| 00:00.21 | [TK]D-Fender | dfgas-cr48: exten => 9203198188,1,NoOp(); <-- you're also running an IVR in a context OFF of a numbered exten which means you could dial it recursively |
| 00:00.25 | dfgas-cr48 | that is just my inbound thats it, its not my whole extensions.conf |
| 00:00.27 | p3nguin | Where extension i in that? |
| 00:00.41 | [TK]D-Fender | dfgas-cr48: We don't see where ANYTHING is dialable in there |
| 00:00.55 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 00:00.58 | [TK]D-Fender | dfgas-cr48: Show us the actual call and the actuall other bits that it could be pointing to |
| 00:01.03 | [TK]D-Fender | because this is masking the problem. |
| 00:01.05 | p3nguin | If you want to know why extension 10 doesn't work, show the entire dial plan. |
| 00:01.14 | [TK]D-Fender | There is no "1" you could dial in there and land on a numbered pattern |
| 00:01.24 | p3nguin | include => internal |
| 00:01.31 | p3nguin | I'd have to see what is in internal. |
| 00:01.36 | *** join/#asterisk nir (~smuxi@192.117.240.253) |
| 00:01.37 | dijib | http://pastebin.com/fZvQkqTe |
| 00:01.38 | [TK]D-Fender | ^ do not see, do not trust. |
| 00:01.52 | dijib | should be the i he has |
| 00:02.05 | dijib | copy pasta powa! |
| 00:02.10 | [TK]D-Fender | There is no "1" there either. |
| 00:02.39 | dfgas-cr48 | p3nguin, dijib posted some code to me one night and in the morning it and it worked with a minor change great but i hadn't realised dijib was in there in saved what he was doing right after me and i could dial anything. its cool but i lost the logs of what he pasted for me to put in there so i had to revert back to a backup file so kids could call today if they needed |
| 00:02.41 | [TK]D-Fender | Oops, separate isse (I hope) |
| 00:02.42 | p3nguin | I'm going to go to the store and get some stuff for supper. When I come back, I want to see the entire dial plan. |
| 00:13.41 | SeRi | dijib: you there? |
| 00:13.45 | citywok | p3nguin: my meetme issue the other day was dahdi was installed but not loaded. whoooops. |
| 00:18.47 | dfgas-cr48 | SeRi, yo |
| 00:18.57 | SeRi | waz up dfgas-cr48 |
| 00:19.03 | dfgas-cr48 | idk |
| 00:19.07 | SeRi | lol |
| 00:19.07 | dfgas-cr48 | lol |
| 00:19.28 | SeRi | you got your stuff fix? |
| 00:19.37 | dfgas-cr48 | not yet |
| 00:20.15 | dfgas-cr48 | i think i know of 2 issues in it yet |
| 00:22.04 | dfgas-cr48 | [Nov 26 18:21:28] WARNING[930][C-000000b8]: pbx.c:11825 pbx_parseable_goto: Priority 'h' must be a number > 0, or valid label |
| 00:22.05 | dfgas-cr48 | <PROTECTED> |
| 00:22.12 | dijib | yes im here SeRi |
| 00:22.23 | SeRi | yeap thats a problem |
| 00:22.27 | SeRi | dijib: conf |
| 00:22.34 | dijib | in it with tony |
| 00:23.25 | dijib | SeRi: i see you trying to connect but its not going throuhg, tony and i are in 2663 |
| 00:31.07 | *** join/#asterisk dfgas (~dfgas@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 00:31.20 | *** join/#asterisk m0spf (~steve@2001:ba8:1f1:f12e::2) |
| 00:34.15 | dijib | http://imgur.com/aT8Eg,iOVYR,XM3Cg,8lT55,WyOKt,Nq0QT |
| 00:35.25 | SeRi | dijib: burn that thing. |
| 00:35.29 | SeRi | Is an alien! |
| 00:40.14 | *** join/#asterisk Echo777 (~echo@71-82-226-158.dhcp.stpt.wi.charter.com) |
| 00:41.06 | Echo777 | ive been trying to accomplish this http://www.dev-random.me/google-voice-asterisk/ for quite a while, downloaded asterisk many different ways but cant get anything to work properly, anyone up to giving me some step by step help, (Linux Mint as Root) |
| 00:44.47 | Echo777 | anyone there? |
| 00:47.41 | Echo777 | hey!? |
| 00:53.04 | dfgas-cr48 | SeRi, where did you go? |
| 01:04.55 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:05.37 | SeRi | dfgas-cr48: one sec |
| 01:09.02 | *** join/#asterisk bmg505 (~leon@196-209-7-128.dynamic.isadsl.co.za) |
| 01:10.41 | p3nguin | It's friggin' cold out there. |
| 01:11.13 | p3nguin | With wind chill, it feels like 32 F. |
| 01:11.59 | p3nguin | Now, where's that dial plan? |
| 01:13.44 | SeRi | p3nguin: his issue was that he had i,n,GoTo(h) |
| 01:13.54 | SeRi | so it was et to hangup on invalid |
| 01:13.57 | p3nguin | Interesting. |
| 01:14.02 | SeRi | s/et/set/ |
| 01:14.45 | p3nguin | Having an invalid priority does not mean it was set to hang up. |
| 01:14.51 | p3nguin | It just means it fails and dies. |
| 01:15.19 | SeRi | agreed. |
| 01:15.34 | SeRi | he wanted to retry again instead. |
| 01:15.46 | p3nguin | I know what it was supposed to do. |
| 01:15.58 | SeRi | Yes Sr. |
| 01:16.06 | SeRi | I need a new job :( |
| 01:19.33 | *** join/#asterisk TSM2 (~the_softw@fw-lon1.wenn.com) |
| 01:23.12 | carrar | Move to the west coast and be a produce picker! |
| 01:23.31 | p3nguin | Lettuce plucker! |
| 01:23.42 | SeRi | fuck the both of ya |
| 01:23.45 | SeRi | LOL |
| 01:24.00 | SeRi | hahahahahaha |
| 01:24.05 | SeRi | That was halarious |
| 01:24.09 | _Corey_ | SeRi: We're hiring... http://www.voneto.com/about-voneto/jobs |
| 01:24.41 | SeRi | _Corey_: can I msg you? |
| 01:24.49 | _Corey_ | sure |
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| 01:35.43 | *** join/#asterisk asteriskmonkey (~philip@206.51.27.151) |
| 01:36.58 | asteriskmonkey | heya wondering if anyone has seen this wierdness before, new install asterisk 10 on centos, did a make config, the serivice start asterisk/status etc.. seems to work, shows its up see it in ps auxx, but i cant connect to it via asterisk -r seems to complain about missing /var/run/asterisk/asterisk.ctl |
| 01:37.03 | asteriskmonkey | anyone had that before? |
| 01:39.31 | *** join/#asterisk deo (~deo@203.177.214.75) |
| 01:42.34 | jpsharp | does /var/run/asterisk/asterisk.ctl actually exist? |
| 01:43.41 | asteriskmonkey | no, but process is running, think i found the cuase |
| 01:43.45 | asteriskmonkey | darn SELINUX :P |
| 01:43.46 | jpsharp | And are you running "asterisk -r" as a user that has read/write privileges to that file. |
| 01:43.54 | SeRi | thats the first mistake of your life |
| 01:43.58 | SeRi | selinux on a pbx |
| 01:44.13 | jpsharp | s/selinux on a pbx/selinux/ |
| 01:44.18 | asteriskmonkey | yeah my bad shoulda check that first, someone else install os on it |
| 01:44.22 | jpsharp | Festering pile of dog shit. |
| 01:44.46 | asteriskmonkey | smacked my head for 10mins until i just twigged in to go check that out :) |
| 01:45.09 | asteriskmonkey | should build a checker into asterisk that blinks red.. WAIT YOU HAVE SELINUX disable it |
| 01:46.24 | jpsharp | Except SELINUX will probably disable that checker. |
| 01:46.26 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-frxiubdgvmbsmqqg) |
| 01:47.44 | asteriskmonkey | lol |
| 01:47.51 | SeRi | lmao |
| 01:58.29 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
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| 02:01.24 | *** join/#asterisk ghost75 (~trechber@dslb-088-064-221-197.pools.arcor-ip.net) |
| 02:02.23 | dijib | back now |
| 02:02.31 | SeRi | ok] |
| 02:06.58 | *** part/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 02:23.45 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
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| 03:03.56 | *** join/#asterisk arfon (~an@adsl-99-184-92-160.dsl.austtx.sbcglobal.net) |
| 03:04.50 | arfon | I have my secret in sip.conf under each extension. When I load asterisk, it ignores secret= how do I stop this? |
| 03:05.06 | p3nguin | There are no extensions found in sip.conf. |
| 03:05.18 | p3nguin | Extensions go in extensions.conf. |
| 03:06.36 | p3nguin | Did you mean something else? |
| 03:06.37 | arfon | Sorry , all the examples had it in sip.conf |
| 03:07.23 | arfon | It's reading it from sip.conf because user 202 connects when I use the OLD (long deleted) secret but not the new one |
| 03:07.28 | p3nguin | Is there an example from the sample sip.conf that you are trying to use? |
| 03:08.03 | p3nguin | What is this 202 device? |
| 03:08.14 | arfon | http://agix.com.au/blog/?p=2656 |
| 03:08.25 | arfon | 202 is a sip softphone |
| 03:09.14 | p3nguin | After you changed the secret and saved the file, did you run "sip reload" in the asterisk CLI? |
| 03:09.32 | arfon | no... I just restarted asterisk |
| 03:09.42 | p3nguin | sip reload is the appropriate thing there.` |
| 03:09.42 | arfon | THAT'S probably the answer I needed |
| 03:09.47 | arfon | TY! |
| 03:09.56 | p3nguin | But restarting totally should have loaded the change. |
| 03:10.10 | arfon | For some reason, it didn't |
| 03:10.16 | arfon | let me try sip reload |
| 03:11.18 | arfon | Hmmm, sip reload worked.... |
| 03:11.32 | arfon | <PROTECTED> |
| 03:13.01 | arfon | Another Q... if I set allowguest=no will that prevent any non-authenicated connections or do I need something else? |
| 03:13.29 | p3nguin | That will reject calls from unknown/unauthenticated devices. |
| 03:13.37 | arfon | Thanks! |
| 03:14.29 | p3nguin | Remember, sip reload to load changes in sip.conf, and dialplan reload to load changes in extensions.conf. |
| 03:15.15 | arfon | Good to know |
| 03:15.53 | arfon | I wonder why they didn't make 'dialplan reload' 'extensions reload'? |
| 03:18.54 | p3nguin | Now that I don't know... but you could create an alias for it if you prefer that. |
| 03:37.48 | arfon | Nah, it seems to be working now... sorta |
| 03:38.47 | arfon | I can call 201 from 202 but 202 to 201 doesn't ring |
| 03:39.45 | p3nguin | Those are, by the way, terrible names for phones. |
| 03:39.53 | arfon | :) thanks |
| 03:40.13 | arfon | How do I dial a phone named "tom"? |
| 03:40.15 | p3nguin | All it does is hinder troubleshooting by confusing people who can't follow configuration and who don't know the difference between phones and extensions. |
| 03:40.30 | p3nguin | What extension does tom have? |
| 03:40.37 | arfon | 201 |
| 03:40.47 | p3nguin | exten => 201,1,Dial(SIP/tom,30) |
| 03:41.03 | arfon | How do I dial tom from a phone keypad? |
| 03:41.14 | p3nguin | You don't. You dial tom's extension. |
| 03:41.32 | arfon | I am VERY new to this |
| 03:41.32 | p3nguin | See example. |
| 03:41.37 | p3nguin | ~devicenames |
| 03:41.37 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 03:43.41 | p3nguin | As a real-life example, my name is Rob. ROB on the keypad is 762. My extension is 762. My SIP phone has a MAC address of 000011112222. To create my extension which dials my phone and rings for 5 rings, use exten => 762,1,Dial(SIP/000011112222,30) |
| 03:44.14 | *** join/#asterisk evharten (~evharten@i.have.xs2us.net) |
| 03:44.19 | p3nguin | 000011112222 is the name that goes in sip.conf for the phone. e.g. [000011112222] |
| 03:44.29 | evharten | Mornin all |
| 03:44.54 | p3nguin | 762 is the extension. exten => 762,1,Dial(SIP/000011112222,30) goes in extensions.conf. |
| 03:46.05 | arfon | So in extensions.conf you put:[rob] username=762? |
| 03:46.22 | p3nguin | This abstract model allows a person to use any phone he wants with minor configuration changes. I can switch to another office where there is a new phone by the name of aaaabbbbcccc. I don't need to reconfigure the phone, just change extension 762 to dial the other device. |
| 03:46.42 | p3nguin | Nah, that's not even close to what I said. |
| 03:46.59 | evharten | arfon: no that goes in sip.conf |
| 03:47.15 | arfon | Blech... |
| 03:47.18 | arfon | Okay |
| 03:47.19 | evharten | sip.conf you define the users in ;) |
| 03:47.22 | p3nguin | sip devices |
| 03:47.23 | p3nguin | not users |
| 03:47.28 | evharten | yeah ok true |
| 03:47.30 | p3nguin | users are people./ |
| 03:47.34 | evharten | wrong choice of words :) |
| 03:47.34 | p3nguin | We don't configure people, unfortunately. |
| 03:47.40 | arfon | What if I don't want users |
| 03:47.44 | evharten | would be lovely if we could tho p3nguin ;) |
| 03:47.46 | arfon | I just want extensions |
| 03:47.50 | p3nguin | Then don't give any people access to anything. |
| 03:47.57 | p3nguin | Extensions are your dialing rules. |
| 03:48.09 | evharten | yep only configure their sip device, but keep the context empty where they are in |
| 03:48.10 | p3nguin | Extensions are not phones. Phones are not extensions. |
| 03:48.27 | arfon | I have a SIP trunk coming in... I want calls to ring at 3 extensions... |
| 03:48.30 | p3nguin | Phones (sip devices) are configured in sip.conf. |
| 03:48.49 | p3nguin | You don't "ring three extensions," because extensions don't have ringers. |
| 03:48.55 | p3nguin | You ring three DEVICES. |
| 03:49.08 | p3nguin | And SIP doesn't trunk. |
| 03:49.14 | arfon | Sorry, I'm used to POTS |
| 03:49.15 | p3nguin | Whoever told you that it does is wrong. |
| 03:49.36 | p3nguin | Step one: configure three devices in sip.conf for your three phones to use. |
| 03:49.47 | arfon | Done for 2 so far |
| 03:49.52 | p3nguin | Name them something unique and significant to the phones if possible. |
| 03:49.56 | arfon | two are up and connected |
| 03:50.00 | arfon | 201 and 202 |
| 03:50.03 | p3nguin | NOT 201, 202, etc. |
| 03:50.09 | p3nguin | That isn't unique to the phone. |
| 03:50.24 | p3nguin | That's the extension number you'll use to reach those phones later. |
| 03:50.43 | p3nguin | Don't confuse yourself any more than necessary by using device names the same as extensions. |
| 03:50.46 | arfon | okay ekiga and sipdroid |
| 03:51.07 | p3nguin | I always us the MAC address of the primary interface. |
| 03:51.23 | p3nguin | But other unique and significant values are also good choices. |
| 03:52.02 | p3nguin | If you feel like ekiga and sipdroid is as unique and device-significant as you can get, I'll work with that. At least it isn't the same as the extension numbers. |
| 03:52.41 | p3nguin | When you're done with step one (configuring devices in sip.conf), let me know. |
| 03:53.36 | arfon | Where do I put ekiga and sipdroid... username or [] |
| 03:53.50 | p3nguin | You won't be using the username= field at all for your phones. |
| 03:53.58 | p3nguin | Phone names go in square brackets. |
| 03:54.13 | p3nguin | Don't use username= at all. |
| 03:54.18 | arfon | so [ekiga] and [sipdroid] in sip.conf |
| 03:54.25 | p3nguin | Okay, good. |
| 03:54.37 | p3nguin | Now you've also defined some good settings under each? |
| 03:54.48 | evharten | username= is replaced by authuser= if i recall correctly |
| 03:54.52 | p3nguin | secret, nat, directmedia, host, etc... |
| 03:55.08 | p3nguin | No, username is what username to send to a device. |
| 03:55.13 | p3nguin | We're not doing that. |
| 03:55.59 | p3nguin | Asterisk isn't going to authenticate to the phone, so you don't use username= in a phone's peer entry. |
| 03:56.30 | arfon | How do you connect the phone to asterisk without user/secret? |
| 03:56.36 | p3nguin | A phone's username is found in the square brackets. |
| 03:56.44 | arfon | ah |
| 03:56.55 | p3nguin | The password is the value in secret= |
| 03:57.10 | arfon | username is now gone |
| 03:57.21 | p3nguin | If the phone is going to be registering, you'll have to set host=dynamic |
| 03:57.31 | arfon | got it |
| 03:57.33 | p3nguin | Most phone register. |
| 03:57.36 | evharten | or define the ip static ;) |
| 03:57.39 | p3nguin | s/phone/phones/ |
| 03:57.42 | arfon | no static ip |
| 03:57.49 | p3nguin | Nope, most phones register. |
| 03:57.59 | evharten | arfon: im using static ip's for phones here internally |
| 03:58.02 | p3nguin | You must set host=dynamic for anything that registers to asterisk. |
| 03:58.15 | evharten | arfon: also define that ip in host and allow rules |
| 03:58.22 | p3nguin | Static network addresses has nothing to do with host=dynamic. |
| 03:58.28 | arfon | evharten: This is MAINLY to get calls on my mobile via sipdroid |
| 03:58.39 | p3nguin | You can use static IP addresses on phones, but they still almost always register. |
| 03:58.46 | arfon | I've got softphones running on my Slackware box to test |
| 03:58.51 | ChannelZ | Maybe you should just get a Google Voice number. |
| 03:59.04 | arfon | Google wouldn't port my landline |
| 03:59.11 | p3nguin | Damn them! |
| 03:59.15 | evharten | :) |
| 03:59.15 | p3nguin | Damn them to hell! |
| 03:59.24 | arfon | I did |
| 03:59.25 | p3nguin | I figured they'd port anything. |
| 03:59.46 | arfon | Nope... you have to port to a mobile THEN port to google |
| 03:59.53 | p3nguin | Icky. |
| 04:00.12 | arfon | Yeah, I CANT lose this number but I gotta get AWAY from stinkrizon |
| 04:00.23 | p3nguin | Okay, so how's the config of the phones going? |
| 04:00.26 | ChannelZ | Think of it as an opportunity to make a clean break from all those collection agencies |
| 04:00.34 | arfon | p3nguin: waiting on the next step |
| 04:00.53 | p3nguin | Feel free to also configure the phones themselves. |
| 04:01.00 | arfon | duh |
| 04:01.08 | arfon | I could be doing that couldn't I |
| 04:01.08 | p3nguin | As in, you did it? |
| 04:01.12 | p3nguin | Oh. |
| 04:01.13 | arfon | not yet |
| 04:01.39 | p3nguin | Remember, the username is in square brackets in sip.conf and the password is the secret. |
| 04:02.08 | p3nguin | If the phone asks for an auth name, that is probably also the username (which is... well, you know where it is). |
| 04:02.39 | arfon | Neither one is now registering |
| 04:02.41 | arfon | :( |
| 04:02.49 | ChannelZ | Progress! |
| 04:02.51 | arfon | No matching peer found |
| 04:03.10 | p3nguin | Check the names (user name and auth name) and try again. |
| 04:03.53 | arfon | in ekiga I have NAME, USER, & AUTHENITICATED USER |
| 04:04.04 | p3nguin | Name is arbitrary. |
| 04:04.18 | p3nguin | user is [phone_name_here] |
| 04:04.19 | arfon | I set them all to ekiga |
| 04:04.31 | p3nguin | auth user should be the same as user. |
| 04:04.42 | arfon | wait |
| 04:04.56 | p3nguin | Tom, ekiga, ekiga |
| 04:05.02 | p3nguin | That's how I'd do it. |
| 04:05.15 | ChannelZ | Who's on first? |
| 04:05.44 | p3nguin | Uh, what? |
| 04:05.46 | p3nguin | :) |
| 04:05.53 | arfon | No matching peer found |
| 04:06.08 | p3nguin | Close the phone and start it again. |
| 04:06.19 | arfon | Can I spam the channel with the SIP.conf section? |
| 04:06.25 | p3nguin | No. |
| 04:06.31 | p3nguin | And don't flood it, either. |
| 04:06.32 | p3nguin | ~pb |
| 04:06.32 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 04:06.37 | p3nguin | Pastebin your confs. |
| 04:08.42 | ChannelZ | Do not eat the paste. |
| 04:08.51 | arfon | http://paste.lisp.org/display/133925 |
| 04:08.55 | arfon | Paste taste good |
| 04:09.48 | *** join/#asterisk timahvo1 (~rogue@41.90.45.221) |
| 04:09.59 | p3nguin | Those configs look okay. |
| 04:10.18 | p3nguin | I would have changed that default context, but it isn't breaking your phones. |
| 04:10.54 | p3nguin | Restart the phone after saving the new user/authuser/password. |
| 04:11.02 | p3nguin | Oh! Did you sip reload? |
| 04:11.04 | ChannelZ | sip reload |
| 04:11.19 | p3nguin | You almost got me there. |
| 04:11.33 | arfon | DAMNIT |
| 04:11.40 | arfon | i I did dialplan reload |
| 04:11.57 | p3nguin | That's after we change extensions.conf. |
| 04:12.12 | p3nguin | So far we haven't created any extensions, only phones. |
| 04:12.23 | arfon | OH look, they register now... :P |
| 04:12.32 | p3nguin | Good deal. |
| 04:12.49 | p3nguin | Okay, now to create those extensions so we can actually call those phones you added. |
| 04:12.57 | p3nguin | extensions.conf <--------- |
| 04:13.04 | arfon | opening extensions.conf |
| 04:13.16 | p3nguin | I noticed that your phones have a context setting of 'internal' |
| 04:13.31 | arfon | Yeah, I don't know what that means |
| 04:13.32 | p3nguin | Do you already have an 'internal' context in extensions.conf? |
| 04:13.34 | *** join/#asterisk ectospasm (~ectospasm@unaffiliated/ectospasm) |
| 04:13.41 | p3nguin | [internal] |
| 04:13.50 | arfon | I'll pastebin the extensions.conf, it's short |
| 04:13.54 | p3nguin | okay |
| 04:14.59 | arfon | http://pastebin.com/cyk1BtX2 |
| 04:15.19 | arfon | That VOIP provider and mobile number are bogus |
| 04:15.32 | p3nguin | You do have the internal context. |
| 04:15.47 | arfon | I don't know whatthat means |
| 04:16.00 | p3nguin | [internal] <----------------- |
| 04:16.04 | arfon | yes |
| 04:16.05 | p3nguin | the 'internal |
| 04:16.09 | p3nguin | ' context |
| 04:16.14 | p3nguin | the 'internal' context |
| 04:16.20 | arfon | no, I have [internal] |
| 04:16.32 | p3nguin | That's where your phones have access. |
| 04:16.44 | arfon | exten => _XXX,1,Dial(SIP/${EXTEN}) |
| 04:16.46 | arfon | that? |
| 04:16.48 | p3nguin | When a call comes from those two phones, the context= value is followed. |
| 04:16.55 | p3nguin | That is a bad extension and should be deleted. |
| 04:17.17 | arfon | What should go there? |
| 04:17.18 | p3nguin | Those phones use a context=internal |
| 04:17.32 | p3nguin | So calls go into the internal context and try to match extensions. |
| 04:17.42 | arfon | should I comment out context=internal? |
| 04:17.48 | p3nguin | No, you need it. |
| 04:17.56 | p3nguin | You have 201 and 202, so let's use those. They need edited to work, though. |
| 04:18.07 | arfon | So [internal] should have nothing after it? |
| 04:18.13 | p3nguin | Stop. |
| 04:18.18 | p3nguin | Pay attention to what I'm saying. |
| 04:18.32 | p3nguin | Your phones will have access to the internal context. |
| 04:18.34 | arfon | listening |
| 04:18.37 | p3nguin | internal context = [internal] |
| 04:19.02 | p3nguin | Whatever comes after that, until the next context in square brackets, is in the internal context. |
| 04:19.24 | p3nguin | Only things in the internal context, and things in context which are included, will be able to be called from those phones. |
| 04:19.33 | p3nguin | (2217.56) <p3nguin> You have 201 and 202, so let's use those. They need edited to work, though. |
| 04:19.40 | p3nguin | 201... |
| 04:19.50 | p3nguin | Which phone do you want to associate with extension 201? |
| 04:20.00 | arfon | ekiga |
| 04:20.11 | p3nguin | This is where we make the relationship between a phone and an extension. |
| 04:20.23 | p3nguin | Look at extension 201. exten => 201.... |
| 04:20.42 | p3nguin | The phone is SIP/ekiga |
| 04:20.51 | dfgas-cr48 | yawn |
| 04:21.05 | p3nguin | Make extension 201 call SIP/ekiga with the Dial command. |
| 04:21.09 | dfgas-cr48 | p3nguin, it is fixed :D dijib got it :D |
| 04:21.11 | arfon | exten => 201,1,Dial(SIP/ekiga,20) |
| 04:21.20 | arfon | ? |
| 04:21.32 | p3nguin | Good. That associates extension 201 with phone named ekiga. |
| 04:21.40 | p3nguin | And it has a ring time of just 20 seconds. |
| 04:22.04 | p3nguin | For extension 202, change that to the other phone name. |
| 04:22.17 | p3nguin | 202 needs to dial SIP/sipdroid |
| 04:22.29 | arfon | one |
| 04:22.32 | arfon | done |
| 04:22.43 | p3nguin | Save. dialplan reload |
| 04:23.05 | arfon | done |
| 04:23.27 | *** join/#asterisk cyborg-one (~cyborg-on@79-140-5-100.broadband.tenet.odessa.ua) |
| 04:23.47 | p3nguin | Now dialing 201 from the sipdroid phone will cause the ekiga phone to receive a call (if ekiga phone is registered). |
| 04:24.42 | arfon | Hmmm, didn't like that |
| 04:24.50 | p3nguin | What happened? |
| 04:25.07 | arfon | I f'd up... deleted a "[" |
| 04:25.13 | p3nguin | oops |
| 04:25.30 | arfon | trying again |
| 04:26.14 | dfgas-cr48 | p3nguin, by chance do you think i could get a copy of you asterisk sound files? or do you know where you got them from so i can get some of them? |
| 04:26.24 | arfon | didn't like it |
| 04:26.28 | arfon | WARNING[32711]: chan_sip.c:3351 __sip_xmit: sip_xmit of 0x8d81be8 (len 869) to 0.0.0.201:5060 returned -1: Invalid argument |
| 04:26.32 | p3nguin | You'd have to google for allison sound files or something. |
| 04:26.34 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 04:26.55 | dfgas-cr48 | alrighty |
| 04:26.58 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:27.42 | p3nguin | I've seen that warning before and I know what it means, but I can't remember what causes it. |
| 04:27.45 | arfon | OK. |
| 04:27.48 | arfon | I'm a moron |
| 04:28.00 | arfon | Another typo in extensions.conf |
| 04:28.05 | p3nguin | Dial(SIP/201) again? |
| 04:28.09 | arfon | 202 ---> 201 works |
| 04:28.20 | p3nguin | 202 isn't a phone! |
| 04:28.25 | arfon | 201 ---> 202 goes right to standby |
| 04:28.32 | p3nguin | 201 isn't a phone! |
| 04:28.43 | p3nguin | Phones call to extensions. |
| 04:29.04 | p3nguin | ekiga -> 202, sipdroid -> 201 |
| 04:29.19 | arfon | Diamy dial pade doesn't have letters |
| 04:29.23 | arfon | my dial pad |
| 04:29.32 | p3nguin | That's why we have extension NUMBERS. |
| 04:29.35 | p3nguin | But phones are not extensions. |
| 04:29.40 | p3nguin | Phones are not extension numbers. |
| 04:29.59 | arfon | ekiga ---> 201 goes to standby |
| 04:29.59 | ChannelZ | An extension is an arbitrary thing you dial. That extension can then call a DEVICE via extensions.conf |
| 04:30.08 | arfon | sipdroid to 202 works |
| 04:30.13 | arfon | Other way around |
| 04:30.31 | arfon | sip droid calls 201 fine |
| 04:30.35 | p3nguin | I only insist that you don't name your devices the same as the extension number used to reach it to avoid MORE confusion. |
| 04:30.45 | arfon | ekiga calls 202 and goes to standby |
| 04:31.04 | p3nguin | I'm not sure what going to standby means. |
| 04:31.47 | ChannelZ | According to your earlier sip.conf, you should be doing Dial(SIP/ekiga) and Dial(SIP/sipdroid) (I think it was.) What extension numbers you hook that up to in extensions.conf is up to you, but stop using them interchangably because we have no idea what you're doing. |
| 04:32.12 | p3nguin | I tried to get that part straightened out. |
| 04:32.19 | ChannelZ | exten => 201,1,Dial(SIP/ekiga) |
| 04:32.31 | ChannelZ | exten => 202,1,Dial(SIP/sipdroid) |
| 04:32.41 | p3nguin | He should have that part. |
| 04:32.45 | arfon | sorry |
| 04:32.52 | arfon | I have: |
| 04:32.54 | p3nguin | I haven't seen the new extensions.conf, but I believe him when he said he did it. |
| 04:33.03 | arfon | exten => 201,1,Dial(SIP/ekiga,20) |
| 04:33.05 | arfon | and |
| 04:33.15 | arfon | exten => 202,1,Dial(SIP/sipdroid,20) |
| 04:33.24 | ChannelZ | and did we reload this time? |
| 04:33.27 | arfon | yes |
| 04:33.32 | arfon | dialplan |
| 04:33.33 | p3nguin | Looks good. Very basic, but should work without problems. |
| 04:33.41 | ChannelZ | so show what the console is actually saying |
| 04:33.44 | arfon | I THINK it's an ekiga quirk |
| 04:33.46 | ChannelZ | core set verbose 3 |
| 04:33.46 | p3nguin | core set verbose 3 |
| 04:33.52 | arfon | vvvvr |
| 04:33.53 | ChannelZ | then make a test call |
| 04:34.09 | arfon | sipdroid to 201 works great |
| 04:34.44 | arfon | ekiga to 202 doesn't show a connection on asterisk and ekiga goes "calling... standby" |
| 04:34.51 | p3nguin | When you increase the verbose and make a call, it should spew useful things. Pastebin everything that shows up. |
| 04:35.04 | arfon | How man v's ? |
| 04:35.07 | arfon | many |
| 04:35.10 | p3nguin | core set verbose 3 |
| 04:35.21 | ChannelZ | I've never used Ekiga. Does it have multiple accounts? Do you have the right one selected? |
| 04:35.38 | p3nguin | If you're connecting to the CLI each time, you can use -rvvv |
| 04:36.06 | arfon | when I call from ekiga NOTHING appears in the cLI |
| 04:36.15 | arfon | I've got ekiga set wrong |
| 04:36.28 | p3nguin | four is also okay, but adds the not-so-useful dnsmgr crap that we don't need to see. |
| 04:38.51 | arfon | ekiga wants an outbound proxy.... |
| 04:38.55 | arfon | NOW I get |
| 04:39.01 | p3nguin | Use asterisk's address or hostname. |
| 04:39.04 | arfon | [Nov 26 22:39:58] NOTICE[32711]: chan_sip.c:22088 handle_request_invite: Call from 'ekiga' (99.184.92.160:5060) to extension '202' rejected because extension not found in context 'internal'. |
| 04:39.14 | p3nguin | This is good. |
| 04:39.32 | ChannelZ | so.. lie-teller! |
| 04:39.46 | *** join/#asterisk radic (~radic@dslb-094-216-243-029.pools.arcor-ip.net) |
| 04:39.48 | p3nguin | Now you need to make sure you have extension 202 in there correctly. |
| 04:40.08 | p3nguin | Feel free to pastebin your changes at any time. |
| 04:40.09 | arfon | exten => 202,1,Dial(SIP/sipdroid,20) |
| 04:40.38 | *** join/#asterisk kingpin8080 (~Javier_I_@c-50-138-105-117.hsd1.fl.comcast.net) |
| 04:40.41 | p3nguin | Is it under [interna] ? |
| 04:40.50 | p3nguin | [internal] ? |
| 04:41.22 | p3nguin | heh... |
| 04:41.24 | p3nguin | [interna] |
| 04:41.26 | p3nguin | include => internal |
| 04:41.33 | p3nguin | :/ |
| 04:41.34 | p3nguin | Don't do it! |
| 04:41.44 | ChannelZ | now you're just baiting him. |
| 04:42.50 | kingpin8080 | Can anyone tell me why I am getting this error when I try to call from sip extension to sip extension: pbx.c:4475 pbx_extension_helper: No such label 'stdexten' in extension '6000' in context 'DLPN_DialPlan1' |
| 04:42.51 | p3nguin | I'm hoping to see the current extensions.conf soon. |
| 04:43.12 | kingpin8080 | I am using asterisk 11.0.0 and using the GUI to configure it. |
| 04:43.18 | arfon | http://pastebin.com/9Sygu50t |
| 04:43.20 | p3nguin | barfs a little |
| 04:43.36 | arfon | I'm slow |
| 04:43.50 | ChannelZ | Iiiiii imaaaagine it's because you don't have a label called stdexten in extension 6000 in your context DLPN_DialPlan1. |
| 04:44.39 | p3nguin | arfon: On the CLI, run this: dialplan show internal |
| 04:44.53 | arfon | http://pastebin.com/ZJZsMsia |
| 04:45.28 | ChannelZ | But "the GUI" is not actually a thing, you're probably referring to FreePBX or something else in which case who knows what it has created. |
| 04:45.38 | p3nguin | asterisk gui |
| 04:45.47 | arfon | http://pastebin.com/CGprsRUE |
| 04:45.49 | p3nguin | I garrrrennnnnnteeeeeeeeeee. |
| 04:46.07 | kingpin8080 | I get this: |
| 04:46.07 | kingpin8080 | There is no existence of 'internal' context |
| 04:46.07 | kingpin8080 | Command 'dialplan show internal' failed. |
| 04:46.08 | p3nguin | 202 looks good to me. |
| 04:46.24 | *** join/#asterisk fleixius (~fleixius@unaffiliated/fleixius) |
| 04:47.15 | ChannelZ | kingpin8080: that wasn't for you |
| 04:47.23 | arfon | I'm gettin grief from the wife... I need to feed the animals... Can I come back tomorrow and finish this? |
| 04:48.07 | p3nguin | Probably. I'll also be here for a while longer tonight if you want me to help. |
| 04:48.14 | p3nguin | Other people will also be here after I go. |
| 04:48.20 | p3nguin | channelz <------- |
| 04:49.49 | ChannelZ | kingpin8080: A label is a way to mark an extension priority with a name instead of a number.. something in your dialplan is trying to call something by that name (presumably) but it doesn't exist. But you're using a GUI (again, FreePBX I assume) which creates a crazy dialplan, so it's a big question mark as to what is supposed to happen. |
| 04:50.40 | ChannelZ | We can solve the error by putting a label somewhere, but I have NO idea where it would even go and function properly. |
| 04:51.34 | *** join/#asterisk ghost75 (~trechber@dslb-178-010-046-035.pools.arcor-ip.net) |
| 04:53.43 | arfon | I would really like to continue (this is the most progress I've made) but the wife aggro is intense |
| 04:54.34 | p3nguin | Once I can teach you the difference between a phone and an extension, the rest will be easy for you. |
| 04:54.34 | ChannelZ | <-- lives alone |
| 04:57.49 | arfon | Thanks for the help |
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| 06:55.24 | *** join/#asterisk Echo777 (~echo@71-82-226-158.dhcp.stpt.wi.charter.com) |
| 06:55.32 | Echo777 | anyone wanna help me with something? |
| 06:56.20 | Echo777 | im using asterisk with google voice and when i try to recieve a call i get |
| 06:56.20 | Echo777 | ] WARNING[18716][C-00000001]: pbx.c:6167 __ast_pbx_run: Channel 'Gtalk/+16087185427-ea38' sent to invalid extension but no invalid handler: context,exten,priority=from-google,echo.spires@gmail.com,1 |
| 07:03.30 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 07:05.22 | Echo777 | ok now i have RROR[19035]: chan_motif.c:815 jingle_add_google_candidates_to_transport: Unable to add Google ICE candidates as ICE support not available or no candidates available == Spawn extension (incoming-motif, s, 2) exited non-zero on 'Motif/+16087185427-3754' == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 |
| 07:09.36 | Echo777 | someone, please ha |
| 07:11.39 | drmessano | You need to add icesupport=yes to rtp.conf |
| 07:11.59 | drmessano | Also, you need an S extension in the from-google context |
| 07:12.41 | Echo777 | foxed it before i saw what you said ha, so now how do i make it call out? |
| 07:13.06 | drmessano | Dial Motif/NUMBER |
| 07:13.09 | *** join/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 07:13.19 | bombev | Hi all :) |
| 07:13.19 | Echo777 | does this go in extensions.conf? |
| 07:13.27 | drmessano | yeah |
| 07:14.08 | Echo777 | something to do with this? NOTICE[19252][C-00000002]: chan_sip.c:25108 handle_request_invite: Call from '101' (192.168.1.134:5061) to extension '6087185427' rejected because extension not found in context 'local'. |
| 07:14.12 | Echo777 | oops no not that |
| 07:14.15 | Echo777 | exten => _1XXXXXXXXXX,1,Dial(Motif/google/${EXTEN}@voice.google.com,,r) |
| 07:14.16 | Echo777 | this |
| 07:15.12 | drmessano | Yep |
| 07:15.52 | Echo777 | do i just paste that under my current stuff in extensions.conf or what do i need to do with it, i apologize i am new to this |
| 07:16.06 | drmessano | ~book |
| 07:16.06 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 07:16.16 | drmessano | Read up on extensions.conf |
| 07:16.32 | Echo777 | i will if you just tell me what to do with this last thing ha |
| 07:17.49 | drmessano | The snippet you pasted is not a complete piece of dialplan. You really need to follow that link |
| 07:25.47 | *** join/#asterisk deo (~deo@122.53.72.218) |
| 07:29.13 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 07:30.46 | Echo777 | im very confused |
| 07:34.08 | *** join/#asterisk Tim_Toady (~fuzzy@178.128.20.156.dsl.dyn.forthnet.gr) |
| 07:36.15 | *** join/#asterisk Tim_Toady (~fuzzy@178.128.20.156.dsl.dyn.forthnet.gr) |
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| 07:47.25 | Echo777 | ok yea, idk, |
| 07:54.17 | Echo777 | so instead of using the dial plan [incoming-motif] |
| 07:54.21 | Echo777 | exten => s,1,NoOp() same => n,Wait(1) same => n,Answer() same => n,SendDTMF(1) same => n,Dial(SIP/malcolm,20) |
| 07:54.32 | Echo777 | one can just use exten => s,1,Dial(SIP/malcolm,20,D(:1)) |
| 07:55.03 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 08:04.40 | Echo777 | uhmmm |
| 08:04.42 | Echo777 | nable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist? |
| 08:04.47 | Echo777 | and it does |
| 08:06.44 | Echo777 | ohhhh nvm god im stupid |
| 08:07.15 | wdoekes | nah, you just lack punctuation |
| 08:07.47 | Echo777 | haha no trust me, i was typing -rvvv instead of -cvvv |
| 08:10.03 | Echo777 | so i changed asterisk to work with another google account but it isnt, what may i need to change that i didnt |
| 08:10.22 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 08:10.24 | Echo777 | i changed jabber.conf and xmpp.conf |
| 08:13.48 | Echo777 | uh why is it not woking |
| 08:14.37 | kaldemar | Echo777: using -c instead of -r was a mistake if you already had asterisk running. you were probably just attaching as the wrong user. |
| 08:15.24 | kaldemar | Echo777: modifying jabber.conf and xmpp.conf won't do you any good if you use chan_motif. what you should be modifying is motif.conf. |
| 08:16.02 | Echo777 | but the only thing there is [google] |
| 08:16.02 | Echo777 | context=incoming-motif |
| 08:16.02 | Echo777 | disallow=all |
| 08:16.02 | Echo777 | allow=ulaw |
| 08:16.02 | Echo777 | connection=google |
| 08:16.24 | Echo777 | all i did by changing google accounts is change the number im using so where is that |
| 08:17.12 | kaldemar | https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 08:17.42 | Echo777 | ha no no i had it up and running a second ago but i want to use it for a different google account |
| 08:19.41 | Echo777 | oh, the sip line is unmonitored, how did that happen |
| 08:20.15 | kaldemar | you didn't configure qualify=yes for it. |
| 08:20.24 | Echo777 | in sip.conf? |
| 08:20.37 | kaldemar | yes. all sip settings are in sip.conf. |
| 08:21.41 | Echo777 | thanks |
| 08:22.20 | Echo777 | calls still arent going to asterisk, wtf |
| 08:23.40 | Echo777 | whats missing here kaldemar? |
| 08:27.11 | kaldemar | < kaldemar> Echo777: using -c instead of -r was a mistake if you already had asterisk running. you were probably just attaching as the wrong user. |
| 08:27.54 | kaldemar | if you did not undo that mistake, you now have two instances of asterisk running, one of which cannot listen on incoming connections. |
| 08:30.40 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 08:31.25 | Echo777 | how do i find out, as far as i know there is only one |
| 08:32.03 | Echo777 | nvm i killed it |
| 08:32.57 | kaldemar | don't start asterisk with -c, it will start a new instance. you should attach to a running one with -r as the right user. |
| 08:33.59 | Echo777 | so whats the correct command to use it verbosely -rvvv? |
| 08:35.11 | kaldemar | Echo777: that command attaches to a running instance with verbosity level 3. |
| 08:36.43 | Echo777 | so then why am i getting == Parsing '/etc/asterisk/asterisk.conf': Found |
| 08:36.43 | Echo777 | <PROTECTED> |
| 08:36.43 | Echo777 | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 08:37.42 | Echo777 | oh damn nvm |
| 08:37.50 | kaldemar | two choices. 1. asterisk is not running 2. you're attaching as the wrong user. |
| 08:37.59 | Echo777 | ugh i need to troublshoot before i ask stupid things, it wasnt running, i got it now |
| 08:39.21 | Echo777 | back to square one though, no ring on the sip phone |
| 08:40.16 | ChannelZ | ring ring ring ring ring ring ring ring bananaphone! |
| 08:42.29 | Echo777 | i was using a google voice account with the number blah blah blah one and then i switched google accounts to one with blah blah blah two and edited xmpp.conf and jabber.conf to update the username and passwords... so whats the problem |
| 08:43.23 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:191b:5d41:775e:15ce) |
| 08:47.23 | Echo777 | ? |
| 08:47.29 | kaldemar | Echo777: with chan_motif, you need to configure motif.conf and xmpp.conf. what happens to the call when it comes to asterisk is configured in extensions.conf. |
| 08:47.38 | kaldemar | show what you see in CLI when making a call. |
| 08:48.04 | Echo777 | nothing at all, its only setup to recieve calls atm |
| 08:48.32 | kaldemar | if you don't see anything, then you're not getting a call. |
| 08:48.38 | Echo777 | right.. |
| 08:49.11 | Echo777 | heres motif.conf |
| 08:49.16 | Echo777 | [google] |
| 08:49.16 | Echo777 | context=incoming-motif |
| 08:49.16 | Echo777 | disallow=all |
| 08:49.16 | Echo777 | allow=ulaw |
| 08:49.16 | Echo777 | connection=google |
| 08:49.22 | ChannelZ | ~pb |
| 08:49.22 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 08:49.37 | Echo777 | fine hang on |
| 08:49.52 | ChannelZ | The toothpaste is out of the tube already |
| 08:50.19 | ChannelZ | Now do you have a [google] entry in xmpp.conf? |
| 08:51.38 | Echo777 | heres all the confs http://pastebin.com/VvXgG6PF they arent labeled so youll have to think ha |
| 08:52.18 | *** join/#asterisk wonderworld (~w@dsdf-4db5dd1e.pool.mediaWays.net) |
| 08:52.21 | ChannelZ | Well I hope either that's not your real gmail address or you put in a new password |
| 08:52.25 | ChannelZ | Otherwise I'd go change that. |
| 08:53.30 | Echo777 | its not |
| 08:54.03 | ChannelZ | You're missing transport=google in motif.conf |
| 08:55.28 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.146) |
| 08:55.46 | Echo777 | just goes at the top? |
| 08:56.10 | ChannelZ | under [google] with the rest |
| 08:57.56 | ChannelZ | actually you might want transport=google-v1 if you're doing this specifically for Google Voice. I didn't go back that far and read what you were doing |
| 08:58.03 | Echo777 | still nothing in the CLI |
| 08:59.27 | ChannelZ | Is everything loaded? does 'xmpp show connections' show your client logged in? |
| 08:59.51 | Echo777 | yes and so does jabber |
| 08:59.59 | ChannelZ | wait... jabber? |
| 09:00.03 | Echo777 | yea? |
| 09:00.10 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 09:00.16 | ChannelZ | What version of Asterisk are you using? |
| 09:00.27 | Echo777 | the newest i believe |
| 09:00.42 | Echo777 | 11.1.0-rc1 |
| 09:01.18 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 09:01.45 | ChannelZ | chan_motif and res_xmpp replace gtalk and jabber. |
| 09:02.01 | Echo777 | it was working fine earlier though |
| 09:02.49 | ChannelZ | It's a problem that it's not working now, right? |
| 09:03.11 | Echo777 | ha correct, so do they take the same settings? |
| 09:03.44 | ChannelZ | they have different configs but they do similar things. res_jabber was the XMPP portion and chan_gtalk was the channel driver. |
| 09:04.28 | ChannelZ | If you actually have jabber/gtalk loaded at the same time as xmpp/motif and they are trying to talk to the same account, I have no idea what would happen. |
| 09:04.49 | ChannelZ | I'd expect more fireworks, but no-worky is not a surprise either. |
| 09:05.24 | Echo777 | so what do i need to do here |
| 09:06.48 | ChannelZ | if you indeed have chan_gtalk and res_jabber build for Asterisk 11, noload them in modules.conf and restart Asterisk completely because who knows what is going on |
| 09:10.00 | Echo777 | done |
| 09:10.24 | ChannelZ | so you shouldn't have any jabber console commands anymore |
| 09:10.32 | ChannelZ | but have xmpp ones |
| 09:10.42 | ChannelZ | and 'core show channeltypes' should list Motif |
| 09:11.03 | Echo777 | good to go |
| 09:11.18 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:11.52 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:12.23 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.163) |
| 09:13.03 | ChannelZ | so if you're still not getting anything perhaps your Google Voice number is being forwarded elsewhere (assuming you're using GV) |
| 09:14.34 | Echo777 | but its not being forwarded elsewhere |
| 09:14.46 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 09:14.59 | ChannelZ | What is your actual gmail address? I can try gtalk and see what it's saying |
| 09:16.09 | Echo777 | apparently i needed to forward calls to google chat |
| 09:16.28 | ChannelZ | (Huh, I swore I said that.) |
| 09:16.42 | Echo777 | i knew that before, brain failure |
| 09:16.59 | ChannelZ | Like, less than a minute ago. |
| 09:17.15 | Echo777 | so is it ok to forward it to my cell at the same time using asterisk with the dialplan exten => s,1,Dial(SIP/malcolm,20,D(:1)) |
| 09:17.40 | ChannelZ | Assuming that's your cell phoen |
| 09:17.44 | Echo777 | so both my cell and the sip rings |
| 09:18.09 | ChannelZ | well you'd want Dial(SIP/whatever&SIP/whateverelse) if you want them both |
| 09:18.33 | Echo777 | well on google voice i have foward to google chat & forward to my cell number |
| 09:18.51 | ChannelZ | and note the whole D(:1) doesn't always work |
| 09:18.52 | Echo777 | and the dialplan is exten => s,1,Dial(SIP/mysipextension,20,D(:1)) |
| 09:18.54 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:19.11 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.82.6) |
| 09:19.19 | ChannelZ | well yes then - otherwise I'm not sure why you'd have gv coming into your Asterisk in the first place |
| 09:19.54 | Echo777 | business line, for when im working because i dont get cell reception here |
| 09:23.11 | Echo777 | i know it seems dumb but it works |
| 09:23.13 | *** join/#asterisk angryuser_laptop (~angryuser@2a02-8422-1230-bb00-a963-b0e0-0281-19ef.rev.sfr.net) |
| 09:23.28 | ChannelZ | That isn't what I meant |
| 09:23.48 | Echo777 | what do you mean then |
| 09:23.58 | ChannelZ | What you're doing is perfectly logical. |
| 09:24.09 | Echo777 | oh ha |
| 09:24.42 | Echo777 | so hypothetically you could use it to run a whole business phone system for free though |
| 09:24.45 | Echo777 | ? |
| 09:25.12 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.82.6) |
| 09:25.14 | ChannelZ | I suppose if it's reliable enough for you |
| 09:25.49 | Echo777 | actually as of using it right now, idk if its my internet connection, or what but the voice is unbelievably choppy |
| 09:26.39 | ChannelZ | How are you testing? If you're calling yourself from another phone in the room its echo cancellation could just be getting confused |
| 09:27.08 | Echo777 | thats true, i am |
| 09:27.36 | ghost75 | there is an echo test |
| 09:28.16 | Echo777 | huh? |
| 09:28.33 | ChannelZ | or to rule out echo cancellation, write a little dialplan to play some sounds to judge that quality, and then record for a few seconds and play that back |
| 09:28.51 | Echo777 | yea |
| 09:28.58 | ChannelZ | anyway have fun, I'm off to bed. |
| 09:29.06 | Echo777 | thanks for the help! |
| 09:29.10 | ghost75 | Playback(demo-echotest) |
| 09:29.20 | ghost75 | Playback(demo-echodone) |
| 09:29.20 | ChannelZ | Or rather to put the clothes in the dryer I left in the washer 4 hours ago and then go to bed |
| 09:29.39 | Echo777 | ghost75 im a noob, where do i put that |
| 09:29.47 | ghost75 | in dialplan |
| 09:30.14 | ghost75 | crazy time zones, i woke up 2h ago |
| 09:31.41 | Echo777 | so how do i set it up to dial out now |
| 09:32.23 | ghost75 | dial out wher |
| 09:32.24 | ghost75 | e |
| 09:33.08 | Echo777 | im using asterisk with google voice and my current dialplan is [incoming-motif] |
| 09:33.12 | Echo777 | exten => s,1,Dial(SIP/101,20,D(:1)) |
| 09:34.51 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 09:35.24 | ghost75 | do you have peer settings in sip.conf |
| 09:35.48 | Echo777 | type=peer |
| 09:36.16 | ghost75 | yes but the whole connection to google voice |
| 09:36.27 | Echo777 | what do you mean |
| 09:37.52 | ghost75 | like here: http://agix.com.au/blog/?p=2656 |
| 09:38.15 | ghost75 | the context [VoIPProvider] you need for google |
| 09:38.30 | ghost75 | i didnt know they offer voip accounts |
| 09:39.18 | Echo777 | this is my sip.conf [101] |
| 09:39.18 | Echo777 | type=peer |
| 09:39.18 | Echo777 | secret=1234 |
| 09:39.18 | Echo777 | host=dynamic |
| 09:39.18 | Echo777 | context=local |
| 09:39.20 | Echo777 | qualify=yes |
| 09:40.06 | ghost75 | thats for the phone |
| 09:40.18 | Echo777 | yep |
| 09:40.20 | ghost75 | you need another one |
| 09:40.30 | Echo777 | ok |
| 09:40.52 | ghost75 | for sip.voice.google.com i guess |
| 09:42.15 | *** join/#asterisk x1user (~User@212.36.13.6) |
| 09:43.04 | x1user | Hi, Is it odbc depricated in 11, asterisk is not build res_odbc.so like earlier versions? |
| 09:45.32 | Echo777 | how should that look? |
| 09:46.00 | kaldemar | x1user: no. |
| 09:46.22 | ghost75 | like in the page above but you have to know what settings you need for google |
| 09:47.46 | *** join/#asterisk fritz09 (~Adium@pop1-224.catv.wtnet.de) |
| 09:48.09 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 09:48.15 | Echo777 | its says this is how the outgoing dialplan should look exten => _1XXXXXXXXXX,1,Dial(Motif/google/${EXTEN}@voice.google.com,,r) |
| 09:48.34 | kaldemar | ghost75: SIP is not used when using google. |
| 09:49.15 | ghost75 | what they are using? |
| 09:50.07 | kaldemar | xmpp and jingle. |
| 09:52.34 | *** join/#asterisk vlad_starkov (~vlad_star@wn1nat29.beelinegprs.ru) |
| 09:53.39 | Echo777 | this is the only section that confuses me is how to make outgoing calls |
| 09:54.41 | ghost75 | https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 09:54.50 | Echo777 | im already there ha |
| 09:56.08 | ghost75 | exten => _1XXXXXXXXXX,1,Dial(Motif/google/${EXTEN}@voice.google.com,,r) |
| 09:56.26 | Echo777 | totally just typed that a sec ago |
| 09:57.34 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 09:59.14 | Echo777 | no clue what to do with that |
| 10:00.03 | ghost75 | go to example xmpp config on that page |
| 10:00.35 | Echo777 | ok im there |
| 10:00.43 | ghost75 | do you have this in sip.conf? |
| 10:00.57 | Echo777 | why would i? |
| 10:01.08 | ghost75 | its mandatory |
| 10:01.19 | kaldemar | ghost75: no. |
| 10:01.29 | kaldemar | ghost75: no Dial goes in sip.conf. |
| 10:01.38 | Echo777 | ha thought so |
| 10:01.42 | ghost75 | i am not saying you dial in sip.conf |
| 10:02.17 | Echo777 | why would i have the same thing i have in xmpp.conf in sip.conf |
| 10:02.24 | ghost75 | maybe i confused you now a bit |
| 10:02.26 | kaldemar | ghost75: and when tech is "Motif", the Dial has nothing to do with sip.conf. |
| 10:03.41 | Echo777 | soo... |
| 10:03.47 | ghost75 | but he needs a peer setting to google, that i meant |
| 10:04.29 | ghost75 | that needs to be setup before dial cmd |
| 10:04.29 | Echo777 | so type=client? |
| 10:05.18 | ghost75 | http://pastebin.com/aQ9kPPR3 |
| 10:05.27 | kaldemar | ghost75: no he does not need a sip peer because SIP IS NOT USED. |
| 10:05.48 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 10:05.55 | Echo777 | kaldemar? |
| 10:06.06 | ghost75 | ah ok its a different file then |
| 10:06.33 | kaldemar | the syntax for app Dial is Dial(Technology/Resource[&Technology2/Resource2[&...]][,timeout[,options[,URL]]]) |
| 10:06.45 | ghost75 | xmpp.conf, not sip.conf sorry |
| 10:07.03 | Echo777 | yes ghost75 i do have that in xmpp.conf |
| 10:07.10 | ghost75 | ok |
| 10:07.20 | Echo777 | next? |
| 10:07.34 | kaldemar | the "Technology" part defines which channel driver is used. if it is SIP, then chan_sip (configured in sip.conf) is used, if it is "Motif", then chan_motif (configured in motif.conf) is used. |
| 10:08.04 | kaldemar | Echo777: next is to enable debug and verbosity and see what happens in CLI when you make a call to get a hint of what is wrong. |
| 10:08.38 | Echo777 | [Nov 27 04:08:25] NOTICE[23103][C-00000001]: chan_sip.c:25108 handle_request_invite: Call from '101' (192.168.1.134:5061) to extension '6087185427' rejected because extension not found in context 'local'. |
| 10:09.10 | ghost75 | you you entered the dial cmd |
| 10:09.12 | kaldemar | Echo777: that says the call lands in context "local" and there is no extension that would match 6087185427. |
| 10:09.35 | ghost75 | show us extensions.conf |
| 10:10.29 | Echo777 | [incoming-motif] |
| 10:10.29 | Echo777 | exten => s,1,Dial(SIP/101,20,D(:1)) |
| 10:10.53 | ghost75 | is this all? |
| 10:10.56 | Echo777 | yep |
| 10:10.59 | kaldemar | Echo777: _1XXXXXXXXXX would match any 11-digit number that starts with 1. 6087185427 is 10 digits long and does not start with 1. |
| 10:11.37 | kaldemar | Echo777: the previous NOTICE is not about incoming call on the motif side, but on the SIP side. |
| 10:12.15 | Echo777 | that was an attempt at an outgoing call |
| 10:12.19 | Echo777 | incoming works fine |
| 10:12.37 | ghost75 | you may read also http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 10:13.17 | WIMPy | Plus the things that are not documented. |
| 10:13.19 | kaldemar | Echo777: all calls are incoming from asterisk's point of view. when you use your SIP phone to make a call, asterisk gets an incoming SIP call. |
| 10:13.45 | *** join/#asterisk PbxMan (c335d968@gateway/web/freenode/ip.195.53.217.104) |
| 10:13.47 | PbxMan | hello |
| 10:14.03 | Echo777 | so what do i need to do, its 4AM here my brain is frying |
| 10:14.16 | ghost75 | you need to add dial cmd |
| 10:15.03 | ghost75 | example: exten => _XXXX,1,Dial(Motif/google/${EXTEN}@voice.google.com,,r) |
| 10:15.23 | Echo777 | just this ? exten => _1XXXXXXXXXX,1,Dial(Motif/google/${EXTEN}@voice.google.com,,r) |
| 10:15.33 | ghost75 | no not this |
| 10:15.56 | ghost75 | (11:11:32) kaldemar: Echo777: _1XXXXXXXXXX would match any 11-digit number that starts with 1. 6087185427 is 10 digits long and does not start with 1. |
| 10:16.14 | Echo777 | and? |
| 10:16.46 | ghost75 | wont work if you dial 6087185427 |
| 10:16.57 | Echo777 | well i can dial a 1 before it |
| 10:17.08 | Echo777 | so thats fine, US numbers work that way ha |
| 10:17.41 | Echo777 | but in the sense of where i put it does it just go on the next line down in extensions.conf? |
| 10:18.03 | ghost75 | make a new context called outgoing |
| 10:18.22 | Echo777 | done |
| 10:18.45 | Echo777 | wait, in extentions or in sip.conf |
| 10:18.55 | ghost75 | extensions |
| 10:18.58 | Echo777 | ok yea done |
| 10:19.00 | ghost75 | you dont touch sip.conf |
| 10:19.10 | Echo777 | [outgoing] exten => _1XXXXXXXXXX,1,Dial(Motif/google/${EXTEN}@voice.google.com,,r) |
| 10:19.13 | Echo777 | right? |
| 10:20.09 | ghost75 | yes but i dont think you need _1XXXXXXXXXX |
| 10:20.19 | ghost75 | make it shorter |
| 10:20.36 | Echo777 | no it needs to be like that |
| 10:20.40 | Echo777 | that much i know |
| 10:20.54 | Blue_Ice | I need to map (using ExecIf or other method) an internal nummer to a certain outbound caller id. No problem for the sip users. But I also have a few analog DAHDI ports (fax etc). How do I map those? For the moment I was matching on the CALLERID(num), but the dahdi devices don't pass a number apparently. Other suggestions? |
| 10:20.59 | kaldemar | Echo777: the NOTICE tells you where to put it. |
| 10:21.49 | Echo777 | i cant scroll up that far kaldemar |
| 10:22.03 | kaldemar | then make a new call to see it again. |
| 10:22.40 | ghost75 | dont forget to reload before |
| 10:23.06 | Echo777 | local? |
| 10:23.28 | ghost75 | dialplan reload if extensions.conf was changed |
| 10:23.52 | Echo777 | seems you were incorrect ghost75 |
| 10:24.21 | *** join/#asterisk vlad_starkov (~vlad_star@wn1nat29.beelinegprs.ru) |
| 10:24.48 | Echo777 | [incoming-motif] |
| 10:24.48 | Echo777 | exten => s,1,Dial(SIP/101,20,D(:1)) |
| 10:24.48 | Echo777 | [outgoing] |
| 10:24.48 | Echo777 | exten => _1XXXXXXXXXX,1,Dial(Motif/google/${EXTEN}@voice.google.com,,r) |
| 10:25.44 | ghost75 | log? |
| 10:26.50 | Echo777 | same thing as before |
| 10:27.56 | ghost75 | show |
| 10:28.28 | Echo777 | NOTICE[23668][C-00000000]: chan_sip.c:25108 handle_request_invite: Call from '101' (192.168.1.134:5061) to extension '6087185427' rejected because extension not found in context 'local'. |
| 10:28.50 | bombev | Does anyone know what does it mean SIP/2.0 401 Unauthorized? |
| 10:28.58 | ghost75 | you called wrong number it doesnt start with 1 |
| 10:29.06 | *** join/#asterisk BorjaGVO (d51beb92@gateway/web/freenode/ip.213.27.235.146) |
| 10:29.24 | Echo777 | still same error when i add a one |
| 10:29.26 | kaldemar | bombev: typically it is a way to require authentication. |
| 10:29.36 | Echo777 | kaldemar, some help here? |
| 10:30.01 | kaldemar | Echo777: you have the extension in [outgoing] and you have told asterisk to look for it in [local]. |
| 10:30.23 | bombev | kaldemar is it bad error or it is not so important? |
| 10:30.42 | kaldemar | bombev: it is not an error at all. it is important however. |
| 10:30.45 | Echo777 | do i need to change something in motif? |
| 10:30.59 | kaldemar | Echo777: no. this is the sip side. |
| 10:31.41 | bombev | kaldemar any idea how to fix that ? |
| 10:31.49 | bombev | authentication issue |
| 10:31.51 | Echo777 | so i need to add a whole new section in sip.conf? |
| 10:31.58 | kaldemar | Echo777: you have two choices. either you put the extension under [local] or include [outgoing] in [local] with "include => outgoing" (better solution). |
| 10:32.21 | kaldemar | bombev: use correct credentials. |
| 10:32.26 | BorjaGVO | hi everyone...anyone knows why, if I do "mv /var/log/asterisk/queue_log /var/log/asterisk/queue_log.back" and "touch /var/log/asterisk/queue_log" calls into queues don't get logged? |
| 10:33.06 | bombev | kaldemar credentials which is? |
| 10:33.12 | Echo777 | give an example? |
| 10:33.19 | kaldemar | bombev: username and secret. |
| 10:33.44 | kaldemar | Echo777: you have all the keywords. read up on dialplan basics in the book. |
| 10:33.46 | kaldemar | ~book |
| 10:33.47 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 10:34.24 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 10:35.58 | bombev | kaldemar well my username is: username (nickname) password: WWf3g312dhhsH something like that |
| 10:36.14 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 10:36.22 | bombev | do you think that in the brackets is wrong (nickname) |
| 10:37.20 | ghost75 | Echo777: as already written put include => outgoing under local context in extensions.conf |
| 10:37.56 | kaldemar | bombev: i don't even know when you're getting that, so no idea. |
| 10:38.17 | ghost75 | its going to local because you defined that context on the phone entry |
| 10:38.23 | Echo777 | ghost75 i dont think so, there is no local context in extensions.conf |
| 10:38.41 | ghost75 | whatever you have called it |
| 10:38.45 | Echo777 | context=local is in sip.conf |
| 10:39.02 | kaldemar | Echo777: context=local in sip.conf points to [local] in extensions.conf. |
| 10:39.04 | ghost75 | then put local in extensions.conf also |
| 10:39.17 | BorjaGVO | ? |
| 10:40.02 | Echo777 | the only context in extensions.conf right now is [incoming-motif] as defined in motif.conf and [outgoing] |
| 10:41.20 | bulkorok | hi |
| 10:42.11 | Echo777 | your not saying to do this are you ghost75? |
| 10:42.15 | Echo777 | [local] |
| 10:42.15 | Echo777 | include => outgoing |
| 10:42.15 | Echo777 | [incoming-motif] |
| 10:42.15 | Echo777 | exten => s,1,Dial(SIP/101,20,D(:1)) |
| 10:42.15 | Echo777 | [outgoing] |
| 10:42.18 | Echo777 | exten => _1XXXXXXXXXX,1,Dial(Motif/google/${EXTEN}@voice.google.com,,r) |
| 10:42.32 | ghost75 | what a mess |
| 10:42.42 | kaldemar | Echo777: use pastebin instead of pasting your configs here. |
| 10:42.45 | Echo777 | ok |
| 10:43.03 | Echo777 | ill take that as a no ghost75 |
| 10:43.27 | kaldemar | Echo777: that is what you're supposed to do. |
| 10:43.41 | kaldemar | ghost75: what's so messy about that? |
| 10:44.08 | Echo777 | yea. no mess there |
| 10:45.08 | Echo777 | worked :) thanks |
| 10:45.14 | Echo777 | i even learned from that too |
| 10:46.21 | ghost75 | here in channel is a mess because no pastebin |
| 10:48.40 | Echo777 | now i need to figure out why the voice is so choppy |
| 10:50.28 | ghost75 | could be because of high delay |
| 10:50.41 | Echo777 | delay where? |
| 10:50.43 | con3x | Echo777: What codecs do you allow? |
| 10:51.09 | Echo777 | ill find out |
| 10:51.38 | Echo777 | ha where do i find that |
| 10:52.01 | ghost75 | guess you share voice with other traffic |
| 10:52.17 | Echo777 | con3x where would i find that |
| 10:53.08 | ghost75 | http://www.voip-info.org/wiki/view/Asterisk+codecs |
| 10:53.41 | kaldemar | delay does not cause choppiness, it causes... delay. :P |
| 10:53.53 | Echo777 | ha yea makes sense |
| 10:55.39 | Echo777 | so kaldemar do you think the choppiness is unavoidable or a bad connection? |
| 10:56.30 | ghost75 | when i have high delay to sip peer then i get choppiness |
| 10:56.42 | kaldemar | Echo777: those are not mutually exclusive. |
| 10:57.25 | kaldemar | ghost75: sounds more like jitter, not delay alone. |
| 10:57.53 | ghost75 | when the connection to isp is fully used |
| 10:57.57 | con3x | Changing the codec to one that forces low delay helped us a little, though the sound quality suffered, do you have connections over WiFi? |
| 10:58.07 | Echo777 | yes i do |
| 10:58.11 | Echo777 | and a crappy one |
| 10:58.30 | ghost75 | especially when there are multiple connections established |
| 10:58.56 | Echo777 | http://pastebin.com/Tj9chQ91 |
| 10:58.58 | ghost75 | i have only 384kbit upload |
| 10:59.13 | Echo777 | im running pandora and a ton of other stuff too |
| 10:59.32 | ghost75 | tried QoS but helps only a bit |
| 10:59.39 | kaldemar | ghost75: a link being saturated does not cause delay only. |
| 11:00.10 | ghost75 | but also |
| 11:00.55 | Echo777 | so no way to really help it then eh? |
| 11:01.16 | con3x | Echo777: Force low bitrate? |
| 11:01.24 | Echo777 | how so? |
| 11:01.28 | ghost75 | what connection do you have and what are u using on it |
| 11:02.03 | Echo777 | wifi and i just paused pandora so nothing but IRC right now |
| 11:02.42 | Echo777 | how do i get rid of the video and text codecs, i dont need those |
| 11:02.46 | con3x | Echo777: http://www.voip-info.org/wiki/view/Asterisk+config+codecs.conf |
| 11:03.36 | con3x | Although thats only for speex |
| 11:04.18 | Echo777 | someone wanna give me a call and see how crappy it is? con3x?? |
| 11:05.34 | ghost75 | never heared speex or plc before |
| 11:06.26 | con3x | I would if I could, in work though |
| 11:06.30 | Echo777 | ok |
| 11:06.40 | Echo777 | well someone give it a try see if they can hear me |
| 11:06.46 | Echo777 | <PROTECTED> |
| 11:06.59 | ghost75 | i am from europe, too far |
| 11:07.09 | Echo777 | kaldemar? |
| 11:07.31 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 11:12.27 | *** join/#asterisk vlad_starkov (~vlad_star@wn2nat49.beelinegprs.ru) |
| 11:16.18 | Echo777 | well im off to bed thanks guys |
| 11:26.27 | con3x | I remember I turned off a bunch of codecs but I can't remember where I done it |
| 11:27.16 | ghost75 | turned off? |
| 11:28.04 | *** part/#asterisk deo (~deo@222.127.13.226) |
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| 11:31.31 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:33.43 | *** join/#asterisk echo777 (~echo@71-82-226-158.dhcp.stpt.wi.charter.com) |
| 11:33.52 | echo777 | damnnit now incoming calls dont work |
| 11:36.46 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 11:38.01 | danfromuk | What PCI cards are recommended for use with asterisk? Are openvox PCI cards any good? |
| 11:38.54 | WIMPy | The official answer is obviousely that you should support Digium. |
| 11:39.02 | WIMPy | But what kind of interface? |
| 11:39.55 | danfromuk | 2 Analog lines |
| 11:42.44 | WIMPy | Dou you really want to do that to yourself? |
| 11:43.16 | danfromuk | The client has unlimited calls from their provider so doesnt want to switch to voip |
| 11:43.54 | WIMPy | What about replacing the two lines with a BRI? |
| 11:44.31 | danfromuk | I'll get them to ask their provider |
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| 11:58.46 | ghost75 | that was quick sleep echo777 |
| 11:59.20 | ghost75 | wish i could recover so fast ;) |
| 12:01.43 | echo777 | damnnit now incoming calls dont work |
| 12:01.58 | echo777 | oh well bed again |
| 12:04.53 | *** join/#asterisk youjelly (~bwahahaha@39.47.49.144) |
| 12:05.43 | youjelly | hey guys, wanted to know if there are any commercial server solutions available for asterisk, I want to use a 16 span PRI card on it |
| 12:08.14 | youjelly | Asterisk is just going to bridge 2 channels (minimalistic IVR), for each call, AGI server is going to be isolated |
| 12:09.30 | x1user | What is the name of odbc module on asterisk 11? I am building asterisk from source with odbc but can see a module named res_odbc.so or smth =/ |
| 12:11.02 | kaldemar | x1user: the odbc resource module is called res_odbd.so. |
| 12:11.13 | kaldemar | x1user: which module do you mean? |
| 12:13.00 | x1user | I want to have odbc in the CLI, i am loading sip users from mysql database. |
| 12:15.28 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 12:15.31 | kaldemar | x1user: you need to select the relevant modules in "make menuselect". |
| 12:16.18 | x1user | [root@localhost asterisk-11.0.1]# menuselect/menuselect --enable app_mysql --ena |
| 12:16.18 | x1user | ble cdr_mysql --enable res_config_mysql --enable cdr_adaptive_odbc --enable cdr_ |
| 12:16.18 | x1user | odbc --enable cel_odbc --enable func_odbc - |
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| 12:17.10 | x1user | and this is everything that contains odbc =/ |
| 12:18.07 | kaldemar | there are others. |
| 12:19.20 | x1user | i´ve grepped this from makeopts, seem to be everything? |
| 12:19.51 | kaldemar | is it working? |
| 12:20.14 | kaldemar | run make menuselect and select the modules you need. |
| 12:20.15 | x1user | asterisk works, but i cant build odbc for reason |
| 12:21.07 | x1user | In menuconfig what XXX means? |
| 12:21.26 | kaldemar | it means you don't have the dependencies installed. |
| 12:22.27 | x1user | Any way to find dependencies for a specific module? |
| 12:22.52 | kaldemar | look at the "Depends on:" line. |
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| 12:23.41 | ghost75 | documenteur-extraordinaire -> lol |
| 12:26.13 | x1user | kaldemar: thank you |
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| 13:00.54 | parasitodelsur | good morning all |
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| 13:40.48 | ghost75 | is google voice reliable? saw its cheaper for mobile calls |
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| 13:47.58 | ghost75 | to neighbor country even 4 times cheaper |
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| 14:05.21 | dfgas-cr48 | what should i set my cid for of each extension? i set it as my phone number (DID number) and that breaks *98 voicemail |
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| 14:05.54 | WIMPy | It's your dialplan. |
| 14:07.24 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 14:07.40 | dfgas-cr48 | i figure that much |
| 14:07.56 | WIMPy | I did a little dirty stuff there as well. |
| 14:07.57 | dfgas-cr48 | here is what it is for my calling my voicemail |
| 14:08.42 | dfgas-cr48 | exten => *98,1,NoOp() |
| 14:08.43 | dfgas-cr48 | exten => *98,n,VoicemailMain(${CALLERID(num)}@default) |
| 14:08.43 | dfgas-cr48 | exten => *98,n,Hangup() |
| 14:08.47 | WIMPy | I wanted to use the phone numbes as mailbox names, but there's a length limit preventing me from doing so. |
| 14:08.51 | WIMPy | (or was?) |
| 14:10.14 | dfgas-cr48 | well the callerid was the extension number but people tell me when i call them my cid is 11 |
| 14:10.44 | WIMPy | I only use real numbers. |
| 14:10.44 | dfgas-cr48 | so i was like, darn it, alright i will try and fix, lol |
| 14:11.22 | jmetro | shouldnt your caller id be separate...and not used in the dialplan for voicemail? |
| 14:11.44 | WIMPy | You don;t always have the chance to use the dialplan to change it when placing calls. |
| 14:12.03 | WIMPy | You can do whatever you like. |
| 14:12.44 | WIMPy | Mailbox names aren't restricted to numerical. |
| 14:13.33 | slav3_kitten | any gui or console editors that have a syntax hilighting plugin for asterisk configs? |
| 14:13.54 | slav3_kitten | i think vim has one |
| 14:14.09 | bulkorok | slav3_kitten: I tried to do one with notepad++ |
| 14:14.36 | slav3_kitten | isn't notepad++ win32 only? |
| 14:14.48 | bulkorok | could be... |
| 14:15.00 | bulkorok | but gui ;-) |
| 14:15.52 | slav3_kitten | totally does not count :| |
| 14:16.17 | dfgas-cr48 | hmmm |
| 14:16.19 | bulkorok | :) |
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| 14:16.41 | jmetro | i believe Vim does. |
| 14:16.44 | jmetro | as well as note++ |
| 14:17.29 | slav3_kitten | as soon as my ssh session starts responding again... |
| 14:18.01 | slav3_kitten | or i turn off caps lock and screen goes back to doing what i ask |
| 14:19.15 | jmetro | if you have vim installed it might have the asterisk language in there by default actually - otherwise you can add one to it. |
| 14:19.19 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 14:19.33 | jmetro | i feel like my recent asterisk test box had the colors built in when I installed vim. |
| 14:21.07 | slav3_kitten | yea vim has it :D |
| 14:23.28 | slav3_kitten | 243 lines of extensions.conf only 136 actually critical |
| 14:25.37 | p3nguin | slav3_kitten: vim is what I use exclusively. |
| 14:26.35 | slav3_kitten | p3nguin, that's what i tend to use, but geany on my laptop offers some nice features like expanding and shrinking sections i'm finished with |
| 14:26.46 | p3nguin | :syntax on |
| 14:27.48 | ghost75 | is it possible to change my callerid on outgoing calls? |
| 14:28.42 | slav3_kitten | p3nguin, yea i got that. i'm trying to figure out how to have it come on when i load the config files automatically |
| 14:28.49 | slav3_kitten | ghost75, yes absolutely |
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| 14:29.02 | slav3_kitten | i know you can do it, just can't remember how |
| 14:30.06 | slav3_kitten | .vimrc! |
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| 14:37.35 | p3nguin | Yep. |
| 14:38.09 | *** join/#asterisk italorossi (~italoross@189.124.196.68) |
| 14:38.22 | p3nguin | ghost75: It depends. If your provider allows it, you can change it. If your provider does not allow it, you can try to change it, but one of two things will happen: the call will fail, or nothing will be changed. |
| 14:39.00 | bulkorok | ghost75: check 'CLIP no screening' |
| 14:39.08 | ghost75 | does google voice allow it? |
| 14:39.12 | p3nguin | no |
| 14:39.19 | ghost75 | :< |
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| 14:48.59 | parasitodelsur | p3nguin: you around? |
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| 14:49.37 | p3nguin | PLEASE LEARN HOW TO USE ASYNCHRONOUS COMMUNICATIONS. |
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| 14:49.58 | parasitodelsur | p3nguin: ? |
| 14:50.01 | p3nguin | PLEASE LEARN HOW TO USE ASYNCHRONOUS COMMUNICATIONS. |
| 14:50.23 | parasitodelsur | and again ? |
| 14:50.33 | parasitodelsur | what am I missing here... |
| 14:50.45 | parasitodelsur | just wanted to let you know that I got the part |
| 14:50.48 | kaldemar | ~ask |
| 14:50.48 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:51.00 | parasitodelsur | I dont have a question |
| 14:51.02 | p3nguin | You could have simply told me what you wanted me to know. |
| 14:51.29 | parasitodelsur | I dont like to bother. |
| 14:51.37 | parasitodelsur | so I rather ask for aviability |
| 14:51.40 | p3nguin | So instead you bothered. |
| 14:52.38 | parasitodelsur | shit really? |
| 14:52.44 | parasitodelsur | that bad. |
| 14:52.51 | [TK]D-Fender | parasitodelsur, He's here, the userlist told you that when you came in. You should probably just ask him whatever you wanted to ask at this point... |
| 14:54.21 | parasitodelsur | is it really that bad to make sure somebody is available when you want to rellay a message directly to them? |
| 14:54.25 | wdoekes | YES |
| 14:54.42 | parasitodelsur | bull shit. |
| 14:56.34 | parasitodelsur | well shit. Thats what I get from trying to be polite. |
| 14:57.10 | parasitodelsur | p3nguin: can you msg me your info. I still have it but not here with me at work. |
| 14:58.06 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:59.15 | parasitodelsur | ok so this goes back to why I ask if somebody is available. |
| 14:59.20 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:59.21 | parasitodelsur | no response. |
| 14:59.48 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 15:00.40 | slav3_kitten | that's odd as hell. with the latest asterisk.vim type=friend is fine but if you put anything after such as a comment it shows it as an error... i have no idea how to fix that |
| 15:01.18 | wdoekes | no, we *don't* get it. because you could've asked penguin *that* question the first time, without asking if he's around first |
| 15:02.05 | parasitodelsur | wdoekes: lay off.... |
| 15:02.17 | jmetro | slav3_kitten : try updating the asterisk language with the current one shown online - i think the vim one is out of date maybe |
| 15:02.50 | slav3_kitten | jmetro, the one included in the latest svn of asterisk 11 is out of date? |
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| 15:13.20 | slav3_kitten | if i have allow guests. i should have type=user in general right? |
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| 15:16.25 | p3nguin | Why would you do that? |
| 15:16.39 | *** join/#asterisk acedia (~rage@unaffiliated/ffs) |
| 15:16.48 | slav3_kitten | good point. it's not like i can call them back |
| 15:16.59 | slav3_kitten | i was over thinking it p3nguin |
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| 15:17.18 | p3nguin | You can't call them with type=user anyway. |
| 15:17.29 | *** join/#asterisk blee (~blee@70.118.107.77) |
| 15:17.34 | slav3_kitten | right, i wanted to prevent calling to them though |
| 15:17.46 | p3nguin | user is for a device calling inbound to asterisk only, and it matches the peer on IP address. |
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| 15:22.45 | [TK]D-Fender | slav3_kitten, You would not have defined entries for them to be able to call them back using so far.... |
| 15:22.46 | dfgas-cr48 | p3nguin, ok, when i call people if shows my extension number as the phone number, now when i goto dial *98 for voicemail it doesn't read the caller id right anymore because all my extensions have been set to the phone number, what do you suggest? |
| 15:23.38 | p3nguin | Fix it. |
| 15:23.46 | slav3_kitten | [TK]D-Fender, right. i had just incorrectly over thought something |
| 15:24.01 | [TK]D-Fender | dfgas, Set another variable to use. Or check the PEERNAME or something else unique |
| 15:24.03 | p3nguin | dfgas-cr48: In your sip.conf, the callerid value should be your INTERNAL CALLERID NUMBER. |
| 15:24.14 | dfgas-cr48 | well i could set it back to extension numbers but that would break callerid but fix voicemail |
| 15:24.18 | p3nguin | dfgas-cr48: callerid=Tony <5> |
| 15:24.19 | [TK]D-Fender | slav3_kitten, If you did what to be able to call then, then "autocreatepeer=yes" |
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| 15:24.34 | slav3_kitten | *nods* |
| 15:24.53 | dfgas-cr48 | k, then how do i set my callerid number for when calling out? |
| 15:24.54 | p3nguin | dfgas-cr48: Will every phone have a different caller ID number? |
| 15:25.08 | dfgas-cr48 | no, they all use the same phone number |
| 15:25.14 | p3nguin | dfgas-cr48: You have two ways to do it, then. |
| 15:25.33 | parasitodelsur | p3nguin: I would like to send out the headset today. can you msg me please. Thanks! |
| 15:25.39 | p3nguin | dfgas-cr48: One, Set(CALLERID(num)=yournumber) before the Dial() on the outbound call... |
| 15:25.53 | dfgas-cr48 | ahh |
| 15:25.57 | dfgas-cr48 | ok |
| 15:25.58 | p3nguin | dfgas-cr48: Two, set it static in the voip.ms customer portal. |
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| 15:26.25 | p3nguin | Either way will be fine for your single phone number. |
| 15:26.47 | parasitodelsur | p3nguin: I also got followme working :) |
| 15:27.15 | p3nguin | Did you add the bugfix so it doesn't ring your cell while you're on the desk phone? |
| 15:28.08 | parasitodelsur | p3nguin: Yes and is awesome |
| 15:28.09 | parasitodelsur | genius! |
| 15:28.30 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 15:28.33 | [TK]D-Fender | dfgas, SetVar=OUTCID=1234567890 |
| 15:28.44 | p3nguin | dfgas-cr48: Now when you need to start using more than one phone number, I have a solution for that as well. |
| 15:28.51 | [TK]D-Fender | dfgas, Set(CALLERID(num)=${OUTCID}) |
| 15:28.55 | p3nguin | Actually very similar to what tk just said. |
| 15:29.04 | [TK]D-Fender | dfgas, So you can set it in your peer |
| 15:29.20 | dfgas-cr48 | yah? |
| 15:29.36 | p3nguin | That requires setting it for each peer, though, and at this point isn't beneficial. |
| 15:29.49 | dfgas-cr48 | i have 2 numbers, but with 25 channels i don't think i need more than 1 |
| 15:29.51 | dfgas-cr48 | yah |
| 15:30.07 | parasitodelsur | p3nguin: check your msg. |
| 15:30.09 | dfgas-cr48 | the second one i can drop |
| 15:30.20 | p3nguin | I set the cidnum conditionally. |
| 15:30.58 | p3nguin | If the outbound number variable is set, use the value in it; otherwise, set it to my primary DID number. |
| 15:36.05 | ghost75 | Calling using Google Voice or via the Google Talk web client requires the use of Asterisk 11.0 or greater. |
| 15:36.07 | ghost75 | oh crap |
| 15:37.42 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 15:37.56 | p3nguin | Weird. I use 1.8 every day. |
| 15:38.33 | ghost75 | good :) |
| 15:39.20 | dfgas-cr48 | p3nguin, should it be exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=920XXXXXXX) or should it be a same line? |
| 15:39.25 | p3nguin | I don't anticipate upgrading to 11 for at least six months. |
| 15:39.47 | p3nguin | dfgas-cr48: Pastebin that whole extension. |
| 15:39.58 | slav3_kitten | is Set(STATUS=$[DB(${CALLERID(num)}/blockid)]); the correct syntax? if i do Set(STATUS=${DB(${CALLERID(num)}/blockid)}); it breaks syntax hilighting |
| 15:40.18 | p3nguin | first line is wrong |
| 15:40.22 | dfgas-cr48 | k |
| 15:40.44 | p3nguin | The $[] will make it evaluate the expression rather than read the value. |
| 15:40.51 | slav3_kitten | ah |
| 15:41.05 | p3nguin | Well, actually... |
| 15:41.13 | p3nguin | Depending on what you're trying to do, THAT could work. |
| 15:41.26 | p3nguin | What are you trying to do with the STATUS var and the db key? |
| 15:42.08 | *** join/#asterisk serafie (~erin@nat/digium/x-idfndiiyilsjgild) |
| 15:42.40 | slav3_kitten | determine weather or not to pass caller id or pass blocked caller id |
| 15:42.49 | dfgas-cr48 | p3nguin, http://pastebin.com/D72Nynyb |
| 15:43.02 | dfgas-cr48 | p3 i tried to add the name in there and it broke it |
| 15:43.09 | dfgas-cr48 | p3nguin, i tried to add the name in there and it broke it |
| 15:43.29 | dfgas-cr48 | so i commented it out and change the number line to 2 |
| 15:43.57 | p3nguin | dfgas-cr48: You can't change your outbound callerid name, so delete that line completely. |
| 15:44.03 | slav3_kitten | the db key returns a 1 or 0 |
| 15:44.15 | dfgas-cr48 | p3nguin, k |
| 15:44.24 | p3nguin | But the other line setting the number looks okay to me. |
| 15:44.49 | parasitodelsur | p3nguin: check your msg. one more thing :) |
| 15:45.53 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 15:45.58 | p3nguin | dfgas-cr48: That will set your callerid number before the call goes out, every single time. |
| 15:46.02 | p3nguin | That's good. |
| 15:46.22 | dfgas-cr48 | cool |
| 15:46.40 | dfgas-cr48 | thank you very much |
| 15:47.14 | p3nguin | slav3_kitten: Would it be sensible to set the key when it needs to be evaluated true and don't set it at all when it needs to be false? |
| 15:47.19 | jmetro | slav3_kitten: i meant the language file for vim itself, not asterisk. just something to check if you were to troubleshoot it. unless its an error in the file you were talking about. |
| 15:47.20 | p3nguin | as opposed to 1 and 0 |
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| 15:48.44 | slav3_kitten | one sec, pastebin |
| 15:48.50 | parasitodelsur | p3nguin: Done. Thank You. |
| 15:49.53 | p3nguin | I don't feel very good today. |
| 15:49.59 | slav3_kitten | http://pastie.org/private/p0ybkuct7hestx6vhgfaq |
| 15:50.07 | slav3_kitten | p3nguin, flu? |
| 15:50.11 | parasitodelsur | p3nguin: I noticed. |
| 15:50.19 | parasitodelsur | :) |
| 15:50.36 | slav3_kitten | makes p3nguin a hot tottie |
| 15:50.46 | p3nguin | Nah, just generally crappy feeling. |
| 15:50.49 | slav3_kitten | because lets face it. booze always helps |
| 15:51.03 | *** join/#asterisk leifmadsen (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:51.03 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:52.06 | p3nguin | That paste site isn't working right. I can't edit your paste. |
| 15:52.20 | p3nguin | There it goes. |
| 15:52.33 | p3nguin | The first time it gave some weird guru meditation error. :/ |
| 15:52.43 | slav3_kitten | hey look i forgot to take my phone number out of it :\ |
| 15:53.29 | *** join/#asterisk blitzrage (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:53.29 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 15:53.42 | p3nguin | You did a great job hiding your name, though! |
| 15:53.55 | slav3_kitten | i know right! |
| 15:54.20 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 15:57.14 | p3nguin | slav3_kitten: http://pastie.org/private/xaruedoofievi3dzljmtq |
| 15:57.55 | parasitodelsur | cya guys! p3nguin pkg should be there any time this week. |
| 15:58.46 | p3nguin | I don't quite understand the purpose of this extension, but there's the better way to do it. |
| 15:59.37 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ixbedtsinyirpiof) |
| 15:59.37 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:59.45 | p3nguin | It looks like you're going to set or hide your outbound caller id number based on the phone you are calling from. That part doesn't make sense to me. |
| 16:00.28 | slav3_kitten | i've got 6 phones in the house, and i'm using asterisk db to allow everyone to determine weather or not to set or hide caller id on their calls |
| 16:00.45 | slav3_kitten | 6 phones, 4 users |
| 16:01.14 | p3nguin | I see. |
| 16:01.41 | slav3_kitten | especially useful since dad has this thing for calling idiots on craigslist |
| 16:02.08 | n3hxs | Personals section? |
| 16:02.15 | p3nguin | The normal way to do that in the US is by prefixing the dialed number with *67 to block (when it is normally not blocked), *82 to unblock. (in cases where it is blocked always). |
| 16:02.49 | slav3_kitten | worse, people selling shit |
| 16:02.49 | n3hxs | Ahh, |
| 16:02.53 | p3nguin | I usually randomize when I call people like that. |
| 16:02.57 | n3hxs | I would make it send a CID of a not in service number. or just the 4 digit extension. |
| 16:03.19 | n3hxs | or 1 digit "0" ;-) |
| 16:04.24 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 16:08.54 | *** join/#asterisk navaismo (~navaismo@189.144.120.135) |
| 16:15.49 | ChannelZ | Get a number for ICE and use that. |
| 16:16.13 | slav3_kitten | ICE? |
| 16:16.38 | ChannelZ | Immigration and Customs Enforcement |
| 16:17.50 | slav3_kitten | ah |
| 16:17.54 | p3nguin | Okay, so what the hell is the purpose of this device? http://www.ebay.com/itm/320354331906 |
| 16:18.23 | slav3_kitten | it's a bnc standoff |
| 16:18.46 | p3nguin | What is the purpose? To simply add 2cm to the length of the connector? |
| 16:18.58 | jmetro | its a dollar peice of metal. |
| 16:19.09 | ghost75 | its a connector |
| 16:19.27 | slav3_kitten | could be useful if you had some like large heliax terminating into a radio and you needed to stand it off the back to allow it not to interfere with something else you have going into the radio? |
| 16:19.34 | slav3_kitten | imho in that case i'd make a small jumper |
| 16:20.07 | p3nguin | People use BNCs on heliax? |
| 16:20.09 | ghost75 | to convert male/female |
| 16:20.15 | p3nguin | ghost75: nope |
| 16:20.18 | p3nguin | Try again. |
| 16:20.49 | p3nguin | I would have thought people would use type N or at least a PL-259 for heliax. |
| 16:20.56 | ghost75 | Product  Description BNC Male to BNC Female Connectors. |
| 16:21.29 | drmessano | p3nguin: My guess would be to put another connector on the end of one that sees heavy use and wouldn't be practical replacing |
| 16:21.29 | p3nguin | ghost75: It connects ONTO a female. It provides a female. It does not change the gender. |
| 16:21.49 | ghost75 | sounds true |
| 16:21.55 | drmessano | It doesnt make total sense.. but I can see people buying it for that much |
| 16:22.17 | p3nguin | But if there is weight on that connector, the connector that it connects onto would also have weight on it. |
| 16:22.27 | drmessano | No, not for weight |
| 16:22.45 | slav3_kitten | p3nguin, not that i've seen but i hear they have such termination options |
| 16:22.47 | drmessano | Connect, disconnect, connect, disconnect. Wears out the contacts |
| 16:22.49 | p3nguin | I couldn't come up with any other purpose but to add 2cm and add loss. |
| 16:22.50 | jmetro | its a good way to make 90 cents on 10 cents of metal? |
| 16:23.01 | slav3_kitten | most heliax i've ever seen was N |
| 16:23.23 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 16:23.24 | p3nguin | Oh, life cycle. I could see that. |
| 16:23.28 | drmessano | Yep |
| 16:23.45 | p3nguin | Change that part every 10,000 cycles. |
| 16:23.54 | drmessano | I can see in SOME cases where maybe the original BNC wouldn't be easy to replace, and perhaps that would extend the life |
| 16:24.00 | drmessano | Exactly |
| 16:24.08 | p3nguin | So that multiples the on-board connector life by a power of 10,000. |
| 16:24.20 | p3nguin | multiplies |
| 16:24.22 | ghost75 | 10000 is a lot |
| 16:24.47 | p3nguin | Okay, drmessano came up with a potential practical use. |
| 16:25.20 | drmessano | However, extending by 2 cm is also useful. If you're setting up a phasing array on FM transmitters, 1/2 cm is enough to make a difference. We keep a box full of jumpers of different short lengths for that.. but same idea |
| 16:25.28 | p3nguin | In all my life, I have never worn out a BNC. |
| 16:26.25 | p3nguin | For a high-power FM broadcast transmitter, I would expect a different connector. |
| 16:26.36 | ghost75 | like to add 10 of those to reach the last one easier? |
| 16:26.49 | p3nguin | yeah, exactly |
| 16:27.02 | p3nguin | Don't have any coax? Buy 20 of these! |
| 16:27.13 | drmessano | The BNC is used in the sample for the combiner |
| 16:27.36 | p3nguin | Ah, right. I wasn't considering the lower powered parts of the station. |
| 16:28.59 | drmessano | The actualing tuning of a combined FM transmitter setup is actually done at very low power stage |
| 16:29.06 | *** join/#asterisk fritz09 (~Adium@pop1-224.catv.wtnet.de) |
| 16:29.25 | drmessano | and its tempermental as shit.. which is why you have the different lengths of coax |
| 16:29.35 | ghost75 | how this is called? bnc extender? |
| 16:29.59 | drmessano | You may as well call it an extender |
| 16:30.04 | p3nguin | Do those stations have any type of antenna tuner between the final amplifier and the feedline? |
| 16:30.51 | ghost75 | if you search google pictures i cant find anything like that |
| 16:31.03 | p3nguin | giant inductor/capacitor -based tuners |
| 16:31.57 | drmessano | Yes, each of the transmitters has tuning on the PA stage. The higher power ones use a tube type PA, so you have plate, grid, and output tuning |
| 16:32.06 | slav3_kitten | mmmm tubes |
| 16:33.02 | slav3_kitten | i love tinkering with tubes, lately i've been big into indicator tubes |
| 16:33.12 | ghost75 | Reverse Polarity BNC Male to BNC Female Adapter |
| 16:33.14 | drmessano | Ugh..and the Filament voltage is also adjustable, which does contribute to the overall Q of the PA |
| 16:33.31 | slav3_kitten | did a whole bunch of tube effects pedals years ago |
| 16:33.35 | slav3_kitten | just something about tubes is fu |
| 16:33.59 | slav3_kitten | fun |
| 16:34.29 | drmessano | http://usr.audioasylum.com/images/4/49339/3cx20000.png <- Thats a typical PA tube |
| 16:35.16 | drmessano | Somewhere I have a pic of one completely melted down |
| 16:36.03 | jmetro | i tinker with PC's. the cyber monday deal at frys had me drooling |
| 16:36.16 | ghost75 | http://www.showmecables.com/product/BNC-Female-Adapter-to-Reverse-Polarity-BNC-Male.aspx |
| 16:36.17 | p3nguin | It's much larger than the tubes I would use: http://www.df6na.de/surplus/tubes/3-500Z.JPG |
| 16:36.59 | drmessano | TV transmitter tubes are pretty awesome |
| 16:37.34 | p3nguin | ghost75: The one I was confused about is a standard polarity BNC male on one side and a standard polarity BNC female on the other side. |
| 16:37.59 | drmessano | Nothing like seeing a 250kw klystron tube sitting halfway in a vat of cooling oil |
| 16:38.19 | ghost75 | the one from ebay? |
| 16:38.36 | p3nguin | yes, that one. |
| 16:38.48 | ghost75 | messed up description i'd say |
| 16:40.38 | p3nguin | I'm not nearly as worried over it now that I have been given a couple sensible uses for it. |
| 16:41.00 | drmessano | What about a bulkhead male BNC? |
| 16:41.06 | drmessano | Name 1 use |
| 16:41.09 | drmessano | I dare you |
| 16:41.54 | p3nguin | I'd guess you could put that on a much larger coaxial cable. |
| 16:42.00 | ghost75 | picture? |
| 16:42.04 | drmessano | In the words of Dwight Shrute. FALSE |
| 16:42.07 | drmessano | lol |
| 16:42.13 | drmessano | I cant see a use for one |
| 16:42.16 | slav3_kitten | drmessano, home made TDR box |
| 16:42.24 | drmessano | http://www.rfstreet.com/images/ProPho/598/BNC_Male_plug_front_mount_bulkhead_with.jpg |
| 16:42.50 | p3nguin | Oh. |
| 16:43.00 | p3nguin | That's kind of... |
| 16:43.03 | ghost75 | mount it in a case |
| 16:43.07 | p3nguin | opposite what you would normally do. |
| 16:43.14 | slav3_kitten | because 2 female 1 male T is easier to find then a Male Male female T |
| 16:43.31 | p3nguin | I disagree with that. |
| 16:44.01 | slav3_kitten | i have boxes upon boxes of T's from thinnet days |
| 16:44.08 | slav3_kitten | so my TRD has a male bulkhead on it |
| 16:44.46 | p3nguin | Speaking of tubes, I really need to find an HF amplifier. |
| 16:45.11 | ghost75 | is that already available in US: http://www.mohrresults.com/wp-content/uploads/2009/07/super-gulp-150x150.jpg |
| 16:45.26 | p3nguin | ha |
| 16:45.29 | p3nguin | I haven't seen that. |
| 16:45.42 | ghost75 | wouldnt surprise me though :) |
| 16:46.36 | slav3_kitten | ghost75, i'd buy one |
| 16:47.30 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 16:47.54 | p3nguin | I wouldn't. It'd go flat before I could consume it all. |
| 16:48.06 | slav3_kitten | p3nguin, i just want the cup |
| 16:48.19 | p3nguin | It would be cool to have. |
| 16:48.26 | ghost75 | yeah |
| 16:48.29 | p3nguin | Don't know where I'd store it, but it would be fun to have it. |
| 16:49.07 | slav3_kitten | game room |
| 16:49.15 | ghost75 | then go to mcdonalds and just put it on the table to see how others look at |
| 16:49.38 | danfromuk | I've got a potential client with fibre optic broadband from Orange. They have a Brightbox router which seems to have a bug which causes lost RTP packets during a call. Has anyone got experience with this router? |
| 16:50.53 | slav3_kitten | mouser, shutup an take my money :D |
| 16:51.54 | *** join/#asterisk svnNB (~svn@host103-70-dynamic.60-82-r.retail.telecomitalia.it) |
| 16:53.25 | p3nguin | I think I could be satisfied with an AL-811 amplifier. |
| 16:53.37 | p3nguin | Maybe an SB-1000. |
| 16:53.48 | slav3_kitten | yay, just ordered 500 crimp terminals, guess what my weekend is going to be spent doing :/ |
| 16:53.52 | p3nguin | I need at least 1kW. |
| 16:54.54 | *** join/#asterisk celord (~celord@201.191.198.57) |
| 16:55.21 | slav3_kitten | dad was all "so i got this game, it's a bit water damaged..." all the connectors have corroded contacts as a result |
| 16:57.01 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 17:01.57 | *** join/#asterisk Widler (47c8d864@gateway/web/freenode/ip.71.200.216.100) |
| 17:03.01 | Widler | anyone know i keep getting the error message "The GUI does not have necessary privileges. Please check the manager permissions for the user !"? |
| 17:03.11 | Widler | i have permission set to all |
| 17:03.32 | p3nguin | Hmm. Wrong channel. |
| 17:03.57 | slav3_kitten | p3nguin, could it be an owner/group problem possibly? |
| 17:04.12 | p3nguin | Don't know. This isn't the channel for it. |
| 17:04.13 | Widler | thanks |
| 17:04.29 | slav3_kitten | it isn't? |
| 17:04.40 | p3nguin | Of course it isn't. |
| 17:04.44 | p3nguin | Why would you think it is? |
| 17:04.46 | Widler | do you know what channel i should be asking |
| 17:04.55 | p3nguin | Maybe #asterisk-gui |
| 17:04.56 | slav3_kitten | i thought there was an asterisk gui thing |
| 17:05.25 | p3nguin | Good luck getting support on a dead piece of software, though. |
| 17:05.38 | p3nguin | Asterisk doesn't have a GUI. |
| 17:05.39 | drmessano | It's deader than dead |
| 17:06.01 | p3nguin | The Asterisk GUI just happens to be a GUI that people use on Asterisk. |
| 17:06.18 | p3nguin | The name is purely a coincidence. |
| 17:06.21 | *** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se) |
| 17:06.27 | jmetro | make your own gui, sell it for lolipops, become king candy. |
| 17:06.45 | drmessano | Asterisk-GUI is laying there next to Asterisk SCF and they're sniffing each other and complaining about how bad the other one smells like rotting meat |
| 17:07.58 | *** join/#asterisk thehar (~thehar@diddlebox.thehar.com) |
| 17:16.13 | *** join/#asterisk qakhan (~qakhan@70-90-90-130-BusName-dc.hfc.comcastbusiness.net) |
| 17:16.17 | qakhan | hi all, |
| 17:16.21 | *** join/#asterisk Tim_Toady (~fuzzy@178.128.20.156.dsl.dyn.forthnet.gr) |
| 17:17.36 | qakhan | can i call on group if exts? like 2312,2313,2314,2315 |
| 17:17.37 | qakhan | ? |
| 17:18.04 | p3nguin | You can use one extension to call a group of phones. |
| 17:18.30 | qakhan | p3nguin plz help me how i can do this |
| 17:18.32 | *** join/#asterisk feeshon (~gaston@ool-45787011.dyn.optonline.net) |
| 17:19.08 | p3nguin | What extension do you want to use to call the group of phones? |
| 17:19.15 | qakhan | 2300 |
| 17:19.24 | feeshon | Can asterisk ring 2 phones that are bound to the same extension? |
| 17:19.40 | p3nguin | exten => 2300,1,Dial(SIP/phone1&SIP/phone2&SIP/phone3,36) |
| 17:19.44 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:19.44 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:20.03 | p3nguin | feeshon: Phones aren't bound to extensions. |
| 17:20.26 | p3nguin | Extensions can dial phones, but phones are not bound. |
| 17:20.44 | qakhan | p3nguin what is 36 at the end of line? |
| 17:20.55 | p3nguin | 36 seconds ring timeout |
| 17:21.11 | p3nguin | You can use any timeout or none. |
| 17:21.13 | feeshon | Ok let me clarify, can a single extension ring > 1 phone? |
| 17:21.19 | p3nguin | It was just an example. |
| 17:21.30 | qakhan | and all phones will ring same time |
| 17:21.31 | *** join/#asterisk fubada (~aamerik@ool-457e3295.dyn.optonline.net) |
| 17:21.44 | p3nguin | feeshon: Of course. exten => 123,1,Dial(SIP/phone1,30) |
| 17:21.54 | p3nguin | qakhan: That is correct. |
| 17:22.04 | fubada | p3nguin: thanks, I will help feeshon with that |
| 17:23.29 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 17:25.44 | feeshon | p3nguin: can you help me understand how I can use my existing http://pastie.org/private/xve8scd2quyif6sot4w to do what you suggest |
| 17:25.59 | navaismo | is there a chance that the r-series utilities(agent resources) will work with Centos 6? |
| 17:26.07 | feeshon | only for one extension, all others (not shown in the pastie) need to remain the same |
| 17:26.46 | p3nguin | This looks like something FreePBX cooked up. |
| 17:27.03 | fubada | no, its not |
| 17:27.11 | fubada | it was written by me, months ago |
| 17:27.18 | fubada | from scratch |
| 17:27.39 | p3nguin | If you could write all that stuff, surely you can take care of this new task. |
| 17:27.56 | fubada | p3nguin: agreed but im teaching lil feeshon here how to use irc |
| 17:27.59 | fubada | and asterisk |
| 17:28.07 | fubada | and this is one of the tasks I want him to do on his own |
| 17:28.12 | fubada | without bothering me |
| 17:32.50 | qakhan | Thanks p3nguin its working fine :) |
| 17:34.45 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 17:34.49 | Ice_Strike | Hi |
| 17:36.01 | feeshon | exit |
| 17:36.28 | jpsharp | stage left |
| 17:38.41 | Ice_Strike | When I execute a reload command |
| 17:38.47 | Ice_Strike | I get warning and error |
| 17:38.48 | Ice_Strike | see http://pastebin.com/b6ViKxA0 |
| 17:38.54 | p3nguin | Which reload command? |
| 17:39.02 | Ice_Strike | on CLI |
| 17:39.21 | ghost75 | [Nov 27 18:31:57] WARNING[15397]: app_fax.c:173 span_message: WARNING T.30 ECM carrier not found <- is this normal ? |
| 17:39.25 | p3nguin | Which reload command? |
| 17:39.35 | p3nguin | Just "reload"? |
| 17:39.39 | Ice_Strike | Yes just reload |
| 17:39.41 | p3nguin | Yeah, stop doing that. |
| 17:40.55 | kaldemar | Ice_Strike: are you using LDAP configs? |
| 17:40.59 | p3nguin | It looks like the only error there is related to not having a host configured for your LDAP. |
| 17:41.38 | Ice_Strike | Where LDAP are setup? |
| 17:42.02 | p3nguin | If you were using LDAP, you would know it. |
| 17:43.23 | [TK]D-Fender | Ice_Strike, [Nov 27 20:36:28] WARNING[23549]: pbx.c:10558 ast_context_verify_includes: Context 'inbound_campaigns' tries to include nonexistent context 'day|9:00-17:00|mon-fri|*|*' |
| 17:43.28 | [TK]D-Fender | Ice_Strike, means exactly what it says |
| 17:46.20 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:52.06 | Ice_Strike | p3nguin I am not using LDAP config |
| 17:52.35 | p3nguin | Then delete the conf and/or noload the module for it. |
| 17:52.51 | Ice_Strike | Hmm odd, I installed Asterisk by default. |
| 17:56.53 | Ice_Strike | <PROTECTED> |
| 17:57.27 | [TK]D-Fender | Ice_Strike, You have a dialplan include error wchi is for a context clearly not included with the sample configs. You've put that in there somewhere |
| 17:57.58 | [TK]D-Fender | Ice_Strike, Asterisk samples don't mention "campaigns". Call center stuff does that, and there is no sample provided that hints at such a purpose |
| 17:58.01 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 17:58.47 | Ice_Strike | I got Asterisk samples from Call center script. |
| 17:59.15 | *** join/#asterisk rampage73 (~rampage73@bob.dctechonline.com) |
| 18:00.00 | [TK]D-Fender | Ice_Strike, Then they've done something wrong. Take it up with whoever supplied them, or go in and fix them yourself. |
| 18:00.30 | *** part/#asterisk rampage73 (~rampage73@bob.dctechonline.com) |
| 18:04.46 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 18:07.18 | Ice_Strike | I had a look all the configs file nothing indicate it using LDAP function |
| 18:07.25 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 18:08.44 | [TK]D-Fender | Ice_Strike, I don't see anything about that dialplan error having anything to do with LDAP |
| 18:09.00 | Ice_Strike | [Nov 27 21:01:31] ERROR[23566]: res_config_ldap.c:1658 parse_config: No directory URL or host found |
| 18:09.02 | p3nguin | Just noload the res_config_ldap.c module in modules.conf to remove the error. |
| 18:09.15 | p3nguin | That was the ONLY error listed in the entire pastebin. |
| 18:09.23 | Ice_Strike | Thanks |
| 18:09.59 | [TK]D-Fender | p3nguin, In thqat PB he also had a dialplan include error |
| 18:10.22 | p3nguin | Nope, just a warning. |
| 18:10.54 | [TK]D-Fender | "mistake" :) |
| 18:11.10 | [TK]D-Fender | I'll leave the gravity of consequence out of it ;) |
| 18:13.20 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 18:13.24 | Ice_Strike | I have added noload => res_config_ldap.so |
| 18:13.30 | Ice_Strike | in the modules.conf |
| 18:13.34 | Ice_Strike | and did reload command |
| 18:13.40 | Ice_Strike | still showing same error |
| 18:16.04 | Ice_Strike | oh restarting asterisk fixed a problem |
| 18:18.00 | p3nguin | I told you to stop using "reload". |
| 18:18.07 | slav3_kitten | [TK]D-Fender, i have my dialplan setup so the only includes are in my phones context, only thing that uses that context are the sip entries for physical phones |
| 18:18.11 | p3nguin | (1139.39) <Ice_Strike> Yes just reload |
| 18:18.13 | p3nguin | (1139.41) <p3nguin> Yeah, stop doing that. |
| 18:19.39 | Ice_Strike | I didnt know you referring that to module. |
| 18:19.53 | p3nguin | I wasn't referring to any module. |
| 18:20.22 | qakhan | all plz let me know is it correct |
| 18:20.25 | qakhan | exten => s,1,GotoIfTime(08:00-19:59,mon-sat,*,*?itc:ncs) |
| 18:21.22 | Ice_Strike | by default where "include" path are located? |
| 18:21.30 | Ice_Strike | For example in the extention.conf file: include => agent |
| 18:22.44 | navaismo | there is no default, you create your own extensions.conf |
| 18:23.41 | p3nguin | qakhan: I think it is okay. |
| 18:24.43 | qakhan | but its doesnt work |
| 18:24.49 | p3nguin | Show me. |
| 18:29.12 | ghost75 | do i backup anything before updating asterisk? |
| 18:30.34 | pabelanger | ghost75, Yes, that would be a good procedure to have |
| 18:31.01 | ghost75 | etc folder enough? |
| 18:31.28 | *** join/#asterisk _Corey_ (~chatzilla@pool-72-78-178-187.phlapa.fios.verizon.net) |
| 18:33.12 | p3nguin | I back up all the directories involved. Just in case. |
| 18:48.37 | *** join/#asterisk w9sh (~chatzilla@64.238.96.125) |
| 18:50.58 | SeRi | waz up guys |
| 18:51.27 | p3nguin | looks at Steven. |
| 18:53.46 | p3nguin | I still feel crummy. I think I should have a nap. |
| 18:55.43 | SeRi | p3nguin: Thats a good idea. |
| 18:59.15 | SeRi | p3nguin: did you recived the confirmation? |
| 18:59.26 | p3nguin | Yes. |
| 18:59.30 | SeRi | Perfect. |
| 18:59.34 | *** join/#asterisk leifmadsen (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:59.34 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:59.47 | p3nguin | I did not check anything else, but I did see the email. |
| 19:00.26 | SeRi | One minor detail.... I for got the lifter cable at home :( |
| 19:00.32 | p3nguin | Oops. |
| 19:00.36 | SeRi | You will need that |
| 19:00.57 | SeRi | SOOOOOO I will drop the pkg at the mail room at work tomorrow |
| 19:01.37 | SeRi | should be there this week still. |
| 19:02.46 | SeRi | I have anothe rpkg going out for dijib's lazy ass. |
| 19:02.48 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 19:04.11 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 19:04.37 | p3nguin | heh |
| 19:05.02 | SeRi | yeap... |
| 19:05.36 | SeRi | old equipment that I was fixing to throw away.... dijib mention you need it a board with pcie slots? |
| 19:06.22 | drmessano | Wait, I want something |
| 19:06.55 | SeRi | drmessano: LOL... all taken |
| 19:07.09 | SeRi | drmessano: you mention you where dumping comcast business... may I ask why? |
| 19:07.40 | SeRi | I just an account with them and it will be nice to know if they suck as much as they do on the home side |
| 19:07.56 | SeRi | s/I just/I just got/ |
| 19:08.43 | drmessano | Because I want cable TV, and Comcast are F&&&KING BEEP BEEP BEEP BEEP, and they are unable to set up a dual residence so I can have a residential TV account and Business Internet. My wife wants Cable TV. Nuff said. |
| 19:09.05 | SeRi | ah. ok |
| 19:09.33 | SeRi | man that sucks but make sense. I dont have cable tv and dont want it ether |
| 19:09.41 | SeRi | I just dumped directv |
| 19:09.41 | drmessano | I had it that way once before, but they can't seem to figure out how to set it up again |
| 19:09.54 | drmessano | So .. GONE |
| 19:09.55 | SeRi | oh that sucks |
| 19:10.24 | SeRi | I understand your frustation |
| 19:10.31 | drmessano | I also moved a couple things I was hosting at home to inexpensive VPS'es, so I have negated some of the need there |
| 19:10.39 | SeRi | I will credit your account with aloyal customer fee |
| 19:10.54 | SeRi | ah. ok. |
| 19:17.17 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 19:19.17 | *** join/#asterisk feeshon (~gaston@ool-45787011.dyn.optonline.net) |
| 19:23.13 | qakhan | all what is wrong with this while loop |
| 19:23.28 | qakhan | exten => s,1,Answer() |
| 19:23.28 | qakhan | exten => s,n,Set(i=0) |
| 19:23.28 | qakhan | exten => s,n,While($[${i} < 3]) |
| 19:23.28 | qakhan | exten => s,n,NoOP() |
| 19:23.28 | qakhan | exten => s,n,Background(WITC) |
| 19:23.28 | qakhan | exten => s,n,WaitExten(10) |
| 19:23.28 | qakhan | exten => s,n,Set(i=$[${i} + 1]) |
| 19:23.29 | qakhan | exten => s,n,EndWhile() |
| 19:23.29 | qakhan | exten => s,n,Hangup |
| 19:23.41 | jmetro | woah woah, pastebin buddy. |
| 19:23.58 | qakhan | ohh sorry |
| 19:24.25 | *** join/#asterisk wonderworld (~w@dsdf-4db5dd1e.pool.mediaWays.net) |
| 19:26.39 | [TK]D-Fender | qakhan, Who says it's wrong? Show us it actually FAILING. |
| 19:27.11 | p3nguin | What, the pastebins quit working today? |
| 19:27.11 | [TK]D-Fender | qakhan, And where is your previous backup for the GotoIfTime problem? |
| 19:28.20 | p3nguin | Here's another seemingly useless connector: http://www.ebay.com/itm/121023980863 |
| 19:28.44 | p3nguin | Convert an SMA female into an SMA female. |
| 19:29.18 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:29.18 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:29.28 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 19:29.47 | [TK]D-Fender | p3nguin, Looks like it swivels, which would validate it.... |
| 19:30.55 | p3nguin | Once you tighten it down, it doesn't. |
| 19:31.00 | qakhan | [TK]D-Fender here is * cli on while loop |
| 19:31.01 | qakhan | http://pastebin.com/7sV1fiDh |
| 19:31.23 | qakhan | while loop is not increasing |
| 19:31.45 | drmessano | I would absolutely justify that one.. SMA sucks. I would love one stacked on top so I didn't F up the connector on an HT trying to conn/disconn in a hurry. |
| 19:32.21 | file | drmessano, I don't follow you on Twitter! *gasp* |
| 19:32.23 | jmetro | qakhan : your statement above tells me you should look at your increment of I then... |
| 19:32.31 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 19:33.06 | qakhan | jmetro can u tell me whats wrong? |
| 19:33.11 | qakhan | plz |
| 19:33.14 | [TK]D-Fender | qakhan, Executing [i@itc:4] Goto("DAHDI/2-1", "s|3") in new stack <-- |
| 19:33.21 | p3nguin | I don't understand why the radio mfgrs decided to change from BNC to SMA. |
| 19:33.28 | p3nguin | BNC worked fine for YEARS. |
| 19:33.41 | [TK]D-Fender | qakhan, You somehow decided that in that flood that you wouldn't provide us the invalid handler that actually has the problem |
| 19:34.06 | [TK]D-Fender | qakhan, The count does go up... but in "i" you jump back to the a point before it sets the count to 0 so i keeps getting reset |
| 19:34.15 | jmetro | exactly. |
| 19:34.19 | [TK]D-Fender | qakhan, You should pay attention to where your Goto's point |
| 19:34.54 | [TK]D-Fender | <PROTECTED> |
| 19:35.52 | drmessano | BNC is too large for newer radios, and BNC makes a really bad connection. Icom had it right when they put TNC's on their commercial stuff. Would have been nice to see that get adopted elsewhere |
| 19:35.58 | *** part/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 19:36.01 | qakhan | all here is my dialplan |
| 19:36.02 | qakhan | http://pastebin.com/WSi2wC26 |
| 19:36.20 | *** join/#asterisk feeshon (~gaston@ool-45787011.dyn.optonline.net) |
| 19:38.20 | p3nguin | If you jump out of the While() loop by entering an extension, does the While() die all by itself, or do you still need to ExitWhile()? |
| 19:39.34 | qakhan | p3nguin if i press correct ext my call goes to that ext, |
| 19:39.43 | p3nguin | That's not what I'm asking. |
| 19:40.18 | p3nguin | I'm asking from an angle much like leaving a Gosub() without ever Return()ing. |
| 19:40.38 | [TK]D-Fender | qakhan, <[TK]D-Fender> qakhan, Executing [i@itc:4] Goto("DAHDI/2-1", "s|3") in new stack <-- |
| 19:40.44 | [TK]D-Fender | qakhan, Clearly NOT all of your dialplan |
| 19:41.00 | [TK]D-Fender | qakhan, I already told you exactly what the problem was. This was a 1 CHARACTER fix. |
| 19:41.23 | [TK]D-Fender | (had you hard-coded priorities) |
| 19:41.34 | [TK]D-Fender | qakhan, Just move the label DOWN. |
| 19:43.17 | qakhan | [TK]D-Fender did u see my dialplan? |
| 19:44.19 | p3nguin | WTF? I have never heard this before. I just called an auto-attendant while testing some things, and one of the sound files played in super-speed-fast-forward mode. What causes that?! |
| 19:44.46 | p3nguin | The phone used g.722 when making the call to asterisk. |
| 19:45.09 | jmetro | p3nguin i had the reverse happen when i was sox'ing media files the wrong way. |
| 19:45.18 | sp00kz | i would guess some clock issue on the server |
| 19:45.55 | p3nguin | I could also hear a difference in some of the sound files switching from a muffled "normal" sound to HD quality. |
| 19:46.09 | p3nguin | Perhaps all the sounds aren't available in all formats. |
| 19:46.45 | qakhan | [TK]D-Fender you know what i want to do? |
| 19:46.48 | [TK]D-Fender | qakhan, exten => s,3(itc),Answer() <----- move this label onto the WHILE() line |
| 19:47.01 | [TK]D-Fender | Your call debug showed a Gotot your "dialplan" did not. |
| 19:47.06 | [TK]D-Fender | qakhan, You are not showing MATCHING bits |
| 19:47.26 | [TK]D-Fender | qakhan, I've now told you THREE times what to fix for this. |
| 19:47.38 | qakhan | 1 min plz |
| 19:50.50 | qakhan | [TK]D-Fender like this http://pastebin.com/iFR71ybh |
| 19:53.05 | p3nguin | Ha! The one that plays in super-fast-forward mode is playing in .slin16 instead of g.722. |
| 19:54.53 | jmetro | =) |
| 19:57.10 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 19:58.23 | file | looks around |
| 19:59.18 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 20:03.49 | ghost75 | how can i install asterisk over apt from debian testing without updating hundreds of other modules |
| 20:05.43 | slav3_kitten | you can't, if asterisk requires x version you need x version |
| 20:05.46 | [TK]D-Fender | qakhan, I said move the LABEL. Why'd you move the WHOLE LINE? |
| 20:06.26 | [TK]D-Fender | qakhan, Do you see point in looping Answer()? |
| 20:06.26 | Nivex | ghost75: I'd personally recommend using the asterisk from backports |
| 20:11.06 | qakhan | [TK]D-Fender sorry i bother u soo much |
| 20:11.12 | qakhan | Thanks i got your point |
| 20:11.21 | qakhan | and its working now :) |
| 20:12.05 | ghost75 | what is it that apt-get check brings no errors and when i want to update asterisk it tells me update-notifier is broken and python-apt has missing dependency |
| 20:12.59 | *** join/#asterisk jsjc (~Adium@48.Red-83-41-72.dynamicIP.rima-tde.net) |
| 20:13.36 | ghost75 | oh, 1.8 is not in backports |
| 20:14.35 | ghost75 | asterisk/squeeze upgradeable from 1:1.6.2.9-2+squeeze5 to 1:1.6.2.9-2+squeeze6 |
| 20:17.32 | *** join/#asterisk celord (~celord@201.191.198.57) |
| 20:21.45 | *** join/#asterisk blee (~blee@70.118.107.77) |
| 20:22.09 | ghost75 | Nivex: do i need to change apt policy to get newer version from backports? |
| 20:22.41 | Nivex | ghost75: http://backports-master.debian.org/Instructions/ |
| 20:23.11 | *** join/#asterisk leifmadsen (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:23.11 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:24.58 | Nivex | I wonder when 11.x will make it into Sid |
| 20:25.17 | ghost75 | thx |
| 20:25.29 | ghost75 | so pid100 isnt active by default |
| 20:26.15 | ghost75 | shouldnt they bring 10 before? |
| 20:26.33 | Nivex | 11 is current and is also an LTS release, so they'll probably jump to it |
| 20:26.40 | Nivex | given that they are currently only tracking 1.8 |
| 20:27.38 | Qwell | In 2 years, they might. |
| 20:27.48 | ghost75 | should i update asterisk.conf or leave old version? |
| 20:27.52 | ghost75 | from 1.6 to 1.8 |
| 20:28.07 | Nivex | I figure they aren't doing much with new versions of software right now because they're trying to get Wheezy out the door |
| 20:28.24 | Nivex | ghost75: for that you'll want to refer to the upgrade documentation |
| 20:28.35 | ghost75 | i cant remember changing anything in asterisk.conf |
| 20:38.49 | p3nguin | Look at your sample file from 1.8 and look at your production file from your 1.6.x box. |
| 20:38.53 | p3nguin | Then decide. |
| 20:39.22 | slav3_kitten | ghost75, just give it a whirl an see where the chips fall |
| 20:47.50 | jmetro | The new versions of ubuntu are pretty bad. |
| 20:48.06 | p3nguin | s/new // |
| 20:48.36 | Nivex | I switched to Xubuntu to get away from Unity. |
| 20:49.16 | jmetro | The old versions of ubuntu werent bad. And the new one is okay if you go with LDXE |
| 20:49.33 | jmetro | mainly unity running on the desktop slowed my box to a standstill. |
| 20:51.47 | ghost75 | i updated now asterisk doesnt start |
| 20:52.00 | ghost75 | maybe because i skipped to update init.d file |
| 20:55.27 | drmessano | Unity isn't bad. I like the interface, I wish it had a few less bugs |
| 20:56.01 | p3nguin | This is a test. |
| 20:56.04 | p3nguin | s/This/It/;s/a/only a/ |
| 20:56.20 | p3nguin | That's what I was afraid of. |
| 20:56.33 | p3nguin | kicks infobot in the bits |
| 20:59.57 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 21:02.28 | ghost75 | [Nov 27 22:00:54] ERROR[31742] ais/clm.c: Could not initialize cluster membership service: Try Again |
| 21:05.06 | ghost75 | what that means, i dont use this |
| 21:06.27 | ghost75 | at least starts now |
| 21:11.52 | ghost75 | does it make sense to put nat=no on phones and nat=yes on peer ? |
| 21:12.12 | ghost75 | or use it only in global |
| 21:15.15 | p3nguin | No. Configure it PER PEER ENTRY as needed. |
| 21:15.28 | *** join/#asterisk greenwolf (425700f6@gateway/web/freenode/ip.66.87.0.246) |
| 21:15.34 | p3nguin | If a phone is behind NAT, put nat=yes and directmedia=no. |
| 21:15.39 | *** part/#asterisk asteriskmonkey (~philip@206.51.27.151) |
| 21:16.15 | ghost75 | i use phones only in lan |
| 21:16.44 | WIMPy | Or just a global directmedia=nonat? |
| 21:16.47 | p3nguin | If an ITSP is not behind NAT, but nat=no. If you are behind a NAT, use directmedia=no. |
| 21:17.54 | p3nguin | s/but/put/ |
| 21:18.30 | ghost75 | now i have nat=no and directmedia=no - asterisk server is behind nat |
| 21:20.50 | p3nguin | Have them where? |
| 21:22.07 | ghost75 | "them" ? |
| 21:23.04 | *** join/#asterisk Tim_Toady (~fuzzy@178.128.20.156.dsl.dyn.forthnet.gr) |
| 21:24.10 | *** join/#asterisk doogienz (~doogienz@support.net24.co.nz) |
| 21:24.12 | p3nguin | them = the settings we are talking about |
| 21:25.08 | ghost75 | sip.conf or what u mean? |
| 21:25.22 | doogienz | Hi all - I'm trying out asterisk11 for the conf bridge function - interesting issue, it's not creating an asterisk.ctl in /var/run/asterisk/ - process is running and listening, but obviously no asterisk -r access. Anyone come across this? astrundir is fine , but only a .pid file being created. |
| 21:28.48 | p3nguin | I mean WHERE IN YOUR sip.conf |
| 21:29.06 | p3nguin | doogienz: Disable SELinux. |
| 21:29.10 | ghost75 | ah lol |
| 21:29.13 | ghost75 | in global |
| 21:29.16 | doogienz | Yeah it is - first thing I check. |
| 21:29.43 | doogienz | That's what has got me stumped - ah well back to reinstall and try again. |
| 21:29.53 | p3nguin | That's silly. This isn't Windows. |
| 21:30.04 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 21:30.05 | jmetro | confbridge is super easy and nice. |
| 21:30.12 | jmetro | also dont forget you can asterisk -r with rasterisk |
| 21:30.20 | p3nguin | Except he can't. |
| 21:30.24 | tonikasch | #indymedia |
| 21:30.26 | tonikasch | ui |
| 21:30.30 | tonikasch | sorry |
| 21:30.48 | doogienz | Yeah it's asterisk 11 - I compiled it from scratch, so I'll just clean it out and try again. |
| 21:30.49 | tonikasch | this wasn't directed to this channel |
| 21:32.51 | ghost75 | i just had problem that asterisk didnt start after update, so i checked /var/log/asterisk/messages |
| 21:33.33 | doogienz | Hmmm... make clean & make install - boomshaka. Wonder what went wrong last time. |
| 21:34.30 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:34.30 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:35.14 | doogienz | Anyone started anything to do with a web gui for confbridge yet? |
| 21:38.16 | fubada | doogienz: fop2 |
| 21:38.23 | doogienz | LOL no. |
| 21:38.29 | *** join/#asterisk Tim_Toady (~fuzzy@178.128.20.156.dsl.dyn.forthnet.gr) |
| 21:38.36 | fubada | p3nguin: sorry about my earlier messages rgarding feeshon |
| 21:38.42 | fubada | i really do need some help now :) |
| 21:38.50 | doogienz | fop2 is fantastic, but it's not a web gui you can use for confbridge to manage conferences, create / delete kick etc. |
| 21:39.24 | fubada | thatd be a cool mobile app |
| 21:39.50 | Qwell | malcolmd: *nudge* |
| 21:40.40 | fubada | can someone suggest a way I can make two phones ring when an extension is called? I understand I can simply Dial two SIP accounts |
| 21:40.55 | fubada | but, im using users.conf and it is not straight forward |
| 21:41.09 | fubada | im using users.conf because back when I wrote this config, phoneprov required ot |
| 21:41.12 | fubada | it |
| 21:41.49 | p3nguin | ~users.conf |
| 21:41.49 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 21:42.36 | p3nguin | If you want to call two phones from one extension, that's simply what you do. Dial(SIP/phone1&SIP/phone2); There is nothing more to it. |
| 21:42.37 | doogienz | @fubada, I wrote macros that take arg |
| 21:42.49 | doogienz | Check the status of the extensions and dial accordingly |
| 21:43.55 | fubada | http://pastie.org/private/0qvkpyptdr1d3lxa4bpvw |
| 21:44.07 | fubada | can someone take a peek at that |
| 21:44.09 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ixbedtsinyirpiof) |
| 21:44.23 | *** join/#asterisk imox (~imox@91-66-32-57-dynip.superkabel.de) |
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| 21:47.26 | fubada | ok i solved my issue |
| 21:47.32 | fubada | thanks doogienz |
| 21:49.38 | *** join/#asterisk jraddin (~jraddin@72-48-177-116.static.grandenetworks.net) |
| 21:50.44 | jraddin | I am using asterisk with freepbx. When I pick up a phone connected to an FXS on a digium 400 card, I get a dialtone, but as soon as I hit a number, asterisk hangs up. Any ideas? |
| 21:51.01 | ghost75 | Remaining options are not specific to users.conf entries but are general -> WTF |
| 21:52.15 | fubada | ghost75: regarding my pastie? |
| 21:52.36 | ghost75 | no, regarding what i just discovered |
| 21:53.03 | *** join/#asterisk Ryushin (chris@2001:1938:2a5:0:224:21ff:fee4:1de3) |
| 21:53.51 | *** join/#asterisk vassilux (~vassilux@31.35.29.247) |
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| 21:55.08 | tompaw | Is there a way for ConfBridge to record directly to mp3? |
| 21:55.35 | Qwell | no |
| 21:55.36 | malcolmd | jraddin: presumably your dial plan is wrong |
| 21:56.19 | ghost75 | are the guys singing in conference? |
| 21:57.55 | SeRi | you can convert when the the out put is done. |
| 21:57.55 | vassilux | hi alls, I have a cluster with cluster address 10.10.1.15, the first node of cluster is 10.10.1.10. I have troubles to call a SNOM phone when the address of register server is the cluster address. When I use the node address 10.10.1.10 it works. Any idea ? |
| 21:58.11 | vassilux | my asterisk version 1.0.8.16 |
| 21:58.12 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:58.18 | SeRi | I am sure it can be accomplish via script or vi context. |
| 22:01.37 | drmessano | vi? |
| 22:02.05 | jmetro | vim |
| 22:02.14 | SeRi | drmessano: sorry |
| 22:02.15 | SeRi | via |
| 22:02.18 | drmessano | wordpad |
| 22:02.24 | SeRi | as in via context |
| 22:02.24 | drmessano | oh |
| 22:02.32 | drmessano | You mean this isn't an editor flame war? |
| 22:02.36 | drmessano | Crap |
| 22:02.40 | SeRi | ROFL |
| 22:02.42 | SeRi | HAHAHA |
| 22:03.03 | gusto | so |
| 22:03.03 | jmetro | psh wordpad.. pros use hex editors |
| 22:03.24 | p3nguin | I use echo and cat. |
| 22:03.25 | gusto | completed some IPsec VPN's |
| 22:03.53 | drmessano | I write all my shell scripts in Microsoft Word |
| 22:04.14 | SeRi | lmao |
| 22:04.17 | SeRi | hahahaha |
| 22:04.23 | SeRi | that was funny for sure |
| 22:04.26 | drmessano | They're all centered |
| 22:04.40 | jmetro | i print out images so i can scan them and email them to people. how else do you email images. |
| 22:04.46 | ghost75 | and make doc in hexeditor |
| 22:05.52 | drmessano | I save all the images from my camera at 4096x2048 for upload to ebay. |
| 22:05.56 | drmessano | Thats not a waste, right? |
| 22:06.49 | ghost75 | (23:05:13) jmetro: i print out images so i can scan them and email them to people. how else do you email images. <- i remember something like this where people were using fax |
| 22:08.22 | jmetro | obv. you print the image and fax it to yourself to make it black and white before scanning it and emailing it, so you dont waste internet ink. |
| 22:08.56 | ghost75 | print mail, put picture on it, scan&print it and then fax it |
| 22:09.50 | ghost75 | to make it b/w is also great to reduce color depth and have smaller attachment size |
| 22:11.09 | *** part/#asterisk doogienz (~doogienz@support.net24.co.nz) |
| 22:15.56 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 22:17.01 | jraddin | thanks @malcolmd |
| 22:17.47 | malcolmd | jraddin: that doesn't help you write a proper one, but it's the most likely cause. good luck |
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