| 00:02.05 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-138-172.chyn.qwest.net) |
| 00:02.21 | wooster | i figured out i was getting unauthorized when registering, so i added domains and now it says not a local domain |
| 00:05.04 | wooster | so domains isn't what i want, but why am i getting 401 unauthorized when register after updating to 10.0.0? |
| 00:08.21 | *** join/#asterisk francisvgarcia (~networker@186.1.90.193) |
| 00:08.29 | francisvgarcia | Well I am back |
| 00:08.54 | francisvgarcia | Here is the log of the call |
| 00:08.55 | francisvgarcia | http://pastebin.com/WaP4j47Z |
| 00:11.12 | WIMPy | And what's wrong wit that? |
| 00:11.59 | p3nguin | Dial and Queue both using tT = bad. |
| 00:12.25 | p3nguin | This means I can call you and transfer calls in your system. |
| 00:12.34 | p3nguin | You want only the callEE to be able to transfer. |
| 00:13.40 | WIMPy | The whole thing doesn't make too much sense to me. So what are you trying to do? |
| 00:13.42 | wooster | here's my conf and issue: http://pastebin.com/U57hrXQV |
| 00:13.45 | wooster | this worked in 1.8 |
| 00:14.28 | p3nguin | wimpy: He's using TDM, Digium card, and there is ringing on incoming calls. He does not care about receiving caller ID; he just wants the line to answer straight away without ringing. |
| 00:15.26 | WIMPy | You mean analog? |
| 00:15.29 | p3nguin | Yes. |
| 00:16.23 | WIMPy | Analog is evil. |
| 00:18.50 | p3nguin | So is there a setting on a TDM card that causes it to wait for two rings instead of answering immediately? |
| 00:18.52 | [TK]D-Fender | francisvgarcia: What kind of device is SIP/100? |
| 00:19.20 | francisvgarcia | GRandstream GXP1450 |
| 00:19.28 | [TK]D-Fender | francisvgarcia: And what do you have that card plugged into exactly? |
| 00:19.34 | p3nguin | He showed the call progress, and there is no Wait(), no Ringing(), and nothing else that I can see that should make it wait before answering. |
| 00:19.57 | francisvgarcia | I have the cord which comes from the wall |
| 00:20.06 | WIMPy | Analog != TDM or at least only extremely rarely. |
| 00:20.17 | francisvgarcia | If I plug the cable to a regular telephone |
| 00:20.17 | [TK]D-Fender | francisvgarcia: standard telco line? |
| 00:20.23 | francisvgarcia | it rings automatically |
| 00:20.25 | p3nguin | Maybe I misunderstood what he said, then. |
| 00:20.28 | francisvgarcia | yes |
| 00:20.29 | WIMPy | It may be DR and/or CID detection. |
| 00:20.51 | francisvgarcia | I disabled the callerid detection |
| 00:21.25 | francisvgarcia | If I plug a regular telephone into the wall It rings automatically |
| 00:21.37 | [TK]D-Fender | francisvgarcia: plug an anlog phon in paralle to the card and do a very fine-tuned test of what you hear VS CLI execution |
| 00:25.03 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 00:26.07 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 00:26.40 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 00:27.26 | francisvgarcia | I made a call from the cell. The Setup is this: Wall---->TDM410p. Results: I hear two rings in the cell phone before the asterisk CLI display "Starting simple switch on 'DAHDI/4-1.. Bla bla bla" |
| 00:27.53 | [TK]D-Fender | francisvgarcia: Yes well the telco could be ringing while your physical line is not |
| 00:27.58 | francisvgarcia | and process the call |
| 00:28.07 | [TK]D-Fender | francisvgarcia: Plug in a physical analog phone line in parallel at prove it |
| 00:28.25 | p3nguin | If that's the case, it is out of your hands as far as asterisk is concerned. |
| 00:28.42 | [TK]D-Fender | Cell's can ring before its left the cell co tower to every interconnecting switch in between |
| 00:28.53 | francisvgarcia | I did it and It rings no time before the phone rings |
| 00:28.54 | [TK]D-Fender | Go prove what is happening at the copper level next to your card |
| 00:30.08 | francisvgarcia | When I plug the analog phone It rings automatically. No pre-answer rings |
| 00:32.53 | wooster | i would really appreciate it if someone could take a look at this and tell me what's wrong: http://pastebin.com/U57hrXQV |
| 00:33.43 | p3nguin | If the caller hears 0 rings before your phone rings, it sounds like configuration with the card. I only know of wait time or ringing application in dial plan. |
| 00:35.39 | francisvgarcia | wooster: what is going on? |
| 00:36.52 | wooster | francisvgarcia: i upgrade to 10.0.0, this config used to work for me, now i get 401 unauthorized when registering |
| 00:41.28 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 00:43.29 | francisvgarcia | Try setting nat=no in sip.conf |
| 00:43.43 | francisvgarcia | and do a sip reload at asterisk CLI |
| 00:44.02 | francisvgarcia | wooster: Try setting nat=no in sip.conf |
| 00:44.02 | wooster | ok |
| 00:44.15 | p3nguin | If your asterisk is behind NAT, you need nat=yes. |
| 00:44.36 | wooster | francisvgarcia: no help there |
| 00:44.43 | p3nguin | As well as the appropriate localnet and externaddr or externhost settings. |
| 00:44.56 | wooster | it's not |
| 00:45.05 | wooster | the clients are |
| 00:45.25 | p3nguin | If asterisk it not behind NAT, nat=no belongs in the general section of sip.conf. |
| 00:45.38 | [TK]D-Fender | NAT has no impact on auth acceptance |
| 00:45.41 | p3nguin | If the phones are behind a NAT, nat=yes belongs in each peer entry. |
| 00:45.44 | [TK]D-Fender | checks for a full-moon |
| 00:45.51 | wooster | right |
| 00:46.00 | wooster | nat=no in general didn't change anything |
| 00:46.03 | p3nguin | I think there's no moon today. |
| 00:46.18 | autofsckk | 10 days to FM |
| 00:46.38 | p3nguin | I'd think it's more than that, since the moon cycle is not 20 days. |
| 00:46.38 | wooster | i've looked at the 10.0.0 sample sip.conf, there's nothing new or special in there |
| 00:47.01 | p3nguin | Should be more like 13 or 14 days till full moon. |
| 00:47.07 | citywok | Why won't an IP650 boot without the mac.cfg file existing on the tftp server? |
| 00:47.25 | p3nguin | There was just a tiny bit remaining on Friday. This is now Sunday. |
| 00:47.26 | WIMPy | Wu says 23% moon. |
| 00:47.29 | autofsckk | no, 10 days, i have a lunar calendar |
| 00:47.31 | p3nguin | No moon was probably yesterday. |
| 00:47.43 | francisvgarcia | wooster: what is your internal ip address |
| 00:47.47 | autofsckk | nov 10 |
| 00:47.57 | p3nguin | Maybe it was waxing when I thought it was waning? |
| 00:48.00 | francisvgarcia | wooster: * subnet |
| 00:48.02 | wooster | francisvgarcia: i don't have an internal IP address |
| 00:48.10 | wooster | this is a dedicated server |
| 00:48.22 | francisvgarcia | with a public ip address |
| 00:48.24 | WIMPy | 10 Nov, 20:15 UTC |
| 00:48.25 | p3nguin | If it was waxing, then no moon would have been like Thursday. |
| 00:48.26 | wooster | yes |
| 00:48.41 | p3nguin | And then 10 more days would be more reasonable. |
| 00:49.25 | p3nguin | Sheesh. Glad we got THAT taken care of. |
| 00:49.29 | wooster | so... i don't know what could possibly be causing auth to fail |
| 00:49.42 | wooster | is anyone running 10.0.0? |
| 00:50.09 | WIMPy | is using SVN |
| 00:50.14 | francisvgarcia | I am using it |
| 00:50.22 | francisvgarcia | with a Cisco 7960 |
| 00:50.37 | francisvgarcia | I have asterisk 10 beta 1 |
| 00:50.45 | WIMPy | At least the version from a few days ago seems fine so far. |
| 00:50.52 | francisvgarcia | virtualized |
| 00:51.02 | francisvgarcia | behind a nat |
| 00:51.07 | francisvgarcia | and is working properly |
| 00:51.30 | francisvgarcia | did you reconfigure your phone after the upgrade |
| 00:52.08 | *** join/#asterisk obnauticus (obnauticus@about/windows/regular/obnauticus) |
| 00:53.01 | francisvgarcia | Try using a softphone |
| 00:53.03 | francisvgarcia | like xlite |
| 00:53.05 | *** join/#asterisk FainaUkraina (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 00:53.13 | francisvgarcia | from ur computer |
| 00:53.29 | francisvgarcia | with the autentication parameters |
| 00:53.58 | wooster | i am trying softphones and a polycom |
| 00:54.05 | wooster | i did not reconfigure my phones |
| 00:54.17 | wooster | all the phones get 401 |
| 00:54.19 | francisvgarcia | even the softphones? |
| 00:54.21 | wooster | yes |
| 00:54.33 | francisvgarcia | hold on |
| 00:54.34 | wooster | all my users auth stopped working |
| 00:55.50 | francisvgarcia | let me paste you a sample |
| 00:55.50 | francisvgarcia | that works for 10.0.1 |
| 00:55.51 | francisvgarcia | sorry |
| 00:55.51 | francisvgarcia | beta 1 |
| 00:56.02 | wooster | ok |
| 00:57.02 | WIMPy | The only issue I have so far is that 'core restart when convenient' once again causes a deadlock instead of a restart. |
| 00:57.57 | WIMPy | And as 'sip reload' still doesn't really work, that's bad. |
| 00:58.15 | p3nguin | When all channels are dead, it didn't restart asterisk? |
| 00:58.37 | WIMPy | No, it just sits there, doing nothing. |
| 00:59.06 | p3nguin | That's what it normally does until channels are no longer. |
| 00:59.08 | WIMPy | Actually it died after some time this time. |
| 00:59.35 | p3nguin | When I use when convenient, it sometimes takes over a minute when there are no channels that I can see. |
| 00:59.45 | WIMPy | With older versions it usually just locked up, requiring a kill -9. |
| 01:00.19 | WIMPy | Invisible channels? |
| 01:00.29 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 01:00.40 | WIMPy | It used to work perfectly for quite some time with 1.8. |
| 01:06.52 | *** join/#asterisk SeRi (~seriosly@c-76-31-169-54.hsd1.tx.comcast.net) |
| 01:07.17 | SeRi | ok finally got a new ups... things seems to be stable now |
| 01:08.17 | francisvgarcia | wooster: Backup your sip.conf file and try this one http://pastebin.com/uSyHfQrV |
| 01:08.43 | francisvgarcia | wooster: and reload asterisk |
| 01:10.10 | wooster | k |
| 01:10.23 | francisvgarcia | wooster: cp /etc/asterisk/sip.conf /etc/asterisk/sip.conf.backup103020112110 |
| 01:11.10 | wooster | autodomain=yes |
| 01:11.14 | wooster | breaks registration |
| 01:11.18 | wooster | more |
| 01:11.23 | wooster | says not a local domain |
| 01:12.25 | francisvgarcia | It should not be |
| 01:12.30 | francisvgarcia | as I understand |
| 01:12.41 | wooster | domain is evil |
| 01:12.42 | wooster | and bad |
| 01:12.48 | wooster | and breaks if you're not only on a lan |
| 01:12.51 | wooster | i think |
| 01:13.00 | francisvgarcia | add this line then |
| 01:13.10 | francisvgarcia | domain=yourpublicip |
| 01:13.27 | wooster | i tried all that stuff, it made things worse |
| 01:13.29 | francisvgarcia | domain=yourdnsname |
| 01:14.07 | francisvgarcia | maybe ur using a dns name at the phones as sip proxy |
| 01:14.08 | p3nguin | I leave all my sip domain crap commented out. |
| 01:14.40 | francisvgarcia | comment ;autodomain=yes |
| 01:14.45 | francisvgarcia | reload sip |
| 01:14.47 | wooster | i did |
| 01:15.06 | francisvgarcia | and still refusing the authentication |
| 01:15.08 | francisvgarcia | ? |
| 01:15.35 | WIMPy | Either unload and load chan_sip or restart Asterisk. |
| 01:15.54 | WIMPy | I wouldn't trust a sip reload for much more than adding peers. |
| 01:16.47 | *** join/#asterisk epaphus (~user@201.199.62.74) |
| 01:16.59 | epaphus | Hello. Is there any place I can get a free or low cost toll free DID ? |
| 01:17.07 | wooster | francisvgarcia: yes, still failing |
| 01:17.17 | wooster | i'll resetart |
| 01:17.34 | francisvgarcia | wooster: /etc/init.d/asterisk restart |
| 01:18.02 | WIMPy | epaphus: If by DID you mean a directory number for inbound calls, probably yes, if you really mean DID, probably no. |
| 01:18.05 | francisvgarcia | or reboot |
| 01:18.23 | epaphus | where.. |
| 01:18.36 | WIMPy | 'core restart ...' is good enough ... unless it hangs. |
| 01:18.42 | wooster | i did core restart now |
| 01:18.47 | wooster | no help |
| 01:18.51 | francisvgarcia | you compiled it without restart |
| 01:18.57 | WIMPy | epaphus: You tell us. |
| 01:19.45 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 01:20.10 | wooster | it restarted, but the problem remains |
| 01:20.42 | wooster | where can i get beta2? |
| 01:20.45 | wooster | maybe that will help |
| 01:21.06 | carrar | heh |
| 01:21.25 | WIMPy | http://downloads.asterisk.org/pub/telephony/asterisk/ |
| 01:22.28 | francisvgarcia | It's a strange situation, I have been upgrading since the 1.8.0 without issues |
| 01:22.50 | p3nguin | sip reload should be enough to reload sip under any circumstances. |
| 01:23.01 | WIMPy | Definitely not. |
| 01:23.11 | p3nguin | Yes, that's what it does. |
| 01:23.18 | wooster | should |
| 01:23.29 | p3nguin | It rereads sip.conf. |
| 01:23.47 | carrar | not always the case |
| 01:23.55 | WIMPy | But it doesn't change all settings that have already existed before. |
| 01:24.17 | WIMPy | Adding peers seems safe. Changing existing ones is not. |
| 01:24.37 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 01:24.59 | p3nguin | epaphus: I pay $0.99 per month for each of my toll-free DIDs, and $0.024/minute for calls on them. |
| 01:25.07 | WIMPy | I just had to restart to remove nat from a peer. Just change nat=yes for nat=no and then 'sip reload' did not change it. |
| 01:25.17 | p3nguin | And the rest of the thought was going to be: Is that cheap enough? |
| 01:25.38 | p3nguin | I'll test that when I don't have any active calls. |
| 01:26.06 | WIMPy | I've also had trouble removing outbound registrations, but that seems to work now. |
| 01:40.29 | p3nguin | I changed nat=yes to nat=no, saved, ran sip reload, and compared the output. It changed two values, Symmetric RTP and Force rport, both from Yes to No. |
| 01:40.41 | p3nguin | I had one active SIP call when I did it. |
| 01:41.03 | WIMPy | But in 'sip show peers' it still showed the N. |
| 01:41.41 | p3nguin | I didn't change it on any peers. |
| 01:41.46 | p3nguin | I'll try one now. |
| 01:42.32 | p3nguin | Worked just fine. |
| 01:42.48 | p3nguin | I changed one peer from nat=yes to nat=no, sip reload, the N went away. |
| 01:43.08 | p3nguin | Changed it back, sip reload, the N is back again. |
| 01:43.23 | WIMPy | It didn't for me. |
| 01:43.31 | p3nguin | I can't think of any time sip reload hasn't worked correctly for me. |
| 01:44.04 | WIMPy | And I've had other situation where I couldn't find out why things didn't work until I did a restart and suddenly they did work. |
| 01:44.33 | francisvgarcia | hey guys |
| 01:44.57 | francisvgarcia | where is located the template file used to send the voicemial alerts to the users |
| 01:45.08 | francisvgarcia | I would like to change it for something else |
| 01:45.21 | francisvgarcia | and traslate it to spanish |
| 01:45.44 | p3nguin | voicemail.conf |
| 01:48.07 | francisvgarcia | Is the one that says "Just wanted to let you know you were just left a...." |
| 01:50.35 | WIMPy | oops. Sorry. I don't use nat= in peers any more. I change template. |
| 01:51.28 | francisvgarcia | opps |
| 01:51.31 | francisvgarcia | I found it |
| 02:21.39 | *** join/#asterisk sacitec (~newbie@189.251.99.53) |
| 02:22.17 | sacitec | good night people |
| 02:24.22 | sacitec | anyone working with patton smartnode FXS/FXS adapters ? |
| 02:24.59 | *** join/#asterisk cyborg-one (1000@188-115-189-185.broadband.tenet.odessa.UA) |
| 02:25.37 | sacitec | web interface is as complicated as CLI |
| 02:25.40 | *** part/#asterisk sacitec (~newbie@189.251.99.53) |
| 02:29.03 | *** join/#asterisk lovetide (~lovetide@211.154.128.135) |
| 02:50.00 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 02:52.50 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 03:08.20 | dijib | anybody in 2663@asterisk.serveirc.com |
| 03:08.21 | dijib | ? |
| 03:11.04 | p3nguin | I'm still connected and someone else is connected, but no one is talking. |
| 03:11.12 | p3nguin | I've just been letting it idle. |
| 03:11.59 | dijib | im here |
| 03:12.03 | dijib | in it |
| 03:13.33 | ChannelZ | Oooh am I missing a phone sex conf party? |
| 03:14.25 | p3nguin | Nope, he's not putting on any shows. |
| 03:16.01 | dijib | i can arrange anything |
| 03:18.25 | p3nguin | I'm not sure if I could pretend you're a 20 year old red headed female. |
| 03:20.31 | dijib | ive got a 22 y/o brunette girlfriend |
| 03:20.34 | dijib | does that help? |
| 03:20.41 | p3nguin | Close enough. |
| 03:21.27 | ChannelZ | http://www.tshirthell.com/images/contestpics/a249_003.jpg |
| 03:25.14 | dijib | more like this |
| 03:25.14 | dijib | http://i.imgur.com/nyVna.jpg |
| 03:25.37 | dijib | im going out for a smoke... 2663 me if you wanna say something to me ChannelZ |
| 03:30.43 | *** join/#asterisk adolfomaltez (~taro@190.62.240.147) |
| 03:32.13 | ChannelZ | what are we testing here |
| 03:32.22 | p3nguin | From whom did you steal the pic? |
| 03:32.58 | ChannelZ | I think that's him |
| 03:33.09 | p3nguin | He's new to having a working asterisk, so he's playing with ConfBridge(). |
| 03:33.30 | p3nguin | Oh? Damn, he's kinda purty. |
| 03:34.25 | p3nguin | Got a cute face. |
| 03:38.24 | SeRi | p3nguin, what is 2663@asterisk.serveirc.com? |
| 03:38.54 | dijib | its my sex show bridge |
| 03:39.01 | SeRi | lol |
| 03:39.02 | SeRi | nice |
| 03:39.08 | dijib | 2663@sexshow.hopto.org |
| 03:40.24 | SeRi | Is that you in that pic or your girlfriend? |
| 03:40.37 | ChannelZ | LOL |
| 03:40.45 | dijib | who knows |
| 03:41.04 | SeRi | rofl @ ChannelZ funny as pic |
| 03:41.06 | dijib | why did someone from riverside cali cal me |
| 03:41.47 | dijib | Seri, you cannot switch between value and premium outbound call routes. |
| 03:41.50 | ChannelZ | On the telemaphone? |
| 03:41.53 | dijib | fyi |
| 03:41.59 | dijib | no i havent opened a ticket |
| 03:42.04 | dijib | me? i am |
| 03:42.06 | dijib | she aint |
| 03:42.13 | dijib | but my shit hotter |
| 03:42.16 | dijib | shits |
| 03:42.39 | SeRi | dijib, what you talking about? |
| 03:42.40 | dijib | i wanna go plinking. |
| 03:43.17 | SeRi | premium and value? |
| 03:43.39 | dijib | you were talking about switching between voip.ms $0.0052/$0.0105 & $0.0125 outbound calling routes... using reseller account |
| 03:43.52 | dijib | value & premium |
| 03:44.18 | SeRi | me? |
| 03:44.21 | SeRi | not me. |
| 03:44.35 | SeRi | I said you can have a reseller account and bump the price up |
| 03:45.06 | ChannelZ | was going to go plinking later |
| 03:45.23 | ChannelZ | sort of anyway |
| 03:46.09 | SeRi | You where trying to do that I told you that you cant. |
| 03:46.53 | p3nguin | I never knew you could only switch the international route in sub accounts. |
| 03:46.59 | SeRi | Man I just cant stress enough how bad CC sucks! |
| 03:47.14 | p3nguin | I even went into the portal earlier looking for it to tell dijib how to do it. |
| 03:47.19 | p3nguin | Found out it wasn't there! |
| 03:47.41 | SeRi | :/ |
| 03:47.51 | p3nguin | Speaking of CallCentric... |
| 03:48.17 | dijib | i wanna go plinking. |
| 03:48.43 | p3nguin | I need to make sure my configuration for it still works. I never had the problems with it that people seem to have with it nowadays. |
| 03:48.52 | SeRi | you mean cock centric? |
| 03:49.23 | SeRi | I am so pist at them. |
| 03:50.35 | SeRi | p3nguin, what issues people seem to be having the most? |
| 03:50.40 | ChannelZ | pssst\ |
| 03:51.07 | p3nguin | It seems like the main complaint is that calls coming in never match the peer entry. |
| 03:51.28 | p3nguin | Outgoing works fine for me. Are you on there now so you can call me? |
| 03:51.48 | SeRi | Yes. Thats because of there redundant DNS. so when a call comes in and it gets match to CC asterisk pukes and declines the server |
| 03:51.52 | SeRi | you have to allow guest |
| 03:51.58 | SeRi | at least that was my experience. |
| 03:52.30 | SeRi | p3nguin, yes |
| 03:52.32 | SeRi | msg me |
| 03:59.16 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:15.08 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-135-115.chyn.qwest.net) |
| 04:17.32 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 04:20.52 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279681716.dsl.bell.ca) |
| 04:41.06 | *** join/#asterisk troyt (~troyt@c-67-166-68-75.hsd1.ut.comcast.net) |
| 04:41.21 | dijib | http://www.youtube.com/watch?v=XtkapY3ZfD4&feature=related |
| 04:47.10 | SeRi | http://www.youtube.com/watch?v=0ABGIJwiGBc |
| 04:47.28 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 04:50.45 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 04:56.04 | *** join/#asterisk radic (~radic@dslb-178-002-225-182.pools.arcor-ip.net) |
| 05:06.53 | SeRi | p3nguin, you still around? |
| 05:08.30 | SeRi | I am curious how do I dial a sip uri |
| 05:12.21 | SeRi | ill try it tomorrow... I have to go to sleep now. work and school tomorrow... ugh... is killing me... no trick or treating with the kids :( |
| 05:12.31 | SeRi | g/n all! |
| 05:16.39 | ChannelZ | Dial(SIP/hostname/exten) |
| 05:19.46 | p3nguin | Dial(SIP/exten@domain.com) |
| 05:26.22 | ChannelZ | It's like the same thing, only different. |
| 05:37.33 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 05:38.33 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 05:57.06 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 06:13.22 | *** part/#asterisk LiuYan (~liu.yan@211.154.128.135) |
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| 07:21.02 | schmidts | good morning |
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| 09:01.39 | *** join/#asterisk tbac (~tbac@p5DE84E60.dip.t-dialin.net) |
| 09:02.27 | tbac | hi. im' looking for a channel variable that holds the source ip address of a peer, does something like this exist? |
| 09:29.40 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 09:30.00 | *** join/#asterisk emate (~marcin@77-255-116-156.adsl.inetia.pl) |
| 09:30.20 | emate | hi |
| 09:31.07 | emate | i have problem with my isdn card and asterisk box |
| 09:31.44 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 09:31.51 | mandla | emate, what problem is that? |
| 09:32.14 | emate | lspci says, that i have isdn card onboard (03:04.0 ISDN controller: Cologne Chip Designs GmbH Unknown device 10b1 (rev 01) |
| 09:32.48 | emate | this is E1 card |
| 09:32.53 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 09:33.05 | emate | so i load hfcmulti driver |
| 09:33.18 | emate | and misnd-init says, that no card were detected |
| 09:34.20 | mandla | emate, did you install the firmware? |
| 09:34.21 | mandla | Oh yah, you did. |
| 09:34.44 | mandla | Is it not a hardware problem? |
| 09:36.01 | emate | on card i have some switches (http://www.atcom.cn/high%20resolution/AX1E.jpg) |
| 09:36.22 | mandla | emate, i have no experience with isdn, im on astribank. |
| 09:36.41 | emate | i tried to turn off and on all of this switches but with no result |
| 09:39.02 | emate | would card be dected it it's hardware problem? |
| 09:39.32 | emate | detected by lspci of course |
| 09:55.13 | mandla | If you do lspci and the isdn card is listed then the h/w is fine. Install the firmware/drivers. |
| 09:57.09 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 09:57.29 | irroot | mandla if they not already loaded can check lsmod too |
| 09:57.44 | irroot | what isdn card you using and what drivers |
| 09:58.09 | irroot | mandla dumelang :P |
| 09:58.48 | emate | Atcom AX-1E |
| 09:58.56 | mandla | irroot, lol, im fine my man, im straggling with call forwading this side. |
| 09:59.57 | irroot | mandla call forwarding is not too complex really |
| 10:00.13 | irroot | you can call the extension normally fine ? |
| 10:01.34 | mandla | irroot, yah i have about 24 extensions and they all can call one another. |
| 10:02.54 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 10:03.08 | mandla | irroot, and they can dial out and one of them is a switchboard, now calls are doing to the switchboard, and i need call forwarding to work, because the switchboard is downstairs. |
| 10:03.33 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 10:12.22 | mandla | irroot, ?? |
| 10:13.16 | irroot | mandla call forwarding or transfering ? |
| 10:19.25 | mandla | irroot, is there a difference. |
| 10:21.04 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 10:21.06 | mandla | Description: When i calls request for me at the switchboard the call should be transfered to my extension. |
| 10:21.06 | mandla | Description: When a caller request for me at the switchboard the call should be transfered to my extension. |
| 10:22.32 | ollii | question...is there a list with all sounds files available/needed by asterisk 1.8 ? |
| 10:33.41 | irroot | mandla yes in my mind at least transfering a call and forwarding it are 2 very different things |
| 10:33.55 | irroot | what switchboard phone you have ? |
| 10:35.36 | mandla | Its just a normal simple analog phone. |
| 10:40.47 | *** join/#asterisk fmota_ (~quassel@adonis.iportalmais.pt) |
| 10:43.25 | *** join/#asterisk fmota_ (~quassel@adonis.iportalmais.pt) |
| 10:43.52 | *** join/#asterisk nW44bsterdam (~Schnitzel@unaffiliated/benwa) |
| 10:44.53 | fmota_ | hello |
| 10:45.18 | fmota_ | a doubt about dahdi with ISDN cards |
| 10:45.32 | fmota_ | I m using openvox isdn cards |
| 10:46.10 | irroot | mandla how you trying to use it ?? also is it a south african type phone they have short flash timers that will mess you arround |
| 10:46.13 | *** join/#asterisk coppice (~chatzilla@m121-203-212-17.smartone-vodafone.com) |
| 10:46.21 | irroot | coppice morning sir |
| 10:46.23 | fmota_ | and I can set up the line if I load the dahdi module without connecting the cable to the card |
| 10:46.33 | fmota_ | is it normal? |
| 10:47.27 | coppice | irroot: hi |
| 10:49.36 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 10:49.53 | irroot | fmota_ no should be able too i dont use the dahdi isdn drivers in production but recently tested them with and without pluged in |
| 10:53.17 | fmota_ | Is there a reason why it should not be able too use the channel if I plug in the cable after loading the drivers? |
| 10:53.55 | fmota_ | I think it is really strange |
| 10:55.03 | stix | When I enable SIP-debugging in the CLI, will it also be logged to full.log as well? |
| 10:55.21 | fmota_ | if I load the drivers with the cables plugged in I m able to plug in and to plug out the cables any times I want and the system is always detecting it |
| 10:55.39 | mandla | Its a normal phone, all incoming calls coming through span_1 they are directed to this phone, now from there the receiptionist should be able to transfer the calls to specific extensions based on who the caller requests. |
| 10:56.44 | *** join/#asterisk maxhbp204 (~chatzilla@122.179.166.43) |
| 10:57.08 | fmota_ | the other way arround, it is like the driver has disabled the unplugged port for ever |
| 10:57.16 | *** join/#asterisk adeel (~adeel@24-246-63-106.cable.teksavvy.com) |
| 10:57.32 | maxhbp204 | Hi everybody, I have installed and configured asterisk 1.6.2.20 latest version and made dialplans to detect 1 digit response with file playback and variable set with read application |
| 10:58.06 | maxhbp204 | but in this version i think read application is not working fine, i have also keep ulaw and auto dtmfmode and i am able to see stmf on asterisk cli with logger enable |
| 10:58.12 | maxhbp204 | but it is not getting with read application |
| 10:58.37 | maxhbp204 | i have also applied patch which i have found in asterisk help for 1.6.2 branch on channel.c |
| 10:58.50 | maxhbp204 | but still having issue with read application detection on dtmf on server |
| 10:58.58 | maxhbp204 | can anybody help me please for this |
| 10:59.05 | *** join/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-228-190.w86-204.abo.wanadoo.fr) |
| 10:59.39 | maxhbp204 | exten => s,n,Read(__ANS,file1,1,,3,5) |
| 11:00.16 | maxhbp204 | i am using this way for read application, but it is not detecting dtmf on variable and it is playing back file as user has nothing entered |
| 11:00.25 | maxhbp204 | can anybody help me for this please |
| 11:18.20 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 11:20.06 | WIMPy | tbac: CHANNEL(peerip) |
| 11:20.08 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 11:22.24 | *** join/#asterisk dom| (~domi@mail.tas.de) |
| 11:23.26 | *** join/#asterisk dr_ (~dr@83.166.214.174) |
| 11:24.36 | dom| | i have teams (a makro that call two extensions and reply busy if one of them is busy), is there any way so get hints of that team? the lamp should light up if anyone from the team is busy |
| 11:24.47 | *** join/#asterisk coppice (~chatzilla@m121-203-212-17.smartone-vodafone.com) |
| 11:25.51 | WIMPy | You can list multiple devices in a hint. |
| 11:27.10 | dom| | like that? |
| 11:27.12 | dom| | exten => 20,1,Macro(team,SIP/12,SIP/13) |
| 11:27.12 | dom| | exten => 20,hint,SIP/12,SIP/13 |
| 11:27.27 | WIMPy | & |
| 11:27.35 | WIMPy | Like in Dial |
| 11:27.36 | dom| | ok, thanks |
| 11:28.51 | dom| | wow it works, thanks again ;) |
| 11:33.20 | irroot | mandla look up dial features and make sure they configured then you type the transfer code to use it |
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| 11:56.48 | mandla | irroot, in features.conf?? |
| 12:06.45 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
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| 12:16.26 | irroot | mandla yip |
| 12:17.41 | *** join/#asterisk nighty- (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 12:17.52 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
| 12:19.37 | mandla | irroot, PM |
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| 12:32.35 | Foxi352 | Hi fellow asterisk'xianers :-) Was there something changed for GotoIf between 1.6 and 1.8 (using Asterisk 1.8.7.1-1digium2~lucid atm) ? The following does no longer work after switching to 1.8: exten => s,15,GotoIf($["${VMBOX}"="novm"]?s-${DIALSTATUS},1)…. I have a NoOp(${VMBOX}) in the line just before the GotoIf, and that prints out novm …. but the GotoIf does not jump to the true label ... |
| 12:34.11 | [TK]D-Fender | Foxi352, Shouldn't have. Show us the complete call trace |
| 12:34.17 | [TK]D-Fender | ~pb |
| 12:34.17 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 12:34.19 | [TK]D-Fender | ^^^ |
| 12:34.57 | Foxi352 | of course… 1 sec |
| 12:36.33 | Foxi352 | [TK]D-Fender: http://pastebin.com/nZKEpDJA The relevant lines are line 63 and 64 |
| 12:37.05 | Foxi352 | I have a 1.6 in parallel, and the same dialplan evals to 1? on line 64 and jumps to _s- |
| 12:37.32 | kaldemar | Foxi352: wm != vm |
| 12:37.40 | [TK]D-Fender | kaldemar, Yup |
| 12:37.55 | [TK]D-Fender | Spieling is prefect! |
| 12:38.19 | Foxi352 | haeh ?? wait ... |
| 12:38.38 | [TK]D-Fender | Executing [307@from-internal:1] Macro("IAX2/306-6227", "exten-vm,nowm,307") <--- |
| 12:38.44 | [TK]D-Fender | "nowm" |
| 12:39.09 | Foxi352 | oh my bad … i guess while porting some of the dialplan to realtime there must have been a spelling mistake *shame* |
| 12:41.36 | *** join/#asterisk LiuYan (~LiuYan@222.125.132.191) |
| 12:42.02 | Foxi352 | ok, that did the trick .. you are the greatest :-) i stuggled nearly the whole day without seeing that stupid spelling mistake … |
| 12:42.10 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:43.08 | *** join/#asterisk francisvgarcia (~francis.g@190.6.137.113) |
| 12:43.20 | francisvgarcia | Good Morning everyone |
| 12:43.32 | francisvgarcia | I'm back |
| 12:43.36 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 12:43.53 | Foxi352 | Good afternoon ;-) |
| 12:44.00 | francisvgarcia | wooster: what's up? did u solve the issue? |
| 12:44.20 | *** join/#asterisk coppice (~chatzilla@m121-202-107-48.smartone-vodafone.com) |
| 12:44.45 | Foxi352 | If already i am with asterisk gods in here: Is it possible with 1.8 to have hints in dialplan ? Read something about -1 priority but iirc that does not yet work in 1.8.7 ? |
| 12:44.55 | Foxi352 | i mean hints in realtime db… sorry.. |
| 12:48.51 | [TK]D-Fender | Foxi352, hints have always been there |
| 12:49.04 | [TK]D-Fender | Ah.. realtime... umm.. no idea |
| 12:49.31 | Foxi352 | yes, i have them working via hints.conf included somewhere. But i would prefer in realtime db |
| 12:49.52 | carrar | realtime scaryness! |
| 12:49.59 | carrar | wooooooo |
| 12:50.07 | carrar | peeekabooooo |
| 12:50.13 | Foxi352 | :-) it works ok for us ... |
| 12:51.24 | carrar | Sounds like you got some research to do |
| 12:54.10 | Foxi352 | well, i googled alot already ….. i read about -1 priority in realtime extensions table, but that did not do it .. then i read that it does not yet work at all because it's in the source but buggy, and some major coding has to be done ….. |
| 12:54.22 | Foxi352 | anyway, i will leave it in .conf file for now i guess ... |
| 13:02.24 | carrar | Your answers, being with such a new release probably won't be in google |
| 13:03.31 | carrar | I'd look at the files supplied with the source, or actually look at the source it's self |
| 13:03.57 | carrar | There is a lot to be learned there |
| 13:04.59 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:06.53 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-gkomfukldokjbsph) |
| 13:07.15 | *** join/#asterisk jacobkiers (~jacobkier@82-168-168-85.ip.telfort.nl) |
| 13:08.11 | Kobaz | how do i do a factory reset on a polycom phonw if it can't boot fully |
| 13:08.30 | Naikrovek | best you can do is format it |
| 13:08.36 | Kobaz | there's 90 other phones on the lan that came up fine |
| 13:08.47 | Kobaz | and this one is waiting for network to initalize |
| 13:09.01 | Kobaz | you can format it from the startup menu? |
| 13:09.27 | *** join/#asterisk wengole (~bcole@178.78.119.76) |
| 13:10.22 | [TK]D-Fender | Kobaz, factory reset instructions are in the admin guide |
| 13:10.28 | *** part/#asterisk wengole (~bcole@178.78.119.76) |
| 13:10.48 | [TK]D-Fender | Kobaz, And you can always reprovision it from a fresh un-tar |
| 13:11.39 | puzzled | Kobaz: an IP670 factory reset is done by pressing all at once 4 6 8 * for 3 seconds or until you hear the beep |
| 13:11.43 | carrar | reset, what model? |
| 13:12.04 | carrar | http://support.polycom.com/PolycomService/support/us/support/voice/index.html |
| 13:12.42 | carrar | Let us, help you, help yourseld! |
| 13:12.45 | carrar | +f |
| 13:12.46 | Naikrovek | if it's waiting for the network to initialize, then you have a network issue. |
| 13:13.12 | Kobaz | 331 |
| 13:13.50 | carrar | http://support.polycom.com/global/documents/support/setup_maintenance/products/voice/spip_ssip_Admin_Guide_SIP_3_1.pdf |
| 13:13.58 | carrar | assuming you are using 3.1 |
| 13:14.00 | Kobaz | hmm |
| 13:14.10 | Kobaz | i've been through the admin guide a bajillion times before for other things |
| 13:14.16 | Kobaz | never noticed a factory reset key sequence |
| 13:15.04 | carrar | What are you saying? |
| 13:15.27 | dom| | i have exten => voicemail,1,VoiceMailMain(${CALLERIDNUM},s) in my context but if i press die VM-Button on the snom m9 it executes: -- Executing [voicemail@from-internal:1] VoiceMailMain("SIP/12-00000090", ",s") in new stack |
| 13:15.42 | dom| | why is the calling extension not submittted? |
| 13:15.53 | carrar | "Resetting to Factory Defaults" |
| 13:16.07 | carrar | 3-5 |
| 13:16.19 | carrar | please |
| 13:16.22 | carrar | take my hand |
| 13:16.35 | carrar | lets run together and face the cool wind in our hair |
| 13:17.49 | *** join/#asterisk clintc (~clintc@n128-227-139-116.xlate.ufl.edu) |
| 13:18.55 | puzzled | Kobaz: how about searching for "factory defaults" in the admin guide? |
| 13:19.19 | Kobaz | hehe |
| 13:19.32 | Kobaz | i've never specifically searched for it, but thanks |
| 13:19.42 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 13:19.59 | Kobaz | i've just never come across it while searching for other things |
| 13:20.06 | *** part/#asterisk clintc (~clintc@n128-227-139-116.xlate.ufl.edu) |
| 13:20.11 | Kobaz | but, that's good to know it's in there, i'll try that |
| 13:20.55 | carrar | It's good to know you are helpless! |
| 13:20.58 | carrar | heh |
| 13:21.06 | carrar | notes this |
| 13:21.08 | puzzled | it's the simple things that make the difference. search for "factory defaults" or wade through 500+ pages of admin guide hoping my eyes will catch "factory defaults" :) |
| 13:21.09 | *** join/#asterisk dmz (~dmz@64.203.235.49.dyn-cm-pool-34.pool.hargray.net) |
| 13:21.27 | dom| | no idea, why the extension is not submitted to the voicemailmain? |
| 13:21.34 | carrar | because you haven't figured out how to use search in PDF's :) |
| 13:21.57 | puzzled | dom|: maybe the Snom does not have the vm extension in its config? |
| 13:22.14 | puzzled | carrar: on my boxes it's always ctrl-f |
| 13:22.30 | puzzled | and I'm lazy so it's one of the first things I do :) |
| 13:22.53 | dom| | puzzled, the extension is called, but ${CALLERIDNUM} seems to be empty |
| 13:22.57 | [TK]D-Fender | dom|, Because you didn't |
| 13:23.14 | [TK]D-Fender | dom|, that variable was deprecated over 5 years ago |
| 13:23.23 | puzzled | speaking of Polycom, anyone have that shiny new 4.0.0 firmware with accompanying BootROM files? |
| 13:23.25 | [TK]D-Fender | dom|, Stop using * 1.0 vars |
| 13:23.37 | Naikrovek | puzzled: no, but i want a copy |
| 13:23.59 | *** join/#asterisk francisvgarcia (~francis.g@190.6.137.113) |
| 13:24.00 | Naikrovek | several people here have access to it. i just haven't asked anyone for it, yet. |
| 13:24.03 | Katty | dear god |
| 13:24.10 | puzzled | Naikrovek: so do I but Polycom will only provide it to distributors or companies with access to that partner site |
| 13:24.11 | carrar | yes |
| 13:24.11 | Katty | my poor liver |
| 13:24.14 | carrar | Katty help us |
| 13:24.20 | puzzled | hi Katty |
| 13:24.20 | dom| | [TK]D-Fender, oh ok, what's the pendant in 1.8? |
| 13:24.26 | Katty | hugs carrar |
| 13:24.31 | Katty | help wif whats |
| 13:24.31 | carrar | w00t!! |
| 13:24.34 | Katty | hugs puzzled |
| 13:24.39 | carrar | does the katty hug! |
| 13:24.39 | [TK]D-Fender | dom|, since * 1.2 <- "core show function CALLERID" |
| 13:24.46 | Katty | morning fender bender. |
| 13:24.48 | carrar | Japanese STYLE!! |
| 13:25.01 | dom| | thx [TK]D-Fender |
| 13:25.23 | puzzled | dom|: I would get the 1.8 book, check the 1.8 wiki and take a peek at the UPGRADE-1.x.txt files |
| 13:25.35 | Kobaz | puzzled: well my point was i never previously needed a factory reset so i never searched for it |
| 13:26.00 | puzzled | but when you needed it, you did not search for it the logical way |
| 13:26.10 | Kobaz | i asked in here first |
| 13:26.23 | [TK]D-Fender | Katty, Mew. |
| 13:26.29 | Kobaz | anyways... |
| 13:26.31 | puzzled | why not learn to fish instead of asking for fish? |
| 13:26.50 | puzzled | that's the point carrar was trying to make |
| 13:26.57 | carrar | use bate! |
| 13:27.02 | Kobaz | i know how to fish, but if you're at a lake and you see someone there, it's usually a good idea to ask the guy "hey, where are the fish" |
| 13:27.24 | carrar | What if that guy is a EX CON |
| 13:27.31 | Kobaz | then that's bad |
| 13:27.33 | puzzled | ha like anyone knows where the fish is |
| 13:27.42 | puzzled | in the lake is about as precise as it gets |
| 13:27.45 | dom| | puzzled, yes is should do... i'm not new to asterisk but didn't use it for some years... now i have to on my new job ;) |
| 13:28.03 | Kobaz | fish hang out at more areas than others |
| 13:28.08 | Kobaz | depends on food source location |
| 13:28.10 | puzzled | dom|: have fun. lots of new goodies to deploy |
| 13:28.17 | Kobaz | and temperature, and etc |
| 13:29.02 | francisvgarcia | Hello Guys, let's do it like yesterday |
| 13:29.13 | francisvgarcia | paste this in your dial plan to join us |
| 13:29.13 | francisvgarcia | Dial(SIP/2663@asterisk.serveirc.com) |
| 13:29.14 | puzzled | Kobaz: only by estimation. doing a simple search in the admin guide would have been much easier :) |
| 13:29.40 | dom| | puzzled, my first task is to port some boxes from 1.4-bristuff to a clean 1.8 ... it sucks ;) |
| 13:30.16 | puzzled | dom|: heh you are in for a treat. bristuff is nasty. which cards are you using? |
| 13:31.04 | dom| | no cards any more. we are selling own hardware with a own os that gates s0 and s2m to sip |
| 13:31.14 | dom| | called "ypsilon" |
| 13:31.59 | puzzled | dom|: ah right. if you want to stay with the Cologne chip ISDN products then mISDN v2 is probably the way to go. check misdn.eu |
| 13:32.39 | irroot | or mISDN 1 with chan_misdn |
| 13:32.50 | Kobaz | puzzled: well if i was using a 650, the answer came back quicker than i could load up my pdf and search it |
| 13:33.04 | Kobaz | er, 670 |
| 13:33.15 | [TK]D-Fender | dom|, Go DL 1.6.0, 1.6.1, 1.6.2, and 1.8.0 and read all of the upgrade docs between each |
| 13:33.25 | Kobaz | speaking of polycoms |
| 13:33.32 | puzzled | dom|: what irroot says :) but only when you run an older kernel. afaik the newer stuff does not work with mISDN v1 |
| 13:33.46 | puzzled | dom|: newer stuff == newer kernels |
| 13:33.47 | Kobaz | anyone ever have a problem where you Dial(SIP/polycomphone) and it reports ringing, but the phone isn't actually ringing |
| 13:33.51 | dom| | puzzled, no need for any channeldriver ... http://www.tas.de/telekommunikation/telefonanlagen-tas-com/ypsilon-survivable-media-gateway.html |
| 13:34.13 | puzzled | Kobaz: you mean you don't always have the Polycom Admin Guide open? :) |
| 13:34.16 | dom| | its own hardware, based on arm and cologne chips |
| 13:34.22 | Kobaz | puzzled: heh, not always |
| 13:34.38 | puzzled | good for you. neither do I or I would have gone completely bonkers already |
| 13:34.56 | puzzled | dom|: only 8 Watt. nice |
| 13:35.21 | Naikrovek | i printed that polycom admin guide, and spiral-bound it. comes in h andy |
| 13:37.28 | *** join/#asterisk mbrevda_ (~mbrevda@unaffiliated/mbrevda) |
| 13:37.33 | puzzled | Naikrovek: yeah was thinking about that too. doublesides printing with 2 pages per page would make it bearable |
| 13:37.42 | mbrevda_ | how can I remove the manager output from core show debug in 1.8? |
| 13:37.50 | Katty | naps on energy drink |
| 13:37.58 | Naikrovek | naps? |
| 13:38.00 | puzzled | dom|: so what are you going to use to drive the HFCS stuff with 1.8? |
| 13:38.11 | Katty | too tired to open it |
| 13:38.42 | Katty | ok it's open now |
| 13:38.56 | puzzled | bottoms up! |
| 13:39.06 | dom| | i do not use andy hfc-stuff, i have no channldiriver at the asterisk ... the ypsolion has its own firmware and gates between internal/external s0 oder s2m and IP |
| 13:39.39 | dom| | s/andy/any |
| 13:40.41 | Katty | christ on a bicycle |
| 13:40.42 | dom| | the ypsilon connects via sip to the * and offers 4 internal and for external channels (s0) or 30 internal and 30 external channels (s3m) |
| 13:40.46 | puzzled | dom|: ok, but what do you need the bristuff for then? |
| 13:41.37 | dom| | the older version was bristuffed for hints, pickup-patches and so on... now with 1.8 there is no need for bristuff anymore |
| 13:43.44 | dom| | the older version it not my work... im new in the company... it's my 3rd week ;) |
| 13:44.11 | puzzled | got it. I thought you were using HFC-S based ISDN chips because that is usually the reason why people use bristuff |
| 13:44.30 | dom| | only used asterisk private in the past ... with misdn and prior with zaptel |
| 13:45.18 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
| 13:48.52 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 13:53.38 | dom| | mhh... my asterisk is missing some sound files... File digits/1F does not exist in any format (in german) but i have asterisk-prompt-de installed |
| 13:55.54 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 13:58.43 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 13:59.38 | schmidts | dom| thats a very common problem cause there is no german file for this |
| 13:59.51 | eppigy | good morning |
| 14:00.22 | dom| | uh ok ... schmidts are there any workarounds for the voicemail-foo? |
| 14:01.47 | [TK]D-Fender | dom|, Go make one |
| 14:02.32 | schmidts | dom| there is a webpage from an university in karlsruhe i think they offer another german package where this file is in there |
| 14:02.53 | dom| | schmidts, free for commercial use? |
| 14:03.04 | dom| | or only private? |
| 14:03.25 | Katty | naps on eppigy |
| 14:04.00 | eppigy | smells Katty's hair |
| 14:04.25 | Katty | i found a new fragrance at macy's i have to have. |
| 14:04.34 | Katty | it's called villian |
| 14:05.14 | n3hxs | Interesting name... |
| 14:05.23 | Katty | smells amazing |
| 14:05.35 | Katty | possibly better than the victoria secret ones |
| 14:05.51 | *** join/#asterisk jacobkiers (~jacobkier@82-168-168-85.ip.telfort.nl) |
| 14:06.28 | n3hxs | hasn't wandered Victoria Secret's aisles in some time. |
| 14:06.29 | eppigy | i wear polo black |
| 14:06.41 | eppigy | ive had it for a while though maybe time to find a new one |
| 14:06.52 | Katty | polo black is ok... |
| 14:06.55 | Katty | but kind of meh |
| 14:07.03 | eppigy | :[ |
| 14:07.12 | Katty | i would recommend an update |
| 14:07.29 | eppigy | i thought so |
| 14:07.30 | Katty | go to macy's to get them, they will give you samples to try |
| 14:07.43 | eppigy | I WILL SAMPLE EVERYTHING |
| 14:07.49 | Katty | ALL THE THINGS |
| 14:07.58 | eppigy | yuesh |
| 14:07.59 | n3hxs | Including the blonde? |
| 14:08.06 | eppigy | especialy the blonde |
| 14:08.10 | Katty | i'll go with you |
| 14:08.15 | eppigy | sweet |
| 14:08.30 | eppigy | i went shopping at bloomngdales over the weekend |
| 14:08.37 | eppigy | they had 30-40% off mens stuff |
| 14:08.42 | eppigy | made out like a bandit |
| 14:08.46 | Katty | sweeeet |
| 14:08.48 | eppigy | new hugo boss coat |
| 14:08.50 | eppigy | BOOYA |
| 14:08.55 | Katty | i wanna see |
| 14:09.52 | *** join/#asterisk master_of_master (~master_of@p57B54FFD.dip.t-dialin.net) |
| 14:10.15 | eppigy | Katty: http://www1.bloomingdales.com/catalog/product/index.ognc?ID=555297&PseudoCat=se-xx-xx-xx.esn_results |
| 14:10.57 | Katty | that's nice. |
| 14:11.00 | Katty | i'd nap on it |
| 14:11.11 | eppigy | :] |
| 14:11.43 | Katty | you should check out the giorgio armani fragrances |
| 14:12.12 | Katty | the gucci guilty intense is pretty nom too |
| 14:13.01 | eppigy | i will then |
| 14:13.26 | Katty | kat von d has a new fragrance out too |
| 14:13.27 | Katty | that i need |
| 14:13.30 | Katty | it's called Saint |
| 14:13.30 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:13.30 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:13.36 | Katty | but it smells like no saint. |
| 14:13.43 | Katty | farrrrrr from it |
| 14:15.14 | eppigy | lol |
| 14:15.30 | eppigy | for the saint who likes to sin |
| 14:16.27 | *** join/#asterisk jeffspeff (~jeffspeff@173-11-144-149-houston.txt.hfc.comcastbusiness.net) |
| 14:17.59 | *** part/#asterisk LiuYan (~LiuYan@222.125.132.191) |
| 14:18.15 | jeffspeff | I'm experiencing an intermittent issue where * uses up all available memory and we have to restart the service. is this a known issue? how can prevent/resolve this? |
| 14:20.05 | [TK]D-Fender | jeffspeff, what version? What circumstances? How much memory do you have on that box? What else is it doing? What is the average airspeed velocity of an unladen swallow? |
| 14:20.08 | *** join/#asterisk mcr (~mcr@2001:4830:116e:1:20d:60ff:fefa:7f03) |
| 14:20.36 | Katty | eppigy: YES |
| 14:20.47 | mcr | I'm trying to forward a call that comes in to a support line to a few mobile phones if no-one in the office replies. I want to avoid voice mail of the mobile phones. |
| 14:20.48 | *** join/#asterisk r1ppa (~McBoingBo@mail.hrsg.ca) |
| 14:20.54 | jeffspeff | [TK]D-Fender, the average airspeed velocity of an unladen swallow = peppermint |
| 14:20.55 | jeffspeff | lol |
| 14:21.01 | r1ppa | Gidday folks! |
| 14:21.04 | Katty | peppermint! |
| 14:21.06 | mcr | So, aside from a rather short ring time, is there anything SIP related that might work at some point? |
| 14:21.57 | Katty | i didn't know that peppermint was a velocity?! |
| 14:22.00 | [TK]D-Fender | mcr, No, nothing SIP related. If the cell company hits VM, too bad, that's an "answer". "core show application dial" ,_ M |
| 14:22.03 | [TK]D-Fender | "M" |
| 14:22.32 | McBoingBoing | Trying to troubleshoot some call quality issues, http://pastebin.com/cPTHQzQR this is what I get with one of my problem extensions, is there anything I can do to resolve this issue? I will be replacing X-Lite for Bria (for the G.729 codec) to help out |
| 14:22.59 | jeffspeff | [TK]D-Fender, version 1.8.5, total ram = 8gb ddr3, box runs appache that locally hosts a few documentation pages (not public), and webmin... i think that's about it |
| 14:23.52 | [TK]D-Fender | McBoingBoing, That is a psycho registration timeout period and your provider sucks for using it as a keep-alive like that. It also doesn't actually show a technical "problem". |
| 14:23.57 | jeffspeff | Katty, i was making fun of the Netflix commercials you hear on radio where they ask these rediculous questions with bizarre answers then the final question has something to do with netflix. lol |
| 14:24.09 | [TK]D-Fender | jeffspeff, 1.8.7.1 is out. You should already have upgraded |
| 14:24.29 | McBoingBoing | [TK]D-Fender: Everytime that notice came up our call experienced a small glitch in sound |
| 14:25.09 | jeffspeff | [TK]D-Fender, so memory leaks are known in 1.8.5? i didn't want to upgrade on this production system just yet until we iron out a few remaining bugs/issues |
| 14:25.22 | [TK]D-Fender | McBoingBoing, stil not conclusive as to what the origin of the problem is |
| 14:25.34 | McBoingBoing | We got remote folks using VPN and VOIP, and I am trying to filter out real issues from PEBKAC and overwhelming the bandwidth |
| 14:26.46 | Katty | NEXT TIME WE EAT KEVIN BACON |
| 14:26.52 | McBoingBoing | [TK]D-Fender: well during every reregistration we experienced a "hiccup" in our call, so you are saying there is another offender? |
| 14:27.01 | [TK]D-Fender | Katty, THAT'S SMART |
| 14:27.16 | McBoingBoing | BACON WEAVE, NICE |
| 14:28.09 | [TK]D-Fender | can't say. No network graph to look at, no link details. Hopefully at this point you can see we're flying blind here. That little reg snippet really doesn't offer anything |
| 14:28.41 | [TK]D-Fender | jeffspeff, You already have critical problems.Upgrade should be near the top of your list |
| 14:28.47 | Faustov | could someone experienced with dandi please let me know what happens if the same extension pattern is advertised by more than one host? What is the default behavior? |
| 14:29.07 | [TK]D-Fender | dandi? Is that like fine? |
| 14:29.16 | [TK]D-Fender | Or just "ok"? |
| 14:29.17 | Faustov | erm |
| 14:29.18 | Faustov | dundi |
| 14:29.19 | jeffspeff | [TK]D-Fender, yeah, and i need to fix DTMF issues too. lol |
| 14:29.20 | Faustov | sorry ;) |
| 14:30.58 | McBoingBoing | [TK]D-Fender: what data should I be gathering to further troubleshoot VOIP call issues? |
| 14:31.12 | McBoingBoing | network on the SIP NIC is quiet, as always |
| 14:31.21 | [TK]D-Fender | McBoingBoing, Network traffic graph |
| 14:31.56 | [TK]D-Fender | McBoingBoing, And why do I get the impression "SIP NIC" is an unreliable fraction of the actual scenario? |
| 14:32.22 | mcr | [TK]D-Fender, yeah, I figured as much. I was hoping that perhaps there was something emerging. |
| 14:32.31 | *** join/#asterisk FinboySlick (~shark@74.117.40.10) |
| 14:32.35 | McBoingBoing | say what what now... |
| 14:32.49 | [TK]D-Fender | mcr, "M" <- |
| 14:32.54 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 14:33.51 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 14:34.47 | FinboySlick | Hello gang. I'm looking for some advice on 'receptionist' phones that would be most compatible with switchvox products. A while back Polycom 601 with the expansion module was apparently a good choice, what would any of you recommend for a 'no coputer' somewhat dummy-proof setup? |
| 14:35.48 | jeffspeff | FinboySlick, we gave ours some Cisco SPA504g's |
| 14:35.58 | p3nguin | Whatever you get, you'll want a sidecar or two. |
| 14:36.14 | jeffspeff | FinboySlick, they're very customizable and easy to work with and use. |
| 14:36.23 | FinboySlick | p3nguin: Sidecar being the expansion module? |
| 14:36.23 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:36.28 | p3nguin | finboyslick: correct |
| 14:36.37 | p3nguin | Aastra phones are nice, but I don't know if they have sidecars. |
| 14:37.00 | Faustov | no one on dundi? |
| 14:37.05 | dom| | schmidts, do you have a link to the german prompts? i only found some dead links (stadt pforzheim, amooma, ....) |
| 14:37.15 | p3nguin | Nope, no one uses DUNDi. |
| 14:37.22 | FinboySlick | jeffspeff: For the SPA504s, would they display who's busy, allow transfers and everything? This has to allow a receptionist to do their job without involving the computer. |
| 14:37.33 | Faustov | p3nguin: is something else recommended instead? |
| 14:37.46 | [TK]D-Fender | FinboySlick, How many phones do you have? |
| 14:37.50 | p3nguin | faustov: I was being facetious. |
| 14:37.59 | Faustov | p3nguin: thought so |
| 14:38.26 | *** join/#asterisk ph8 (~ph8@unaffiliated/ph8) |
| 14:38.27 | jeffspeff | FinboySlick, they don't show who's busy or on the phone like PSTN phones; they're sip |
| 14:38.28 | Faustov | but I'm also a bit surprised, I thought the implementation is much more common, yet for most questions I had to build a test system |
| 14:38.30 | FinboySlick | [TK]D-Fender: This setup will have under 15 and be built around a switchvox SMB305 |
| 14:38.56 | [TK]D-Fender | FinboySlick, Polycom is still a good choice, but the 601 has been disco'd for years now |
| 14:38.57 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 14:39.30 | FinboySlick | jeffspeff: Yeah, that's sort of expected, but since it's replacing a traditional phone system for non-techies, I'd like to keep the re-learning minimal. |
| 14:39.46 | FinboySlick | [TK]D-Fender: 650 seems to be its successor, right? |
| 14:40.09 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 14:40.12 | wcselby | o/ |
| 14:40.17 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:40.17 | jeffspeff | FinboySlick, we just did the same thing. they're likeing the new phones a lot better than old ones |
| 14:40.18 | FinboySlick | I think the important bit is that some sort of indicator lights up when someone is already on the phone. |
| 14:40.25 | wcselby | So did 10 get released this weekend? |
| 14:40.34 | wcselby | without an annoucement email? |
| 14:40.48 | [TK]D-Fender | FinboySlick, Amongst them |
| 14:40.53 | Naikrovek | Polycom 650 is where it's at for receptionist phones |
| 14:41.01 | p3nguin | wcselby: The beta1 is still the one on the download page. |
| 14:41.06 | [TK]D-Fender | wcselby, No. |
| 14:41.09 | wcselby | odd |
| 14:41.19 | wcselby | i got a twitter message from digium saying 10 was released |
| 14:41.40 | [TK]D-Fender | wcselby, Plenty of "jump the gun" schmuck articles though talking as though it were in full release |
| 14:41.50 | p3nguin | And the beta2 is the only one I see in the downloads. |
| 14:41.52 | wcselby | but yeah I wasn't seeing 10 on the download page, so I thought maybe I was missing something |
| 14:42.02 | wcselby | 10 full I mean |
| 14:42.09 | wcselby | meh, I'm talking faster than I'm typing |
| 14:42.17 | FinboySlick | Naikrovek: Can they be programmed to 'watch buddies', and light up who's busy? |
| 14:42.23 | Naikrovek | FinboySlick: yeppers |
| 14:42.37 | [TK]D-Fender | Heck the topic here says "beta2" and the webpage is still on beta1 |
| 14:42.46 | Naikrovek | that's how my receptionist uses hers |
| 14:43.22 | p3nguin | But the beta2 is the only one I see in the downloads. |
| 14:43.24 | Naikrovek | i imagine the topic here would change first; though leifmadsen is away ATM and he usually updates that I think. |
| 14:43.27 | FinboySlick | Naikrovek: That's promising. You think that'd work well with switchvox hardware? (I expect so, but not having that feature would be a no-go for this project) |
| 14:43.28 | wcselby | lol @ http://www.digium.com/en/mediacenter/viewpress/digium-and-open-source-community-release-asterisk-10-at-astricon |
| 14:43.33 | leifmadsen | Naikrovek: I'm here |
| 14:43.42 | Naikrovek | FinboySlick: yeah it should work fine, i'd think. |
| 14:44.03 | leifmadsen | Just to tell everyone -- Asterisk 10 is not released. Someone got a bit trigger happy on a release announcement. |
| 14:44.09 | Naikrovek | leifmadsen: nice. |
| 14:44.18 | wcselby | :) |
| 14:44.42 | p3nguin | There hasn't even been an RC given out yet! |
| 14:44.45 | leifmadsen | I will update the website to point to beta2 |
| 14:44.47 | Faustov | leifmadsen: thought so, what are your plans regarding the beta however: how many do you expect or how long do you see the stabilization taking place? |
| 14:45.07 | leifmadsen | I'm going to aim for RC1 today. We hope to have a full release ready in about 2 weeks. |
| 14:45.12 | leifmadsen | No one will test until the full release anyways :) |
| 14:45.19 | Naikrovek | heh |
| 14:45.19 | Faustov | correct |
| 14:45.25 | Faustov | at least not in production... |
| 14:45.49 | Naikrovek | "I'll wait until release to test." *finds bugs in release* "no one tests anything anymore" |
| 14:46.07 | wcselby | I'm still getting used to 1.8 |
| 14:46.08 | wcselby | :) |
| 14:46.24 | leifmadsen | that's pretty much the deal. For some reason people think a release is something different from any other snapshot in time :) |
| 14:46.44 | leifmadsen | Asterisk 10 and 1.8 are fairly similar except Asterisk 10 has some nice things in it 1.8 doesn't. |
| 14:46.56 | leifmadsen | you shouldn't really have to make many changes between 1.8 and 10 |
| 14:47.03 | Faustov | leifmadsen: is version 10 going to be any more long-term than 1.8? |
| 14:47.04 | wcselby | 10 has some cool features in regards to Confrencing, but I didn't see any other major enhancements over 1.8. I admittedly haven't looked very deeply either. |
| 14:47.11 | leifmadsen | ~asteriskversioning |
| 14:47.11 | infobot | Asterisk versions and their support levels are documented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 14:47.16 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 14:47.20 | wcselby | Faustov- no. 1.8 is LTS, 10 is normal 1 year support |
| 14:47.26 | pabelanger | ^ |
| 14:47.33 | leifmadsen | wcselby: just media handling in chan_sip (which permits the better conferencing in Asterisk 10) |
| 14:47.39 | wcselby | 1.8 is LTS means 4 years of support plus an extra year of security fixes |
| 14:47.47 | Naikrovek | video conferenceing |
| 14:47.47 | leifmadsen | it's all documented too :) |
| 14:47.57 | wcselby | leifmadsen- :) |
| 14:48.02 | wcselby | Naikrovek- yep |
| 14:48.04 | Faustov | sorry for asking, I saw that document many times |
| 14:48.15 | Faustov | got confused at some point |
| 14:48.31 | wcselby | Naikrovek- but no brady bunch conferencing, but it does auto-switch to the video feed of whoever is talking (or it's supposed to, I haven't tested it yet lol) |
| 14:48.38 | Naikrovek | it happens, no worries |
| 14:48.45 | Naikrovek | wcselby: brady bunch conf would be neat |
| 14:48.51 | leifmadsen | I doubt we'll see multiplexed video conferencing anytime soon |
| 14:49.06 | Naikrovek | it would take 0.2 seconds for a group of people to recreate the intro to that show with asterisk |
| 14:49.11 | Faustov | I have to admit video confs are in HIGH demand ;) |
| 14:49.27 | Naikrovek | indeed |
| 14:49.32 | wcselby | yeah that's what they said during the "future of asterisk" talk at astricon |
| 14:50.03 | wcselby | pabelanger- didn't you just get married? |
| 14:50.06 | Faustov | is there a serious blocker? |
| 14:50.14 | Naikrovek | doing a video wall would not be hard, I don't think. I don't know why it would take so long. surely there's a good reason |
| 14:50.19 | wcselby | pabelanger- congrats and all, but go enjoy yourself! :) |
| 14:50.20 | Faustov | or are there simply more important things to do? |
| 14:50.33 | Naikrovek | that would be it, I think |
| 14:50.36 | Naikrovek | other priorities |
| 14:50.46 | Naikrovek | VLC does what they need, and it's open source |
| 14:50.56 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 14:51.04 | Naikrovek | so there must be other priorities, or VLC doesn't really translate to Asterisk's architecture |
| 14:51.09 | wcselby | Naikrovek- they mentioned at astricon that it would put an exponential load on the server to do the mixing of a brady bunch-style video call |
| 14:51.16 | jeffspeff | [TK]D-Fender, thanks for the recommendation of upgraded. Just doing a keyword search through the recent change-log shows that they have fixed a LOT of memory leaks since 1.8.5 |
| 14:51.23 | Naikrovek | wcselby: that's what GPUs are for. |
| 14:51.33 | Naikrovek | they're REALLY good at that kind of thing |
| 14:51.35 | Faustov | exactly |
| 14:51.38 | Faustov | plus it is optional |
| 14:51.59 | Faustov | most people got machines more than capable of running asterisks |
| 14:52.13 | Faustov | for those who want video confs, higher spec hardware is available |
| 14:53.05 | FinboySlick | Naikrovek: Got experience with an equivalent SNOM setup? |
| 14:53.14 | leifmadsen | it's not just the processing requirements, it's all the coding involved in making that happen, or finding a library that is license compatible |
| 14:53.16 | Naikrovek | FinboySlick: I don't. |
| 14:53.30 | Naikrovek | leifmadsen: yeah there's a good reason, i'm sure |
| 14:53.38 | leifmadsen | it's not going to be trivial |
| 14:53.53 | leifmadsen | so the priority is really quite low |
| 14:55.42 | *** join/#asterisk cerberus_za (~coert@8ta-151-134-105.telkomadsl.co.za) |
| 15:01.47 | leifmadsen | btw: I did update the downloads page on asterisk.org to point to beta2 earlier today |
| 15:02.01 | *** join/#asterisk francisvgarcia (~francis.g@190.6.137.113) |
| 15:02.36 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 15:02.39 | [TK]D-Fender | leifmadsen, website admin is on Steve Sokol's list, right? |
| 15:03.07 | leifmadsen | not that page |
| 15:03.24 | leifmadsen | and not Steve Sokol directly -- you need to use webmaster@digium.com |
| 15:03.36 | leifmadsen | there is a web dev team that handles all that stuff |
| 15:04.48 | McBoingBoing | [TK]D-Fender: network graphs do not exceed 60KB/s, and still experience slight hiccup when the "reregistration" takes place |
| 15:06.04 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:07.49 | wcselby | hmmmmm |
| 15:07.52 | wcselby | can't login to one of my servers |
| 15:08.00 | *** join/#asterisk epaphus (~epaphus@200.122.149.9) |
| 15:08.12 | wcselby | that's always fun |
| 15:08.14 | wcselby | afk |
| 15:09.35 | epaphus | Hello. So does putting an asterisk server behind a NAT cause issues for SIP clients who want to connect remotely which are also under a NAT? |
| 15:11.11 | [TK]D-Fender | epNot if configured properly |
| 15:11.14 | [TK]D-Fender | ~sipnat |
| 15:11.14 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 15:19.07 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 15:20.55 | *** join/#asterisk Twitchnln (~Adium@adsl-184-36-49-49.asm.bellsouth.net) |
| 15:21.02 | Twitchnln | morning |
| 15:23.52 | p3nguin | It also depends on your routers between the phones and asterisk. |
| 15:24.21 | p3nguin | Some routers just do not play well. |
| 15:25.19 | p3nguin | My Cisco SOHO router, for example. There was nothing I could do to get RTP to work correctly through the NAT. |
| 15:26.10 | coppice | some SOHO routers play even worse, compromise your firewall, and cost you lots of money |
| 15:26.26 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:29.57 | wcselby | was this weekend the old fall back time? |
| 15:30.04 | wcselby | for daylight savings? |
| 15:30.11 | p3nguin | Should be next weekend. |
| 15:30.25 | wcselby | i know this year it's next weekend |
| 15:30.29 | wcselby | but there was a change a few years back |
| 15:30.37 | wcselby | it used to be different dates for fallback |
| 15:30.41 | p3nguin | Oh, yeah, it was this weekend. |
| 15:30.47 | wcselby | huh |
| 15:30.51 | p3nguin | It changed six years ago. |
| 15:30.57 | wcselby | now I have to figure out how to tell this linux box to get with the times |
| 15:31.11 | p3nguin | You need to update tzdata. |
| 15:31.16 | p3nguin | What distro? |
| 15:31.53 | epaphus | p3nguin, ok but in that example of your SOHO cisco.. if the server would have had a public IP even though you are under a NAT it should play nicely right |
| 15:32.16 | p3nguin | That statement doesn't make sense to me. |
| 15:32.31 | wcselby | centos 5.5 |
| 15:32.42 | p3nguin | If you are behind NAT, you don't have public IP addressing on your server behind the NAT. |
| 15:33.11 | p3nguin | yum -y update tzdata |
| 15:33.15 | p3nguin | (I think) |
| 15:36.14 | p3nguin | And after that package is updated, you may need to copy your new tz file into place. For me, I'd use cp /usr/share/zoneinfo/America/Chicago /etc/localtime |
| 15:36.44 | p3nguin | I don't know if CentOS copes it on startup or not. I'd guess not. |
| 15:37.21 | *** join/#asterisk catphish (~catphish@2001:9d8:2005:11:222:15ff:fe88:aae2) |
| 15:37.48 | p3nguin | And that's all there is to it. |
| 15:38.32 | catphish | can anyone shed any light on this: ERROR[30143] astobj2.c: refcount -1 on object 0x19173d8 |
| 15:38.46 | catphish | unfortunately it's not reproducible |
| 15:39.56 | francisvgarcia | Hi P3nguin |
| 15:40.12 | francisvgarcia | I tell you that retest yesterday |
| 15:40.37 | francisvgarcia | and the line is only ringing once before asterisk takes the call |
| 15:40.52 | francisvgarcia | not 3 times like before |
| 15:41.13 | p3nguin | But if you hook a phone to the wall jack, it rings zero times before you hear the phone? |
| 15:41.21 | *** join/#asterisk hudony (~chatzilla@modemcable078.217-70-69.static.videotron.ca) |
| 15:41.23 | francisvgarcia | yes |
| 15:42.19 | francisvgarcia | but now it's better |
| 15:42.26 | francisvgarcia | before were 3 times |
| 15:42.27 | francisvgarcia | not 1 |
| 15:42.29 | francisvgarcia | now 1 |
| 15:42.43 | francisvgarcia | I'll be calling the Digium Support |
| 15:42.47 | [TK]D-Fender | francisvgarcia, * also needs to hear the ring fininsh before acknowledging that it was indeed a ring in the first place |
| 15:42.58 | [TK]D-Fender | francisvgarcia, You'll never get "realtime" |
| 15:43.15 | [TK]D-Fender | francisvgarcia, If you got it to one.. that's it. Nothing more to do. |
| 15:43.20 | p3nguin | That's a feature of the TDM card? |
| 15:43.30 | hudony | Hi there : really simple question but somehow, cant find the answer on google : plan to use asterisk to manage our 16 phones here in the office I work in. Plan to buy 1 trunk and 12 channels (12 is enough according to people here). From what I understand : 12 people will be able to talk at the same time with the outside world but how many can call the office at the same time from the... |
| 15:43.31 | francisvgarcia | maybe a feature |
| 15:43.31 | hudony | ...outside workl? |
| 15:43.38 | hudony | How my question my sense... |
| 15:43.50 | p3nguin | With SIP or IAX2 calls, Asterisk can answer immediately without any ringing to the caller. |
| 15:44.21 | francisvgarcia | yes, but for now I have the limitant |
| 15:44.29 | francisvgarcia | of TDM circuit |
| 15:44.34 | [TK]D-Fender | hudony, 12 calls with the outside world. |
| 15:44.44 | p3nguin | hudony: You need to know how many channels you get for inbound calls (for your DIDs). If it's 12 channels, you can have 12 calls coming in at one time. |
| 15:45.02 | hudony | Oh...yes... we have 1 did but 12 channels |
| 15:45.11 | p3nguin | hudony: You can have 12 calls at one time. |
| 15:45.17 | [TK]D-Fender | hu12 channels = 12 call to the outside. |
| 15:45.23 | p3nguin | from |
| 15:45.24 | hudony | ok..no matter which way it was initiated |
| 15:45.30 | p3nguin | to = termination |
| 15:45.42 | [TK]D-Fender | hudony, correct. |
| 15:45.47 | hudony | Thank you all |
| 15:45.49 | p3nguin | They may give you unlimited channels for outbound. |
| 15:45.54 | p3nguin | Mine does. |
| 15:45.57 | hudony | oh |
| 15:45.59 | *** join/#asterisk tapout (~gamer@unaffiliated/tapout) |
| 15:46.05 | hudony | I'll have a look at it....good day all |
| 15:46.11 | p3nguin | I can call as many times as my bandwidth will allow. |
| 15:46.12 | [TK]D-Fender | hudony, Go find out what yuo paid for |
| 15:46.51 | tapout | Hello all... hey p3nguin, was it you that I got the fax help from? You sent me the tiff2pdf lines and stuff that extensions.conf gets |
| 15:47.04 | p3nguin | probably |
| 15:47.23 | francisvgarcia | hundony: don't foget to edit the file /etc/asterisk/asterisk.conf and to set the maxcall option to the number desired by you maxcalls = 24 |
| 15:47.55 | tapout | i'm pretty sure it was you... I'm using your fax-in-new setup, and you have .. h-SUCCESS and h-APPERROR... |
| 15:47.58 | *** join/#asterisk TimeRider (~steve@92.41.234.217.threembb.co.uk) |
| 15:48.00 | [TK]D-Fender | francisvgarcia, Why on earth would you want to do that? |
| 15:48.12 | tapout | the goto line says: exten => h,1,Goto(h-${SYSTEMSTATUS},1); .... |
| 15:48.20 | p3nguin | timerider: http://pastebin.com/Piqv4Egj 90-111 |
| 15:48.21 | tapout | it's calling "h-" instead of h-UNKNOWN |
| 15:48.32 | p3nguin | timerider: sorry |
| 15:48.43 | p3nguin | tapout: http://pastebin.com/Piqv4Egj 90-111 |
| 15:48.54 | p3nguin | Use this new method documented here. |
| 15:49.01 | p3nguin | I think you were using a "beta" design. |
| 15:49.47 | tapout | ahh ok, thank you! i will try that |
| 15:50.10 | tapout | your business-inbound is off the hook |
| 15:51.13 | francisvgarcia | [TK]D-Fender: It's just a suggestion |
| 15:51.24 | *** join/#asterisk troyt (~troyt@PariahZero.broker.freenet6.net) |
| 15:51.53 | [TK]D-Fender | francisvgarcia, Anything can be a suggestion. My question is "why this"? What possible benifit does he have artificailally limiting himself via asterisk.conf? |
| 15:52.03 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-135-115.chyn.qwest.net) |
| 15:52.27 | p3nguin | tapout: That part is very similar to what I use on my production system. |
| 15:52.49 | tapout | well it's awesome, i'm gonna steal some of it :) |
| 15:52.58 | p3nguin | That's why I put it in the pastebin. |
| 15:53.07 | tapout | when ReceiveFAX is done, it calls extension 'h' ? |
| 15:53.19 | p3nguin | When any call hangs up, it goes to extension h. |
| 15:53.21 | tapout | i notice you do... fax,* .. and then h,* ... it must be coming from receivefax eh? |
| 15:53.24 | tapout | ohh |
| 15:53.30 | francisvgarcia | Because of the CPU transcoding because maybe he'll be using g729 codec for the sip channel |
| 15:53.31 | tapout | OHH, i see |
| 15:53.58 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 15:54.08 | wcselby | p3nguin- thanks that worked (the tzdata stuff, plus copying the new tz file) |
| 15:54.27 | tapout | p3nguin, i'm probably brain dead in your eyes, but i see.. ${DB(fax/fax-manager/email)} ... lol where do you set that? |
| 15:54.32 | p3nguin | Okay, good. That would annoy me to have my clock wrong. |
| 15:55.06 | catphish | does "ERROR[30143] astobj2.c: refcount -1 on object 0x19173d8" mean anything useful or am i going to have to get a better stack trace somehow |
| 15:55.11 | p3nguin | tapout: I enter it manually in the AstDB. You can either enter an email address in that family/key in the DB, or just replace that variable with your real email address. |
| 15:55.32 | tapout | how do you enter it? |
| 15:55.35 | [TK]D-Fender | francisvgarcia, First what CPU can't hand transcoding that many calls? next... you'd cripple the ability to place call sat all? Who said he'd even be transcoding at all? He didn't even say it was using SIP in the first place. |
| 15:55.50 | p3nguin | database put fax fax-manager/email rob@mydomain.com |
| 15:56.14 | tapout | oh so clean!!! |
| 15:56.17 | tapout | very nice |
| 15:56.48 | p3nguin | To change to a different email address, the operation is the same -- it will overwrite. |
| 15:56.54 | tapout | going to my dropbox :) this is being saved |
| 15:57.06 | p3nguin | And you don't have to edit the dial plan when you do it that way. |
| 15:57.34 | p3nguin | I have all kinds of crap in my AstDB to keep from having to edit dial plan. |
| 15:58.04 | catphish | p3nguin: not worth using realtime for that? |
| 15:58.26 | p3nguin | I use an embedded system. |
| 15:58.37 | p3nguin | Realtime would be the death of me. |
| 15:58.41 | catphish | ah ok |
| 15:59.20 | p3nguin | It's bad enough that I put CDR/CEL into postgresql. :/ |
| 15:59.49 | catphish | i have the luxury of a dual 6-core xeon with 24GB of RAM for my main pbx |
| 16:00.16 | catphish | mysql is happy on there |
| 16:00.28 | tapout | i really appreciate this p3nguin. |
| 16:04.29 | tapout | p3nguin, are you using the ReceiveFAX that came with asterisk or the digium ? |
| 16:05.24 | tapout | http://pastebin.com/Pf8Q0Ka8 |
| 16:05.52 | tapout | p3nguin, I keep getting UNKNOWN instead of FAILED |
| 16:06.09 | tapout | of course i am not even trying to send a fax, just calling it and letting it error out |
| 16:06.25 | p3nguin | tapout: I use app_fax that comes in asterisk 1.8.7.1. |
| 16:07.08 | tapout | <PROTECTED> |
| 16:07.36 | tapout | do you think I have to upgrade to get FAILED from ReceiveFAX? |
| 16:07.40 | *** join/#asterisk jacobkiers (~jacobkier@82-168-168-85.ip.telfort.nl) |
| 16:08.13 | tapout | i am using app_fax.so |
| 16:08.15 | p3nguin | I wouldn't think so. |
| 16:08.32 | tapout | p3nguin, if you call your fax line and sit there, does it give you unknown? |
| 16:08.38 | tapout | or does it go to failed? |
| 16:08.44 | p3nguin | Although when I used 1.4, I don't think I had the FAXOPT() thing just like it complains about in yours. |
| 16:08.47 | *** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net) |
| 16:09.03 | p3nguin | I'll check it; one moment. |
| 16:09.59 | p3nguin | <PROTECTED> |
| 16:10.20 | *** join/#asterisk r0m|u (~are@darkstar.rice.edu) |
| 16:10.22 | p3nguin | And I got an email that says a fax to me failed. |
| 16:10.46 | r0m|u | waz up guys |
| 16:10.57 | tapout | is there an easy way on debian to get the latest asterisk withotu having to compile/play around? |
| 16:11.12 | p3nguin | Check the repo. |
| 16:11.19 | p3nguin | asterisk/digium repo, that is. |
| 16:12.09 | r0m|u | p3nguin, do you remember the CID that displayed on your side during the test calls yesterday? |
| 16:12.26 | p3nguin | I don't remember it, but I could look for it. |
| 16:12.41 | p3nguin | I'd just have to scroll up a bit. |
| 16:12.44 | r0m|u | If you can please |
| 16:12.47 | r0m|u | Thanks |
| 16:13.40 | p3nguin | I see that it is an invalid NANP caller ID number. :) |
| 16:14.25 | r0m|u | is it? |
| 16:14.33 | *** join/#asterisk ChkDigit (~mike@static24-72-71-175.r.rev.accesscomm.ca) |
| 16:14.36 | p3nguin | Yes, it's 11 digits. |
| 16:14.55 | p3nguin | NANP is only 10. |
| 16:15.15 | r0m|u | can you msg me the number that came out in your cid please. |
| 16:15.16 | p3nguin | But anyway, I see the number. |
| 16:15.18 | p3nguin | Yes. |
| 16:15.34 | r0m|u | thanks |
| 16:15.49 | r0m|u | Got it thanks. |
| 16:15.56 | r0m|u | another fuck up from CC. |
| 16:16.02 | p3nguin | heh |
| 16:16.06 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 16:16.10 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:16.20 | p3nguin | Are you setting any caller id number on calls going out through them? |
| 16:16.25 | r0m|u | I have my own CID setup in asterisk but CC does not allow you to overwrite it |
| 16:16.47 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 16:16.51 | r0m|u | I have no way to turn of there cid. |
| 16:16.58 | p3nguin | They require you to prove to them that you have "control" over the number in order to use it as CID. |
| 16:17.22 | r0m|u | is that right? |
| 16:18.06 | p3nguin | Such as call their support line from the number, then later open a ticket with them to tell them the exact time/date that you called from the number, and then tell them you want to use it as CID. They will add it to your list of allowed numbers. |
| 16:18.13 | r0m|u | how can I do that? |
| 16:18.27 | r0m|u | ah I see |
| 16:19.26 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 16:20.34 | tapout | yay p3nguin, it's doing 1?FAILED now.. I just have to figure out why it's not letting me do the /bin/echo |
| 16:20.58 | *** join/#asterisk [Outcast] (~anonymous@pool-96-252-45-211.bstnma.fios.verizon.net) |
| 16:21.05 | tapout | system_exec_helper:unable to execute .... that line you gave me. I'm wondering if it's .. sudo -u asterisk? |
| 16:21.39 | p3nguin | It's probably not the echo that is causing the problem. |
| 16:21.48 | tapout | ahh, missing "sudo" |
| 16:21.50 | tapout | trying again :) |
| 16:22.09 | tapout | why did you do.. sudo -u asterisk ... doesn't it inherit 'asterisk' user anyways? |
| 16:22.12 | p3nguin | If you need sudo to make it work, you'll have to also configure sudoers. |
| 16:22.25 | tapout | can you show me the sudoers? |
| 16:22.55 | p3nguin | In 1.4, asterisk was running as asterisk, but it still would not execute mutt as asterisk because of System(). |
| 16:23.05 | p3nguin | asterisk ALL=(ALL) NOPASSWD: /usr/bin/mutt |
| 16:23.16 | r0m|u | visudo |
| 16:23.38 | tapout | r0m|u, beautiful :) |
| 16:23.45 | tapout | p3nguin, amazing as well |
| 16:23.58 | tapout | thanks!! i put that in there.. gonna restart asterisk (probably not needed) and retry... i'm so CLOSE |
| 16:23.59 | p3nguin | You can try without the sudo part. |
| 16:24.15 | p3nguin | If it works without, no reason to keep it. |
| 16:24.53 | p3nguin | To be honest, I haven't tested without sudo in 1.8. |
| 16:25.02 | p3nguin | I should do that. |
| 16:26.28 | tapout | OH MY GOD, got the email! |
| 16:26.28 | tapout | yay |
| 16:26.39 | tapout | woot, now to actually test the fax receiving :) |
| 16:26.48 | p3nguin | I got the email without using sudo. |
| 16:27.02 | p3nguin | I guess that was a 1.4 thing. |
| 16:27.09 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-181-48.netvisao.pt) |
| 16:28.45 | r0m|u | p3nguin, you hang around often on that conf from dijib? |
| 16:28.58 | p3nguin | No, yesterday was the first time it was online. |
| 16:29.07 | r0m|u | ah I see |
| 16:29.33 | tapout | hey p3nguin, how do you control the 'from' email and all that from mutt? did you setup a mutt.conf or something? i'm getting... from asterisk@localhost |
| 16:29.35 | p3nguin | You know how asterisk virgins are -- always playing with the toys. |
| 16:29.46 | r0m|u | rofl! |
| 16:29.49 | r0m|u | so tur |
| 16:29.50 | r0m|u | lol |
| 16:29.53 | r0m|u | true* |
| 16:30.12 | tapout | mine worked without sudo as well |
| 16:30.35 | r0m|u | I want to test sipuri dialing but I dont know of one except for dijib's |
| 16:30.47 | r0m|u | once I get home* |
| 16:30.51 | p3nguin | tapout: I relay via gmail, so it comes from the my asterisk's gmail address. The name is set in /etc/passwd, though. |
| 16:31.30 | p3nguin | Asterisk daemon <asterisk@gmail.com> |
| 16:32.03 | r0m|u | fuck you scored that address? |
| 16:32.34 | tapout | mutt will use the email address from /etc/passwd? hrmm.. |
| 16:32.42 | p3nguin | No, I said the name. |
| 16:32.51 | p3nguin | "Asterisk daemon" |
| 16:32.55 | tapout | oh just the name |
| 16:33.02 | p3nguin | The address is that of the account I am using. |
| 16:33.16 | p3nguin | What MTA are you using? |
| 16:33.23 | tapout | postfix |
| 16:33.35 | tapout | i didn't even set it up tho tbh |
| 16:33.35 | p3nguin | Okay, so you can set any email address you want... |
| 16:33.39 | r0m|u | postfix rules |
| 16:33.43 | r0m|u | main.mc |
| 16:33.51 | tapout | on this box, i have it setup on my other one |
| 16:33.53 | p3nguin | mc? |
| 16:33.58 | p3nguin | It's postfix, not sendmail. |
| 16:34.04 | r0m|u | rofl |
| 16:34.05 | r0m|u | lol |
| 16:34.20 | tapout | do you get warnings of T.30 ecm carrier not found when receiving a fax p3nguin ? |
| 16:34.21 | r0m|u | .cf* |
| 16:34.24 | r0m|u | my bad :P |
| 16:34.39 | p3nguin | I guess you can set the address either in .muttrc or maybe in the mutt command. |
| 16:34.48 | tapout | WARNING[24923]: res_fax_spandsp.c:368 spandsp_log: WARNING T.30 ECM carrier not found |
| 16:34.56 | r0m|u | I was a sendmail junkie back in the days... Thank God for post fix! :) |
| 16:35.00 | r0m|u | tapout, Thats normal |
| 16:35.05 | tapout | sweet |
| 16:35.12 | p3nguin | tapout: I've never seen it. |
| 16:35.30 | tapout | that's me doing; asterisk -rvvvvv |
| 16:35.34 | r0m|u | p3nguin, I get that error on my incoming faxes |
| 16:35.38 | *** join/#asterisk dtascom (~david@98-24-18-72.static.tierzero.net) |
| 16:35.45 | p3nguin | I don't. |
| 16:35.49 | tapout | LOL i love you p3nguin |
| 16:35.50 | tapout | serious |
| 16:35.54 | p3nguin | Or at least never have before. |
| 16:35.56 | tapout | I GOT FAXES working!!! |
| 16:35.59 | r0m|u | I found that is norml if you have it set to look fot T.30 |
| 16:36.12 | p3nguin | I recently turned on T.38 support, though. |
| 16:36.24 | tapout | p3nguin, is that something I should do? |
| 16:36.28 | r0m|u | :) |
| 16:36.33 | p3nguin | But I haven't had a single fax since I turned on T.38. |
| 16:36.45 | r0m|u | tapout, T.38 is for error correction |
| 16:36.51 | tapout | http://www.interpage.net/sub-wwwfax.html |
| 16:37.08 | tapout | how do I see if I have t.38 enabled ? |
| 16:37.27 | r0m|u | it can be tuned in res_fax.conf |
| 16:37.29 | tapout | p3nguin, that interpage is where i used to send myself a fax |
| 16:38.39 | tapout | my res_fax.conf only talks about statusevents/modems/and t.30 ecm.. i'll google :) |
| 16:39.40 | r0m|u | p3nguin, do recive faxes threw a fax machine or using asterisk to email? |
| 16:39.45 | r0m|u | you* |
| 16:39.50 | p3nguin | ... T.30 is error correction. |
| 16:40.17 | p3nguin | Fax over SIP to asterisk to email. |
| 16:40.55 | r0m|u | ops thats right :) had them backwords :P |
| 16:41.07 | r0m|u | backwards* |
| 16:41.13 | p3nguin | I guess I need to turn off T.38 again. |
| 16:41.14 | tapout | p3nguin, how did you enable t.38? can you show me pls? |
| 16:41.35 | tapout | why are you disabling t.38? |
| 16:41.37 | *** join/#asterisk benlangfeld (~ben@unaffiliated/benlangfeld) |
| 16:41.43 | p3nguin | [Oct 31 11:40:39] WARNING[27068]: res_fax.c:1508 receivefax_t38_init: channel 'SIP/sipgate-00000026' refused to negotiate T.38 |
| 16:41.46 | p3nguin | [Oct 31 11:40:39] WARNING[27068]: res_fax.c:1529 receivefax_t38_init: Audio FAX not allowed on channel 'SIP/sipgate-00000026' and T.38 negotiation failed; aborting. |
| 16:41.49 | p3nguin | [Oct 31 11:40:39] ERROR[27068]: res_fax.c:1734 receivefax_exec: error initializing channel 'SIP/sipgate-00000026' in T.38 mode |
| 16:41.50 | r0m|u | p3nguin, have you tested it? Last I play with T.38 broke faxing for me |
| 16:42.00 | r0m|u | AH! |
| 16:42.05 | r0m|u | there you go :P |
| 16:42.12 | p3nguin | turning it back off. |
| 16:42.19 | tapout | man this is amazing |
| 16:42.22 | tapout | amazing |
| 16:42.23 | tapout | a |
| 16:42.24 | tapout | mazing |
| 16:42.53 | tapout | so happy I found asterisk, had great help from you guys and got it rocking... so damn amazing |
| 16:43.00 | r0m|u | nothing better than the feeling of getting your dial plan to do what ever you wanted to do for the first time... :) |
| 16:44.05 | tapout | lol |
| 16:44.45 | p3nguin | I wish someone would cut quintana's cord so he'd stop doing that. |
| 16:44.46 | *** part/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-228-190.w86-204.abo.wanadoo.fr) |
| 16:44.56 | p3nguin | It's really annoying to see him changing his nick all fucking day long. |
| 16:45.02 | p3nguin | back and forth, back and forth |
| 16:45.40 | *** join/#asterisk BillyFred (~smithbd@128.187.234.225) |
| 16:47.16 | francisvgarcia | I have a question for u guys if some of you had this kind of issue |
| 16:47.50 | francisvgarcia | does anyone has an issue with the PC port of the grandstream GXP1450 which suddenly stop working |
| 16:48.07 | francisvgarcia | and don't make the bridging anymore |
| 16:48.49 | *** join/#asterisk xpot-mobile (~james@dhcp69.emcb.utah.edu) |
| 16:49.07 | r0m|u | lol @ p3nguin |
| 16:50.34 | r0m|u | francisvgarcia, I have no clue. I dont use grandstream. contact there tech support maybe? |
| 16:50.50 | hardwire | p3nguin: to be fair.. the city he is in has gone back and forth between france and germany a lot. |
| 16:50.53 | hardwire | it's just in his nature. |
| 16:51.09 | r0m|u | lol |
| 16:51.34 | catphish | what's the simplest SIP ping i can use to test asterisk's aliveness? |
| 16:51.47 | catphish | i'm guessing an options request |
| 16:51.51 | catphish | but not sure what url to use |
| 16:51.52 | hardwire | option |
| 16:52.00 | hardwire | same |
| 16:52.07 | hardwire | you just issue an option vs anything else. |
| 16:52.36 | hardwire | options can change based on the destination. So throwing the destination onto the URL (even a test one) is good practice. |
| 16:53.06 | catphish | right now all i can get back are 404s |
| 16:53.15 | catphish | not sure what URL i should be sending |
| 16:53.25 | hardwire | using sipsak? |
| 16:53.28 | catphish | i'm just sending sip:[incoming number]@ip |
| 16:53.46 | catphish | no, just a perl script that generates an options request |
| 16:53.52 | hardwire | ah |
| 16:54.01 | hardwire | maybe use a program that works well first.. do some packet captures |
| 16:54.05 | hardwire | and cross reference. |
| 16:54.21 | catphish | but i don't actually know what url to use :( |
| 16:54.27 | catphish | even if the program worked |
| 16:54.30 | [TK]D-Fender | catphish, All * ever sends back to OPTIOSN is 404. This is NORMAL |
| 16:54.42 | catphish | oh ok |
| 16:54.48 | p3nguin | You measure the time it takes between sending and receiving the 404. |
| 16:54.49 | [TK]D-Fender | catphish, And proves a successful test |
| 16:54.57 | catphish | i'll use that then :) |
| 16:55.01 | p3nguin | If it sends a 404, you know asterisk got it. |
| 16:55.05 | catphish | yeah |
| 16:55.10 | catphish | i just need to check it hasnt crashed |
| 16:55.12 | p3nguin | What more do you need? |
| 16:55.29 | catphish | nothing |
| 16:55.32 | catphish | i'm good with a 404 |
| 16:55.38 | catphish | i just felt i was doing something wrong |
| 16:56.01 | hardwire | catphish: I'm going to laugh if your probe program crashes chan_sip. |
| 16:56.08 | hardwire | that would be exactly what you don't want. : |
| 16:56.10 | hardwire | ) |
| 16:56.24 | catphish | hardwire: why would that happen? |
| 16:56.30 | hardwire | with chan_sip? |
| 16:56.32 | catphish | if it does i'll raise a very urgent bug report :) |
| 16:56.35 | hardwire | phase of the moon during compiling. |
| 16:57.45 | hardwire | should put in a bug report that moon phases be taken into consideration while releasing binaries. |
| 16:57.48 | catphish | my asterisk crashed today with no explanation |
| 16:57.56 | catphish | so im a bit concerned and adding some monitoring |
| 16:58.02 | catphish | as well as throwing it a -g |
| 17:01.35 | Katty | grooves |
| 17:09.48 | Katty | why is it so quiet |
| 17:11.16 | navaismo | monday |
| 17:11.40 | Katty | :< |
| 17:11.44 | Katty | that's no excuse! |
| 17:11.51 | Katty | monday is the new thursday, donchaknow |
| 17:12.05 | eppigy | i am just wilin out |
| 17:13.22 | *** join/#asterisk tonsofpcs (~tonsofpcs@cpe-69-205-240-64.stny.res.rr.com) |
| 17:14.54 | Katty | ALL OF THE PCS |
| 17:15.42 | Faustov | c u 'n thursday ;( |
| 17:17.24 | Katty | hai Faustov! |
| 17:17.50 | *** join/#asterisk cusco (~tralala@88.157.128.26) |
| 17:17.53 | Faustov | hai Katty ;) |
| 17:17.54 | cusco | hi folks |
| 17:18.01 | Katty | how're you dear |
| 17:18.03 | Katty | hugs cusco |
| 17:18.08 | cusco | :)) |
| 17:18.13 | cusco | someone is happy today! |
| 17:18.21 | Faustov | that's definitely not me |
| 17:18.27 | cusco | heh, troubles? |
| 17:18.31 | Faustov | I forgot about the 1h sleepover |
| 17:18.37 | Faustov | got to work 1h early |
| 17:18.42 | cusco | haha! you went to work early |
| 17:18.42 | Faustov | could it be worse? |
| 17:18.58 | Faustov | not that I mind going to work |
| 17:19.03 | Faustov | but sleeeep... |
| 17:19.28 | Katty | nomnom sleep nomnom |
| 17:19.35 | cusco | next time you can compensate by forgetting to wake up earlier in summer :) |
| 17:19.39 | Katty | i am always in a good mood! |
| 17:19.46 | cusco | thats excelent! |
| 17:19.47 | Katty | and i had 4 hours of sleep last night |
| 17:19.57 | cusco | thats not so great... |
| 17:20.01 | Katty | so i will likely DIE in about..ehn...2 hours hehe |
| 17:20.10 | cusco | ew.. |
| 17:20.17 | Katty | oh well ;) |
| 17:20.30 | cusco | I have a probably common question... |
| 17:20.57 | Faustov | Katty: kids? |
| 17:21.02 | cusco | list of pstn numbers, asterisk with PRI, would like to dial and check wich numbers are good to re-use (the ones that rang) |
| 17:21.08 | Katty | pbbffftttt kids. |
| 17:21.15 | Katty | no. |
| 17:21.27 | cusco | thing is dialstatus will most likelly be NOANSWER all the time |
| 17:21.36 | Faustov | hands Katty a tissue to wipe the screen from coffee |
| 17:21.44 | Katty | lol!!! |
| 17:21.46 | WIMPy | cusco: HANGUPCAUSE |
| 17:23.15 | cusco | WIMPy: hangup cause 0 |
| 17:23.46 | WIMPy | cusco: That means you hung up. |
| 17:23.56 | WIMPy | i.e. Dial() timed out. |
| 17:23.59 | cusco | but I dunno if the far end rang |
| 17:24.05 | cusco | yex... |
| 17:24.07 | cusco | yes... |
| 17:24.34 | WIMPy | If you get any other respones, HANGUPCAUSE will be >0. |
| 17:24.57 | cusco | even if I hangup after someone answered? |
| 17:25.27 | WIMPy | Or you listen on AMI. IIRC you get a channelstatus event when you receive an alerting message. |
| 17:25.32 | WIMPy | Yes. |
| 17:27.13 | cusco | hmm... ami |
| 17:27.49 | WIMPy | Or use soemthing else for scanning. |
| 17:30.04 | *** join/#asterisk becca_r (~becca_r@72.165.148.230) |
| 17:30.32 | WIMPy | You could also try to scan with different BC if you don;t want to upset ppl and get in to trouble for that. |
| 17:34.45 | *** join/#asterisk mpe (~mpe@31.25.23.177) |
| 17:35.10 | cusco | WIMPy: BC ? |
| 17:36.56 | WIMPy | Bearer Capability. |
| 17:37.34 | WIMPy | Make data calls and phones will ignore your call. |
| 17:43.43 | cusco | really? how? |
| 17:44.33 | cusco | like sending a fax? |
| 17:44.42 | catphish | of course you really shouldn't be calling lists of people and hanging up when you get a call progress confirmation |
| 17:44.53 | catphish | but i guess that goes without saying |
| 17:45.17 | cusco | well that is a problem, today with ringing songs, they send a Progress instead of 'ringing' |
| 17:45.29 | catphish | why is that a problem? |
| 17:45.59 | cusco | well I want to hangup the call as soon as possible. and if it is progress'ing how do I know it is not voicemail? |
| 17:46.11 | cusco | I would rather not let them answer... |
| 17:46.21 | catphish | i'm afraid i'm going to speculate that what you're proposing will a) annoy people and b) cause the same people to be annoyed again later |
| 17:47.17 | cusco | well... |
| 17:47.28 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 17:47.30 | cusco | not sure what you want me to say |
| 17:47.38 | catphish | but you should be able to just run this over a sip connection and see what responses you get back, ringing or progressing should mean theres a good chance it's going to connect and you can end the call immediately |
| 17:48.02 | cusco | we're using PRI |
| 17:48.08 | cusco | dahdi |
| 17:48.10 | catphish | oh i see |
| 17:48.27 | wcselby | i need a new book to read |
| 17:48.27 | catphish | why not just wait until its time to spam / call the people |
| 17:48.30 | catphish | and do it then |
| 17:48.40 | catphish | simply erase the failed number at that point |
| 17:48.45 | catphish | and speak to the ones that connect |
| 17:49.07 | cusco | and I was even thinking about parsing the PRI DEBUG and search for some indication where the number would be valid befor ringing, lol |
| 17:49.10 | catphish | but WIMPy's idea seems sane too |
| 17:49.27 | cusco | that is the actual scenario, but there are too many wrong numbers |
| 17:49.32 | catphish | you can make an isdn data call and i guess it will refuse to route to a ptsn with varying errors |
| 17:49.54 | cusco | how do I change the bearer cap.? |
| 17:49.58 | catphish | but i've never used it myself |
| 18:04.08 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 18:04.20 | cusco | WIMPy: how do I change the BC? |
| 18:04.54 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279681716.dsl.bell.ca) |
| 18:05.14 | r0m|u | waz up dijib |
| 18:05.26 | dijib | 2663@asterisk.serveirc.com |
| 18:05.33 | dijib | add that to your dialplan and asl me |
| 18:05.39 | dijib | r0m|u, who are you anyways |
| 18:05.39 | dijib | ? |
| 18:05.47 | r0m|u | SeRi |
| 18:06.02 | r0m|u | I am at work |
| 18:06.12 | r0m|u | you going to keep that chann up? |
| 18:06.22 | r0m|u | I will call in once I get home |
| 18:08.28 | dijib | yeah its up |
| 18:08.35 | dijib | its got a limit of 6 users. |
| 18:08.44 | dijib | but invite whoever you want in |
| 18:08.49 | r0m|u | cool |
| 18:09.05 | dijib | its all good |
| 18:10.48 | WIMPy | cusco: CHANNEL(transfercapability) |
| 18:11.15 | WIMPy | And yes, you will get different errors, depending on what you're calling. |
| 18:12.33 | *** join/#asterisk lordvadr (~something@jose-tc.ctc.biz) |
| 18:12.35 | cusco | No application 'CHANNEL' for extension (status, 0909932485457, 1) |
| 18:12.49 | WIMPy | function |
| 18:14.05 | r0m|u | does dialing sipuris charge for minutes like dialing a regular land line? |
| 18:14.43 | *** join/#asterisk elemenopy (~elemenopy@64.194.139.236) |
| 18:16.12 | elemenopy | hi everyone, has anyone in here had experience using CEL and FreeTDS? |
| 18:16.33 | cusco | WIMPy: so should I set CHANNEL(audiowriteformat) ?? |
| 18:17.06 | cusco | im feeling lost, I'm not seeing how will I be able to do this lol |
| 18:17.29 | lordvadr | I have a stumper for everyone. We have a turn-key asterisk cluster (so can't upgrade) based on asterisk 1.4 serving customers. One customer is using a custom soft-phone solution based on x-lite, and has an interesting complaint. They say they hear no progress (ringing) indication on outbound calls. After investigation, the developer informed me that they always expect a 183 with in-band indication. No problem, progressinband=always... |
| 18:18.30 | WIMPy | cusco: That not what I said. |
| 18:18.43 | cusco | dooh sorry im tired I guess |
| 18:19.19 | lordvadr | No, on most calls from which we only get a 180 from our upstream carrier, the 183 gets sent to the client after the 180, but the early media audio usually only opens one-way, and in the wrong direction, and usually this doesn't resolve when the call is answered. Both my end and their client set the media stream as "sendrecv", AND they are sending the packets to me from port 0, indicating they aren't expecting a response. |
| 18:19.22 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
| 18:20.31 | lordvadr | It's my job to find out which end is dicking up, and blame either their developer, or our vendor, but I've never seen anything like this. It sounds to me like this is the way its supposed to work because I get nothing at all from the asterisk debug logs, as well as the traffic--they don't appear to expect anything, nor do I send them anything. Anybody have any ideas as to what's going on? |
| 18:20.38 | cusco | WIMPy: I tryed setting it to 0x18 (video, right?) |
| 18:20.48 | cusco | but PRI DEBUG shows: > Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability: Speech (0) |
| 18:21.32 | WIMPy | cusco: For example, yes. Just not audio. I'd try 'digital'. |
| 18:22.05 | WIMPy | You use the words. See 'core show function CHANNEL'. |
| 18:22.42 | WIMPy | So that's DIGITAL not digital. NFI if case matters there. |
| 18:23.12 | *** join/#asterisk GameGamer43 (~GameGamer@li119-60.members.linode.com) |
| 18:24.00 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 18:26.38 | cusco | WIMPy: digital worked lol... anyways I cannot see if the number is good to re-use |
| 18:26.41 | cusco | :/ |
| 18:26.57 | cusco | ${DIALSTATUS} comes empty |
| 18:27.39 | *** join/#asterisk Vince-0 (~AndChat@41-132-156-89.dsl.mweb.co.za) |
| 18:27.40 | cusco | hangup cause comes 0 |
| 18:27.56 | WIMPy | Use HANGUPCAUSE. |
| 18:28.01 | WIMPy | That should be ok then. |
| 18:28.30 | Katty | them |
| 18:28.41 | cusco | ow? |
| 18:28.45 | cusco | let me put my mobile off |
| 18:28.56 | WIMPy | 'incompatible destination' and 'no response from user' would mean the number was valid. |
| 18:29.13 | WIMPy | That won't make the number invalid. |
| 18:29.51 | cusco | aw... |
| 18:30.29 | WIMPy | If it doesn't go to VM, you probably get a 'destination out of order'. |
| 18:31.48 | cusco | right I tried another number and ggot hc 20 |
| 18:32.37 | WIMPy | 'subscriber absent' |
| 18:33.13 | WIMPy | But definitely a valid number. |
| 18:35.38 | cusco | I see. is there any way i can get that kind of info? subscriber absent, or incompatible destination, in asterisk?? |
| 18:36.18 | WIMPy | google for cause codes. |
| 18:36.53 | cusco | ow hangup cause ok |
| 18:37.20 | cusco | ok that is great really, but we still have some number that keep going to voicemail, how would I filter out those in a round2 |
| 18:37.52 | p3nguin | Is it okay to express NANP toll-free numbers in E.164 format? |
| 18:40.17 | *** join/#asterisk francisvgarcia (~francis.g@190.6.137.113) |
| 18:40.17 | WIMPy | Depends on the VM implemntation. usually you get a redirection information. |
| 18:40.44 | *** join/#asterisk devianTz (~deviant@166.206-62-69.ftth.swbr.surewest.net) |
| 18:42.05 | WIMPy | If you don't you can only guess by the timing. |
| 18:42.30 | WIMPy | But if you make data calls, you shouldn't be able to reach VM. |
| 18:42.39 | r0m|u | p3nguin, dialing sipuris charges my account? |
| 18:44.10 | WIMPy | r0m|u: If you send it ti an ITSP, theyr price list will apply. If you do it yourself, you only pay for IP traffic. |
| 18:44.15 | p3nguin | That would depend how you dial. |
| 18:45.11 | r0m|u | as a test here is a context |
| 18:45.17 | r0m|u | exten => 2000,1,Dial(SIP/2663@asterisk.serveirc.com) |
| 18:45.56 | p3nguin | That does not invole your ITSP. |
| 18:46.08 | p3nguin | It's a direct call from asterisk to asterisk.serveirc.com. |
| 18:46.24 | p3nguin | And that's not a context. |
| 18:46.28 | p3nguin | It is an extension. |
| 18:46.34 | p3nguin | Extension 2000. |
| 18:46.55 | r0m|u | cool! I will try it when I get home. yea I still mix that up. sorry. |
| 18:46.57 | p3nguin | Phones are not extensions, THAT is an extension. |
| 18:47.01 | *** join/#asterisk scubes13 (~scubes13@rrcs-70-60-211-241.midsouth.biz.rr.com) |
| 18:48.17 | r0m|u | got it. a context is what you declare to a peer as dialing rule correct? |
| 18:48.30 | *** join/#asterisk becca_r (~becca_r@adsl-99-21-18-162.dsl.ksc2mo.sbcglobal.net) |
| 18:48.37 | p3nguin | A context is a container of extensions. |
| 18:48.55 | r0m|u | ie : context=voipms-inbound |
| 18:49.02 | p3nguin | [voipms-inbound] is a context by the name of voipms-inbound. |
| 18:49.18 | p3nguin | And then you assign a context to a peer. |
| 18:49.42 | p3nguin | Extensions are "dialing rules." |
| 18:49.53 | r0m|u | got it. for some reason I tend to confuse that. I will make it clear now |
| 18:50.04 | FinboySlick | Anyone know of an ATA that will talk vlan on its wan port but let you filter it out of its lan port? |
| 18:50.27 | p3nguin | Most people think that phones are extensions. In asterisk, phones are phones, and extensions are dialing rules. |
| 18:51.38 | becca_r | yeppers, took me a while to make that "click" |
| 18:51.39 | r0m|u | I actually dont tend to confuse phones with ext. but I tend to call context to dialing rules. thats the confusion I have. |
| 18:51.54 | r0m|u | Is all clear now. |
| 18:53.14 | p3nguin | I guess people think that phones are called extensions due to some legacy terminology from old telephony models. Something about "extension phones," or whatever. Does that sound familiar? |
| 18:57.02 | WIMPy | Aren;t they still calld that in other products? |
| 19:01.06 | r0m|u | get a kick out of this one... I found this on a forum... The word "extension" can be a bit confusing when you are used to Asterisk. In Asterisk an extension is mainly used to register a SIP device. |
| 19:01.23 | p3nguin | pffft |
| 19:01.40 | p3nguin | Leave a comment to tell the person he is an idiot, and give him the link to the book. |
| 19:01.55 | r0m|u | needle's to say... he got burned.... :P |
| 19:02.06 | r0m|u | http://www.dslreports.com/forum/r26444199-Mega-Cheaper-Than-Dirt-Worldwide-Sale-Anveo-US-DID-0.99 |
| 19:02.14 | p3nguin | Point out the chapter on dialplan. |
| 19:02.21 | *** join/#asterisk darkdrgn2k3 (~darkdrgn@208.124.232.58) |
| 19:02.23 | darkdrgn2k3 | morning all |
| 19:02.28 | *** join/#asterisk zyphlar (~zyphlar@wsip-68-14-229-148.ph.ph.cox.net) |
| 19:02.31 | r0m|u | m |
| 19:02.35 | darkdrgn2k3 | is there a way to do modem DIALIN over a voip line |
| 19:02.40 | darkdrgn2k3 | (like FAX but for dialup) |
| 19:03.13 | WIMPy | darkdrgn2k3: Forget it. Try T.38 for fax. |
| 19:03.23 | darkdrgn2k3 | WIMPy: not looking for fax.. |
| 19:03.31 | darkdrgn2k3 | actualy that was my second question, how bad is it |
| 19:03.38 | darkdrgn2k3 | looking for Dialup # |
| 19:03.50 | [TK]D-Fender | darkdrgn2k3, No way carrier would survive |
| 19:03.58 | WIMPy | Almost no chance. |
| 19:04.01 | darkdrgn2k3 | hmm to bad :() |
| 19:04.22 | darkdrgn2k3 | well it was worth a shot |
| 19:04.48 | darkdrgn2k3 | was thjinking of implementing a backup dialin option for my ciscos |
| 19:04.50 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 19:05.20 | becca_r | well modem over voip line is possible depending on latency, jitter, delay. I have had it successfully implemented for alarm systems even though I strongly cautioned against it. |
| 19:05.20 | WIMPy | What kind of sense would that make? |
| 19:06.00 | darkdrgn2k3 | well looking at 2 options |
| 19:06.00 | [TK]D-Fender | becca_r, that isn't constant communication. That is a momentary burst |
| 19:06.12 | [TK]D-Fender | Even then you'd have to be very "lucky" |
| 19:06.18 | becca_r | agreed |
| 19:07.41 | darkdrgn2k3 | looks like i need to find a cheap dialup provider around here :-P |
| 19:07.59 | darkdrgn2k3 | ok lets take this back |
| 19:08.07 | darkdrgn2k3 | what are the chances of doing VOIP over DIALUP |
| 19:08.10 | darkdrgn2k3 | for like 3 users |
| 19:08.57 | [TK]D-Fender | 1 call. |
| 19:08.57 | becca_r | cringes at the thought of VoIP over dialup. |
| 19:09.01 | [TK]D-Fender | not "X users" |
| 19:09.05 | [TK]D-Fender | 1 call. |
| 19:09.11 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 19:09.13 | darkdrgn2k3 | Yhe... thast what i thought.. |
| 19:09.19 | *** join/#asterisk mpe (~mpe@31.25.23.177) |
| 19:09.22 | darkdrgn2k3 | im betting of looking at the GSM smart hubs as a backup |
| 19:09.58 | [TK]D-Fender | darkdrgn2k3, Cell Data <- |
| 19:10.00 | darkdrgn2k3 | i wonder how bad the jitter would be over a GSM or EDGE or whatever its called now network |
| 19:10.14 | darkdrgn2k3 | yeh... cell data |
| 19:10.25 | r0m|u | darkdrgn2k3, not bad at all. I do it |
| 19:10.36 | r0m|u | I have a portech mv-370 |
| 19:10.47 | *** join/#asterisk murdock_ut (~chatzilla@mail.kimballequipment.com) |
| 19:10.57 | darkdrgn2k3 | i just need to get bell to take its head out of its A _ _ and give me a demo model for a moneth.. telus will do it.. |
| 19:10.59 | darkdrgn2k3 | i hate bell |
| 19:11.09 | r0m|u | all my calls to tmobile go over it. SIP/GSM/CELL |
| 19:11.22 | WIMPy | CSD is without jitter. PSD can be interesting, but is usually ok. |
| 19:12.29 | darkdrgn2k3 | whats the diff between csd and psd |
| 19:12.47 | murdock_ut | Is there a way to find out what the valud of the "mailbox" setting is for a specific device in sip.conf via dial plan? |
| 19:12.48 | darkdrgn2k3 | cell swiched data vs packet swiched data right? |
| 19:13.02 | coppice | continuously screwedup data vs partially screwedup data |
| 19:13.05 | [TK]D-Fender | murdock_ut, "core show functions" <- |
| 19:13.10 | darkdrgn2k3 | LMAO |
| 19:13.20 | WIMPy | darkdrgn2k3: Circuit ... |
| 19:14.52 | murdock_ut | [TK]D-Fender: Thanks... found it. |
| 19:15.40 | *** join/#asterisk scubes13 (~scubes13@rrcs-70-60-211-241.midsouth.biz.rr.com) |
| 19:25.45 | francisvgarcia | Gurus |
| 19:25.56 | francisvgarcia | is there any click to dial application |
| 19:26.07 | becca_r | php code on your web page |
| 19:26.08 | francisvgarcia | open source for |
| 19:26.16 | becca_r | using the AMI |
| 19:26.18 | citywok | francisvgarcia: telnet to the AMI and tell it to originate a call |
| 19:26.54 | francisvgarcia | what about anveo |
| 19:28.01 | *** join/#asterisk jasonbassett (~jasonbass@cpc3-basl7-0-0-cust155.basl.cable.virginmedia.com) |
| 19:28.15 | *** join/#asterisk FinboySlick (~shark@74.117.40.10) |
| 19:28.18 | jasonbassett | Good evening folks |
| 19:28.25 | FinboySlick | Curse you nvidia :P |
| 19:28.38 | eppigy | hold your tongue blasphemer |
| 19:29.10 | jasonbassett | I have the following scenario and am trying to find the easiest (most manageable long term) solution, as follows: |
| 19:30.02 | McBoingBoing | troubleshooting voip call quality, is it best to use "sip debug" or will I get more info from tcpdump -> wireshark ? |
| 19:30.28 | citywok | call quality won't have anything in sip debug |
| 19:30.34 | WIMPy | McBoingBoing: Sip debug will not help you at all. |
| 19:30.39 | McBoingBoing | ok |
| 19:30.39 | *** join/#asterisk Ionic (ionic@ionic.de) |
| 19:30.41 | citywok | calls are RTP streams, SIP is just the conversation layer |
| 19:30.44 | [TK]D-Fender | McBoingBoing, SIP doesn't have "quality". RTp does |
| 19:31.02 | hardwire | kwality |
| 19:31.06 | McBoingBoing | :P |
| 19:31.32 | WIMPy | You can use rtp debug, but wireshark is probably a lot more helpful. |
| 19:31.35 | [TK]D-Fender | hardwire, That's the name of an Indian restarant at the other end of the mall from my favourite one :) |
| 19:31.56 | FinboySlick | [TK]D-Fender: They have to keep 'em separated? |
| 19:31.57 | jasonbassett | I call into my system using my DID number, I know to press a key sequence such as #9 which should then add the number I am calling from to the asterisk database and from then on, all calls coming into the system, should be forwarded via voip to that number. Calling in from another number and pressing #9 should change the divert to the new number, dialling #9 again from the already diverted to number, should canc |
| 19:31.57 | jasonbassett | the divert altogether. ? |
| 19:32.02 | jasonbassett | A bit of a mouthful |
| 19:32.38 | jasonbassett | I have been trying features.conf but the #9 is only read if the call has been answered, not when it is ringing |
| 19:32.49 | [TK]D-Fender | FinboySlick, http://rlv.zcache.com/b4i_screw_u_ru_over_18_qt_pi_postcard-p239988175163436951z8iat_400.jpg |
| 19:33.01 | McBoingBoing | WIMPy, still learning what I need to do but tcpdump data payload and analyzing with Wireshark is the way to go then? |
| 19:33.04 | [TK]D-Fender | FinboySlick, MATHS y0 |
| 19:33.37 | WIMPy | jasonbassett: You need a 2nd extension for that. Or you have to answer the call before forwarding. |
| 19:33.39 | Katty | i can haz nap nao plz?! |
| 19:33.42 | elemenopy | can i post code into this window? |
| 19:33.59 | becca_r | use pastebin or something for code |
| 19:34.04 | eppigy | i would use paste |
| 19:34.06 | eppigy | yes |
| 19:34.07 | eppigy | that |
| 19:34.09 | SparFux | elemenopy: preferably the whole asterisk code for sure :-) |
| 19:34.11 | [TK]D-Fender | jasonbassett, that isn't some global forward. What you seem to be asking for is just basic dialplan logic |
| 19:34.23 | WIMPy | McBoingBoing: There are probably other tools, but IIRC wireshark does have some features for that kind of stuff. |
| 19:34.26 | jasonbassett | I dont want my inbound call to be diverted (as I am just really calling in to inform the system of my location), only subsequent calls from other parties need to be forwarded |
| 19:34.28 | FinboySlick | [TK]D-Fender: Oh my, you flirting? ;) |
| 19:34.34 | Katty | i am. |
| 19:34.42 | McBoingBoing | meow! |
| 19:34.45 | Katty | ohai |
| 19:34.47 | Katty | hugs McBoingBoing |
| 19:34.50 | Katty | how're you dear. |
| 19:34.52 | [TK]D-Fender | FinboySlick, Hey, you went all Offspring... context .... |
| 19:35.11 | McBoingBoing | same old same old |
| 19:35.12 | jasonbassett | hmm, i'm making it harder than it needs to be then |
| 19:35.22 | McBoingBoing | jasonbassett: thats what she said? |
| 19:35.28 | Katty | jasonbassett: that's not what she said. |
| 19:35.32 | becca_r | lol |
| 19:35.33 | jasonbassett | hehe |
| 19:35.34 | McBoingBoing | hehe |
| 19:35.45 | elemenopy | "tcpdump -s0 host $ipaddress -w /tmp/$filename -c 65000" you can pinpoint a specific ip with that and write it out to a file which can be used in wireshark, remove the $filename and replace with <filename.pcap> |
| 19:35.51 | WIMPy | jasonbassett: What about a web interface? |
| 19:36.07 | Katty | McBoingBoing: jinx! |
| 19:36.16 | McBoingBoing | elemenopy, cool thanks, yeah I was just looking at an article suggesting something similar |
| 19:36.20 | eppigy | buy me a coke! |
| 19:36.27 | jasonbassett | likely to have access from the phone I want to divert to, but not a PC with web access |
| 19:36.33 | Katty | coke? pfff |
| 19:36.42 | eppigy | jim beam neat? |
| 19:36.52 | Katty | why soda when you can find someone to spoil you absoultely rotten?! |
| 19:37.18 | eppigy | i am already rotten but i like being spoiled |
| 19:37.21 | FinboySlick | [TK]D-Fender: Oh, right... And to think I was flattered for a moment. |
| 19:37.22 | WIMPy | jasonbassett: If you send the call to VM after some time, you can do it that way. |
| 19:37.26 | Katty | me too |
| 19:37.32 | Katty | tho i'm probably both |
| 19:37.36 | jasonbassett | for example, when I arrive at my brothers house, I want to dial my system using his phone and have all my calls forwarded to his phone. I may be at anyone else's phone though, not just his. |
| 19:38.00 | *** part/#asterisk SparFux (~raoul@rl2-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 19:38.02 | McBoingBoing | I am tired of users that have way less of a clue than me shitting on our VOIP setup, decent connection here, and the system is quiet in terms of CPU/Network IO, so I want to learn how to PROVE that VOIP is fine...but so far it is not an easy task |
| 19:38.45 | Katty | Nugget: ping |
| 19:38.49 | WIMPy | Are you sure, you want to try? |
| 19:38.50 | Katty | Nugget: telnet. |
| 19:39.00 | Katty | aww i wasn't first :< |
| 19:39.00 | Katty | boo |
| 19:39.28 | elemenopy | McBoingBoing: there are several ways to do this but there some cost in time |
| 19:39.52 | McBoingBoing | elemenopy, what you talking bout Mr. D? |
| 19:39.54 | jasonbassett | I dont see how I can do it via voicemail |
| 19:40.28 | elemenopy | McBoingBoing: when you say "PROVE that VOIP is fine." are you trying to prove call quality? |
| 19:40.34 | WIMPy | You can exit VM with 0 or *. |
| 19:40.41 | McBoingBoing | elemenopy, yes |
| 19:41.26 | p3nguin | jasonbassett: What happens if someone, including you from someone else's house, call your DID number? |
| 19:41.49 | becca_r | I'd personally just set it up to answer and do a waitexten, then listen for #9 compare what is in the DB for CID, if different then update. If no #9 then forward to dbentry. |
| 19:41.54 | elemenopy | McBoingBoing: maybe start by compairing timestamps with your wireshark tool, showing recordings to the people interested by sending .wav files of call quality that's been recorded, gather statistics about average system load and resource consumption |
| 19:42.00 | p3nguin | What you're asking for is basic dial plan logic. There just has to be a hook of some sort to be able to make the change. |
| 19:42.37 | jasonbassett | They will do often, that is my normal DID for the world to call me on. They would need to know to press the #9, which I would also look into passwording once I have the base setup working. |
| 19:43.06 | p3nguin | I'm just asking what happens when someone calls the number. |
| 19:43.32 | wcselby | o/ |
| 19:43.39 | wcselby | gotta love comcast technical support |
| 19:43.43 | wcselby | ........or not |
| 19:43.48 | becca_r | ewwww |
| 19:44.00 | p3nguin | Possible answers: it rings on their side while my phone rings, awaiting my answer; it answers and provides an attendant; et cetera. |
| 19:44.05 | jasonbassett | Someone calls number and my phone rings, but during ringing I can dial #9 to divert all new inbound calls to the phone I was calling in from. |
| 19:44.32 | p3nguin | So it is a true DID -- direct to dialing your phone. |
| 19:44.33 | Katty | hi p3nguin |
| 19:44.38 | p3nguin | Hello. |
| 19:44.41 | Katty | how're you dear |
| 19:44.45 | p3nguin | Typical. |
| 19:44.46 | Katty | and the wifey |
| 19:45.05 | jasonbassett | Picked up by my Asterisk box |
| 19:45.08 | p3nguin | I'm sure she's typical, but I haven't seen her since 6 AM. |
| 19:45.49 | Katty | that's unfortunate |
| 19:46.03 | p3nguin | She always comes back eventually. |
| 19:46.13 | Katty | i recommend a shower for catching up on Lost Time |
| 19:46.14 | p3nguin | (each day, that is) |
| 19:46.23 | Katty | extra bubbly |
| 19:46.25 | wcselby | Lost Time? |
| 19:46.31 | Katty | it's my favorite wallaby! |
| 19:46.34 | Katty | i mean wcselby |
| 19:46.35 | wcselby | :) |
| 19:46.39 | wcselby | o/ Katty |
| 19:46.40 | Katty | hugs wcselby |
| 19:47.25 | Katty | wcselby: and how're you? :> |
| 19:47.39 | wcselby | i'm mad and I don't want to take it anymore |
| 19:47.47 | p3nguin | jasonbassett: Is there any opposition to having asterisk actually answer the call before sending to your phone? I think it was wimpy that mentioned you'll have to have an answer before you can enter any special code. |
| 19:48.27 | Katty | wcselby: aww i am sorries :< |
| 19:48.34 | WIMPy | Yes, or wait for the call to go to VM and take it from there. |
| 19:48.35 | Katty | wcselby: who's pissin you off, i'mma go kick their tail |
| 19:48.39 | jasonbassett | No opposition, I think I see what your saying, will just try something.... |
| 19:48.40 | p3nguin | Some people oppose it due to billsec. |
| 19:49.20 | jasonbassett | when in VM i tried my #9 but it didnt work, will try answering line first.... |
| 19:49.22 | wcselby | comcast |
| 19:49.31 | p3nguin | If you had an attendant, it would be super-easy to add. |
| 19:49.46 | p3nguin | Once you are in VM, the line should already be "Up." |
| 19:50.04 | p3nguin | And by line, I mean channel. |
| 19:50.05 | Katty | shakes fist at comcast |
| 19:50.23 | p3nguin | shakes katty's fist at comcast |
| 19:50.34 | Katty | kthx for the help. |
| 19:50.49 | WIMPy | You can only exit VM with 0 or *. |
| 19:51.06 | *** join/#asterisk azv4 (~azv4@50-73-175-237-pennsylvania.hfc.comcastbusiness.net) |
| 19:51.15 | jasonbassett | I put an Answer() line in at beginning of inbound exten but #9 etc. still not being picked out |
| 19:51.19 | *** join/#asterisk jimi_ (~jimi@unaffiliated/tuxguy) |
| 19:51.33 | p3nguin | Did you enable the dynamic features? |
| 19:51.34 | jimi_ | Is there a repo for Centos 6? I only see Centos 5 on yhe asterisk yum page. |
| 19:51.40 | p3nguin | not that I know of. |
| 19:51.51 | p3nguin | Last I looked, 5 was the latest. |
| 19:52.31 | jasonbassett | exten => 01298567567,1,Answer() |
| 19:52.31 | jasonbassett | exten => 01298567567,1,n,set(__DYNAMIC_FEATURES=toggleforwarding) |
| 19:52.46 | jasonbassett | hm |
| 19:52.52 | jasonbassett | that dont look good |
| 19:53.09 | jasonbassett | Oh no, just a cut n paste issue |
| 19:53.09 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v019-127.mobile.uci.edu) |
| 19:53.24 | jasonbassett | extra 1 is not really there |
| 19:53.34 | p3nguin | weird |
| 19:54.33 | Nugget | silly katty |
| 19:54.48 | Katty | :> |
| 19:54.51 | Katty | glomps Nugget |
| 19:55.40 | *** part/#asterisk Twitchnln (~Adium@adsl-184-36-49-49.asm.bellsouth.net) |
| 19:55.42 | *** join/#asterisk epaphus (~epaphus@200.122.149.9) |
| 19:56.04 | p3nguin | I guess wimpy's idea is a reasonable one. Once in voice mail, exit voice mail by pressing the necessary key, which takes you to another "section" that allows entering special extensions to run certain things, which, in this case, would be your toggle. |
| 19:56.15 | epaphus | Hello. So If I administrate asterisk via freepbx... I cant get support from here right? is that because freepbx has its own format of altering /etc/asterisk ? |
| 19:56.27 | p3nguin | I wouldn't have done it via features, but that's just me. |
| 19:56.35 | p3nguin | I would do it with regular extensions. |
| 19:56.45 | jasonbassett | just looking at the waitexten option too |
| 19:56.46 | p3nguin | ~freepbx |
| 19:56.46 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 19:57.08 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
| 19:57.26 | p3nguin | WaitExten() will provide silence while it waits; BackGround() could be used to play some sounds while waiting for input. |
| 19:58.01 | p3nguin | It could be used so that if you do not enter the necessary extension during the playing of the sound file, you miss your chance to authenticate. |
| 19:59.04 | jasonbassett | noted, thank you |
| 19:59.28 | p3nguin | I have an idea how I would write the dial plan if I wanted to implement that type of system on my on equipment. |
| 20:01.49 | [TK]D-Fender | epaphus, Depends what you need. Most call flow & configuring = not here. |
| 20:01.53 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
| 20:02.56 | p3nguin | It looks like VoiceMail()'s option d could be useful to do it using the voice mail idea. |
| 20:03.13 | *** join/#asterisk azv4 (~azv4@50-73-175-237-pennsylvania.hfc.comcastbusiness.net) |
| 20:03.14 | p3nguin | That's probably the route I'd take. |
| 20:03.50 | Katty | eppigy: proof that i am spoiled...getting paid to sit here and paint my nails. |
| 20:03.58 | eppigy | HAHA |
| 20:04.03 | eppigy | whoops caps |
| 20:04.12 | eppigy | yeah i am basically going to just go home |
| 20:04.16 | eppigy | no one is here |
| 20:04.27 | Katty | do eet |
| 20:04.30 | Katty | i'm leaving in an hour |
| 20:04.32 | Katty | it's dead here too |
| 20:06.07 | Katty | eppigy: are you handing out candy tnight |
| 20:06.13 | eppigy | haha no |
| 20:06.24 | eppigy | no one wants to go to our house |
| 20:06.27 | eppigy | we have big dogs |
| 20:06.36 | eppigy | two jeeps and a trans am in the driveway |
| 20:06.52 | eppigy | and we are always carrying guns in and out of the house |
| 20:06.59 | eppigy | no one goes near our place |
| 20:07.18 | eppigy | the fedex guy drops packages are the end of the car port and runs back to his trucki |
| 20:07.21 | p3nguin | That sounds like most residences around here. It's the ones that are NOT like that that people don't go near. |
| 20:08.01 | eppigy | yeah that should be every household |
| 20:08.13 | eppigy | AMERIKA 4 LYFE |
| 20:10.05 | p3nguin | Got a gun rack on the back window of the family sedan, even. |
| 20:12.30 | r0m|u | same down south |
| 20:12.55 | jasonbassett | Looks like the Dial() apps d option will do what I need |
| 20:13.23 | *** join/#asterisk azv4 (~azv4@50-73-175-237-pennsylvania.hfc.comcastbusiness.net) |
| 20:17.11 | Katty | awww |
| 20:17.14 | Katty | trans am and doggies?! |
| 20:17.16 | Katty | i am SO there |
| 20:17.19 | wcselby | lol |
| 20:17.24 | Katty | i will bring riddick |
| 20:17.42 | wcselby | as in, the chronicles of? |
| 20:17.45 | Katty | i don't like guns tho. |
| 20:17.48 | Katty | wcselby: yes'r |
| 20:18.21 | wcselby | you know, he turned down some role to do that sequel |
| 20:18.40 | wcselby | i'm trying to think what it was |
| 20:18.42 | Katty | vin disel is HOT |
| 20:19.01 | Katty | i'd chew on im |
| 20:19.03 | Katty | him |
| 20:20.11 | devianTz | so I come to #asterisk just to look.....people are talking about vin diesel. |
| 20:20.27 | Katty | nom. |
| 20:21.12 | devianTz | they're already making a F&F6 so |
| 20:21.14 | devianTz | you're in luck |
| 20:21.15 | devianTz | -__- |
| 20:21.52 | Katty | suuhhhweeeet |
| 20:22.19 | *** join/#asterisk rdahlin_1_ (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 20:22.32 | jasonbassett | Thanks everyone, on my way to getting this forwarding thing working now - opted for the dial d option route. Seems to be working ok. |
| 20:22.34 | p3nguin | katty: You don't like guns?! I don't know you anymore. |
| 20:22.37 | *** part/#asterisk doug (doug@breakout.telerama.com) |
| 20:23.29 | Katty | newp don't like them |
| 20:23.30 | Katty | not a fan. |
| 20:23.52 | Katty | i support your right to own them, regardless. |
| 20:23.57 | p3nguin | So I guess we're not going shootin', then. |
| 20:24.05 | Katty | mmm probably not |
| 20:24.08 | Katty | unless they're water guns |
| 20:24.22 | p3nguin | Would you at least carry my ammo? |
| 20:24.25 | wcselby | lol |
| 20:24.27 | r0m|u | lol |
| 20:24.30 | r0m|u | hahahaha |
| 20:25.23 | Katty | idk, you gonna carry my shoes when my feet start hurting?? |
| 20:25.48 | p3nguin | If you carry my ammo, I should have room for them in my range bag. |
| 20:26.23 | p3nguin | or you could piggy back, and leave the shoes on. |
| 20:27.01 | r0m|u | lol..... rofl.... |
| 20:27.10 | Katty | that works. |
| 20:27.22 | r0m|u | You are a good woman! |
| 20:27.40 | Katty | computer people are the best, donchaknow |
| 20:27.45 | r0m|u | you are a keeper! |
| 20:27.49 | p3nguin | :) |
| 20:27.49 | r0m|u | lol |
| 20:28.09 | p3nguin | So... I think I might order a new pistol. |
| 20:28.15 | Katty | now if i could only find someone worth keeping me! |
| 20:28.42 | r0m|u | :/ |
| 20:28.50 | p3nguin | I'm looking at a P226, chambered in 9mm. |
| 20:29.15 | r0m|u | p3nguin, NightHawk. I have a GRP and ladyhawk |
| 20:29.18 | Katty | r0m|u: all the people in southern missouri are crazy. |
| 20:29.25 | r0m|u | Katty, LOL |
| 20:29.34 | *** part/#asterisk serafie (~erin@nat/digium/x-wriptwiwzxfudjbo) |
| 20:29.36 | p3nguin | Maybe you're looking for the wrong kind of person. |
| 20:29.44 | Katty | mmm, no. |
| 20:29.58 | r0m|u | we are nice here in Texas :) |
| 20:30.01 | Katty | i know what i want. |
| 20:30.09 | Katty | i liked dallas when i went to visit |
| 20:30.24 | wcselby | if dallas is the only part of texas you've been too |
| 20:30.25 | wcselby | to* |
| 20:30.28 | wcselby | you're missing out |
| 20:30.33 | Katty | orly |
| 20:30.53 | wcselby | i'm down in Houston, much nicer town than Dallas |
| 20:30.54 | wcselby | :) |
| 20:31.11 | r0m|u | I am in Spring :P |
| 20:31.26 | r0m|u | Dallas does suck |
| 20:31.30 | r0m|u | :P |
| 20:32.42 | p3nguin | I think San Antonio is the only city I've been to down there. |
| 20:33.15 | p3nguin | If that's on the route back from Phoenix, that's the place. |
| 20:33.48 | p3nguin | I stopped there to eat at a little Mexican restaurant called Amber's II. |
| 20:34.55 | r0m|u | Amber's? That does not sound mexican.... Maybe TexMex |
| 20:35.05 | p3nguin | She was Mexican. |
| 20:35.21 | r0m|u | Don Julio's or El Churro... |
| 20:35.31 | r0m|u | lol |
| 20:35.42 | navaismo | need a churro both of them |
| 20:35.45 | wcselby | The thought of trying to figure out which Mexican restaurant he went to in San Antonio is kind of funny |
| 20:36.19 | wcselby | there's probably more Mexican restaurants in San Antonio than there are white people in SA |
| 20:36.24 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-135-115.chyn.qwest.net) |
| 20:36.28 | devianTz | funny, I fly into San Antonio for a job interview on thursday :O |
| 20:36.33 | devianTz | never been. |
| 20:36.39 | wcselby | i hope you like tex-mex food :) |
| 20:36.45 | p3nguin | I was just passing through and it was near supper time, so I found a place that looked inviting. |
| 20:37.04 | wcselby | there's some great places there, you should watch Man V. Food for the SA episode |
| 20:37.16 | r0m|u | lol @ wcselby |
| 20:37.29 | wcselby | even the italian places are mexican |
| 20:37.34 | r0m|u | lmao |
| 20:37.35 | wcselby | there's a place with beef fajita pizza |
| 20:37.43 | p3nguin | I don't know how many Walmart SuperCenters there are in SA, but Amber's was not far from Walmart. |
| 20:37.45 | wcselby | etc etc |
| 20:37.54 | p3nguin | I stopped there to get gas. |
| 20:38.13 | wcselby | lol, that's another thing, I don't kmnow about SA, but in Houston, there's like 3 Super Wal-Marts within a 5 mile radius of where I live |
| 20:38.33 | wcselby | I spent some weeks in Kansas City a few years back, and there were 2 |
| 20:38.37 | wcselby | in all of KC |
| 20:38.41 | wcselby | in both states! |
| 20:38.53 | wcselby | i don't know how those poeple shop |
| 20:39.11 | *** join/#asterisk azv4 (~azv4@50-73-175-237-pennsylvania.hfc.comcastbusiness.net) |
| 20:39.15 | p3nguin | That's closer to my neck of the woods, so you understand why I would think Walmart could be used as a landmark. |
| 20:39.23 | p3nguin | potentially |
| 20:39.32 | wcselby | :) |
| 20:39.38 | *** join/#asterisk Syrex (~syrex@dsl-165-146-17-16.telkomadsl.co.za) |
| 20:39.54 | p3nguin | We have no more than one per city in these parts. |
| 20:41.07 | p3nguin | I remember cruising down some streets that looked very questionable trying to find a place to eat. |
| 20:41.28 | *** part/#asterisk Steavis (Steavis@lib-stf-lst-121.lib.asu.edu) |
| 20:42.15 | p3nguin | I never saw any questionable people, but I expected to at any moment. |
| 20:42.21 | wcselby | yeah, that sounds like SA, but then again, that could have happened in Houston or Dallas too |
| 20:42.30 | wcselby | just if you got off on the wrong exit, etc |
| 20:42.36 | p3nguin | yeah |
| 20:42.50 | r0m|u | agree |
| 20:42.58 | p3nguin | I think every big(ger) city has at least one area like that. |
| 20:44.05 | *** join/#asterisk ariel_ (u3533@pdpc/supporter/active/abatista) |
| 20:44.23 | wcselby | did you find the highways in SA confusing? I remember having to exit I10 to stay on I10 multiple times |
| 20:44.23 | r0m|u | one? try many! houston is full of area you really dont want to exit or even cruse by |
| 20:44.31 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 20:44.33 | r0m|u | lol |
| 20:45.46 | p3nguin | I really don't recall having too much trouble. It was like five years ago. |
| 20:47.19 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 20:47.22 | r0m|u | http://maps.level3.com/default/ |
| 20:47.29 | r0m|u | cool link I guess |
| 20:48.32 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279681716.dsl.bell.ca) |
| 20:48.35 | dijib | http://images.4chan.org/k/src/1320086446703.png |
| 20:49.08 | p3nguin | I don't get it. |
| 20:49.16 | r0m|u | me nether |
| 20:49.35 | r0m|u | dijib, does that mean you run a gay conference line? |
| 20:49.42 | p3nguin | heh |
| 20:49.48 | dijib | yes |
| 20:50.01 | r0m|u | lol j/k :) |
| 20:50.15 | r0m|u | worked has been killing me today |
| 20:50.36 | p3nguin | At least make it bi-sexual so there are SOME females on it. |
| 20:50.39 | r0m|u | work* |
| 20:50.44 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 20:50.50 | r0m|u | LOL @ p3nguin |
| 20:53.07 | p3nguin | I just can't decide if I want that P226 or not. It's a two-tone with night sights. |
| 20:53.19 | p3nguin | DAK trigger |
| 20:53.34 | p3nguin | Much better than the DAO trigger on my P250. |
| 20:54.04 | *** join/#asterisk nafg_ (~quassel@ool-4355e4a2.dyn.optonline.net) |
| 20:54.19 | nafg_ | I'm having a strange problem. Using the Festival dialplan application via FastAGI, when I connect to my server via a SIP phone everything is fine. |
| 20:54.24 | r0m|u | Buy It and sell your P250 |
| 20:54.25 | nafg_ | But when I call via IPKall, the beginning of many sentences are dropped. |
| 20:54.34 | nafg_ | Any ideas? |
| 20:54.59 | p3nguin | I'll probably keep the P250 for a while. I've only had it for a little over a year. |
| 20:55.45 | r0m|u | cool. IMHO the P226 is the way to go. |
| 20:56.00 | dijib | francis was in the conf before i left to go get more halloween candy |
| 20:56.51 | p3nguin | I've also been looking for a P220 carry, a P225/P6, a P229, and a P239. |
| 20:57.35 | r0m|u | As a carry I use a STI Escort |
| 20:58.40 | r0m|u | I also Carry a Glock 26 |
| 20:59.12 | ariel_ | Does anyone have any good dial plan examples for a good replacement for AgentCallbackLogin that has been removed? AEL is not an option. And it should have never been an option for replacement of a good command |
| 21:00.57 | p3nguin | I'm not a fan of Glock. |
| 21:01.08 | p3nguin | I don't like striker-fired pistols very well. |
| 21:01.39 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:01.40 | p3nguin | With my P250, I never have to worry about it shooting myself. |
| 21:02.16 | p3nguin | With a striker, there's nothing to guarantee that it can't be fired accidentally. |
| 21:02.28 | nafg_ | I'm having a strange problem. Using the Festival dialplan application via FastAGI, when I connect to my server via a SIP phone everything is fine. |
| 21:02.31 | nafg_ | But when I call via IPKall, the beginning of many sentences are dropped. |
| 21:02.35 | p3nguin | And yes, I do know how Glock's mechanism is built inside. |
| 21:03.04 | p3nguin | nafg_: Add some silence and see if that problem goes away. |
| 21:03.15 | nafg_ | p3nguin: How do I do that? |
| 21:03.16 | p3nguin | Playback(silence/2) |
| 21:03.22 | r0m|u | If you have to worry about that than you have bigger issues. I carry @ condition 1 |
| 21:03.59 | p3nguin | I don't understand what you're saying. |
| 21:04.00 | nafg_ | Is that the name of a wav file containing two seconds of silence? |
| 21:04.04 | r0m|u | which can be argued as 0 in the glocks |
| 21:04.15 | p3nguin | Yes, that is a two-second silent file. |
| 21:04.36 | *** join/#asterisk MariusKarthaus (~quassel@5418654F.cm-5-1b.dynamic.ziggo.nl) |
| 21:05.01 | nafg_ | So it will work with AGI's STREAM FILE? |
| 21:05.08 | p3nguin | I have no idea. |
| 21:05.24 | p3nguin | I'm just saying play the file before you play whatever other thing was having trouble. |
| 21:05.37 | elemenopy | nafg_: if you can stream .gsm then probably yes |
| 21:05.59 | p3nguin | I often pad playback with silence in front to allow RTP to get established. |
| 21:06.27 | p3nguin | I usually use silence/1, but if 1 is not enough, I use 2. |
| 21:07.14 | elemenopy | nafg_: /var/lib/asterisk/sounds/en/silence/2.gsm is the file |
| 21:07.30 | MariusKarthaus | Hi. I have a grandstream gxv3140 behind NAT connecting to an asterisk server on a 'realworld' IP on a server in a datacenter. Since a few days i'm getting exactly 30 missed calls in a very short time through account 1. But I do not see anything going wrong on the asterisk server. No mention in the messages or CDR files. Does anyone know this strange behaviour or have any tips where I should start looking? |
| 21:07.56 | elemenopy | MariusKarthaus: possible toll fraud |
| 21:08.05 | p3nguin | peer configuration, dial plan |
| 21:08.42 | jasonbassett | Done |
| 21:09.01 | jasonbassett | Heres my dialplan I have used in case it is useful to anyone: |
| 21:09.09 | jasonbassett | ;Toggle on/off and change call forwarding numbers for inbound calls |
| 21:09.09 | jasonbassett | exten => 9,1,Authenticate(1234,,4) |
| 21:09.10 | jasonbassett | exten => 9,n,GotoIf($["${DB(forwarding/01368123456)}" = ""]?needtoset) |
| 21:09.10 | jasonbassett | exten => 9,n,GotoIf($["${DB(forwarding/01368123456)}" != "${CALLERID(num)}"]?needtoset) |
| 21:09.10 | jasonbassett | exten => 9,n,Set(DB(forwarding/01368123456)=) |
| 21:09.11 | jasonbassett | exten => 9,n,Hangup() |
| 21:09.11 | jasonbassett | exten => 9,n(needtoset),Set(DB(forwarding/01368123456)=${CALLERID(num)}) |
| 21:09.12 | jasonbassett | exten => 9,n,Hangup() |
| 21:09.16 | r0m|u | p3nguin, some times I have a very low echo in my line.... I can barely hear it but is there.... Any ideas what could be causing this? |
| 21:09.26 | r0m|u | jasonbassett, use PB! |
| 21:09.32 | r0m|u | ~pastebin |
| 21:09.32 | infobot | [~pastebin] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 21:09.50 | jasonbassett | ah right, will do in future, sorry |
| 21:09.59 | r0m|u | :) |
| 21:10.48 | p3nguin | r0m|u: Usually it is from feedback from speaker to microphone. If you turn down the speaker volume a little, it may go away. |
| 21:11.43 | r0m|u | p3nguin, even on a handset? |
| 21:11.57 | p3nguin | yes |
| 21:11.58 | MariusKarthaus | elemenopy & p3nguin those two were directed at me right? How would I start find out if toll fraud is what they are trying ? And how are they making my phone at home ring? I'm not seeing any strange peers logged in when I do sip show peers btw |
| 21:12.07 | r0m|u | p3nguin, ok ill try that. Thanks |
| 21:12.39 | p3nguin | mariuskarthaus: Unless you need to accept calls from people you do not know, change allowguest to no in sip.conf. |
| 21:12.47 | wcselby | r0m|u- especially on a headset |
| 21:13.40 | r0m|u | wcselby, I guess I meant handset but I guess it all applies |
| 21:13.44 | r0m|u | Thanks guys |
| 21:13.49 | elemenopy | mariuskarthaus: turn on logging during the periods which are getting spammed. "full" should do it, take note of any calls which come in from " incoming call from ' ' to '' " and anything from output which does not contain information |
| 21:13.51 | wcselby | oh sorry :) |
| 21:13.59 | MariusKarthaus | p3nguin: and by calls from people I do not know you mean "anyone on the web" and not "people dialing into my phone number(s) that I have with nudgetphone" right> |
| 21:14.21 | p3nguin | mariuskarthaus: That's correct. Calls to your phone number are from a known peer, your ITSP. |
| 21:14.23 | MariusKarthaus | nudget => budget |
| 21:14.39 | elemenopy | mariuskarthaus: full logging i believe can be setup through logger.conf |
| 21:15.47 | p3nguin | allowguest=no will stop the anonymous calls. |
| 21:17.49 | MariusKarthaus | I checked the allowguest and it was indeed turned to on |
| 21:18.03 | MariusKarthaus | i shut that off and tested, i can still receive calls :P |
| 21:18.34 | MariusKarthaus | I've also enabled 'full' in logger config. And I now have /var/log/asterisk/full |
| 21:18.37 | wcselby | MariusKarthaus- are the anonymous calls coming in through asterisk or direct to your phone? |
| 21:18.51 | MariusKarthaus | but not much is being logged, not even my own test call from my mobile |
| 21:19.35 | MariusKarthaus | wcselby: I do not know if they are comming from asterisk. My phone says that are calling on account 1, which is asterisk. But on asterisk i do not see anything |
| 21:19.36 | wcselby | MariusKarthaus- add "verbose" to the logging for full and then reload logger, then up your cli verbosity to between 7 and 10 |
| 21:20.32 | p3nguin | There is no reason to go beyond 3. |
| 21:20.43 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:20.49 | p3nguin | It does not get more verbose at 5, 6, 7, ... 10. |
| 21:20.55 | MariusKarthaus | wcselby: ah yes I did not know the file logger was bound to the debug settings from CLI |
| 21:20.57 | wcselby | p3nguin- i remember we came up with a reason to go past 3 at some point, it was like up to 6 I think? |
| 21:20.58 | MariusKarthaus | Did that |
| 21:21.08 | MariusKarthaus | stuff is now comming in |
| 21:21.15 | p3nguin | At 4, it adds the dnsmgr flood. |
| 21:21.18 | p3nguin | At like 11, it adds CDR. |
| 21:21.19 | wcselby | some app that only logged at 6, but I don't remember |
| 21:21.30 | wcselby | it's been a while since we had the discussion |
| 21:21.46 | p3nguin | Did you remember to sip reload after you changed allowguest to no and saved the file? |
| 21:22.19 | MariusKarthaus | p3nguin: because I needed the new full logfile i assumed i needed a full restart anyway |
| 21:22.26 | MariusKarthaus | so i did a restart af asterisk |
| 21:22.26 | wcselby | i'm jumpin gin the middle here, but it sounds like you've either got a phone on a public IP, or port 5060 routed directly to a phone on a LAN somewhere, and you're getting scanned |
| 21:22.35 | p3nguin | Are you using other channel drivers? |
| 21:22.40 | p3nguin | Maybe it isn't coming in on SIP. |
| 21:22.50 | MariusKarthaus | only sip |
| 21:22.58 | wcselby | what do the calls look like on your phone? |
| 21:23.22 | p3nguin | With allowguest=no, the call would have to match a peer entry to continue to dial plan. |
| 21:23.39 | p3nguin | Maybe you should pastebin your entire sip.conf. Hide ONLY PASSWORDS. |
| 21:24.07 | wcselby | p3nguin- that's what I'm saying, it sounds like his home phone is getting random scans, if all it does is ring the phone. if you answer, is anyone there? this happens to my desk phone at home quite frequently. |
| 21:24.12 | MariusKarthaus | like exactly 30 missed calls that get done in 1 second. flooding my phone (sometimes it crashes), after a while it restores and gives me the regular missed call screen with the 30 missed calls from 'unknown' |
| 21:24.35 | wcselby | yeah, that's exactly what I've seen. it's not coming from your asterisk box at all |
| 21:24.41 | wcselby | it's directly on your IP address |
| 21:24.43 | p3nguin | Why would someone have access to your phone at home? |
| 21:25.03 | hardwire | squirrels |
| 21:25.04 | hardwire | cats |
| 21:25.05 | hardwire | dogs |
| 21:25.06 | wcselby | my phone establishes a connection, my home router opens port 5060 and leaves it open for a short time, i get a scan |
| 21:25.07 | hardwire | etc... |
| 21:25.13 | MariusKarthaus | my desk phone is not on a real IP. is on 192.168.1.34 and my router does not staticly port forward any hosts on my lan |
| 21:25.37 | wcselby | MariusKarthaus- same thing for me. i've never worried about it |
| 21:25.51 | p3nguin | It magically forwards 5060 to your phone, regardless of what started it? |
| 21:25.57 | wcselby | no |
| 21:26.04 | wcselby | well hell,I dunno |
| 21:26.10 | wcselby | i've never really looked that deep into it |
| 21:26.15 | wcselby | i just assumed nat keepalive or something |
| 21:26.16 | MariusKarthaus | wcselby: I think that if my phone makes a connection to my server the router does not 'open 5060' to the world... it opens 5060 only to the IP of the voip server |
| 21:26.19 | hardwire | p3nguin: sometimes routers are idiotic |
| 21:26.28 | p3nguin | But THAT idiotic? |
| 21:26.36 | hardwire | and instead of hashing out conntracks with a source and dest.. it only hashes out the source |
| 21:26.41 | p3nguin | That doesn't really seem practical. |
| 21:26.52 | wcselby | i'm on a u-verse router at home, nothing fancy |
| 21:26.53 | hardwire | yet it's been going on for a LONG time. |
| 21:27.38 | hardwire | newer conntrack solutions make it more difficult for traffic to flow ingress just because an egress happened. |
| 21:27.41 | hardwire | which I love. |
| 21:27.48 | MariusKarthaus | I'm on a pretty good cisco, should be ok |
| 21:27.52 | hardwire | but older/off-the-shelf solutions just open up an outside port and nat it in. |
| 21:27.57 | *** join/#asterisk jerware (~jerryg@c-71-58-179-44.hsd1.pa.comcast.net) |
| 21:27.58 | jerware | moin. |
| 21:28.07 | jerware | Are there bugs in asterisk? |
| 21:28.12 | hardwire | jerware: no |
| 21:28.15 | hardwire | not that I'm aware of |
| 21:28.16 | MariusKarthaus | hehe |
| 21:28.20 | jerware | Someone is recomending freeswitch for me. |
| 21:28.24 | MariusKarthaus | there are bugs in any software :P |
| 21:28.54 | hardwire | jerware: freeswitch has bugs.. asterisk however has been running stable with no bugs for.. minutes.. |
| 21:28.59 | wcselby | jerware- no bugs, just undocumented features |
| 21:29.55 | [TK]D-Fender | If they completely lock your system they're SPECIAL FEATURES |
| 21:30.05 | wcselby | VERY SPECIAL FEATURES |
| 21:30.16 | wcselby | now with added CAPSLOCK |
| 21:30.46 | [TK]D-Fender | Extra Special Features are ones you should have seen coming.... |
| 21:30.48 | *** join/#asterisk BlackBishop (~dexter@ipv6.d3xt3r01.tk) |
| 21:31.54 | MariusKarthaus | according to the bugtracker there are about 90 bugs that crash and 170 bugs of major importance.... not too bad :P |
| 21:32.21 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
| 21:33.10 | MariusKarthaus | anyway I sure hope that the phantom calls now go away or at least that I have enough info inthe logs to dive deeper into it next time. |
| 21:34.05 | MariusKarthaus | thank you wcselby & p3nguin |
| 21:34.30 | MariusKarthaus | and elemenopy ! |
| 21:35.46 | wcselby | np |
| 21:36.04 | jerware | I hear freeswitch is superiour to asterisk |
| 21:36.27 | p3nguin | Like a banana is superior to a watermelon. |
| 21:36.40 | r0m|u | i hear you are a troll |
| 21:36.50 | wcselby | i hear shoes are superiour to gloves |
| 21:36.54 | [TK]D-Fender | jerWhat else do your Rice Crispies say to you? |
| 21:37.02 | hardwire | jerware: I really hope there's an actual point to what you're saying. :) |
| 21:37.08 | wcselby | it depends on what the need is you're trying to satisfy |
| 21:37.23 | hardwire | comparing the two in an channel about asterisk seems like bad karma. |
| 21:37.25 | *** join/#asterisk datarecal (~data@S0106c43dc7876e60.cg.shawcable.net) |
| 21:37.32 | [TK]D-Fender | hardwire: Some people take a long time to make their pointless :) |
| 21:37.37 | datarecal | any one know a good DID provider for AU 1800's |
| 21:37.46 | hardwire | datarecal: voip.ms? |
| 21:37.56 | [TK]D-Fender | hardwire: ... my karma ran over your dogma :p |
| 21:38.16 | hardwire | [TK]D-Fender: my catma peed on your karma. |
| 21:38.37 | wcselby | and on that note |
| 21:38.42 | wcselby | :) |
| 21:38.50 | datarecal | hardwire ill take a look |
| 21:38.51 | hardwire | datarecal: http://voip.ms/intldids.php?CountryID3=13&countryselected=AUSTRALIA |
| 21:38.52 | elemenopy | datarecal: voip.ms make you sign a waiver stating any toll fraud your 100% responsible for there should be some other provider with more relaxed terms i think |
| 21:39.09 | wcselby | adios folks, happy halloween! |
| 21:39.23 | hardwire | elemenopy: why should they be respo.. |
| 21:39.25 | hardwire | oh.. fine.. |
| 21:39.27 | hardwire | just quit then. |
| 21:39.33 | wcselby | lol |
| 21:39.35 | wcselby | o/ |
| 21:39.47 | hardwire | fine! |
| 21:41.51 | dijib | bisexual confrence line yall 2663@asterisk.serveirc.com |
| 21:41.59 | hardwire | uh.. hi |
| 21:42.23 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 21:42.35 | datarecal | hardwire, anywhere else you can recommend : The end users location (place of residence/business) is required for geographical numbers. This info will be sent to VoIP.ms within 24 hours upon request, otherwise the number may be disconnected. The location should correspond to the geographical zone of the number used by the end user. |
| 21:42.37 | *** join/#asterisk hipitihop (~denis@202.153.71.36) |
| 21:42.41 | datarecal | so i cant get an aus number if i dont live there |
| 21:43.04 | hardwire | datarecal: ask them anyways. |
| 21:43.13 | hardwire | they wanna sell you DIDs right? |
| 21:43.28 | datarecal | yeah thats voip.ms TOS there |
| 21:43.31 | hardwire | show them you have a business oriented around exactly what they are doing. |
| 21:43.40 | hardwire | shrugs |
| 21:44.21 | hardwire | I'm guessing they don't have an address in all of those locations.. but an affiliate does. |
| 21:44.33 | hardwire | so it's sort of contrary if they have that in their TOS |
| 21:44.37 | hardwire | let em know you have business. |
| 21:44.40 | hardwire | go get em tiger! |
| 21:44.46 | hardwire | RAWR! |
| 21:45.53 | datarecal | lol |
| 21:45.59 | datarecal | talking to their live chat guys now |
| 21:47.01 | hardwire | just remember to wave your hand in front of their face and say "you want to do business with me" |
| 21:47.04 | hardwire | over and over |
| 21:49.19 | MariusKarthaus | hmm that voip.ms seems pretty expensive (at least for me / netherlands) |
| 21:49.44 | MariusKarthaus | too bad having a good DID provider is always nice to have spare :P |
| 21:51.10 | r0m|u | hardwire, By AU law you have to show prof you are from the originating DID |
| 21:51.33 | r0m|u | Its not voip.ms is the AU's LAW. |
| 21:51.47 | hardwire | r0m|u: you can't even get a permit? |
| 21:51.49 | r0m|u | same goes for Germany and Russia that I know off |
| 21:52.12 | hardwire | r0m|u: I remember somebody got a permit for a place with similar laws. |
| 21:52.48 | r0m|u | hardwire, that would be a question for voip.ms. And if you are serious I would call them do not write to them. |
| 21:53.13 | hardwire | sure nuff |
| 21:53.25 | *** part/#asterisk dtascom (~david@98-24-18-72.static.tierzero.net) |
| 21:53.35 | r0m|u | good luck. please report back if it worked out for you. |
| 21:54.21 | hardwire | not me.. datarecal |
| 21:55.36 | r0m|u | o ok. well he quit all ready. he might be in for a nasty surprise :P |
| 21:56.41 | *** join/#asterisk pietro (~pietro@88-149-227-165.dynamic.ngi.it) |
| 21:57.22 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 21:58.01 | hardwire | he'll just curse at me for wasting his time for a while. |
| 21:58.10 | r0m|u | lol |
| 21:58.18 | hardwire | I'm always challenging one of our buyers to find other ways to deal with DID problems. |
| 22:01.33 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-gkomfukldokjbsph) |
| 22:03.57 | hardwire | so far voip.ms is the only provider I trust within reason for Hawaii DID |
| 22:14.42 | *** join/#asterisk henk (~henk@leonardo.netwichtig.de) |
| 22:16.36 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 22:27.05 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 22:30.41 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v019-127.mobile.uci.edu) |
| 22:31.23 | p3nguin | They want you to be in AU to have an AU DID? |
| 22:32.06 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 22:32.12 | hardwire | p3nguin: you'd think that there would be consultation services to help get permits if you really need DID in AU but are not within it. |
| 22:32.23 | hardwire | I'd gladly pay for somebody to deal with it. |
| 22:32.42 | hardwire | Worth more than setting up shop and moving an employee :) |
| 22:33.14 | [TK]D-Fender | That's not what he said |
| 22:33.33 | [TK]D-Fender | In order to present an AU DID as your callerID you are required to have proved that you own it |
| 22:33.51 | [TK]D-Fender | You can't jsut fake an AU CID legally |
| 22:33.52 | *** join/#asterisk beccara (~beccara@mail.ubergroup.co.nz) |
| 22:33.53 | p3nguin | If I buy it from the ITSP, I own it. Problem solved. |
| 22:33.56 | hardwire | but would that get in the way of ordering new service? |
| 22:34.03 | [TK]D-Fender | p3nguin: Correct. |
| 22:34.09 | [TK]D-Fender | hardwire: No |
| 22:34.21 | p3nguin | It sounded like the problem was obtaining the DID to begin with. |
| 22:34.30 | p3nguin | I can't imagine it should be a problem. |
| 22:34.43 | p3nguin | I should see if I can order an AU DID. |
| 22:34.43 | [TK]D-Fender | Not from what's been said |
| 22:34.48 | hardwire | yeh. I thought you couldn't obtain it without showing a legal pres. within the country. |
| 22:34.59 | beccara | is anyone able to help with a dialstatus question? |
| 22:35.01 | [TK]D-Fender | Silly people don't read here :) |
| 22:35.10 | hardwire | bah! |
| 22:35.13 | hardwire | I didn't even read! |
| 22:35.19 | [TK]D-Fender | beccara: Your odd improve drastically after asking it ;) |
| 22:35.24 | beccara | :P |
| 22:35.46 | beccara | I'm using dialstatus to do things based on the return but the status of cancel never seems to be hit |
| 22:36.06 | beccara | If i call in and hit Dial() and then hangup it exit's right there |
| 22:36.12 | beccara | doesn't jump to s-CANCEL |
| 22:36.53 | [TK]D-Fender | beccara: If you the caller hangup then the call will never continue on. It will always hit "h" |
| 22:37.04 | p3nguin | The guy wanted an AU DID... but was his place of business/residence not in AU? |
| 22:37.04 | [TK]D-Fender | beccara: "CANCEL" is not for that scenario |
| 22:37.12 | beccara | hmm thats what I thought |
| 22:37.17 | beccara | oh really? whats it for? |
| 22:37.35 | [TK]D-Fender | beccara: "core show application dial" <- read the instructions and make another guess as to how such a status might have a reason to be returned. |
| 22:38.40 | *** join/#asterisk freeedrich| (friedrich@2a01:4f8:130:2023:1:151:0:babe) |
| 22:38.42 | beccara | you mean "CANCEL: Call is cancelled. The dial command reached its number but the caller hung up before the callee picked up." |
| 22:39.05 | r0m|u | [TK]D-Fender, I read that some country's require you to show proof that you reside in that country's to get a DID from that country. |
| 22:39.39 | [TK]D-Fender | r0m|u: Conceivable though I've never actually seen a case of it yet |
| 22:40.18 | [TK]D-Fender | beccara: Look at Dial's options. The answer is there |
| 22:40.18 | r0m|u | I see. |
| 22:40.41 | p3nguin | If the setup fee for area code 1800 in AU was not $38 with a $6.25/mo fee, I'd order one and see how long it took them to request my location of use. |
| 22:40.51 | r0m|u | most carriers do take it seriously I guess. |
| 22:40.53 | beccara | I've looked at the options, I'm not sure what your getting at |
| 22:41.12 | beccara | the cancel status is defined as this exact situation, Caller hanging up before callee picks up |
| 22:41.44 | beccara | the closest option I can see is "g" but thats for the called party |
| 22:42.04 | [TK]D-Fender | beccara: Not it. Look on... |
| 22:42.28 | beccara | or you could just say which option it is, I have actually read all the options |
| 22:42.54 | [TK]D-Fender | beccara: And nothing else there says "stop the call" to you? |
| 22:43.35 | r0m|u | out to the library... cya! |
| 22:43.35 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 22:43.37 | beccara | nothings standing out to me which is why I came in here to ask |
| 22:44.03 | [TK]D-Fender | h: Allow the called party to hang up by sending the '*' DTMF digit. |
| 22:44.05 | [TK]D-Fender | <PROTECTED> |
| 22:44.10 | [TK]D-Fender | Not clear enough? |
| 22:44.19 | beccara | let me reexplain |
| 22:44.21 | [TK]D-Fender | "hang up" |
| 22:44.32 | [TK]D-Fender | [18:38]beccarayou mean "CANCEL: Call is cancelled. The dial command reached its number but the caller hung up before the callee picked up." <- hung up |
| 22:44.53 | beccara | yep |
| 22:45.00 | [TK]D-Fender | There would be a perfectly valid place to expect "CANCEL" as a ${DIALSTATUS} |
| 22:45.24 | beccara | and yet not the place for a simple hangup before it was answered |
| 22:45.57 | beccara | h isn't the place for a hangup prior to the call connecting |
| 22:46.01 | [TK]D-Fender | You press * before it is answered = CANCEL |
| 22:46.33 | beccara | You hangup before it's anwered = NOT CANCEL in asteriskland |
| 22:47.03 | [TK]D-Fender | If the caller literally hangs up the the phone (lets say your typical SIP phone), then that channel dies. Just dies. Goes to "h". |
| 22:47.24 | [TK]D-Fender | Always has. |
| 22:47.33 | beccara | the channel is torn down with a cancel SIP message |
| 22:47.43 | [TK]D-Fender | That dialstatus is for a healthy termination of Dial |
| 22:47.57 | [TK]D-Fender | having the carpet ripped out is not it |
| 22:48.13 | beccara | it's still a graceful shutdown of the call |
| 22:48.18 | p3nguin | A DIALSTATUS of ANSWER would seem sensible for a Dial() which was answered. |
| 22:49.13 | [TK]D-Fender | But only if other options are provided |
| 22:50.26 | beccara | I would expect asterisk to treat the canceling of a call as a graceful tear down and jump to the dialstatus of cancel rather than the catchall of h |
| 22:51.06 | beccara | it certainly makes treating call failures differently very tricky |
| 22:51.06 | [TK]D-Fender | beccara: this behaviour pre-dates * 1.0 |
| 22:51.41 | beccara | which goes to show nobodys really looked at it in detail |
| 22:51.46 | [TK]D-Fender | beccara: Which is to say I'm tempted to say "always", but I did only start around .98 |
| 22:52.18 | beccara | since with h you can end up with a completed call and a failed call ending up in the same area |
| 22:52.43 | [TK]D-Fender | Yes, we've looked at it. We've acknowledged "yes, this is how it work, and pretty much always has" and we go on with our affairs just fine knowing it |
| 22:53.07 | [TK]D-Fender | beccara: yes, and there are things you can test at that point |
| 22:53.08 | beccara | lol |
| 22:53.31 | [TK]D-Fender | beccara: it simply isn't a "problem or "news" for anyone |
| 22:54.02 | beccara | You shouldn't have to test anything, The status of a call torn down by a SIP cancel before answer should be different to the status of a call torn down by a completed call |
| 22:54.11 | beccara | it's hardly rocket science |
| 22:54.37 | [TK]D-Fender | beccara: So both say "CANCEL" in your tests? |
| 22:55.10 | [TK]D-Fender | I've never had a completed call say "CANCEL" before... |
| 22:55.11 | beccara | what? |
| 22:55.15 | [TK]D-Fender | That'd be a neat trick... |
| 22:55.24 | beccara | I didn't say cancel on a completed call |
| 22:55.31 | beccara | I said a call torn down by a completed call |
| 22:55.47 | carrar | dialstatus doesn't give you cancel as a status? |
| 22:56.02 | beccara | not if the caller simply hangups up |
| 22:56.11 | [TK]D-Fender | You end up in "h" just the same... doesn't mean you can't see why |
| 22:56.37 | carrar | hangup send a BYE |
| 22:57.13 | SwK | i thought dialstatus was canceled on abandoned calls |
| 22:57.34 | SwK | am i wrong on that [TK]D-Fender ? |
| 22:57.48 | SwK | i honestly havent looked at that in a while heh |
| 22:58.06 | [TK]D-Fender | SwK: Just not the sort of thing most people even have to think about.... |
| 22:58.21 | beccara | I wish my employer would just fork out for a hiQ |
| 22:59.27 | SwK | i wish my employer would fork out for me to take one of those around the world booze cruises with the full sized suite but he's a cheap SOB |
| 22:59.39 | beccara | lol |
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