar | roshan08, so you'll have to use a pri line, using which you can reroute the call |
| 04:51.57 | roshan08 | aruntomar, what is pri? ok me googles. |
| 04:52.57 | *** join/#asterisk jstapleton (~jstapleto@173-15-197-75-BusName-Richmond.hfc.comcastbusiness.net) |
| 04:53.08 | aruntomar | roshan08, but if you want to reroute to internal locations,then you'll have to buy voip package from any service provider and reroute the traffic via them, in both case you'll be charged, how much, it depends on your service provider |
| 04:53.57 | roshan08 | aruntomar, ok. am looking at pri lines. |
| 05:03.02 | roshan08 | aruntomar, So if i have an incoming PRI line to my PBX, I can then reroute it to a pstn line in india using voip? |
| 05:03.26 | roshan08 | Or do I have to use the PRI line outgoing as well? |
| 05:06.37 | aruntomar | roshan08, you can receive a call either on voip or pri line, but if you want to call a normal phone (pstn network ) in india then you'll have to route the call via pri line |
| 05:07.21 | aruntomar | roshan08, but let me warn you, this may not go down well with telecom authorities or providers |
| 05:07.48 | roshan08 | aruntomar, ah. Why so? Is it illegal again? |
| 05:09.01 | aruntomar | roshan08, no, it's not illegal, but this is what the telecom providers also do, so basically you are using there infra and competing against there business interest |
| 05:13.38 | roshan08 | aruntomar, ok. |
| 05:16.03 | *** join/#asterisk coppice (~chatzilla@210.17.255.115) |
| 05:34.47 | *** join/#asterisk CheBuzz_Home (~CheBuzz_H@ip68-104-9-125.lv.lv.cox.net) |
| 05:35.56 | CheBuzz_Home | Any hints on what this means? |
| 05:35.57 | CheBuzz_Home | [Jan 7 13:52:43] WARNING[10879]: chan_sip.c:8160 process_sdp: Insufficient information for SDP (m= not found) |
| 05:44.22 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 05:44.23 | *** part/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 05:49.50 | *** join/#asterisk Maximo (~maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 05:52.34 | *** part/#asterisk Maximo (~maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 06:15.21 | *** part/#asterisk harovali (~harovali@r190-134-144-28.dialup.adsl.anteldata.net.uy) |
| 06:35.28 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 06:38.52 | *** join/#asterisk [8none1] (~8none1]@ps14528.dreamhost.com) |
| 06:52.40 | *** join/#asterisk [8none1] (~8none1]@ps14528.dreamhost.com) |
| 07:12.47 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
| 07:25.12 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 07:25.46 | IsUp | hello |
| 07:29.29 | *** join/#asterisk DJClean (~djclean@unaffiliated/djclean) |
| 07:36.22 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 07:42.02 | *** join/#asterisk Tim_Toady (~moi@62.1.174.14.dsl.dyn.forthnet.gr) |
| 07:42.23 | *** join/#asterisk salimb (~chatzilla@81.5.139.17) |
| 07:42.45 | ChannelZ | aloha |
| 07:52.35 | CheBuzz_Home | Interesting. I was using Asterisk 1.8 and simply could not get my ATA to work. But it works just fine with asterisk 1.6. It seems some "features" were introduced in the latest version. |
| 07:56.09 | *** join/#asterisk lfod- (~email@unaffiliated/lfod-) |
| 07:56.49 | *** join/#asterisk lfod- (~email@unaffiliated/lfod-) |
| 07:57.12 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 08:13.35 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
| 08:16.32 | *** join/#asterisk Polysics (~Polysics@host1-98-static.106-82-b.business.telecomitalia.it) |
| 08:16.35 | Polysics | hello |
| 08:17.03 | Polysics | interesting use case: company has roughly 70 shops and wants to move to a VoIP infrastructure to save on costs |
| 08:17.20 | Polysics | would it simply be a case of a good Asterisk installation and a softphone in each shop? |
| 08:17.39 | Polysics | plus a contract with a compatible provider for calling out? |
| 08:17.43 | Kobaz | depends |
| 08:18.13 | Polysics | each shop needs its own PSTN number, but i can get a hundred of them for a few euros |
| 08:18.43 | Polysics | i am more worried about redundancy, but that should be achievable |
| 08:19.57 | Polysics | on that i could build any kind of service, voicemail, web interfaces |
| 08:20.47 | Polysics | all shops have a good ADSL line with 256kb guaranteed bw because of their ERP |
| 08:26.16 | Polysics | if the above would work, what would be the easiest way to have redundancy? |
| 08:26.38 | Polysics | all i need is a ready-to-go backup server |
| 08:31.30 | Kobaz | depends which part(s) you want to be redundent |
| 08:33.19 | Polysics | i would say all of it, i could even just do backups |
| 08:33.40 | Polysics | but asterisk never "breaks", so redundancy would involve a different location |
| 08:34.00 | Polysics | because 99% of failures involve the network or hardware |
| 08:35.22 | Kobaz | asterisk has it's problems |
| 08:35.39 | Kobaz | i crash asterisk pretty often, although that's probably because i'm doing weird stuff with it |
| 08:35.49 | Kobaz | although i haven't tried 1.8 yet |
| 08:37.23 | Kobaz | but you can't guarantee it will never break |
| 08:38.02 | Polysics | well, asterisk breaking is the easiest part as i can have an hot-swap virtual image ready in seconds |
| 08:38.11 | Kobaz | main redundency would be internet connection, if you're doing all voip |
| 08:38.25 | Polysics | well, shops CAN go down, they have cellphones |
| 08:38.38 | Polysics | they also have IP based credit card processors and their ERP |
| 08:38.44 | Kobaz | and depending on the voip provider, you can set up automatic reroutes if a site is down |
| 08:38.46 | Polysics | so i am not worrying about lines |
| 08:39.01 | ChannelZ | Someone fetch me some tacos! |
| 08:39.07 | Polysics | i guess i could just use a domain name instead of an IP |
| 08:39.13 | Kobaz | anyway... nap time |
| 08:39.14 | Polysics | that alone would ease things up |
| 08:39.25 | Polysics | then replicate the mysql contents and a few config files |
| 08:39.49 | Polysics | anything happens, a script switches the domain to the other server |
| 08:40.08 | Polysics | still, i remember reading somewhere about a very easy cluster setup using 2 * boxes |
| 08:46.52 | *** part/#asterisk jplank1 (~G_Bove@208-104-67-26.dyn.fttp.comporium.net) |
| 09:04.31 | *** join/#asterisk [o_o]owl (~o_o]owl@unaffiliated/goldroger) |
| 09:05.10 | [o_o]owl | hi, any way to capture Q931/Q921 pcap dumps for wireshark use while using dahdi with digium h/w ? |
| 09:12.08 | *** join/#asterisk gamedna (~gamedna@cpe-173-173-109-171.satx.res.rr.com) |
| 09:12.15 | gamedna | evening |
| 09:12.23 | [o_o]owl | hi gamedna |
| 09:12.34 | [o_o]owl | are you into game development or something ? |
| 09:12.38 | [o_o]owl | lol |
| 09:12.53 | *** join/#asterisk gamedna (~gamedna@cpe-173-173-109-171.satx.res.rr.com) |
| 09:18.42 | WIMPy | [o_o]owl: If you're using misdn you can use misdn_log. It you're using dahdi, try to search the issue tracker. I think I've seen a patch there. |
| 09:24.42 | WIMPy | Is wireshark useful for isdn? |
| 09:25.50 | [o_o]owl | WIMPy: yes, pretty much |
| 09:26.58 | [o_o]owl | well, not _that_ much, but we can see outgoing messages like ALERTING/DISCONNECT/RELEASE/RELEASE COMPLETE and their causes, etc and their directions - it helps debugging by like 1-5% more :P |
| 09:27.16 | [o_o]owl | thanks, I'll have a look at the issue tracker |
| 09:27.40 | WIMPy | Doesn't sound that much. Maybe I should try it. |
| 09:31.33 | WIMPy | Looks like you get the info but in an rather inconvenient way. |
| 09:36.22 | WIMPy | Maybe I should look for some documentation on wireshark files. |
| 09:53.41 | *** join/#asterisk friendly12345 (~friendly@ppp118-209-132-36.lns20.mel6.internode.on.net) |
| 10:01.41 | *** join/#asterisk Da-Geek (~Da-Geek@11.74.155.90.in-addr.arpa) |
| 10:06.58 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 10:19.25 | WIMPy | [o_o]owl: Have you looked in to it? I get the impression that it's all about the commercial EyeSDN product and the option for wireshark to read the capturs of that software. Or is there more to that? |
| 10:26.04 | [o_o]owl | WIMPy: with FreeSWITCH, there is an isdn signalling module that allows packets to be dumped in pcap format |
| 10:26.29 | [o_o]owl | however the signalling library itself is not mature and not encouraged to be used |
| 10:27.11 | [o_o]owl | libpri is the one that is recommended for use, but unfortunately, libpri doesn't export packets in the format that is required for pcap ( I'm assuming this based on a quick view of the code ) |
| 10:27.57 | WIMPy | So it looks there are two possible formats. |
| 10:28.45 | WIMPy | I'm interested in the other direction. I'd like to read the files, not write them :-) |
| 10:29.03 | [o_o]owl | heheee |
| 10:31.02 | *** join/#asterisk moltar_net (~Roman@180.183.201.91) |
| 10:36.11 | *** join/#asterisk moltar_net (~Roman@180.183.201.91) |
| 10:40.08 | *** join/#asterisk iscario (~quassel@LMontsouris-156-24-36-77.w217-128.abo.wanadoo.fr) |
| 11:12.05 | *** part/#asterisk friendly12345 (~friendly@ppp118-209-132-36.lns20.mel6.internode.on.net) |
| 11:30.55 | *** join/#asterisk frocostrocos (~chatzilla@80.172.172.49) |
| 11:31.02 | frocostrocos | Hi guys! |
| 11:31.13 | frocostrocos | really have a problem with asterisk-java |
| 11:31.41 | frocostrocos | Anyone, ever got this: INFO: No event class registered for event type 'bridge', attributes: |
| 11:31.53 | frocostrocos | I'm using 1.0.0.M3 |
| 11:31.59 | frocostrocos | still got these issues |
| 11:32.17 | *** join/#asterisk ruben23 (~Innermax@119.92.78.124) |
| 11:40.10 | *** join/#asterisk Da-Geek (~Da-Geek@11.74.155.90.in-addr.arpa) |
| 11:41.42 | *** join/#asterisk Da-Geek (~Da-Geek@11.74.155.90.in-addr.arpa) |
| 11:47.36 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 11:47.36 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.1.1 (2010/12/15), 1.6.2.15 (2010/12/08), 1.4.38 (2010/12/08), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 12:03.46 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 12:06.33 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
| 12:27.15 | *** join/#asterisk Da-Geek (~Da-Geek@11.74.155.90.in-addr.arpa) |
| 12:45.44 | *** join/#asterisk bio-tty (~c@2.3.34.95.customer.cdi.no) |
| 12:58.00 | *** join/#asterisk Da-Geek (~Da-Geek@11.74.155.90.in-addr.arpa) |
| 12:58.33 | *** join/#asterisk Da-Geek (~Da-Geek@11.74.155.90.in-addr.arpa) |
| 13:06.33 | *** join/#asterisk puzzled (~patrick@535335AA.cm-6-4a.dynamic.ziggo.nl) |
| 13:16.27 | *** join/#asterisk Da-Geek_ (~Da-Geek@11.74.155.90.in-addr.arpa) |
| 13:17.15 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:17.49 | *** part/#asterisk Da-Geek_ (~Da-Geek@11.74.155.90.in-addr.arpa) |
| 13:19.43 | *** join/#asterisk Da-Geek (~Da-Geek@11.74.155.90.in-addr.arpa) |
| 13:24.07 | *** join/#asterisk sourcode (~code@ppp-58-8-182-3.revip2.asianet.co.th) |
| 13:29.24 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 13:38.05 | *** join/#asterisk mpe (~mpe@212.45.120.202) |
| 13:43.17 | *** join/#asterisk af_ (~getsmart@78.134.22.194) |
| 13:46.52 | *** join/#asterisk digilink (~digilink@irc.stephennet.net) |
| 14:23.08 | *** join/#asterisk digilink (~digilink@70.90.228.196) |
| 14:29.47 | *** join/#asterisk RWan (~kvirc@AMontpellier-553-1-67-128.w90-0.abo.wanadoo.fr) |
| 14:31.48 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 14:41.41 | *** join/#asterisk digilink (~digilink@70.90.228.196) |
| 14:45.27 | *** join/#asterisk Mhaddog_ (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 14:48.28 | IsUp | hi |
| 14:58.04 | *** join/#asterisk Mhaddog_ (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 15:07.55 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 15:16.31 | *** join/#asterisk v1s (~v1s@202.84.107.67) |
| 15:16.52 | v1s | is there way to boost call volume? |
| 15:17.00 | v1s | for just one person/ |
| 15:17.13 | IsUp | whats your channel technology? |
| 15:17.19 | IsUp | SIP ZAP IAX2? |
| 15:18.23 | v1s | sip or iax |
| 15:18.45 | IsUp | probably you have to edit your phone settings |
| 15:18.55 | IsUp | i dont know any way to "boost" volume on Asterisk |
| 15:26.42 | *** join/#asterisk RypPn (~TuMbL@unaffiliated/ryppn) |
| 15:35.54 | *** join/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 15:36.30 | z4nD4R | [Jan 8 15:41:30] WARNING[10845]: res_rtp_asterisk.c:2017 ast_rtp_read: RTP Read too short - when i want to call from iphone.... somebody??? |
| 15:49.43 | *** join/#asterisk bn-7bc (bjarne@2001:470:dc32:0:c62c:3ff:fe2d:b7db) |
| 15:52.22 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 15:58.42 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-088-066-137-246.pools.arcor-ip.net) |
| 16:09.57 | *** join/#asterisk chasing`Sol (~rc4@smtp.master-zone.net) |
| 16:10.03 | *** join/#asterisk Tim_Toady (~moi@62.1.174.14.dsl.dyn.forthnet.gr) |
| 16:18.03 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
| 16:19.01 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 16:44.30 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 16:48.50 | *** join/#asterisk markit (~marco@88-149-177-66.staticnet.ngi.it) |
| 16:53.03 | *** join/#asterisk vasanthv16 (~Nithya@216.195.17.249) |
| 16:57.15 | vasanthv16 | I am looking to buy some hardware for my asterisk hobby project. I am looking at probably an adapter that has one FXS port. Maybe also one FXO port. Any suggestions? |
| 16:59.56 | carrar | Digium TDM411 |
| 17:00.16 | carrar | optionally with the echo cancellation |
| 17:00.29 | carrar | but probably wanted |
| 17:01.01 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
| 17:01.16 | vasanthv16 | I am looking for something standalone. Not a pci card that connects to a computer. Something like http://www.amazon.com/Linksys-Cisco-SINGLE-ANALOG-ADAPTER-SPA1001/dp/B000BW0REG (which seems to be a very old model) |
| 17:02.09 | vasanthv16 | The idea is that I will run asterisk on my router (if at all that is possible) and have this adapter also connected to my intranet, so that I dont have to have a computer running all the time. |
| 17:02.24 | carrar | Lot of people use (Cisco) Linksys SPA2102 |
| 17:02.32 | frocostrocos | PAP2 |
| 17:02.59 | frocostrocos | linksys PAP2 |
| 17:03.53 | *** join/#asterisk vasanthv161 (~Nithya@216.195.17.249) |
| 17:04.37 | carrar | Here a config of a PAP2 |
| 17:04.38 | carrar | http://ui.linksys.com/files/PAP2/Admin-Basic.htm |
| 17:04.51 | carrar | well a actual login that is |
| 17:05.22 | *** join/#asterisk Denial- (Denial@drgi.co.uk) |
| 17:09.21 | vasanthv161 | carrar: The linksys one seems decent. It doesnt seem to have an FXO port though. Will check it out. thx. |
| 17:09.32 | carrar | You probably want Cisco SPA3102 |
| 17:09.38 | carrar | if you are going with that line |
| 17:09.44 | carrar | since that has 1 FXS, 1 FXO |
| 17:10.06 | carrar | They are really LinkSys |
| 17:10.13 | carrar | but Cisco just recently bought them out |
| 17:12.41 | vasanthv161 | Yup. That has an FXS and FXO port. Do you have any such hardware yourself? I am not even sure if the route that I am taking makes the best sense. |
| 17:13.02 | carrar | No, I use the internal Digium cards and or T1 with a ADIC600's |
| 17:13.30 | carrar | Adtran work well too |
| 17:13.43 | carrar | LinkSys is the low end stuff |
| 17:13.55 | carrar | but cheaper |
| 17:14.33 | frocostrocos | carrar: have u ever tested asterisk-java? |
| 17:14.38 | carrar | no |
| 17:14.50 | vasanthv161 | I am still finding my way around. So I think I will probably play with an SPA3102 first. |
| 17:14.56 | frocostrocos | anyone tested asterisk-java? |
| 17:14.58 | carrar | I don't use any GUI with Asterism |
| 17:15.00 | carrar | ASterisk |
| 17:15.23 | frocostrocos | it's a framework, not gui.. u can do a gui with it, if u want. |
| 17:19.48 | drmessano | I'm not real big on Java. I don't really see it taking off and going anywhere |
| 17:20.57 | carrar | besides Java is kinda going downhill |
| 17:21.21 | carrar | Apache is out of it because of Oracle |
| 17:21.40 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
| 17:22.03 | drmessano | Oracle is probably going to sue me for even mentioning it |
| 17:22.08 | carrar | yeah |
| 17:22.38 | carrar | I used to like SUN |
| 17:22.59 | WIMPy | Many of us did. |
| 17:24.28 | JunK-Y | lo guys |
| 17:24.32 | drmessano | Oracle was gonna sue the MySQL project before one of their accountants tapped the lawyer on the shoulder and told him "It looks like we own it, sir" |
| 17:24.56 | drmessano | Oracle is like the Vogons from the HHGG series |
| 17:24.59 | carrar | PostgreSQL is better anyways |
| 17:26.02 | drmessano | SQLite is where it's at |
| 17:27.56 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 17:30.24 | *** part/#asterisk vasanthv161 (~Nithya@216.195.17.249) |
| 17:44.13 | frocostrocos | i don't think java is going downhill.... there's to much big projects round it, including google, IBM, ... |
| 17:44.25 | *** join/#asterisk russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 17:45.27 | drmessano | I think Java as we know it won't be around much longer |
| 17:46.15 | drmessano | I expect several large forks that give Oracle the middle finger |
| 17:58.30 | raden_work | lol |
| 18:00.33 | *** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn101.78-98-155.t-com.sk) |
| 18:11.23 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 18:12.01 | *** join/#asterisk jetlag (~jetlag@pool-173-61-216-174.cmdnnj.east.verizon.net) |
| 18:12.09 | bio-tty | how is the content of an object of indefinite length escaped -- i mean, how may it contain two zero octets itself, without it being decoded as end-of-content |
| 18:12.36 | bio-tty | that was a question about BER encoding |
| 18:12.37 | *** join/#asterisk engrxyz (~cbvvvvv@host81-150-217-173.in-addr.btopenworld.com) |
| 18:13.54 | bio-tty | basic encoding rules |
| 18:39.30 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 18:41.57 | xSmurf | I wonder, is there a way I could pipe an espeak stream to a channel? |
| 18:42.16 | WIMPy | Does Zoiper barf on 10ms packet size? i got complaints about unusable audio quality and that's the only thing I can think of that I changed. |
| 18:42.18 | xSmurf | I know it's possible to pipe "live" (tail -f) text to espeak, |
| 18:42.35 | WIMPy | Use an EAGI? |
| 18:46.37 | xSmurf | more specifically this would ideally be done from AMI |
| 18:47.59 | WIMPy | To answer my question: Yes. Now I don't like zoiper any more. |
| 18:48.24 | xSmurf | also "The EAGI extension will let you receive sound from the channel into your application. It will not let you send sound" |
| 18:48.47 | WIMPy | You can't stream audio via AMI. But you can redirect to some dialplan location. |
| 18:49.07 | xSmurf | how can I stream audio from the dialplan? |
| 18:49.17 | xSmurf | does Playback allow a stream? |
| 18:49.30 | raden_work | where does asterisk store voicemail ? |
| 18:50.01 | WIMPy | Right. A normal AGI is enough. |
| 18:50.37 | WIMPy | mixed that up. |
| 18:50.51 | xSmurf | maybe http://www.voip-info.org/wiki/view/Asterisk+cmd+MP3Player ?? |
| 18:50.59 | xSmurf | but mp3? probably easier to stream a wav |
| 18:53.35 | *** join/#asterisk geneg1 (~gene@173-230-163-176.cable.teksavvy.com) |
| 18:58.59 | *** join/#asterisk sourcode_ (~code@ppp-58-8-182-3.revip2.asianet.co.th) |
| 18:59.05 | atan | So for some odd reason voip.ms servers try to pass a call over to my Asterisk box but fail. The first call will get a busy tone, and then the second, third, after it will ring through fine. And, every now & then the first call will be just fine as well. Is there something in here that might cause this? http://pastie.org/private/l36y7smk07kaua66lacolg |
| 19:01.14 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 19:15.02 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 19:16.07 | atan | Err my issue seems to be resolved by taking out that nat=yes part. Of course. |
| 19:19.59 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 19:25.40 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 19:26.19 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 19:26.19 | *** join/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 19:26.19 | *** join/#asterisk jezier (~jezier@nat.zgierz.net.pl) |
| 19:26.19 | *** join/#asterisk tvc123 (~tcameron@host-148.pl1071220.fiber.net) |
| 19:26.19 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 19:26.19 | *** join/#asterisk tris (~tristan@CAMEL.ETHEREAL.NET) |
| 19:26.19 | *** join/#asterisk akoma1s (quasselcor@unaffiliated/akoma1s) |
| 19:29.30 | *** join/#asterisk moltar_net (~Roman@180.183.201.91) |
| 19:35.27 | xSmurf | is there a way I can stream the output of text2wave? |
| 19:38.40 | *** part/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 19:49.03 | *** part/#asterisk Pimmetje (~Pimmetje@s5146b5f9.adsl.wanadoo.nl) |
| 19:55.51 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:57.29 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 20:05.36 | *** join/#asterisk fenrus (fenrus@drickheroin.se) |
| 20:11.08 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 20:17.16 | pabelanger | xSmurf: *CLI> core show application Festival |
| 20:17.28 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 20:18.22 | xSmurf | pabelanger: I don't think that allows a fifo for text input |
| 20:18.22 | antiwire | hey, If I'm using Background() for an AA, how can I stop it from immediately disconnecting the call after the menu prompts play? |
| 20:18.43 | antiwire | I get no time to enter a digit after all the prompts play but I can enter a digit while they are still playing |
| 20:19.18 | antiwire | i tried exten => s,n,Set(TIMEOUT(response)=60) but it doesn't seem to work |
| 20:20.32 | WIMPy | antiwire: core show application background |
| 20:21.38 | antiwire | WIMPy: doh! thanks |
| 20:21.49 | antiwire | WaitExten :) |
| 20:24.23 | antiwire | rad thanks WIMPy, go tit going |
| 20:24.25 | antiwire | it |
| 20:24.36 | xSmurf | this almost works with application=nc -l 9999 |
| 20:24.46 | *** join/#asterisk bn-7bc (bjarne@2001:470:dc32:0:c62c:3ff:fe2d:b7db) |
| 20:24.52 | xSmurf | and then echo "Hello there\!" | espeak --stdin --stdout | nc localhost 9999 |
| 20:24.56 | xSmurf | but the sound is distorted |
| 20:28.29 | *** join/#asterisk DelphiWorld (~VoIpGuy@41.200.21.135) |
| 20:28.32 | DelphiWorld | hey all |
| 20:28.38 | DelphiWorld | anyone worked with aculab? |
| 20:28.55 | *** join/#asterisk sshock (~sshock@sshock.net) |
| 20:33.58 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:38.07 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 20:42.42 | *** join/#asterisk Corydon76-home (six@c-69-137-80-31.hsd1.tn.comcast.net) |
| 20:42.42 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 20:43.49 | *** join/#asterisk bjhaid (~IceChat7@41.220.69.10) |
| 20:44.44 | DelphiWorld | welcome back Corydon76-home |
| 20:44.58 | Corydon76-home | Firewall rebooted |