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| 08:21.06 | qloogkm | does anyone have a recommended client for Hylafax that is nice? |
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| 09:11.40 | phpboy | I've tried every method known to man (google) to get audio working with WEBRTC and still it doesn't want to work... can somebody please help me |
| 09:14.07 | Chainsaw | phpboy: What Asterisk version are you trying WebRTC with? 11 or 13? |
| 09:16.08 | phpboy | Chainsaw: 12 |
| 09:16.34 | Chainsaw | phpboy: That is a development release. I would recommend that you try on 13 instead. |
| 09:16.45 | phpboy | that I can do |
| 09:17.49 | phpboy | let me install it quick |
| 09:18.11 | phpboy | going to remove /usr/lib64/asterisk and build asterisk13 |
| 09:18.57 | Chainsaw | phpboy: Essentially the PJSIP support in 13 is better. 12 was caught in the middle between chan_sip & PJSIP if you ask me. |
| 09:19.18 | Chainsaw | phpboy: I'm expecting tutorials for "Asterisk 12 with PJSIP" to work well on 13 with PJSIP. Just better. |
| 09:19.37 | phpboy | Well, I have pjproject installed on the server but I cannot for the life of me figure out where it fits into the picture |
| 09:19.47 | phpboy | been drowning myself in reading :( |
| 09:20.00 | Chainsaw | There is a lot of reading material out there, yes. |
| 09:20.19 | Chainsaw | You're going to have to cut things into modules, and test small parts. |
| 09:20.47 | phpboy | certified-asterisk-13.1-cert2 <--- good enough? |
| 09:20.55 | Chainsaw | i.e. try with a SIP soft phone from the machine you're trying WebRTC from, and get that to the point where you get audio both ways. Then try WebRTC, so that you know you're not fighting a NAT problem elsewhere. |
| 09:21.00 | Chainsaw | Yes, certified Asterisk will do nicely. |
| 09:21.40 | phpboy | ok, the one thing I can confirm from the beginning is there is NO NAT involved here and there will never be a NAT requirement for this specific project I'm working on |
| 09:23.46 | Chainsaw | phpboy: That does make things a little easier. |
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| 09:26.51 | phpboy | ok, I'm looking through menu select.. I can confirm all pjsip related modules are checked.... app_ices enabled... and strp enabled |
| 09:27.03 | phpboy | anything else I do think of before I make and make install? |
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| 09:28.23 | phpboy | I don't want to waste your time, but I'd love for you to quickly help me through... I'll be quick! |
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| 09:37.58 | Chainsaw | Unfortunately I'm in an office full of people, they do ask me questions as well. |
| 09:38.07 | Chainsaw | So honest admission, you do not have my undivided attention. |
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| 09:38.44 | Chainsaw | I'd make sure you have STUN, even if you think you don't need it. Clients may try it unconditionally. |
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| 09:42.20 | Padma3475 | Hi there , I have an outbound IVR with grand stream HT503 as PSTN Gateway asterisk start playing recorded mesage before user pickup the phone and miss first few words of the recorded message, how do i correct this. could any one help in this regard |
| 09:55.37 | phpboy | Chainsaw: I'm going to be writing the webclient myself and besides that I cannot get STUN working properly either |
| 09:55.49 | phpboy | asterisk 13 installed and still not working |
| 09:56.14 | Chainsaw | phpboy: The two least descriptive words in the world... "not working". |
| 09:56.59 | phpboy | I've configured pjsip ONLY, both clients I'm testing with register (Firefox sip.js) and (Chrome sipxl5) |
| 09:58.07 | phpboy | so lets start with chome, when I click call it doesn't seem to do anything with asterisk... what I see in chrome debug console is all the SIP chatter and then a final message --> The FSM is in the final state |
| 09:59.23 | Chainsaw | phpboy: And what have you set up in the dial plan? echo test? |
| 10:00.11 | phpboy | no, a welcome playback |
| 10:00.21 | phpboy | the thing is it doesn't even attempt the dialplan |
| 10:00.41 | Chainsaw | phpboy: You'd only see it traverse the dial plan if you're on a high "verbose" level. Are you? |
| 10:01.11 | phpboy | asterisk -vvvvvvvvvvvvvvvvvvvvvvr |
| 10:01.18 | Chainsaw | phpboy: core set debug 10 |
| 10:01.20 | Chainsaw | phpboy: core set verbose 10 |
| 10:01.25 | Chainsaw | phpboy: Then try. |
| 10:01.57 | Chainsaw | is wanting to rule out the call working and just not lasting beyond the voice prompt |
| 10:02.09 | Chainsaw | (This is why the echo test is better as it lasts forever) |
| 10:02.19 | phpboy | absolutely nothing |
| 10:02.29 | phpboy | well add echo test but I need to get to the dialplan first |
| 10:04.08 | Padma3475 | can any one help me to rectify my outbound calling issues |
| 10:05.32 | phpboy | Padma3475: sure |
| 10:05.38 | phpboy | Chainsaw: I've enabled pjsip logging |
| 10:05.51 | phpboy | getting some unauthorized messages which is not correct |
| 10:06.28 | Chainsaw | phpboy: Is it wanting a TLS transport that you are not providing? |
| 10:07.37 | phpboy | Chainsaw: it is possible but I believe it's setup correctly |
| 10:07.41 | phpboy | how can I tell for sure? |
| 10:07.54 | Padma3475 | Hi phpboy, My set up is aserisk call pstn number and plays a recorded message before user picks up the phone, is there a way to correct it. thanks |
| 10:08.03 | Chainsaw | phpboy: I'd expect your client end to tell you if is abandoning TLS. |
| 10:08.15 | Chainsaw | phpboy: So the browser. |
| 10:09.13 | phpboy | nothing stands on in the browser |
| 10:09.20 | phpboy | can asterisk debug or some sort help? |
| 10:10.55 | phpboy | I am seeing a Not Acceptable msg |
| 10:11.40 | phpboy | http://pastebin.ca/3223018 the data closer to the bottom is a more recent call |
| 10:11.45 | Chainsaw | phpboy: That could happen for various reasons, including an audio codec that is not to its liking. |
| 10:13.18 | phpboy | http://pastebin.ca/3223019 <--- pjsip debug of a failed call |
| 10:14.10 | Chainsaw | We may need Fender, the SIP whisperer. |
| 10:14.28 | phpboy | lol |
| 10:20.46 | Chainsaw | phpboy: The codec list that the client is offering seems rather comprehensive, but the response suggests it likes none of them. |
| 10:20.53 | Chainsaw | phpboy: Are you restricting available codecs? |
| 10:25.00 | phpboy | ok, I've managed to get into the dialplan... was a misconfiguration in pjsip.conf |
| 10:25.10 | phpboy | so now I'm hitting the dialplan but no audio |
| 10:25.25 | Chainsaw | phpboy: Hitting the dial plan is good. (Was it the context?) |
| 10:26.34 | phpboy | no, it was the webserver setup in pjsip.conf |
| 10:27.08 | phpboy | ok, so now I do a rtp debug and it's sending the rtp traffic to my public IP which is completely incorrect as there's no NAT setup |
| 10:28.21 | phpboy | hmm, icesupport=false set |
| 10:28.30 | Chainsaw | I'd let it use ICE. |
| 10:28.43 | phpboy | does it matter that I don't have stun set? |
| 10:28.45 | Chainsaw | Better to have NAT awareness and not use it then to explicitly say "no NAT ever" and be wrong, now or later. |
| 10:29.03 | phpboy | ok |
| 10:29.28 | phpboy | ice enabled... |
| 10:30.56 | phpboy | Sent RTP packet to 192.168.7.248:49816 (via ICE) (type 00, seq 059628, ts 023360, len 000160) |
| 10:31.01 | phpboy | one of many rtp debugs |
| 10:31.08 | phpboy | still no audio |
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| 10:34.49 | Padma3475 | my asterisk is connected to a grandstream pstn gateway. asterisk dials a pstn number and plays a recorded message.the problem is that, asterisk plays the message before the user pickup the phone, so he missess first few words |
| 10:35.27 | phpboy | Padma3475: put in a Wait(2) before the message is played |
| 10:36.51 | Padma3475 | <PROTECTED> |
| 10:38.52 | phpboy | what you may want to consider then is using ORIGINATE |
| 10:39.01 | phpboy | that way it will play the message when the call is answered |
| 10:39.14 | Padma3475 | I am using originate already |
| 10:42.48 | phpboy | then it should only be playing the message when the call is connected? |
| 10:43.00 | phpboy | or rather, when the client answers the call |
| 10:43.45 | phpboy | so how it should work is you origniate to the clients number being (12345678) and when it connects the call it will execute the other end (your playback file) |
| 10:44.13 | Padma3475 | grandstream pstn gateway is registered to asterisk as a sip account (eg:100), asterisk dial the pstn number like this: originate sip/100/pstnnumber when the gateway responds asterisk start playing the message |
| 10:45.01 | phpboy | I would dial the orignate number directly from asterisk |
| 10:45.14 | phpboy | see asterisk thinks it's answered the call hence doing its job |
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| 10:46.13 | phpboy | it should be SIP/gateway/123456(client number) ----> answer ----> hit asterisk context with playback msg |
| 10:47.02 | Padma3475 | exactly, but how do i directly dial the originate number |
| 10:47.32 | phpboy | setup a sip connection to your grandstream |
| 10:47.52 | phpboy | and dial via that sip connection |
| 10:48.21 | phpboy | so it would look like sip/grandstream/123456 |
| 10:49.41 | Padma3475 | i have registered the gateway ,as a sip account is that you mean? |
| 10:49.57 | Padma3475 | now i dial sip/100/12345 |
| 10:50.17 | Padma3475 | 100 is the sip account |
| 10:50.25 | phpboy | yes |
| 10:50.46 | Padma3475 | but still the problem occurs |
| 10:52.15 | phpboy | then it thinks the call as been answered because it's answered |
| 10:52.23 | phpboy | *before |
| 10:53.03 | Padma3475 | you are correct, but is there a way to solve this scenario |
| 10:53.03 | phpboy | pastbin the asterisk console between initiating the call and playing back the file |
| 10:53.17 | phpboy | let me see exactly what's happening |
| 10:53.30 | Padma3475 | ok |
| 10:55.43 | Padma3475 | can two stage dialing be a solution to this, |
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| 11:08.51 | phpboy | Padma3475: meaning? |
| 11:08.58 | phpboy | just pastebin the colsole output |
| 11:09.17 | phpboy | Chainsaw: got some time on your hands? |
| 11:15.00 | Padma3475 | phpboy:http://pastebin.com/5t3V7LuJ |
| 11:16.23 | phpboy | hmmmm, that doesn't look like a plain old originate |
| 11:16.34 | phpboy | how are you originating a call exactly? |
| 11:18.25 | Padma3475 | from the dialplan |
| 11:20.00 | Padma3475 | every thing is ok if the user pich up the phone |
| 11:20.21 | Padma3475 | <PROTECTED> |
| 11:20.37 | Padma3475 | <PROTECTED> |
| 11:20.55 | Padma3475 | if he wait for long he may miss more words |
| 11:27.10 | Padma3475 | phpboy:same => n,originate(sip/100/12345,exten,users,xxxx,1) |
| 11:31.36 | phpboy | try this on your cli |
| 11:32.01 | phpboy | originate SIP/100/123456 extension 123@your_dialplan_context |
| 11:34.24 | Padma3475 | sorry,extension 123@ means.? |
| 11:48.18 | phpboy | so in your extensions.conf |
| 11:48.31 | phpboy | your playback file falls under a context |
| 11:48.39 | phpboy | let's call it [my-context] |
| 11:48.55 | phpboy | and within that context the file is played by a dialing plan |
| 11:50.34 | Padma3475 | I am doing the same . Let me share my doubts with u ? |
| 11:50.48 | phpboy | ok |
| 11:51.32 | Padma3475 | I think when asterisk dial sip/100/123456 in a single shot actually it is like |
| 11:51.49 | Padma3475 | asterisk call sip 100 gateway responds |
| 11:52.05 | phpboy | that's fine, it's the answer portion that's important |
| 11:52.22 | Padma3475 | asterisk think ok other end pickedup let me play the message |
| 11:52.50 | file | Asterisk isn't what determines that it has answered, the gateway responds that it has answered |
| 11:52.54 | phpboy | that could very well be |
| 11:53.01 | Padma3475 | by this moment gateway is dialing the pstn no |
| 11:54.21 | phpboy | that's fine |
| 11:54.27 | phpboy | it doesn't matter what is making the call |
| 11:54.35 | phpboy | it just matters what asterisk interprets |
| 11:57.19 | Padma3475 | i guess when gateway responds asterisk interpret it as line is answered, because if i call a sip phone,message plays exactly after user pick the phone |
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| 11:58.22 | file | Asterisk is told that it is answered. It doesn't do any analysis on the audio or anything. The gateway likely responded with a 200 OK which means answered. |
| 11:59.50 | phpboy | yeah |
| 11:59.53 | Padma3475 | @file: then how we tell asterisk, wait untill pstn user pick the line? |
| 11:59.57 | phpboy | or there's something funky in the dialplan |
| 12:00.34 | file | you can't, unless you try to use something like AMD (answering machine detection) to determine that a person has picked up |
| 12:00.34 | Padma3475 | but to sip phone it is working fine |
| 12:00.48 | file | a SIP phone won't respond that it has answered until you pick up |
| 12:01.18 | Padma3475 | if i get rid of gateway and instead use a pstn card, will this help? |
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| 12:04.19 | Padma3475 | will two stage dialing help to resolve this? |
| 12:09.05 | file | analog doesn't provide call progress so the audio has to be analyzed to determine things, Asterisk has really experimental code which may or may not work for PSTN cards |
| 12:09.12 | file | otherwise the gateway has to do it |
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| 12:09.48 | phpboy | Anybody got some time on their hands to help me with a WebRTC audio issue that I'm having? |
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| 12:12.32 | Padma3475 | @file:phpboy: Ok. thanks for your time |
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| 13:52.02 | Reapster | heya, I'm having some odd trouble with bridging. I'm creating a bridge and adding a channel to PJSIP andpoint and another to Local/*43@from-internal (echo test for testing), but keep ending up with Unable to find a codec translation path: (ulaw) -> (slin). core show codecs shows both slin and ulaw available though, and there seems to be a translation in core show translations |
| 13:52.27 | Reapster | it seems to be creating a simple_bridge, do these not transcode? |
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| 14:21.16 | vandyk | I have an Asterisk 11 running with medium load of calls per day. Some customers are complaining about call quality, they say that the call is too bad, the sound is low. All calls only use u-law codec. How can I troubleshoot this issues and discover what could be wrong? |
| 14:21.37 | robmal | Packetloss or jitter. |
| 14:22.15 | vandyk | robmal: and how can I identify that? |
| 14:22.39 | robmal | ping |
| 14:24.19 | vandyk | I did a small search now about jitter. It says to use jitter buffer. How can I do that? Probably is a setting on SIP that should be set on both sides, right? |
| 14:24.40 | robmal | You shouldn't use jitterbuffer. |
| 14:24.56 | phpboy | robmal: happy to see you here |
| 14:25.08 | phpboy | I've reinstalled from asterisk 12 to asterisk 13 |
| 14:25.17 | phpboy | identical issues with WebRTC |
| 14:25.28 | robmal | Because it's on the client side ;-> |
| 14:26.27 | robmal | You have to find out where those weird 192.0.x.x addresses come from. |
| 14:27.06 | *** join/#asterisk Kaian (~kaian@6.62-99-78.static.clientes.euskaltel.es) |
| 14:28.24 | phpboy | robmal: I'm onsite now on a different pc |
| 14:28.33 | phpboy | no more funny 192.0 address |
| 14:28.58 | phpboy | what's weird is I see the RTP packets going to my IP but no audio :\ |
| 14:31.51 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 14:38.21 | *** join/#asterisk mac_ified (~mac_ified@67-9-150-210.res.bhn.net) |
| 14:38.37 | robmal | wireshark and debug the rtp stream. |
| 14:40.16 | phpboy | do you have any tips on debugging the rtp stream? |
| 14:40.24 | phpboy | I see the udp packets hitting my PC |
| 14:40.31 | phpboy | but further than that I don't know what to do |
| 14:41.23 | vandyk | robmal: this is the result of sip show channelstats |
| 14:41.26 | vandyk | Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter |
| 14:41.27 | vandyk | 262b8b05-f8 00:04:00 0000010880 0000000092 ( 0.84%) 0.0000 0000010987 0000000879 ( 8.00%) 0.0007 |
| 14:41.27 | vandyk | NjI3Y2FhYjU 00:00:08 0000000254 0000000000 ( 0.00%) 0.0000 0000000113 0000000000 ( 0.00%) 0.0030 |
| 14:41.27 | vandyk | 6af70ee8079 00:01:06 0000002994 0000000000 ( 0.00%) 0.0000 0000003280 0000000001 ( 0.03%) 0.0000 |
| 14:41.28 | vandyk | 0b51895d04c 00:00:08 0000000113 0000000001 ( 0.88%) 0.0000 0000000254 0000000000 ( 0.00%) 0.0000 |
| 14:41.29 | vandyk | 21821254321 00:01:06 0000002378 0000000000 ( 0.00%) 0.0000 0000002402 0000000000 ( 0.00%) 0.0000 |
| 14:41.30 | vandyk | ae52e8-0-13 00:01:06 0000003280 0000000013 ( 0.39%) 0.0000 0000002994 0000000000 ( 0.00%) 0.0037 |
| 14:45.45 | *** join/#asterisk dummys (~dummys@unaffiliated/dummys) |
| 14:46.35 | dummys | hi [TK]D-Fender, I come to you because we speak together about the perfect way to use the sip trunk behind nat |
| 14:46.42 | dummys | and you speak about vpn tunnel |
| 14:47.04 | dummys | my question is, how do you say to asterisk to use the tunnel ? only with ip route ? |
| 14:47.16 | dummys | but with this, all my traffic will be routed thru the vpn |
| 14:47.33 | dummys | there is a way to tell to asterisk to use the tunnel for outgoing call ? |
| 14:47.40 | phpboy | dummys: do you only want traffic to an IP routed via the VPN? |
| 14:47.53 | phpboy | is the server that hosts asterisk doing the actual VPN connection? |
| 14:47.57 | dummys | yes |
| 14:48.09 | phpboy | is it connected to the VPN right now? |
| 14:48.10 | dummys | with openvpn and tun |
| 14:48.12 | dummys | yes |
| 14:48.14 | phpboy | ok |
| 14:48.30 | phpboy | please pastebin the output of this command -> netstat -rn |
| 14:48.41 | phpboy | and which IP should it use via the VPN? |
| 14:48.51 | dummys | in fact I need to do that because when speaking with [TK]D-Fender he told me to do it because is a proper way to do it and will work on every networks |
| 14:48.58 | dummys | ok |
| 14:49.49 | dummys | http://pastebin.com/ivpNG21e |
| 14:50.18 | dummys | I don't want to have inbound calls |
| 14:50.31 | dummys | just that my outbound calls are routed correctly |
| 14:50.58 | phpboy | ok so let me exaplin what's happening here |
| 14:51.00 | dummys | I think if I put the default route for the vpn it will be ok |
| 14:51.08 | dummys | but all my traffic will go thru the vpn |
| 14:51.09 | phpboy | you have set openvpn to route all traffic via the vpn |
| 14:51.13 | dummys | yes |
| 14:51.16 | phpboy | so go into the config and disable that |
| 14:51.35 | phpboy | then add only the route for the IP that your asterisk server uses for it's SIP traffic |
| 14:51.59 | dummys | hmm |
| 14:52.13 | dummys | I think it use the ip of the eth0 |
| 14:52.16 | dummys | 192.168.1.26 |
| 14:52.19 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:52.19 | dummys | it's the dhcp ip |
| 14:52.47 | phpboy | what is the IP your use on the VPN? the sip gateway IP |
| 14:53.11 | dummys | <PROTECTED> |
| 14:53.14 | dummys | <PROTECTED> |
| 14:53.16 | dummys | <PROTECTED> |
| 14:53.17 | dummys | this is the ip of the tun |
| 14:53.23 | phpboy | no |
| 14:53.23 | dummys | the sip gateway is voip.ms |
| 14:53.30 | phpboy | ok |
| 14:53.38 | dummys | yes so I just add a route |
| 14:53.39 | phpboy | so you want voip.ms to route via the vpn? |
| 14:53.41 | dummys | for this one |
| 14:53.43 | dummys | yes |
| 14:53.57 | dummys | with this solution and an iptables route, I must be ok for a optimal SIp setup right ? |
| 14:54.02 | phpboy | ok |
| 14:54.07 | phpboy | no set it up in openvpn |
| 14:54.10 | dummys | iptables rule sorry |
| 14:54.21 | phpboy | in your open vpn config |
| 14:54.25 | dummys | yes |
| 14:54.31 | phpboy | disable adding default route |
| 14:54.35 | phpboy | comment it out |
| 14:54.38 | dummys | the problem it's network-manager of debian doing that |
| 14:54.47 | dummys | I will try to find where I can change it |
| 14:54.54 | Ikikay | sorry, I know it's the wrong channel... but, does someone know jitsi-virtualbridge ? :x |
| 14:55.09 | phpboy | all you need to do is disable default route and add a route for 74.63.41.222 via the vpn |
| 14:55.13 | phpboy | and that will solve your problem |
| 14:55.22 | dummys | yes |
| 14:55.57 | phpboy | your issue is with openvpn and NOT with asterisk |
| 14:56.27 | dummys | yes |
| 14:56.51 | *** join/#asterisk vandyk (~vandyk@191.187.216.136) |
| 14:57.22 | dummys | http://pastebin.com/ZFMvaex6 |
| 14:57.25 | dummys | look better now |
| 14:59.55 | dummys | phpboy: |
| 15:00.00 | dummys | so now I put a route like this: |
| 15:00.03 | dummys | 159.8.85.128 141.101.134.196 255.255.255.192 UG 0 0 0 tun0 |
| 15:06.31 | robmal | Can't you just bind asterisk to the vpn address? |
| 15:09.12 | dummys | hmm i dunno how |
| 15:09.30 | dummys | is it possible to do that ? |
| 15:09.41 | dummys | even if the vpn address is dhcp I think ? |
| 15:11.17 | dummys | bindaddr with this settings robmal ? but is there a way to say bind asterisk to tun0 ? |
| 15:11.48 | robmal | No. |
| 15:12.56 | *** join/#asterisk eofster (~eofster@213.61.153.26) |
| 15:13.10 | dummys | ok so it's better to add a route when the tun0 is up |
| 15:13.12 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 15:15.04 | *** join/#asterisk c0ldg0ld (~c0ldg0ld@unaffiliated/c0ldg0ld) |
| 15:17.06 | c0ldg0ld | so I've established that there is an Asterisk selinux module out there. How would one go about activating it? I'm hoping that it's not a bunch of semange fcontext for all the asterisk directories and rather an all encompasing command for the application |
| 15:17.17 | WIMPy | You don't bind to an interface. You bind to an IP. |
| 15:17.42 | dummys | yes |
| 15:17.56 | dummys | I'm creating python script to retrieve the ip and do the binding and routing |
| 15:18.12 | dummys | WIMPy: did you have some pre configured iptables rule for sip ? |
| 15:18.21 | dummys | like the one we spoke together yesterday |
| 15:18.48 | WIMPy | Err, what was that for? |
| 15:20.31 | WIMPy | Or what do you want to do? |
| 15:21.24 | dummys | just to add rule for iptables to accept sip/rtp port from the vpn |
| 15:22.41 | WIMPy | Unless you blocked it, I don't see a need for special configuration. |
| 15:23.02 | dummys | I will block all incoming from the vpn |
| 15:23.08 | dummys | for security purpose |
| 15:23.12 | dummys | so I need to accept it |
| 15:23.48 | WIMPy | Accept udp/5060 and ESTABLISHED,RELATED. |
| 15:24.14 | [TK]D-Fender | Shouldn't even need that unless you have conntrack on |
| 15:24.25 | [TK]D-Fender | Since otherwise UDP = stateless |
| 15:24.32 | dummys | yes |
| 15:24.32 | [TK]D-Fender | and "established" does not exist |
| 15:24.45 | dummys | and you speak about 10000-20000 right ? |
| 15:25.06 | [TK]D-Fender | about ALL |
| 15:25.40 | dummys | hmm ? |
| 15:25.53 | WIMPy | Ne, but ESTABLISHED does. |
| 15:26.22 | WIMPy | RELATED will take care of RTP ports. |
| 15:26.55 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-uhkftvdyzknjogwb) |
| 15:26.55 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:27.05 | [TK]D-Fender | that's just an "Accept" anyway. So far I haven't seen any policy to REJECT anything |
| 15:35.35 | *** join/#asterisk cyford (~support@c-73-137-1-6.hsd1.ga.comcast.net) |
| 15:35.57 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-pwqztultvfzbevlq) |
| 15:36.22 | *** join/#asterisk zaf (~zaf@76.72.92.37) |
| 15:53.06 | dummys | WIMPy: so only this should be enough ? |
| 15:53.12 | dummys | iptables -A INPUT -p tcp -i eth0 --dport 5060 -j ACCEPT |
| 15:53.36 | [TK]D-Fender | It shouldn't even matter unless you are REJECTING somewhere else |
| 15:53.50 | dummys | http://pastebin.com/pCx2AUJw |
| 15:53.53 | WIMPy | No. |
| 15:53.54 | [TK]D-Fender | and NO, that should not be TCP either way |
| 15:53.57 | dummys | like this is ok ? |
| 15:54.03 | dummys | yes sorry i put udp |
| 15:54.05 | dummys | don't worry |
| 15:54.05 | WIMPy | Unless you want to do SIP via TCP. |
| 15:54.12 | dummys | just a bad cc |
| 15:54.21 | dummys | can you look at the paste please, is it correct for you? |
| 15:54.48 | WIMPy | Looks ok. |
| 15:54.55 | dummys | ok cool |
| 15:55.16 | dummys | I would like to say sorry to you guys, [TK]D-Fender and WIMPy to be rude yesterday with you |
| 15:55.29 | dummys | but i was really thinking that you take me for a noob that didn't want to rtfm and so on |
| 15:55.43 | WIMPy | Now you just have to make sure you have the nf_conntrack_sip module loaded and the nf_nat_sip module isn't loaded. |
| 15:55.54 | dummys | let me check |
| 15:56.02 | WIMPy | I didn't really read much here yesterday. |
| 15:56.12 | dummys | I didn't change the original module file |
| 15:56.18 | dummys | this must be enabled ? |
| 15:56.55 | WIMPy | OS modules, not Asterisk modules. |
| 15:57.08 | dummys | oh yeah |
| 15:57.12 | dummys | it's already ok |
| 15:57.22 | dummys | oh wait you speak about sip |
| 15:57.24 | dummys | let me check |
| 15:57.31 | dummys | it's a normal debian with xfce |
| 15:57.39 | dummys | it's by default or not ? |
| 15:58.01 | dummys | nf_conntrack_ipv4 18448 |
| 15:58.22 | WIMPy | No idea what debian does. Usually not what I want. |
| 15:58.36 | dummys | hmm there is no nf_conntrack_sip |
| 15:59.48 | WIMPy | Then load it or allow the RPT ports manually and statically. |
| 15:59.54 | dummys | nf_conntrack 87424 4 xt_conntrack,nf_conntrack_sip,nf_conntrack_ipv4,nf_conntrack_ipv6 |
| 15:59.56 | dummys | it's ok now |
| 16:08.17 | *** join/#asterisk Voyage (~user1@unaffiliated/voyage) |
| 16:12.48 | Voyage | Hi, |
| 16:13.34 | Voyage | I need some help with finding correct operator() in dialplan. I want to ring only 1 time (one ring tone) and then hangup. |
| 16:14.13 | [TK]D-Fender | you aren't even dialing via the dialplan currently |
| 16:14.49 | Voyage | sorry? |
| 16:14.55 | [TK]D-Fender | Ther is also no such thing as "ring 1 time". calls can be limited by "time", not "rings" |
| 16:15.03 | Voyage | oh ok |
| 16:15.06 | [TK]D-Fender | You are using ORIGINATE. |
| 16:15.22 | [TK]D-Fender | you are not even IN the dialplan until that call-out gets answered |
| 16:15.41 | Voyage | then lets say 3 seconds "after" its starts ringing (being picked up by callee is now irrelevant) |
| 16:15.45 | craigify | He did ask for operator() |
| 16:16.10 | craigify | https://wiki.asterisk.org/wiki/display/AST/Application_Dial |
| 16:16.24 | [TK]D-Fender | craigify: He isn't USING Dial() |
| 16:16.28 | Voyage | [TK]D-Fender> Ther is also no such thing as "ring 1 time". <-- should be as a suggestion |
| 16:18.15 | Voyage | [TK]D-Fender, I am in the extension that is used for dialing.... |
| 16:18.16 | Voyage | [outgoing] |
| 16:18.16 | Voyage | exten => _1NXXXXXXXXX,1,Dial(SIP/${EXTEN}@flowroute) |
| 16:18.16 | Voyage | same => n,Hangup() |
| 16:18.24 | craigify | Voyage: the problem with counting rings is that it's a very ambiguous way to do things. Not all rings are the same. When you're actually writing up telephony applications, you deal with seconds |
| 16:18.34 | Voyage | craigify, ah |
| 16:18.38 | [TK]D-Fender | Voyage: And how are you even calling that? |
| 16:19.09 | *** join/#asterisk sgriepentrog (sgriepentr@nat/digium/x-fnljdhdndurxovjs) |
| 16:19.47 | Voyage | [TK]D-Fender, I think , as far as I have read the book you suggested, by what ever means you call either softphone or CLI or AMI, the call goes to the extension and that ext decides "HOW" to call |
| 16:19.58 | [TK]D-Fender | NO |
| 16:20.05 | Voyage | so, for learning purposes, I am making the ext right |
| 16:20.19 | [TK]D-Fender | [12:18][TK]D-FenderVoyage: And how are you even calling that? |
| 16:20.21 | Voyage | hm |
| 16:20.24 | lvlinux | vandyk: clarify a few things: approximately how many is "medium load"? 10? 100? 500? What type of phones and/or interfaces? What does "call is too bad" mean? Breaking up? Silence? Odd sounds? |
| 16:20.26 | [TK]D-Fender | Show us EXACTLY how you are calling that |
| 16:20.29 | WIMPy | Voyage : You get an event when the called party starts ringing. |
| 16:20.51 | Voyage | WIMPy, ok, how to know "its ringing to the callee"? |
| 16:20.58 | Voyage | [TK]D-Fender, ok |
| 16:21.00 | WIMPy | lvlinux: That all sounds rather meaningless. |
| 16:21.22 | vandyk | lvlinux: 35 calls simultaneous. The issue they say the volume at call is to low nad sometimes there is too much noise on it |
| 16:21.31 | WIMPy | That's what I just wrote. |
| 16:21.44 | Voyage | WIMPy, first I will detect if it IS ringing and then hangup after 3 seconds |
| 16:21.52 | vandyk | they are using SIP phones and softphones |
| 16:22.16 | [TK]D-Fender | No point in detection |
| 16:23.24 | Voyage | [TK]D-Fender, this is how I am calling. originate SIP/+132numberhere7@flowroute extension s@outgoing Any more questions? |
| 16:23.37 | [TK]D-Fender | that has NOTHING to do with Dial() then |
| 16:23.50 | [TK]D-Fender | And you are still using CLI Originate which you should NOT be using |
| 16:24.08 | Voyage | [TK]D-Fender, ok, the said will be true for AMI too? |
| 16:24.13 | [TK]D-Fender | You have a serious problem following instructions |
| 16:24.32 | *** join/#asterisk DragonAzul (~DragonAzu@187.208.14.46) |
| 16:24.33 | Voyage | [TK]D-Fender, stop responding me if you want to start that again |
| 16:24.39 | [TK]D-Fender | originate SIP/+132numberhere7@flowroute <- this is NOT "Dial()" |
| 16:24.45 | Voyage | [TK]D-Fender, sorry to say, we are not on the same page |
| 16:25.08 | [TK]D-Fender | IT IS NOT DIAL() <- Are we on the same page now? |
| 16:25.23 | Voyage | may yes on that but not on [TK]D-Fender> You have a serious problem following instructions |
| 16:25.25 | [TK]D-Fender | You CANNOT tell CLI-based Originate a TIMEOUT |
| 16:25.30 | dummys | voyage voyage tutututututu |
| 16:25.38 | [TK]D-Fender | STOP USING CLI ORIGINATE. Clear? |
| 16:25.44 | [TK]D-Fender | Same page? |
| 16:25.47 | [TK]D-Fender | it has LIMITATION |
| 16:25.59 | Voyage | I am not going to argue with you. Lets take AMI then |
| 16:26.14 | Voyage | can AMI do what I am trying to do? |
| 16:26.28 | [TK]D-Fender | Read the page for AMI Originate |
| 16:26.35 | Voyage | <PROTECTED> |
| 16:26.44 | [TK]D-Fender | What else is there besides ringing? |
| 16:26.49 | Voyage | nothing |
| 16:26.57 | Voyage | ringing for now. thats it |
| 16:27.07 | [TK]D-Fender | [12:22][TK]D-FenderNo point in detection |
| 16:27.20 | [TK]D-Fender | called + not yet answered = ringing |
| 16:27.28 | [TK]D-Fender | So ring for TIME |
| 16:27.31 | Voyage | ok |
| 16:27.32 | Voyage | fine |
| 16:27.35 | Voyage | ring for time |
| 16:27.35 | Voyage | how? |
| 16:27.46 | [TK]D-Fender | Read the instructions for Originate |
| 16:27.52 | [TK]D-Fender | I gave you the page for it already |
| 16:27.56 | WIMPy | If you limit by time, it will cut the call no matter if it even reached ringing stage. |
| 16:27.56 | [TK]D-Fender | Others have as well |
| 16:28.07 | WIMPy | So you have to act on the channelstate event. |
| 16:28.24 | [TK]D-Fender | WIMPy: Realistically he's calling US48 off a decent ITSP |
| 16:28.41 | [TK]D-Fender | WIMPy: Not like we're dealing with international interconnects |
| 16:29.05 | Voyage | if that is one problem. heres what I think happens. you dial > call goes out > it connects callee > it rings > it gets picked up. I want to wait for 3 seconds on step 3 |
| 16:29.16 | [TK]D-Fender | incorrect |
| 16:29.17 | Voyage | " it connects callee" |
| 16:29.22 | WIMPy | I don't see an issue with international calling, but usiong SIP makes things ratehr unpredictable. |
| 16:29.35 | [TK]D-Fender | you do not CONNECT the callee ... and THEN ring |
| 16:30.07 | Voyage | [TK]D-Fender, how do you draw the chat? |
| 16:30.12 | WIMPy | Connecting is done after (optional) ringing. |
| 16:30.17 | [TK]D-Fender | Your ORIGINATE creates a channel. ITSP rings (or not) for your amount of time or until answered. |
| 16:30.31 | lvlinux | vandyk: are you using all VoIP or is there PSTN inteface involved? |
| 16:30.32 | [TK]D-Fender | After answer it gets dumped to the place you told it to dump to. |
| 16:30.41 | Voyage | WIMPy, oh so in *, connected means when callee picks up? |
| 16:30.56 | [TK]D-Fender | ANSWERED. |
| 16:31.07 | Voyage | WIMPy, oh so in *, connected means when callee answered? |
| 16:31.08 | [TK]D-Fender | "connected" = vague choice of words |
| 16:31.15 | [TK]D-Fender | thus bad |
| 16:31.16 | vandyk | lvlinux: my customers connect to my Asterisk server and I have GSM gateways to complete the calls |
| 16:31.17 | WIMPy | Not only with Asterisk. |
| 16:31.39 | Voyage | ok, so its ringing > answered |
| 16:31.53 | Voyage | dial > ring > answered |
| 16:32.08 | Voyage | I want to wait for 3 seconds on step 2 "ringing" |
| 16:32.16 | Voyage | and then hangup() |
| 16:32.42 | [TK]D-Fender | Originate > Channel: called > (maybe get ringing, still just waiting for answer) > Is time up, if not keep waiting for answer > ANSWER > Goto Dialplan/Application |
| 16:32.42 | Voyage | WIMPy, this is an automated call by AMI or CLI not by soft/hard phone |
| 16:32.58 | [TK]D-Fender | So tell it to wait LESS for answer |
| 16:32.58 | WIMPy | told you how to do that. |
| 16:33.03 | Voyage | [TK]D-Fender, ok |
| 16:33.15 | lvlinux | vandyk: ah ok. Methinks your low volume problem may be related to your GSM gateways possibly. I would think the noise would be there too. |
| 16:33.24 | WIMPy | Even though that smells quite extreme like some illegal activity. |
| 16:33.49 | Voyage | [TK]D-Fender, I believed those instructions to wait/hangup etc would be in the [extension] either I do AMi or CLI |
| 16:33.58 | lvlinux | WIMPy: hehe depends on where he is. |
| 16:34.03 | [TK]D-Fender | And I told you 10 times "no" |
| 16:34.06 | [TK]D-Fender | CHANNEL gets called |
| 16:34.10 | vandyk | livlinux: I don't think so, because I have 64 channels at total, and there are 3 distinct gsm gateways |
| 16:34.15 | Voyage | WIMPy, ya, but I just want to understand things, experiment |
| 16:34.16 | [TK]D-Fender | Everything else is AFTER the call is actually answered |
| 16:34.28 | vandyk | I mean lvlinux: I don't think so, because I have 64 channels at total, and there are 3 distinct gsm gateways |
| 16:34.29 | WIMPy | Voyage: No dialplan so far. |
| 16:35.03 | Voyage | ok, WIMPy where do I need to put those instructions in case I am doing it by CLI , AMI |
| 16:35.13 | Voyage | will be back in 5 |
| 16:35.34 | WIMPy | Read what I told you earlier. |
| 16:35.36 | [TK]D-Fender | Voyage: in the ORIGINATE. Told you repeatedly, it is in the ORIGINATE ITSELF. Read the instructions again |
| 16:36.31 | lvlinux | vandyk: You can try increasing your volume, but that wouldn't fix the noise. |
| 16:39.05 | WIMPy | Great. Looks like my soldering iron just broke :-( |
| 16:39.45 | lvlinux | WIMPy: a nice one? |
| 16:40.01 | lvlinux | WIMPy: what are you soldering? |
| 16:41.05 | WIMPy | No, just a simple regulated one. Have got the station over now. Doing a 12V power distribution box. |
| 16:41.26 | lvlinux | ah, ok. You a ham operator? |
| 16:41.34 | WIMPy | Nope. |
| 16:41.46 | WIMPy | It's mainly for lighting. |
| 16:41.55 | lvlinux | what kind of lighting? |
| 16:42.15 | WIMPy | gives lvlinux one guess. |
| 16:42.19 | lvlinux | video? |
| 16:42.46 | WIMPy | Oh, that "kind". No, room lighting. |
| 16:43.30 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 16:44.28 | lvlinux | admires other people that use DIY electronic skills :-) |
| 16:47.37 | *** join/#asterisk trewq (~trewq@li70-4.members.linode.com) |
| 16:47.40 | lvlinux | |?"}[1;2A[1;2B |
| 16:47.51 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:48.14 | lvlinux | lol sry wow that was close---I knocked my laptop off the top of a ladder and barely saved it by grabbing the keyboard... |
| 16:48.39 | WIMPy | What is your Laptop doing on a ladder? |
| 16:49.05 | trewq | hi Folks - I've been using voip.ms and they are great, but I want to have some more control over voice lines.. If I were to use asterisk, I imagine I need to use some SIP provider - Is there someone reliable that you guys use with asterisk? |
| 16:49.41 | lvlinux | WIMPy: I'm working on a phone sys at a church and have a console cable connected to a Cisco switch and an Adtran FXS box...which are mounted in the ceiling. |
| 16:50.12 | WIMPy | Get a longer cable. Or use bluetooth. |
| 16:50.21 | Voyage | [TK]D-Fender, WIMPy ok. What i understood from the conversation is that if I am using normal soft/hard phones, the thing I want to do would be configured in dialplan extension but if I am using CLI or AMI to originate calls, I need to look into origination and I assume all or most of the things done in [extensions] can be done in origination as well. (regarding dailing applications itc). I will come back after reading about origination form the book |
| 16:50.47 | lvlinux | WIMPy: lol well when I get IP connectivity setup I'll just ssh into them. |
| 16:50.59 | [TK]D-Fender | You should not be "assuming". We have explained how Originate works over a dozen times now |
| 16:51.01 | Voyage | craigify, thanks |
| 16:51.03 | lvlinux | or telnet (gasp) for the Adtran lol... |
| 16:51.21 | [TK]D-Fender | Dialplan does not EXIST until the call is actually answered |
| 16:51.24 | WIMPy | Voyage: No. the calling is done in one place ot the other. What happens after the call is answered is always dialplan. |
| 16:51.29 | Voyage | [TK]D-Fender, hm |
| 16:51.38 | Voyage | got it WIMPy |
| 16:51.42 | [TK]D-Fender | How many more times are we going to have to tell you? |
| 16:51.46 | lvlinux | trewq: voip.ms works fine with Asterisk. Any other provider will be similar |
| 16:51.54 | Voyage | [TK]D-Fender, respect my newbieness |
| 16:52.13 | Voyage | [TK]D-Fender, I am bloody TRYING to follow your sarcastic help. |
| 16:52.21 | [TK]D-Fender | I am not being sarcastic |
| 16:52.27 | [TK]D-Fender | Everything I have told you is literal |
| 16:52.39 | Voyage | you are. and also disrespectful. Help me does not give you that right either |
| 16:52.51 | Voyage | [TK]D-Fender, ok. let me read the book then |
| 16:52.59 | lvlinux | Voyage: newbiness has nothing to do with following instructions and doing your own research when it is clearly laid out for you. If you are having trouble following what TK is saying, it shows that you haven't really read much in the * book. |
| 16:53.29 | Voyage | lvlinux, I have problems understanding where to go/read. anyways. nvm |
| 16:54.16 | [TK]D-Fender | You were told repeatedly that dialplan does no exist until after than Channel: gets answered |
| 16:54.24 | Voyage | ok fine |
| 16:54.25 | [TK]D-Fender | This has nothing to do with "newb" |
| 16:54.28 | Voyage | fine. |
| 16:54.31 | lvlinux | Voyage: TK isn't being overly hard on you---he's just frustrated. You need to read the book. You can skip the part about installation, but read the rest, especially the core concepts like the dialplan until you fully understand it. |
| 16:54.32 | WIMPy | Voyage: that's normal. You read and then you understand. It doesn't work the other way round. |
| 16:54.34 | [TK]D-Fender | we told you a SIMPLE FACT |
| 16:54.41 | [TK]D-Fender | 10 times or more |
| 16:54.44 | Voyage | ok STOP |
| 16:55.11 | Voyage | WHAT ARE YOU TRYING TO PROOF. ? whatever it is , i admit it in advance. CAN I GO READ ORIGINATION PLEASE? |
| 16:55.14 | [TK]D-Fender | If you want to restrict how long that call goes out go then you have to do it in the Originate. |
| 16:55.34 | Voyage | LOOK, I DONT HAVE TIME TO PROOVE MY SELF |
| 16:55.38 | Voyage | WHAT ARE YOU TRYING TO PROOF. ? whatever it is , i admit it in advance. CAN I GO READ ORIGINATION PLEASE? |
| 16:56.05 | [TK]D-Fender | Go read |
| 16:56.48 | WIMPy | Voyage: My impression is that you don't have time to learn, but do have time to argue. |
| 16:57.03 | Voyage | WIMPy, good. |
| 16:57.09 | craigify | heh |
| 16:57.15 | WIMPy | No. Bad. |
| 16:57.20 | Voyage | WIMPy, can can I step back from this chat and do something now? |
| 16:57.32 | Voyage | WIMPy, ok, bad. happy? |
| 16:57.39 | WIMPy | See. That's exactely the point. |
| 16:57.49 | [TK]D-Fender | Why are you still chatting in here? You said you were going to go read. Just stop and do it. |
| 16:57.57 | [TK]D-Fender | And come back with some new questions |
| 16:57.58 | WIMPy | Stop arguing and DO it. It's YOUR choice! |
| 16:58.13 | [TK]D-Fender | Hopefully more informed..... |
| 16:58.14 | Voyage | WIMPy, [TK]D-Fender what ever you are doing and saying is correct. |
| 16:58.36 | [TK]D-Fender | Go read. |
| 16:58.44 | Voyage | [TK]D-Fender, I know what to do |
| 16:58.53 | [TK]D-Fender | Good. Go do it. |
| 16:58.58 | Voyage | [TK]D-Fender, I know what to do |
| 16:59.04 | Voyage | or go or not |
| 17:04.24 | dummys | omg |
| 17:04.28 | dummys | lol Voyage |
| 17:04.49 | *** join/#asterisk cyford (~support@c-73-137-1-6.hsd1.ga.comcast.net) |
| 17:04.50 | dummys | you are taking one of the most ownage I ever seen in an irc channel |
| 17:06.06 | *** join/#asterisk vinrock (~vin@unaffiliated/vinrock) |
| 17:06.09 | Voyage | dummys, I got pissed of man. they could just say in the start "dialplan comes after the call is connected and not before - you need to read origination" |
| 17:06.17 | Voyage | thats one line answer |
| 17:06.46 | Voyage | everyday, i get " you dont read, you dont folow instructions, you are nt smart, you dont read we say" |
| 17:07.06 | [TK]D-Fender | [12:32][TK]D-FenderOriginate > Channel: called > (maybe get ringing, still just waiting for answer) > Is time up, if not keep waiting for answer > ANSWER > Goto Dialplan/Application |
| 17:07.10 | Voyage | and I get CAPS words like abusive thigns. e.g FUKCING |
| 17:07.12 | [TK]D-Fender | And I gave uyou exactly that in 1 line |
| 17:07.26 | [TK]D-Fender | You were given exactly that |
| 17:07.27 | Voyage | after a 3 dozen lines. yes |
| 17:07.29 | WIMPy | Voyage: I'm pretty sure I DID tell you 3 days ago. |
| 17:07.42 | Voyage | ah. you guys again win |
| 17:07.52 | Voyage | wonders why he even replied to dummys |
| 17:07.56 | [TK]D-Fender | [12:51][TK]D-FenderDialplan does not EXIST until the call is actually answered |
| 17:07.56 | Voyage | steps back |
| 17:08.00 | [TK]D-Fender | Told you here too |
| 17:08.08 | lvlinux | Voyage: it has nothing to do with being smart. You can be the most intelligent person in the world, but you have to be able to take instructions and be humble enough to take some hard words when someone else is taking time to help you. |
| 17:08.29 | [TK]D-Fender | We were clear |
| 17:08.43 | [TK]D-Fender | And said exactly what you now say we should have |
| 17:08.49 | [TK]D-Fender | And that's just today's chat |
| 17:09.28 | Voyage | lvlinux, i really appreicate all the help i am geting. from bottom of my heart. but being harassed is not impressing me either |
| 17:09.54 | Voyage | wel. I should really stop responding. no offence. |
| 17:11.33 | [TK]D-Fender | Voyage: Now that you just went to read up on Originate again do you have new questions on it? |
| 17:15.38 | Voyage | [TK]D-Fender, do you think I will ask if I have any more questions |
| 17:15.58 | [TK]D-Fender | No idea. |
| 17:16.10 | [TK]D-Fender | I'm trying to see if you're just looking to move forward |
| 17:18.42 | *** join/#asterisk theron (~theron@2620:10d:c090:200::e:4179) |
| 17:21.37 | *** join/#asterisk znf (~ibm86@unu.card-sharing.eu) |
| 17:24.44 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 17:29.50 | *** join/#asterisk ThomasKeller (~Thomas@vmx.ethz.ch) |
| 17:39.05 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:47.10 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 17:47.18 | Voyage | looked at https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_Originate and p 572 of asterisk definitive guide book. Cant understand how to limit call time to 3 seconds and hangup. |
| 17:48.05 | [TK]D-Fender | That is Application Originate |
| 17:48.08 | [TK]D-Fender | that is not AMI Originate |
| 17:48.10 | [TK]D-Fender | Not the same |
| 17:48.15 | [TK]D-Fender | just like CLI Originate is not the same |
| 17:48.20 | [TK]D-Fender | that is not the page you should be reading |
| 17:48.49 | [TK]D-Fender | I gave you this link for this yesterday and probably earlier as well |
| 17:48.50 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_Originate |
| 17:49.06 | [TK]D-Fender | Actually bookmark it. |
| 17:49.09 | Voyage | ok. going to read that |
| 17:49.59 | Voyage | oh, what ever is in that page, I read similar in the book. p 572 |
| 17:50.35 | Voyage | maybe I need Data - Data to use (requires Application). but how, I dont know |
| 17:50.55 | lvlinux | well, they may be similar in that they both originate calls, but different in what you can do with them. |
| 17:51.13 | Voyage | maybe Timeout - How long to wait for call to be answered (in ms.). |
| 17:51.48 | [TK]D-Fender | Voyagemaybe I need Data - Data to use (requires Application). but how, I dont know <- you don't |
| 17:52.36 | [TK]D-Fender | AI answered this in your question yesterday about the fixed word "extension" in CLI originate |
| 18:01.16 | *** join/#asterisk jonno11 (~Jon@cpc1-walt12-2-0-cust582.13-2.cable.virginm.net) |
| 18:03.46 | *** join/#asterisk ModFather (~ModFather@unaffiliated/modfather) |
| 18:11.47 | ModFather | hi robmal :) |
| 18:12.11 | Voyage | [TK]D-Fender, can we talk without hints and riddles/ |
| 18:13.00 | [TK]D-Fender | I haven't given any |
| 18:13.15 | [TK]D-Fender | I told you what that fixed word "extension" meant in relation to CLI Originate |
| 18:13.26 | Voyage | ya you did |
| 18:13.37 | [TK]D-Fender | ALL originates have one of two destinations |
| 18:13.58 | [TK]D-Fender | Either a single application, or a fully qualified target in the dialplan |
| 18:14.12 | Voyage | right, I know that. |
| 18:14.22 | [TK]D-Fender | [13:51][TK]D-FenderVoyagemaybe I need Data - Data to use (requires Application). but how, I dont know <- you don't |
| 18:14.34 | [TK]D-Fender | You don't need Datat because you aren't going to be using Application |
| 18:14.38 | [TK]D-Fender | this is not a riddle |
| 18:14.45 | Voyage | ok |
| 18:14.47 | [TK]D-Fender | I told you you would only be sending your cal into the dialplan |
| 18:14.48 | Voyage | then I need timout? |
| 18:15.03 | [TK]D-Fender | What does Timeout say it does? |
| 18:15.14 | Voyage | <PROTECTED> |
| 18:15.31 | Voyage | that means, its not in the dialplan yet |
| 18:15.39 | Voyage | oh wait. it is |
| 18:15.39 | [TK]D-Fender | So if you want to ring for a limited amount of time, then that is telling you extremely clearly that this is the setting for that |
| 18:15.47 | Voyage | wait its not |
| 18:16.12 | Voyage | [TK]D-Fender, ok, what is the setting for it then? |
| 18:16.22 | [TK]D-Fender | [14:15][TK]D-FenderSo if you want to ring for a limited amount of time, then that is telling you extremely clearly that this is the setting for that |
| 18:16.39 | [TK]D-Fender | Voyage Timeout - How long to wait for call to be answered (in ms.). <- |
| 18:16.45 | Voyage | ah |
| 18:17.02 | Voyage | Question: |
| 18:17.22 | Voyage | The call goes to dialplan AFTER it had been answered or after it had been dialed? |
| 18:17.37 | [TK]D-Fender | AFTER answered |
| 18:17.43 | Voyage | hmmmm |
| 18:17.57 | Voyage | thanks for being informative |
| 18:18.06 | [TK]D-Fender | I did tell you that a lot |
| 18:18.06 | Voyage | gets back to read |
| 18:18.35 | Voyage | sir, either you didn't, or you did and I missed (sorry for that), or i mixed up things. either way. thanks |
| 18:19.00 | Voyage | is trying to learn.. trying to try to try learning |
| 18:20.25 | [TK]D-Fender | I didn't? |
| 18:20.28 | [TK]D-Fender | [12:15][TK]D-Fenderyou are not even IN the dialplan until that call-out gets answered |
| 18:20.30 | [TK]D-Fender | [12:34][TK]D-FenderEverything else is AFTER the call is actually answered |
| 18:20.31 | [TK]D-Fender | [12:51][TK]D-FenderDialplan does not EXIST until the call is actually answered |
| 18:20.33 | [TK]D-Fender | [12:51]WIMPyVoyage: No. the calling is done in one place ot the other. What happens after the call is answered is always dialplan. |
| 18:20.34 | [TK]D-Fender | [12:54][TK]D-FenderYou were told repeatedly that dialplan does no exist until after than Channel: gets answered |
| 18:20.36 | [TK]D-Fender | [13:07][TK]D-Fender[12:51][TK]D-FenderDialplan does not EXIST until the call is actually answered |
| 18:20.44 | [TK]D-Fender | There is 7 times in the past 2 hours alone you were told |
| 18:21.04 | [TK]D-Fender | And from multiple people |
| 18:21.22 | Voyage | ok. you win again |
| 18:21.50 | [TK]D-Fender | Let's move on. |
| 18:21.54 | Voyage | :) |
| 18:22.37 | robmal | Voyage: You're paying [TK]D-Fender to be this patient, right? |
| 18:22.43 | Voyage | [TK]D-Fender, you should know I dont have ALLLLL terminologies in my head yet. And I cant do that before a month. |
| 18:22.47 | [TK]D-Fender | No, we're all paying |
| 18:22.52 | Voyage | robmal, I might pay someont to be impatient on him though |
| 18:22.55 | [TK]D-Fender | Voyage: You were told 7 rtimes in 2 hours |
| 18:23.28 | Voyage | [TK]D-Fender, you should know I dont have ALLLLL terminologies in my head yet. And I cant do that before a month. e.g call connected vs call answered vs call picked up. just on example |
| 18:23.31 | *** join/#asterisk mjordan (mjordan@nat/digium/x-fbbaxfdfpdyzmkvi) |
| 18:23.31 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 18:23.36 | Voyage | eh |
| 18:24.19 | [TK]D-Fender | VoyageThe call goes to dialplan AFTER it had been answered or after it had been dialed? <- this EXACT thing was what we answered extremely specifically repeatedly today alon. We also did all this yesterday. There is no excuse to say we didn't clearly tell you |
| 18:24.30 | Voyage | ...... |
| 18:24.41 | [TK]D-Fender | So continuing on.... |
| 18:25.19 | drmessano | What happens in a month? |
| 18:25.30 | drmessano | Are you being paid to set this up? |
| 18:25.34 | [TK]D-Fender | Black Friday? |
| 18:25.49 | [TK]D-Fender | Advent? |
| 18:26.10 | lvlinux | Squirrel season! |
| 18:26.29 | [TK]D-Fender | In what territory? |
| 18:26.41 | drmessano | It sounds like someone took a job setting this up and has resorted to IRC to get them to do it for him |
| 18:26.57 | [TK]D-Fender | I know just they guy he should have drinks with.... |
| 18:27.19 | lvlinux | lol actually it's been squirrel season here (TN) for a while but we always wait till it's miserable cold to go... |
| 18:27.46 | [TK]D-Fender | lvlinux: Because death should always be a somber thing... |
| 18:28.08 | lvlinux | :-) |
| 18:28.17 | Voyage | I have been years in irc and never saw this attitude |
| 18:28.41 | lvlinux | Voyage: who's attitude? TK? |
| 18:32.03 | lvlinux | Voyage: TK is helping you. He's not perfect but he is giving you accurate and followable information. Like many on IRC and elsewhere, he does expect cooperation and extra effort on your part. |
| 18:32.20 | [TK]D-Fender | let's not even ask for "extra". |
| 18:32.27 | lvlinux | lol |
| 18:32.32 | [TK]D-Fender | I tell people specifically where to go and they go looking at everything else |
| 18:33.11 | [TK]D-Fender | I answer specific question extremely clearly and specifically and get told "why did you just tell me this?". |
| 18:33.54 | [TK]D-Fender | I literally hand the links to the specific pages for documentation and then come back to find they aren't reading them and are somewhere else |
| 18:34.33 | Voyage | lvlinux, I REALY appreciate all including [TK]D-Fender helping me. Its PRICELESS. but .. |
| 18:36.08 | Penguin | I'm glad I didn't also try to help. He would have probably killed himself. |
| 18:36.39 | Voyage | :) |
| 18:37.49 | Voyage | ok, timeout works, now deleteing any timout in origination and I am making sure (in dialplan) that the call gets hungup() after 5 seconds. (no matter what). Will read and get back |
| 18:38.10 | Voyage | next, play some recording. ask for press 1/2/3.. |
| 18:40.01 | Voyage | ah, great, I dont have any sound file. I wonder why * didnt installed any while I installed it via ubuntu repositories |
| 18:40.23 | [TK]D-Fender | because it probably isn't all in one package |
| 18:40.31 | [TK]D-Fender | And specific sounds is not a requirement |
| 18:44.02 | Voyage | hm |
| 18:45.38 | [TK]D-Fender | free hint : http://packages.ubuntu.com/search?keywords=asterisk |
| 18:46.19 | Voyage | apt-get install asterisk-core-sounds-en-wav |
| 18:46.48 | Voyage | still :/var/lib/asterisk/sounds# ls |
| 18:46.48 | Voyage | custom |
| 18:48.49 | [TK]D-Fender | Where does the package say it puts them? |
| 18:49.09 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 18:49.09 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 18:49.16 | Voyage | [TK]D-Fender, hm let me see |
| 18:49.27 | Voyage | should be /var/lib/asterisk/sounds but let me see |
| 18:49.29 | *** join/#asterisk jroseAtDigium (jon@nat/digium/x-bznjgncxvxmwknfd) |
| 18:49.37 | [TK]D-Fender | re-evaluate "should be". |
| 18:49.45 | craigify | Voyage: what would you do without irc? |
| 18:49.46 | craigify | heh |
| 18:49.56 | [TK]D-Fender | Welcome to Debian. Puttuing things where they feel like, not necessarily the author |
| 18:50.59 | ldc | [ot] I've managed to fit an entire asterisk PBX plus sounds and codecs (except g729) in a 16 MB flash, 32 MB RAM device |
| 18:51.28 | ldc | the largest thing was the default /etc/asterisk :p |
| 18:51.32 | craigify | what cpu architecture? |
| 18:51.36 | lvlinux | ldc: cool! |
| 18:51.37 | craigify | intel? |
| 18:51.48 | Voyage | ./usr/share/asterisk/sounds/ |
| 18:51.48 | ldc | dual core bmips |
| 18:51.49 | ldc | BCM6358KFBG SoC |
| 18:51.59 | craigify | 32MB |
| 18:52.01 | craigify | that's crazy |
| 18:52.19 | lvlinux | yes that's kindof scary :-) |
| 18:52.34 | lvlinux | what exactly are you trying to do? or just for fun? |
| 18:53.01 | ldc | lvlinux: oh just for fun. I've had this MIPS board laying around for too long :) |
| 18:53.14 | ldc | tested 4-5 concurrent calls just fine, of course high cpu usage if performing translations |
| 18:53.26 | ldc | but no drops |
| 18:54.17 | lvlinux | cool. That SoC is an ADSL transceiver isn't it? What format do you have? a dev board? |
| 18:55.05 | ldc | lvlinux: it was from a very old Telecom Italia do-it-all box, you know those CPE devices that perform router, firewall, wireless, SIP ATA, whatever |
| 18:55.29 | ldc | and ADSL modem too |
| 18:55.54 | [TK]D-Fender | Plenty of people doing this with crappy Linksys routers, etc for a long time now. |
| 18:56.05 | lvlinux | yep. wow that's neat. reminds me of the days when I ran linux with X windows on a 486dx33 laptop with 3.5MB RAM. |
| 18:56.27 | [TK]D-Fender | sure it's possible, but a painful amount to sacrifice. If you have the gear I suppose "why not" |
| 18:56.44 | lvlinux | at the time that was the best machine I had. |
| 18:56.52 | [TK]D-Fender | "640K ought to be enough for anybody" |
| 18:56.55 | ldc | yeah, nothing I'd put in production use of course, but a nice experiment |
| 18:56.57 | lvlinux | lol |
| 18:57.02 | craigify | I remember the first time I got X to run |
| 18:57.08 | craigify | I had 4MB of total ram |
| 18:57.10 | [TK]D-Fender | I remember ... before X |
| 18:57.18 | craigify | I just had X running |
| 18:57.20 | craigify | no window manager |
| 18:57.26 | [TK]D-Fender | "mega"? What are you the gov't? |
| 18:57.28 | craigify | and all it did was thrash my swap partition |
| 18:57.34 | drmessano | lol |
| 18:57.51 | craigify | this might have been on slackware |
| 18:57.59 | craigify | and linux kernel 1.2 maybe |
| 18:58.00 | lvlinux | craigify: yes that was about what mine was like. took 15 minutes or so to open Netscape Navigator lol |
| 18:58.07 | craigify | trying to remember.... |
| 18:58.21 | lvlinux | i was at 2.0 i think |
| 18:58.22 | [TK]D-Fender | Nutscrape *shudder* |
| 18:58.26 | lvlinux | yup |
| 18:58.33 | craigify | wait |
| 18:58.35 | craigify | 4MB ram |
| 18:58.38 | craigify | is tht even right? |
| 18:58.42 | craigify | I guess it is |
| 18:58.45 | craigify | it seems so small |
| 18:58.46 | craigify | heh |
| 18:58.58 | lvlinux | mine only had 3.8MB or so. |
| 18:59.10 | craigify | why such an odd number> |
| 18:59.15 | craigify | something else using it? |
| 18:59.21 | craigify | I didn't think that went on then |
| 18:59.35 | craigify | video cards using system ram, etc... |
| 18:59.45 | lvlinux | not sure, it was a toshiba 1910 and that's what it had available. Maybe there was some sort of BIOS junk using the extra. I'm sure it was nominally 4MB |
| 19:00.02 | drmessano | I remember booting QNX from a Floppy disk |
| 19:00.15 | drmessano | and browsing the internet |
| 19:00.20 | ldc | ahh the two bootable QNX demo disks :D |
| 19:00.21 | craigify | well, it could be that the same thing happened for me, but I honestly don't remember if all of that 4MB was available directly to ths OS |
| 19:00.22 | drmessano | So elite |
| 19:00.29 | drmessano | ldc, you bet |
| 19:01.11 | craigify | does anybody remember something that could have been called TSX-11? It was some kind of os that implemented MS-DOS but had multitasking? |
| 19:01.19 | ldc | I'm still using QNX on my phone, although it's being phased out in favour of Android :( |
| 19:01.46 | craigify | I guess it implemented the DOS API (is that a thing?) on top of something else |
| 19:02.31 | Voyage | Is there anyything illogical here? http://pastie.org/10514196 |
| 19:02.42 | [TK]D-Fender | craigify: I user I used to use DESQview for that.... |
| 19:03.38 | Voyage | when i call (my skype number), it rings, I pickup and I hear my own skype (callee) call recording instead of hello world.. |
| 19:04.04 | lvlinux | ldc: QNX? on what phone? |
| 19:04.12 | lvlinux | bbos? |
| 19:04.14 | ldc | lvlinux: BlackBerry Classic |
| 19:04.18 | ldc | yes, BB OS 10 (RIP) |
| 19:04.27 | [TK]D-Fender | Voyage: You are calling right back out there. |
| 19:04.35 | lvlinux | ldc: it could have been so nice... |
| 19:04.38 | [TK]D-Fender | Voyage: You are not "just playing a message" |
| 19:04.39 | Voyage | [TK]D-Fender, sorry? |
| 19:04.49 | [TK]D-Fender | [15:04][TK]D-FenderVoyage: You are calling right back out there. <- what's unclear? |
| 19:04.53 | craigify | Voyage: you'll never get to the part after Dial |
| 19:04.55 | *** join/#asterisk ModFather (~ModFather@unaffiliated/modfather) |
| 19:05.01 | [TK]D-Fender | After your callee answers you are issuing a DIAL |
| 19:05.13 | [TK]D-Fender | Why are you doing that? |
| 19:05.20 | craigify | actually |
| 19:05.23 | craigify | yeah why are you doing that? |
| 19:05.30 | Voyage | [TK]D-Fender, You are calling right back out there. <-- am... i am calling my other phone (for testing). Is that what you mean? yes then. |
| 19:05.45 | craigify | Originate takes the place of a human |
| 19:05.49 | Voyage | [TK]D-Fender, OH. got it |
| 19:06.00 | craigify | you can Originate, hit the dialplan, then call another person and bridge the call |
| 19:06.01 | craigify | if you want |
| 19:06.02 | [TK]D-Fender | same => n,Playback(/usr/share/asterisk/sounds/en/hello-world) |
| 19:06.02 | Voyage | [TK]D-Fender, I got my mistake. |
| 19:06.08 | [TK]D-Fender | This is nto goign to happemn in the background |
| 19:06.13 | [TK]D-Fender | Dial is a blocking app |
| 19:06.17 | [TK]D-Fender | Like almost all |
| 19:06.26 | *** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com) |
| 19:07.31 | Voyage | [TK]D-Fender, so whats happening is -> I call via AMI, it calls the number, but when the call is answered, it hits dialplan [outgoing]. Now the first line says DIAL, it dials again to the same number, this time the voice recording speaks up (as th e skype number is already on the first call). So I am having 2 calls. |
| 19:07.35 | Voyage | correct? |
| 19:07.46 | [TK]D-Fender | Dialing the same number again is pointless. |
| 19:07.56 | Voyage | right. it was a mistake. |
| 19:08.29 | [TK]D-Fender | You said you wanted a message. You put a dial first. |
| 19:09.09 | [TK]D-Fender | Technically you are cannibalizing taht other dilplan taht was provided as a sample for how to let a "phone user" cdial to go out your provider. |
| 19:09.48 | [TK]D-Fender | that context was meant for phone-type user's to use |
| 19:09.54 | [TK]D-Fender | not for your automated process |
| 19:09.57 | Voyage | right |
| 19:10.00 | [TK]D-Fender | something we did go over before |
| 19:10.16 | Voyage | right. |
| 19:10.25 | craigify | The way you have it, you could originate a call to person A, then have asterisk call person B and bridge the call |
| 19:10.33 | Voyage | I am going to put only 2 lines in [outgoing] now. |
| 19:10.34 | Voyage | exten => playm,1,Playback(/usr/share/asterisk/sounds/en/hello-world) |
| 19:10.34 | Voyage | same => n,Hangup() |
| 19:11.04 | Voyage | craigify, yes. now I get it. as [TK]D-Fender said it was s default config for phone/soft/hard dialing |
| 19:11.15 | [TK]D-Fender | a SAMPLE |
| 19:11.32 | [TK]D-Fender | let's see how round 5001 goes |
| 19:11.37 | Voyage | sample. yes |
| 19:11.45 | Voyage | round 50001? |
| 19:11.47 | craigify | why don't you just pay him to write this for you? |
| 19:12.10 | Voyage | [TK]D-Fender, that made me wonder how I got out of PSTN balance so quick yesterday :( |
| 19:12.22 | [TK]D-Fender | 5001 was mild sarcasm. I've pre-payed this one. |
| 19:12.36 | Voyage | craigify, coz I dont want [TK]D-Fender to earn from me at least |
| 19:12.43 | craigify | lol |
| 19:12.46 | Voyage | :) |
| 19:13.42 | *** join/#asterisk Hypfer (~hypfer@unaffiliated/hypfer) |
| 19:13.56 | Hypfer | is there any sip client for linux which tells me at least SOMETHING about the connection status? |
| 19:14.17 | Hypfer | jitsy sflphone ekiga linphone. they all just assume that everything works and I don't need to know |
| 19:14.18 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-iiambygmzhledgjt) |
| 19:14.18 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 19:14.29 | Hypfer | whats the recommended software client for an asterisk? |
| 19:14.32 | craigify | zoiper.com ? |
| 19:14.51 | lvlinux | yate |
| 19:15.15 | lvlinux | but mainly you can get the most info from your * console |
| 19:15.43 | Hypfer | lvlinux: but I'll need some monitoring or something |
| 19:16.03 | craigify | what do you want to monitor? |
| 19:16.05 | lvlinux | Hypfer: huh? what do you mean? |
| 19:16.21 | Hypfer | also, at work we have a unify pbx whatever. with that I can dial numbers in the softclient while my hardware phone dials. can astersik do that? |
| 19:16.33 | Hypfer | lvlinux: well.. notify me when there is no connection and I won't receive calls |
| 19:16.43 | Hypfer | like the indicator every normal gsm phone has |
| 19:17.04 | lvlinux | you can setup asterisk to do that for you. |
| 19:17.04 | craigify | Zoiper will tell you if it cannot connect to Asterisk |
| 19:17.15 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-snrvdizccyknqwri) |
| 19:17.15 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 19:17.25 | craigify | a red box pops up in the UI |
| 19:17.31 | Hypfer | oh and hardware phone.. whats the recommendation for a hardware phone? |
| 19:17.45 | craigify | there's all kind of vendors |
| 19:17.49 | Voyage | [TK]D-Fender, UNBELIEVABLE |
| 19:17.51 | Hypfer | i'll try yate and zoiper |
| 19:18.03 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-zbvflwhnurbmpuvs) |
| 19:18.03 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 19:18.04 | Hypfer | is it possible to dial on my computer and use the phone? |
| 19:18.18 | lvlinux | Hypfer: I like Polycom, Yealink, and Snom. But to each his own. |
| 19:18.21 | lvlinux | yes you can do that. |
| 19:18.23 | [TK]D-Fender | what phone? |
| 19:18.32 | lvlinux | he means a hardphone |
| 19:18.33 | craigify | I've been wanting to try Yealink's portable sip phones |
| 19:18.33 | Hypfer | [TK]D-Fender: to be decided and bought |
| 19:18.50 | [TK]D-Fender | Why would you be dialing on a computer? |
| 19:18.56 | Voyage | [TK]D-Fender, I heard hello world. |
| 19:19.00 | lvlinux | [TK]D-Fender: click-to-call, etc. |
| 19:19.04 | lvlinux | Voyage: yay! |
| 19:19.14 | *** join/#asterisk jroseAtDigium (jon@nat/digium/x-qvtsaayphoxpbhpc) |
| 19:19.42 | craigify | Hypfer: Perhaps a commercial PBX that implements Asterisk would have what you want out of the box |
| 19:20.00 | Hypfer | craigify: but i'm just a single student living in a small flat :-) |
| 19:20.03 | lvlinux | or you could hire me to configure it :-) |
| 19:20.09 | craigify | otherwise, you can do all of those things, but you'll have to either a) use some kind of software, or b) program it |
| 19:20.12 | Hypfer | so that would most likely be insanely overpowered |
| 19:20.17 | Voyage | though the caller id was stil "undisclosed number" despite of out.println("CallerID: 321 <321>"); but it worked |
| 19:20.21 | lvlinux | Hypfer: then you get to learn and do it yourself! |
| 19:20.31 | lvlinux | Hypfer: which will greatly benefit you! |
| 19:20.40 | Hypfer | as long as its somehow possible I think i could do it, yes |
| 19:20.47 | [TK]D-Fender | Voyage: taht does not look like a legit # to your provider. |
| 19:20.51 | Hypfer | I just need help on deciding what to buy exactly |
| 19:20.56 | craigify | Asterisk as a development platform is pretty cool, and powerful |
| 19:21.02 | lvlinux | yes it can be done and it's not that hard |
| 19:21.03 | [TK]D-Fender | Voyage: They'll probably look at it and go "nope!" |
| 19:21.13 | Voyage | [TK]D-Fender, hm I will make a long number then |
| 19:21.27 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-acfmkvhatskmpeut) |
| 19:21.35 | [TK]D-Fender | a LEGIT NUMBER |
| 19:21.49 | lvlinux | Voyage: which provider? |
| 19:21.49 | [TK]D-Fender | because they might also limit you to a number you arranged through them |
| 19:22.08 | craigify | I would start with soft phones Hypfer |
| 19:22.12 | craigify | you can get 'em for free |
| 19:22.47 | Voyage | lvlinux, flrowroute and [TK]D-Fender was right. more meaningfull full number worked instead of 321 |
| 19:22.57 | Hypfer | i need my fritzbox to install asterisk on first |
| 19:23.01 | [TK]D-Fender | craigify: He's already been using several |
| 19:23.08 | Hypfer | because i need to convert my analog phone to.. well.. whatever asterisk likes |
| 19:23.16 | Hypfer | analog phone line |
| 19:23.39 | lvlinux | Hypfer: get an Obihai OBI110 |
| 19:23.56 | *** join/#asterisk malcolmd (malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:23.56 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:24.00 | lvlinux | Asterisk likes lots of stuff---SIP, IAX2, MGCP, etc |
| 19:24.08 | Hypfer | lvlinux: pots -> asterisk -> analog phone |
| 19:24.15 | Hypfer | i need pots -> asterisk |
| 19:24.22 | lvlinux | Obihai Obi110 |
| 19:24.23 | Hypfer | location is germany :) |
| 19:24.38 | Hypfer | lvlinux: that thing looked like its a sip to analog phone bridge |
| 19:24.40 | Hypfer | not vice versa |
| 19:24.46 | lvlinux | it is both |
| 19:24.58 | lvlinux | you have FXS and FXO ports on it. |
| 19:25.09 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 19:25.13 | Hypfer | whatever that is |
| 19:25.21 | Hypfer | will it work in germany? |
| 19:25.25 | Voyage | [TK]D-Fender, one question out of curiousity. My skype number has a call recording facility if the call is not answered. So I am guessing the * would have called, completed all the rings and the call recording would have spoken. But I didnt go any voice recording. Maybe because by the time recording "beep" started, * had played helloworld and hungup) |
| 19:25.38 | Hypfer | Sale:$59.99 + $23.27 Shipping & Import Fees Deposit to Germany |
| 19:25.39 | Hypfer | jesus |
| 19:25.54 | lvlinux | FXO = phone line from oFFICE. FXS = phone line going to sTATION analog (phone). |
| 19:26.00 | Hypfer | ah! |
| 19:26.12 | Hypfer | any alternative to that obi110 thing which is also available in germany? |
| 19:26.21 | lvlinux | yes there are others. |
| 19:26.46 | lvlinux | really just about anything that has an FXO port and supports SIP, and isn't locked to a particular provider. |
| 19:26.52 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-xmabsypkxhdbmgqe) |
| 19:26.52 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 19:27.19 | lvlinux | Sipura makes some, and Multitech MultiVOIP MVP410 (nice box but pain to config), and others. |
| 19:28.29 | Hypfer | hm |
| 19:28.43 | Hypfer | but i'll need to run asterisk on some other machine then, right? |
| 19:28.51 | lvlinux | yes |
| 19:29.35 | lvlinux | your other choice is get an FXO interface from Rhino, Digium, or Sangoma that comes in a PCI card, or a USB interface. |
| 19:29.40 | lvlinux | Put that on your * box. |
| 19:29.44 | Hypfer | I don't understand why its so hard to find information on this topic on the internet |
| 19:30.05 | lvlinux | Hypfer: huh? it's not hard. But you have to know what you are looking for. |
| 19:30.29 | Hypfer | ah, that could be the reason why I don't find stuff :-) |
| 19:30.32 | lvlinux | Hypfer: the book will help you understand how it all works. |
| 19:30.36 | lvlinux | ~book |
| 19:30.37 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:30.54 | Hypfer | woah |
| 19:30.56 | Hypfer | books |
| 19:31.00 | craigify | I just searched for "How to hook up Asterisk to a phone line" |
| 19:31.07 | craigify | got some blogs, wiki and other articles |
| 19:31.16 | Hypfer | oh |
| 19:31.27 | Hypfer | apparently I'm unable to use google properly then |
| 19:31.41 | craigify | http://www.asterisk.org/products/telephony-interface-cards |
| 19:31.44 | craigify | that was in the list |
| 19:31.59 | lvlinux | all it takes is some thinking. Figure out what you are trying to ask and then punch it in to google and it will get you at least close. |
| 19:32.01 | craigify | probably of interest to you |
| 19:32.17 | [TK]D-Fender | Voyage[TK]D-Fender, one question out of curiousity. My skype number has a call recording facility if the call is not answered. So I am guessing the * would have called, completed all the rings and the call recording would have spoken. But I didnt go any voice recording. Maybe because by the time recording "beep" started, * had played helloworld and hungup) <- no |
| 19:32.33 | Hypfer | main problem i that I don't have any pcie slots left in my server |
| 19:32.38 | lvlinux | Hypfer: but read the book---it's free online so no excuses. Will cover all the basics that you want to know. |
| 19:32.42 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-xoffeazlvopxixmc) |
| 19:32.47 | Voyage | [TK]D-Fender, then? |
| 19:32.54 | lvlinux | Hypfer: Then you'll have to go with an external device. |
| 19:32.55 | [TK]D-Fender | if the call is not answered then you aren't getting to the dialplan. Remember we just went over the fact we said this about 8 times now? |
| 19:33.14 | [TK]D-Fender | answerd call > dialplan |
| 19:33.15 | Voyage | [TK]D-Fender, but call waiting/beep recording == call answered |
| 19:33.18 | craigify | Also |
| 19:33.23 | craigify | if you are using google.de or something |
| 19:33.30 | craigify | perhaps your search results are different |
| 19:33.31 | Hypfer | lvlinux: the definition of fxoand fxs was very useful |
| 19:33.51 | Voyage | [TK]D-Fender, if skype says "please record your message after the beep", it is an attended call. |
| 19:33.54 | [TK]D-Fender | Voyage: remember the call is to a phone number. Voicemail is still an "answer" |
| 19:33.54 | Voyage | no? |
| 19:34.04 | *** join/#asterisk kharwell (kharwell@nat/digium/x-ndzrgmvxoipfnrxb) |
| 19:34.19 | lvlinux | Hypfer: yw |
| 19:34.25 | [TK]D-Fender | my first answer was more from a different direction |
| 19:34.37 | [TK]D-Fender | So lets disregard that for now |
| 19:34.49 | Voyage | [TK]D-Fender, yes, voicemail or manual answer/pickup. isnt that a call and the hello-world should have been played in both cases? |
| 19:34.59 | [TK]D-Fender | Channel: calls a phone. Voicemail is an answer. But your Timeout is still a limit |
| 19:35.06 | craigify | Voyage: I can tell you, the fact that there is no difinitive way for the telephone netwok to send back a regular answer, or voicemail answer is highly annoying, but answers are just answers. It might not even be a human that answers. |
| 19:35.15 | [TK]D-Fender | And yes.. if could start playing a sound file to the answering machine |
| 19:35.15 | Voyage | [TK]D-Fender, I deleted timout |
| 19:35.21 | [TK]D-Fender | You shouldn't |
| 19:35.27 | [TK]D-Fender | All dialouts should ahve a limit |
| 19:35.37 | [TK]D-Fender | something sensible |
| 19:35.42 | Voyage | creativx, yes , I agree " answers are just answers" |
| 19:35.44 | craigify | 24 years |
| 19:36.29 | WIMPy | Well, often voicemail is achieved through diversion. In that case you'd know that part. |
| 19:36.30 | *** join/#asterisk areski (~areski@80.174.128.118.dyn.user.ono.com) |
| 19:37.08 | Voyage | [TK]D-Fender, ya, I would limit call to be = all ring time + 10 sec recording time. But thats a separate story. For now I did NOT made a limit. answering machine attended call. no recording recorded. |
| 19:37.27 | Hypfer | are the us and DE landlines the same? |
| 19:37.35 | Voyage | what i will do is I would record my calls and see when it spoke 'hellow world' or it didnt in the first place |
| 19:37.41 | Hypfer | because then I'd order one of those obi whatever from the us |
| 19:37.52 | craigify | WIMPy: how so? |
| 19:38.07 | craigify | some kind of event you mean? |
| 19:38.15 | Voyage | I wish ther would be a universal recording facility rather exten => s,n,Monitor(wav,,b) |
| 19:38.58 | lvlinux | Hypfer: not sure if they are completely the same as far as rings and stuff, but should work fine I believe. WIMPy probably knows the definitive answer to that. |
| 19:39.22 | WIMPy | craigify: A divesion notification. |
| 19:40.15 | WIMPy | They are not the same. |
| 19:40.20 | craigify | WIMPy: I'm lokoing into thisâ¦. |
| 19:40.31 | WIMPy | Hypfer: What kind of line are you dealing with? |
| 19:40.44 | *** join/#asterisk sgriepentrog (sgriepentr@nat/digium/x-uqdrxremjzmeoodh) |
| 19:40.45 | craigify | It looks like it comes in a SIP header, for one. This is news to me. This might solve a problem I have... |
| 19:40.52 | Hypfer | WIMPy: the one that comes out of my hitron DOCSIS 3 modem |
| 19:41.17 | WIMPy | Urgs |
| 19:41.40 | Hypfer | yep. kabeldeutschland :-)) |
| 19:42.03 | WIMPy | Can't they be hacked? |
| 19:42.14 | Hypfer | not that I'd know of |
| 19:43.04 | Hypfer | so what do you think? |
| 19:43.43 | WIMPy | You made a bad decision when signing that contract. |
| 19:43.48 | Hypfer | yes. |
| 19:44.56 | Hypfer | but i'll run that modem in bridge mode so internet wise everything is fine. just the telephony feature is crippled |
| 19:45.46 | WIMPy | I know. But using Asterisk will probably not make the packet loss any better. |
| 19:45.58 | Hypfer | there is no packet loss |
| 19:46.03 | WIMPy | Unless you don't suffer from that. |
| 19:46.09 | Hypfer | when did anyone mention packet loss? |
| 19:46.39 | WIMPy | I just hear it every tiome I talk to KDG users. |
| 19:46.46 | [TK]D-Fender | Voyage[TK]D-Fender, ya, I would limit call to be = all ring time + 10 sec recording time. But thats a separate story. For now I did NOT made a limit. answering machine attended call. no recording recorded. <- some providers bill you for RINGING TIME TOO. |
| 19:46.56 | [TK]D-Fender | Imagine you screw up and it rings for HOURS and you get billed |
| 19:47.26 | [TK]D-Fender | Or you are making your mass call-out system and lines get tied up because you were expecting them to always get answered one way or another ... andd they aren't |
| 19:47.31 | Voyage | [TK]D-Fender, oh come on... NO. if providers bill for ringing time, thats not goood. |
| 19:47.33 | [TK]D-Fender | BAD THINGS HAPPEN |
| 19:47.45 | Hypfer | 20:47 < [TK]D-Fender> BAD THINGS HAPPEN |
| 19:47.51 | Hypfer | this concludes the news for today |
| 19:47.55 | [TK]D-Fender | This is a Real World warning for you |
| 19:47.57 | Voyage | flowroute? |
| 19:47.57 | craigify | heh |
| 19:48.03 | [TK]D-Fender | 60 seconds TOPS |
| 19:48.09 | Voyage | sorry? |
| 19:48.13 | Voyage | 60 sec tops? |
| 19:48.22 | [TK]D-Fender | unless you have specific people you need to make exceptions for. |
| 19:48.29 | [TK]D-Fender | 60s before giving up a dial |
| 19:48.31 | WIMPy | Hypfer: I think the best you can do is use another provider and wait for the new law to pass. |
| 19:48.56 | Hypfer | WIMPy: still 1 year contract. but I will after that because the telekom provides 100/40 here :-) |
| 19:49.11 | Voyage | what do you mean by "60s before giving up dial"? you mean pstn will keep on dialing for 60 seconds if not picked up |
| 19:49.29 | WIMPy | Hypfer: 40? Is that a new thing? |
| 19:49.51 | Hypfer | WIMPy: yes. the new vdsl whatever it is |
| 19:50.04 | Hypfer | "magentaZuhause L" |
| 19:50.07 | [TK]D-Fender | You should force a timeout limit of 60s on YOUR side in case the other side DOESN'T give up on you automatically |
| 19:50.28 | Voyage | hm |
| 19:50.29 | Voyage | ok |
| 19:50.46 | Voyage | I will ask flowroute if they charge on dialtones |
| 19:50.48 | Voyage | is there something wrong with this syntax? exten => _1NXXXXXXXXX,1,Monitor(wav,myfilename) |
| 19:50.48 | Voyage | same => n,Playback(/usr/share/asterisk/sounds/en/hello-world) |
| 19:51.32 | [TK]D-Fender | Is there something wrong in the result of that? |
| 19:51.52 | [TK]D-Fender | Does it fail to do what you ask? |
| 19:53.01 | Voyage | [TK]D-Fender, nothing wrong, call is being made but no recording |
| 19:53.13 | Voyage | in /var/spool/asterisk/outgoing |
| 19:53.42 | [TK]D-Fender | Show us exxact code and the CLI output of your call |
| 19:53.53 | *** join/#asterisk jrrose (jon@nat/digium/x-gjkyjiltkqxhufkg) |
| 19:54.05 | Voyage | ok. one minut |
| 19:57.11 | Voyage | [TK]D-Fender, is it safe to show sip debug and manager debug logs? does it have username/ password in it? |
| 19:57.34 | [TK]D-Fender | we don't care about AMI at this stage |
| 19:57.48 | Voyage | sip debug? |
| 19:57.59 | [TK]D-Fender | Sure |
| 19:58.11 | [TK]D-Fender | just in case inbound audio is a problem |
| 19:58.55 | *** join/#asterisk kharwell (~kharwell@216.186.189.18) |
| 20:01.02 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 20:01.39 | Voyage | [TK]D-Fender, sent you private message |
| 20:02.18 | Voyage | providing that I promptly picked up the call and yelled at it. (also heard the hellowrold from other side. ) |
| 20:03.11 | [TK]D-Fender | [15:53]Voyage[TK]D-Fender, nothing wrong, call is being made but no recording |
| 20:03.13 | [TK]D-Fender | [15:53]Voyagein /var/spool/asterisk/outgoing |
| 20:03.29 | [TK]D-Fender | You should very quickly reconsider why you assumed it would go in THAT folder |
| 20:03.56 | Voyage | correct |
| 20:04.29 | Voyage | right. because its a default one? |
| 20:04.38 | [TK]D-Fender | not default for recordings... |
| 20:04.46 | [TK]D-Fender | that is for call-files |
| 20:05.06 | [TK]D-Fender | which is the text-file equivalent of AMI Originate |
| 20:05.13 | [TK]D-Fender | Not the folder we want |
| 20:05.25 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 20:05.25 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 20:05.36 | [TK]D-Fender | goes under astspooldir /monitor by default |
| 20:05.42 | [TK]D-Fender | "core show settings" |
| 20:05.52 | *** join/#asterisk rmudgett (~rmudgett@216.186.189.18) |
| 20:05.52 | Voyage | hm |
| 20:06.00 | [TK]D-Fender | if you don't provide an absolute path |
| 20:08.12 | Voyage | hm |
| 20:09.04 | *** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com) |
| 20:09.11 | Voyage | it has myfilename-out.wav and myfilename-in.wav. the out one saying 'hello world" but not my yelled voice which I did. the 'in' one says 'click sound' |
| 20:11.49 | [TK]D-Fender | You should try making a call that lasts longer. Also one where you do an explicit answer first, etc |
| 20:22.37 | Voyage | hm |
| 20:23.08 | Voyage | the last one was an explicit answer |
| 20:23.33 | Voyage | and I wonder why theres an name'in'.wave |
| 20:23.48 | Voyage | out is what is should have for outgoing calls |
| 20:28.29 | [TK]D-Fender | because that is a call |
| 20:28.35 | [TK]D-Fender | sounds goes in both directions |
| 20:29.05 | [TK]D-Fender | packs up to head home |
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| 20:34.39 | Voyage | ah, so I cant get one recording of both way conversation unless I merge the 2 in,out.wav files in asterisk? |
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| 20:37.43 | drmessano | Voyage, they added this thing called MixMonitor about 12 years ago.. |
| 20:37.46 | drmessano | https://wiki.asterisk.org/wiki/display/AST/Application_MixMonitor |
| 20:38.24 | Voyage | great. thanks |
| 20:41.30 | Voyage | drmessano, i wonder if theres some universal call monitory settings rather declaring it in every extensions |
| 20:42.29 | drmessano | The chapter in the Asterisk book on pattern matching will tell you how to do just that |
| 20:42.47 | Voyage | ok |
| 20:43.05 | WIMPy | If there is any pattern matching involved... |
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| 20:53.40 | drmessano | True, there doesnt HAVE to be |
| 20:53.50 | drmessano | Few different ways of doing it |
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| 20:58.39 | trewq | Folks - I've been using voip.ms and would like to find out if there is someone cheaper or more of a favorite here in the channel |
| 21:00.31 | Penguin | They're pretty cheap already. |
| 21:00.57 | ldc | trewq: didlogic.com |
| 21:05.02 | drmessano | Flowroute |
| 21:05.02 | lvlinux | trewq: I second didlogic.com |
| 21:06.16 | trewq | thank you - I am looking for providing phone service to businesses and was looking for sip terminating options |
| 21:06.30 | trewq | been happy with voip.ms |
| 21:07.50 | trewq | when a phone rings, I want my app to know what it says on the caller id |
| 21:08.11 | trewq | voip.ms has no ability to do an api callback with the number |
| 21:08.23 | trewq | are there any creative ways of doing this? |
| 21:09.31 | WIMPy | Try again. I'm not sure what you want to do there. |
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| 21:10.21 | trewq | when someone calls my phone, I would like my web application to know who is calling (caller id) |
| 21:11.41 | WIMPy | Well, the information is available in the dialplan. How to get it in to your app, only you can know. |
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| 21:29.46 | craigify | trewq: There are many ways to accomplish this in Asterisk with AGI scripts to push a notification somewhere, or potentially a program listening for events via AMI that can make a push request |
| 21:30.48 | craigify | trewq: If you are running a boxed product, open source or commercial, perhaps they already have such a solution. |
| 21:34.12 | craigify | WIMPy: I wonder how consistently the Diversion: header gets sent back⦠|
| 21:34.54 | WIMPy | Not |
| 21:35.16 | craigify | I figured it's not 100% |
| 21:35.23 | WIMPy | Nothing in SIP is in any way consistent or reliable. Everybody should know that. |
| 21:35.27 | craigify | I'm guesisng not even close |
| 21:35.33 | craigify | yes |
| 21:35.44 | WIMPy | And off course not everyone is using a diversion to VM. |
| 21:35.47 | craigify | I'm wondering if there is anything like that in SS7 at all |
| 21:35.54 | WIMPy | Yes |
| 21:36.08 | craigify | hmm |
| 21:36.39 | craigify | that's good news |
| 21:36.43 | craigify | I'm going to run some tests |
| 21:36.56 | craigify | I want to see how many I'm getting out of total call volume per day |
| 21:37.04 | craigify | and see if it will even be useful to me |
| 21:49.22 | *** join/#asterisk Dpunkt (~Dpunkt@p5DE5F73C.dip0.t-ipconnect.de) |
| 21:51.18 | drmessano | I HAZ ASTERISK IDEA AND NO SKILLZ CAN U HELP ME BUILD IT RITE NOW OR ESLE? |
| 21:52.27 | WIMPy | s/ASTERISK/BUSINESS/ |
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| 21:58.36 | craigify | that seems to be the majority of poeple asking questions in here |
| 21:59.30 | WIMPy | No, but they might ask the most questions. |
| 22:03.01 | drmessano | Start to finish setup |
| 22:03.28 | drmessano | Help me install, help me configure, help me dialplan, help me get caught inappropriately touching my underage assistant |
| 22:03.31 | drmessano | All the same |
| 22:05.03 | robmal | I can help with the last one. |
| 22:05.10 | robmal | If she's 16+ |
| 22:05.23 | robmal | And in polen. |
| 22:05.27 | robmal | And cute. |
| 22:06.00 | drmessano | 16+ is "old meat" in Uzbekiraqipalistadia |
| 22:07.14 | robmal | I'm a connoisseur. |
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| 22:54.13 | craigify | haaah |
| 22:54.20 | craigify | Uzbekiraqipalistadia |
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| 23:20.22 | craigify | silly irc client |
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| 23:57.00 | Micc | does sipp show rtp packet loss? |
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