| 00:04.27 | powerunits | guysssss please can some one can tell me why.. |
| 00:04.34 | powerunits | voicemail are not going to email |
| 00:12.42 | Zopsi | Has anyone done the Digium D70 firmware upgrade? it keeps saying that the firmware upgrade has failed for me and I have no idea why. |
| 00:27.37 | *** join/#asterisk _Corey_ (~chatzilla@pool-72-78-178-17.phlapa.fios.verizon.net) |
| 00:34.45 | *** join/#asterisk pyther (~pyther@unaffiliated/pyther) |
| 00:34.55 | pyther | Hi. What is the difference between a friend and a peer? |
| 00:39.12 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 00:44.56 | Nivex | SIP is driving me crazy! |
| 00:56.02 | SteelReign | i here ya |
| 00:57.00 | Nivex | Cisco 7960 on the far end of a VPN. Diddling with the NAT directives (despite there being no NAT in the way because of the VPN) in order to get it back |
| 00:57.02 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 00:57.02 | SteelReign | my asterisk server wont pass IM's over SIP and I don't know if it's the Openfire Asterisk Plug-in or Asterisk itself. |
| 00:57.33 | pyther | is it possible to set a outbound caller id for a user? |
| 01:02.40 | *** part/#asterisk SteelReign (~steel@72-28-219-021-dhcp.mia.fl.atlanticbb.net) |
| 01:03.44 | *** part/#asterisk shadowapex (~William@adsl-99-61-86-59.dsl.lsan03.sbcglobal.net) |
| 01:05.12 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 02:09.10 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 02:09.10 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:20.45 | EmleyMoor | Is there anything I can do to block a specific IP from making SIP calls to me? |
| 02:20.56 | Kobaz | deny=x.x.x.x |
| 02:21.54 | EmleyMoor | ... in sip.conf? |
| 02:22.30 | Kobaz | yeah |
| 02:22.38 | Kobaz | you can see the sip.conf.sample for examples |
| 02:22.50 | carrar | Put that phone in a context that doesn't have access to your extension |
| 02:23.33 | carrar | I assume you still want that phone to be able to make calls |
| 02:23.34 | Kobaz | EmleyMoor: if it's something that's being abusive then you can add it to your firewall |
| 02:23.45 | carrar | You can also route the call basaed on it's caller ID |
| 02:24.06 | Kobaz | callerid based routing for security is terrible |
| 02:24.20 | carrar | then you have bad security |
| 02:24.40 | carrar | but it is a viable option |
| 02:24.54 | EmleyMoor | Kobaz: Tried adding it to my firewall already - didn't seem to work |
| 02:25.06 | Kobaz | EmleyMoor: then your firewall config isn't correct |
| 02:25.21 | EmleyMoor | ... an[d nor did that "deny" |
| 02:25.26 | Kobaz | callerid filtering/router can be part of a security scheme, but shouldn't be *the* security scheme |
| 02:25.35 | carrar | true |
| 02:25.45 | EmleyMoor | Kobaz: Either its config or its operationality. yes |
| 02:26.02 | Kobaz | if you're using iptables for your firewall then it's most certainly your configuration |
| 02:26.07 | EmleyMoor | I'm not |
| 02:26.09 | Kobaz | k |
| 02:26.29 | Kobaz | even if it's a cheapo linksys, it's probably the config |
| 02:26.46 | carrar | EmleyMoor, do you want to block the phone from your system entirely? or just calling your extension? |
| 02:27.24 | EmleyMoor | carrar: I want to block the "whatever it is" altogether |
| 02:27.41 | Kobaz | then you should really use the firewall instead of asterisk |
| 02:27.44 | carrar | You need find the IP of whatever it is then |
| 02:28.00 | carrar | Whats the IP? |
| 02:28.12 | EmleyMoor | 37.8.30.194 |
| 02:28.27 | carrar | I would then block 37.8.16.0/20 |
| 02:28.54 | carrar | since that IP belongs to larger block and they may change their IP |
| 02:29.08 | carrar | person: Walid Kassab |
| 02:29.08 | carrar | address: Palestinian Internet Services |
| 02:29.08 | carrar | address: P. O. BOX 5111 Gaza City, Palestine |
| 02:29.36 | Kobaz | why /20? |
| 02:29.53 | Kobaz | why not block the class c with a /24 |
| 02:29.57 | carrar | Cause thats the aggregatyed block that IP belongs too |
| 02:30.05 | carrar | and if they are on some dhcp |
| 02:30.12 | carrar | then they can come back as a different IP |
| 02:30.13 | EmleyMoor | My problem is not what to block but how to block |
| 02:30.18 | carrar | but still withing that range more then likely |
| 02:30.26 | Kobaz | oh the whole block is a /20 assigned to the walid guy |
| 02:30.31 | carrar | yeah |
| 02:30.33 | carrar | via RIPE |
| 02:30.40 | Kobaz | yeah, k that's fine |
| 02:31.15 | Kobaz | EmleyMoor: iptables -I INPUT --source 37.8.16.0/20 -j REJECT |
| 02:31.16 | Kobaz | done and done |
| 02:32.09 | EmleyMoor | I am not using iptables |
| 02:32.15 | Kobaz | well you can start |
| 02:32.16 | carrar | Your not!!! |
| 02:32.25 | carrar | What are you using? |
| 02:32.31 | Kobaz | since your other method of firewalling isn't working apparently |
| 02:32.45 | EmleyMoor | Firewalling in a proprietary router |
| 02:32.56 | Kobaz | cisco? |
| 02:33.03 | EmleyMoor | Technicolor |
| 02:33.08 | Kobaz | heh |
| 02:33.11 | Kobaz | what the heck is that |
| 02:33.18 | Kobaz | that's probably your problem :P |
| 02:33.19 | carrar | I thought they made studio cameras |
| 02:33.28 | EmleyMoor | I'm trying to get my supplier to help |
| 02:33.31 | Kobaz | you're better off using iptables then |
| 02:33.35 | carrar | yeah |
| 02:33.39 | Kobaz | just type that in your root prompt on your linux box |
| 02:33.41 | carrar | ise iptables |
| 02:33.44 | Kobaz | poof, no more traffic |
| 02:33.50 | carrar | heh |
| 02:33.51 | Kobaz | from that whole ip block |
| 02:34.37 | carrar | [root@zotac ~]# iptables -I INPUT --source 37.8.16.0/20 -j REJECT |
| 02:34.38 | carrar | [root@zotac ~]# iptables -n -L |
| 02:34.38 | carrar | Chain INPUT (policy ACCEPT) |
| 02:34.38 | carrar | target prot opt source destination |
| 02:34.38 | carrar | REJECT all -- 37.8.16.0/20 0.0.0.0/0 reject-with icmp-port-unreachable |
| 02:34.40 | carrar | see |
| 02:34.42 | carrar | thats proof |
| 02:36.05 | carrar | removes the block so his Palestinian friends can get back in |
| 02:36.10 | Kobaz | haha |
| 02:36.57 | Kobaz | never know, might miss an important call |
| 02:37.18 | carrar | NSA will get it for me |
| 02:37.37 | EmleyMoor | You're advising me to use something I just haven't got at present - fair enough |
| 02:37.38 | carrar | CIA or whever happens to be saving it at the time |
| 02:37.49 | Kobaz | EmleyMoor: why do you say you don't have it? |
| 02:38.03 | Kobaz | EmleyMoor: iptables --version |
| 02:38.24 | EmleyMoor | Command not found |
| 02:38.32 | Kobaz | oh wow |
| 02:38.34 | Kobaz | surprising |
| 02:38.35 | carrar | RU root |
| 02:38.40 | EmleyMoor | carrar: Yes |
| 02:38.49 | Kobaz | /sbin/iptables --version |
| 02:38.50 | carrar | /sbin/iptables --version |
| 02:38.51 | Kobaz | ? |
| 02:38.52 | Kobaz | heh |
| 02:39.14 | Kobaz | apt-get install iptables ? |
| 02:40.25 | EmleyMoor | Kobaz: Could do that I guess but have never used iptbles before so will probably need some help with setting it up |
| 02:40.34 | Kobaz | there's nothing to set up |
| 02:40.43 | Kobaz | you just feed it commands for what you want to do |
| 02:40.57 | Kobaz | and then if you want to keep those settings, then you add them to your bootup |
| 02:41.00 | Kobaz | and that's it |
| 02:41.27 | EmleyMoor | So if I just plain install it, it won't get in the way and I can then block that IP? |
| 02:41.31 | Kobaz | yeah |
| 02:41.35 | Kobaz | it doesn't do anything by default |
| 02:41.40 | Kobaz | it only does what you tell it to do |
| 02:41.53 | Kobaz | and when it's first installed, nothing is telling it anything to do |
| 02:42.11 | Kobaz | ie: default rules are allow |
| 02:43.16 | Kobaz | unless for some crazy retarded reason the package maintainer for your distribution decided otherwise |
| 02:43.38 | Kobaz | but if you're using anything 'mainstream' like debian, redhat, ubuntu, etc, then it wont be an issue |
| 02:44.15 | EmleyMoor | Well, that's stopped it for now... |
| 02:44.19 | carrar | that should be your next project, UPDATE |
| 02:44.24 | carrar | err UPGRADE |
| 02:44.37 | Kobaz | EmleyMoor: yay, step one complete |
| 02:44.55 | Kobaz | so what's the problem, is it like failed registrations? like some sort of attack? |
| 02:45.27 | EmleyMoor | Attempting to call numbers by sending them with random prefixes as anonymous SIP calls |
| 02:45.32 | Kobaz | ah |
| 02:45.33 | Kobaz | yeah |
| 02:45.43 | Kobaz | so your next project, is install, and learn fail2ban |
| 02:45.55 | Kobaz | no asterisk install should be without it |
| 02:46.23 | carrar | correct |
| 02:46.38 | carrar | if not for the ability to block something on a whim if need be |
| 02:46.52 | Kobaz | well, iptables is for whims |
| 02:47.09 | Kobaz | basically fail2ban looks at log files for bad stuff, failed logins, etc, and will add iptables rules to block the attacker |
| 02:47.24 | carrar | yeah I don't mean fail2ban, iptables |
| 02:47.35 | Kobaz | yeah |
| 02:47.53 | EmleyMoor | So I'd need to write something for it that reads the asterisk log and deals with this situation? |
| 02:48.30 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 02:48.30 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 02:50.05 | Kobaz | there are samples you can plug in for asterisk |
| 02:50.09 | Kobaz | and it will do a decent job |
| 02:50.25 | carrar | http://www.fail2ban.org/wiki/index.php/Asterisk |
| 02:50.28 | Kobaz | the syntax is simple enough, it's just regular expressions to match against the log files |
| 02:50.44 | Kobaz | but you should do enough reading to understand what it does and how it works so you can tailor it to your needs |
| 02:52.05 | carrar | well thats a incorrect example |
| 02:52.20 | carrar | but close |
| 02:52.24 | Kobaz | heh |
| 02:52.36 | Kobaz | exactly a good reason to learn the configs |
| 02:55.52 | Kobaz | depends on the asterisk version too |
| 03:02.31 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 03:29.28 | *** join/#asterisk li3p (~li3p@195.230.99.40) |
| 04:09.04 | *** join/#asterisk ilj (~ilj@sourcemage/grimoire/apprentice/ilj) |
| 04:23.29 | *** join/#asterisk bmg505 (~leon@196-209-171-201.dynamic.isadsl.co.za) |
| 05:18.00 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 05:18.06 | linocisco | #join #freepbx |
| 05:18.45 | linocisco | hi all |
| 05:18.57 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 05:19.27 | linocisco | i have installed dd-wrt , I could also telnet to it, i saw its busybox, I dont know how to install asterisk using which command and which file |
| 05:19.27 | Nugget | telnet is eeeeeeevil! |
| 05:19.39 | fling | what call-out service are you using? |
| 05:20.31 | *** join/#asterisk kayfox (~kayfox@xheotris.zerda.net) |
| 05:21.35 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 05:21.44 | ChannelZ | Spam: "We provide a program that will allow someone with sufficient work experience to obtain a fully verifiable Dgeree: Bachleors, Masetrs or Doctoarte." |
| 05:22.08 | ChannelZ | I'm sure the education is second to none. Dgerees are hard to get! |
| 05:23.58 | kaldemar | ChannelZ: but you're guaranteed to get a good jbo with a dgeree! or at least some temporary wrok! |
| 05:25.55 | ChannelZ | Swaeeet! |
| 05:26.28 | kaldemar | linocisco: dd-wrt uses a package manager called opkg. use "opkg -h" to get help on available commands. dd-wrt has versions of asterisk in their repositories. "opkg update && opkg list" will update the package list and list available packages for you. |
| 05:28.47 | linocisco | kaldemar, so I would need internet for that?? i have no internet on that router |
| 05:29.31 | kaldemar | turns out i was full of crap. i was thinking about openwrt. |
| 05:31.47 | kaldemar | linocisco: http://www.dd-wrt.com/wiki/index.php/Asterisk |
| 05:32.00 | kaldemar | that link was thrown at you yesterday already. |
| 05:32.16 | fling | kaldemar: hello! |
| 05:39.52 | kaldemar | fling: howdy. |
| 06:15.40 | linocisco | kaldemar |
| 06:16.05 | linocisco | kaldemar, I was looking at that link. I dont know which command is to be used to install asterisk which file |
| 06:17.33 | linocisco | kaldemar, I have no usb drive on my router |
| 06:26.30 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:26.30 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:26.32 | schmidts | good morning |
| 06:26.40 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:26.57 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:31.07 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 06:31.43 | v0lZy | lo |
| 06:32.01 | jpsharp | hi |
| 06:35.16 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 06:35.28 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 06:50.49 | EmleyMoor | has set allowguest=no for now at least - never had a legitimate call that way anyway |
| 06:54.44 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:7d6f:fefe:efff:67ff) |
| 06:56.31 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-lwoogbwxdudornsc) |
| 07:08.24 | *** join/#asterisk hebber (~hebber@118.175.66.17) |
| 07:11.53 | linocisco | kaldemar, hello. I dont know how to install asterisk on my dd-wrt using your link |
| 07:12.02 | linocisco | kaldemar, all commands are not working |
| 07:13.19 | *** join/#asterisk bulkorok (~bulkorok@217.110.197.225) |
| 07:13.37 | bulkorok | hi |
| 07:16.35 | kaldemar | linocisco: maybe you should consider using something different from dd-wrt then. check that your router even has the sufficient memory for what you're trying to do. |
| 07:18.57 | linocisco | kaldemar, i m exploring openwrt and freewrt and tomato. but dd-wrt permanently took positon in place. no other firmware will replace dd-wrt using firmware upgrade |
| 07:21.50 | kaldemar | linocisco: what device are you using? |
| 07:24.54 | linocisco | kaldemar, WRT54G2V1 |
| 07:25.39 | kaldemar | where did you come up with that you can't replace dd-wrt with anything else? |
| 07:28.22 | kaldemar | http://www.dd-wrt.com/wiki/index.php/Linksys_WRT54G2 <- for more info, use the proper support channels for dd-wrt. this has really nothing to do with asterisk. |
| 07:39.49 | linocisco | kaldemar, i have installed dd-wrt well |
| 07:40.10 | linocisco | kaldemar, the thing is how to install asterisk on it . guide or reference did n't work |
| 07:41.42 | kaldemar | and why is that? |
| 07:46.03 | *** join/#asterisk Neptu (~Neptu@c-af90e255.113-1-64736c14.cust.bredbandsbolaget.se) |
| 07:54.52 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 07:55.40 | *** join/#asterisk kombi (~kombi@port-92-198-15-96.static.qsc.de) |
| 08:11.00 | linocisco | kaldemar, to install asterisk, what do we do? |
| 08:12.16 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 08:15.06 | kaldemar | linocisco: do what the guide says. |
| 08:15.27 | linocisco | kaldemar, the guide didn't say how to install asterisk |
| 08:16.45 | kaldemar | linocisco: how much available space do you have on the device? |
| 08:17.06 | linocisco | kaldemar, how can I check? |
| 08:18.54 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 08:19.18 | kaldemar | ssh in the device and command "df". the jffs block is what tells you the available space. |
| 08:21.04 | v0lZy | btw |
| 08:21.13 | v0lZy | anyone know how ddrwt compares to say... pfsense? |
| 08:22.49 | ChannelZ | isn't pfsense just a firewall implementation? |
| 08:23.26 | ChannelZ | oh.. it's a whole distro.. |
| 08:23.46 | kaldemar | they have very little in common. |
| 08:24.13 | linocisco | kaldemar, can telnet be ok? or must use ssh? |
| 08:24.49 | kaldemar | linocisco: use telnet if that's what you have. |
| 08:24.59 | v0lZy | i find pfsense pretty awesome, but i run it on a full blown pc |
| 08:24.59 | kaldemar | linocisco: as long as you get console access to it. |
| 08:25.08 | v0lZy | some kind of alternative hardware would be better |
| 08:25.12 | v0lZy | like alix stuff |
| 08:25.13 | dax_roc | pfsense would be more stable, based on PF / FreeBSD |
| 08:25.18 | v0lZy | if for nothign other than power consumption |
| 08:25.27 | v0lZy | but then... if that was 50 bucks not 300 ... |
| 08:25.53 | linocisco | kaldemar, /dev/root 1280(1K blocks) 1280(used) 0(avaliaible) 100(Use%) /(mounted on) |
| 08:26.32 | kaldemar | dax_roc: and will it fit on a SOHO router? |
| 08:26.35 | dax_roc | I have pfsense in production on Alix boards, and Soekris. I would say it's awesome too |
| 08:26.52 | kaldemar | linocisco: is that all it tells you? |
| 08:26.55 | linocisco | v0lZy, why is pfesense so awlsome? what pfsense is related to asterisk? |
| 08:27.00 | linocisco | kaldemar, yes |
| 08:27.22 | v0lZy | linocisco: its awesome because people that develope it really try to make it that wa |
| 08:27.23 | dax_roc | kaldemar: depends, you can convert an old pc to do it |
| 08:27.24 | v0lZy | way* |
| 08:27.37 | v0lZy | as for asterisk, i think i saw a beta package that can install it on pfsense |
| 08:27.37 | kaldemar | dax_roc: so that would be a no? |
| 08:27.46 | kaldemar | linocisco: i don't believe you. |
| 08:27.51 | v0lZy | but its a whoel different thing that a linksys router etc... never mind linocisco |
| 08:28.09 | kaldemar | linocisco: but if it really does, you can stop with the dd-wrt right there. you will not be able to install asterisk on that. |
| 08:28.13 | dax_roc | kaldemar: yes -> http://www.ebay.com/sch/i.html?_trksid=p5197.m570.l1313&_nkw=alix&_sacat=0 |
| 08:29.19 | linocisco | kaldemar, ls shows there is folder called jffs. |
| 08:29.21 | dax_roc | kaldemar: It wont run on off the shelf routers from <insert brand> tho |
| 08:29.47 | kaldemar | dax_roc: that's what i was after. no sense comparing the two really. |
| 08:29.50 | v0lZy | yeah, wish it did. |
| 08:30.00 | v0lZy | putting it ona linksys would be awesome i think |
| 08:30.08 | v0lZy | alas, no such option |
| 08:30.50 | linocisco | kaldemar, so should I try openwrt ? |
| 08:30.51 | dax_roc | kaldemar: if you want a full featured scalable firewall os use pfSense if you want a hackable router os use ddwrt / openwrt |
| 08:30.52 | kaldemar | linocisco: irrelevant. if it's just a folder, it has no use. |
| 08:31.14 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 08:31.29 | kaldemar | dax_roc: i'm happy without any of those. :) |
| 08:31.40 | v0lZy | i recommend pfsense too |
| 08:31.43 | dax_roc | kaldemar: sorted :D |
| 08:31.48 | v0lZy | its really good. |
| 08:32.06 | v0lZy | <- fan |
| 08:32.23 | kaldemar | dax_roc: though i do have devices running openrwt and dd-wrt. |
| 08:32.47 | v0lZy | blasphemy! |
| 08:32.49 | v0lZy | :S |
| 08:32.55 | v0lZy | (kidding) |
| 08:33.15 | v0lZy | whats openrwt compared to dd-wrt? |
| 08:34.20 | linocisco | kaldemar, I found http://downloads.openwrt.org/sources. I saw many. what to choose? |
| 08:34.23 | dax_roc | kaldemar: there interface is much cleaner and imo is better for joe-user. I need some of the features that are only available in pfsense or costly alternatives. |
| 08:34.39 | dax_roc | *the ddwrt interface. |
| 08:35.05 | dax_roc | dd-wrt is based on open-wrt |
| 08:35.11 | kaldemar | linocisco: first find out if your WRT54G2 is even supported. stop random googling and actually read the openwrt documentation. |
| 08:35.22 | v0lZy | i like the options pfsense gives you... tons of stuff |
| 08:35.38 | kaldemar | linocisco: and keep away from those sources at this point, you'd need to setup a build environment for openwrt first. |
| 08:36.04 | kaldemar | dax_roc: interface as in web interface? |
| 08:37.02 | mariusno | linocisco: if I am correct, you can't install asterisk on any of those. The wrt54 has to litle internal memmory. You will need to use the image-generator to customize the packages to be installed in the image. This is kind of a complex operation, and you should have some linux and googleling experience first. |
| 08:37.09 | dax_roc | kaldemar: Yes |
| 08:37.38 | kaldemar | mariusno: depends on the HW version. i've personally run asterisk on an older WRT54G. |
| 08:38.03 | dax_roc | kaldemar: I have yet to see a better UI for a firewall. |
| 08:38.12 | kaldemar | dax_roc: i don't even use a web interface on my openwrt. |
| 08:38.16 | mariusno | kaldemar: with one of the pre-build images? |
| 08:38.22 | kaldemar | mariusno: yes. |
| 08:39.07 | mariusno | I could not even install openssl before it was out of internal memory, I had to use the image-generator. But as I said, i could be wrong:) |
| 08:39.09 | kaldemar | mariusno: some version of whiterussian, iirc. |
| 08:43.34 | mariusno | kaldemar: okay:) |
| 08:44.24 | mariusno | but running asterisk on wrt54 hardware, i thought there would be performance issues, or is it no problem? |
| 08:44.48 | kaldemar | mariusno: depends on what you want to do with it. |
| 08:44.50 | v0lZy | btw... virtualization and asterisk... bad idea? |
| 08:45.23 | mariusno | worked good for me with virtualbox |
| 08:47.44 | *** join/#asterisk _zoom_ (~Eissa@196.1.219.122) |
| 08:57.53 | v0lZy | i kind of like virtualbox |
| 08:58.06 | v0lZy | but it stinks of oracle a bit |
| 08:59.37 | mariusno | I started of with virtualbox, migrated to bare hardware, and will soon try to migrate to KVM |
| 09:04.18 | *** join/#asterisk orn (~orn@2a01:8280:10:4:44b:f353:b5f8:ffd) |
| 09:06.25 | v0lZy | kvm seems interesting |
| 09:06.29 | v0lZy | but qemu.. ergh |
| 09:06.45 | mariusno | libvirt takes care of the machines |
| 09:17.09 | *** join/#asterisk danfromuk (~IceChat77@2.30.230.183) |
| 09:17.38 | danfromuk | Hi, has anyone managed to connect a cisco phone thats behind a nat, to a remote asterisk box? |
| 09:18.56 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 09:19.09 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 09:26.40 | Roelt | danfromuk, yes |
| 09:27.53 | danfromuk | Ok, i'll give it another bash. Last time i tried, the cisco phone wasn't providing the correct port information. cant remember exactly what happened. |
| 09:28.30 | *** join/#asterisk Tim_Toady (~fuzzy@77.49.3.242.dsl.dyn.forthnet.gr) |
| 09:30.09 | dax_roc | Does the hangup stop execution in the dialplan ? |
| 09:31.32 | dax_roc | I'm trying to do a callback after the hangup(), How should I do this. Do I need a script checking for callbacks to be made |
| 09:35.12 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 09:36.07 | danfromuk | dax_roc: http://www.voip-info.org/wiki/view/Asterisk+h+extension |
| 09:36.53 | WIMPy | Before the hangup and yes Asterisk really needs some concept of disconnected calls. |
| 09:41.16 | v0lZy | dax_roc: world of pain :D |
| 09:41.34 | v0lZy | its the dial thats the problem |
| 09:41.47 | v0lZy | theres flags g and f (or F, not sure) |
| 09:42.02 | v0lZy | anyway, its supposed to continue the dialplan execution |
| 09:42.03 | dax_roc | so before hangup i would dial from exten h ? |
| 09:42.03 | v0lZy | but |
| 09:42.08 | v0lZy | depends on who hangs up. |
| 09:42.35 | dax_roc | I don't answer the call, I'm just getting the callerid hangup callback and place in queue |
| 09:43.17 | v0lZy | perhaps i can help here |
| 09:43.19 | v0lZy | sec |
| 09:46.23 | v0lZy | a while people here helped me come up with a solution |
| 09:46.36 | v0lZy | where as long as my phone is on the line, its calls are redirected |
| 09:46.42 | v0lZy | when it hangsup, the redirect terminates |
| 09:46.55 | v0lZy | its done in such a way that its sent to loop in some context |
| 09:47.07 | v0lZy | and then i have something written in my s extenison... |
| 09:47.34 | v0lZy | exten => s,1,Wait(2) |
| 09:47.35 | v0lZy | exten => s,n,SayDigits(${REDIRECTED_TO}) |
| 09:47.37 | v0lZy | exten => s,n,Playback(vm-saved) |
| 09:47.39 | v0lZy | exten => s,n(waitloop),Wait(10) |
| 09:47.40 | v0lZy | exten => s,n,SayDigits(${REDIRECTED_TO}) |
| 09:47.42 | v0lZy | exten => s,n,Playback(vm-saved) |
| 09:47.43 | v0lZy | exten => s,n,Goto(waitloop) |
| 09:47.45 | v0lZy | exten => h,1,Noop(Removed ${DB_DELETE(CF/${CALLERID(num)})})})})}) |
| 09:47.56 | kaldemar | v0lZy: you should know not to paste here. |
| 09:48.06 | v0lZy | Sorry... i thought if its just a little that i can. |
| 09:48.17 | v0lZy | though yeah, i don tsee others do it much.. |
| 09:48.36 | v0lZy | anyway... you see the h,1, there.. |
| 09:48.39 | v0lZy | that happens on hangup |
| 09:49.59 | v0lZy | dax_roc: maybe u can use that... |
| 09:54.12 | *** join/#asterisk xoveruk (~rich@196.200.85.125) |
| 09:55.16 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 09:56.56 | kaldemar | dax_roc: you could set a variable in the extension that you want to trigger the callback and then execute the callback in the hangup extension if the variable is set. |
| 09:57.32 | xoveruk | is there a command that will allow me to monitor SIP and IAX connections in realtime, similar to the watch command in BASH? |
| 09:58.53 | li3p | asterisk -rx 'any command' ))) |
| 09:59.30 | kaldemar | xoveruk: monitor how? show channels go up and down or show signalling traces? |
| 10:01.29 | salz212 | how to make asterisk work on multi core.. or does it ... by default works on multicores? |
| 10:01.49 | kaldemar | salz212: it works. |
| 10:02.18 | salz212 | so is the load.. equally divided? lets say I hae 8 core processor.. |
| 10:06.32 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 10:06.51 | kaldemar | salz212: wouldn't guarantee anything about load equality, but it will use multiple cores. |
| 10:07.00 | xoveruk | both, but what is the latter? |
| 10:07.18 | xoveruk | what is the command for signal tracing? |
| 10:08.28 | kaldemar | xoveruk: "sip set debug on", "iax2 set debug on" |
| 10:09.51 | kaldemar | xoveruk: for channels, enable verbosity in the CLI or use something like: watch -n 1 'asterisk -rx "core show channels"' |
| 10:10.16 | li3p | xoveruk, tcpdump -v |
| 10:10.58 | xoveruk | ah great. |
| 10:11.42 | xoveruk | I did not realise that i could execute commands only. brilliant |
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| 10:24.35 | xoveruk | how can i save that to an alias? |
| 10:51.54 | xoveruk | sorry wrong channel. |
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| 11:58.52 | bulkorok | how can I reduce the volume of MOH !? |
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| 12:32.10 | leifmadsen | bulkorok: use the VOLUME() function? |
| 12:32.17 | leifmadsen | or you can modify the sound files with sox or something else |
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| 13:00.10 | bulkorok | leifmadsen: what are the possible values!? -8 to 8 or 0 to 9 ?! |
| 13:00.14 | *** join/#asterisk orn (~orn@2a01:8280:10:4:7126:181d:5bb7:5e4a) |
| 13:00.21 | leifmadsen | unknown |
| 13:00.25 | leifmadsen | you'll have to test and find out |
| 13:00.37 | leifmadsen | and look at the documentation for VOLUME() |
| 13:03.00 | bulkorok | I took a look... there is only 3 or 2 mentioned... but I tried and can hear that VOLUME(TX)=5 is mouch louder than -6 so -6 to 5 is verified ;-) |
| 13:03.16 | leifmadsen | well that makes perfect sense... |
| 13:03.20 | leifmadsen | the values are DB I think |
| 13:03.27 | bulkorok | k |
| 13:03.42 | leifmadsen | which the docs I'm pretty sure mention |
| 13:04.19 | bulkorok | core show function VOLUME says only rx/tx gain |
| 13:04.56 | bulkorok | doesn't like the standard moh anymore :-/ |
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| 13:05.06 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:06.16 | leifmadsen | bulkorok: good thing you can change it then |
| 13:06.53 | bulkorok | :) |
| 13:09.29 | bulkorok | wiki says gain uses dB... |
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| 13:13.40 | *** join/#asterisk gusto (~gusto@ppp-62-216-201-5.dynamic.mnet-online.de) |
| 13:19.31 | Katty | hello my asterisk does not work at all how to fix plz?? answer plz. |
| 13:20.19 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 13:21.59 | drmessano | Katty, Start > Shutdown > Reboot |
| 13:22.10 | Katty | heard it. |
| 13:22.13 | Katty | next! |
| 13:22.20 | drmessano | Hmmm |
| 13:23.08 | drmessano | Remove the outer wrapper, place Asterisk at opening. Slide Asterisk into opening, gently remove applicator. |
| 13:23.31 | Katty | ^_- |
| 13:23.35 | Katty | well that's certainly new. |
| 13:24.47 | drmessano | It was a simple substitution based on a process familiar to your gender. I would say it's "amateurish" and/or "sexist" at best. That's all I have this early. |
| 13:25.47 | Katty | hands drmessano coffee |
| 13:27.08 | drmessano | Why, thank you! |
| 13:28.20 | drmessano | I don't want to go to work. Yesterday was bad and those people make me sadface |
| 13:28.38 | [TK]D-Fender | A friend in need's a friend indeed. A friend with beans is better.... |
| 13:29.31 | drmessano | Coffee, the second best thing to come out of Colombia |
| 13:30.16 | *** join/#asterisk serafie (~erin@nat/digium/x-uhujccfdnlohslea) |
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| 13:46.18 | *** join/#asterisk bchia (~Adium@nat/digium/x-yajxpyzdzxeccdnm) |
| 13:47.51 | mariusno | Hi. I have 4 phones in a ring group. If I receive the incomming call on one of them, the other 3 displayes a "missing" call on the screen. The call is not missing, just taken by someone else. Any neat trick to solve this? |
| 13:48.22 | WIMPy | Get decent phones. |
| 13:48.35 | mariusno | cisco 7912 phones |
| 13:49.58 | WIMPy | Asterisk will tell the phones that the call was answered elsewehere. Whether the phone case is up to the phone. |
| 13:50.42 | mariusno | ah, so no specific settings to asterisk then |
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| 13:51.39 | WIMPy | Nope. It just works. Or at least it has been working for quite some time. |
| 13:51.47 | mariusno | tried to add this "c" to "Asterisk Dial command" |
| 13:52.05 | mariusno | which i found after some googeling, but it did not work |
| 13:52.09 | *** join/#asterisk gusto (~gusto@ppp-62-216-201-5.dynamic.mnet-online.de) |
| 13:52.10 | mariusno | so i thought to ask here |
| 13:52.44 | WIMPy | Your phones seem to ignore it. |
| 13:53.06 | WIMPy | Err. |
| 13:53.13 | WIMPy | You are using SIP firmware? |
| 13:53.19 | mariusno | on the cisco, yes |
| 13:53.27 | WIMPy | ok |
| 13:54.48 | *** join/#asterisk gusto (~gusto@ppp-62-216-201-5.dynamic.mnet-online.de) |
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| 13:56.11 | hebber | Hi, when asterisk just stops working what would be the first thing to do? |
| 13:56.25 | WIMPy | restart it |
| 13:56.37 | hebber | I have restarted, reinstalled |
| 13:56.41 | WIMPy | And look for a core dump to find out what happened. |
| 13:56.51 | WIMPy | ~collectdebug |
| 13:56.51 | infobot | well, collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 13:58.26 | hebber | hmm |
| 13:58.43 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 13:58.43 | [TK]D-Fender | First thing is to actually look at what you're using ... and tell us before asking... |
| 13:58.55 | mariusno | WIMPy: thank you for the answer, then I will not follow up on the issue anymore:) |
| 14:04.14 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
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| 14:06.21 | *** join/#asterisk slidesinger-lt (~jtatum@c-174-57-5-70.hsd1.nj.comcast.net) |
| 14:08.01 | *** join/#asterisk Lann (47162f0c@gateway/web/freenode/ip.71.22.47.12) |
| 14:08.07 | Lann | Hello |
| 14:08.09 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
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| 14:10.41 | hebber | Fender: I'm not able to reproduce the issue as described in Collectdebug - Asterisk 1.8 only respond to open console, nothing else |
| 14:11.40 | Lann | Could anyone reccomend the most accurate voice recognition engine for asterisk? |
| 14:14.19 | *** join/#asterisk coppice (~chatzilla@14.140.218.146) |
| 14:16.23 | Lann | Second question: could somebody point me in the right direction to perform multiple simultaneous conference calls for a single user? |
| 14:17.25 | WIMPy | hebber: Try to describe what's happening. |
| 14:20.38 | hebber | Wimpy: I use asterisk 1.8, been pretty smooth setup until today. Was just trying to get parking to work, so did some changes in features.conf. Then I restarted asterisk using service and CLI:core reload didn't work. Then my SIP got disconnected. Checked CLI and all commands are renderes unusable. SIP, dahdi etc. Restarting the server solves nothing, but I can still log into the CLI. I have reinstalled from scratch and everything remains the same. |
| 14:21.31 | *** join/#asterisk atekinalp (~alper@88.247.181.163) |
| 14:21.40 | mjordan | hebber: what version of 1.8? |
| 14:22.05 | hebber | I just downloaded the latest branch from SVN |
| 14:22.20 | WIMPy | If the last change was to features.conf, you should try to move that out of the way. |
| 14:22.42 | hebber | I agree, but that doesn't work either |
| 14:24.06 | WIMPy | Then you should try to move them one by one or disable autoloading of modules and load them one by one until you find out which one causes the trouble. |
| 14:24.37 | *** join/#asterisk dax_roc (~dax_roc@pdpc/supporter/bronze/daxroc) |
| 14:25.19 | hebber | Thanks for advice Wimpy |
| 14:25.33 | dax_roc | How would I go about creating an outbound call and on progress / answer transfer to an internal queue? |
| 14:26.01 | dax_roc | originate and use the queue as the destination app ? |
| 14:26.30 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-dxslpxubtzalhcib) |
| 14:27.38 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:28.14 | Lann | anyone? is there a way to put a user in a conference but also continue to do voice recog on him privately? |
| 14:28.21 | Lann | simultaneously |
| 14:29.08 | WIMPy | dax_roc: Sounds logical. |
| 14:31.17 | atekinalp | hi my asterisk server doesn't create answer to register request. what could be the problem? |
| 14:31.29 | Katty | checks CAH |
| 14:31.51 | Katty | atekinalp: Global Warming |
| 14:31.59 | Katty | infobot: forget CAH |
| 14:32.00 | infobot | i forgot cah, Katty |
| 14:32.26 | Katty | Qwell: does a full string of text go in quotes? |
| 14:32.30 | Katty | Qwell: to infobot |
| 14:32.32 | WIMPy | atekinalp: Are you sure it receives the request? |
| 14:32.48 | Katty | Qwell: "foo bar" is |
| 14:35.34 | *** join/#asterisk atekinalp (~alper@88.247.181.163) |
| 14:36.07 | tzafrir_laptop | notes that the Babbage difference engine used decimal digits rather than binary digits |
| 14:36.48 | atekinalp | WIMPy: when i use tcpdump to listen traffic i see register packets but when i use 'sip set debug on' i can't see anything |
| 14:37.31 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 14:37.56 | WIMPy | Did you bind to a specific address? |
| 14:38.06 | coppice | tzafrir_laptop: lots of computers in the 60s used a variety of non-binary number schemes |
| 14:41.24 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 14:41.46 | [TK]D-Fender | lann: Originate a local channel to chanspy on him |
| 14:42.24 | Lann | that won't work in this case |
| 14:42.25 | [TK]D-Fender | atekinalp, pastebin BOTH for us to see... |
| 14:42.25 | [TK]D-Fender | ~pb |
| 14:42.26 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 14:42.27 | [TK]D-Fender | ^^^ |
| 14:42.33 | [TK]D-Fender | lann: ... because? |
| 14:42.42 | Lann | Heh *deep breath* |
| 14:43.10 | Lann | Well it's basically a game where the user can be in a chat room |
| 14:43.16 | Lann | but there is also a command channel |
| 14:43.22 | Lann | the user can switch between command mode and chat mode |
| 14:43.25 | atekinalp | <WIMPy>: when i netstat i can see asterisk is listenin port 5060 |
| 14:43.57 | Lann | I need the user to always be in command mode even when chatting because it is looking for something specific to decide when to enter command mode |
| 14:44.10 | Lann | err well, to always be running sphinx on the user's voice |
| 14:44.22 | [TK]D-Fender | Lann: so... this interferes with my suggestion how exactly? |
| 14:44.43 | Lann | Well, I need the user to be in some place that I can run speech synthesis on him, and also be in a chat |
| 14:44.51 | Lann | is that possible? |
| 14:45.03 | Lann | by chat I mean conference |
| 14:45.14 | [TK]D-Fender | Lann: Originate teh Chanspy channel to run the Sphinx in the background and dump him into the conference. The End. |
| 14:45.27 | Lann | ah, thanks |
| 14:45.43 | [TK]D-Fender | Lann: Still seems to apply just fine |
| 14:45.46 | Lann | can i play background audio to the user via chanspy? |
| 14:46.00 | [TK]D-Fender | "core show application chanspy" <- |
| 14:46.21 | Lann | ...not sure what you mean |
| 14:47.25 | WIMPy | atekinalp: On which address? |
| 14:49.00 | [TK]D-Fender | Lann: Means "read the apps instructions" |
| 14:49.39 | Lann | I have before coming here |
| 14:49.45 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-mcdaowjvmqtivxck) |
| 14:49.45 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:51.48 | [TK]D-Fender | Lann: and what does it say? |
| 14:51.55 | Lann | Ah, it seems that chanspy in whisper mode can play background audio if i'm not mistaken |
| 14:56.28 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 14:58.04 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 14:58.13 | dax_roc | What's the correct way to initiate a call from the cli with originate. This works for internal extensions but not a pstn number "channel originate <number> application queue MyCallQueue" |
| 14:58.44 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-rgqjvofoniehiynl) |
| 14:58.44 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:00.37 | Katty | hi jordan |
| 15:00.45 | Katty | mjordan. |
| 15:02.09 | mjordan | hello Katty |
| 15:02.18 | Katty | how'rechu |
| 15:02.49 | mjordan | good! making ... things. |
| 15:03.20 | Katty | making things without scowly face, i hope |
| 15:03.37 | mjordan | just enough scowly face to be appropriate |
| 15:04.40 | Katty | ^_^ |
| 15:08.27 | WIMPy | dax_roc: Channel, not number. |
| 15:09.10 | [TK]D-Fender | dax_roc, "channel originate <number> application queue MyCallQueue" <- that is not what the instructions say |
| 15:09.48 | dax_roc | mybad, got it |
| 15:18.21 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:28.19 | dax_roc | Can you use originate from h extention? |
| 15:29.58 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 15:31.40 | dax_roc | I don't get why the skype originate fails here, http://pastebin.ca/2176175 |
| 15:32.13 | Qwell | Why are you using quotes? |
| 15:32.38 | dax_roc | sorry just tested them |
| 15:32.55 | Qwell | show us the failure, and the real dialplan |
| 15:34.38 | dax_roc | http://pastebin.ca/2176179 |
| 15:35.38 | Qwell | You aren't even getting to the originate. |
| 15:36.35 | Katty | infobot: cah |
| 15:36.35 | infobot | extra, extra, read all about it, cah is http://cah.kired.net:81/ |
| 15:39.49 | [TK]D-Fender | dax_roc, Stop shoving quotes everywhere |
| 15:40.03 | [TK]D-Fender | dax_roc, * is has no data-types |
| 15:40.58 | dax_roc | habit .. |
| 15:41.11 | [TK]D-Fender | dax_roc, and you are not permitted to use "wait" in "h" |
| 15:41.22 | dax_roc | oh |
| 15:42.04 | [TK]D-Fender | dax_roc, If you want a delay then originate a Local channel that will wait before dialing out. |
| 15:42.52 | dax_roc | Yeah, I need the wait as the call hasn't hung up on the recieving end when it tries to call |
| 15:52.34 | *** join/#asterisk r00t-Ed (~no@74.42.252.2) |
| 16:12.54 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:596f:7ffe:e7fe:4fff) |
| 16:17.14 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 16:31.16 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 16:33.08 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-dumvcfjaopyoltuv) |
| 16:33.58 | Qwell | malcolmd: I hadn't seen this. Neat. http://blogs.digium.com/2012/07/30/digium-phones-and-asterisk-whats-new/ |
| 16:37.27 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 16:38.57 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 16:41.34 | *** join/#asterisk navaismo (~navaismo@189.191.27.69) |
| 16:41.40 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 16:42.22 | dax_roc | [TK]D-Fender: Can I do a originate in a non-blocking way |
| 16:42.44 | [TK]D-Fender | dax_roc, AMI, CLI, or Call Files instead |
| 16:46.32 | *** join/#asterisk uskerine (~Uske@231.Red-79-157-169.dynamicIP.rima-tde.net) |
| 16:46.33 | uskerine | hi |
| 16:46.36 | uskerine | Specify the channel definitions. The format is: |
| 16:46.36 | uskerine | <device> = <channel list> |
| 16:46.50 | uskerine | hi, i am trying to configure a ISDN PRI card (Digium TE121) |
| 16:47.05 | uskerine | the pdf from DIGIUM provides you channel definition for T1 |
| 16:47.12 | uskerine | bchan = 1-23 |
| 16:47.15 | uskerine | dchan =24 |
| 16:47.22 | uskerine | what should be used for ISDN PRI? |
| 16:47.34 | pigpen | seems digium listened to some of my comments on this last release. (well, contributed to) |
| 16:47.38 | [TK]D-Fender | that IS for a T1 ISDN PRI |
| 16:47.39 | uskerine | bchan = 1-15,17-31 |
| 16:47.44 | pigpen | hopefuly the jabra ehs is next |
| 16:47.45 | uskerine | and dchan= 16? |
| 16:47.57 | uskerine | i mean E1 ISDN PRI |
| 16:47.58 | [TK]D-Fender | uskerine, your 2nd list is for an E1 PRI |
| 16:48.03 | uskerine | ok thanks |
| 16:49.39 | Katty | infobot: cah |
| 16:49.39 | infobot | well, cah is http://cah.kired.net:81/ |
| 16:49.54 | navaismo | Using DPMA the Phones cant download the firmware says "could not be reached", the path exist, the firmware files exist the firmware.conf exist |
| 16:50.32 | Qwell | navaismo: It fetches the firmware files via http. Do you have that part setup? |
| 16:51.03 | Qwell | see file_url_prefix= |
| 16:51.21 | navaismo | you mean if I have http running? If so, yes its running and I can enter via web to http:/10.100.210.10/digium_phones_firmware |
| 16:51.26 | uskerine | [TK]D-Fender, I have T121 with echo cancelling, what should I put in /etc/dahdi/system.conf for "echocanceller"? |
| 16:51.34 | navaismo | the last url is the file_url_prefix |
| 16:51.44 | [TK]D-Fender | usemg2. |
| 16:51.49 | uskerine | what is mg2, kb1, sec2 and sec? |
| 16:51.49 | fling | Hello! :p Where should I read about Call Deflection? |
| 16:51.50 | Qwell | navaismo: What is the full path to one of the .eff files? |
| 16:52.01 | [TK]D-Fender | uskerine, mg2. HWEC will take precedence anyway |
| 16:52.19 | [TK]D-Fender | uskerine, those are all SOFTWARE EC. And won't functioally matter |
| 16:52.20 | uskerine | i want HW EC |
| 16:52.27 | [TK]D-Fender | it's AUTOMATIC |
| 16:52.28 | uskerine | ok |
| 16:52.29 | [TK]D-Fender | ^ |
| 16:52.30 | uskerine | thanks |
| 16:52.37 | navaismo | Qwell, is /var/www/html/digium_phones_firmware/1_0_5_1_47270_D40_firmware.eff |
| 16:52.39 | uskerine | so if i don't even add echocanceller line to system.conf |
| 16:52.42 | uskerine | is that correct? |
| 16:52.45 | Qwell | navaismo: sorry, full URL |
| 16:52.49 | [TK]D-Fender | fling, Define your use of the term more clearly and we'll see... |
| 16:53.53 | navaismo | Qwell, http://10.100.210.10/digium_phones_firmware/1_0_5_1_47270_D40_firmware.eff |
| 16:54.05 | fling | [TK]D-Fender: I have four phone lines (with four separate phone numbers) over ip from my isp |
| 16:54.16 | navaismo | Qwell, if i click on it the browser start the download |
| 16:54.21 | Qwell | navaismo: And digium_phones show firmware has that as the path/filename? |
| 16:54.30 | Qwell | err, not path |
| 16:54.32 | [TK]D-Fender | fling, First don't call them "lines".... |
| 16:55.01 | Qwell | I forget what older versions of DPMA showed there.. |
| 16:55.12 | fling | [TK]D-Fender: I want one of the numbers to be the _master_ number; so if someone calls, call deflected to another number if possible |
| 16:55.13 | [TK]D-Fender | fling, DID's, and what pool of channels they permit to be used between them are 2 very different things |
| 16:55.42 | [TK]D-Fender | fling, well yuo should SEE what # the call is coming in on. Just dial out. |
| 16:57.12 | fling | [TK]D-Fender: umm? I want to make this master number to not to be busy most of the time |
| 16:57.26 | *** join/#asterisk gusto (~gusto@2001:470:6d:213::42:4) |
| 16:57.33 | navaismo | Qwell, nope, only the name of the firmware |
| 16:57.49 | [TK]D-Fender | fling, again you have not clearly defined how channels are ALLOCATED to your DID's |
| 16:58.09 | Qwell | navaismo: what version of DPMA? 1.2, I would hope |
| 16:58.20 | [TK]D-Fender | fling, I can support 24 23 channels any which way I want ... amongst my 102 DIDs <---- |
| 16:58.25 | [TK]D-Fender | 23* |
| 16:59.30 | navaismo | Qwell, doing a digium_phones show version it show "Digium Phone Module for Asterisk Version 1.8.11_1.0.2" |
| 16:59.57 | Qwell | Then the option is not file_url_prefix |
| 17:00.29 | uskerine | without the ISDN PRI E1 actually connected to the board, should I see the TE121 as GREEN in dahdi_tool? |
| 17:00.47 | fling | [TK]D-Fender: sorry; is it enough to show you my config files? > http://dpaste.com/778771/ |
| 17:00.55 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:01.10 | navaismo | Qwell, weird it was working until I changed to the production enviroment, in home it work, but let me see what is the right setting and will change it. |
| 17:03.31 | WIMPy | uskerine: No. Only if it works. |
| 17:04.39 | uskerine | thanks WIMPy, so without having the actual E1 ISDN PRI provisioned, as far as i see everything ok in dmesg and dahdi_tool shows the 30 channels with the RED Alarm, should I assume that it has been properly pre-configured? |
| 17:05.08 | WIMPy | Possibly |
| 17:05.11 | uskerine | ok thanks |
| 17:05.32 | WIMPy | You can try a loopback, if you have one handy. |
| 17:06.03 | WIMPy | But the line state doesn't mean that much anyway. |
| 17:06.56 | navaismo | Qwell, i have this settings in the res_digium_phone.conf: firmware_url_prefix=http:/10.100.210.10/digium_phones_firmware, firmware_package_directory=/var/www/html/digium_phones_firmware and file_directory=/var/lib/asterisk/digium_phones |
| 17:07.07 | navaismo | and it match with the sample configuration given in the tarball |
| 17:07.15 | fling | [TK]D-Fender: DID is the inward number, right? I have it in extensions.conf |
| 17:07.20 | [TK]D-Fender | fling, Your configs tell me nothing of what actual services you are paying for or how they allocate channels to you |
| 17:07.45 | [TK]D-Fender | fling, DID's have nothing to do with how many channels are allocated to each or if they share a POOL of channels |
| 17:08.06 | Qwell | navaismo: and digium_phones show firmware shows the right firmware version? |
| 17:08.25 | Qwell | version/filename, as it were |
| 17:09.58 | uskerine | WIMPy, board came with a lopback RJ45 |
| 17:10.01 | navaismo | Qwell, i think so, I downloaded this: res_digium_phone-1.8.11_1.0.2-x86_64.tar.gz and show version show: Digium Phone Module for Asterisk Version 1.8.11_1.0.2 |
| 17:10.05 | uskerine | can i use network UTP cable? |
| 17:10.15 | Qwell | navaismo: firmware version, not DPMA version |
| 17:10.34 | navaismo | sorry |
| 17:11.08 | *** join/#asterisk slidesinger-lt (~jtatum@c-174-57-5-70.hsd1.nj.comcast.net) |
| 17:11.22 | fling | [TK]D-Fender: each did is using it's own channel, I have four channels and four dids if I understand you right :P |
| 17:11.46 | [TK]D-Fender | fling, More like if you understand what you PAID FOR right |
| 17:12.07 | navaismo | files are: [D40] version=1_0_5_1_47270 file=1_0_5_1_47270_D40_firmware.eff |
| 17:12.16 | [TK]D-Fender | fling, You do not been able to definitively answer this |
| 17:12.34 | Qwell | navaismo: digium_phones show firmware |
| 17:12.53 | Qwell | navaismo: pastebin the output of that |
| 17:12.57 | navaismo | jesus why im so stupid some times |
| 17:14.13 | navaismo | Qwell, here is the PB http://pastebin.com/5CX5UN1E |
| 17:14.50 | malcolmd | Qwell: :D it's basically a rehash of what i gave to marketing for them to e-mail to people that's registered dpma keys, so that they know there's new stuff. |
| 17:14.51 | Qwell | much better. And you're certain your phones can access that address? |
| 17:15.02 | fling | [TK]D-Fender: my isp is not providing any info on the things I pay for |
| 17:15.13 | Qwell | malcolmd: I probably aught to actually register myself a key, so I get that stuff. |
| 17:15.45 | malcolmd | i've got a registered key and i haven't gotten the e-mail yet :( |
| 17:15.50 | fling | [TK]D-Fender: I have spent few weeks to figure it out how to connect to their server |
| 17:16.20 | navaismo | Qwell, nope im not sure I guess the phones can if my PC can. Same LAN same switch |
| 17:16.37 | navaismo | but now you give another clue to look at the problem |
| 17:16.46 | navaismo | lets dig |
| 17:17.25 | bobb_WU | can someone help me troubleshoot a SIP problem? The connection to my voice mail server starts but it takes audio 15-20 seconds to start streaming (thereby missing the greeting or the opportunity to enter one's own extension) |
| 17:17.45 | bobb_WU | i have sip debug output and can describe the system as needed |
| 17:20.00 | [TK]D-Fender | .... |
| 17:20.19 | fling | [TK]D-Fender: hehe :p |
| 17:20.33 | [TK]D-Fender | moves on to more productive matters |
| 17:26.44 | fling | [TK]D-Fender: what info do I need? I can ast ISP guys |
| 17:26.45 | Katty | liek lunch! |
| 17:27.40 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 17:28.53 | jeffspeff | I'm trying to implement free fax for asterisk (FFA) and am having trouble locating current documentation. I'm wanting the faxes to go straight to regular old fax machines, I already have the Linksys PAP devices in place... |
| 17:33.59 | [TK]D-Fender | jeffspeff, FFA isn't for sitting between your PAP & anything else. |
| 17:34.12 | navaismo | jeffspeff, I think FFA create the receive the fax and create the tiff file |
| 17:34.31 | navaismo | damm again too late |
| 17:34.46 | navaismo | bobb_WU, PB your issues |
| 17:34.54 | navaismo | ~pb |
| 17:34.55 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 17:35.47 | jeffspeff | so, FFA recieves the fax, saves it as a .tff file in a temp directory then takes the .tif file and sends it to the fax machine? |
| 17:36.01 | bobb_WU | http://pastebin.com/kXrPBCtD |
| 17:36.15 | bobb_WU | that's the sip debug output and other related dial plan stuff from the CLI |
| 17:36.28 | navaismo | jeffspeff, nope save the tiff in the directory the you can send to an email or download it |
| 17:36.52 | jeffspeff | that's not what i'm trying to do |
| 17:36.59 | bobb_WU | the .110 address is our relay box, its the central server that passes between our Mitel 3300 controller and the rest of our Asterisk boxes |
| 17:37.19 | jeffspeff | in a corporate environment i can't have every fax go to everybody in this office |
| 17:37.52 | *** join/#asterisk coppice (~chatzilla@14.140.218.146) |
| 17:38.00 | jeffspeff | they had this setup on a freepbx box previously, but i'm not sure about all the under-the-hood stuff |
| 17:38.05 | jeffspeff | [TK]D-Fender, any ideas? |
| 17:38.18 | [TK]D-Fender | Get an ATA that actually supports T.38 |
| 17:38.23 | [TK]D-Fender | And FFA has nothing to do with this. |
| 17:38.52 | jeffspeff | [TK]D-Fender, ok... what's FFA for then? My ATA has fax settings |
| 17:39.09 | jeffspeff | FAX CED Detect Enable: |
| 17:39.32 | [TK]D-Fender | PAP2 does not support T.38 as per numerous citings including coppice who is our resident authority. |
| 17:40.00 | jeffspeff | Ok, how can i get this work over SIP/ulaw then? |
| 17:40.36 | navaismo | jeffspeff, read the brief of FFA here http://store.digium.com/productview.php?product_code=804-00007 |
| 17:41.22 | jeffspeff | ok, i knew that part |
| 17:41.42 | jeffspeff | can the .tif image not then be sent to a fax machine? |
| 17:44.56 | Katty | http://cah.kired.net:81/1.html?q=10&a=423 |
| 17:45.13 | *** part/#asterisk pgrace (pgrace@vsix.me) |
| 17:45.17 | jaytee | PAP2 doesn't support T.38 but the Linksys SPA-2102 does. |
| 17:45.54 | jeffspeff | ok, forget T.38 |
| 17:46.13 | jeffspeff | this setup was recently working with a standard fax machine, a PAP2 and freepbx |
| 17:46.31 | jeffspeff | i'm migrating away from the freepbx box and am trying to figure out how to implement this in my own dialplan |
| 17:48.35 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 17:50.54 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 17:51.00 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 17:51.18 | bobb_WU | any advice on my delay issue? |
| 17:52.18 | Katty | infobot: forget CAH |
| 17:52.18 | infobot | i forgot cah, Katty |
| 17:52.35 | Katty | infobot: CAH is Cards Against Humanity! http://cah.kired.net:81/1.html?q=10 |
| 17:52.35 | infobot | Katty: okay |
| 17:53.06 | navaismo | bobb_WU, without the timestamps i can't see the delay but maybe the delay is in the audio itself. Is the default audio or custom audio? |
| 17:53.13 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 17:53.14 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:53.23 | bobb_WU | we run ulaw, is that what you mean? |
| 17:54.15 | bmoraca_work | if I see a message that says "disconnecting call '<channelname>' for lack of RTP activity in 31 seconds", does that mean that the lack of RTP was coming from the named channel, or could it mean that the lack of RTP was from the channel to which the named channel is bridged? |
| 17:54.22 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 17:54.32 | navaismo | Did you uploaded this audio or you use the default audio --> /var/spool/asterisk/voicemail/winthrop/4558/unavail.slin' (language 'en') |
| 17:54.39 | navaismo | bobb_WU, ^ |
| 17:54.53 | bobb_WU | ah that is custom for each user |
| 17:55.04 | bobb_WU | they are able to set that through the voicemail prompts |
| 17:55.38 | bobb_WU | ok the interesting part about this problem is its only an issue from the mitel side. if i call from an asterisk-based number, it starts the audio at the beginning like it should |
| 17:57.00 | r00t-Ed | noob question, is fail2ban the best way to protect against hacks to asterisk? |
| 17:57.19 | navaismo | bobb_WU, maybe the delay is in the mitel's phone timeout |
| 17:57.43 | Qwell | r00t-Ed: It is a way. |
| 17:58.06 | [TK]D-Fender | r00t-Ed, No, removing the black cable between your power-supply & the wall is. But fail2ban is pretty decent too |
| 17:58.59 | r00t-Ed | i run asterisk on a wrt54g router, but i don't have much room for fail2ban |
| 17:59.17 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:59.17 | r00t-Ed | and all i have found is that |
| 17:59.33 | navaismo | try with blockhosts |
| 17:59.56 | r00t-Ed | it works great, but I have noticed tons of bots guessing users/passwords |
| 18:00.05 | Qwell | r00t-Ed: Using passwords that can't be guessed would be a good start. |
| 18:00.23 | bobb_WU | there isn't an issue when calling asterisk phones from mitel, it seems to be secluded to the voice mail server |
| 18:00.51 | r00t-Ed | i have done all the basic stuff, but that doesn't stop the bots running all the dictionary for random users |
| 18:01.02 | navaismo | bobb_WU, can you check it alive? When the mitel send the voicemail number you see immediately the activity in the cli? |
| 18:01.16 | bobb_WU | yes i'm doing that right now |
| 18:01.32 | r00t-Ed | passwords are pretty hard to guess, I doubt they find them, but that doesn't stop them from traying |
| 18:01.35 | navaismo | bobb_WU, can you paste the full log of that call, full log have the timestamps. |
| 18:01.54 | bobb_WU | it plays my message, then the 'vm-intro.gsm' and it starts playing audio half way through |
| 18:02.19 | bobb_WU | sure can |
| 18:02.23 | bobb_WU | you want sip debug in that? |
| 18:02.44 | navaismo | sure |
| 18:03.22 | bobb_WU | how can i turn logging verbosity up? it doesn't have all the output it needs |
| 18:03.35 | [TK]D-Fender | r00t-Ed, fail2ban <- |
| 18:03.51 | leifmadsen | core set verbose 10 |
| 18:03.59 | r00t-Ed | i have a few kb's to spare |
| 18:04.03 | leifmadsen | for debug level logging, you need to enable that in logger.conf |
| 18:04.12 | leifmadsen | then core set debug 10 |
| 18:04.12 | r00t-Ed | looking in to blockhost |
| 18:04.26 | r00t-Ed | navaismo mentioned above |
| 18:04.26 | bobb_WU | not logger set level debug on? |
| 18:04.45 | leifmadsen | that may work, never saw that command before |
| 18:05.30 | bobb_WU | i just tabbed a bunch to find it |
| 18:05.50 | leifmadsen | then sounds like you already know the answer |
| 18:05.52 | bobb_WU | testing the new logging level, pastebin to come |
| 18:06.25 | navaismo | r00t-Ed, http://www.aczoom.com/blockhosts/ |
| 18:06.41 | r00t-Ed | yup, checking it out right now, thanks |
| 18:07.37 | bobb_WU | sip debugging info is not in the logs |
| 18:07.55 | navaismo | r00t-Ed, I use that for ssh and sip attacks, so far work fine, isnt the panacea but it work with bots and ssh attacks |
| 18:14.10 | *** join/#asterisk coppice (~chatzilla@14.140.218.146) |
| 18:15.01 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 18:16.45 | r00t-Ed | at least the binary is tiny.. fail2ban bin was about 400k |
| 18:19.54 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 18:21.15 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 18:21.59 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 18:23.43 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 18:24.08 | Tuju | hi, please help me. somehow my config changed and now i cannot place calls anymore with cisco 7975. |
| 18:24.15 | bobb_WU | http://pastebin.com/nWfxW2Uh |
| 18:24.33 | Tuju | it still rings when someone calls and i can receive calls, but if i try to call myself, nothing happens. |
| 18:25.00 | Tuju | if i dump with tcpdump, it shows that asterisk sends responses back to 5060 port as it should. |
| 18:25.19 | bobb_WU | taking a quick 5 min break |
| 18:25.56 | rrittgarn | anyone ever have issues getting Caller ID to work with IAX? I'm trying to use the SET(CALLERID="Name<8009009000>" in the dialplan for the IAX users however it is just coming through as IAXModem2, etc. |
| 18:26.44 | WIMPy | rrittgarn: 'core show function CALLERID' |
| 18:28.20 | rrittgarn | sorry exact syntax i'm using is: same => n,Set(CALLERID(all)="Name<Number>" |
| 18:28.37 | rrittgarn | (with the closing paren etc) |
| 18:28.48 | Qwell | Why quotes? Where's your space? |
| 18:29.16 | rrittgarn | tried with space and without, and quotes because it usually has a space |
| 18:29.20 | WIMPy | I haven't tried to set (all) for ages, but I'm pretty sure it won't like the quotes and might want a space between the values. |
| 18:29.52 | [TK]D-Fender | rrittgarn, No quotes. |
| 18:32.14 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 18:32.25 | bobb_WU | back |
| 18:34.09 | Tuju | is dash, - character forbidden in user 'part' in sip url? |
| 18:34.31 | Tuju | like tuju-c7975 |
| 18:36.10 | *** join/#asterisk lorsungcu_ (~lorsungcu@65.103.31.36) |
| 18:38.01 | lorsungcu_ | i need to transfer a call, but hold on to it until it is hung up so I can record it. |
| 18:38.05 | lorsungcu_ | any suggestions? |
| 18:38.55 | Katty | lorsungcu_: http://cah.kired.net:81/1.html?q=10 refresh liberally. |
| 18:41.31 | [TK]D-Fender | Tuju, No |
| 18:42.07 | Tuju | not forbidden? |
| 18:42.18 | [TK]D-Fender | Correct |
| 18:45.36 | Tuju | http://fpaste.org/Z53Q/ that's debug output of that failed call. |
| 18:46.18 | Tuju | ha, unautohraised? |
| 18:46.23 | Tuju | 401 |
| 18:46.42 | Tuju | howcome i can make calls with other phones |
| 18:48.42 | bobb_WU | any further suggestions on my delay issue after seeing the pastebin'd stuff? |
| 18:51.43 | [TK]D-Fender | bobb_WU, just waiting doesn't guarantee that RTP is set up. and you should be wasting time waiting BEFORE you answer. Instead, ANSWER first, then playbacka few seoncds of SILENCE, not "wait" |
| 18:52.19 | Qwell | Tuju: 401 is the correct response for the first INVITE Asterisks ends. We need to see more debug. |
| 18:53.06 | Tuju | Qwell: it works now |
| 18:53.18 | Tuju | i added insecure=invite under peer configuration. |
| 18:53.39 | bobb_WU | how do i playback silence? |
| 18:53.49 | [TK]D-Fender | Playback(silence/2) |
| 18:53.50 | Tuju | puuuuuuuh, have i suffered for that for past weeks. |
| 18:53.57 | [TK]D-Fender | there is a FOLDER full of them. |
| 18:54.01 | [TK]D-Fender | Yuo really should look |
| 18:54.03 | Tuju | did that change in some update from default? |
| 18:58.45 | *** part/#asterisk r00t-Ed (~no@74.42.252.2) |
| 18:59.13 | *** join/#asterisk Jinxed- (~Administr@147.177.62.73) |
| 18:59.41 | Jinxed- | how similar are cisco's call control discovery to dundi |
| 19:06.15 | *** join/#asterisk shadowapex (~William@adsl-99-61-86-59.dsl.lsan03.sbcglobal.net) |
| 19:06.36 | shadowapex | Hi, I'm using Asterisk 1.8.12.0 (Elastix) and am currently running into a problem where Asterisk is sending the RTP audio to the wrong server. I currently have a trunk configured to go to an OpenSIPS server which acts as a signaling server and have a separate server that handles media. Right now when an inbound call comes from OpenSIPS, Asterisk is sending the RTP audio to the signaling server instead of the media server. In |
| 19:07.36 | shadowapex | I thought that Asterisk might be thinking that there is a NAT, so I set "nat=no" in the trunk configuration, but the problem is still persisting. I confirmed with a packet capture that Asterisk is sending the RTP audio to the signaling server instead of the media server specified in the SDP of the invite. Any thoughts? |
| 19:08.08 | bobb_WU | can i get asterisk to ring one more time so there isn't 5 seconds of silence? |
| 19:08.46 | [TK]D-Fender | bobb_WU, do a playback of 1s silence, and then Ringing() |
| 19:08.49 | [TK]D-Fender | for a few more |
| 19:09.12 | [TK]D-Fender | shadowapex, thought : SHOW US the configs and complete call debug. |
| 19:09.14 | [TK]D-Fender | ~pb |
| 19:09.14 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 19:09.16 | [TK]D-Fender | ^^^ |
| 19:10.33 | shadowapex | [TK]D-Fender: Sure, I can give you a PasteBin. Give me a second to compile it. |
| 19:13.50 | *** join/#asterisk ncrollo (d0357f0e@gateway/web/freenode/ip.208.53.127.14) |
| 19:14.00 | bobb_WU | it doesn't ring again after answering |
| 19:14.11 | [TK]D-Fender | And I see nothing |
| 19:14.54 | bobb_WU | Ringing(), Answer(),Playback(silence/3),Ringing(),Playback(silence/2) |
| 19:16.34 | *** part/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 19:19.03 | fling | kaldemar: can you please tell me how can I do Call Deflection or what info do I need to get from ISP guys for this? :p |
| 19:19.37 | [TK]D-Fender | fling, "core show application transfer" <- if they even support it |
| 19:19.45 | [TK]D-Fender | fling, Which I wouldn't bet on |
| 19:19.53 | fling | [TK]D-Fender: thanks |
| 19:20.23 | [TK]D-Fender | fling, And doesnt' answer any of the previous questions regarding exactly what service you are being provided. |
| 19:21.04 | fling | [TK]D-Fender: which kind of service? |
| 19:21.15 | [TK]D-Fender | the one you're using |
| 19:21.26 | fling | [TK]D-Fender: sip? |
| 19:22.10 | fling | [TK]D-Fender: or service I need to use for deflection? |
| 19:22.11 | [TK]D-Fender | that is teh TECH, not the service |
| 19:23.35 | fling | [TK]D-Fender: then I don't understand :( |
| 19:23.46 | *** part/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
| 19:27.38 | ncrollo | how is the status time determined when you have sip endpoints? |
| 19:28.22 | ncrollo | ex. sip show peer 111 |
| 19:28.32 | [TK]D-Fender | ncrollo, what "status time"? |
| 19:28.47 | ncrollo | aka "Natted qualification ping" |
| 19:29.00 | ncrollo | "Status : OK (259 ms)" |
| 19:29.07 | [TK]D-Fender | that is QUALIFY time |
| 19:29.21 | [TK]D-Fender | and is the time the device took to answer a SIP OPTIONS packet * sent out |
| 19:34.11 | lorsungcu_ | anybody using swift()? |
| 19:34.20 | ncrollo | thank you for that clarification |
| 19:48.14 | Kobaz | http://www.wimp.com/babymoose/ |
| 19:52.09 | shadowapex | [TK]D-Fender: sip.conf: http://pastebin.com/feR6qcNr, asterisk.log: http://pastebin.com/CDEPZY8b, Packet Capture: http://pastebin.com/asHx5tTP |
| 19:53.40 | shadowapex | [TK]D-Fender: Oddly enough, it looks like the media contact shown in the Asterisk log is showing as the signaling server, but in the packet capture, the media contact is correctly showing as the media server. |
| 19:53.50 | sawgood | If a SIP peer is inside of sip.conf with a host=IP address (and no other authentication) ... what could be a reason outbound calls fail to find the correct context (which is 100% correct) ... every outbound calls fails with the message: not found in context 'default' |
| 19:54.03 | sawgood | inbound calls to the box work 100% only outbound is failing |
| 19:54.31 | sawgood | all other peers are working (and setup nearly identical) |
| 19:55.29 | [TK]D-Fender | because there isn't a match where it's looking |
| 19:55.32 | [TK]D-Fender | just like it says |
| 19:55.42 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:56.19 | sawgood | I have a context=outbound-calls (inside of sip.conf) other entries use this same context (no mis-spelling here) |
| 19:57.01 | [TK]D-Fender | sawgood, We are not taking your word for any of this |
| 19:57.10 | [TK]D-Fender | The dialplan works |
| 19:57.13 | [TK]D-Fender | You've done something wrong |
| 19:57.27 | sawgood | I can show you better via Pastebin |
| 19:57.29 | sawgood | hold one pleae |
| 20:01.57 | shadowapex | [TK]D-Fender: If you need the extensions.conf with the relevant dialplan, I can provide that too |
| 20:02.24 | [TK]D-Fender | shadowapex, c=IN IP4 10.0.0.86 |
| 20:02.32 | [TK]D-Fender | so far everything points to that IP as the call comes in. |
| 20:02.43 | shadowapex | That's the signaling server |
| 20:02.43 | [TK]D-Fender | dialplan has nothing to do with this |
| 20:02.47 | shadowapex | Yeah, I know |
| 20:02.54 | [TK]D-Fender | yes and that's what your Freeswitch is telling * to use |
| 20:03.02 | shadowapex | That's why I posted the packet capture. in the packet capture it shows 10.0.0.101 |
| 20:03.08 | shadowapex | Which is the media server |
| 20:03.14 | [TK]D-Fender | o=FreeSWITCH 1343745579 1343745580 IN IP4 10.0.0.86 |
| 20:03.23 | shadowapex | o=FreeSWITCH 1343745414 1343745415 IN IP4 10.0.0.101. |
| 20:03.32 | shadowapex | c=IN IP4 10.0.0.101. |
| 20:03.56 | shadowapex | Only in the Asterisk log does it show c=10.0.0.86 |
| 20:04.13 | shadowapex | but in the raw packet you can see it's showing .101 |
| 20:04.38 | [TK]D-Fender | :/ |
| 20:06.11 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 20:07.35 | shadowapex | I'm positive Freeswitch is sending .101 as the media IP in the SDP. Packet captures from both the Asterisk side and the Freeswitch side show it correctly. |
| 20:08.10 | [TK]D-Fender | Well you've also mentioned an OpenSIPS in there somewhere. |
| 20:08.13 | shadowapex | Only within Asterisk does it show the media IP set as .86, which makes me think that Asterisk is changing it |
| 20:08.17 | shadowapex | Yeah, OpenSIPS is .86 |
| 20:08.17 | [TK]D-Fender | Guess what I'm not trusting in this picture? |
| 20:08.29 | [TK]D-Fender | the MIDDLE piece |
| 20:08.32 | shadowapex | Well OpenSIPS is just acting as a proxy |
| 20:08.50 | shadowapex | Forwarding the INVITE from FS to Asterisk |
| 20:09.12 | shadowapex | A packet capture from OpenSIPS ALSO shows the media IP set to .101 |
| 20:10.09 | [TK]D-Fender | Ok, not sure what I could advise here myself on this.... |
| 20:10.54 | *** join/#asterisk gusto (~gusto@2001:470:6d:213::42:4) |
| 20:11.26 | Qwell | shadowapex: Run the packet capture on the Asterisk box. |
| 20:11.29 | drmessano | OpenSIPS, FreeSWITCH and Asterisk.. What, no MS Lync server? |
| 20:11.35 | Qwell | Asterisk is absolutely not changing the header on you. |
| 20:11.50 | shadowapex | Qwell: The pastebin above is from the Asterisk server |
| 20:12.05 | shadowapex | drmessano: We do use MS Lync too, but not in this situation ;P |
| 20:12.13 | drmessano | O.o |
| 20:12.45 | Qwell | shadowapex: These are not the same call. |
| 20:13.24 | shadowapex | Qwell: I did the packet cap on a subsequent call, but if you'd like I can do the packet cap and log for the same call |
| 20:13.58 | lorsungcu_ | i need to transfer a call, but hold on to it until it is hung up so I can record it. |
| 20:14.08 | lorsungcu_ | any suggestions |
| 20:14.21 | *** join/#asterisk wonderworld (~ww@dsdf-4db5132b.pool.mediaWays.net) |
| 20:18.06 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 20:19.27 | uskerine | hi, is there any predefined stats in asterisk associated to a queue? |
| 20:22.06 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 20:23.32 | *** join/#asterisk danfromuk (~IceChat77@2.30.230.183) |
| 20:23.47 | *** join/#asterisk b0ot (~Administr@147.177.62.73) |
| 20:24.45 | b0ot | I know this isn't cisco support but they are busy arguing politics and I have been working on a cisco voice problem related to call control discovery and extension mobility for 2 days straight if someone could pm me I could explain more or read this post I made: https://supportforums.cisco.com/thread/2163061 |
| 20:26.36 | uskerine | which is the easiest way to get stats from a queue? number of calls received, average waiting time, average call time, etc. |
| 20:26.43 | uskerine | or |
| 20:26.48 | uskerine | can i get a complete CDR from a call? |
| 20:27.00 | [TK]D-Fender | CDR is completely separate from queue stats |
| 20:27.03 | shadowapex | Qwell: Asterisk.log: http://pastebin.com/LHhaky7n, Packet Capture: http://pastebin.com/2QUhHBUM |
| 20:27.05 | [TK]D-Fender | BOTH are described in THE BOOK |
| 20:27.08 | [TK]D-Fender | ~book |
| 20:27.08 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:27.09 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 20:27.11 | shadowapex | Qwell: Those are from the same call |
| 20:27.20 | [TK]D-Fender | checkout time, later all |
| 20:28.09 | Qwell | Your proxy is clearly changing things on you. |
| 20:28.37 | shadowapex | How? The packet capture on the Asterisk server itself is showing .101 |
| 20:28.49 | Qwell | Look at the first line in the INVITE. |
| 20:29.04 | Qwell | That is different. Asterisk is not changing your packet. It prints what it receives. |
| 20:31.15 | shadowapex | How can the proxy change the packet when it's already been sent to the Asterisk server? |
| 20:31.32 | Qwell | I didn't say your proxy was changing it. |
| 20:31.36 | Qwell | Something very clearly is. |
| 20:32.12 | shadowapex | I can't think of anything other than Asterisk that might be modifying it. |
| 20:32.20 | Qwell | I can think of lots of things. |
| 20:32.23 | Qwell | pastebin the output of lsmod |
| 20:33.25 | shadowapex | http://pastebin.com/rZ4WfZyt |
| 20:33.33 | Qwell | HEY LOOK AT THAT |
| 20:33.35 | *** join/#asterisk gusto (~gusto@2001:470:6d:213::42:4) |
| 20:33.51 | shadowapex | ip_nat_sip |
| 20:34.00 | shadowapex | son of a... |
| 20:36.14 | *** join/#asterisk Greek-Boy (~Greek-Boy@196.46.120.105) |
| 20:37.17 | uskerine | is there any easy way to get some stats about queues? |
| 20:37.24 | uskerine | in ascii |
| 20:38.07 | *** join/#asterisk Pan3D (~Pan3D@63.208.160.190) |
| 20:39.27 | navaismo | uskerine, have you tried with asternicstats http://www.asternic.biz/ |
| 20:39.34 | shadowapex | Qwell: I stopped the firewall before, but the problem was still persisting, but after completely stopping iptables prevented it from modifying the SIP INVITE. Works like a charm now :) |
| 20:39.45 | shadowapex | Qwell: Thanks for your help! |
| 20:40.37 | uskerine | have you tried it navaismo? |
| 20:42.13 | uskerine | i would prefer to understand a bit how to configure it through config files |
| 20:45.58 | *** join/#asterisk scubes13 (~scubes13@cpe-098-025-028-033.sc.res.rr.com) |
| 20:46.15 | navaismo | then use the documentation of queue.conf and queue_log |
| 20:46.22 | scubes13 | Katty... hmmm… not sure why I didnt auto connect here to begin with… |
| 20:46.42 | Katty | (= |
| 20:49.18 | *** join/#asterisk ks3 (~ks3@74.115.41.6) |
| 20:52.50 | uskerine | navaismo |
| 20:52.58 | uskerine | seems that all the logs for the queue are enabled |
| 20:53.03 | uskerine | when they enter in the queue |
| 20:53.11 | uskerine | if they abandon and how much did they wait |
| 20:53.11 | uskerine | etc. |
| 20:53.17 | uskerine | i am also having a look to asternic |
| 20:53.21 | uskerine | did you actually try it? |
| 20:53.29 | *** join/#asterisk ks3 (~ks3@74.115.41.6) |
| 20:55.43 | *** join/#asterisk Korolev (~Korolev@204.88.28.115) |
| 20:56.23 | *** join/#asterisk ks3 (~ks3@74.115.41.6) |
| 20:58.01 | uskerine | it looks like easy to install and nice to have |
| 21:03.12 | navaismo | yes I installed once, usually I use queuemetrics for customers |
| 21:03.58 | *** join/#asterisk pyther (~pyther@unaffiliated/pyther) |
| 21:05.55 | uskerine | pl thanks |
| 21:06.03 | uskerine | ok thanks |
| 21:06.05 | uskerine | i will give it a try |
| 21:06.13 | uskerine | i have another question |
| 21:06.15 | uskerine | what |
| 21:06.24 | uskerine | <PROTECTED> |
| 21:06.24 | uskerine | <PROTECTED> |
| 21:06.34 | uskerine | "playing periodic announcement" does actually mean? |
| 21:06.45 | uskerine | periodic-announce = queue-periodic-announce,your-call-is-important,please-wait |
| 21:06.53 | uskerine | that's from queues.conf |
| 21:07.17 | uskerine | i see that |
| 21:07.19 | uskerine | -- Playing periodic announcement |
| 21:07.19 | uskerine | <PROTECTED> |
| 21:07.36 | uskerine | every 10 seconds the "playing periodic announcement" message appears |
| 21:07.54 | uskerine | but only one out of three times the actual 'playing queue-periodic-announce.gsm' is played by asterisk |
| 21:08.00 | uskerine | what happens the other two times? |
| 21:08.04 | *** join/#asterisk sebastianpersic (~chatzilla@ua-85-227-32-4.cust.bredbandsbolaget.se) |
| 21:08.27 | navaismo | quote from queue.conf at voip-info: "Periodic announcements are available in queues using the new periodic-announce and periodic-announce-frequency options. This allows a message like "Thank you for holding, your call is important to us." to be played at regular intervals while a caller is in the queue" |
| 21:08.29 | sebastianpersic | Is there any guide how to install on ubuntu 12.04 server lts |
| 21:08.57 | Qwell | sebastianpersic: ./configure; make; make install |
| 21:09.04 | navaismo | sebastianpersic, yeah the generic one, that ^ |
| 21:09.21 | uskerine | navaismo, i understand that, but why "your-call-is-important" and "please-wait" are not played? |
| 21:09.40 | uskerine | it seems that only "queue-periodic-announce" is played |
| 21:10.54 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 21:10.57 | navaismo | yes, weel i only used one file in that definition |
| 21:11.12 | navaismo | s/weel/well/ |
| 21:12.45 | navaismo | the other files exist? |
| 21:12.55 | navaismo | s/other/others/ |
| 21:15.22 | uskerine | oops i found the folder |
| 21:15.23 | sebastianpersic | but is there any step by step guide setting up asterisk |
| 21:15.27 | uskerine | there were no such files |
| 21:15.34 | Qwell | ~book |
| 21:15.34 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:15.56 | Katty | how long till christmas? |
| 21:16.06 | Katty | i'll settle for halloween :< |
| 21:17.28 | drmessano | I'm waiting for the 4th edition |
| 21:17.36 | drmessano | I have an autographed second edition |
| 21:17.44 | drmessano | Think I am gonna stick with even numbers |
| 21:17.55 | Katty | no one even said hi on mine. |
| 21:18.00 | Katty | where is that book... |
| 21:18.22 | Katty | it doesn't even have an edition on it |
| 21:18.26 | drmessano | Russell autographed mine and put my name on it. |
| 21:18.30 | drmessano | Swoon |
| 21:18.36 | Qwell | drmessano: I have a signed digital copy of the second edition |
| 21:18.40 | Qwell | probably the only one in existence. |
| 21:18.41 | drmessano | lol |
| 21:18.54 | Qwell | <PROTECTED> |
| 21:18.56 | Katty | printing history, september 2005: first edition |
| 21:18.57 | Qwell | That happened. |
| 21:19.01 | drmessano | Ohhh |
| 21:19.33 | Katty | someone named meggelen, smithh, and madsen |
| 21:19.37 | Katty | whoever those are |
| 21:19.53 | drmessano | Never heard of them. Sound like lawyers |
| 21:20.22 | Katty | that should be part of the xmas card exchange. |
| 21:20.27 | Katty | send each other books for writing notes on |
| 21:20.41 | Katty | that'd probably get expensive to ship books to each other tho |
| 21:21.03 | Katty | maybe photos with the books instead?! |
| 21:21.27 | Katty | and a recipe, of course. |
| 21:23.16 | scubes13 | of course |
| 21:23.21 | scubes13 | and… a CAH card.. |
| 21:23.50 | Katty | YES |
| 21:23.51 | scubes13 | maybe even a Jayne hat, eh Katty? o.O |
| 21:23.57 | Katty | ..maybe. |
| 21:24.01 | Katty | i make no promises |
| 21:24.05 | jaytee | "I'll be in my bunk!" |
| 21:24.15 | Katty | hi john |
| 21:24.27 | jaytee | I have a Jayne hat. A friend of mine from Ohio made it for me. |
| 21:24.30 | jaytee | hi Katty |
| 21:24.34 | Katty | that's wonderful! |
| 21:24.39 | Katty | what a wonderful friend. |
| 21:24.51 | Katty | you should put it on a penguin, and send me a picture of it |
| 21:24.52 | jaytee | my old boss at the zoo hated it |
| 21:24.54 | Katty | or do you work at the zoo anymore? |
| 21:25.00 | navaismo | uskerine, you found the problem |
| 21:25.05 | jaytee | I left the zoo in 2010 |
| 21:25.12 | Katty | oh where has the time gone |
| 21:25.27 | Katty | where are you at now dear |
| 21:26.13 | fling | <PROTECTED> |
| 21:26.16 | fling | <PROTECTED> |
| 21:26.33 | fling | what does this mean 'UNKNOWN'? |
| 21:26.55 | lorsungcu_ | unknown |
| 21:27.08 | fling | oh! ok |
| 21:27.11 | lorsungcu_ | np |
| 21:27.39 | Katty | http://cah.kired.net:81/1.html?q=10 |
| 21:27.47 | Katty | refresh liberally. |
| 21:28.30 | lorsungcu_ | fling |
| 21:28.34 | lorsungcu_ | first google result |
| 21:28.34 | lorsungcu_ | http://forums.whirlpool.net.au/archive/885099 |
| 21:29.19 | lorsungcu_ | anyone have swift() installed and want to test something for me |
| 21:31.09 | lorsungcu_ | im seeing this issue in 1.8.14.1 : https://issues.asterisk.org/jira/browse/ASTERISK-17173?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel#issue-tabs |
| 21:32.42 | drmessano | lorsungcu_: Seems like it was an issue with app_swift |
| 21:33.33 | *** join/#asterisk lorsungcu (~lorsungcu@209.118.96.2) |
| 21:33.43 | drmessano | Nice |
| 21:34.29 | uskerine | navaismo, more question about queues, is it possible for an agent to put the call on hold so the customer listens again to the music until the agent recovers that call? |
| 21:37.21 | *** join/#asterisk lorsungcu (~lorsungcu@209.118.96.2) |
| 21:39.07 | sebastianpersic | sudo apt-get install build-essential \ |
| 21:39.08 | sebastianpersic | subversion libncurses5-dev libssl-dev \ |
| 21:39.08 | sebastianpersic | libxml2-dev vim-nox |
| 21:39.36 | sebastianpersic | should I just change the vim to nano if I want to use nano? |
| 21:40.07 | lorsungcu | no if you want to use nano, just type vim and hope really hard that nano comes up. |
| 21:40.13 | drmessano | lol |
| 21:40.14 | fling | sebastianpersic: you should change it to emacs :p |
| 21:40.24 | drmessano | wine + notepad |
| 21:40.32 | lorsungcu | ms word 97 |
| 21:40.38 | sebastianpersic | and am I supposed to type those \ |
| 21:41.00 | drmessano | You could type it all as one long line |
| 21:41.02 | lorsungcu | the \ keeps it on a single line. |
| 21:41.04 | drmessano | But thats up to you |
| 21:41.31 | sebastianpersic | ok thx |
| 21:43.22 | sebastianpersic | could not find nano-nox |
| 21:44.10 | navaismo | uskerine, yes but the customer only will the music without the announce, because the MOH is triggered by the pone, like |
| 21:44.15 | navaismo | phone** |
| 21:44.24 | navaismo | aaaaaaaaaaaaaa |
| 21:44.35 | uskerine | i am fine with that |
| 21:44.43 | uskerine | as the customer is not actually returning to queue |
| 21:44.51 | uskerine | it is attanded, but has been put on hold by the agent |
| 21:45.01 | uskerine | could you point me in the right direction to implement that? |
| 21:45.48 | navaismo | press the MOH button on your phone |
| 21:45.56 | uskerine | what is the MOH button? |
| 21:46.02 | uskerine | i am using softphone |
| 21:46.56 | uskerine | is it required to configure something in extensions.conf or somewhere else? |
| 21:47.50 | navaismo | your softphone soesnt have a hold button? |
| 21:48.05 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 21:48.36 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 21:48.44 | uskerine | i don't think so |
| 21:49.26 | fling | Can you please help me with my setup? /etc/asterisk/sip.conf > http://bpaste.net/show/37821/ ; /etc/asterisk/extensions.conf > http://bpaste.net/show/37822/ |
| 21:50.10 | *** join/#asterisk vlad_starkov (~vlad_star@ppp91-79-109-246.pppoe.mtu-net.ru) |
| 21:54.57 | navaismo | uskerine, what softphone you are using? |
| 21:55.20 | *** join/#asterisk lorsungcu (~lorsungcu@209.118.96.2) |
| 21:55.31 | uskerine | linphone |
| 21:56.05 | uskerine | if the softphone has the on-hold and transfer buttons, do i have to change something in the config files for asterisk? |
| 21:57.15 | navaismo | nope |
| 21:57.45 | uskerine | any recommendation for softphone? |
| 21:57.58 | malcolmd | blink for mac; jitsi for everything else |
| 21:58.18 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-rgqjvofoniehiynl) |
| 21:59.14 | navaismo | uskerine, linphone has the pause button |
| 21:59.23 | navaismo | it trigger the MOH |
| 21:59.40 | navaismo | or im crazy |
| 21:59.43 | navaismo | let see |
| 21:59.45 | navaismo | 1 sec |
| 22:00.25 | uskerine | i can't check right now |
| 22:00.38 | uskerine | i am working on remote asterisk but i am not in the LAN |
| 22:01.03 | uskerine | i will try anyway |
| 22:01.04 | uskerine | it is ok |
| 22:02.04 | navaismo | yep pause button put onhold |
| 22:03.24 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-unclzwdkrymgggce) |
| 22:09.11 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 22:18.55 | *** part/#asterisk newtonr (~newtonr@nat/digium/x-ngskfuuulcfrougs) |
| 22:21.37 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 22:23.14 | *** join/#asterisk iamcorbin (~corbin@c-68-83-202-225.hsd1.mi.comcast.net) |
| 22:24.14 | iamcorbin | I've never used asterisk before and I am piecing together a new computer to set it up on. Is there a list of supported network cards somewhere? I am thinking about getting this board: http://www.newegg.com/Product/Product.aspx?Item=N82E16813153239 |
| 22:25.41 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 22:25.49 | uskerine | thanks navaismo |
| 22:30.41 | navaismo | no problem |
| 22:32.04 | navaismo | iamcorbin, the drivers is related with your OS, not asterisk. |
| 22:34.25 | iamcorbin | so that board should work just fine? |
| 22:35.35 | leifmadsen | iamcorbin: well your link doesn't go to anytrhing but the main page for me |
| 22:35.53 | leifmadsen | looks like a weird redirect for me at least |
| 22:36.11 | leifmadsen | iamcorbin: but ya, the network card and it's operation has nothing to do with asterisk |
| 22:36.12 | *** part/#asterisk millsu2 (~brad@mail.serverplus.com) |
| 22:36.21 | leifmadsen | that's the OS's problem |
| 22:37.29 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 22:37.33 | *** part/#asterisk shadowapex (~William@adsl-99-61-86-59.dsl.lsan03.sbcglobal.net) |
| 22:44.12 | lorsungcu | has anyone used drbd with asterisk for failover? |
| 22:49.03 | lorsungcu | working on some existing macros, what is the best way to exit to a specific context/priority |
| 22:49.13 | uskerine | which are the basic steps to: |
| 22:49.24 | uskerine | 1) record a call when it is answered in a queue |
| 22:49.38 | uskerine | 2) allow the agent to not store the call |
| 22:49.53 | uskerine | ? |
| 22:50.22 | lorsungcu | you mean allow the agent to not keep the recording? |
| 22:51.20 | lorsungcu | probably enable the |
| 22:51.48 | lorsungcu | membergosub=, and have it do what you're asking |
| 22:52.56 | navaismo | ah? |
| 22:53.21 | uskerine | i mena that all calls will be recorded by default |
| 22:53.22 | lorsungcu | you might be able to do it with the set(MONITOR_EXEC()) command, but it sounds like it might be easier the first way. |
| 22:53.27 | uskerine | i mean that all calls will be recorded by default |
| 22:53.43 | uskerine | but when the call starts, the customer might ask (as per legal requirement) the agent not to record the call |
| 22:53.55 | uskerine | so the agent should have a way to discard the record |
| 22:54.13 | lorsungcu | why not prompt the caller before connecting them |
| 22:54.49 | uskerine | you mean like asking them "press * if you do not want to be recorded" |
| 22:54.50 | uskerine | ? |
| 22:54.55 | lorsungcu | sure |
| 22:55.03 | uskerine | it would be better the other way |
| 22:55.12 | navaismo | hmmm usually we prompt an audio to let them know the call will be recorder if they dont want the call hangup |
| 22:55.16 | uskerine | these customers will not be so smart |
| 22:55.29 | uskerine | navaismo i like that way, but i don't think i can do that |
| 22:55.58 | lorsungcu | read the documentation for the membergosub= in queues.conf |
| 22:56.37 | uskerine | i will do that |
| 22:59.39 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 23:02.49 | lorsungcu | you could even just ask that the queue members press a dedicated button to initiate recording |
| 23:02.57 | lorsungcu | rather than recording by default. |
| 23:03.43 | uskerine | can't find membergosub in the book (http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id288901.html) |
| 23:03.47 | lorsungcu | hmmm |
| 23:04.15 | navaismo | memebermacro |
| 23:04.16 | uskerine | i would precisely prefer the opposite, the agent being able to invalidate the recording |
| 23:04.45 | lorsungcu | ok |
| 23:04.51 | navaismo | i think its easier start or not start the recording |
| 23:05.28 | lorsungcu | in features.conf |
| 23:05.41 | lorsungcu | tehre are examples for pausing/unpausing monitoring |
| 23:05.47 | lorsungcu | that might do it for you |
| 23:07.32 | uskerine | this sounds like the most difficult part |
| 23:07.45 | lorsungcu | ? |
| 23:09.15 | uskerine | http://www.fosslc.org/drupal/node/643 |
| 23:09.28 | uskerine | i think something like this might be a good start |
| 23:11.15 | uskerine | what does pauseMonitor do? |
| 23:11.23 | uskerine | sounds like it stops more things than jsut recording |
| 23:12.19 | lorsungcu | do you happen to have swift installed, uskerine? |
| 23:12.37 | uskerine | i don't know what swift is |
| 23:12.45 | uskerine | i am learning the hard way |
| 23:12.52 | lorsungcu | :) |
| 23:14.14 | uskerine | do i need it? |
| 23:14.28 | lorsungcu | no |
| 23:14.36 | uskerine | it is text-to-speech, right? |
| 23:14.37 | lorsungcu | i just need someone to test something involving mixmonitor |
| 23:14.39 | lorsungcu | yes |
| 23:15.09 | uskerine | if i can help |
| 23:15.26 | lorsungcu | nah itd be a pain for you |
| 23:15.52 | uskerine | ok i will continue tomorrow, thanks a lot for your guidance, it saves me lot of time |
| 23:16.20 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 23:16.46 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-3-7.bois.qwest.net) |
| 23:25.08 | *** join/#asterisk SeRi (~wtf@c-98-200-177-50.hsd1.tx.comcast.net) |
| 23:25.21 | SeRi | p3nguin: You arround? |
| 23:26.52 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 23:26.58 | SeRi | I am trying to enable incomming calls recording and remote voicemail support and I been stucked in limbo |
| 23:27.26 | SeRi | so %^&* called my house today with a false cid and obtain private information |
| 23:27.40 | SeRi | the cid display my own number and my name |
| 23:28.07 | SeRi | my 9yr old not knowing better gives all the information been asked thinking is me... |
| 23:29.55 | *** join/#asterisk HyperNerdV2 (~FoughtThe@71-95-164-90.static.mtpk.ca.charter.com) |
| 23:29.58 | lorsungcu | where are you stuck? |
| 23:31.07 | SeRi | well on issue number one. I have no idea how to enable incomming call recording. |
| 23:31.27 | jpsharp | Use the MixMonitor command |
| 23:31.28 | lorsungcu | https://wiki.asterisk.org/wiki/display/AST/Application_MixMonitor |
| 23:31.48 | lorsungcu | can you post your extensions.conf for the relevant extension? |
| 23:31.56 | *** join/#asterisk HyperNerdV2 (~FoughtThe@71-95-164-90.static.mtpk.ca.charter.com) |
| 23:32.15 | SeRi | lorsungcu: Sure. Thanks for the help!. One sec |
| 23:34.15 | SeRi | http://pastebin.com/RCWYu3ZH |
| 23:34.34 | SeRi | Thats my inbound extension from voip.ms |
| 23:34.41 | SeRi | lorsungcu: ^^ |
| 23:36.53 | *** join/#asterisk mbrit (~mbrit@186.120.97.194) |
| 23:36.57 | lorsungcu | try that? |
| 23:38.00 | SeRi | lorsungcu: Thanks! |
| 23:46.11 | *** join/#asterisk sustav (~vpp@76.73.166.16) |
| 23:47.54 | lorsungcu | its too hot. |
| 23:51.10 | jpsharp | Summer sucks. |
| 23:54.10 | lorsungcu | in my office with 4x dell 1850s |
| 23:54.12 | lorsungcu | no ac |
| 23:54.15 | lorsungcu | windows open |
| 23:54.19 | lorsungcu | 85 degrees out |
| 23:54.32 | lorsungcu | probably 100 inside :< |
| 23:55.14 | *** join/#asterisk lorsungcu_ (~lorsungcu@65.103.31.36) |