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| 00:29.43 | juliocesarlhg | could someone tell me the correct configuration for mgcp |
| 00:29.49 | juliocesarlhg | gateway |
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| 01:00.28 | UnixDev_ | I have a peer with an outboundproxy tag, outbound is working perfectly, but when an inbound call comes from the outboundproxy it is rejected with a 407... how can I tell asterisk to also accept calls from the outboundproxy of a peer? |
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| 01:47.20 | postconf | anyone else having problems with outgoing callerid in 1.6? |
| 01:48.18 | postconf | i mean outgoing callerid(name) or callerid(all), callerid(num) seems to work fine |
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| 02:05.16 | juliocesarlhg | alguien me pudiera ayudar con un gateway |
| 02:05.18 | juliocesarlhg | sip |
| 02:05.46 | juliocesarlhg | mi gateway tiene puertos fxs, pero cuando llamo a un anexo este no suena |
| 02:05.57 | juliocesarlhg | la luz del telefono prende pero no suena nada |
| 02:06.05 | juliocesarlhg | si levanto el telefono puedo hablar sin problemas |
| 02:06.07 | juliocesarlhg | pero no suena |
| 02:10.49 | roxdragon | italian? |
| 02:11.45 | *** join/#asterisk codefreeze-lap (~Steve_Mur@174-22-48-242.chyn.qwest.net) |
| 02:15.42 | roxdragon | juliocesarlhg, |
| 02:16.10 | raden | its spanish |
| 02:16.17 | roxdragon | ok |
| 02:16.21 | roxdragon | i am italian |
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| 02:16.49 | raden | lol |
| 02:21.01 | roxdragon | radeb you? |
| 02:21.14 | roxdragon | raden, * |
| 02:21.46 | juliocesarlhg | hi |
| 02:21.48 | juliocesarlhg | sorry |
| 02:22.02 | juliocesarlhg | i will write in english |
| 02:22.16 | juliocesarlhg | i have a gateway with sip protocol |
| 02:22.47 | juliocesarlhg | when i call to a telephone, the telefono does not ring |
| 02:22.49 | raden | got that much |
| 02:23.01 | raden | call which way in or out |
| 02:23.05 | juliocesarlhg | in |
| 02:23.10 | juliocesarlhg | i can call out |
| 02:23.57 | juliocesarlhg | when i receive a call on the gateway none of the telephones ring |
| 02:24.35 | raden | are the ports routed ? |
| 02:24.38 | raden | are you behind a nat ? |
| 02:24.44 | juliocesarlhg | no |
| 02:24.45 | raden | there a lot of things |
| 02:24.57 | raden | well route port 5060 to your asterisk box for one |
| 02:25.06 | juliocesarlhg | my gateway has 4 ports |
| 02:25.15 | raden | huh |
| 02:25.22 | juliocesarlhg | i have connect 2 telephones, |
| 02:25.51 | juliocesarlhg | when i call from one phone to the other it doenst ring |
| 02:26.11 | juliocesarlhg | thats why i think is a parameter in the router |
| 02:34.47 | juliocesarlhg | ??????? |
| 02:55.56 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 02:55.56 | ChannelZ | you've mentioned nothing about your config or what the console might be telling you. Are they even registering? |
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| 03:06.15 | Maliuta | does anyone know of an easy way to debug voicemail in 1.6.2 ?? my conf that was working is now not working :) |
| 03:13.03 | *** part/#asterisk postconf (~postconf@msfree.com) |
| 03:13.48 | juliocesarlhg | could someone tell me how to register a mgcp gateway in asterisk 1.8? |
| 03:19.15 | fauxalliance | http://www.voip-info.org/wiki/view/Asterisk+config+mgcp.conf |
| 03:19.46 | juliocesarlhg | i have already try |
| 03:19.55 | juliocesarlhg | but i can't get it work |
| 03:20.01 | juliocesarlhg | thats why i ask |
| 03:32.27 | Freeaqingme | "Call from '' to extension 'AGI' rejected because extension not found in context 'defaul" << What would be the most logical reason for the origin of the call to be empty? |
| 03:32.58 | juliocesarlhg | who know about mgcp |
| 03:32.59 | juliocesarlhg | please |
| 03:33.02 | juliocesarlhg | i need help |
| 03:53.49 | *** join/#asterisk kaushal (~kaushal@115.118.253.55) |
| 03:53.51 | kaushal | hi |
| 03:54.03 | kaushal | Does asterisk support VoiceXML ? |
| 04:19.07 | kaushal | checking in again for the query ? |
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| 05:08.41 | kaushal | checking in again for the query ? |
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| 06:36.57 | zeo111 | hello all |
| 06:38.28 | shapr | Howdy zeo111 |
| 06:39.16 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 06:39.35 | zeo111 | :-) so, I'm a complete n00b as far as anything telephony is concerned. Tasked with updating our offices Asterisk 1.2 on fedora core 4 to something current. |
| 06:39.47 | shapr | I'd suggest Asterisk 1.8 |
| 06:39.56 | shapr | But you will need to make changes to the config. |
| 06:40.03 | zeo111 | That's what I was thinking, since its the new LTS |
| 06:40.14 | shapr | Unless you have expensive telephony hardware, I'd suggest setting up a whole 'nother box for the new system |
| 06:40.24 | shapr | That way you can rollback quickly in case of dire need :-) |
| 06:40.37 | zeo111 | I read the new "Asterisk Definitive guide", and thought I'd ask for suggestions what else to consider |
| 06:40.56 | zeo111 | I think we want to build of pretty raw asterisk, not like trixbox |
| 06:41.03 | shapr | What do you have at the moment? SIP only? PRI? |
| 06:41.05 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 06:41.15 | zeo111 | SIP only, but its all subject to change. |
| 06:41.22 | zeo111 | we're just a small consulting group of about 30 |
| 06:41.33 | zeo111 | need a few conference lines, VM, the regular. |
| 06:41.48 | zeo111 | maybe the ability to hook some functionality to our ticketing system down the road... |
| 06:42.23 | zeo111 | are there any killer tools I should be looking into for managing / configuring the basics? read a lot of good things about freePBX |
| 06:42.54 | shapr | I can't help you with that, but I will say that freepbx makes text-editor debugging difficult. |
| 06:43.08 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.144) |
| 06:43.22 | zeo111 | ok |
| 06:44.26 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 06:44.40 | zeo111 | I'm pretty comfortable working with scripting languages, so I think I'll be alrigh twith dial-plan, but the big boss isn't not technical, and I would like to give him the ability to do some things, hopefully in a GUI based environment |
| 06:45.44 | shapr | zeo111: FreePBX is the most popular front-end I've seen, and for the most part it's great for end users. |
| 06:45.56 | shapr | If something goes wrong, it's very difficult to debug all the added functionality. |
| 06:46.00 | shapr | Or maybe that's just me... |
| 06:46.29 | shapr | I'm sure my employer won't endorse my opinions, but I have been working in Digium's tech support department for some time. |
| 06:46.52 | zeo111 | lol, I can see how that would get out of hand. Is it controllable? Like say could I allow him to create a new extension for a contract employee, without too much fuss. But not let him mess with critical settings? |
| 06:47.46 | shapr | I don't know about limiting access via freePBX |
| 06:47.54 | zeo111 | ok, thought I'd ask. |
| 06:48.05 | zeo111 | it's nice to see there are some folks on here overnight |
| 06:48.30 | shapr | I'm at work, but I don't mind answering questions when I have a moment, and the knowledge to give an answer :-) |
| 06:48.45 | zeo111 | the last project I worked on migrating some of our apps to ec2 was a hella time finding folks to talk to |
| 06:49.11 | zeo111 | and I find IRC so much better than forums. |
| 06:49.12 | shapr | Oh, I've done some EC2 work, it's pretty nifty. |
| 06:49.14 | shapr | Me too |
| 06:49.18 | juliocesarlhg | how to configure mgcp gateway with asterisk? |
| 06:50.35 | zeo111 | yeah. I want to convice them to put our phone system in there too. |
| 06:50.57 | shapr | I've not heard much about people running Asterisk on EC2, does that work well? |
| 06:51.01 | shapr | thinks |
| 06:51.08 | zeo111 | allegedly... |
| 06:51.11 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 06:51.15 | shapr | I'd bet you could dynamically trunk everything together and it'd work smoothly. |
| 06:51.59 | zeo111 | that's my hope. our ISP isn't the best, but we don't use phones too much. I think i could recover some bandwidth if we didn't have to split our t1 in half ALL the time |
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| 06:54.29 | zeo111 | but like I said... I'm a n00b in the VOIP world. didn't even know how a voip phone worked until a few days ago. I was a lowly java programmer before I got this new job. |
| 06:55.18 | shapr | VoIP is way nifty, that's for sure. |
| 06:55.33 | shapr | Asterisk is more open sourcey than Java for the most part. |
| 06:56.31 | zeo111 | yeah, again, just found out it's more than drop and go software this week, but I'm looking forward to this project. Had way too much fun for about an hour playing with the built in sound bytes |
| 06:58.39 | shapr | zeo111: Yah, tt-weasels is great :-) |
| 06:59.18 | zeo111 | ROTFL hadn't heard that one. |
| 07:01.01 | zeo111 | I liked adding because-paranoid to the end of all the automated alerts we send out every night |
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| 07:39.52 | sawgood | 2165/2165 99.19.4.231 D N 5060 OK (63 ms) |
| 07:39.52 | sawgood | 2166/2166 67.18.17.188 D N 61001 OK (28 ms) |
| 07:39.52 | sawgood | 2167/2167 69.109.249.27 D N 5060 OK (14 ms) |
| 07:40.17 | sawgood | In the above example, why do two hosts come in on PORT 5060 and the one host comes in on port 61101 |
| 07:40.42 | wdoekes2 | depends entirely on their settings |
| 07:40.57 | sawgood | The port is not the * box, but rather the host box? |
| 07:40.58 | wdoekes2 | and probably nat-routers that are in between |
| 07:41.07 | wdoekes2 | yes |
| 07:41.12 | sawgood | perfect ... |
| 07:41.13 | sawgood | ty! |
| 07:41.25 | wdoekes2 | asterisk listens on exactly one port |
| 07:41.33 | sawgood | 100% understood |
| 07:41.36 | wdoekes2 | (if you're talking about sip) |
| 07:45.43 | sawgood | Is there a way using * (with a true SBC registration server) to have (say 5 SIP phones) all have the same extension (example 501) |
| 07:45.56 | sawgood | I meant without a registration server |
| 07:46.32 | wdoekes2 | well.. you can create a "callgroup".. in your dialplan, you do exten => 501,1,Dial(SIP/201&SIP/202&SIP/203) |
| 07:46.43 | wdoekes2 | where 201, 202 and 203 are sip phones |
| 07:46.46 | sawgood | well, yes, and that is what I've done ... |
| 07:46.55 | sawgood | but for outbound calling they all still have their own extension number |
| 07:47.03 | sawgood | unless I 'mask' caller ID at the extension level |
| 07:47.21 | wdoekes2 | (1) is that a problem? (2) you can set callerid= in the sip context |
| 07:48.14 | wdoekes2 | so yes.. you cannot register from multiple locations at once, and no, that is not an unsolvable problem in most (if not all) cases |
| 07:48.43 | sawgood | Its not a 'problem' ... more so its a 'why' ... I have quite a few situations where 3 or 4 extensions (SIP phones) all do the same job (for example they sit in a compay warehouse) and if they were all the same extension, it would make programming much smoother |
| 07:49.17 | wdoekes2 | there is very much you can automate using asterisk |
| 07:49.31 | sawgood | wdoekes2: are you in the USA? |
| 07:49.46 | wdoekes2 | look at realtime dynamic config and func_odbc if you like programming over typing |
| 07:49.51 | wdoekes2 | NL |
| 07:50.00 | sawgood | Netherlands? |
| 07:50.06 | wdoekes2 | correct |
| 07:50.14 | sawgood | what GMT is it there? |
| 07:50.18 | wdoekes2 | +1 |
| 07:50.19 | sawgood | I am GMT -8 |
| 07:50.34 | sawgood | 10 AM? |
| 07:50.37 | wdoekes2 | 8:50 |
| 07:50.45 | sawgood | cool |
| 07:51.08 | sawgood | I have a cool PoE clock on the wall (6 digit timing) |
| 07:51.43 | sawgood | It is a SIP extension with a built in speakerphone |
| 07:51.48 | shapr | That is cool. |
| 07:52.14 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
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| 07:56.50 | Dovid | morning EV1 |
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| 08:27.39 | juliocesarlhg | someone knows about mgcp configuration file? |
| 08:30.59 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 08:34.19 | kaldemar | juliocesarlhg: did you take a look at the sample one? |
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| 08:39.56 | juliocesarlhg | yes |
| 08:40.29 | juliocesarlhg | who can connect to my pc remotly to check? |
| 08:40.32 | juliocesarlhg | pleaseeeeeeeee |
| 08:41.02 | juliocesarlhg | i havent sleep in 3 days |
| 08:42.15 | *** join/#asterisk Tim_Toady (~moi@77.49.233.214) |
| 08:42.38 | kaldemar | juliocesarlhg: why don't you connect yourself? |
| 08:43.12 | juliocesarlhg | i have try everything i know |
| 08:43.19 | juliocesarlhg | but i cant get i work |
| 08:46.16 | juliocesarlhg | someone can help me with mgcp.conf |
| 08:46.16 | juliocesarlhg | ? |
| 08:46.34 | tzafrir | juliocesarlhg, could you actually provide some information? |
| 08:47.14 | juliocesarlhg | what do u wanna know? |
| 08:47.25 | juliocesarlhg | my mgcp.conf? |
| 08:48.17 | tzafrir | Does 'mgcp show endpoints' give any output? |
| 08:48.26 | tzafrir | This is a command in the Asterisk CLI |
| 08:48.33 | juliocesarlhg | yes |
| 08:48.39 | juliocesarlhg | i know that |
| 08:49.00 | tzafrir | So does it give output? |
| 08:49.09 | juliocesarlhg | yes |
| 08:49.14 | tzafrir | If so, please post it in a pastebin |
| 08:49.16 | tzafrir | ~pb |
| 08:49.16 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 08:53.01 | juliocesarlhg | i paste it |
| 08:54.36 | juliocesarlhg | http://asterisk.pastey.net/147925 |
| 08:55.24 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 08:56.32 | juliocesarlhg | ? |
| 08:58.27 | juliocesarlhg | any help? |
| 09:00.26 | juliocesarlhg | ? |
| 09:00.43 | juliocesarlhg | who knows about mgcp.conf? |
| 09:00.57 | tzafrir | juliocesarlhg, when you try to call from the endpoint, what happens? |
| 09:01.04 | tzafrir | core set verbose 3 |
| 09:01.09 | tzafrir | do you see anything? |
| 09:01.21 | juliocesarlhg | core set verbose 10 |
| 09:01.37 | juliocesarlhg | nothing happens |
| 09:01.45 | juliocesarlhg | i can call from endpoint |
| 09:02.21 | juliocesarlhg | i can't call from endpoint |
| 09:03.32 | tzafrir | what do you mean by "can't call"? What does happen? |
| 09:04.03 | tzafrir | can you try 'mgcp audit endpoint <whatever>"? |
| 09:04.25 | juliocesarlhg | have u ever use mgcp.conf? |
| 09:05.34 | juliocesarlhg | http://asterisk.pastey.net/147926 |
| 09:05.48 | juliocesarlhg | sip to mgcp result there |
| 09:06.37 | tzafrir | "timed out". I'm not really familiar with mgcp. But you're not really talking to it |
| 09:06.46 | tzafrir | Is the device on the same network with you? |
| 09:07.05 | juliocesarlhg | yes |
| 09:08.03 | juliocesarlhg | this gateway has gateway name |
| 09:08.10 | juliocesarlhg | 138.0.60.1 |
| 09:08.22 | juliocesarlhg | gateway ip address 138.0.60.1 |
| 09:08.30 | juliocesarlhg | gateway port 2427 |
| 09:08.31 | tzafrir | The IP address is 138.0.60.1 ? |
| 09:08.40 | tzafrir | Is that a private range? |
| 09:08.45 | juliocesarlhg | no |
| 09:08.54 | juliocesarlhg | thats what i dont understand |
| 09:09.07 | tzafrir | Can you ping it? |
| 09:09.09 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 09:09.20 | juliocesarlhg | i ping it on 192.168.1.80 |
| 09:09.29 | juliocesarlhg | thats my lan range |
| 09:09.34 | tzafrir | So why would the IP address be that? |
| 09:09.45 | tzafrir | MGCP is over IP. |
| 09:10.03 | juliocesarlhg | yes |
| 09:10.09 | tzafrir | If you can't send IP packets to it (and recieve them back), how can you talk MGCP with it? |
| 09:10.38 | juliocesarlhg | i access web configuration trough 192.168.1.80 |
| 09:10.47 | tzafrir | How can you tell that the IP address is 138.0.60.1 ? |
| 09:11.06 | juliocesarlhg | can u see it? |
| 09:11.19 | juliocesarlhg | i has voice ip |
| 09:11.31 | juliocesarlhg | 138.1.60.1 |
| 09:11.42 | juliocesarlhg | data ip 138.0.60.1 |
| 09:11.52 | juliocesarlhg | and lan ip 192.168.1.80 |
| 09:13.47 | tzafrir | juliocesarlhg, workaround: add an extra IP to your system: |
| 09:14.14 | tzafrir | ip addr add 138.0.60.5/8 dev eth0 |
| 09:14.29 | tzafrir | And configure everything related to mgcp with it |
| 09:14.39 | juliocesarlhg | i am running debian |
| 09:14.53 | juliocesarlhg | i have two network cards |
| 09:15.13 | tzafrir | That command adds it manually |
| 09:15.45 | tzafrir | You can add it permanently with a /etc/network/interfaces static ip address for the "device" eth0:0 |
| 09:15.55 | tzafrir | (make sure it is not the default route |
| 09:16.08 | juliocesarlhg | but if i change it i lost internet |
| 09:16.20 | tzafrir | iface eth0:0 inet static |
| 09:16.29 | tzafrir | <PROTECTED> |
| 09:16.36 | tzafrir | <PROTECTED> |
| 09:16.48 | tzafrir | (I assume this is the netmask to use) |
| 09:17.24 | juliocesarlhg | wouldn't it be better if i change ip of the gateway? |
| 09:17.58 | tzafrir | sure. If you can get rid of that 138.<whatever> - better |
| 09:18.15 | juliocesarlhg | can u explain me something |
| 09:18.53 | juliocesarlhg | my gateway has 4 lan ports and 1 wan port |
| 09:19.01 | juliocesarlhg | an 2 fsx ports |
| 09:19.51 | juliocesarlhg | on the network configuration in the gateway it says WAP IP = 138.0.60.1 |
| 09:19.52 | *** join/#asterisk [netman] (~netman@75.Red-83-41-1.dynamicIP.rima-tde.net) |
| 09:19.56 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 09:20.18 | juliocesarlhg | nat server = data and voice transmit with data net port |
| 09:20.39 | juliocesarlhg | then lan ip 192.168.1.80 |
| 09:21.01 | juliocesarlhg | now my question is where do i connect the cable? |
| 09:21.05 | juliocesarlhg | on wan or lan? |
| 09:21.45 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 09:24.00 | juliocesarlhg | ? |
| 09:26.22 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 09:28.39 | tzafrir | do you connect to it from the LAN or from the WAN? |
| 09:28.49 | juliocesarlhg | lan |
| 09:29.09 | juliocesarlhg | i have connected 1 ip telephone to lan 1 |
| 09:29.16 | juliocesarlhg | i works ok |
| 09:29.26 | tzafrir | So just use that local address. But be sure to disable any DHCP server it has (if you already have one) |
| 09:30.03 | juliocesarlhg | is disable |
| 09:30.58 | *** join/#asterisk imox1234 (~imox1234@p4FC5C742.dip0.t-ipconnect.de) |
| 09:34.31 | benngard | weird, execif "0" and still execute: -- Executing [0317998975@inputinterior.se:4] ExecIf("OOH323/Avaya2-115", ""0"?Goto("",1)") in new stack |
| 09:34.31 | benngard | -- Goto (inputinterior.se,"",1) |
| 09:36.10 | shapr | Check the number of quotes? |
| 09:36.28 | shapr | ser ut lite konstigt, kanske förmånga? |
| 09:36.32 | benngard | guess it must be an error there, but i cat se where |
| 09:36.57 | benngard | another swed in the channel :) |
| 09:36.59 | kaldemar | "0 <--- |
| 09:37.18 | kaldemar | show the command and we'll take a look. |
| 09:37.21 | benngard | sec |
| 09:37.34 | shapr | benngard: Jag vad född i och bor Alabama och arbetar i tech support med Digium :-) |
| 09:37.54 | *** join/#asterisk drcode (~user1@bzq-84-111-89-77.red.bezeqint.net) |
| 09:37.56 | drcode | hi all |
| 09:37.58 | drcode | I need help |
| 09:38.00 | shapr | And my spelling in Swedish is crap somedays :-/ |
| 09:38.22 | drcode | I connect asterisk with mysql by odb |
| 09:38.23 | joobie | drcode, that's nice |
| 09:38.26 | drcode | odbc |
| 09:38.33 | drcode | I can loged it |
| 09:38.38 | joobie | man I miss TK |
| 09:38.42 | joobie | and his sarcasm |
| 09:38.51 | joobie | he would have smacked drcode down :) |
| 09:38.57 | shapr | What happened to him? |
| 09:39.02 | drcode | I want to build simple dial plan to call from each user |
| 09:39.06 | joobie | he had a falling out wiht russelb and got banned |
| 09:39.22 | shapr | In my opinion he was way too caustic. |
| 09:39.31 | drcode | I use switch => Realtime,@extentions |
| 09:39.57 | benngard | exten => 0317998975,n,ExecIf($["${DB_EXISTS(CF/0317998975/Extension)}"]?Goto("${DB(CF/0317998975/Extension)}",1)) |
| 09:40.09 | juliocesarlhg | i made the change on my gateway |
| 09:40.19 | shapr | benngard: Anyway, I lived in Boden and Stockholm for five years, and now am back in Alabama. |
| 09:40.42 | benngard | shapr: Boden, thats far north i Sweden |
| 09:40.45 | benngard | in* |
| 09:40.48 | drcode | benngard, can U simple expline |
| 09:41.01 | drcode | I want to build simple dial plan |
| 09:41.04 | *** join/#asterisk oelewapperke (wapper@85-158-215-1.powered-by.benesol.be) |
| 09:41.10 | drcode | in extension db |
| 09:41.19 | oelewapperke | anyone know how you're supposed to do actual call control in AMI ? Anyone have an example ? |
| 09:41.20 | shapr | drcode: Your question is not clear. |
| 09:41.20 | benngard | extension 1,1,Dial(SIP/123) |
| 09:41.26 | benngard | :) |
| 09:41.34 | benngard | exten => 1,1* |
| 09:41.39 | shapr | benngard: Nothing obviously wrong with that ExecIf |
| 09:41.52 | benngard | shapr: no, not that i can se either |
| 09:42.12 | drcode | and how can I get dialplan from table? |
| 09:42.58 | shapr | benngard: Does the DB value contain quotes also? |
| 09:43.51 | benngard | database show CF/037998975 |
| 09:43.51 | benngard | 0 results found |
| 09:43.59 | benngard | not even any data |
| 09:44.38 | kaldemar | benngard: set the expression to "${DB_EXISTS(CF/0317998975/Extension)}" = "1" instead. |
| 09:45.11 | benngard | but then i will jump to inputinterior.se,1,1 and i dont wanna do that |
| 09:45.27 | oelewapperke | can you do Asterisk Management Interface call control without using a dialplan ? Just pure event-based call control |
| 09:45.34 | kaldemar | benngard: the gotoif app does not know that you expect a plain 1 to be true and 0 false. |
| 09:45.57 | benngard | execif u mean |
| 09:46.14 | kaldemar | benngard: so ExecIf($["${DB_EXISTS(CF/0317998975/Extension)}" = "1"]?Goto("${DB(CF/0317998975/Extension)}",1)) |
| 09:46.21 | *** part/#asterisk cashback (~mac@ip68-2-140-46.ph.ph.cox.net) |
| 09:46.43 | juliocesarlhg | hi |
| 09:47.05 | benngard | oki, got u |
| 09:47.22 | benngard | will test, just need to fix a cell phone first |
| 09:48.09 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-uvgwnmmiztlhyobw) |
| 09:52.48 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 09:54.38 | *** join/#asterisk cashback (~cashback@ip68-2-140-46.ph.ph.cox.net) |
| 09:55.56 | benngard | kaldemar: that worked, thx |
| 09:56.38 | benngard | starnge that i have like hundeds of other line like that, i did assume that 1 was true and 0 was false |
| 09:57.37 | shapr | Hrm, what's the lowest power Asterisk gateway system? |
| 09:58.49 | shapr | As in, what Asterisk hardware would you guys suggest be installed in the parts of Japan that currently don't have power or phone? |
| 10:00.23 | *** join/#asterisk juliocesarlhg (~jcesarg@190.234.250.59) |
| 10:00.25 | kaldemar | benngard: the quotes make it a string which gets interpreted as true. without quotes, 1 would be true and 0 would be false. |
| 10:00.32 | juliocesarlhg | help with mgcp |
| 10:02.09 | benngard | kaldemar: thx for updating me |
| 10:02.33 | *** join/#asterisk cashback (~cashback@ip68-2-140-46.ph.ph.cox.net) |
| 10:02.34 | shapr | Aha, so it was too many quotes! |
| 10:05.05 | juliocesarlhg | help with mgcp.conf |
| 10:05.06 | juliocesarlhg | please |
| 10:06.15 | tzafrir | juliocesarlhg, ask more specific questions |
| 10:08.43 | juliocesarlhg | http://asterisk.pastey.net/147927 |
| 10:09.07 | juliocesarlhg | my wan ip now is 192.168.1.90 |
| 10:09.27 | juliocesarlhg | wan netmask 255.255.0.0 |
| 10:09.34 | juliocesarlhg | lan ip 192.168.1.80 |
| 10:10.00 | juliocesarlhg | lan net mask 255.255.255.0 |
| 10:10.24 | juliocesarlhg | gateway ip 192.168.1.90 |
| 10:10.56 | dandre | hello, |
| 10:11.23 | *** join/#asterisk drcode (~tp131@bzq-84-111-89-77.red.bezeqint.net) |
| 10:11.57 | dandre | when using Set(__FOO="bar"), can I modify ${FOO} in inherited channels? |
| 10:12.00 | drcode | hi all |
| 10:12.01 | drcode | back |
| 10:12.03 | drcode | I got this: |
| 10:12.18 | drcode | [Mar 21 11:07:39] WARNING[3350]: app_dial.c:2041 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown) |
| 10:12.31 | kaldemar | dandre: define modify. |
| 10:13.05 | drcode | and I use this: |
| 10:13.06 | drcode | exten => 105,1,Dial(IAX2/105) |
| 10:13.06 | drcode | exten => 106,1,Dial(IAX2/106) |
| 10:16.43 | kaldemar | drcode: does iax2 show peers list an ip address for the peer you were trying to dial? |
| 10:19.02 | juliocesarlhg | i got this when calling from sip to mgcp |
| 10:19.52 | juliocesarlhg | http://asterisk.pastey.net/147929 |
| 10:20.11 | drcode | I am checking |
| 10:21.38 | drcode | no |
| 10:21.42 | dandre | kaldemar: change the value of this variable |
| 10:21.52 | drcode | it dosnt show |
| 10:21.59 | drcode | but I can login |
| 10:22.12 | drcode | I think I am doing somthing wrong |
| 10:22.32 | drcode | i am using frinend |
| 10:22.37 | drcode | not peers |
| 10:23.00 | kaldemar | dandre: where do you change it and where do you expect to see the results? show your dialplan. |
| 10:26.04 | juliocesarlhg | ? |
| 10:26.57 | dandre | I expect to have the changes in the first defined variable |
| 10:27.22 | kaldemar | drcode: then make 105 and 106 register to your asterisk box of define an address for them. |
| 10:27.48 | kaldemar | drcode: a friend is both user and a peer. |
| 10:28.15 | kaldemar | dandre: are you going to show something? |
| 10:28.49 | drcode | I see |
| 10:28.50 | drcode | <PROTECTED> |
| 10:28.59 | drcode | I am checking if it work without odbc |
| 10:33.25 | wdoekes2 | "0" != 0 |
| 10:33.31 | wdoekes2 | oops.. scrollback |
| 10:33.48 | drcode | ok |
| 10:33.56 | drcode | if I am using iax.conf |
| 10:33.58 | drcode | it wirk |
| 10:33.59 | drcode | work |
| 10:34.20 | drcode | but If I try to use users from odbc , it dosn't show in iax2 show peers |
| 10:35.15 | dandre | I am trying to simplify my dialplan to show |
| 10:36.24 | skrusty | morning all |
| 10:42.17 | drcode | what it mean: |
| 10:42.18 | drcode | NOTICE[3908]: chan_iax2.c:4643 __auto_congest: Auto-congesting call due to slow response |
| 10:44.48 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 10:46.47 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 10:50.00 | tzafrir | juliocesarlhg, "No command found on [192.168.1.80] for transaction 65. Ignoring..." |
| 10:50.22 | tzafrir | I'm not really familiar with mgcp. Can you try calling from the gateway to asterisk? |
| 10:51.43 | dandre | kaldemar: some explanation here: http://pastebin.fr/10807 |
| 10:54.20 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-lgunticdgtskddxl) |
| 11:04.17 | kaldemar | dandre: if you set a variable like __FOO=value, a channel created by the master channel gets a variable by the value. but, the created channel has its own copies of the variables. if you change the value in the created channel, i won't affect the master channel, only the created channel and possible channels that it creates. |
| 11:04.46 | kaldemar | dandre: you can use func MASTER_CHANNEL to access variables of the channel that created yours. |
| 11:04.52 | drcode | why do I got this:Call rejected, CallToken Support required. If unexpected, resolve by placing address 10.20.107.54 in the calltokenoptional list or setting user 105 requirecalltoken=no |
| 11:06.19 | kaldemar | drcode: http://downloads.asterisk.org/pub/security/AST-2009-006.html |
| 11:06.43 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 11:07.47 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 11:10.15 | dandre | ok thanks |
| 11:11.40 | dandre | I can't, I am still using 1.6.2 |
| 11:18.08 | *** part/#asterisk drcode (~tp131@bzq-84-111-89-77.red.bezeqint.net) |
| 11:25.19 | kaldemar | dandre: then you could use global variables for instance. |
| 11:26.04 | kaldemar | or shared variables. |
| 11:26.29 | kaldemar | functions GLOBAL and SHARED |
| 11:38.30 | *** join/#asterisk neurosys (~neurosys@216.199.210.217.nw.nuvox.net) |
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| 12:02.18 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 12:02.27 | *** join/#asterisk TeknoJuce (~TeknoJuce@xbmc/staff/TeknoJuce) |
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| 12:04.49 | TeknoJuce | Hello trying to use a Nortel i2004 with asterisk 1.8.3.2 and google/voice so far I got the phone working screen shows asterisk/setitings and when I pick up hear the dialtone |
| 12:05.43 | TeknoJuce | entered all the settings for google voice with the help of some mario documentation |
| 12:06.02 | TeknoJuce | but the two dont seem to link together am I missing something? |
| 12:07.23 | benngard | whats the easiest way to test if first digit of a number is a zero? |
| 12:08.04 | TeknoJuce | used this http://www.arctangent.net/~superm1/gv_configs/ |
| 12:08.04 | kaldemar | benngard: $["${EXTEN:0:1}" = "0"] |
| 12:08.10 | *** join/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 12:09.12 | kaldemar | TeknoJuce: so you can dial in the asterisk box with the phone, but not out via google voice? |
| 12:10.22 | TeknoJuce | yup seems that way says hit dial tried to dail my home number then pushed call and it just hangs up |
| 12:11.30 | kaldemar | TeknoJuce: sounds like a dialplan problem. you need something like talk-numeric-outbound in your dialplan. |
| 12:13.32 | saxa | hi ppl, can somebody enlighten me what means this message i see in CLI ? |
| 12:13.34 | saxa | [Mar 21 12:35:47] WARNING[5285]: chan_iax2.c:1123 iax_error_output: Information element length exceeds message size |
| 12:13.35 | TeknoJuce | minus most of this documentation for trixbox as I am not using that I just used it to check out how they set the config file |
| 12:13.37 | TeknoJuce | http://fonality.com/trixbox/forums/trixbox-forums/trixbox-endpoints/your-one-stop-guide-nortel-i2002-i2005-install |
| 12:13.37 | saxa | [Mar 21 12:35:47] WARNING[5285]: chan_iax2.c:9904 socket_process: Undecodable frame received from '189.105.92.184' |
| 12:13.37 | benngard | kaldemar: i my case: exten => 040206612,n,ExecIf($["${CALLERID(num):0:1}"="0"]?Set(CALLERID(num)=${CALLERID(num):1})) worked as a clock :) thx |
| 12:13.45 | TeknoJuce | for the phone |
| 12:14.00 | TeknoJuce | after doing all that it got the phone working |
| 12:15.18 | TeknoJuce | I will pastebin |
| 12:22.03 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-lkbswwzubekfpale) |
| 12:23.42 | *** join/#asterisk [netman] (~netman@75.Red-83-41-1.dynamicIP.rima-tde.net) |
| 12:26.39 | *** join/#asterisk wdoekes2 (~walter@wjd.osso.nl) |
| 12:27.10 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 12:28.24 | felimwhiteley | hi folks |
| 12:28.49 | *** join/#asterisk drcode (~user1@bzq-84-111-89-77.red.bezeqint.net) |
| 12:28.50 | drcode | hi all |
| 12:29.04 | felimwhiteley | anyone know why Dial(SIP/2001,15) where you set it greater than 15 casues it to go invalid? |
| 12:29.07 | drcode | I did install asteriks in ubuntu 10.04 |
| 12:29.10 | drcode | I want to use odbc |
| 12:29.33 | drcode | it say no odbc engine |
| 12:29.45 | drcode | do I need to rebuild asteriks from source? |
| 12:30.16 | TeknoJuce | kaldemar http://pastebin.com/YeEJDXHL |
| 12:30.39 | TeknoJuce | or anyone else willing to point out my n00bness |
| 12:31.12 | drcode | any idea? |
| 12:32.20 | TeknoJuce | drcode, when you run the setup interface there is an option for odbc i would start there |
| 12:32.25 | TeknoJuce | enable it |
| 12:32.58 | TeknoJuce | so I would say yes to the recompile |
| 12:32.59 | drcode | how? |
| 12:33.06 | drcode | I see |
| 12:33.07 | drcode | ok |
| 12:33.27 | drcode | in other pc I have recompile it and it worked |
| 12:33.41 | drcode | I thote ubuntu add it by default |
| 12:33.48 | TeknoJuce | this is the guide i followed http://forums.plugpbx.org/index.php/topic,247.0.html |
| 12:34.06 | TeknoJuce | using the latest build number where it says to... |
| 12:34.13 | *** join/#asterisk garymc (~chatzilla@host81-148-15-59.in-addr.btopenworld.com) |
| 12:34.17 | drcode | thnx TeknoJuce |
| 12:34.19 | TeknoJuce | i didnt use odbc but I saw the option in there |
| 12:34.21 | TeknoJuce | for it |
| 12:34.25 | wdoekes2 | felimwhiteley: it shouldn't "go invalid".. you'll need a sip trace to know what's going on |
| 12:34.49 | wdoekes2 | fetch one for when it "works" and one for when it doesn't |
| 12:35.02 | felimwhiteley | wdoekes2: ok |
| 12:35.06 | felimwhiteley | I'll give that a go |
| 12:35.27 | felimwhiteley | 15 or below it's fine.. after that it.. weird |
| 12:36.01 | kaldemar | TeknoJuce: what do you see in CLI when you make a call? |
| 12:36.49 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 12:38.06 | TeknoJuce | kaldemar is there a certain command i need to run with astrisk -vvvvr after I see the CLI? |
| 12:39.31 | TeknoJuce | n/m one moment |
| 12:41.51 | TeknoJuce | kaldemar, http://pastebin.com/28wChLYP |
| 12:43.28 | kaldemar | TeknoJuce: what is SIP/1337? |
| 12:43.36 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 12:44.26 | kaldemar | TeknoJuce: whatever it is, you need to make is register to asterisk in order to dial it, since you have host=dynamic in sip.conf. |
| 12:45.03 | TeknoJuce | isnt 1337 the extension |
| 12:45.16 | *** join/#asterisk MatthewPowell (~powell@c-71-228-186-47.hsd1.al.comcast.net) |
| 12:45.21 | *** join/#asterisk [netman] (~netman@75.Red-83-41-1.dynamicIP.rima-tde.net) |
| 12:46.48 | kaldemar | TeknoJuce: -- Executing [s@default:2] Dial("1337@black-0", "SIP/1337,10") <--- you're dialing SIP/1337 |
| 12:48.43 | TeknoJuce | hmmm why is it dialing it sell it shows the number 15555555500 that its trying to dail first |
| 12:48.54 | TeknoJuce | sell = self |
| 12:49.21 | kaldemar | see lines 5 and 6. |
| 12:50.20 | kaldemar | TeknoJuce: you have context=from-internal in the unistim conf file, but there is no such context in extensions.conf. |
| 12:50.22 | TeknoJuce | says it failed to dail the number so its falling back |
| 12:52.41 | TeknoJuce | a kid on youtube said to change that so I will put that back to default he said something will fail if you dont put that might had something to do with trix as thats what he was using but i am not |
| 12:53.18 | TeknoJuce | think he was talking about voice mail but gv I think has its own vm so dont think that matters |
| 12:54.59 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 12:56.09 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 12:57.34 | *** join/#asterisk mzahariev (adminimini@piem-nafta-v.unixsol.org) |
| 12:58.48 | kaldemar | TeknoJuce: in fact, you shouldn't have a default context at all. it can be a bit of a security hole. anyway, you need to change the context setting in unistim config to something that exists is your extensions.conf. |
| 12:59.24 | TeknoJuce | so just coment it out |
| 12:59.38 | kaldemar | no no no. |
| 12:59.57 | kaldemar | ah, the default context. yes, comment it out. |
| 13:01.08 | TeknoJuce | how do you restart the server in cli doesnt seem to work in this latest version |
| 13:01.15 | TeknoJuce | restart now |
| 13:02.33 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:03.23 | TeknoJuce | says no availible command |
| 13:03.47 | kaldemar | core restart now |
| 13:05.02 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 13:06.00 | *** join/#asterisk blackcat73 (~Blackcat@adonis.iportalmais.pt) |
| 13:06.38 | TeknoJuce | same thing after commenting that line out |
| 13:06.49 | TeknoJuce | (same error output) |
| 13:07.17 | *** join/#asterisk dimm (~admin@unaffiliated/dimm) |
| 13:07.19 | kaldemar | did you change the context? |
| 13:07.51 | ssureshot | does an operator panel come with asterisk source or do I need to compile something else? |
| 13:07.53 | TeknoJuce | we comented it out |
| 13:08.13 | dimm | hello! not hear sound in softphone when dialing to some number, but mixmonitor recording sound. Is it common problem? |
| 13:08.22 | kaldemar | ssureshot: there is no operator panel with asterisk. |
| 13:08.27 | TeknoJuce | ssureshot do I know you |
| 13:08.34 | TeknoJuce | xbmc? |
| 13:08.57 | ssureshot | thanks kaldemar: |
| 13:09.12 | TeknoJuce | guess thats a no |
| 13:09.14 | ssureshot | TeknoJuce: I don't believe we've met |
| 13:09.31 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 13:09.38 | TeknoJuce | used to be a guy in my channel xbmc that had the same name as you |
| 13:09.44 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 13:09.49 | TeknoJuce | oh well |
| 13:09.51 | ssureshot | right on |
| 13:09.57 | TeknoJuce | carry on |
| 13:10.13 | retentiveboy | Is there a way to add entries to the Directory app without setting up voicemail boxes? |
| 13:10.38 | TeknoJuce | ssureshot if you want a silly gui you could use trix |
| 13:11.59 | TeknoJuce | kaldemar any other suggestions? |
| 13:12.29 | ssureshot | TeknoJuce: I just need a switchboard for our receptionist.. We currently use op-panel, so I guess Ill just recompile that and hope it's compat with 1.8 |
| 13:12.51 | *** join/#asterisk JonathanRose (~jonathan@nat/digium/x-qfcjejvxxssueiij) |
| 13:12.56 | kaldemar | TeknoJuce: did you change the context in unistim configuration? |
| 13:13.08 | TeknoJuce | I commented it out |
| 13:13.36 | TeknoJuce | ;;context=default ; context, default="default" |
| 13:15.52 | kaldemar | TeknoJuce: i told you to set it to something that exists in your extensions.conf, for example outbound. |
| 13:17.23 | TeknoJuce | context=outbound |
| 13:17.26 | TeknoJuce | restarting |
| 13:19.10 | TeknoJuce | i think its working now :) |
| 13:19.47 | TeknoJuce | one moment |
| 13:20.58 | TeknoJuce | so its dailing now but I get no speaker/mic |
| 13:21.10 | TeknoJuce | just dead space when someone picks up |
| 13:23.30 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:23.30 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:23.36 | dandre | kaldemar: ok I'll try to use GLOBAL |
| 13:24.22 | dandre | but is there some cleaner way to solve my issue than usingi some complicated variable setup? |
| 13:25.19 | *** join/#asterisk russellb (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 13:25.20 | *** mode/#asterisk [+o russellb] by ChanServ |
| 13:27.24 | TeknoJuce | kaldemar, http://pastebin.com/WErGnJfS |
| 13:28.05 | dimm | what does mean this line? |
| 13:28.11 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 13:28.16 | dimm | -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/117-00000011", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 38") in new stack |
| 13:29.02 | *** part/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 13:31.55 | kaldemar | dandre: don't dial a local channel to get into the queue in the first place. |
| 13:33.20 | kaldemar | dimm: a network issue for some reason. |
| 13:34.46 | dimm | kaldemar, can i do something like this - can asterisk always do hookup with fxo (i meant i not hearing anything on some dialing, but mixmonitor recording sound well) |
| 13:36.25 | dandre | I don't directly enter the queue dialing the local channel. In fact dialing the local channel ends in trying an sip device and if it is busy then go to the queue |
| 13:37.07 | TeknoJuce | Thank you for your help so far Kaldemar |
| 13:38.00 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 13:38.21 | kaldemar | TeknoJuce: you have the "chan_unistim.c:2073 start_rtp: Unable to create RTP session" part of the debug that looks like a reason for no audio. i've never used unistim though. |
| 13:39.56 | kaldemar | dimm: i don't quite understand what you mean. |
| 13:40.25 | kaldemar | dandre: you better give a more accurate description on what your dialplan is up to at this point. |
| 13:40.32 | TeknoJuce | so do you think that this phone doesnt support that rtp audio? |
| 13:41.07 | dimm | kaldemar, my softphone work fine, but when i dialing on some numbers then i not hear sound in softphone, but mixmonitor recording sounds. why it can be? (PSTN, fxo) |
| 13:44.14 | kaldemar | TeknoJuce: i don't know about that, it just looks like the unistim channel driver does not like the address for reason. see this: https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18540 |
| 13:44.34 | kaldemar | dimm: no idea. |
| 13:44.52 | dimm | kaldemar, :-) |
| 13:46.19 | *** join/#asterisk bn-7bc (~bjarne-im@213.236.252.25) |
| 13:46.38 | *** join/#asterisk Carlos_mundoR (~Carlos@246.193.165.83.static.mundo-r.com) |
| 13:46.56 | Carlos_mundoR | hello |
| 13:47.25 | Carlos_mundoR | im just new to asterisk and may i ask if someone can help me with the install? |
| 13:48.29 | TeknoJuce | you might not want to use google talk but this is basically the setup install steps |
| 13:48.29 | Carlos_mundoR | just bought an analog card Digium 410P, installed it on the pc. Read the users manual "https://www.digium.com/en/supportcenter/documentation/viewdocs/TDM400P" |
| 13:48.30 | TeknoJuce | http://forums.plugpbx.org/index.php/topic,247.0.html |
| 13:48.45 | *** join/#asterisk Zathraz (~Zzz@a83-163-198-186.adsl.xs4all.nl) |
| 13:48.49 | Carlos_mundoR | and still no luck to start |
| 13:49.03 | kaldemar | TeknoJuce: what version of asterisk are you using? |
| 13:49.34 | TeknoJuce | one in the topic |
| 13:49.56 | TeknoJuce | 1.8.3.2 |
| 13:50.03 | *** part/#asterisk dimm (~admin@unaffiliated/dimm) |
| 13:50.11 | *** join/#asterisk dimm (~admin@unaffiliated/dimm) |
| 13:50.25 | Carlos_mundoR | AsteriskNOW 1.7.1 32-bit |
| 13:50.44 | *** part/#asterisk dimm (~admin@unaffiliated/dimm) |
| 13:51.03 | Carlos_mundoR | seems this old year 2005 manul is not valid... |
| 13:53.47 | *** join/#asterisk cerberus_za (~coert@dsl-185-78-136.dynamic.wa.co.za) |
| 13:59.04 | kaldemar | Carlos_mundoR: what exactly are you trying to install? asterisk, asterisknow or the card on a machine that already has asterisk or asterisknow? |
| 13:59.20 | Carlos_mundoR | i installed the ISO |
| 13:59.36 | Carlos_mundoR | days after (today), installed the card. |
| 14:00.07 | Carlos_mundoR | i managed to create a new analog trunk. |
| 14:00.36 | Carlos_mundoR | Asterisk detects the card because when i Create the trunk (via Web), i can select the 4 analog channels |
| 14:01.04 | Carlos_mundoR | my problem is -- How do i check via CLI if card is well recognized. |
| 14:01.29 | Carlos_mundoR | <PROTECTED> |
| 14:01.55 | TeknoJuce | from your question kaldemar should I try Asterisk 1.8.4-rc2? people say it works with 1.6 but I dont want to use that as all the updates for gv are in 1.8 |
| 14:02.00 | Carlos_mundoR | last version 1.7.1 32-bit |
| 14:02.13 | TeknoJuce | guess I am stuck between a rock and a hard place :D |
| 14:02.35 | kaldemar | Carlos_mundoR: "dahdi show channels" should list you the channels if they are properly configured on the asterisk side. |
| 14:03.34 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 14:03.42 | Carlos_mundoR | ok, i do not have dahdi show channels command instead ive |
| 14:04.10 | Carlos_mundoR | dahdi_cfg dahdi_genconf dahdi_hardware --- etc and a lot of more |
| 14:04.19 | Carlos_mundoR | will try with dahdi_hardware |
| 14:04.36 | Carlos_mundoR | yey |
| 14:04.39 | Carlos_mundoR | pci:0000:01:06.0 wctdm24xxp+ d161:8005 Wildcard TDM410P |
| 14:04.46 | Carlos_mundoR | its ther |
| 14:06.23 | *** join/#asterisk serafie (~erin@nat/digium/x-lrjhmtqixypnycup) |
| 14:06.44 | *** join/#asterisk mawhii (~mawhii@123.219.119.70.cfl.res.rr.com) |
| 14:08.01 | Carlos_mundoR | input Dahdi_scan and prompt issued |
| 14:08.02 | Carlos_mundoR | [root@Sip asterisk-addons-1.6.2.3]# dahdi_scan |
| 14:08.03 | Carlos_mundoR | [1] |
| 14:08.03 | Carlos_mundoR | active=yes |
| 14:08.03 | Carlos_mundoR | alarms=OK |
| 14:08.03 | Carlos_mundoR | description=Wildcard TDM410P Board 1 |
| 14:08.03 | Carlos_mundoR | name=WCTDM/0 |
| 14:08.03 | Carlos_mundoR | manufacturer=Digium |
| 14:08.04 | Carlos_mundoR | devicetype=Wildcard TDM410P (VPM100M) |
| 14:08.04 | Carlos_mundoR | location=PCI Bus 01 Slot 07 |
| 14:08.05 | Carlos_mundoR | basechan=1 |
| 14:08.05 | Carlos_mundoR | totchans=4 |
| 14:08.06 | Carlos_mundoR | irq=233 |
| 14:08.06 | Carlos_mundoR | type=analog |
| 14:08.07 | Carlos_mundoR | port=1,FXO |
| 14:08.07 | Carlos_mundoR | port=2,FXO |
| 14:08.32 | kaldemar | Carlos_mundoR: dahdi_cfg and so one are commands in the OS shell, "dahdi show channels" is an asterisk CLI command, which you attach to with "asterisk -vvvr". |
| 14:08.35 | kaldemar | ~pb |
| 14:08.36 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 14:08.48 | Carlos_mundoR | ahh i see |
| 14:08.58 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 14:09.56 | Carlos_mundoR | issued command dahdi show channels and ive info |
| 14:10.05 | Carlos_mundoR | Parsing '/etc/asterisk/asterisk.conf': == Found |
| 14:10.05 | Carlos_mundoR | <PROTECTED> |
| 14:10.06 | Carlos_mundoR | Connected to Asterisk 1.6.2.11 currently running on Sip (pid = 2727) |
| 14:10.06 | Carlos_mundoR | Verbosity is at least 3 |
| 14:10.06 | Carlos_mundoR | Sip*CLI> dahdi show channels |
| 14:10.06 | Carlos_mundoR | <PROTECTED> |
| 14:10.06 | Carlos_mundoR | <PROTECTED> |
| 14:10.07 | Carlos_mundoR | <PROTECTED> |
| 14:10.07 | Carlos_mundoR | <PROTECTED> |
| 14:10.08 | Carlos_mundoR | <PROTECTED> |
| 14:10.08 | Carlos_mundoR | <PROTECTED> |
| 14:10.18 | Carlos_mundoR | so card is there... and recognised. |
| 14:10.40 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 14:11.37 | Carlos_mundoR | now how can i configure asteriskNow to use the analog card?. I created a new analog trunk and i was able to configure options. |
| 14:11.53 | Carlos_mundoR | next step is to create Outgoing Calling rules? |
| 14:12.28 | kaldemar | Carlos_mundoR: raed that pastebin stuff by infobot |
| 14:13.34 | *** join/#asterisk dimm (~admin@unaffiliated/dimm) |
| 14:13.37 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 14:13.44 | kaldemar | Carlos_mundoR: the channels seem to be properly configured. for help with the GUI, ask in #asterisknow or #freepbx (assuming that your version of asterisknow has freepbx). |
| 14:14.18 | Carlos_mundoR | Kaldemar: Ok will switch to #asterisknow channel. Thank you very muvh. |
| 14:21.33 | pabelanger | Carlos_mundoR: yes, in the future please use pastebin |
| 14:24.41 | dimm | sip provider - nat - asterisk-1.6.2.16.1 - peer am i right that in this case i must use nat=yes in general section of asterisk, and nat=no in peer properties ? |
| 14:27.56 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:29.03 | kaldemar | dimm: nat=no is not really needed if you define localnet along nat=yes in general. also remember to set externaddr. |
| 14:30.07 | Kobaz | p3nguin: poke |
| 14:33.13 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 14:34.06 | Carlos_mundoR | sorry new to irc do not know waht pastebin is. Sorry. Will remember. |
| 14:39.49 | dandre | kaldemar: here is a striped down version of my dialplan: http://pastebin.fr/10809 |
| 14:39.49 | dandre | the inbound context is kwin-000000 |
| 14:40.17 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 14:40.37 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 14:41.09 | dandre | the background instruction is never called |
| 14:41.36 | *** join/#asterisk sizzers (~k_killah@c-68-41-174-94.hsd1.mi.comcast.net) |
| 14:42.37 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 14:45.17 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 14:48.13 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:48.13 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:48.42 | leifmadsen | morning all |
| 14:48.45 | creativx | yo o |
| 14:49.24 | leifmadsen | so I'm trying to think of the best approach for this queue. I have a receptionist with 2 line buttons, and I want to place her into a queue, but I want her to be able to receive multiple calls. I'm thinking I probably need to give her two separate registrations for those line buttons in order to effectively control whether she is busy or not, and add both registrations as members to the queue. |
| 14:49.29 | leifmadsen | that sounds right, right? |
| 14:52.38 | _Corey_ | leifmadsen: What kind of phone? |
| 14:52.43 | leifmadsen | Polycom 335 |
| 14:53.30 | _Corey_ | Well, you could just tell the phone to use 1 line key per call, providing you don't use ringinuse on the queue |
| 14:53.44 | _Corey_ | the phone will just BUSY back when she's on 2 calls |
| 14:54.27 | _Corey_ | We typically do a 550 or 650 for a receptionist and pretty much the same thing |
| 14:54.56 | leifmadsen | gotcha, ok that makes sense |
| 14:55.39 | leifmadsen | thanks for thinking it through with me :) |
| 14:55.46 | _Corey_ | Those 330/335's are a PITA for higher call volume though :( |
| 14:56.26 | _Corey_ | no prob |
| 14:58.07 | leifmadsen | ya I only need to handle 2 lines at a time though |
| 14:58.16 | leifmadsen | I might get them to upgrade her phone to something with more lines at some point |
| 14:58.33 | _Corey_ | The issue with those is the screen size and lack of color LED to indicate line status |
| 14:58.35 | leifmadsen | but I figure if there is a queue, then just doing 2 lines at a time makes it so she isn't just answering calls and putting them on hold |
| 14:58.57 | leifmadsen | ya I'm not using SLA at this location so the line status doesn't change |
| 14:59.17 | _Corey_ | ouch, i hope not :) |
| 14:59.22 | _Corey_ | i mean in use vs. holding |
| 14:59.26 | *** join/#asterisk pgrace (~pgrace@hermes.vsix.me) |
| 14:59.48 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 14:59.54 | leifmadsen | ah |
| 14:59.55 | _Corey_ | never underestimate a red/green light |
| 14:59.57 | pgrace | Is it possible in asterisk 1.4 to register via TCP in sip? having a devil of a time figuring out how to specify sip in the register line. It appears "tcp://" prefix does not work in 1.4 |
| 15:00.04 | leifmadsen | pgrace: no |
| 15:00.12 | leifmadsen | pgrace: TCP is only 1.8 and above |
| 15:00.20 | pgrace | TLS you mean |
| 15:00.30 | leifmadsen | I mean what I said |
| 15:00.31 | pgrace | TCP is supported in 1.6 fine? |
| 15:00.36 | _Corey_ | i had a receptionist threaten to quit over a cisco 7960 once... we gave her a polycom and those red/green lights calmed her down |
| 15:00.44 | leifmadsen | _Corey_: :) |
| 15:01.03 | leifmadsen | _Corey_: know what Polycom changes linekey to in 3.3.1? |
| 15:01.07 | _Corey_ | suppresses memory |
| 15:01.26 | titter | I just have each line key as multiple registrations, and use busylevel=1 and ringinuse=no in the queue with the Polycom's. Although I usually use 501/550 so I have more than 2. |
| 15:01.39 | _Corey_ | leifmadsen: hmm, no i'm on 3.2.3 pretty much everywhere |
| 15:01.45 | leifmadsen | gotcha |
| 15:01.49 | leifmadsen | ok I'll keep looking:0 |
| 15:02.45 | pgrace | leif: You're saying TCP is only supported in 1.8 and above; are you trying to indicate in an obtuse way that tcp support in 1.6 is not stable? Because we use a whole lot of tcp in 1.6 and it seems fine. |
| 15:03.06 | leifmadsen | pgrace: ok then I was wrong and it is in 1.6.2 and above, but definitely not in 1.4 |
| 15:03.17 | leifmadsen | check the CHANGES file to find out what features were added when |
| 15:03.20 | pgrace | leif: ok, good enough. Thanks |
| 15:03.44 | leifmadsen | _Corey_: ah it's the same thing, I just need the reg-advanced.cfg and not reg-basic.cfg :) |
| 15:04.11 | _Corey_ | just when i think we've got those damn config files mastered... |
| 15:04.26 | titter | Ya 3.3.1 is silly |
| 15:04.55 | _Corey_ | seriously, we just re-wrote our config template script like 2 weeks ago... bah |
| 15:05.06 | *** part/#asterisk pgrace (~pgrace@hermes.vsix.me) |
| 15:05.17 | leifmadsen | ugh... I need a new config script too for 3.3.1 |
| 15:05.30 | leifmadsen | doing it all manually at this point as I didn't have time to write one |
| 15:05.36 | leifmadsen | and only 20 phones, so..... |
| 15:06.13 | _Corey_ | I'd offer to share, but apparently it wouldn't work :) |
| 15:06.26 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 15:06.51 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 15:08.30 | titter | I wrote mine in ASP.net and have it to the point where it creates the sip and dialplan I need ... so I just manually check the conf files once I am done, and reload. |
| 15:08.55 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 15:08.55 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 15:13.06 | *** join/#asterisk sizzers (~k_killah@c-68-41-174-94.hsd1.mi.comcast.net) |
| 15:13.33 | kaldemar | dandre: if you mean that the one on line 51 is never called, i have to say no idea why because you're not telling what Dial(Local/6000@from-internal/n) does. |
| 15:18.53 | elb | TeknoJuce: you have Google Talk inbound working? |
| 15:19.19 | elb | I've been unable to make it go ... the dialing party just keeps hearing ringing even after I pick up |
| 15:19.37 | elb | and I get Unable to han |
| 15:19.44 | elb | dle indication 3 for 'Gtalk/+<number>' on the console |
| 15:20.23 | elb | outbound is also giving me fits ... the dialed number never rings, and after a few rings Asterisk says the remote extension is busy/unavailable |
| 15:22.25 | dandre | kaldemar: here it is: |
| 15:22.27 | dandre | [ Context 'from-internal' created by 'pbx_config' ] |
| 15:22.27 | dandre | <PROTECTED> |
| 15:22.27 | dandre | <PROTECTED> |
| 15:24.41 | *** join/#asterisk Janos (~chatzilla@190.10.52.113) |
| 15:25.17 | dimm | is it a magic? when i dialing then i go to router and look at " tcpdump -i eth 0 'net <address of my sip provider>' " after pair or four seconds i can see udp traffic (rtp) |
| 15:25.51 | dimm | then i try to dialing to another number and not see the same traffic |
| 15:26.02 | dimm | i'm stupid :) |
| 15:26.20 | dimm | second time the dialing is go via dahdi :) |
| 15:27.34 | *** join/#asterisk diatonic (~diatonic_@mail.clearwater-research.com) |
| 15:27.35 | dimm | btw second time i not hearing any sound of conversation. Then i go to mixmonitor and i can hearing all. Why it can be? What start point to start diagnostic? |
| 15:29.03 | Janos | hi there, i need to get a specific value of the channel variable, example in this channel variable "SIP/2000-00063f19" i need to get the 2000, is there any other variable that is set to that value ? "CALLERID(num)" is not good since it's been overwritten along the way. Otherwise which would be a simple way to split that variable into it's 3 parts ? |
| 15:32.00 | *** join/#asterisk jmchado (~jmchado@75-145-240-36-waynesboro-va.hfc.comcastbusiness.net) |
| 15:32.14 | *** join/#asterisk chazzam (~chazz@173-24-239-247.client.mchsi.com) |
| 15:32.19 | jmchado | anyone available to help with a sip channel setup? |
| 15:33.00 | Janos | looks like regex would be the way to go if it supported groups |
| 15:34.11 | jmchado | Buehler? |
| 15:35.15 | *** join/#asterisk rlankfo (rlankford@hahainyourface.com) |
| 15:35.26 | rlankfo | hello, can someone point me in the right direction for sending/recving faxes, asterisk/freepbx setup |
| 15:35.47 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:37.00 | wdoekes2 | Janos: you could ${CUT} |
| 15:37.17 | Janos | wdoekes2: thanks, will check that one out |
| 15:37.28 | wdoekes2 | rlankfo: #freepbx |
| 15:37.37 | jmchado | I am trying to setup my PBX to make outgoing calls, and keep receiving NOTICE[1327] call rejected because extension not ofund |
| 15:37.39 | jmchado | found* |
| 15:37.42 | rlankfo | wdoekes2: thanks |
| 15:37.59 | wdoekes2 | jmchado: core set verbose 10 |
| 15:38.05 | jmchado | I did |
| 15:38.19 | wdoekes2 | in that case you're not entering the right context |
| 15:38.35 | jmchado | right, but im not sure where I went wrong may I pm you> |
| 15:38.41 | wdoekes2 | no you may not |
| 15:38.51 | wdoekes2 | account needs to have a context= var, right? |
| 15:39.11 | wdoekes2 | in that [context] in the dialplan (extensions.conf) you need a pattern that catches the outbound number |
| 15:39.45 | wdoekes2 | say you're calling 01234, you could go with exten => _0X!,1,Dial(SIP/${EXTEN}@my-sip-trunk) |
| 15:39.54 | jmchado | [outbound-long-distance] |
| 15:41.12 | jmchado | exten => _9NXXNXXXXXXX,1,Dial (${OUTBOUNDTRUNK}/${EXTEN:1}) |
| 15:41.43 | wdoekes2 | and context=outbound-long-distance ? |
| 15:41.49 | jmchado | yes |
| 15:42.01 | wdoekes2 | and you're dialing 920020000000 ? |
| 15:42.25 | jmchado | no X isnt caps |
| 15:43.10 | jmchado | 9 is extension for line out |
| 15:43.51 | wdoekes2 | whatever.. if you're not dialing something what matches the pattern, you'll get the "extension not found" |
| 15:44.12 | wdoekes2 | try with exten => 123,1,NoOp(hi.. I'm calling 123) |
| 15:44.14 | wdoekes2 | and dial 123 |
| 15:46.32 | jmchado | ok, Autofallthrough, channel SIP/1001-00000161 status is 'UNKNOWN' |
| 15:46.44 | wdoekes2 | b.t.w. you might want to remove that space between Dial and ( |
| 15:46.53 | wdoekes2 | did you see the NoOp then? |
| 15:47.02 | jmchado | yes |
| 15:47.14 | wdoekes2 | ok.. then everything works, except for your dial skills |
| 15:47.27 | wdoekes2 | try a phone number that matches the pattern to dial out |
| 15:47.40 | jmchado | one sec...still laughing |
| 15:47.49 | jmchado | you're right about above |
| 15:48.39 | jmchado | ok as a test I want to place a phone call to a U.S. number how should I format? |
| 15:49.41 | wdoekes2 | depends on your ${OUTBOUNDTRUNK} I suppose.. over here in Europe we dial 0-regioncode-number |
| 15:51.03 | *** join/#asterisk _omer (~omer@182.178.198.31) |
| 15:51.42 | _omer | anyone who has used A2billing? I just have a question about it's features |
| 15:52.30 | *** join/#asterisk UnixDev (~UnixDev@unaffiliated/unixdev) |
| 15:52.38 | _omer | does a2billing have CALL FORWARDING, CALL RECORDING, VOICE mail ? |
| 15:53.47 | jmchado | ok wdoeskes2, new issue No channel type registered for $OUTBOUNDTRUNK |
| 15:56.44 | jmchado | ok wdoekes2, new issue No channel type registered for $OUTBOUNDTRUNK, I know this to mean my channel may not be registered, so I do show channeltypes and its there |
| 16:02.02 | jmchado | New issue No channel type registered for $OUTBOUNDTRUNK, I know this to mean my channel may not be registered, so I do show channeltypes and its there |
| 16:06.49 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:07.20 | wdoekes2 | jmchado: what is the value of ${OUTBOUNDTRUNK}? |
| 16:07.51 | wdoekes2 | it should be something like SIP/my-sip-trunk, where [my-sip-trunk] is a context in your sip.conf with the details for the outbound trunk |
| 16:08.06 | jmchado | aww junk, that may be the issue |
| 16:08.08 | jmchado | SIP/1 |
| 16:08.44 | _omer | anyone who have used A2billing? |
| 16:08.59 | wdoekes2 | nope, still not _omer |
| 16:09.08 | _omer | oops |
| 16:10.23 | jmchado | ok, so I'm using s3.voipvoip.com as my trunk... |
| 16:10.37 | jmchado | \and the trunk context is [VOIPVOIP] |
| 16:12.26 | *** join/#asterisk Janos (~chatzilla@190.10.52.113) |
| 16:15.10 | wdoekes2 | and in [globals] in extensions.conf you have OUTBOUNDTRUNK=SIP/VOIPVOIP ? |
| 16:15.28 | jmchado | yes |
| 16:16.08 | wdoekes2 | if you replace Dial with NoOp |
| 16:16.14 | wdoekes2 | what does it say? |
| 16:16.48 | jmchado | status is 'UNKNOWN' |
| 16:16.52 | eject_ck | [Mar 21 17:14:17] WARNING[13989]: res_fax.c:1994 sendfax_t38_init: Audio FAX not allowed on channel 'Local/46933xxxxx@outbound-allroutes-4523;1' and T.38 negotiation failed; abo rting. [Mar 21 17:14:17] ERROR[13989]: res_fax.c:2223 sendfax_exec: error initializing channel 'Local/46933xxxxx@outbound-allroutes-4523;1' in T.38 mode |
| 16:17.16 | eject_ck | I have alaw passthrough and t38 support on provider's side |
| 16:17.49 | wdoekes2 | jmchado: before that? you did have core set verbose 10? |
| 16:17.59 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 16:18.13 | jmchado | full message read... NoOp ("SIP/1001-0000016a") in new stack ==Auto fallthrough, channel 'SIP/1001-0000016a' status is unknown |
| 16:18.20 | jmchado | yes, core verbose set to 10 |
| 16:18.42 | wdoekes2 | you never did remove that excess space, did you? |
| 16:19.02 | jmchado | no... |
| 16:19.33 | jmchado | wait the space was gone |
| 16:19.51 | jmchado | same error |
| 16:20.31 | eject_ck | any ideas why I'm getting this ? |
| 16:20.57 | jmchado | so its still staying No channel type registered for $OUTBOUNDTRUNK, which makes me think somethinng is wrong in my sip file |
| 16:23.19 | *** join/#asterisk cyborg-one (1000@85-238-125-95.broadband.tenet.odessa.ua) |
| 16:25.46 | jmchado | SIP lloks right to me... |
| 16:26.04 | *** join/#asterisk JonnyD_work (~Jon@173.226.80.154) |
| 16:30.28 | jmchado | ok so I think I had a typo which i corrected |
| 16:30.36 | p3nguin | kobaz: 'sup? |
| 16:30.36 | jmchado | no I am getting a congestion message |
| 16:32.43 | p3nguin | If you don't solve it by a couple of hours, I'll try to help you out. (I have to be afk for a while, or I'd help now.) |
| 16:35.10 | Janos | ok this is driving me crazy, can anyone tell me if you see any error in this line "exten => s,n,Set(channel_temp=${CUT(${CHANNEL},-,1-)})" ? |
| 16:35.56 | jmchado | your brackets dont look right Janos |
| 16:36.55 | pigpen | hmm..I don't think it is brackets. |
| 16:37.20 | _Corey_ | } seems missing |
| 16:37.22 | pigpen | Janos, you may want stick in pastebin of what you are trying to do with that command. |
| 16:37.36 | pigpen | maybe it is my font on the screen....... |
| 16:37.43 | pigpen | moves it to a better monitor. |
| 16:37.50 | jmchado | did you try {CUT}(${CHANNEL},-,1-}) |
| 16:39.26 | Janos | _Corey_: i have two { and two }, i woke up stupid this morning and that's prolly why i don't see it |
| 16:39.41 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 16:40.00 | Janos | jmchado: according to the example in voip-info.org my setup should be right |
| 16:40.22 | wdoekes2 | CUT(varname,char-delim,range-spec) |
| 16:40.26 | wdoekes2 | varname, not value |
| 16:40.39 | wdoekes2 | i.e. ${CUT(CHANNEL,... |
| 16:41.07 | jmchado | ahh well |
| 16:41.16 | Janos | wdoekes2: thanks, i think that's the right answer |
| 16:41.18 | wdoekes2 | or.. if that's unset for some reason.. Set it to a temp variable (I believe EXTEN has that problem) |
| 16:41.33 | Janos | goes out for a coffee ... |
| 16:41.53 | Janos | thinks it must be the sun or something |
| 16:41.56 | jmchado | ok so Everyone is busy/congested at this time.... I've seen many people post on this.. anyone have an answer |
| 16:42.19 | wdoekes2 | jmchado: sip set debug peer VOIPVOIP |
| 16:43.14 | *** join/#asterisk wire[speed] (~wirespeed@unaffiliated/wirespeed/x-6092358) |
| 16:43.29 | jmchado | unable to get Ip address of peer VOIPVOIP |
| 16:44.02 | wdoekes2 | that's your problem right there ;) |
| 16:44.21 | jmchado | I have it listed in the fromdomain: |
| 16:45.13 | wdoekes2 | ok.. and if you sip set debug ip <ip-of-your-sip-trunk> |
| 16:45.18 | jmchado | also when I do a sip reload, I get the message Section 'peers' lacks type |
| 16:45.59 | nestAr | anyone doing anything with Vitelity's SMS service and *? |
| 16:46.08 | wdoekes2 | (the fromdomain is not needed btw.. do you register=> to VOIPVOIP?) |
| 16:47.05 | jmchado | register=> useraccount:password@69.90.209.57/username |
| 16:47.14 | eject_ck | another strange error [Mar 21 17:46:08] NOTICE[17501]: channel.c:4046 __ast_read: Dropping incompatible voice frame on Local/46933XXXXX@outbound-allroutes-f457;1 of format alaw since our native format has changed to 0x40 (slin) |
| 16:47.34 | eject_ck | how should I translate it ? |
| 16:47.50 | jmchado | sip set debug 69.90.209.57 show Usage: sip set debug.... |
| 16:48.00 | wdoekes2 | sip set debug ip 69.90.209.57 |
| 16:48.08 | jmchado | Enables dumping of SIP packets for debugging purposes etc. |
| 16:48.44 | jmchado | SIp debugging enabled |
| 16:49.06 | jmchado | Really destroying dialog ...................:Register |
| 16:50.51 | jmchado | do I need to change my VOIPVOIP context to IP? |
| 16:51.23 | wdoekes2 | no |
| 16:51.41 | wdoekes2 | you need to look at sip output when dialing |
| 16:52.25 | jmchado | where? |
| 16:52.58 | wdoekes2 | when sip debugging is enabled, you'll see lots of it in your console |
| 16:53.25 | wdoekes2 | or it's not calling (the right device) |
| 16:54.10 | *** join/#asterisk JonnyD_work_ (~Jon@cpe-071-075-036-057.carolina.res.rr.com) |
| 16:54.58 | jmchado | the sip debug is showing me ping to the sip trunk it seems |
| 16:55.10 | jmchado | but when I call outbound all I get is congestion message |
| 16:57.34 | wdoekes2 | add host=69.90.209.57 to the [VOIPVOIP] |
| 16:58.35 | jmchado | no change |
| 16:58.53 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 16:59.25 | wdoekes2 | replace the Dial with Dial(SIP/${EXTEN:1}@VOIPVOIP) |
| 17:00.54 | jmchado | that worked |
| 17:00.58 | jmchado | what was the issue? |
| 17:01.59 | jmchado | duude, I've been working on this thing for a week you want work? |
| 17:03.03 | wdoekes2 | your ${OUTBOUNDTRUNK} variable is still wrong for some reason |
| 17:03.39 | wdoekes2 | and the register=> only makes them know you, not the other way around |
| 17:03.46 | wdoekes2 | (that's what the host= fixed) |
| 17:03.52 | jmchado | ok |
| 17:04.06 | jmchado | so will using SIP instead of OUTBOUND work |
| 17:04.06 | titter | wdoekes2: thanks for that link Friday ... saved me a lot of work, until our Exchange cluster crashed this weekend lol ... so it was a long weekend regardless. |
| 17:04.48 | wdoekes2 | jmchado: yes.. this works fine |
| 17:05.22 | jmchado | ok so how can I setup up a variable so that I can use this trunk to call any number instead of the programmed number i put in my dialplan |
| 17:05.23 | wdoekes2 | haha titter, good to know |
| 17:05.45 | wdoekes2 | that's what the ${EXTEN} does along with the pattern matching |
| 17:06.16 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 17:06.43 | wdoekes2 | exten => _0!,1,Dial(SIP/${EXTEN}@VOIPVOIP) will match any number starting with a zero and pass it along to VOIPVOIP |
| 17:06.51 | jmchado | pattern matching in the beginning of the exten => line? |
| 17:07.02 | wdoekes2 | ~pattern |
| 17:07.05 | wdoekes2 | hm.. |
| 17:07.11 | jmchado | you explained it |
| 17:07.20 | wdoekes2 | http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 17:08.39 | jmchado | I have it open, must have missed that bit along with everything you walked me through |
| 17:08.48 | jmchado | <PROTECTED> |
| 17:10.05 | jmchado | seriously yo uwant work? |
| 17:10.44 | wdoekes2 | haha.. I'm set, but thanks :) |
| 17:11.09 | jmchado | I figure I'm going to have more issues as I set up |
| 17:11.16 | dimm | can i just update to 1.8 from 1.6 without new config files? |
| 17:11.20 | dimm | i use freepbx |
| 17:11.21 | jmchado | I can pay for support |
| 17:12.37 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 17:15.12 | *** join/#asterisk [netman] (~netman@40.Red-79-156-254.staticIP.rima-tde.net) |
| 17:19.08 | wire[speed] | Hi, Does anyone have a working config for v1.6 to allow call forwarding and forward release. I have tried several examples both from books and online within my extensions.conf file but nothing seems to work. |
| 17:21.21 | *** join/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 17:24.39 | *** join/#asterisk [netman] (~netman@80.31.243.134) |
| 17:27.43 | dandre | can anyone help me with my dialplan problem: http://pastebin.fr/10814 |
| 17:28.17 | *** join/#asterisk sequencer (~something@196.218.255.29) |
| 17:28.46 | sequencer | hi, am still having a huge problem with calls being dropped randomly, am pastin the call og just now |
| 17:31.38 | *** join/#asterisk voipnet-tech (~voipnet-t@66.63.72.130) |
| 17:32.33 | voipnet-tech | hi all- has anyone ever encountered an asterisk crash following the first 200 OK to a register message after doing a commandline reload |
| 17:33.39 | sequencer | here we go http://pastebin.com/n64va2af |
| 17:35.17 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 17:37.01 | *** join/#asterisk [netman] (~netman@15.Red-83-41-12.dynamicIP.rima-tde.net) |
| 17:39.50 | *** join/#asterisk juliocesarlhg (~jcesarg@190.234.250.59) |
| 17:40.29 | juliocesarlhg | help with mgcp gateway |
| 17:42.19 | sequencer | anyone here? |
| 17:47.15 | JerJer | nobody but us chickens |
| 17:48.07 | tzafrir | juliocesarlhg, still haven't learned how to ask more specific questions? |
| 17:49.14 | juliocesarlhg | hehe |
| 17:49.19 | juliocesarlhg | sorry tzafrir |
| 17:49.36 | juliocesarlhg | i just woke up |
| 17:51.41 | WIMPy | Unable to locate coffee - Operator halted. |
| 17:57.46 | carrar | Default route to kitchen espresso machine |
| 18:10.57 | juliocesarlhg | i have a gaoke gateway |
| 18:11.03 | juliocesarlhg | with mgcp protocol |
| 18:11.13 | juliocesarlhg | on network configuration says |
| 18:13.31 | juliocesarlhg | nat server= nat data and voice transmit with data net port |
| 18:13.37 | juliocesarlhg | way ip= 192.168.1.80 |
| 18:13.51 | juliocesarlhg | wan netmas= 255.255.0.0 |
| 18:14.00 | juliocesarlhg | lan ip= 192.168.1.90 |
| 18:14.08 | juliocesarlhg | lan netmask 255.255.255.0 |
| 18:14.15 | juliocesarlhg | in my mgcp configuration says |
| 18:14.42 | juliocesarlhg | gateway name 192.168.1.80 it means that it uses wan ip |
| 18:14.51 | juliocesarlhg | gateway ip= 192.168.1.80 |
| 18:15.13 | juliocesarlhg | now my question is, where do i connect the cable on the lan port, o the wan port? |
| 18:18.40 | juliocesarlhg | ?? |
| 18:19.53 | *** join/#asterisk DanFloun (~DanFloun@60.13.gr6.adsl.brightview.com) |
| 18:20.57 | eject_ck | [Mar 21 19:20:20] ERROR[9572]: res_fax.c:1223 generic_fax_exec: channel 'Local/46933xxxxx@outbound-allroutes-3833;1' FAX session '0' failure, reason: 'fax session timed-out' (TIMEOUT) |
| 18:21.16 | eject_ck | what timeout is above ? |
| 18:21.17 | russellb | 6 |
| 18:21.21 | eject_ck | what does it mean ? |
| 18:21.35 | eject_ck | I'm using free fax for asterisk |
| 18:21.50 | eject_ck | it works from one server with exactly same settings and not work from another |
| 18:28.15 | sequencer | could any one tell me what a CNG means ? |
| 18:30.31 | WIMPy | sequencer: CalliNG tone. It was used by modems to tell the calld party what kind of service was expected. |
| 18:30.59 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 18:31.06 | *** join/#asterisk hfb (~hfb@96.247.52.84) |
| 18:32.31 | WIMPy | Today it might be a Comfort Noise generator. |
| 18:41.59 | DanFloun | CuNnilinGuis? |
| 18:42.08 | *** join/#asterisk millsu2 (~brad@mail.serverplus.com) |
| 18:42.21 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 18:43.05 | elb | I've set up Google Talk/Voice on my Asterisk 1.8.3.2 installation, but it doesn't quite work right ... incoming calls ring in, but the calling phone is never signalled that there was a pickup (and Asterisk complains with "Unable to handle indication 3" on the gtalk channel); outgoing calls ring for a few seconds, then report "Everyone is busy/congested at this time" without the dialed phone actually ringing ... any ideas what I'm doi |
| 18:43.26 | *** join/#asterisk dr__ (~duckz@78.96.101.150) |
| 18:46.10 | DanFloun | anyone familiar with uk caller id / tdm410p / asterisk 1.6? |
| 18:46.15 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:50.59 | *** join/#asterisk e-squizo (~fheinz@190.189.199.159) |
| 18:56.46 | *** join/#asterisk Cadey (~x@host81-135-124-94.range81-135.btcentralplus.com) |
| 18:58.32 | Cadey | Hi guys, I think I know the answer to this but not the detail. Why is it when you pick up a call and watch the AMI output it some times fires multiple Bridge, Unlink, Bridge messages. I think its to do with getting the two channels into sycn and so aborts and retrys if they are out of sync and keeps retrying until they are in sync? |
| 19:05.00 | *** join/#asterisk kb3ien (~kb3ien@static-72-80-25-34.nycmny.fios.verizon.net) |
| 19:05.50 | kb3ien | i'm finding no viable means to access local_lostpackets in ast 1.8 CHANNEL(rtpqos,audio,lost_lostpackets) is not understood. what are we using now? |
| 19:07.33 | *** join/#asterisk mersault (~mersault@acihip.tor4.dsl4u.ca) |
| 19:09.19 | kb3ien | maybe if i could get NoOp to print to the console the way it did in 1.4 ? anyone know that ? |
| 19:09.20 | *** join/#asterisk FlaPer87 (~FlaPer87@unaffiliated/flaper87) |
| 19:10.08 | FlaPer87 | In case anyone is interested and willing to contribute: https://github.com/FlaPer87/asterisk-zmq-manager |
| 19:12.44 | *** part/#asterisk voipnet-tech (~voipnet-t@66.63.72.130) |
| 19:13.02 | kb3ien | RTPAUDIOQOS should allow me to monitor the channel still, but i cant findout what takes its place in ast 1.8 |
| 19:13.39 | dimm | When I run the call through an analog line, the tube softphone I can not hear beeps and voice subscribers. Then he hangs up, and I hear sirens, "busy ". In the mixmonitor hear and whistle, and a voice subscriber. Do not tell in what could be the problem? |
| 19:16.19 | kb3ien | CHANNEL(rtpqos,audio,all) is now working ; and i swear I didn't tocuh ANYTHING ... olh weel. |
| 19:17.09 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 19:20.36 | elb | kb3ien: core set verbose 3 will cause NoOp to print to the console |
| 19:21.07 | kb3ien | it was set to 5 a typo in my label was causing the NoOp() not to be in my loop. |
| 19:21.13 | Freeaqingme | How does the asterisk project relate to voip-info.org ? |
| 19:21.45 | kb3ien | curiously. sometimes voip-info.org is right with its documentation, often it is slightly inaccurate, but a good starting place. |
| 19:25.05 | kb3ien | 1300735421,ssrc=1727054931;themssrc=3925670453;lp=0;rxjitter=0.001805;rxcount=1858;txjitter |
| 19:25.05 | kb3ien | 000;txcount=0;rlp=1;rtt=0.000000 |
| 19:25.05 | kb3ien | rtt is not 0! |
| 19:25.48 | kb3ien | max is (all) not really all ? |
| 19:27.45 | *** join/#asterisk JonnyD_work (~Jon@173.226.80.154) |
| 19:31.24 | kb3ien | looks like remote_lostpackets is not available to me. |
| 19:31.35 | *** join/#asterisk MrNemus (~MrNemus@firewall.drgutah.com) |
| 19:31.39 | *** join/#asterisk rkill (~james@nat/digium/x-bslhqybpjsqfaqtr) |
| 19:36.28 | MrNemus | Hi would anyone know why I keep getting this compilie error with centos 5.5 x86_64 and asterisk 1.8.3.2 http://pastebin.com/486ZzQnK |
| 19:41.35 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 19:48.08 | rkill | ~newbook |
| 19:48.08 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 19:54.55 | *** join/#asterisk bullium (~wbradshaw@216.68.250.30) |
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| 20:07.23 | *** join/#asterisk cashback (~cashback@ip68-2-140-46.ph.ph.cox.net) |
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| 20:14.30 | elb | hah that book gives the same gtalk instructions that continue to not work for me :-/ |
| 20:16.08 | *** join/#asterisk raden_work (~jon@66-191-96-74.static.eucl.wi.charter.com) |
| 20:22.22 | *** join/#asterisk bmg505 (~leon@196-209-44-243.dynamic.isadsl.co.za) |
| 20:24.23 | elb | huh, it's messaging mogorman@astjab.org |
| 20:24.29 | elb | and I definitely did not configure that |
| 20:24.30 | p3nguin | elb: What are you having problems with? I've set up Google Voice calling direct to/from Asterisk and it works quite well from what feedback I've received. |
| 20:25.02 | elb | p3nguin: outgoing calls never ring the dialed phone |
| 20:25.08 | p3nguin | Paste up your _entire_ gtalk.conf and jabber.conf for me, masking only passwords. |
| 20:25.19 | elb | one moment |
| 20:28.04 | elb | p3nguin: http://www.pidgin.im/nopaste/148 |
| 20:29.36 | elb | JabberSend() works on the gmail jabber account |
| 20:30.05 | elb | incoming calls ring the incoming phone, but the dialing phone never knows that the sip phone picked up (keeps ringing until it goes to gvoice voicemail) |
| 20:30.24 | elb | outgoing calls ring on my end, then terminate 503, the dialed phone never rings |
| 20:32.45 | elb | p3nguin: actually, as of about 20 minutes ago, incoming calls ring the incoming line briefly, then fail with an internal server error from gtalk ... but that is new -- previously the internal phone would ring through until picked up |
| 20:34.44 | *** join/#asterisk nighty^ (~nighty@tin51-1-82-226-147-104.fbx.proxad.net) |
| 20:35.30 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 20:37.47 | *** part/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 20:40.05 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
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| 20:41.08 | tuxxie | can i set inbound callerid name by number dialed? |
| 20:41.34 | Winkie | so i'm having an annoying problem with sipgate, it seems calls in from them do not supply any authentication, and the only way i can receive calls is to remove the 'secret' line, but they require it on outbound, and setting up seperate peers/users doesn't seem to be working very well for me |
| 20:42.12 | wdoekes2 | insecure=invite |
| 20:42.41 | Winkie | hmm, i swear that is in their list and is on |
| 20:42.44 | Winkie | let me go check |
| 20:43.17 | saliak | I'm having some troubles setting up a sip peer from broadvoice for my incoming calls.. I'm registering successfully (per sip show registry), and can make outbound calls. where i define my sip peer, i put the context as "from-broadvoice". the relevant section of my dialplan looks like http://pastebin.com/xx2SiHcz. when i call i get a "party you are trying to reach is not available to take your call..." message. it does look like the call |
| 20:43.18 | saliak | into asterisk, as i see a "== Using SIP RTP CoS mark 5" message when the call is answered. |
| 20:43.42 | Winkie | wdoekes2: oh man, it wasn't and it has solved the problem i think, thanks |
| 20:43.50 | Winkie | i had an older one, which wa sinsecure=very |
| 20:44.14 | Winkie | oh wait, now i'm getting outbound problems, great :D |
| 20:44.46 | Winkie | nope it does seem fine, thanks again wdoekes2 |
| 20:44.47 | wdoekes2 | tuxxie: you can do lots of things from the dialplan, including setting the callerid |
| 20:44.55 | wdoekes2 | :) |
| 20:46.19 | wdoekes2 | saliak: it's probably calling with a different destination than 's' or '1000' |
| 20:46.24 | p3nguin | elb: Outgoing calls don't work? |
| 20:46.32 | elb | p3nguin: correct |
| 20:46.47 | wdoekes2 | sip set debug on |
| 20:47.06 | jaytee | http://imgur.com/eCkyv |
| 20:47.16 | elb | p3nguin: actually ... after five days of not working, one just worked |
| 20:47.31 | p3nguin | very peculiar |
| 20:47.34 | elb | and all I did since the last that did *not* was remove the [kb8ojh-gtalk] stanza from gtalk.conf |
| 20:47.35 | wdoekes2 | saliak: you should see an see an "INVITE sip:...." when you're dialing in.. |
| 20:47.59 | elb | (since I don't intend to call myself anyway) |
| 20:48.18 | elb | so now outgoing is working, actually, at least with that brief test |
| 20:48.23 | elb | incoming, still not so much |
| 20:50.22 | p3nguin | elb: If I were you, I would strip down both of those configs. You have way more in them than necessary. https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 20:50.42 | p3nguin | elb: I used this wiki page and had google voice calls working in just a few minutes. |
| 20:50.54 | elb | I started with dead simple configs |
| 20:50.54 | saliak | wdoekes2: how do i see what extension it's set to? sip show peer sip.broadvoice.com spits out a lot of stuff, but nothing that looks like "extension" |
| 20:50.54 | elb | they didn't work |
| 20:51.00 | elb | which is where a lot of the extra stuff came from |
| 20:51.07 | elb | note also the second jabber account -- I actually need that |
| 20:51.14 | elb | it's used for JabberSend() for call status notifications |
| 20:51.23 | saliak | wdoekes2: hrm. ok, i see the same message i see if i'm making an outgoing sip call. |
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| 20:51.41 | wdoekes2 | did I say 'sip show peer'? you we're talking about inbound calls right? |
| 20:51.47 | wdoekes2 | never mind the SIP RTP CoS message |
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| 20:54.17 | saliak | wdokes2: yeah, sorry. I'm just wondering if the "sip rtp.." message should tell me about what i might have done wrong. this is my sip.conf for the peer http://pastebin.com/cJzTGWWb .. i believe that it's set to extension 1000 |
| 20:54.54 | wdoekes2 | saliak: if you're talking about the inbound calls failing.. you should 'sip set debug on' and do a call |
| 20:56.16 | p3nguin | elb: I'm just saying I followed the wiki and had it going in minutes. If I can do it, you can do it as well. |
| 20:56.38 | wdoekes2 | yes.. the register line says the default contact is 1000@yourhost.. but sipgate may certainly choose to replace that with a telephone number or perhaps your account-id |
| 20:57.09 | wdoekes2 | so.. unless you watch the traffic, you'll be working in the dark |
| 20:57.26 | elb | p3nguin: yeah, I did too ... and it didn't work :-) |
| 20:57.45 | elb | but ... removing the allow/disallows from [general] and moving them to [guest] seems to have fixed inbound |
| 20:57.57 | elb | outbound |
| 20:57.58 | elb | sorry |
| 20:58.03 | elb | the rest of the stuff is unchanged |
| 20:58.18 | elb | let me see if outbound still works |
| 20:59.32 | saliak | |
| 21:00.33 | dimm | p3nguin, hello, please look my question. may be you know some words related to my question? |
| 21:00.35 | dimm | When I run the call through an analog line, the tube softphone I can not hear beeps and voice subscribers. Then he hangs up, and I hear sirens, "busy ". In the mixmonitor hear and whistle, and a voice subscriber. Do not tell in what could be the problem? |
| 21:00.54 | elb | p3nguin: let me take that back ... inbound worked *once* |
| 22:38.07 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 22:38.07 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3.2 (2011/03/17), 1.6.2.17.2 (2011/03/17), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 22:38.13 | raden_work | TeknoJuce, sorry bro LOL |
| 22:38.42 | raden_work | TeknoJuce, just wore out just seems odd that asterisk and aastra which are used alot together are not working together |
| 22:39.00 | Freeaqingme | <PROTECTED> |
| 22:39.06 | TeknoJuce | the first thing I would suggest when you come to a place for support dont imediatly fly off the handle |
| 22:39.44 | TeknoJuce | keep your cool, some words maybe taking off key like you are dissing the thing you are trying to get support for |
| 22:39.48 | raden_work | Tech_Travis, I have been posting this issue in here for 2 weeks |
| 22:40.05 | TeknoJuce | second come to the table with pastebin logs of errors etc from cli |
| 22:40.12 | raden_work | Im not dissing anything, I love asterisk ... I just dont understand how something like this is happening |
| 22:40.26 | raden_work | TeknoJuce, there are no logs |
| 22:40.31 | raden_work | I mean errors |
| 22:40.41 | raden_work | lemee pastbin when i put someone on hold the SIP debug |
| 22:40.42 | TeknoJuce | its open source ever time you update you are risking stuff just not working right out of the box |
| 22:41.01 | elb | p3nguin: incidentally, the comment by 'Lakeside tech' at the bottom of the gtalk wiki page describes what I see, and the symtpoms I have, precisely |
| 22:41.21 | elb | NAT behind STUN, advertising 192.168.33.1 as well as the external IP, works once, then doesn't work for some time |
| 22:41.23 | TeknoJuce | did he have a conclusion elb |
| 22:41.39 | elb | no |
| 22:41.42 | TeknoJuce | :( |
| 22:41.48 | elb | he modified his chan_gtalk to not advertise the internal IP |
| 22:41.54 | elb | but it didn't fix everything |
| 22:42.20 | TeknoJuce | I brought my i2004 to work today to play with to see if I can get these patches working in vm |
| 22:42.37 | TeknoJuce | as I will be here till 1am :( |
| 22:43.15 | raden_work | http://pastebin.com/ygrnAEy6 |
| 22:43.21 | raden_work | Call placed and removed from hold no music |
| 22:44.27 | raden_work | On the same asterisk box i can place a call on hold with my PAP2T adapter no problem |
| 22:44.32 | TeknoJuce | are you using astrisk -vvvvvr |
| 22:44.35 | raden_work | my polycom at home works fine |
| 22:44.42 | raden_work | TeknoJuce, YES SIR |
| 22:45.06 | TeknoJuce | also word to the wise never use caps lock. |
| 22:45.37 | TeknoJuce | I know its cruise control for cool |
| 22:46.52 | raden_work | lol |
| 22:46.54 | TeknoJuce | also irc seems like an instant answer zone but you can wait and wait till someone reads their backlog and says hey i know that prob happen to me etc |
| 22:47.09 | raden_work | I have posted this at least 15 times |
| 22:47.13 | raden_work | I have this on forums |
| 22:47.18 | TeknoJuce | just dont stop here post to the mailing lists and forums and wait for an answers |
| 22:47.20 | raden_work | I have sent emails to other techs I know |
| 22:47.28 | raden_work | already did all that |
| 22:47.33 | TeknoJuce | mailing list? |
| 22:47.43 | raden_work | I have been in IT for 15 years I know a person needs to have patience but this is nuts |
| 22:48.00 | raden_work | your telling me every aastra phone running on 1.8 doesnt have MOH and no one has a clue ? |
| 22:48.22 | raden_work | i cannot get a single aastra phone to work with any of the 5 1.8 boxes I have used |
| 22:48.52 | raden_work | if debugging is off I dont even see anything happen on CLI when i hit hold like i normally would |
| 22:48.53 | *** join/#asterisk Deeewayne (~dwayne@68-64-220-110.static.forethought.net) |
| 22:48.53 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 22:49.01 | raden_work | I have read through change logs multiple times |
| 22:49.07 | raden_work | Just seems nuts |
| 22:49.11 | TeknoJuce | maybe start small get another box not related one ata and one phone and see if you can get that or use a vm |
| 22:49.19 | TeknoJuce | work around from there |
| 22:49.33 | raden_work | not related ? |
| 22:49.42 | raden_work | howso ? |
| 22:49.46 | TeknoJuce | and in the mean time put your box back to 1.6 until you figure out the testlab |
| 22:50.06 | raden_work | if i go back ill just keep everything at 1.6 till it gets worked out |
| 22:50.13 | raden_work | cause no one seems to know anything |
| 22:50.22 | TeknoJuce | sounds like a plan my man |
| 22:50.44 | TeknoJuce | it will eliviate the pressure of your waiting for an answer |
| 22:51.06 | TeknoJuce | and maybe on the way back to 1.6 you will find out you were maybe doing something silly |
| 22:51.22 | raden_work | why do all the other phone brands work |
| 22:51.38 | raden_work | why do I know 2 other companies having the same problem ? |
| 22:51.53 | TeknoJuce | with such a big jump in versioning if your system is vital always do a testlab with vm's and 1 phone etc |
| 22:51.54 | raden_work | its music on hold not rocket science |
| 22:52.06 | raden_work | TeknoJuce, I agree |
| 22:52.27 | raden_work | this was a last minute forced to do something many issues with other software redo type deal |
| 22:52.34 | raden_work | AKA a major fing mess |
| 22:53.44 | TeknoJuce | Yeah I understand but I would still go back once you came (1.6), then do a test lab till the issue is solved |
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| 22:55.04 | TeknoJuce | takes like 5 mins to setup a vm with virtualbox:ubuntu + astrisk 1.8 and plug a phone in your network make sure you put the vm adapter to bridged so you get an ip from your dhcp and not nated from your computer |
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| 22:56.04 | *** join/#asterisk freakazoid (~sean@208.185.212.98) |
| 22:56.05 | TeknoJuce | or whatever os you are using... |
| 22:56.51 | freakazoid | A friend of mine is running freepbx and thinks he might have been compromised - they see a bunch of extensions with leading underscores and one with "blackhole" in the name getting created in the logs - anyone know if that's normal freepbx admin stuff or if there's a known attack that does something like that? |
| 23:00.17 | raden_work | hmmm |
| 23:01.06 | freakazoid | Hmm, I see app-blackhole on google |
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| 23:29.02 | TeknoJuce | becareful when searching blackhole on google with the safterfilter off freakazoid |
| 23:29.10 | freakazoid | haha thanks |
| 23:29.19 | freakazoid | I just looked for blackhole and freepbx |
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| 23:33.16 | p3nguin | freakazoid: You did ask your FreePBX question in the FreePBX channel before you came here, didn't you? |
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| 23:35.01 | freakazoid | I wasn't even aware there was a FreePBX channel. |
| 23:35.34 | freakazoid | Oh hey look at that? |
| 23:36.35 | p3nguin | Well, this is an Asterisk channel. If your question is regarding Asterisk, someone will probably be along shortly to help you. |
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