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| 00:57.12 | carbinemonoxide | Hello. I am having a small issue getting asterisk to pick up the call when transfering to voicemail using Google Voice: http://pastie.org/8521784 |
| 00:57.40 | carbinemonoxide | Typo in there... |
| 00:58.09 | carbinemonoxide | http://pastie.org/8521786 |
| 01:03.02 | carbinemonoxide | And... Forgot to have it sendDTMF(1), thanks! |
| 01:03.07 | *** part/#asterisk carbinemonoxide (~none@outonjupiter.com) |
| 01:06.04 | ChannelZ | You might try D(:w1) to make it pause before dialing the 1 |
| 01:07.13 | ChannelZ | Oh nevermind I see what you meant, if you don't answer the * voicemail doesn't work |
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| 03:35.10 | XandriX | im in an odd situation where i have a cisco ip phone with sip firmware and it can make outgoing calls with my current setup but cannot recieve calls at all all incoming calls are essentialy dropped |
| 03:37.17 | *** part/#asterisk stormerider (~stormerid@unaffiliated/stormerider) |
| 03:40.11 | XandriX | its as if the incoming call wasnt routing to the proper extension or wasnt routing ot any extension for that matter |
| 03:42.58 | Penguin | Show us something relevant to your problem. |
| 03:46.15 | XandriX | like my sip and extension.conf ? |
| 03:46.21 | Penguin | Is the phone registered? |
| 03:46.49 | Penguin | What extension is being called? Who is calling it? Does the extension exist? What does the extension do? |
| 03:48.09 | XandriX | the phone appears registerd when i make an outgoing call but outside of that it disapears when i call the did associated to that box it calls and asterisk sees the incoming call with a duration of 1 second cuz it gets dropped and in the to column that i can see it says that my cell phone number is the from and the did is the to |
| 03:49.58 | Penguin | The fact that you can make a call from the phone has little to do with registration. |
| 03:50.17 | XandriX | noted |
| 03:50.26 | Penguin | So let us look at the extensions. |
| 03:50.49 | Penguin | You've indicated the call makes it to asterisk, but then dies 1 second later. |
| 03:51.15 | Penguin | How is the call getting to asterisk? Are you using an ITSP? |
| 03:51.53 | XandriX | i use a cell phone to call my number registerd with voipms and my asterisk server is registerd to the voipms sip server |
| 03:52.03 | XandriX | thats the extent of my knoledge on that part to be honest |
| 03:52.27 | Penguin | We'll get through it. It may take a few minutes, but we'll get there. |
| 03:52.35 | XandriX | that and the phone does not appear in the sip peer list |
| 03:53.01 | Penguin | Is there an entry in the list for it at all? |
| 03:54.47 | Penguin | If you run sip show peers, does the phone not appear at all, or does it appear on the list but show that it is unspecified and unknown/unmonitored? |
| 03:56.44 | XandriX | now it appears cuz i specified the phones ip address in the sip.conf file i was about to test incoming calls and lost signal on my cell phone ... |
| 03:57.44 | XandriX | still not recieving calls |
| 03:58.22 | Penguin | In the peer entry in sip.conf, if you specify the host as dynamic, asterisk requires the phone to register. If you specify the ip address in the host line, the phone must not register. |
| 03:59.09 | XandriX | Lin01/109991 192.168.1.87 N 5060 Unmonitored |
| 03:59.11 | XandriX | voipms/accountname 67.205.74.184 N 5060 Unmonitored |
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| 03:59.19 | Penguin | Usually, phones will register. It makes the most sense to me for phones to register, so use host=dynamic and make sure the phone will register. |
| 03:59.19 | XandriX | it used to be dynamic now i set it to an ip |
| 03:59.32 | XandriX | oh ok |
| 04:00.14 | Penguin | The next problem appears to be that you might not understand how the peer's name is set/determined. |
| 04:00.41 | Penguin | It looks like you set a peer entry [Lin01] in sip.conf. |
| 04:01.04 | XandriX | yes |
| 04:01.07 | Penguin | Lin01 is the name you use on the phone. That will be the username the phone uses. |
| 04:01.59 | Penguin | It looks like you also tried to define a user name within the peer entry, probably by specifying defaultuser or username. |
| 04:02.07 | Penguin | For a phone, that is incorrect. |
| 04:03.09 | XandriX | how would i do it for a phone ? |
| 04:03.16 | Penguin | defaultuser (username in older asterisks) is the username that asterisk is to send when configuring asterisk as a peer for another system. |
| 04:03.49 | Penguin | The name in the square brackets is the phone's username. If you created [Lin01], Lin01 is the name you configure as the username in the phone's config settings. |
| 04:04.08 | XandriX | and what about secret ? |
| 04:04.22 | Penguin | secret is the password required by the phone. |
| 04:04.36 | XandriX | kk so i remove the defaultuser entry |
| 04:04.52 | Penguin | If you set secret=12345 in sip.conf under [Lin01], the phone is to be configured with username Lin01 and password 12345. |
| 04:06.05 | Penguin | And then to send a call to that phone, you will Dial SIP/Lin01 in the extension. |
| 04:06.27 | Penguin | e.g., Dial(SIP/Lin01,32) |
| 04:07.39 | Penguin | If you have trouble with the phone registering after you reconfigure the username and password, check to be sure the phone is set to send registration. Phones can be set to register or not register. |
| 04:08.29 | Penguin | I have to go for about 20 minutes. I'll be back. |
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| 04:15.13 | *** join/#asterisk smirker (~x@CPE-58-169-177-185.lns5.cha.bigpond.net.au) |
| 04:15.53 | smirker | Howdy. With Asterisk 11, is there an easy way to find the number of concurrent channels being used? And possible, the maximum concurrent channels that have been used? |
| 04:18.31 | XandriX | phone does not seem to be registering |
| 04:19.40 | XandriX | it doesnt appear in peers list but in channel list it does |
| 04:20.59 | smirker | Nevermind first question ;o, core show channels count. Any solution to max concurrent channels, or would I have to set up some type of logging/history? |
| 04:25.19 | Penguin | I've written dial plan to show maximum calls ever. |
| 04:25.28 | Penguin | I can share that with you, smirker. |
| 04:25.51 | Penguin | xandrix: Did you ensure that the phone is set to send registration? What is the phone model? |
| 04:27.41 | XandriX | the register to proxy setting is set to yes on the phone |
| 04:30.50 | XandriX | oh wait not it wasnt anymore holdon leme restart the phone to make sure the settings are applied properly |
| 04:31.46 | XandriX | its not done booting up yet but now its showing up in the peer list |
| 04:31.49 | XandriX | progress :) |
| 04:34.36 | XandriX | yeah now the phone appears in the peer list and in the channel list |
| 04:36.13 | XandriX | Penguin: what whouls be my next step / |
| 04:36.17 | XandriX | * ? |
| 04:36.50 | Penguin | Did you already create an extension which dials the phone? |
| 04:37.24 | XandriX | i have the phone register to the line maybe i did not create a proper entry in the extensions.conf |
| 04:38.21 | XandriX | exten => mydidnumber,1,Answer() and exten => 109991,1,Answer() |
| 04:38.40 | Penguin | Looks all wrong to me. |
| 04:39.10 | Penguin | You most likely do not need to Answer() the call before you pick up the phone. So we'll do away with that... |
| 04:39.37 | Penguin | When someone calls your DID, do you want it to start ringing your phone straight away? |
| 04:39.44 | XandriX | yeah |
| 04:40.05 | Penguin | What did you end up naming the phone? It was Lin01 earlier. |
| 04:40.18 | XandriX | i ended up naming it 109991 |
| 04:40.36 | Penguin | How long do you want it to ring before moving on to the next step? |
| 04:40.51 | XandriX | 7 times |
| 04:41.18 | Penguin | exten => yourDIDnumber,1,Dial(SIP/109991,44) |
| 04:42.08 | Penguin | Actually, I may have calculated too many seconds for 7 rings. |
| 04:42.42 | Penguin | Yeah, change that to 40. |
| 04:43.07 | Penguin | exten => yourDIDnumber,1,Dial(SIP/109991,40) |
| 04:43.15 | XandriX | aha kk |
| 04:43.31 | XandriX | but thx that did the trick things brings me to other things i have yet to understand |
| 04:43.46 | XandriX | setting up a local voicemail and setup call forwarding to an external number |
| 04:44.48 | Penguin | Did you look at The Book? |
| 04:44.50 | Penguin | ~book |
| 04:44.50 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 04:45.30 | XandriX | thx for the link :) and thx for letting me get incoming calls to work |
| 04:45.47 | Penguin | I'll summarize what you need to do for voicemail. |
| 04:45.54 | Penguin | Create a mail box in voicemail.conf |
| 04:46.32 | Penguin | After the Dial(), you'll execute the Voicemail() application. |
| 04:46.55 | Penguin | To get MWI to work, associate your mailbox with your peer in sip.conf using the mailbox configuration setting. |
| 04:47.21 | Penguin | I prefer to create a mail box with the same extension number used to reach my phone. |
| 04:48.33 | Penguin | VoiceMail(${VMBOX}@default,u); where VMBOX is your mailbox that you created |
| 04:49.12 | Penguin | I actually set that variable in my extension routine, but you can do it statically. |
| 04:50.05 | XandriX | im still reading on howto create it in voicemail.conf |
| 04:50.26 | Penguin | Look at the samples in the sample voicemail.conf file. |
| 04:50.36 | XandriX | going through them now |
| 04:50.51 | Penguin | It is probably also in the book. |
| 04:53.09 | XandriX | so something in my case like 109991 => 10991,Example Mailbox,root@localhost |
| 04:53.44 | Penguin | I would expect something similar, yes. |
| 04:54.56 | XandriX | VoiceMail(${VMBOX}@default,u); would go in my sip peer config ? |
| 04:55.12 | Penguin | no |
| 04:55.45 | Penguin | VoiceMail() is a dialplan application. Applications are executed by extensions. |
| 04:57.30 | Penguin | To get voice mail, the only things necessary are 1) configure a mail box in voicemail.conf, and 2) execute the VoiceMail() application in dialplan using the mail box created. |
| 04:57.54 | XandriX | so after the line you gave me to get it to ring essentialy |
| 04:57.55 | Penguin | To get MWI to work, that's when you add the mailbox= line to your sip.conf peer entry. |
| 04:58.19 | Penguin | Yes, exactly. |
| 04:58.34 | XandriX | and i replace logicaly VMBOX by 109991 |
| 04:58.48 | Penguin | If your mail box is 109991, yes. |
| 04:58.55 | XandriX | cool ok so ive got that |
| 04:59.10 | XandriX | now if i could only get signal on my cell phone to test my voice mail box |
| 04:59.17 | Penguin | I literally have the variable VMBOX in my dial plan, but I also set the variable earlier. |
| 05:00.53 | XandriX | and what is this MWI you speak of |
| 05:01.08 | Penguin | I do things like Set(_VMBOX=${EXTEN}), in the most simple terms. |
| 05:01.12 | Penguin | ~mwi |
| 05:01.12 | infobot | Message Waiting Indicator |
| 05:01.19 | Penguin | aka, the red light on your phone |
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| 05:04.08 | XandriX | yeah i set that up to but hmmm when i call my sip number from my cell it rings for a while but does not seem to hit a voicemailbox |
| 05:05.02 | Penguin | Did you remember to run "voicemail reload" after changing voicemail.conf and "dialplan reload" after changing extensions.conf? |
| 05:05.31 | XandriX | i just restarted asterisk after doing the changes |
| 05:05.39 | Penguin | That's silly. |
| 05:05.46 | Penguin | But that'll do it. |
| 05:09.15 | Penguin | If you aren't sure what is going on, increase core verbosity and make the call again. core set verbose 3 |
| 05:10.20 | XandriX | [voipms-inbound] |
| 05:10.22 | XandriX | exten => mydid,1,Dial(SIP/109991,40); |
| 05:10.24 | XandriX | VoiceMail(${109991}@default,u); |
| 05:10.32 | XandriX | or do i not need the ; or did i not do that right |
| 05:10.34 | Penguin | Fail. |
| 05:10.37 | XandriX | cuz it rings and then dies |
| 05:11.03 | Penguin | If you look at the CLI when you run dialplan reload, you'll see where the problem lies. |
| 05:12.02 | Penguin | The semi-colon is not necessary nor required. It is typically used when a comment will follow a particular line. I, on the other hand, terminate every line with a semi-colon just like is required in other programming languages. |
| 05:13.27 | Penguin | And in addition to the syntax failure, your mailbox is invalid. You are telling VoiceMail() to look at a variable by the name of 109991, not the mailbox named 109991. |
| 05:16.06 | Penguin | Do you follow me? |
| 05:16.58 | XandriX | so more like exten => mydid,1,Dial(SIP/109991,40) |
| 05:17.00 | XandriX | VoiceMail(109991@default,u) |
| 05:17.21 | Penguin | That takes care of that part of it, but your syntax is still a fail. Run dialplan reload so you can see why it's wrong. |
| 05:18.33 | XandriX | errr i am not sure how to run dialplan reload this is asterisk running on pfsense |
| 05:18.50 | Penguin | Are you not connecting to the asterisk CLI? |
| 05:20.23 | XandriX | No '=' (equal sign) in line 20 of /conf/asterisk/extensions.conf |
| 05:21.06 | Penguin | There's your hint. |
| 05:22.52 | Penguin | Here's another hint: |
| 05:23.26 | Penguin | Every extension line you write needs to contain the extension and a priority. |
| 05:25.22 | XandriX | the priority order 1 being higher than 2 or oposite ? |
| 05:26.06 | Penguin | Priority starts at 1 and increases either with explicit numbers or by using 'n' as the priority to indicate "next" in sequence. |
| 05:26.42 | Penguin | I prefer n over the number because it makes later edits much easier. |
| 05:26.57 | XandriX | is confused |
| 05:27.03 | Penguin | ~book |
| 05:27.03 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 05:27.12 | Penguin | Look at the extensions section. |
| 05:27.44 | Penguin | Or if you can't bother to look in the book that contains all of the information, look at a quick asterisk primer... |
| 05:27.47 | Penguin | ~primer |
| 05:27.48 | infobot | methinks primer is http://burner.com/asterisk-primer |
| 05:29.20 | XandriX | Penguin: thx for that link aswell |
| 05:29.27 | XandriX | ive just successfully left myself a message |
| 05:29.36 | XandriX | just cant currently go and listen to it |
| 05:29.50 | Penguin | You'll have to create some other extensions to make that possible. |
| 05:29.55 | XandriX | but the mwi seems to be working it poped up a flashing envelope on my phone |
| 05:31.42 | XandriX | oh so an extension that from my phone i dial to access my voicemailbox |
| 05:32.47 | Penguin | You can create an extension that will execute VoiceMailMain(), or you can create an extension that allows you to press * when listening to the outgoing message to take you to VoiceMailMain(). |
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| 05:35.37 | Penguin | I use *86, which is *VM, to take me to my own voicemail main menu. |
| 05:36.29 | XandriX | oh well i created an extension 199 using VoiceMailMain() but it prompts me for a login i give it the voicemailid and then a password wich i dont know what it is to be honest |
| 05:36.56 | Penguin | You set it in voicemail.conf earlier... how can you not know it? |
| 05:39.00 | XandriX | i had a typo in it |
| 05:39.10 | XandriX | you say u set it to *86 |
| 05:39.22 | XandriX | i would have an easier time remembering that i think |
| 05:39.46 | Penguin | Verizon Wireless uses *VM to call voicemail. That is *86. |
| 05:39.56 | Penguin | It made sense to use it. |
| 05:40.48 | XandriX | cool it works is there a way i can get it to autodetect my voicemailbox id ? |
| 05:41.24 | Penguin | Of course. That's what my *86 does. I use either accountcode or callerid number, or validate one with the other. |
| 05:41.54 | XandriX | exten => *86,1,VoicemailMain(s${CALLERIDNUM}) |
| 05:42.04 | Penguin | In simplest terms, VoiceMailMain(${CALLERID(num)}@default) |
| 05:42.14 | XandriX | woops |
| 05:42.37 | Penguin | I'm not sure, but you may need to Answer() the channel first. |
| 05:43.09 | XandriX | nope that work straight off the bat |
| 05:43.13 | XandriX | just prompted for password :) |
| 05:43.27 | Penguin | I have Answer(500) before my VoiceMailMain(), so I must have used that to add some delay after starting RTP. |
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| 05:44.17 | XandriX | Penguin: all of your help is much apreciated :) |
| 05:44.42 | Penguin | Yeah, that was probably why I did that. I was probably hearing "ssword" or "assword" when calling *86. |
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| 05:54.19 | Penguin | Set(VMBOX=${IF($[${EXISTS(${CDR(accountcode)})}]?${CDR(accountcode)}:${CALLERID(num)})}) |
| 05:54.26 | Penguin | Answer(500) |
| 05:54.32 | Penguin | VoiceMailMain(${VMBOX}@default) |
| 05:56.11 | Penguin | I'm thinking about doing that a little different in the future. |
| 05:57.22 | Penguin | I'll use CALLERID(num) and validate it against accountcode, and if it validates I will skip the password. If it does not validate, go ahead and require password. |
| 05:57.47 | Penguin | Maybe. I don't know yet. It works like it is. |
| 06:02.34 | Penguin | I'm out. |
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| 08:21.48 | bitfury | hey guys, I have an outbound google voice trunk set up and working for local calling. However, I can't seem to get international calling working through this trunk |
| 08:22.31 | bitfury | is it possible to route international calls through a GV trunk ? |
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| 08:49.35 | ChannelZ | bitfury: I can't imagine why not |
| 08:51.14 | bitfury | ChannelZ: same here, when I try dialing out I get "the number is not answering" |
| 08:51.47 | bitfury | my GV account balance is still 0.00 though.. they haven't credited it yet |
| 08:51.51 | bitfury | that might be why? |
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| 08:53.13 | ChannelZ | Probably a pretty good one yes |
| 08:53.22 | ChannelZ | No pay, no play. |
| 08:54.39 | bitfury | hehe |
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| 10:52.47 | nextime | hi all, asterisk 11.5 from debian package ( on debian sid ): it must receive calls coming from internet and redirect calls to an internal asterisk server, nothing more, so, it doesn't have any sip account registered, it just receive guest calls on a context and on this context make a Dial() to an internal sip account proxying the call. It work great with "common" sip clients. |
| 10:53.22 | nextime | now i'm trying it with the demo of jssipo ( tryit.jssip.net ), and the call come correctly but it try to use avfp codec |
| 10:53.52 | nextime | i have avfp=yes in my sip.conf ( not on peer based, as i don't have peers, in the global section ) |
| 10:54.12 | nextime | but when i try to call it i see on the console: |
| 10:54.14 | nextime | [2013-12-02 11:41:17] WARNING[1939][C-00000004]: chan_sip.c:10107 process_sdp: Received SAVPF profle in audio offer but AVPF is not enabled: audio 45307 RTP/SAVPF 111 103 104 0 8 107 106 105 13 126 |
| 10:54.29 | nextime | so, it seems that avfp=yes make no effects at all |
| 10:54.45 | nextime | is there something i can do about this issue? |
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| 11:04.34 | *** join/#asterisk rox (~rox@mx.abraxas.si) |
| 11:05.00 | rox | good morning, can anybody tell me, is how can i get hinting logged? |
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| 13:23.49 | BeachBall | dont all say hi at once |
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| 13:25.15 | nextime | o/ |
| 13:25.41 | BeachBall | :/ |
| 13:25.55 | BeachBall | (~._.~) |
| 13:25.58 | BeachBall | ()-*-() |
| 13:26.00 | BeachBall | (_)-(_) |
| 13:26.07 | coppice | h |
| 13:26.41 | nextime | can anyone try to make a call to sip:casa@casa.nexlab.it to see if it is working? |
| 13:26.58 | coppice | i |
| 13:27.39 | [TK]D-Fender | coppice++ |
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| 13:34.04 | XandriX | penguin if for some reason ur ever in the montreal area and wana grab a beer give me a holla |
| 13:34.47 | BeachBall | XandriX ur in montreal? |
| 13:35.26 | BeachBall | might get a free beer |
| 13:35.27 | BeachBall | :} |
| 13:35.41 | BeachBall | doesn't drink beer :{ |
| 13:38.51 | [TK]D-Fender | BeachBall: Yes |
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| 13:47.10 | rox | I have the following setup: 2 SIP telephones registered on Asterisk, one is Polycom 430, the other is Polycom 450, both have presence configured, and both have each other configured for Buddy Watch, and they are in the same callgroup/pivkupgroup. My problem is, when one telephone is called, the other should light up its led, but it doesn't. |
| 13:47.18 | rox | does anybody have an idea, what could be wrong? |
| 13:47.54 | [TK]D-Fender | rox: And the part where you set up the dialplan for this? |
| 13:48.13 | rox | oh, and Asterisk is displaying each other's subscriptions, and when calls are in progress, the subscription status turns from "idle" to "in use" |
| 13:48.20 | [TK]D-Fender | rox: callgroup/pickupgroup is unrelated |
| 13:48.38 | rox | [TK]D-Fender: in dialplan both are set for hint |
| 13:49.29 | rox | the phones are Station205 and Station306, in dialplan each have the hint line: exten => 306,hint,SIP/Station306 |
| 13:49.59 | [TK]D-Fender | rox: place a call and show the full CLI output with SIP debug enabled |
| 13:51.34 | nextime | uhmm |
| 13:51.49 | nextime | it seems that avpf=yes doesn't work in [general] for guesr calls |
| 13:52.47 | nextime | so, is there a way to define a "catch all" user sip account where i can put avpf=yes for guest calls? |
| 13:54.55 | rox | [TK]D-Fender: the call output: I called from third phone to 306, logs say 205 should have been notified: http://pastebin.com/ic084TVP |
| 13:56.32 | [TK]D-Fender | rox: SIP DEBUG <--- |
| 13:57.01 | rox | debug of the called phone or the phone that was supposed to be notified? |
| 13:57.06 | rox | [TK]D-Fender: or both? |
| 13:57.37 | [TK]D-Fender | both |
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| 14:00.42 | BeachBall | Ziggs!!! |
| 14:00.48 | BeachBall | do you play LoL? |
| 14:00.49 | BeachBall | ;D |
| 14:03.00 | rox | [TK]D-Fender: this is the notify debug: http://pastebin.com/fxcB8gH3 |
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| 14:06.03 | [TK]D-Fender | rox: Sure does look fine there... odd... |
| 14:06.23 | rox | [TK]D-Fender: http://pastebin.com/nstm0HUy and this is teh call debug |
| 14:06.57 | [TK]D-Fender | rox: your firmware is a little behind.. I'd probably update that right away |
| 14:07.03 | rox | [TK]D-Fender: it looks fine to me too, the users say there is no LED on the phones, and i'm frustrated like hell, because it's a remote site and I can't verify it |
| 14:07.29 | [TK]D-Fender | rox: I'm not sure if the 430 actually lights the LED for it.. do you see the indicator on-scren change? |
| 14:08.03 | [TK]D-Fender | rox: I know the 30X didn't light... and I'm not sure on the 32X/33X |
| 14:10.00 | rox | [TK]D-Fender: 3.2 shows leds, I have setups on other sites, i just can't figure out why this one isn't functioning |
| 14:12.10 | rox | [TK]D-Fender: so, to sum it up: in sip.cfg presence is on, in phone.cfg presence is on, in directory file bw is on, in dialplan hint is on, is there anything i'm missing? |
| 14:12.43 | [TK]D-Fender | rox: If that debug is at is appears.. nothing on the * side... |
| 14:13.26 | rox | [TK]D-Fender: the debug is just copy/paste, so firmware upgrade is the last thing to do? or telephone configuration perhaps? |
| 14:13.40 | rox | i guess something has to be suppressing this |
| 14:15.20 | [TK]D-Fender | rox: Again, NTOHIGN on screen at all? |
| 14:16.13 | rox | that is what they are claiming |
| 14:16.18 | *** join/#asterisk serafie (~erin@nat/digium/x-ukvlfcuyxedsrhbs) |
| 14:16.44 | nextime | uhmm |
| 14:16.44 | rox | [TK]D-Fender: the users are telling me "no changes, phone is completely idle" |
| 14:17.03 | nextime | so, i can't really find a way to make a guest sip user use avpf=yes |
| 14:17.10 | nextime | this is sad. |
| 14:17.22 | [TK]D-Fender | rox: so image next to the spedd-dial for it is unchanged? |
| 14:19.40 | *** join/#asterisk delete (~delete@fedora/delete) |
| 14:20.37 | delete | hi, I have troubles to build dahdi modules after kernel update on Centos 6 |
| 14:20.39 | delete | http://pastebin.ca/2488121 |
| 14:22.48 | boom^time | Hello, every so often with an AMI call origination I won't receive a hangup event. It's really frustrating. I was wondering if anyone else ran into this bug? |
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| 14:38.10 | tomodachi | delete: are you using your distributions dahdi sources, or the ones from digium? |
| 14:41.37 | delete | the ones from digium |
| 14:42.14 | *** join/#asterisk mbrit (~mbrit@186.120.97.194) |
| 14:42.59 | delete | the ones from here http://www.asterisk.org/downloads/dahdi |
| 14:44.54 | *** part/#asterisk sereal (~sereal@unaffiliated/sereal) |
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| 14:46.09 | *** join/#asterisk boratynskikamil (5311a3f3@gateway/web/freenode/ip.83.17.163.243) |
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| 14:46.59 | boratynskikamil | Hello. |
| 14:47.04 | boratynskikamil | I have one small question. |
| 14:47.37 | boratynskikamil | Let's assume I have 2 Asterisks connected via LAN. Is it possible to call between them without using external connection? |
| 14:47.56 | [TK]D-Fender | yes |
| 14:48.16 | boratynskikamil | [TK]D-Fender: Any suggestion how to do it? |
| 14:48.28 | [TK]D-Fender | boratynskikamil: However you feel like |
| 14:49.12 | boratynskikamil | Suggestion? |
| 14:49.20 | [TK]D-Fender | boratynskikamil: Setup whatever kind of channel you want between them |
| 14:49.33 | [TK]D-Fender | boratynskikamil: how about SIP. That seems to be popular... |
| 14:49.49 | boratynskikamil | I use SIP to connect to Asterisk as client. |
| 14:49.50 | Penguin | Or IAX2... which is an inter-asterisk type of thing. |
| 14:50.10 | [TK]D-Fender | Penguin: Too obvious.... |
| 14:50.17 | Penguin | Good point. |
| 14:50.41 | Penguin | I forgot this is supposed to be a difficult decision. |
| 14:50.43 | boratynskikamil | But main question is how to connect from client1@asterisk1 to client2@asterisk2? |
| 14:50.52 | XandriX | BeachBall: yes i am in montreal |
| 14:51.04 | [TK]D-Fender | boratynskikamil: DIAL .. just like everything else |
| 14:51.06 | XandriX | if u dont drink beer i can get your i dunno free scotch or juice ? :P |
| 14:51.07 | BeachBall | how can 1 bitcoin be worth 1000$? |
| 14:51.26 | [TK]D-Fender | boratynskikamil: Asterisk talking to another asterisk is no different than Asterisk talking to a phone. |
| 14:51.47 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:51.47 | [TK]D-Fender | boratynskikamil: In sip.conf you make the same kind of peer and you DIAL just the same. |
| 14:51.47 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:51.48 | boratynskikamil | So I am able to call IP@Asterisk1? |
| 14:51.54 | boratynskikamil | O.O |
| 14:51.55 | boratynskikamil | Great. |
| 14:51.57 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:51.58 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:52.04 | [TK]D-Fender | boratynskikamil: make a peer like you do with any other provider and dial. |
| 14:52.23 | XandriX | Penguin: thx again for last night :) |
| 14:52.34 | [TK]D-Fender | boratynskikamil: Never just dial by IP in the dial command. Alqways make a proper peer that specifies codecs, secret, etc. |
| 14:52.41 | boratynskikamil | So, as far as I understand I have to create exension. |
| 14:53.25 | [TK]D-Fender | boratynskikamil: Dialing doesn't just happen on its own... |
| 14:53.43 | boratynskikamil | What do you mean? |
| 14:53.48 | [TK]D-Fender | boratynskikamil: No different than any ITSP setup example you should ever have seen |
| 14:53.59 | [TK]D-Fender | boratynskikamil: extension = line in extensions.conf |
| 14:54.23 | boratynskikamil | Mhm. |
| 14:54.26 | [TK]D-Fender | boratynskikamil: you know.. DIALPLAN. the thing that says "Oh they dialed 123456 that means I should send it to the OTHER Asterisk server...." |
| 14:54.44 | boratynskikamil | Yes, yes. |
| 14:54.47 | *** join/#asterisk smirker (Smirker@CPE-58-169-177-185.lns5.cha.bigpond.net.au) |
| 14:54.58 | boratynskikamil | But nowadays I have one extension called outgoing. |
| 14:55.08 | boratynskikamil | And it uses 1st channel in dahdi. |
| 14:55.19 | boratynskikamil | And uses all connections, I mean _X. |
| 14:55.28 | [TK]D-Fender | boratynskikamil: Make better ones now. |
| 14:55.42 | [TK]D-Fender | boratynskikamil: And consider your contexts |
| 14:55.46 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-21-140.dynamic.qsc.de) |
| 14:56.08 | boratynskikamil | Mhm. |
| 14:58.55 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
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| 15:10.13 | boratynskikamil | One more question. |
| 15:10.27 | boratynskikamil | Could you be so kind and advice me some GSM card for Asterisk? |
| 15:10.51 | *** join/#asterisk chris_n (~Chris@koha/developer/chris-n) |
| 15:13.28 | *** join/#asterisk saliak (~saliak@ip72-195-155-142.ri.ri.cox.net) |
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| 15:23.25 | XandriX | BeachBall: what area of mtl you in / from ? |
| 15:28.48 | *** join/#asterisk CeBe (~CeBe@port-92-206-89-192.dynamic.qsc.de) |
| 15:33.05 | saliak | I keep getting a "WARNING[18096]: chan_sip.c:3370 __sip_xmit: sip_xmit of 0xb6e56460 (len 517) to 192.168.1.170:5060 returned -1: Operation not permitted" warning that seems to be related to a phone not registering with my server. sip debug at http://pastebin.com/RYLm7ssR |
| 15:36.02 | [TK]D-Fender | saliak: And what do you have in the peer for that? |
| 15:40.48 | boom^time | Does anyone with AMI experience know if adding a special hangup extension which simply ran Hungup, ie exten => h,1,Hangup() would fix the occasional lack of a Hangup AMI event? |
| 15:41.33 | [TK]D-Fender | boom^time: 'h" doesn't create another |
| 15:42.21 | boom^time | I had a feeling that was the case. The aggravating part is that this problem happens so infrequently it's hard to try and debug it. |
| 15:45.27 | saliak | [TK]D-Fender: what you mean by "have in the peer"? |
| 15:45.54 | [TK]D-Fender | saliak: ... in your entry for them for sip.conf |
| 15:46.34 | *** join/#asterisk cyberfab007 (cyberrfab0@cpe-72-231-250-192.buffalo.res.rr.com) |
| 15:46.47 | cyberfab007 | ok today write my own custome dial plan ! Yay! |
| 15:47.10 | saliak | [TK]D-Fender: sorry. http://pastebin.com/sXZLu1Cs. added to top of pastebin |
| 15:47.41 | XandriX | anyone here use voipms as there provider ? |
| 15:47.56 | [TK]D-Fender | saliak: "sip show peer 8002" |
| 15:48.20 | [TK]D-Fender | XandriX: probably several..... |
| 15:49.29 | saliak | [TK]D-Fender: ok. http://pastebin.com/gHGdcDwk |
| 15:51.27 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 15:51.59 | Penguin | xandrix: yes |
| 15:52.05 | XandriX | trying to figure out howto get the check balance thing to work properly |
| 15:52.21 | Penguin | Via key pad? |
| 15:52.22 | [TK]D-Fender | [2013-12-02 10:28:05] WARNING[18096]: chan_sip.c:3370 __sip_xmit: sip_xmit of 0xb6e56460 (len 517) to 192.168.1.170:5060 returned -1: Operation not permitted |
| 15:52.30 | XandriX | yeah there *225 thing |
| 15:52.38 | Penguin | It's easy enough to log in and look in the customer portal if that doesn't work. |
| 15:52.41 | [TK]D-Fender | Wondering on this... firwall block stopping the challenge from being sent perhaps? |
| 15:52.59 | XandriX | Penguin: this account is loaned to me there for i do not have access to said user panel |
| 15:55.07 | XandriX | but the person in question did set me up to have access to balance via the *225 function |
| 15:55.51 | Penguin | Did you create an extension that sends *225 to the provider? |
| 15:55.58 | Penguin | My guess is no. |
| 15:56.18 | XandriX | no cuz i was unsure of the syntax to get it to tie in to that function on voipms's side |
| 15:56.30 | saliak | [TK]D-Fender: that's possible. weird thing is that it did work at one point, and i have another sip phone on the same network that works |
| 15:56.45 | Penguin | It's just a number. It requires no special attention. |
| 15:56.50 | Penguin | It's a regular phone call. |
| 15:59.00 | XandriX | ooh |
| 15:59.07 | XandriX | tild |
| 16:03.25 | XandriX | so i will probably fail again but something like this ? exten => *225,1,Dial(SIP/109991,40) |
| 16:04.40 | XandriX | err no |
| 16:04.44 | XandriX | thats not what i meant to paste |
| 16:06.01 | Penguin | Paste something different, then. |
| 16:09.39 | XandriX | exten => *225,1,Dial(SIP/*225@voipms) is what i meant to paste clipboard stupidity on my part sorry |
| 16:10.33 | Penguin | *225@voipms is wrong. voipms is a defined peer in your sip.conf. |
| 16:10.53 | Penguin | Dial(SIP/voipms/*225) |
| 16:11.23 | Penguin | Or, since you are using extension *225, you can use Dial(SIP/voipms/${EXTEN}). |
| 16:11.27 | XandriX | it works but i will correct that thx for the heads up |
| 16:12.01 | *** join/#asterisk xytis (~xytis@5.20.223.100) |
| 16:12.14 | Penguin | Dialing something@host is for URI dialing. If you have a peer configured, you don't need to dial by URI. And voipms isn't a full host name anyway. |
| 16:14.10 | XandriX | fuck my inet at home just went down -_- |
| 16:19.37 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:19.37 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:22.12 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:22.12 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:22.23 | XandriX | pfahahaha so they changed my profile on my inet connection to upgrade me and accidentaly put my account on hold |
| 16:23.19 | XandriX | now there is one feature i need to get working on asterisk that a friend of mine had a while ago you call the did number and if u send it say a 4 or a 5 it gives you the machines current ip address that was cool hehe |
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| 16:46.07 | Kobaz | anyone using sangoma/wanpipe on a fairly new kernel/dahdi? |
| 16:46.18 | Kobaz | i'm having trouble building 7.0.8 |
| 16:46.31 | Kobaz | fun kernel api changes |
| 16:48.22 | *** join/#asterisk hardwire (~hardwire@cl-101.anc-01.us.sixxs.net) |
| 16:49.56 | *** join/#asterisk xytis (~xytis@88-223-36-62.meganet.lt) |
| 16:52.27 | Penguin | xandrix: It's all fairly easy to do in the dial plan or you can script it in php and run it with AGI(). But if your asterisk is not on a public IP address that changes, there is no point in doing it. |
| 16:53.47 | *** join/#asterisk serafie1 (~erin@nat/digium/x-tjdoaeahdwradyqn) |
| 16:54.06 | XandriX | Penguin: asterisk is currently running straight off my asterisk box i just want to make so i can call the number and dial an extension and it gives me the machines current wan address |
| 16:54.57 | Penguin | That wouldn't be too difficult to write up. |
| 16:55.31 | *** join/#asterisk xytis (~xytis@88-223-36-62.meganet.lt) |
| 16:55.40 | XandriX | cool |
| 16:55.53 | Penguin | I might even do that later just for the hell of it. It won't serve me any purpose since my asterisks are on private IP addresses, but I might write it up just for the sake of doing it. |
| 16:56.18 | XandriX | Penguin: keep me posted on that :) |
| 17:06.22 | *** part/#asterisk Brixius (~kmurphy@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 17:08.09 | *** join/#asterisk serafie (~erin@nat/digium/x-yxvsdtcdfmfcldeq) |
| 17:10.38 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.38) |
| 17:12.19 | Penguin | xandrix: Done. |
| 17:12.31 | Penguin | xandrix: I couldn't wait until later; I had to do it right now. |
| 17:13.50 | Penguin | http://pastebin.com/v3gQufkt |
| 17:15.22 | XandriX | Penguin: <3 no homo |
| 17:17.39 | *** join/#asterisk xytis (~xytis@88-223-36-62.meganet.lt) |
| 17:20.40 | Penguin | I should consider stripping off the /24 to make it more friendly. |
| 17:20.49 | *** join/#asterisk Greenlight (wluke@cpc1-dund9-0-0-cust142.16-4.cable.virginm.net) |
| 17:21.38 | Greenlight | Why is it so hard to get Queue in Asterisk to behave like a queue on a traditional phone system |
| 17:21.52 | Penguin | Asterisk isn't a traditional phone system. |
| 17:21.57 | Penguin | ^reason |
| 17:22.22 | Greenlight | Yea, but the way it's Queue system works is lacking |
| 17:22.28 | Greenlight | It's hugely frustrating |
| 17:22.38 | Penguin | I'd imagine if you state what behavior you are trying to get out of it, someone might be able to provide some tips on making it work that way. |
| 17:23.14 | Greenlight | Well it's this whole timeout of members in the queue before we retry them |
| 17:23.38 | Greenlight | If a caller is in the queue, I want to just keep the members ringing |
| 17:23.52 | Greenlight | And if a currently busy member becomes available, I want their phone to ring. |
| 17:24.50 | Greenlight | Unless I'm totally missing something, it seems with Queue, we have to set a low agent timeout or be calling a 2nd line to get this behaviour |
| 17:25.26 | Greenlight | Problem with a low agent timeout (say 10 seconds) is that it takes about 1 second for it to retry all the callers, and if a member picks up their phone, no one is there... |
| 17:26.21 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 17:28.32 | Greenlight | I mean I just don't see why we have to stop a members phone from ringing it there's a caller still queued. It seems backwards, and I've no explanation to offer to customers when they ask why it behaves like this. |
| 17:33.31 | XandriX | Penguin: im gonna need to decipher a fiew things from your script hehe |
| 17:34.23 | [TK]D-Fender | [12:28]GreenlightI mean I just don't see why we have to stop a members phone from ringing it there's a caller still queued. It seems backwards, and I've no explanation to offer to customers when they ask why it behaves like this. <- because that's how it works |
| 17:34.44 | [TK]D-Fender | Greenlight: You don't have to justify the design, only explain its behaviour |
| 17:35.02 | XandriX | [Dec 2 12:34:19] WARNING[93787]: app_playback.c:475 playback_exec: ast_streamfile failed on SIP/109991-00000001 for silence/1&your&address&is&silence/1 |
| 17:35.09 | *** part/#asterisk matt-moretalk (~matthew@88.96.27.150) |
| 17:35.12 | Greenlight | [TK]D-Fender: I realise that's how it works, but I get complaints for almost all my customers that use the Queue module. These are small business nothing out of the ordinary |
| 17:35.34 | [TK]D-Fender | Greenlight: They are nagging. You should get them to stop doing that :) |
| 17:35.35 | Greenlight | It's counter intuative |
| 17:35.52 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 17:36.04 | Greenlight | Thing is, I see their point. They pick up the phone, and it just says "disconnected". Only to ring half a second later. |
| 17:36.07 | [TK]D-Fender | Greenlight: * doesn't poll all members constantly to add. think of it as "ringing campaigns" |
| 17:36.31 | [TK]D-Fender | [12:36]GreenlightThing is, I see their point. They pick up the phone, and it just says "disconnected". Only to ring half a second later. <- this is ANY call that is ringing and ends as they are reaching for it... Too bad. |
| 17:36.36 | Greenlight | I don't want it to poll all membrers, but when a channel clears, why can't it then check if that channel is a MEMBER in any queues... |
| 17:37.01 | Greenlight | <[TK]D-Fender> [12:36]GreenlightThing is, I see their point. They pick up the phone, and it just says "disconnected". Only to ring half a second later. <- this is ANY call that is ringing and ends as they are reaching for it... Too bad. <-- Buuuut the call hasn't ENDED..., |
| 17:37.21 | Penguin | xandrix: You're missing some sound files. |
| 17:37.24 | [TK]D-Fender | Greenlight: When I add LOCAL channels.... it has no psychic way of knowing that the SIP device that just ended a call would be the one called by it |
| 17:37.46 | Greenlight | [TK]D-Fender: WEll you could use hints |
| 17:38.12 | XandriX | Penguin: probs and also well on pfsense i apparently cannot check my ip that way |
| 17:38.20 | XandriX | cuz apparently the command ip does not exist wich i doubt |
| 17:38.38 | Greenlight | [TK]D-Fender: I mean, what would make sense is just "rechecking" so see if any members have now become available. Rather than restarting all members calls. |
| 17:38.41 | Penguin | I didn't write it for you... I wrote it for myself. And I use Linux. |
| 17:39.13 | Penguin | xandrix: But you can easily change the commands to what FreeBSD does use. |
| 17:39.17 | XandriX | Penguin: on most of my machines i use lin |
| 17:39.26 | XandriX | and thats what i am about to do |
| 17:39.32 | Greenlight | Most of the times I use "timeout" and "retry", what I really mean is that I want to recheck for any new possible members. |
| 17:39.48 | XandriX | Penguin: about the missing sound files tho what would be your solution to that ? |
| 17:40.00 | Penguin | I'd install the sound packs. |
| 17:40.02 | Greenlight | But currently only way to do that is to HANGUP all calls to get members, and restart them. That's backwards. |
| 17:40.53 | XandriX | Penguin: ill go read up on where they are stored and so forth since in my case its an odd installation on pfsense |
| 17:41.26 | Greenlight | I fully understand that this is *how it works*, what I don't see is WHY. |
| 17:42.02 | Greenlight | Queuing seems like such a core feature, and I see so many complaints from both online message posts, and my own customers |
| 17:42.59 | Greenlight | On my larger installs, where they use our CRM software that integrates via AMI, we don't use Asterisk's built in queuing at all, and I wrote my own queuing logic |
| 17:44.28 | XandriX | Penguin: your script gave me a good idea on how to do it just need to set up the proper awk and cut syntax to get my ip and i should be good that and install the sound packs |
| 17:44.44 | Penguin | If you use awk, why would you also need to use cut? |
| 17:45.24 | Penguin | You'll notice I didn't use cut. |
| 17:45.36 | XandriX | good point i haveto do it via ifconfig's output |
| 17:45.47 | Penguin | Of course. |
| 17:45.53 | XandriX | ill just fiddle around till i get it to get me the proper value |
| 17:45.56 | Penguin | I could have used ifconfig, but I used ip instead. |
| 17:46.42 | boom^time | If I wanted to use Dial to dial a local extension in a specific context is that possible? What would I use for the technology argument as opposed to SIP Zap etc? |
| 17:47.02 | Penguin | Local |
| 17:47.03 | Greenlight | Dial(Local/exten@context) |
| 17:47.07 | boom^time | Thanks guys |
| 17:47.24 | *** part/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 17:48.26 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 17:48.56 | Penguin | xandrix: What is your interface's name? |
| 17:49.37 | *** join/#asterisk onizo (~onizo@cpe-75-80-122-116.san.res.rr.com) |
| 17:49.45 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 17:49.55 | Penguin | ifconfig le0|awk '$1 == "inet" {print $2}' |
| 17:50.33 | Penguin | Mine is le0. |
| 17:51.05 | Penguin | No fiddling required. |
| 17:52.50 | XandriX | ow time to read up on the sound packs hehe |
| 17:55.56 | XandriX | as i am not entirely sure where this installation stores them by default |
| 18:00.45 | Penguin | You can do a search for echo-test and see where it is. |
| 18:01.50 | *** join/#asterisk mjordan (~matt@15.natext-192-249-1-0-27.utk.edu) |
| 18:01.50 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 18:03.34 | Penguin | xandrix: You don't even have to use those sound files. I just added those to make it nice. |
| 18:03.52 | Penguin | You could just let it play the numbers and dots. |
| 18:04.05 | XandriX | but i want sounds it would be nicer |
| 18:04.07 | XandriX | :P |
| 18:05.43 | Penguin | http://pastebin.com/v3gQufkt |
| 18:07.06 | Penguin | You may need to refresh it. |
| 18:07.46 | *** join/#asterisk xytis (~xytis@88-223-36-62.meganet.lt) |
| 18:11.12 | XandriX | hmm so this install does not use /var/lib/asterisk/sounds |
| 18:14.41 | [TK]D-Fender | "core show settings" <--------- |
| 18:17.56 | *** join/#asterisk leifmadsen (~lmadsen@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:17.56 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:18.34 | XandriX | Penguin: thx for the code without the sounds it works like a charm :) |
| 18:19.06 | leifmadsen | If anyone wants the Asterisk: TDG 4e book for 50% off, O'Reilly has it on sale today. Just use code CYBER3 at check out |
| 18:20.18 | BeachBall | does BeachBall need that? |
| 18:20.22 | XandriX | [TK]D-Fender: no mention for a sound folder but reveals alot of info |
| 18:20.59 | BeachBall | I have the 3rd e |
| 18:21.03 | BeachBall | :{ |
| 18:21.33 | BeachBall | leifmadsen: whats new in the 4th one? |
| 18:21.38 | *** join/#asterisk mjordan (~matt@15.natext-192-249-1-0-27.utk.edu) |
| 18:21.38 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 18:21.39 | drmessano | Everything is new |
| 18:21.53 | leifmadsen | BeachBall: Asterisk 11 stuffz |
| 18:21.53 | drmessano | Asterisk 11 makes 10 look like 1.8 |
| 18:22.01 | TechSmurf | I'm trying to hack the blacklist module to recognize CID names as well as numbers |
| 18:22.04 | leifmadsen | big update on all the Asterisk 11 stuff |
| 18:22.16 | leifmadsen | Also about 150 pages more than the 3rd edition |
| 18:22.36 | TechSmurf | can anyone identify any problems with this line? |
| 18:22.38 | TechSmurf | exten => s,n,GotoIf($["${DB_EXISTS(blacklist/${TOUPPER(${CALLERID(name)})})}"="1"]?blacklisted) |
| 18:22.48 | BeachBall | You did not meet the criteria for this discount. |
| 18:22.49 | BeachBall | :{ |
| 18:22.51 | drmessano | leifmadsen, so it makes one taller? |
| 18:23.11 | TechSmurf | I'm pretty sure my first problem is that blacklist is replacing spaces with "\" |
| 18:23.23 | TechSmurf | but beyond that... |
| 18:23.48 | BeachBall | I THINK I need to buy another book |
| 18:23.52 | [TK]D-Fender | [13:20]XandriX[TK]D-Fender: no mention for a sound folder but reveals alot of info <- VARLIB/sounds |
| 18:23.54 | BeachBall | to get the discount |
| 18:23.59 | BeachBall | whats a good one to get? |
| 18:25.00 | leifmadsen | TechSmurf: at least the syntax looks right |
| 18:25.04 | drmessano | BeachBall, did you get the ebook or the print? |
| 18:25.07 | leifmadsen | I see no issue it the brackets or braces |
| 18:25.12 | BeachBall | print |
| 18:25.18 | BeachBall | it's cheaper at amazon |
| 18:25.22 | drmessano | Read the banner |
| 18:25.36 | leifmadsen | drmessano: at least 100 pages taller |
| 18:25.44 | XandriX | [TK]D-Fender: so varlib folder is the folder that should contain a subfolder called sounds is that corrent ? |
| 18:25.55 | BeachBall | hugs leifmadsen for being so smart |
| 18:26.01 | [TK]D-Fender | TechSmurf: you should start by not assuming the line is wrong.. and instead proving whey it should be working... |
| 18:26.07 | [TK]D-Fender | XandriX: Correct |
| 18:26.15 | mjordan | hugs > $$$ |
| 18:26.21 | leifmadsen | I disagree |
| 18:26.25 | leifmadsen | I can hug money |
| 18:26.34 | leifmadsen | Scrooge McDuck styles |
| 18:26.37 | XandriX | leifmadsen: yeah but money doesnt hug you back bro |
| 18:26.37 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.117) |
| 18:26.38 | mjordan | leifmadsen: whoever said money doesn't keep you warm at night hasn't tried my money blanket |
| 18:26.48 | leifmadsen | XandriX: it does when I call the escort service |
| 18:26.58 | XandriX | leifmadsen: well played my friend :) |
| 18:27.02 | leifmadsen | :D |
| 18:27.09 | leifmadsen | snappy comback Monday ftw |
| 18:27.14 | drmessano | Whoever said money can't buy you happiness has never paid for a divorce |
| 18:27.30 | mjordan | snaps |
| 18:27.43 | XandriX | that or who said money cant buy happiness obviously never bought a jetski have you ever seen anyone sad on a jetski ? |
| 18:28.19 | drmessano | XandriX, have you ever seen anyone sad eating a cupcake? |
| 18:28.30 | XandriX | drmessano: actualy yes thats the worst part |
| 18:29.38 | TechSmurf | yep, the problem is the gui. |
| 18:29.51 | TechSmurf | the backslash is killing it |
| 18:30.13 | XandriX | hmm odd so i do have default soundfiles so i take it Penguin that the files you referenced to are extra sound files ? |
| 18:31.12 | TechSmurf | if I add a db entry through cli it works |
| 18:31.33 | file | crackles |
| 18:32.12 | XandriX | removes file from the stove |
| 18:32.18 | *** join/#asterisk navaismo (~navaismo@189.241.125.73) |
| 18:41.01 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 18:42.43 | *** join/#asterisk zigg (~matt@unaffiliated/zigg) |
| 18:43.44 | TechSmurf | hrm |
| 18:43.50 | *** join/#asterisk xytis (~xytis@88-223-36-62.meganet.lt) |
| 18:44.18 | TechSmurf | so... any ideas on how to modify that line so that it will match a cid of "NAME HERE" to a db entry of blacklist/NAME\HERE ? |
| 18:44.41 | TechSmurf | regex wildcard for the space? |
| 18:45.58 | Penguin | What command are you using to write the db entry? |
| 18:46.50 | *** join/#asterisk felipealmeida (~user@177.98.67.55) |
| 18:47.10 | TechSmurf | freepbx blacklist module gui, and fixing whatever it's doing to replace the space with a backslash isn't much within the scope of my hack... |
| 18:48.03 | tm1000 | TechSmurf: advice: |
| 18:48.04 | tm1000 | first |
| 18:48.07 | tm1000 | ~freepbx |
| 18:48.07 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:48.09 | tm1000 | then |
| 18:48.12 | tm1000 | ?bugs |
| 18:48.17 | TechSmurf | I'm not asking for freepbx help. |
| 18:48.26 | tm1000 | TechSmurf: and I |
| 18:48.31 | tm1000 | and Im not saying you are |
| 18:48.46 | tm1000 | I am telling you how to apply yourself and report bugs so these things can be fixed |
| 18:48.53 | TechSmurf | It's not a bug, either. |
| 18:49.00 | tm1000 | Penguin: isn't going to walk you through hacking a gui he/she never uses |
| 18:49.03 | Penguin | It's a ... feature? |
| 18:49.14 | tm1000 | Penguin: no its a bug. that needs to be fixed |
| 18:49.19 | tm1000 | (from my opinion) |
| 18:49.27 | Penguin | You're probably correct on that. |
| 18:49.34 | file | watches |
| 18:49.35 | TechSmurf | It's not designed to support using cid names instead of numbers :P |
| 18:49.50 | tm1000 | I guess what I am getting at here |
| 18:50.09 | tm1000 | is that you are asking how to go about modifying a gui in a channel where everyone uses straight dialplan |
| 18:50.15 | tm1000 | so my advice is lets get it working in freepbx |
| 18:51.27 | tm1000 | I would assume, (perhaps wrongly) that Penguin is probably slightly annoyed to know you are working the constraints of freepbx when you didn't initally state that fact |
| 18:52.14 | Penguin | I'm just simply not going to be able to help "solve" the issue. It is outside of the scope of asterisk. |
| 18:52.24 | navaismo | Penguin and how was the taste of the DIY tortillas? |
| 18:52.39 | Penguin | navaismo: Did not work so well. |
| 18:53.29 | TechSmurf | i.e. the proper solution is to have it add the database entry without replacing spaces with backslashes, and attempting to fix this after the fact by somehow using a wildcard in the database check is a horrible idea |
| 18:53.45 | navaismo | Penguin, :( |
| 18:54.16 | Penguin | navaismo: We only had all purpose flour, not corn flour. The flour soaked the water too much, and when we added more water, we got gluten problems and didn't have good flavor. |
| 18:54.58 | navaismo | :S |
| 18:55.04 | Penguin | navaismo: We added some baking powder and shortening, and then it was okay with all purpose flour. |
| 18:55.56 | navaismo | now you need a video for making your own flour using corn |
| 18:57.10 | TechSmurf | wonders if the calories in the tortilla make up for the calories output making it this way. :P |
| 18:59.49 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 19:00.00 | *** join/#asterisk serafie (~erin@nat/digium/x-jdlutboscrgzcgnz) |
| 19:00.53 | *** join/#asterisk Doyanole (~Doyanole@vodsl-11062.vo.lu) |
| 19:01.04 | Doyanole | hello |
| 19:01.49 | Doyanole | is there anyone who can help me with a small issue i have with my asterisk running on my synology regarding internal numbers not workin ? |
| 19:02.25 | BeachBall | can help you |
| 19:02.47 | BeachBall | please descript your problem in full detail |
| 19:02.52 | BeachBall | describe* |
| 19:03.09 | Doyanole | well , i installed asterisk today on my synology. inbound outbound works. only thing i have left that is not working is internal calling between phones |
| 19:03.47 | Doyanole | it says busy |
| 19:03.54 | Doyanole | when trying to reach an internal number |
| 19:04.52 | [TK]D-Fender | Doyanole: Last I checked Synology was running a GUI and there no no supported GUI's in this channel and virtually no-one using yours specifically. |
| 19:05.11 | [TK]D-Fender | Doyanole: So the abstractions of "internal number" etc are specific to their implementation. |
| 19:05.13 | Doyanole | indeed , it is running a GUI |
| 19:05.29 | [TK]D-Fender | Doyanole: We can't support their GUI |
| 19:05.51 | Doyanole | here's a paste of the error |
| 19:05.52 | Doyanole | http://pastebin.com/upmKA0PL |
| 19:05.53 | Doyanole | hmm |
| 19:06.30 | [TK]D-Fender | That looks like Digium's old AsteriskGUI dialplan... |
| 19:06.42 | [TK]D-Fender | Doyanole: -- Executing [6002@DLPN_521906:1] Macro("SIP/6000-000000cc", "trunkdial-failover-0.3,SIP/trunk_1/6002,,trunk_1,") in new stack <-- no NUMBER being passed. |
| 19:06.53 | [TK]D-Fender | Doyanole: That is your problem most likely |
| 19:06.53 | Doyanole | ok |
| 19:07.06 | WIMPy | Whatever it is it doesn't think that "6002" is an "internal" number. |
| 19:07.15 | [TK]D-Fender | Doyanole: 2nd parameter is blank there... |
| 19:07.28 | [TK]D-Fender | Doyanole: Actually... scratch that |
| 19:07.50 | Doyanole | uuhm meaning ? |
| 19:07.52 | [TK]D-Fender | <PROTECTED> |
| 19:08.03 | Doyanole | yes, and it shouldn't |
| 19:08.08 | Doyanole | <PROTECTED> |
| 19:08.10 | Doyanole | right ? |
| 19:08.11 | [TK]D-Fender | And isn't a legit looking number for the PSTN. |
| 19:08.16 | Doyanole | yes |
| 19:08.19 | [TK]D-Fender | Doyanole: I'd like to think "no". |
| 19:08.30 | WIMPy | It doesn't matter. You need to find a suipport channel for whatever GUI you're using. Ur bin it and install plain Asterisk. |
| 19:08.35 | [TK]D-Fender | Doyanole: But we can't say why things aren't as you think they should be. |
| 19:08.58 | Doyanole | why are there such differences between GUI nad CLI ? |
| 19:09.10 | Doyanole | besides clicking and typing |
| 19:09.27 | WIMPy | Noone here knows what they configure. |
| 19:09.48 | Doyanole | so you guys just use simple cli to configure it ? |
| 19:09.49 | WIMPy | Asterisk doesn't have any kind of standard way of working. |
| 19:10.03 | WIMPy | No. A text editor. |
| 19:10.07 | Doyanole | damn |
| 19:10.11 | Doyanole | that's a hassle |
| 19:10.14 | Doyanole | for me at least |
| 19:10.51 | WIMPy | It is, but debugging what someone else wrote is certainly no easier. |
| 19:10.56 | [TK]D-Fender | Doyanole: Thing is you're using some chopped up version hacked into a toy platform |
| 19:11.22 | Doyanole | ok, than tell me what the ideal configuration would be (and safe) to use asterisk |
| 19:11.24 | Doyanole | please |
| 19:11.48 | [TK]D-Fender | Doyanole: Anything else |
| 19:11.51 | Doyanole | lol |
| 19:11.54 | WIMPy | The one you create for your needs. |
| 19:12.09 | [TK]D-Fender | Doyanole: Roll your own or use a GUI that actual has an active support community |
| 19:12.16 | Doyanole | i am talking hardware wise |
| 19:12.24 | [TK]D-Fender | Doyanole: #freepbx <- |
| 19:12.33 | [TK]D-Fender | Doyanole: Anything might do. |
| 19:12.39 | [TK]D-Fender | Doyanole: Depends on your needs |
| 19:12.39 | WIMPy | Any PC. |
| 19:12.45 | Doyanole | i just had the synology running and unfortunately no freepbx is supported on that |
| 19:12.56 | WIMPy | And depending on what you plan do do even much less may be ok. |
| 19:13.17 | [TK]D-Fender | Doyanole: Then get an actual computer |
| 19:13.43 | Doyanole | i have a server running that is used as a file server, can i use freepbx on that ? |
| 19:13.55 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 19:13.56 | [TK]D-Fender | probably |
| 19:14.11 | Doyanole | k |
| 19:14.31 | Doyanole | thanks for the tips, will try that |
| 19:14.56 | [TK]D-Fender | Doyanole: Still not smart as spikes could hurt you on the uadio side, etc |
| 19:15.06 | Doyanole | aha |
| 19:15.07 | Doyanole | ok |
| 19:15.14 | [TK]D-Fender | Might do OK depending |
| 19:15.17 | [TK]D-Fender | YMMV |
| 19:15.33 | Doyanole | so ideal would be a standalone PC ? |
| 19:15.37 | Doyanole | or server ofc |
| 19:15.44 | WIMPy | I haven't had any issue like that even when I tried really hard. |
| 19:16.00 | [TK]D-Fender | Doyanole: Ideal is standalaone |
| 19:16.56 | WIMPy | I went on a strict one PC is enough policy some years ago. That thing is doing quite a lot of stuff, but I have never had any RTP delays or such. |
| 19:17.25 | [TK]D-Fender | Does depend on load. Might work well... might not. |
| 19:17.44 | WIMPy | With TDM minor glitvhes are possible, but you only notice then when listening to MOH or a dialtone. |
| 19:17.50 | [TK]D-Fender | Feel free to try setting up on yours and see what happens |
| 19:17.57 | Doyanole | k |
| 19:18.09 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.131) |
| 19:18.18 | WIMPy | Copying videos or starting lots of compilers doesn't do any harm for me. |
| 19:19.15 | WIMPy | There cann be small delays in playing samples when under heavy load. |
| 19:19.23 | *** join/#asterisk simNIX (~simNIX@156-60.bbned.dsl.internl.net) |
| 19:20.23 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 19:21.22 | Doyanole | will it run on a rasp ? |
| 19:21.44 | TechSmurf | exten => s,n,GotoIf($["${DB_EXISTS(blacklist/${TOUPPER(${REPLACE(CALLERID(name),\ ,/)})})}"="1"]?blacklisted) |
| 19:21.53 | TechSmurf | well, that was easier than fixing fpbx |
| 19:21.56 | WIMPy | Yes. |
| 19:22.13 | WIMPy | There are even multiple distros for it by now, IIRC. |
| 19:22.46 | [TK]D-Fender | Doyanole: What are you actually looking to do on it? |
| 19:23.53 | Doyanole | well, not much , it's basically for my 2 homenumbers, my office number, and 5 internal numbers. if possible , i would like to hook up a GSM Gateway , so i can reroute the Mobile calls via that |
| 19:24.01 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.131) |
| 19:25.49 | WIMPy | There are several way to do the GSM thing. But you should check if any of them are supported by GUIs. |
| 19:26.11 | Doyanole | what could you suggest ? |
| 19:26.23 | [TK]D-Fender | WIMPy: Shouldn't be an issue with FreePBX... |
| 19:26.25 | WIMPy | Need more input. |
| 19:26.37 | Doyanole | what kind of input ? |
| 19:27.01 | WIMPy | I'm pretty sure FreePBX will at least not support using osmocomm. And what about chan_dongle? |
| 19:27.03 | [TK]D-Fender | Doyanole: Specifics on the gateway you're loking at |
| 19:27.18 | [TK]D-Fender | WIMPy: Dongle you can doa custom trunk for easily enough |
| 19:27.18 | WIMPy | What you want to do, how and with what hardware. |
| 19:27.30 | Doyanole | ok, i am working from home |
| 19:27.42 | Doyanole | so i have 2 home numbers an 1 office number |
| 19:27.47 | Doyanole | everything is VOIP |
| 19:27.52 | Doyanole | i have a 200mbit connection |
| 19:28.01 | Doyanole | mostly i'm at customers |
| 19:28.10 | Doyanole | so i route my homenumber to my mobile |
| 19:28.19 | Doyanole | which costs me a fortune every month |
| 19:28.30 | Doyanole | and therefor i would like to use a gsm gateway |
| 19:28.33 | Doyanole | additionaly |
| 19:28.38 | *** join/#asterisk Micc (~Micc@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 19:28.44 | WIMPy | Cant you get a mobile with a geo number? |
| 19:28.46 | Doyanole | i have 2 simple SIP phones and some dect phones |
| 19:28.49 | WIMPy | Where are you? |
| 19:28.53 | Doyanole | in Luxembourg |
| 19:29.32 | WIMPy | If you want to use DECT phones, you should take a look at the AVM FritxBox series of routers. |
| 19:29.40 | Doyanole | i have one |
| 19:29.44 | Doyanole | 7390 |
| 19:30.01 | WIMPy | Great. What are you missing on it? |
| 19:30.25 | Doyanole | should i install asterisk on the fritz ? |
| 19:30.39 | WIMPy | Why do you want Asterisk? |
| 19:30.51 | Doyanole | what else ? |
| 19:31.06 | WIMPy | You could do it if you need more features than it offers by itself. |
| 19:31.14 | Doyanole | just the simple fritz ? |
| 19:31.39 | WIMPy | That's why I asked what you missing from it. |
| 19:31.43 | Doyanole | ok |
| 19:31.52 | Doyanole | well |
| 19:32.07 | Doyanole | i don't know if the gsm gateway will work |
| 19:32.10 | Doyanole | that's the first |
| 19:32.14 | Doyanole | and which one to buy |
| 19:32.17 | Doyanole | and 2nd |
| 19:32.25 | WIMPy | Well, no. |
| 19:32.33 | Doyanole | i have 2 sip phones i bought , just to use them internally |
| 19:32.35 | WIMPy | Unless you use a SIP gateway, off course. |
| 19:33.02 | Doyanole | and unfortunaltely the 2 sipphones can't do **620 for example |
| 19:33.06 | WIMPy | But if you just want cheap rates to the GSM, get the right ITSP for that. |
| 19:33.06 | Doyanole | the ** doesn't work |
| 19:33.08 | Doyanole | only * |
| 19:33.18 | WIMPy | Configure them. |
| 19:33.22 | Doyanole | you can't |
| 19:33.29 | Doyanole | it's standard in the phone |
| 19:33.30 | WIMPy | huh? |
| 19:33.35 | Doyanole | and fritz can't change it |
| 19:33.44 | Doyanole | fritz only does ** |
| 19:33.51 | Doyanole | and the phones only do * |
| 19:33.56 | Doyanole | so duuuh, i'm fucked |
| 19:33.58 | Doyanole | ;-) |
| 19:33.59 | WIMPy | I haven't seen a phone where you can't change the dialplan. |
| 19:34.10 | Doyanole | it's a cheap 25 euro sip phone |
| 19:34.15 | WIMPy | Well, I have, but they let you dial anything, which is just as good. |
| 19:34.35 | Doyanole | i looked all over the place and i got the confirmation that the phone can't change it |
| 19:34.51 | WIMPy | Looks like you bought crap. |
| 19:34.55 | Doyanole | yep |
| 19:34.56 | Doyanole | :-D |
| 19:34.59 | WIMPy | Have you tried *#**? |
| 19:35.15 | Doyanole | when you hit * two times , it makes *. |
| 19:35.26 | Doyanole | it doesn't do 2 stars |
| 19:35.42 | WIMPy | Nowhere? |
| 19:35.47 | WIMPy | Or just not at the beginning? |
| 19:35.57 | Doyanole | nowhere |
| 19:36.05 | Doyanole | * is the beginning and than ..... |
| 19:36.23 | WIMPy | Get a decent phone then. |
| 19:36.26 | Doyanole | hehe |
| 19:36.27 | Doyanole | yeah |
| 19:36.31 | Doyanole | that's the other option |
| 19:36.47 | Doyanole | but main thing i guess would be my gsm issue |
| 19:37.03 | WIMPy | >>But if you just want cheap rates to the GSM, get the right ITSP for that. |
| 19:37.11 | Doyanole | what's ITSP ? |
| 19:37.18 | WIMPy | ~itsp |
| 19:37.18 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 19:37.31 | Doyanole | hehe |
| 19:37.33 | Doyanole | yes |
| 19:37.38 | Doyanole | i'm in europe |
| 19:37.44 | Doyanole | specifically in Luxebourg |
| 19:37.46 | Doyanole | forget that |
| 19:37.50 | WIMPy | So what? |
| 19:38.11 | WIMPy | Check voipratetracker.com for example |
| 19:38.35 | Doyanole | i can get a Mobile card for 5 euros'a month with unlimited MObile calls |
| 19:38.45 | Doyanole | i doubt any ITSP can beat that |
| 19:38.50 | lvlinux | [6~[6~[B |
| 19:39.11 | WIMPy | 6.2¢/min |
| 19:40.03 | WIMPy | So that's only 80 minutes indeed. |
| 19:40.06 | Doyanole | 6.2 ceents a minute |
| 19:40.09 | Doyanole | cents |
| 19:40.15 | Doyanole | let me check how much i pay now |
| 19:40.18 | Doyanole | gimme a sec |
| 19:41.29 | Doyanole | 105 minutes were 15 euros |
| 19:41.52 | Doyanole | so that's 14cts |
| 19:42.13 | WIMPy | Or if you can manage to log in to a german network at least once every four weeks, get a SIM from sipgate and call for free. |
| 19:42.43 | Doyanole | na |
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| 19:45.19 | Doyanole | ok, but i could use the onevoip as a start |
| 19:45.33 | Doyanole | how does that work , and how do i need to configure it , never did that |
| 19:46.01 | WIMPy | You should probably be able to use Asterisk with chan_donge on the FB. |
| 19:46.23 | WIMPy | Didn't you say tha't you're all voip alread? So justlike you did with the others. |
| 19:46.52 | Doyanole | and how do i configure it to use the voipone just for mobile calls |
| 19:47.20 | WIMPy | By configuring the prefix(es) in the routing table. |
| 19:47.45 | Doyanole | k |
| 19:48.58 | Doyanole | fuck |
| 19:49.04 | Doyanole | Only businesses or registered organizations may apply for the Onevoip VoIP solutions. |
| 19:49.13 | Doyanole | grrrr |
| 19:53.02 | Doyanole | what's chan_donge , WIMPy |
| 19:53.03 | Doyanole | ? |
| 19:53.56 | WIMPy | A channel for Asterisk that makes use of USB GSM donges (AKA surfsticks). |
| 19:54.29 | Doyanole | k |
| 19:56.28 | hardwire | :( |
| 19:56.33 | Doyanole | http://www.raspberry-asterisk.org/documentation/gsm-voip-gateway-with-chan_dongle/ |
| 19:57.18 | WIMPy | I guess you should be able to run that on the FritzBox directly. |
| 19:58.52 | Doyanole | k |
| 19:58.58 | Doyanole | with freepbx ? |
| 19:59.08 | Doyanole | i like gui's ;-) |
| 19:59.23 | WIMPy | No |
| 19:59.26 | Doyanole | damn |
| 19:59.27 | Doyanole | ok |
| 19:59.49 | WIMPy | For ony a gateway that would be complete overkill. |
| 19:59.54 | WIMPy | only |
| 20:00.16 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 20:01.28 | WIMPy | Or if you don't have a phone line connected, look for an old GSM ISDN gateway. That would be a real plug and play thing. |
| 20:02.02 | Doyanole | http://www.amazon.de/Huawei-B260-gateway-Original-Artikel-Vodafone/dp/B00DNZQXB4/ref=sr_1_3?ie=UTF8&qid=1386012425&sr=8-3&keywords=gsm+gateway |
| 20:02.04 | Doyanole | like that ? |
| 20:03.37 | WIMPy | As usual with Amazon, the description isn't clear at all. |
| 20:03.45 | Doyanole | lol |
| 20:04.08 | Doyanole | http://www.mobilityminded.com/6685/ |
| 20:06.01 | Doyanole | i can get a Mobile SIM for 9 euros that includes 4000 minutes |
| 20:06.04 | WIMPy | Maybe that's only uasble as an analogue line. |
| 20:06.09 | Doyanole | damn |
| 20:06.47 | WIMPy | Well, it might be ok. |
| 20:08.30 | Doyanole | http://www.amazon.de/Sagem-RL302-Voicebox-Mobiltelefon-GSM/dp/B001CSMKNE/ref=sr_1_4?ie=UTF8&qid=1386014777&sr=8-4&keywords=gsm+gateway |
| 20:08.33 | Doyanole | that should do it |
| 20:10.01 | WIMPy | Again I don't find any information about how you can connect to that thing. |
| 20:11.00 | Doyanole | all i find is in german |
| 20:11.35 | WIMPy | I'm, german :-) |
| 20:11.51 | Doyanole | aaah |
| 20:11.52 | Doyanole | lol |
| 20:11.55 | Doyanole | dann is gut |
| 20:11.57 | Doyanole | ;-) |
| 20:12.05 | Doyanole | Ich habe das Gerät als GSM-Gateway an der Fritzbox angeschlossen. Telefonate mit Mobiltelefonen werden jetzt rausgeleitet und gehen über die RL302. Das Geräte hängt hierbei im EG an der Außenwand wo der Empfang recht gut ist. Über eine Multicard vom Handy erfolgt die Mitnutzung des bestehenden Mobilfunkvertrages. Ich bin recht zufrieden damit. Sprachqualität passt, Ausfälle bisher keine, Verbindungsaufbau über Fritzbox und GSM-Gateway dauer |
| 20:12.06 | Doyanole | deutlich länger aber OK, ein Nachteil ist halt immer da. |
| 20:12.07 | Doyanole | Inbetriebnahme ohne jedes Problem |
| 20:12.32 | WIMPy | #asterisk-de |
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| 20:27.41 | Qwell | nowai, it's a bkruse |
| 20:27.51 | Qwell | I hear that guy is a big deal. |
| 20:28.17 | bkruse | Qwell: you must be thinking of tcruise - tom cruise, he regulars #asterisk every once in awhile |
| 20:28.51 | bkruse | Qwell: I also heard bkruse drives like a total _dick_ |
| 20:29.06 | Qwell | bkruse: only in his lambo |
| 20:38.12 | bkruse | Qwell: you ever in hsv anymore? |
| 20:38.16 | bkruse | we should do lunch sometime! |
| 20:38.23 | Qwell | bkruse: I still live here |
| 20:39.50 | BeachBall | gives Qwell a friendly kick in the nuts |
| 20:39.56 | BeachBall | :} |
| 20:40.10 | bkruse | ouch! |
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| 20:56.09 | ChannelZ-Wk | Those were perfectly good walnuts. |
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| 21:30.18 | lvlinux | anybody know how what "unknown callerid data type 'nam'" means? I get it when getting a SIP call in from a UK DID. The callerid shows as unknown |
| 21:30.53 | lvlinux | is that a callerid type that asterisk cannot handle or something? Is there a way I can work around it (still get the callerid info)? |
| 21:31.18 | WIMPy | Show us |
| 21:33.18 | lvlinux | show you what? |
| 21:33.34 | WIMPy | The call with that message. |
| 21:33.38 | lvlinux | k |
| 21:36.32 | lvlinux | http://pastebin.com/X8sCB33x |
| 21:37.53 | WIMPy | wonders where that callerid_read comes from. |
| 21:38.15 | WIMPy | But a Progress() after Answer() doesn't make any sense. |
| 21:38.53 | lvlinux | oh? why? |
| 21:39.19 | lvlinux | oh I know why---nevermind |
| 21:39.33 | lvlinux | i just had that in there from the other day trying to fix something else lol |
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| 21:39.48 | lvlinux | what do you mean about the callerid_read? |
| 21:40.10 | WIMPy | I do only see a write, but no read. |
| 21:40.44 | WIMPy | Do you read CLLERID() there as well? |
| 21:45.07 | lvlinux | not sure I understand what you are asking ---after the call is answered, i have "same => n,Set(CALLERID(name)=UK_caller${CALLERID(nam)})" so that I can at least see on th ephone that they dialed the UK DID. |
| 21:45.34 | navaismo | wondering why he didint see the typo |
| 21:46.20 | lvlinux | typo? where? |
| 21:46.53 | navaismo | ,Set(CALLERID(name)=UK_caller${CALLERID(nam)}) |
| 21:46.58 | navaismo | nam<---- |
| 21:47.16 | lvlinux | oh my goodness lol |
| 21:47.20 | lvlinux | thanks |
| 21:47.29 | lvlinux | kicks myself |
| 21:48.11 | navaismo | I have been there a lot of time because my stupid twinkie fingers |
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| 22:44.19 | smirker | mmm twinkie fingers |
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| 23:27.41 | navaismo | want to suck it... my fingers? LOL |
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| 23:36.29 | Penguin | You want to suck what? |
| 23:37.35 | navaismo | argh again that sounds bad for me |
| 23:37.58 | navaismo | supposed to be for smirker not for me :'( |
| 23:38.14 | Penguin | Ah, then you probably should not start your statement with /me |
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| 23:39.47 | Penguin | This is what we all saw: |
| 23:39.51 | Penguin | navaismo want to suck it |
| 23:40.06 | navaismo | yep i just re-read that |
| 23:40.11 | navaismo | and sounds very bad |