| 00:02.06 | p3nguin | wimpy: I keep trying to start it and then I add another noload for the last module it shows anything about each time. I have a huge list of noloads now, but it just keeps on going. I think it might be easier to not load anything and then manually load the things I do need. |
| 00:03.22 | WIMPy | I've once had it with codec settings. |
| 00:05.29 | p3nguin | Another day, another way. |
| 00:06.03 | p3nguin | I might even decide to try it on another OS, maybe in a virtual machine or something. |
| 00:06.44 | WIMPy | Did you use any old config files? |
| 00:06.54 | p3nguin | yes |
| 00:07.11 | p3nguin | I have a 1.8.2.2 system running for someone else... it didn't have any problem when I compiled, installed, configured, or ran it. |
| 00:07.12 | WIMPy | It's probably one of them. |
| 00:07.48 | WIMPy | had to remove codecs.conf. |
| 00:08.30 | p3nguin | I'll keep that in mind during the next round. |
| 00:09.07 | WIMPy | Ja, I think it's a good time for some sleep. |
| 00:14.09 | NetAtWorld | what is X-Asterisk-HangupCause: Protocol error, unspecified ? |
| 00:14.19 | NetAtWorld | or X-Asterisk-HangupCauseCode: 111 ? |
| 00:18.05 | WIMPy | What it says. |
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| 00:38.10 | NetAtWorld | X-Asterisk-HangupCauseCode: 111 --> What does it means ---> X-Asterisk-HangupCauseCode: 111 |
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| 00:56.45 | nestAr | Is there a good up to date replacement for voip-info.org? |
| 01:07.57 | p3nguin | How about the offial asterisk wiki? |
| 01:09.39 | p3nguin | http://wiki.asterisk.org/ |
| 01:13.23 | *** join/#asterisk shapr (~shapr@nat/digium/x-bwkxzicliyxsspmm) |
| 01:17.21 | shapr | Good Morning Asterisk people! |
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| 01:20.45 | nestAr | p3nguin: lol.. yeah, i guess that would work. maybe they need better SEO. :) |
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| 02:06.07 | tashAFK | WIMPy: still around? |
| 02:06.48 | tashAFK | I'm going to try those things we talked about earlier here in a bit. Was hoping you might be around just in case I have a couple of questions. |
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| 02:29.54 | XDS|Work | can i make my own sip accounts with asterisk and freepbx |
| 02:33.01 | ChannelZ | that's kind of the point |
| 02:35.48 | XDS|Work | got them up and running , where can i find a guide how to do exactly that. |
| 02:36.03 | XDS|Work | ? , and thanks for the reply. |
| 02:37.23 | shapr | I don't understand the question? |
| 02:37.35 | shapr | XDS|Work: asterisk is both client and server for SIP |
| 02:37.51 | shapr | So you could connect two SIP phones to your asterisk server and they could call each other. |
| 02:37.54 | XDS|Work | server |
| 02:38.26 | shapr | Then you could purchase SIP to phone minutes from a SIP provider and use that too. |
| 02:49.24 | *** join/#asterisk prowl (~prowl@unaffiliated/prowl) |
| 02:49.35 | p3nguin | Asterisk is a Back-to-Back User Agent (B2BUA). |
| 02:50.13 | p3nguin | What are you trying to do, and what's the problem you are experiencing while trying to accomplish it? |
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| 03:10.46 | XDS|Work | p3nguin setup a sip account to just do internet calling, i dont need a DID |
| 03:10.58 | XDS|Work | ive got fp/as up and running |
| 03:11.09 | p3nguin | xds|work: Do you need to have termination to the PSTN? |
| 03:11.11 | XDS|Work | logged in the admin account |
| 03:11.45 | XDS|Work | no. |
| 03:12.10 | p3nguin | What kind of phone will you be using? |
| 03:12.36 | XDS|Work | cSip |
| 03:12.41 | p3nguin | soft phone? |
| 03:12.42 | XDS|Work | or sipdroid |
| 03:12.46 | XDS|Work | yes |
| 03:13.00 | XDS|Work | sp on nix |
| 03:13.09 | XDS|Work | (respectively) |
| 03:13.14 | p3nguin | If you don't need to do any PSTN termination, you don't even need Asterisk... you can use direct SIP URI dialing from your softphone. |
| 03:13.44 | XDS|Work | what about receiving calls ? |
| 03:13.57 | XDS|Work | same thing ? |
| 03:13.57 | p3nguin | And really, you can even do PSTN termination through an ITSP without Asterisk. |
| 03:14.26 | p3nguin | You said you don't need a DID, so that means you will not be receiving any calls from the PSTN in any way. |
| 03:14.32 | XDS|Work | p3nguin the softphone requires a configured account |
| 03:14.44 | XDS|Work | can i simply setup a dummy account in fp? |
| 03:14.51 | XDS|Work | (freepbx) |
| 03:15.10 | XDS|Work | if so, how? |
| 03:15.28 | p3nguin | I guess you could... but I don't see why you need any type of "account." |
| 03:15.55 | XDS|Work | it has a bottleneck that i have no control over |
| 03:15.57 | p3nguin | I wouldn't know anything about that, because this isn't a FreePBX channel. |
| 03:16.59 | p3nguin | I'm just thinking of something like X-lite... I think you can do SIP URI dialing without any account information. I could be mistaken about this, since I really don't like X-lite and therefore do not use it. |
| 03:22.45 | leifmadsen | XDS|Work: sounds like you want to use X-Lite 4 or Bria 3.0 which doesn't require an "account" to place calls. You can use SIP URI dialing and port forwarding I believe. |
| 03:23.04 | XDS|Work | leifmadsen i just said cSip |
| 03:23.14 | leifmadsen | I just told you what I know |
| 03:23.15 | XDS|Work | this has nothing to do with xlite or bria |
| 03:23.21 | leifmadsen | you didn't listen to me |
| 03:23.34 | leifmadsen | I'm saying if you happen to use that phone then I know it supports a no-account setup |
| 03:23.39 | leifmadsen | cSip sounds like it doesn't |
| 03:24.03 | p3nguin | If that softphone requires an "account" to make calls by SIP URI, maybe that's not the best phone. Or maybe you can get a free ekiga account and use that. |
| 03:24.10 | leifmadsen | anyways I was going to bed since i wasn't feeling well, and you've already frustrated me enough to not care about your problem |
| 03:24.11 | leifmadsen | night all |
| 03:24.12 | XDS|Work | i want to use the client on my android phone, its built directly into the native dialer leifmadsen |
| 03:24.47 | XDS|Work | p3nguin my question stands |
| 03:25.14 | p3nguin | We've discussed enough stuff, I don't know what question you're talkin' about. |
| 03:25.34 | XDS|Work | then why answer ? |
| 03:25.58 | XDS|Work | why not just say that from the beginning ? |
| 03:27.10 | p3nguin | You're beginning to give me the impression that you might be a moron. |
| 03:27.20 | p3nguin | You asked some things, I gave you some answers. |
| 03:27.44 | p3nguin | You said a whole bunch of stuff that didn't even make sense, and I STILL tried to help you... while being polite. |
| 03:28.38 | p3nguin | If you aren't satisfied with the answers you got, consider asking a different question. |
| 04:01.40 | XDS|Work | p3nguin you're giving me the impression that you don't care either way so why should i ? |
| 04:02.14 | p3nguin | You keep saying things that don't make any sense. |
| 04:02.39 | XDS|Work | if something doesn't make sense to you maybe im not the one who is ignorant here |
| 04:03.14 | XDS|Work | you can by all means , ignore me |
| 04:10.15 | tash | I could use a little help. I set a COUNTER in my call file, SetVar: COUNTER=0. In my dialplan, I want to verify that COUNTER is less than or equal to 5 before moving to the next line in my context. Would this do the trick? I'm assuming not based on me never having done this before and just looking at some examples that I don't entirely understand: exten => mycaller,7,GotoIf($["${COUNTER}"<="5"]?mycaller:8) |
| 04:15.02 | p3nguin | I'd rather see that 7 become an n and :8 be removed. |
| 04:15.44 | p3nguin | $[${COUNT} <= 5] should be okay, though. |
| 04:16.34 | tash | p3nguin: I was just thinking that myself actually ... I'm tired of changing the numbers every time I add a line to the context |
| 04:17.19 | tash | do I not need the quotations before and after ${COUNTER} ? |
| 04:17.27 | p3nguin | You need to decide what is going to happen if it evaluates to true. The ?mycaller would try to send it to a priority with the label of mycaller... and based on your choice of extension name, I doubt that's what you meant. |
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| 04:19.03 | p3nguin | It might look something like this: exten => mycaller,n,GotoIf($[${COUNTER} <= 5]?labeliftrue:labeliffalse) |
| 04:19.09 | tash | if it evaluates to true, does the go to exten => mycaller,8,whateverIsAtMyCaller8? |
| 04:19.17 | tash | oh |
| 04:19.21 | p3nguin | If labelisfalse is the very next line, omit :labelisfalse |
| 04:19.23 | tash | I didn't know what the colon meant |
| 04:19.43 | tash | so, the colon separates true from false then I presume? |
| 04:19.54 | p3nguin | s/labelisfalse/labeliffalse |
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| 04:20.44 | p3nguin | exten => mycaller,n,GotoIf($[${COUNTER} <= 5]?go:die) |
| 04:20.48 | p3nguin | etc. |
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| 04:21.59 | p3nguin | You can also omit the true part if the true label is the very next line... |
| 04:22.07 | tash | ok |
| 04:22.08 | p3nguin | exten => mycaller,n,GotoIf($[${COUNTER} <= 5]?:hangup) |
| 04:22.40 | tash | the more I think about this, the more I'm not even sure it is what I want/am trying to accomplish :P |
| 04:23.00 | p3nguin | What *are* you trying to accomplish? |
| 04:23.41 | p3nguin | If you can describe it well, someone can possibly help you devise a plan of action. |
| 04:24.42 | tash | :) Well, I'm placing outbound calls and want users to press 1 to confirm they heard the message played back to them. If I don't get any DTMF back, I essentially move the .call to a temporary directory, touch it for 5 minutes in the future, then move it to the Asterisk outgoing directory. So, the user will get another call in 5 minutes. But, I don't want to flood the user's phone. I only want to attempt a call, say 5 times, then give up. |
| 04:25.16 | tash | It will save me money to prevent my server from placing too many outbound calls too. |
| 04:25.21 | p3nguin | The call file should have that capability built in. |
| 04:25.34 | tash | WIMPy was helping me earlier, then I had to break, and lost some things in translation |
| 04:26.10 | p3nguin | MaxRetries: 5 |
| 04:26.37 | tash | Yes, I have Maxretries: 5 (except I see that I don't have it camel-cased like you just typed) |
| 04:26.42 | tash | does that matter? |
| 04:26.45 | p3nguin | nah |
| 04:26.55 | tash | And, doesn't MaxRetries assume the call was never answered? |
| 04:27.01 | p3nguin | yes |
| 04:27.10 | p3nguin | as far as I know, anyway. |
| 04:27.34 | p3nguin | Your scenario allows the line to be answered, but just never got the right key pressed? |
| 04:27.35 | tash | So, what if the call is answered and the user is just dumb and doesn't press 1...and instead hangs up. Or, the call goes to voicemail. In either case, nobody presses "1" to confirm they got the call. |
| 04:28.40 | tash | If answering machine detection worked well enough, I'd forget all of this nonsense and just try to configure that. But, I've read mixed reviews. |
| 04:29.05 | p3nguin | I think I'm on the same page as you, now. You need it to retry until they finally push the key or until ${COUNTER} is reached. Basically. |
| 04:29.16 | tash | well, hmm ... no, b/c there are still cases where someone may answer the phone and just hangup after they hear the message.... |
| 04:29.22 | tash | that no wasn't to your last message |
| 04:29.23 | tash | reading now |
| 04:29.34 | tash | YES to your last message :) |
| 04:30.32 | p3nguin | They could answer the line 3 times and never press the right key... and the 4th call they don't answer at all, and the 5th call, who knows what happens, but counter has expired so it ends. |
| 04:30.38 | tash | My challenges that I face: increasing ${COUNTER} by 1 in the .call file itself ... then doing the check in the dial plan |
| 04:30.46 | tash | yes |
| 04:30.58 | tash | after 5 attempts, TOO BAD basically |
| 04:31.31 | p3nguin | I would use the dialplan and possibly the AstDB to do the counting. |
| 04:32.02 | p3nguin | The counting and checking the counter is probably the easy part. |
| 04:32.24 | tash | not sure how to increment the COUNTER by 1 each time |
| 04:32.31 | tash | I could probably do it via AGI |
| 04:32.40 | p3nguin | Set(COUNT=$[${COUNT} + 1]); |
| 04:32.54 | p3nguin | s/COUNT/COUNTER/ |
| 04:33.01 | tash | would that go in the dial plan or in the call file? |
| 04:33.01 | p3nguin | dammit |
| 04:33.09 | p3nguin | Set(COUNTER=$[${COUNTER} + 1]); |
| 04:33.18 | tash | i knew what you meant, btw |
| 04:33.19 | p3nguin | I keep wanting to use COUNT. |
| 04:33.56 | tash | and I keep thinking of dracula every time you say count |
| 04:33.58 | p3nguin | I'm eating my supper and trying not to make a mess and losing focus of the task. |
| 04:34.14 | tash | what's for supper tonight? |
| 04:34.31 | p3nguin | Tyson brand honey battered chicken tenders |
| 04:34.43 | tash | yum |
| 04:35.03 | p3nguin | with hickory brown sugar KC Masterpiece sauce |
| 04:35.05 | tash | Set is to be used in the dial plan right? |
| 04:35.10 | p3nguin | yes |
| 04:35.12 | tash | I love me some KC Masterpiece |
| 04:35.28 | p3nguin | The counting and checking of the counter will be dialplan and DB. |
| 04:36.03 | tash | so, when I create the .call file to begin with, should I SetVar: COUNTER=0 in it? |
| 04:36.42 | p3nguin | I would probably do without it. |
| 04:36.57 | tash | k, going to take care of that now |
| 04:37.44 | p3nguin | When you create the call file, are you putting the target's phone in the tech/data portion or in the extension number@context portion? |
| 04:38.22 | tash | I'm putting it in the .call ... Channel: sip/provider_name/phone_number |
| 04:38.38 | p3nguin | And then connect it to an extension and context? |
| 04:39.06 | tash | yes, I have the following in the call file: Maxretries, Retrytime, Context, Callerid, Extension, SetVar, and Waittime |
| 04:39.25 | p3nguin | How are you getting the victim's phone number into the call file? |
| 04:39.28 | tash | ignore the SetVar, it is for something totally separate from this |
| 04:39.39 | tash | i like that, victim... |
| 04:39.54 | tash | it's a PHP script, so via PHP variables |
| 04:40.07 | p3nguin | okay, so it's generated with the file. |
| 04:40.10 | tash | yes |
| 04:40.20 | p3nguin | That makes a difference on how the counting stuff works. |
| 04:40.33 | tash | ok |
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| 04:41.43 | p3nguin | If you were connecting to asterisk with the Chan/number and then connect to your victim with the Extension and Context, that would simplify it, in that Asterisk gets the call first and has a chance to process the counter and end the call. |
| 04:42.06 | p3nguin | The way you're doing it, the victim has to answer before asterisk gets the call into the dialplan. |
| 04:42.28 | tash | yeah, makes sense |
| 04:43.07 | p3nguin | So I'm thinking there has to be some way for the dialplan to abort the call file process when the counter has expired. |
| 04:44.55 | tash | or...on the 5th attempt if the counter is at 5, skip to hangup instead of the next couple of lines...the next couple of lines being the lines in the dial plan that move the .call to a tmp dir, touch the file, and make it ready to be moved to /var/spool/asterisk/outgoing |
| 04:45.10 | tash | or, whatever attempt it would be when the counter is reached.... |
| 04:46.27 | p3nguin | Yes, that's exactly what I had in mind when I was thinking abort. |
| 04:47.05 | p3nguin | Now that we've got the brainstorming part done, it's time to write some extensions. |
| 04:47.28 | tash | I know the priority numbers bug ya, but here is what I have at the moment: |
| 04:47.28 | tash | exten => mycaller,7,System(mv /var/spool/asterisk/outgoing/${PLAYFILE}.call /usr/local/tmp_asterisk/retry_calls/) |
| 04:47.29 | tash | exten => mycaller,8,System(mv /usr/local/tmp_asterisk/retry_calls/${PLAYFILE}.call /usr/local/tmp_asterisk/retry_calls/${PLAYFILE}) |
| 04:48.07 | tash | so, retry_calls.sh actually uses sed to find a line "StartRetry: ... " and deletes it |
| 04:48.31 | tash | b/c for some reason, whey that line exists in the .call file after the 1st failure, the call never gets placed again when I move it to the Asterisk outgoing directory. |
| 04:48.50 | tash | So, by removing that line, touching the file for 5 minutes later, then moving to /var/spool/asterisk/outgoing it all works like I want it to |
| 04:49.32 | tash | so, if the counter is higher than 5 ... I can just skip those lines and hangup, thus not moving the call file around again ... and the user will no longer get called .... follow? |
| 04:49.54 | p3nguin | If you've got a few minutes, I'll rig up some dialplan that does exactly that. |
| 04:52.04 | tash | I got plenty of time |
| 04:52.20 | tash | in the meantime, I can probably get rid of the numbered priorities |
| 05:05.08 | p3nguin | still here |
| 05:05.24 | p3nguin | I'm still here, I mean. |
| 05:05.39 | p3nguin | Thinking about this dialplan. |
| 05:06.07 | tash | ok, np |
| 05:06.15 | tash | I'm working on some other things as well ... |
| 05:06.23 | tash | what time is it where you are? |
| 05:06.28 | tash | 11:06PM here |
| 05:09.54 | p3nguin | same |
| 05:09.56 | p3nguin | CST |
| 05:10.20 | Kobaz | crazy |
| 05:10.31 | Kobaz | you guys are weird... it's actually 12:10 |
| 05:10.53 | Kobaz | OKT |
| 05:10.56 | Kobaz | Official Kobaz Time |
| 05:11.12 | tash | p3nguin: you near KS? |
| 05:11.37 | p3nguin | I'm by St. Louis, MO. |
| 05:12.13 | tash | nice, I'm near KC |
| 05:13.02 | tash | I got side tracked with something else, you having any luck with that dial plan? |
| 05:13.07 | tash | I'm going to think through it now too |
| 05:14.08 | p3nguin | I'm just trying to make it work like I want it to. I know what I want, I just have to type it and get it like it is in my head. |
| 05:15.44 | tash | doest this set COUNTER to 0 to begin with? exten => mycaller,n,Set(__COUNTER="0") |
| 05:19.29 | tash | hmm, I'm just confusing myself I think. If I set COUNTER in the dial plan, and the GotoIf(${$COUNTER}<="5"]?:Hangup evaluates to true, it goes to the next line. The next line moves the call file to my tmp directory, touches it for 5 minutes later, then moves to asterisk outgoing directory to be placed in 5 minutes. Now, this will loop forever unless something is put in the call file, right? I mean, how will the dialplan ever know that the |
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| 05:32.31 | p3nguin | tash: This is what I came up with: http://pastebin.com/eqGqKtEV |
| 05:34.02 | tash | k, looking now |
| 05:35.20 | p3nguin | That allows it to run 5 times, and then on the 6th time the script to move the call file can be executed. |
| 05:36.05 | *** part/#asterisk titter (~Justin@c-98-208-153-148.hsd1.fl.comcast.net) |
| 05:36.13 | p3nguin | I was having issues with DBdel() and DB_DELETE(), so I ended up just setting the value back to where it starts each time. |
| 05:36.29 | tash | can you explain line 1 please? |
| 05:37.07 | p3nguin | ExecIf($[${ISNULL(${DB(counter/counter)})}],Set,DB(counter/counter)=5); |
| 05:37.38 | p3nguin | If the value of counter/counter in the DB is null, execute Set(DB(counter/counter)=5). |
| 05:37.49 | tash | which it would be b/c it doesn't exist yet ... ok |
| 05:37.54 | p3nguin | Otherwise, just go to the next line. |
| 05:38.00 | p3nguin | The first time it runs, it will be null. |
| 05:38.04 | tash | right |
| 05:38.05 | tash | ok |
| 05:38.33 | tash | question though .... |
| 05:38.53 | tash | do you recall the 3 lines I pasted earlier that move the .call around and things? |
| 05:38.54 | p3nguin | I was going to end the "paragraph" with deleting the key from the DB, which would make it null every time it finishes, but DBdel() and DB_DELETE() were's working out for me. |
| 05:39.00 | p3nguin | yeah |
| 05:39.23 | p3nguin | Just replace line 7 with as many lines as you need. |
| 05:40.12 | p3nguin | same goes for line 4. |
| 05:40.35 | p3nguin | Those two System() calls I used are basically just placeholders for whatever you need to happen. |
| 05:40.48 | tash | alright, so, say the first time the user doesn't press "1" ... my current dialplan moves the .call to the tmp dir, touches, it, then moves it back to the asterisk outgoing directory. Doesn't the dialplan forget about that .call at that point, then this starts all over when it picks up the call again? As in, wouldn't this be a loop without something checking for a var in the .call itself? |
| 05:41.54 | *** join/#asterisk Russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 05:43.10 | tash | here is the full context: http://pastebin.com/ZcNtEcnV |
| 05:43.26 | tash | if my question doesn't make sense, let me know and I'll try to word it better. |
| 05:44.00 | tash | for sake of what we are working on, you can just pretend line 17 of that pastebin isn't even there |
| 05:44.55 | tash | retry_calls.sh looks in the .call, finds a line with StartRetry and removes it ... it also touches the file for 5 minutes later. Then a cron runs every 1 minute on my box and moves all .call files in that retry_calls dir to the asterisk outgiong dir |
| 05:46.47 | p3nguin | As in, wouldn't this be a loop without something checking for a var in the .call itself? <--- there is no var in the call file that I am concerned with. |
| 05:47.28 | p3nguin | I am creating the counter in the AstDB and then counting in the dialplan and AstDB. |
| 05:48.22 | p3nguin | Is that the part of it you're having trouble understanding? |
| 05:49.43 | tash | maybe =\ ... let me try to explain what I'm having a hard time understanding.... |
| 05:49.58 | p3nguin | I decided to completely do away with using a variable for the counter. I felt like it wasn't needed. |
| 05:50.18 | p3nguin | okay |
| 05:51.58 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 05:52.32 | p3nguin | I'm ignoring line 17. What do lines 16 and 18 do? |
| 05:52.36 | tash | alright, so a call is placed to the user, they answer, don't press 1 ... I mv the file, touch it, and a cron runs every minute to move .call files from that tmp dir to the asterisk outgoing dir ... so, what is keeping a counter for *that* call file? I mean, when the user answers and we are in the dialplan ... when I move the .call to the asterisk tmp dir, doesn't the dial plan sorta "forget" about that .call file? And, when the call is plac |
| 05:53.38 | p3nguin | You truncated at: And, when the call is plac |
| 05:53.58 | tash | And, when the call is placed again in 5 minutes, is the counter being affected for that .call file? am I making sense? |
| 05:54.07 | p3nguin | I'm using the AstDB to keep count. Do you know what the AstDB is? |
| 05:54.12 | tash | no :( |
| 05:54.20 | p3nguin | It's the Asterisk DataBase. |
| 05:54.23 | p3nguin | It is built in. |
| 05:54.26 | tash | well, I figured that much :) |
| 05:54.37 | p3nguin | It's a persistent database. |
| 05:54.37 | tash | but ... is there a unique ID or something for that call? |
| 05:55.09 | p3nguin | I've made the assumption that this will only be processing one call at a time. |
| 05:55.38 | p3nguin | to only a single victim. |
| 05:57.50 | tash | I think that is a safe assumption...I think :) Every call file will only be placed to 1 phone number |
| 05:57.54 | p3nguin | If you could be processing multiple victims during the same time period, we'd need to add some type of unique identifier per call file so that we can give each victim his own unique ID and keep count per unique ID rather than just one instance of it. |
| 05:57.56 | tash | but, I could have many concurrent calls |
| 05:58.54 | p3nguin | My assumption was that victims would get calls consecutively rather than concurrently. |
| 05:59.11 | tash | so, what I'm struggling to understand is ... say 1234.call is a call to my cell phone. I answer, don't press "1" ... 1234.call is moved to that tmp dir, touched, and moved back to /var/spool/asterisk/outgoing to be placed in 5 minutes...what is keeping track and noting that 1234.call is now in attempt #2 |
| 05:59.31 | p3nguin | the AstDB |
| 05:59.46 | tash | how does it keep track of it? By name? |
| 05:59.56 | tash | sorry so many questions, this is obviously new to me |
| 06:00.12 | tash | wow, I hit enter at precisely 12:00:00AM our time :) |
| 06:00.34 | p3nguin | counter/counter is the DB key and the value is read and changed with each time run of the extension. |
| 06:01.11 | p3nguin | ${DB(counter/counter)} reads the value like a variable. |
| 06:01.29 | tash | so, this logic would support multiple calls going out to different users, possibly at the same time? |
| 06:01.36 | tash | I think you said no |
| 06:01.52 | tash | but, I could test it for a single, non-concurrent call right now .... |
| 06:01.54 | p3nguin | So if the value in the DB is 2, then $[${DB(counter/counter)} = 2] would be true. |
| 06:02.25 | p3nguin | With the way I configured it, it is for calls consecutively, not concurrently. |
| 06:03.00 | tash | I will have many voip trunks, so I will have concurrent calls eventually ... so I need to support that. Is it a pain to write that into the dial plan? |
| 06:03.34 | p3nguin | Run a call file, it runs 5 times to 5551212. Then you can run another call file, and it'll run 5 times to 5552232. Etc. |
| 06:04.09 | tash | there will be times, though, that I have more than 1 call being placed at once ... |
| 06:04.15 | p3nguin | If you can generate some form of a unique identifier PER CALL FILE PER VICTIM, write it as a var into the call file. |
| 06:04.24 | tash | so, damnit, I should have told you that to begin with :) I'm sorry... |
| 06:04.46 | tash | I can do that ... actually I already have a var set in the call file that is unique |
| 06:05.14 | tash | if you look at my dialplan ${CALLFILE} is unique to each call |
| 06:05.20 | p3nguin | As long as it is unique per call file per victim, it'll be simple to add. |
| 06:05.33 | tash | yes, that is unique per call per victim |
| 06:06.00 | p3nguin | One unique thing per victim is his phone number. |
| 06:06.05 | tash | it comes from some php vars ... without looking at the code, it's something like $phone$timestamp$user_id, etc |
| 06:06.07 | *** join/#asterisk benngard (~mabe@213.88.138.230) |
| 06:06.07 | p3nguin | I'll use that as the key. |
| 06:06.28 | tash | well, let's just use ${CALLFILE} if you don't mind ... it is definitely unique |
| 06:06.37 | tash | and has the victim's phone number in the var |
| 06:06.56 | p3nguin | okay |
| 06:08.32 | tash | would you mind putting it in the dialplan I pasted? |
| 06:08.52 | tash | btw, I'm very thankful for your help :) |
| 06:19.00 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 06:19.00 | tash | p3nguin: gonna be on for a bit? |
| 06:19.17 | p3nguin | tash: In that pastebin, what do lines 16 and 18 do? I'm still ignoring 17, by the way. |
| 06:19.32 | p3nguin | Yes, I'm going to make sure this does what you want before I go. |
| 06:19.36 | tash | sorry if I never answered that earlier, going to explain nnow ... |
| 06:22.29 | tash | line 16 moves /var/spool/asterisk/outgoing${PLAYFILE}.call to a temporary directory ... ${PLAYFILE} is the unique identifier I was telling you about. It is a variable set in the .call file with SetVar. So, SetVar: PLAYFILE=$a$bunch$of$php$variables$unique$to$the$user (including their phone number and user ID...its basically a long string of numbers separatated at certain places by a dash or two, an example is +17854142216201102220700.00-305-2 |
| 06:22.53 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-mmqqwobeucwqpire) |
| 06:23.07 | tash | so, that is for sure unique...then line 18 .... |
| 06:24.35 | tash | The script that is executed on line 18 looks in that call file for a line that starts with 'StarRetry' and deletes it. The reason I do that is b/c for whatever reason, if that line exists when I touch the file and move it to the asterisk outgoing directory, the call never gets placed again, period. I don't know why. I just know that when I remove that line, the call is placed when it should be. |
| 06:24.37 | p3nguin | I'm going to go ahead and leave all three of those System() calls like you have them and you can adjust as needed. Since you know how it works and I know how I want to make it work in the dialplan, you'll be able to change it like you need it after I integrate my dialplan with yours. |
| 06:24.40 | tash | does all that make sense? |
| 06:25.02 | tash | ok |
| 06:25.39 | tash | if you'd like to see the contents of retry_calls.sh I can pastebin |
| 06:25.46 | tash | if you don't care, then I won't :P |
| 06:26.02 | p3nguin | tash: http://pastebin.com/4GK6ADmT |
| 06:26.58 | p3nguin | Actually, line 5 isn't needed. |
| 06:27.33 | p3nguin | http://pastebin.com/JTwgtbp9 |
| 06:29.12 | p3nguin | and 19 has an error. |
| 06:29.31 | p3nguin | http://pastebin.com/zWXUhA3a |
| 06:30.24 | tash | k |
| 06:30.30 | tash | going to test, hang tight |
| 06:31.08 | p3nguin | Remember that ${CALLFILE} is the variable that is unique per victim per call file. |
| 06:31.34 | tash | hmm, server is hanging when stopping asterisk ... gonna have to restart the booger or something |
| 06:31.35 | tash | one sec |
| 06:35.53 | tash | testing now.... |
| 06:35.54 | *** join/#asterisk timahvo1 (~rogue@41.90.129.35) |
| 06:36.19 | p3nguin | final bugfix release: http://pastebin.com/pm5Lz8MF |
| 06:37.38 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 06:38.10 | tash | if a user doesn't press a digit at line 7 WaitExten, do lines 10, 11, and 12 get executed? I mean, should they/will they? |
| 06:39.08 | p3nguin | exten => mycaller,n,WaitExten(5); |
| 06:39.09 | p3nguin | exten => mycaller,n,Hangup(); |
| 06:39.28 | p3nguin | If they wait more than 5 seconds at that WaitExten(), the next line is executed. |
| 06:39.37 | p3nguin | That next line says to Hangup(). |
| 06:39.54 | tash | I see that, does that mean any line after hangup doesn't get executed? |
| 06:40.04 | tash | if so, I need to put my System calls before Hangup |
| 06:40.09 | p3nguin | Once it hangs up, there is nothing else to be done. |
| 06:40.12 | tash | ok |
| 06:40.38 | p3nguin | So you wanted those System()s to run either way? |
| 06:41.34 | tash | the system calls move the .call to the tmp dir I'm working with, touch it, and then a cron runs every 1 minutes ... looks in that dir for .call files and moves them to /var/spool/asterisk/outgoing |
| 06:42.01 | tash | so, if at WaitExten(5) the time expires without user input, I want my System calls to execute |
| 06:42.14 | p3nguin | I guess I thought those should only run when the counter expired. |
| 06:42.17 | tash | does that make sense? |
| 06:42.31 | tash | no, those are what is actually moving the .call around, etc |
| 06:42.37 | p3nguin | Yes, I just didn't get it. |
| 06:42.59 | tash | so, http://pastebin.com/vzrtTjwj look right based on that? |
| 06:43.15 | p3nguin | no |
| 06:43.18 | tash | dman |
| 06:43.23 | tash | s/dman/damn |
| 06:43.23 | p3nguin | Once Hangup() runs, that's the end of the line. |
| 06:43.28 | tash | eff |
| 06:43.31 | tash | forgot to delete that |
| 06:43.38 | tash | so, if I remove line 8, I'm good? right? |
| 06:43.46 | p3nguin | You need to move it down, not remove it. |
| 06:43.59 | tash | down to just before line 12? |
| 06:44.15 | p3nguin | You have to have the Hangup() before (done) to prevent that part from running. |
| 06:44.45 | p3nguin | Otherwise it'll reset the count back to 5 which is the start. |
| 06:45.08 | tash | http://pastebin.com/UKY9z72Z |
| 06:46.07 | p3nguin | So now you have 5 seconds to enter 1 or 2. If you fail to press 1 or 2 within the WaitExten time, the thing plays around with the files and runs a script, then hangs up. |
| 06:46.22 | tash | yeah, sounds like what I want |
| 06:46.24 | tash | going to test |
| 06:48.25 | tash | ok, got the call, didn't press 1 or 2 ... call file got manipulated around via my scripts and cron and is back in /var/spool/asterisk/outgoing |
| 06:48.42 | tash | which is what I want...now I need to wait for a few more calls to come through and see if the counter is working |
| 06:48.59 | p3nguin | "database show counter" in your asterisk CLI |
| 06:49.22 | tash | database show counter |
| 06:49.23 | tash | /counter//counter : 2 |
| 06:49.23 | tash | 1 results found. |
| 06:49.31 | tash | sorry for the crap paste |
| 06:49.42 | p3nguin | ${CALLFILE} is null |
| 06:50.26 | tash | shit |
| 06:50.42 | tash | I meant PLAYFILE the whole time I was telling you CALLFILE |
| 06:50.51 | tash | was I telling you CALLFILE was the unique ID? |
| 06:51.13 | tash | scrolls up |
| 06:51.15 | tash | indeed I was |
| 06:51.17 | tash | I'm sorry |
| 06:52.08 | tash | fixing that now |
| 06:54.44 | *** join/#asterisk tyman_ (~tyler@173-12-219-179-Fresno.hfc.comcastbusiness.net) |
| 06:55.34 | tash | p3nguin: still there? looks better now |
| 06:55.42 | p3nguin | I'm here. |
| 06:55.58 | tash | schedulife01*CLI> database show counter |
| 06:55.58 | tash | /counter/+17854102216-201102220033.00-305-20110222005341/counter: 4 |
| 06:55.58 | tash | /counter//counter : 2 |
| 06:56.16 | tash | getting the second call now ... I changed to every 2 minutes to test |
| 06:56.51 | tash | and the counter went down by 1 :) |
| 06:57.09 | tash | 6 mores minutes or so I s'pose and I'll know if it works the way I want it to |
| 07:06.56 | tash | p3nguin: it *mostly* worked |
| 07:07.37 | tash | when the counter got to 0, I still got 1 more call. But, it was a call with no audio, I answered then it hung up. I was watching the asterisk console too ... maybe I need to tweak something else in the dial plan |
| 07:07.43 | tash | but, I really need to get some zzz's |
| 07:07.47 | tash | you gonna be on tomorrow? |
| 07:08.22 | p3nguin | I'm sure I'll be on at some point. I'm usually in and out during the day. |
| 07:08.32 | tash | k |
| 07:09.04 | p3nguin | I think I know what needs to be changed. |
| 07:09.18 | tash | oh? |
| 07:09.30 | p3nguin | You're currently calling the phones, and THEN connecting them to the dialplan. |
| 07:09.35 | tash | I'll stay up if you think it's quick ... look at me, you are the one helping and I'm running off to bed :P |
| 07:09.38 | tash | yes |
| 07:10.02 | p3nguin | So the phone can answer, then the dialplan says, "You're at zero, we're ending now." But it's too late; the phone has already answered. |
| 07:10.12 | tash | right |
| 07:10.26 | p3nguin | This is where I would reverse my call file. |
| 07:11.16 | p3nguin | I'd connect it to a local channel which starts the dialplan, and then dial an extension which is a victim's phone number. |
| 07:11.40 | mzb | bashes out a p3nguin medal |
| 07:11.51 | tash | no shizz |
| 07:11.54 | p3nguin | It could affect the logic of what we devised, but it shouldn't be too hard to correct. |
| 07:11.57 | tash | p3nguin has been the bomb tonight |
| 07:12.06 | tash | well, another night? |
| 07:12.11 | tash | I'm freakin' tired :P |
| 07:12.28 | tash | you can gmail me if you'd like too |
| 07:12.38 | p3nguin | Can we reverse it and see if that solves it? If not, go to bed and fix it later. |
| 07:12.53 | tash | yeah ... I wonder if I'll have to change my PHP script |
| 07:13.05 | tash | it is creating the .call dynamically and moving to the asterisk outgoing dir |
| 07:13.23 | p3nguin | I would think so. Your script will need to change the info it is writing. |
| 07:13.43 | p3nguin | Show me an example of your call file again. |
| 07:13.53 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 07:14.30 | tash | http://pastebin.com/rtRzFvDN |
| 07:14.44 | tash | that is directly from my PHP script that is dynamically generating the .call |
| 07:14.54 | tash | later in the script, I move it to /var/spool/asterisk/outgoing |
| 07:15.19 | tash | ignore all the vars if you'd like |
| 07:15.50 | tash | you can at least see what parameters I'm putting in the .call (if parameters is even the correct asterisk terminology for things like "Retrytime") |
| 07:17.20 | p3nguin | http://pastebin.com/11GUBFsr |
| 07:17.54 | p3nguin | Then we create an extension pattern which dials outbound via "bandwidth" peer. |
| 07:18.16 | tash | I just added that code to my php script |
| 07:19.02 | p3nguin | Is your php script generating $prefix$cc$multiple_phone as +17854102216 ? |
| 07:19.13 | tash | yes |
| 07:19.14 | p3nguin | or just 7854102216 ? |
| 07:19.37 | tash | prefix is +, $cc is 1, $multiple_phone is 7854102216 |
| 07:19.45 | p3nguin | okay |
| 07:19.56 | tash | it is only $multiple_phone due to some arrays and things |
| 07:20.13 | tash | ignore my lame variable naming :P |
| 07:22.30 | p3nguin | I guess we need to add some pattern that matches all numbers the script could generate. Hopefully this'll work: exten => _+1NXXNXXXXXX,1,Dial(SIP/bandwidth/${EXTEN}) |
| 07:22.44 | p3nguin | I hope you're only dialing North America. :/ |
| 07:22.49 | tash | I am for now |
| 07:22.55 | tash | no plans on INTL for a long time |
| 07:23.11 | p3nguin | This may be a problem later and you'll end up making more patterns. |
| 07:23.17 | tash | yeah |
| 07:23.23 | tash | where do I put that ^ |
| 07:23.54 | p3nguin | You can put it below all the mycaller extension lines in the outbound-calls context. |
| 07:23.55 | *** join/#asterisk _zoom_ (~zero@196.1.219.211) |
| 07:24.22 | _zoom_ | hello, is it doable to integrate asterisk with stunnel? |
| 07:24.48 | p3nguin | Well shit. |
| 07:25.00 | tash | p3nguin: ? |
| 07:25.01 | p3nguin | I just realized that you don't have any AMD() stuff. |
| 07:25.02 | ChannelZ | shits |
| 07:25.40 | p3nguin | This is going to be another issue... Asterisk is going to start mycaller before the victim answers the phone. |
| 07:26.12 | tash | yeah ... |
| 07:26.21 | tash | let's talk another night? I'm sorry ... losing focus |
| 07:26.23 | tash | need some zzz's |
| 07:26.25 | tash | sleep on it? |
| 07:26.33 | p3nguin | I lost focus a couple hours ago. |
| 07:26.36 | tash | and thanks a million for what you've helped me with tonight! |
| 07:26.37 | tash | haha |
| 07:26.54 | p3nguin | At least you've got some ideas now. |
| 07:27.10 | tash | definitely |
| 07:27.36 | tash | g'night p3nguin |
| 07:27.40 | tash | maybe we'll talk tomorrow |
| 07:27.44 | p3nguin | If you're around in the daytime, other people can help you solve this whole scenario. |
| 07:27.54 | tash | yep, i should be |
| 07:28.20 | tashZZZs | less available than after 5PM, but I'll be around |
| 07:28.27 | tashZZZs | nite |
| 07:28.52 | WIMPy | Good morning. |
| 07:28.57 | *** join/#asterisk topriddy (~Seamfix@41.58.59.97) |
| 07:29.12 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:29.14 | schmidts | good morning |
| 07:29.42 | topriddy | good morning |
| 07:30.57 | topriddy | i asked this question yesterday: i want to create a solution where i can make phone calls to thousands of gsm phone nos pooled from a database, about 8 calls simultaneously. when users pick the call all i do is play a prerecorded audio for them. |
| 07:31.41 | topriddy | i know this may require some hardware and writing of codes, (i do java and heard of AGI protocol), anyone here to guide me to making a cost-effective solution??/ |
| 07:32.10 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 07:32.41 | WIMPy | There are probably quite a few ready made solutions for that. |
| 07:33.03 | schmidts | topriddy search for war dialer software, this is what you need |
| 07:33.38 | topriddy | schmidts: how about hardware? dont i have to pay for call stoo? |
| 07:34.09 | WIMPy | There are many ways to interface to the PSTN or GSM directly. |
| 07:34.26 | WIMPy | You either need hardware or an ITSP. |
| 07:34.30 | WIMPy | ~ITSP |
| 07:34.31 | 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. |
| 07:36.23 | schmidts | topriddy sure you have to pay for calls, as wimpy said it depends on your connection to the PSTN or GSM Network. You can use an ITSP to call out, or you can use a GSM Channel bank with several Sim cards from your GSM Network Provider, but you will not find anyone who will terminate calls to the gsm network for free |
| 07:37.43 | topriddy | ITSP would imply i'll probably need internet connectivity to route calls to them. |
| 07:39.01 | WIMPy | Indeed |
| 07:39.48 | topriddy | well i'll rather buy a gsm channel bank with sims...got any to recommend? |
| 07:40.50 | WIMPy | http://www.portech.com.tw/p3-product1.asp?Cid=6 |
| 07:41.12 | WIMPy | You can find these things on ebay. |
| 07:41.27 | schmidts | portech are cool and easy to use |
| 07:42.25 | schmidts | topriddy BUT and thats really a big BUT you have to make sure, you dont break the contract with your gsm provider if you use this in a channel bank, most of them dont want this |
| 07:43.45 | WIMPy | Shouldn't be a problem if you pay the standard fees, i.e. the highest price. |
| 07:43.45 | topriddy | schmidts: why? |
| 07:43.47 | *** join/#asterisk Tim_Toady (~moi@77.49.3.6.dsl.dyn.forthnet.gr) |
| 07:44.32 | WIMPy | But if you think you can get it cheaper via some on-net flat or whatever, they will probably pull the virtual plug pretty fast. |
| 07:45.04 | topriddy | WIMPy: i intend paying standard fees. what do you think of me interacting with asterisks through standard java, http, and AGI protocol? dynamically generating dial plans? |
| 07:45.37 | WIMPy | Why do you want to dynamically generate dialplans? |
| 07:45.58 | topriddy | WIMPy: i am dynamically calling a list of numbers |
| 07:46.31 | topriddy | i have an algorithm in my head where i simultaneously queue of numbers, depending on networks....same network calls are usually cheaper. |
| 07:46.36 | WIMPy | That's just calling a fixed dialplan, unless you intend to do different things on every call. |
| 07:46.49 | *** join/#asterisk qjb (~qjb@a83-163-158-168.adsl.xs4all.nl) |
| 07:49.36 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 07:50.25 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:51.55 | topriddy | WIMPy: i;ll check my asterisks book again |
| 07:52.03 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:52.28 | topriddy | WIMPy: but of course i wont definitely generate the plans for one million nos at once would i? doesnt sound right |
| 07:52.49 | topriddy | so am probably generating fixed dial plans then....dynamically. :D |
| 07:53.12 | WIMPy | That might only be one single extension rule. |
| 07:53.27 | topriddy | i just want to be sure AGI protocol and communicating through http with my java is way to go. |
| 07:53.31 | WIMPy | Only once, before you start. |
| 07:54.02 | WIMPy | But AGI is only for active calls. |
| 07:54.22 | WIMPy | If you want to generate calls, you need to use AMI or generate call files. |
| 07:55.35 | topriddy | WIMPy: still doesnt sound nice, if a number isnt available at the moment, my program should mark it as such and queue it up for later trying again. i can give a number a max of 5 trials before trying again...intellisense. active calls? |
| 07:56.05 | WIMPy | You probably should use AMI. |
| 07:56.08 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 07:56.15 | WIMPy | You can get the results that way as well. |
| 07:56.51 | topriddy | WIMPy: thanks...information about AMI is in asterisk book right? |
| 07:56.57 | topriddy | ~book |
| 07:56.58 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 07:57.07 | eMBee | good afternoon |
| 07:57.10 | WIMPy | I think so. |
| 07:57.37 | topriddy | WIMPy: or you have a book suggestion? |
| 07:58.17 | WIMPy | I'm not in to books any more. |
| 07:58.59 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 07:59.02 | eMBee | got asn ISDN line today and is now struggeling to set it up |
| 08:00.35 | topriddy | WIMPy: i'll just google then |
| 08:01.21 | WIMPy | eMBee: Was that a question? |
| 08:01.49 | *** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 08:02.05 | eMBee | WIMPy: yes, sort of, i don' |
| 08:02.18 | eMBee | i am not even sure what to ask |
| 08:02.38 | eMBee | i am mainly wondering about the hardware config settings |
| 08:02.43 | WIMPy | Maybe you tell us what you've got then. |
| 08:03.43 | eMBee | well, which details are helpful? it's an ISDN10 line in singapore, and a digium card |
| 08:04.19 | eMBee | the hardware is recognized, and after i plugged in a cat5 the isdn box confirmed that there is a connection |
| 08:04.26 | topriddy | WIMPy: yesterday i browsed through portech site, kinda heard getting price information from them without a card, poorly designed site i'll say. i'll probably try ebay. |
| 08:05.07 | eMBee | but when i try incoming calls i get a ringing signal on the phone but asterisk shows nothing in the logs about a call coming in |
| 08:05.31 | WIMPy | eMBee: start dahdi_genconf and then manually edit chan_dahdi.conf to restrict the cahnnels to your partial line. And make sure you've got the right signalling, wahtever is used in Singapore. |
| 08:07.47 | eMBee | ok,thanks, i'll try that. how would i go about to find the infortmation about which signaling i need to use. had no luck with google so far |
| 08:08.48 | WIMPy | Your telco should be able to tell you. |
| 08:09.56 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:10.29 | WIMPy | From a quick peek at google, it looks like you use DSS1 there. |
| 08:11.32 | eMBee | would you mind telling me what search string you used? :-) |
| 08:11.45 | WIMPy | Err, thats euroisdn in Asterisk terms. |
| 08:11.53 | eMBee | ah |
| 08:11.57 | WIMPy | 'singapore isdn signalling' |
| 08:13.13 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 08:18.23 | WIMPy | According to Wikipedia: Europe, Australia, Brazil, Iran, India, Israel, New Zealand, Peru, Singapore, South Africa, Sri Lanka, Taiwan, Palestine and United Arab Emirates |
| 08:21.02 | eMBee | i could not make anything out of the search, which pages are you looking at? |
| 08:21.38 | WIMPy | http://en.wikipedia.org/wiki/DSS1 |
| 08:23.28 | *** join/#asterisk Siblor (~putthibon@58.136.24.13) |
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| 09:25.59 | benngard | stateinterface parameter in AddQueueMember drives me crazy :( |
| 09:26.27 | benngard | i am trying to "monitor" 2 extensions but cant figure out howto do it |
| 09:28.14 | benngard | exten => 0424449647,hint,SIP/0424449647&SIP/0106024647 |
| 09:28.42 | benngard | that hint works as far as i can se |
| 09:29.40 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 09:29.57 | benngard | if i dial from one of the extensions: core show hint 0424449647, 0424449647@support : SIP/0424449647&SIP/0 State:InUse Watchers 0 |
| 09:31.47 | benngard | but i cant figure out howto get that hint into AddQueueMember |
| 09:40.43 | *** join/#asterisk lftsy (~lftsy@pul-lav-fw-so-01-x1.vtxnet.net) |
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| 10:42.43 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 10:43.41 | eject_ck | Hi, all. What * application supports playing ogg network stream ? |
| 10:47.02 | eject_ck | I need something like this exten => _999,1,Answer() |
| 10:47.17 | eject_ck | exten => _999,2,MP3Player(http://stream.m-1.fm/ogg) |
| 10:49.01 | *** join/#asterisk mpe (~mpe@office.ipvision.dk) |
| 10:49.24 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 10:50.34 | eject_ck | How can I run mplayer instead ? |
| 10:52.03 | XDS|Work | what would be put in the content field for a dns record being sent to _udp_.domain.com , for SRV |
| 10:55.33 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
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| 11:09.01 | *** join/#asterisk saxa (~sasa@host242-95-static.223-217-b.business.telecomitalia.it) |
| 11:10.46 | *** part/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 11:11.00 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 11:11.44 | BlackBishop | anyone knows a pdu cli decoder for linux ? :/ |
| 11:12.01 | shapr | pdu? |
| 11:12.55 | BlackBishop | yeah .. stuff like "07914400000000F001000B811000000000F000001A2E97CBE572B95C2E97CBE572B95C2E97CBE572B95C2E17" :) |
| 11:13.26 | BlackBishop | found a nice javascript one .. not an option though... |
| 11:13.32 | BlackBishop | ports it to php |
| 11:15.02 | shapr | What's the javascript decoder? |
| 11:15.37 | BlackBishop | http://www.rednaxela.net/pdu.php |
| 11:20.45 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 11:29.36 | bjhaid | i would like to know if i can get to put outgoing call through a group of sip channels, say i am connected to an ITSP and i have 20 sip channels, how can i make sure the outgoing calls are sent to different sip channels. |
| 11:46.45 | *** join/#asterisk domedan (~domedan@c83-253-183-67.bredband.comhem.se) |
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| 12:11.48 | wdoekes2 | eject_ck: you can use musiconhold for this quite easily |
| 12:12.52 | *** join/#asterisk topriddy (~Seamfix@41.58.59.97) |
| 12:15.31 | topriddy | i need a little help brainstorming. i am researching a gps solution where i can get a user to carry a gps device around with him/her and then device sends info to a server i can retrieve info from and use in a bigger app. my question is does devices currently support this kind of network call? does it use mobile network or how exactly is this done? |
| 12:17.31 | eject_ck | wdoekes2: let me try |
| 12:17.43 | leifmadsen | topriddy: well the only way I can think of this is to use a cell phone with GPS built in, which then sends the data via the data network (cellular). See Google Maps with Latitude as an example. |
| 12:18.56 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 12:19.07 | leifmadsen | XDS|Work: look at this page for SIP URI DNS configuration: http://ofps.oreilly.com/titles/9780596517342/ch12.html#asterisk-ICR-SECT-1 |
| 12:19.35 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 12:19.50 | *** join/#asterisk OlafsenM (~mark.olaf@193.198.31.85) |
| 12:19.57 | OlafsenM | "10 of 10 threads accounted for with 101 dynamic threads" |
| 12:20.04 | topriddy | leifmadsen: i know of that and implemented this a blackberry app based on it http://watchdog-oaife.appspot.com/wicket/home |
| 12:20.19 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 12:20.31 | OlafsenM | my astbox has 47 active IAX channels, but threadpool is used up |
| 12:20.54 | OlafsenM | [12:42:07] Saša Obadić: a kaj onaj spy zeza ast i dalje? |
| 12:20.54 | OlafsenM | [12:42:13] Kresimir Grofelnik: cini se :/ |
| 12:20.55 | OlafsenM | [12:42:17] Kresimir Grofelnik: moram to rijesit sad |
| 12:20.55 | OlafsenM | [12:42:26] Kresimir Grofelnik: cim emisija prodje |
| 12:20.55 | OlafsenM | [12:42:56] Kresimir Grofelnik: random |
| 12:20.55 | OlafsenM | [12:43:07] Kresimir Grofelnik: nakon 2 ili vise puta spyanja istog kanala :/ |
| 12:21.05 | OlafsenM | oops |
| 12:21.08 | OlafsenM | sorry, wrong paste |
| 12:21.47 | OlafsenM | http://pastebin.com/r8q6YSd6 |
| 12:28.14 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
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| 13:27.41 | *** join/#asterisk nicola_pav (~nicola@mail2.tikalnetworks.com) |
| 13:27.45 | *** join/#asterisk qvsqvs (~anonymous@196.214.133.227) |
| 13:28.46 | nicola_pav | hello. i ran dmesg to find a lot of the following random messages: |
| 13:28.51 | nicola_pav | hello dahdi: Disabled echo canceller NLP because of CED rx detected on channel xx |
| 13:28.58 | qvsqvs | Hi Guys, how can i fix the save button on creating new trunk on the asterisk gui |
| 13:29.03 | nicola_pav | each time different chaneel of course |
| 13:29.11 | nicola_pav | any ideas how to stop those messages? |
| 13:30.37 | Tim_Toady | nicola_pav u get these messages when u receive a fax and echo canceler gets disabled |
| 13:30.43 | Tim_Toady | so its nothing to worry about |
| 13:31.26 | nicola_pav | Tim_Toady: no faxes :S |
| 13:31.35 | Tim_Toady | thes false fax detection |
| 13:31.58 | Tim_Toady | if u dont use fax then disable fax suport in chan_dahdi.conf |
| 13:32.41 | Tim_Toady | setting something like faxdetect=no in the general section of chan_dahdi.conf |
| 13:32.58 | nicola_pav | iax trunk between two PBXs |
| 13:33.09 | nicola_pav | Tim_Toady: does it affect the calls? |
| 13:33.26 | nicola_pav | quality of calls? |
| 13:33.44 | Tim_Toady | the message u pasted is reffering to dahdi channels |
| 13:34.18 | nicola_pav | so nothing to worry about? |
| 13:34.38 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 13:34.49 | Tim_Toady | depends on ur setup, u use these dahdi channels at all, or its just iax? |
| 13:35.30 | nicola_pav | i use the dahdi channels |
| 13:35.45 | nicola_pav | two asterisk servers |
| 13:35.48 | nicola_pav | in two countries |
| 13:35.54 | nicola_pav | both have pri cards |
| 13:36.02 | nicola_pav | connected together with iax trunk |
| 13:36.14 | Tim_Toady | then disable the fax detection if u dont use faxes |
| 13:36.15 | nicola_pav | on of the servers, i am getting those messages |
| 13:36.32 | nicola_pav | i will try it yeah |
| 13:36.33 | nicola_pav | thanks |
| 13:36.45 | nicola_pav | i hope it does not affect the quality |
| 13:36.57 | qvsqvs | hi any way to fix the save button in the gui |
| 13:37.06 | Tim_Toady | only the quality of the faxes :P |
| 13:37.06 | nicola_pav | lately, we r facing loew waulity calls |
| 13:38.20 | nicola_pav | which we dont use :) |
| 13:38.24 | *** join/#asterisk qjb (~qjb@2001:888:17b4:0:21f:5bff:fef1:f005) |
| 13:38.24 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 13:40.16 | kaldemar | qvsqvs: which gui? |
| 13:40.23 | qvsqvs | Asterisk Build: |
| 13:40.23 | qvsqvs | Asterisk/1.6.2.16.1 |
| 13:40.23 | qvsqvs | Asterisk GUI-version : SVN--rexported |
| 13:40.32 | qvsqvs | this is what i have in the home screen |
| 13:40.42 | qvsqvs | how do i check the guo version |
| 13:41.10 | Tim_Toady | asterisk gui is still mainteined/supported/working? |
| 13:41.21 | nicola_pav | Tim_Toady: what about those msgs: dahdi: Disabled echo canceller because of tone (tx) on channel xx |
| 13:41.32 | nicola_pav | it happens for rx also |
| 13:41.40 | kaldemar | iirc, not maintained anymore. |
| 13:41.49 | pabelanger | Tim_Toady: Yes, espiceland is currently maintaining it |
| 13:41.59 | Tim_Toady | and it works? :P |
| 13:44.29 | alvdav | greetings |
| 13:44.30 | pabelanger | yes |
| 13:44.51 | nicola_pav | Tim_Toady: any hints on dahdi: Disabled echo canceller because of tone (tx) on channel xx |
| 13:45.32 | Tim_Toady | im not sure about this one nicola_pav disable fax detection and check if they still appear |
| 13:45.48 | Tim_Toady | also monitor ur cals to see if echo canceler is active etc |
| 13:45.50 | nicola_pav | those msgs r different |
| 13:46.01 | nicola_pav | on another server |
| 13:46.08 | nicola_pav | not related to the fax |
| 13:46.13 | *** join/#asterisk chasing`Sol (~chasingSo@82.201.135.150) |
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| 13:49.30 | *** join/#asterisk garymc (~chatzilla@host81-148-15-59.in-addr.btopenworld.com) |
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| 13:56.00 | *** part/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 13:59.09 | *** join/#asterisk moy (~moy@CPE003048b11058-CM0026f396812d.cpe.net.cable.rogers.com) |
| 14:01.11 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 14:01.13 | Katty | morning |
| 14:01.24 | beek | hugs Katty |
| 14:02.21 | schmidts | morning katty |
| 14:07.33 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:19.15 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:21.44 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 14:22.34 | killown | If I got registry rejected of my provider, how I do to retry to registry again without to have to restart the asterisk service? |
| 14:24.34 | killown | sorry I was talking about "VOIP" registry |
| 14:24.37 | pabelanger | killown: For SIP? It should automatically retry after x seconds (usually 300) |
| 14:24.39 | *** join/#asterisk zpertee (~yaaic@111.sub-174-252-219.myvzw.com) |
| 14:25.17 | killown | pabelanger I'd like to retry after 30 seconds, how to change the retry settings? |
| 14:26.25 | pabelanger | killown: defaultexpiry=30, sip.conf |
| 14:26.40 | killown | pabelanger, thank you |
| 14:27.10 | Katty | hugs beek |
| 14:27.13 | Katty | hugs schmidts |
| 14:27.15 | Katty | Qwell: ping |
| 14:27.16 | *** join/#asterisk erinspice (~erin@nat/digium/x-sssddinyeuaezhgm) |
| 14:27.19 | pabelanger | but your ITSP may not let you. Since they also have the ability to control the default time for registration attempts |
| 14:28.07 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 14:28.39 | killown | pabelanger, there is a way to retry it manually? |
| 14:28.57 | killown | SIP retry registry, something like that. |
| 14:29.05 | pabelanger | no |
| 14:29.50 | *** join/#asterisk Tim_Toady (~moi@77.49.3.6.dsl.dyn.forthnet.gr) |
| 14:30.24 | retentiveboy | Perhaps I'm paranoid but I'd like to use random usernames and passwords for SIP stations setup via users.conf. Is this possible? I want to end up with SIP entries like [1001] but have stations login using the randomly generated credentials. |
| 14:32.05 | killown | pabelanger, very stranger, this has a sip unregister Unregister , but not a sip register Resgistry or register retry |
| 14:32.34 | pabelanger | killown: what is your actual issue? |
| 14:33.21 | killown | pabelanger, I have got registry rejected and it stuck in that.. so I needed to restart the asterisk service to registry my account |
| 14:34.08 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 14:38.01 | *** part/#asterisk tashZZZs (~Tommy@ks-76-7-1-196.sta.embarqhsd.net) |
| 14:38.51 | *** part/#asterisk topriddy (~Seamfix@41.58.59.97) |
| 14:39.43 | *** join/#asterisk tash (~Tommy@ks-76-7-1-196.sta.embarqhsd.net) |
| 14:42.29 | *** join/#asterisk Mhaddog__ (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 14:44.12 | schmidts | killown sip reload helps you |
| 14:44.48 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:46.15 | killown | schmidts thank you |
| 14:48.19 | schmidts | your welcome ;) |
| 14:52.04 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 14:54.39 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:59dd:ad2:e592:4f46) |
| 14:58.33 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 14:58.48 | *** join/#asterisk dajhorn (~dajhorn@adsl-75-17-124-41.dsl.rcsntx.sbcglobal.net) |
| 15:00.10 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:06.15 | *** join/#asterisk matrix3000 (~matrix@rrcs-70-61-255-226.central.biz.rr.com) |
| 15:06.21 | matrix3000 | Hey everyone |
| 15:06.33 | matrix3000 | why would asterisk ignore port 5060 requests |
| 15:07.06 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-gnwcswpyvspakiiv) |
| 15:07.23 | matrix3000 | the router and forwarding them without a problem we can see them being sent to the asterisk box but asterisk isn't acknowledging them |
| 15:07.40 | matrix3000 | its not showing up in logs, but it is showing up in tcpdump and iptables |
| 15:08.17 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 15:08.21 | retentiveboy | matrix3000: tried "sip set debug on"? |
| 15:08.37 | LemensTS | What do you guys prefer for an 800 sip number? |
| 15:09.13 | matrix3000 | k |
| 15:09.48 | matrix3000 | retentiveboy: thanks I will do that right now, we are also currently running 50 extensions |
| 15:09.58 | matrix3000 | and 17 active channels |
| 15:11.04 | retentiveboy | matrix3000: may look at using "sip set debug peer" instead |
| 15:11.43 | matrix3000 | retentiveboy: what is teh difference, one will output more than the other? |
| 15:11.55 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:12.15 | retentiveboy | matrisx3000: later is limited to a single peer |
| 15:12.43 | retentiveboy | matrix3000: i.e. "sip set debug peer 1001" |
| 15:13.38 | matrix3000 | its the calls in only |
| 15:14.11 | matrix3000 | everything in the network can connect |
| 15:14.13 | retentiveboy | I'm no SIP expert. Just sharing the tools I've used to troubleshoot begore |
| 15:14.17 | matrix3000 | ah |
| 15:14.24 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 15:14.47 | matrix3000 | what is the "number" from |
| 15:14.59 | retentiveboy | peer name |
| 15:15.44 | retentiveboy | I find the tab-completion in the console pretty damn handy |
| 15:16.14 | retentiveboy | Whoever whipped that up deserves a big gold star :) |
| 15:22.19 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 15:22.21 | *** join/#asterisk jkroon (~jkroon@dsl-241-225-112.telkomadsl.co.za) |
| 15:24.44 | matrix3000 | how do I set up asterisk itself to operate properly behind a firewall, all ports needed are being forwarded properly |
| 15:25.52 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 15:26.10 | matrix3000 | when i call in it says "all circuits are busy" |
| 15:26.15 | matrix3000 | what does that mean on asterisk |
| 15:26.44 | _Corey_ | Depends altogether on where the "all circuits busy" message is coming from |
| 15:27.10 | Katty | hmm. |
| 15:27.17 | Katty | i need help picking out a derby name. Katty something |
| 15:27.20 | Katty | what do you guys think |
| 15:27.31 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:28.18 | beek | Katty-nine tails? |
| 15:28.31 | _Corey_ | Katty Scratches |
| 15:28.38 | Katty | beek: ooohmy |
| 15:28.39 | matrix3000 | when you call asterisk from outside |
| 15:28.41 | Katty | beek: YES |
| 15:28.44 | Katty | beek: love it! |
| 15:28.46 | _Corey_ | lol |
| 15:28.48 | matrix3000 | when you call the external number |
| 15:28.54 | matrix3000 | but it was working a few hours ago |
| 15:28.58 | matrix3000 | and nothing has changed |
| 15:29.00 | beek | :D |
| 15:29.56 | carrar | http://rollergirlthebook.com/name_generator |
| 15:30.01 | matrix3000 | _Corey_: when you call from the outside into asterisk, you would get that message, but it was working earlier today |
| 15:30.28 | matrix3000 | _Corey_: we also are using trixbox asterisk server |
| 15:30.48 | _Corey_ | matrix3000: Well, your carrier is probably playing the message because it can't reach Asterisk. You may want to check your config to make sure it's registering properly, etc |
| 15:30.56 | *** join/#asterisk pressureman (~daniel@2001:470:1f0b:8a6:222:68ff:fe0b:4757) |
| 15:31.08 | pressureman | anybody here using yealink/tiptel phones with ipv6? |
| 15:31.20 | matrix3000 | _Corey_: we are registering and able to make outbound calls without any problem |
| 15:33.37 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:34.04 | _Corey_ | matrix3000: It worked at one point, I would suspect a timeout of some sort |
| 15:34.18 | _Corey_ | either way, sniff sip traffic and verify what you're getting |
| 15:35.24 | matrix3000 | _Corey_: well, the firewall is sending the 5060 packets to asterisk properly and tcpdump on the asterisk server is logging them as well as iptables, but for some reason we think asterisk is ignoring them |
| 15:36.12 | _Corey_ | debug sip in asterisk to see what asterisk sees |
| 15:37.21 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 15:40.10 | *** join/#asterisk fofware (~fofware@wdctf.siup.gov.ar) |
| 15:52.35 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 15:52.44 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 15:56.13 | *** join/#asterisk ubuspire (~ubuspire@41.190.129.207) |
| 16:00.16 | *** join/#asterisk StaRetji (~BigAll@91.143.222.166) |
| 16:01.00 | StaRetji | folks, I'm having problem where dtmf tone coming from my dect phone (connected trough dahdi) are not recognized |
| 16:01.22 | StaRetji | I've found this article http://astrecipes.net/index.php?n=248 Changing DTMF tone frequency in Asterisk |
| 16:01.49 | StaRetji | I'm desperate so I thought maybe this is my problem |
| 16:01.57 | *** join/#asterisk calhoun (~calhoun@host90-152-3-218.ipv4.regusnet.com) |
| 16:02.15 | StaRetji | Will applying that patch recognize tone codes (buttons) I press on my dect phone? |
| 16:02.28 | StaRetji | Any help would be highly appreciated |
| 16:02.31 | StaRetji | thx :) |
| 16:03.00 | calhoun | hey, I'm trying to figure out asterisk, got the now iso installed, and got two phones connected via skinny, but how do I write rules that will allow me to call between each other? |
| 16:05.41 | *** join/#asterisk marienz (~marienz@freenode/staff/marienz) |
| 16:05.52 | Tozz_ | calhoun: that is very basic setup. Perhaps you can read a HOWTO |
| 16:06.57 | nestAr | hrmm.. asterisk crashed last night. that's like.. never happened before. |
| 16:07.32 | Tozz_ | good think we have safe_asterisk :> |
| 16:07.41 | Tozz_ | s/think/thing/ |
| 16:08.12 | calhoun | Do you have an example of a howto that covers that, cause the ones ive found stop at connecting the phone. |
| 16:09.03 | StaRetji | http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf |
| 16:09.10 | chazzam | ~newbook |
| 16:09.10 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 16:09.43 | Tozz_ | http://jeremy-mcnamara.com/asterisk/how-to-configure-asterisk/ |
| 16:09.45 | Tozz_ | first page on google |
| 16:09.55 | Tozz_ | but i know google is difficult |
| 16:10.43 | ubuspire | Hi, am running asterisk 1.6.2 with libss7 and sangoma card. How do I advertise a pointcode in hexadecimal notation to an ss7 media gateway |
| 16:11.14 | *** join/#asterisk skrusty (~ben@93-97-20-22.zone5.bethere.co.uk) |
| 16:11.46 | StaRetji | How about my problem. I'm having problem that sometimes (after reboot) DTMF tones are not recognized. |
| 16:12.13 | StaRetji | How to debug that? |
| 16:12.33 | ezq | hi, if there a posibly to disable moh for one queue? |
| 16:13.11 | calhoun | so should be dial(skinny/<line>) ? |
| 16:13.14 | *** join/#asterisk digitolx (~ssureshot@dynamic-acs-24-154-156-11.zoominternet.net) |
| 16:13.49 | StaRetji | I suspect that my phone's DTMF (ordinary analog phone) is not recognized properly by asterisk. Sometimes I press 1234 and it will accept only tone 1 |
| 16:13.55 | StaRetji | any ideas? |
| 16:14.33 | *** join/#asterisk salimb (~chatzilla@83-244-177-2.cust-83.exponential-e.net) |
| 16:14.49 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 16:16.13 | digitolx | can anyone point me in the direction of why I'm getting this error.. I'm testing upgrade from 1.2 to 1.8.. it has to do with the extensions.conf file but I can't put my finger on it |
| 16:16.22 | digitolx | WARNING[3981] pbx.c: Maximum PBX stack exceeded |
| 16:16.56 | StaRetji | bah, never got help on this channel. I'm leaving never to come back here again :P |
| 16:17.10 | *** part/#asterisk StaRetji (~BigAll@91.143.222.166) |
| 16:17.27 | leifmadsen | excellent |
| 16:17.52 | Tozz_ | he should get his money back too! |
| 16:18.36 | leifmadsen | perhaps he should wait more than a whole 3 minutes |
| 16:19.13 | wdoekes2 | digitolx: gosub or macro recursion too deep? |
| 16:19.25 | leifmadsen | wdoekes2: +1 |
| 16:19.37 | leifmadsen | well, it'd just be Macro() likely |
| 16:19.46 | leifmadsen | GoSub() shouldn't have a recursion issue |
| 16:19.47 | wdoekes2 | yeah ;) |
| 16:19.54 | wdoekes2 | did it even exist in 1.2? |
| 16:20.04 | leifmadsen | Macro(... Macro(.... func_odbc)) will kill the stack |
| 16:20.08 | leifmadsen | GoSub()? No. |
| 16:20.27 | digitolx | hmm, I guess Ill have to filter through one by one that sux lol |
| 16:20.40 | wdoekes2 | digitolx: you should probably 'core set verbose 20' and look at the verbose output before it complains about the recursion |
| 16:20.55 | leifmadsen | digitolx: well just look at the logging output with console logging enabled prior to the crash and it should show you exactly where it was executing when it dies |
| 16:21.04 | leifmadsen | what wdoekes2 said :) |
| 16:21.08 | leifmadsen | goes back to new issue triage |
| 16:21.11 | tash | hi guys, got a dialplan configured to set a counter for .call files - and after 5 calls to a user without their input (pressing 1 to confirm they heard the call), I no longer want to call them. This works, mostly, except after the counter is down to 0, I still get 1 more call without audio. I need to find a way to set the counter and prevent that 6th call instead of placing the call then realizing the counter is 0 and hanging up. Sorry for t |
| 16:21.25 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 16:21.37 | leifmadsen | tash: how are you setting the counter? |
| 16:21.46 | leifmadsen | if you're using GROUP() and GROUP_COUNT() you wouldn't have that issue |
| 16:22.00 | leifmadsen | as you're going to increment the counter with GROUP() prior to checking with GROUP_COUNT() |
| 16:22.08 | digitolx | roger that I will take another look.. |
| 16:22.23 | tash | leifmadsen: http://pastebin.com/jxiC6vHm |
| 16:22.36 | tash | I got help from p3nguin last night - I am not smart enough to figure this out on my own :P |
| 16:23.01 | tash | sorry, I need to step away for a few minutes, I'll be right back |
| 16:24.24 | *** join/#asterisk OneNarrowWay (~OneNarrow@ip4da1344b.direct-adsl.nl) |
| 16:25.55 | Katty | Qwell |
| 16:25.58 | Katty | QWELL |
| 16:27.33 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 16:27.33 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:28.27 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 16:28.35 | *** join/#asterisk Letoric (~Letoric@tpi-dfw-uc-f2-69-94-238-26.totalprocess.net) |
| 16:29.52 | leifmadsen | Katty: I killed him |
| 16:30.33 | Katty | :< |
| 16:30.36 | Katty | bad leif! |
| 16:33.28 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:33.41 | Faustov | A JIRA account has been created for you at: (...) <--- I got an email like this from digium but the server does not respond |
| 16:34.04 | Faustov | dear admins! |
| 16:34.42 | pabelanger | http://lists.digium.com/pipermail/asterisk-dev/2011-February/048117.html |
| 16:34.52 | pabelanger | Faustov: ^ |
| 16:35.06 | *** join/#asterisk neurosys (~neurosys@69.198.172.57) |
| 16:35.19 | leifmadsen | Faustov: ya, see mailing list |
| 16:35.19 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 16:36.11 | Faustov | ah ok |
| 16:36.17 | Faustov | thanks |
| 16:37.17 | LemensTS | Say I buy two 900mhz cordless phones, exact same model. I hook them both up in the same room. Each phone knows its base station, I assume because they have slightly different 900.0001mhz and 900.0002mhz or something? |
| 16:37.55 | *** join/#asterisk Godfather_ (~godfather@90.162.102.229) |
| 16:39.13 | digitolx | how can I list the modules that are installed and working ? |
| 16:39.22 | digitolx | installed and loaded I should say |
| 16:40.05 | leifmadsen | module show |
| 16:41.30 | digitolx | ah thank you |
| 16:42.29 | digitolx | here is the error just before the slew of pbx stack exceeded.. madd_pri_lockoptESC[0m: Unable to register extension '002', priority 2 in 'office-ext', already in use |
| 16:42.33 | tash | leifmadsen: am back |
| 16:44.13 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 16:52.59 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 16:53.14 | *** part/#asterisk pressureman (~daniel@2001:470:1f0b:8a6:222:68ff:fe0b:4757) |
| 16:54.07 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 16:55.02 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 16:57.36 | *** join/#asterisk puzzled_ (~patrick@puzzled.xs4all.nl) |
| 16:58.50 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 16:59.02 | kfife | Hey folks. Anyone have a recommendation for [real] CNAM? I've been using Metrostat for years, but looking to possibly change. Recommendatiosn? |
| 16:59.57 | leifmadsen | CNAM? |
| 17:00.19 | leifmadsen | oh CID lookup? |
| 17:00.36 | kfife | leifmadsen: You Canadians and your funny terms! :-) |
| 17:00.36 | *** join/#asterisk Ad-Hoc (~nimbus@193.92.82.70) |
| 17:01.00 | leifmadsen | kfife: \o/ |
| 17:02.10 | kfife | I know there are lots of 'free' name lookup type services that make a best-effort to get somethign, but the actual CNAM/LIDB database costs money to dip because it's licensed. That's the one I'm looking for. |
| 17:03.22 | leifmadsen | ya I've never done that kind of lookup so I'm not sure who to contact... wonder if one of the VoIP providers would offer that as a service to you? |
| 17:04.26 | kfife | Some do, most don't. Vitelity does, but they pre-populate it on the invite for specific DID's. Therefore it's more expensive if you have hundreds of DID's vs a straight CANM dip service. |
| 17:04.38 | kfife | (in my experience) |
| 17:05.45 | kfife | Metrostat allows me to dip via HTTP i.e. curl http://cnam1.edicentral.net/getcnam?dn=8285871000 |
| 17:06.27 | kfife | Anybody can dip that number for free without an account. |
| 17:08.05 | *** join/#asterisk Hanumaan (~Hanumaan@132.230.17.77) |
| 17:08.50 | leifmadsen | kfife: why are you changing? (just curious) |
| 17:08.59 | _Corey_ | kfife: How much does Metrostat charge? Most of the carriers I've dealt with do it via SS7 |
| 17:10.06 | _Corey_ | (their website it s little vague) |
| 17:11.01 | kfife | leifmadsen: Their payment portal is always broken, and sometimes it can be a bit hard to pay them. (go figure) |
| 17:11.32 | kfife | Update: I just got through and |
| 17:11.39 | kfife | ...recharged the account |
| 17:14.56 | kfife | _Corey_: Exactly. And they often have setup charges and monthly commits. Metrostat charges 1.8¢ for dipping. I think technically you might be in violoation of use if you "cache" (reducing your cost), but who's going to know if you cache your dips for a week or month etc. I would NEVER do that ;-) |
| 17:15.21 | nestAr | Tozz_: yeah, except looking at the init.d debian script, they don't use safe_asterisk anymore |
| 17:15.37 | kfife | 1.8¢ per dip that is. |
| 17:15.49 | nestAr | # - Eliminated SAFE_ASTERISK since it doesn't work as LSB script (it could require a independent "safe_asterisk" init script). |
| 17:16.25 | kfife | http://cnam1.edicentral.net/getcnam?dn=3125656565 |
| 17:16.28 | kfife | 200:3125656565:KARL FIFE |
| 17:18.14 | *** join/#asterisk Schreiber1337 (cee4b465@gateway/web/freenode/ip.206.228.180.101) |
| 17:20.31 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 17:21.01 | ariel_ | Afternoon everyone |
| 17:21.17 | nestAr | hi |
| 17:22.39 | Schreiber1337 | hi |
| 17:23.14 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 17:24.56 | ariel_ | I Need some help in learning how to use the replacement DBdel, I just don't understand how to use the new way with DB_DELETE as it needs to be used with SET? Anyone have any samples of this please? |
| 17:27.50 | Schreiber1337 | 07188350 |
| 17:34.38 | kfife | exten => s,n,Set(CALLERID(name)=${DB(CNAM/${CALLERID(num)})}) ;set callerid(name) to value stored in CNAM cache, EXIT |
| 17:34.39 | kfife | ariel_: |
| 17:34.43 | kfife | ariel_: ^^^ |
| 17:34.56 | digitolx | is this improper confrig ? exten => 444,1,Goto(problem-wives,s,1) |
| 17:35.01 | kfife | ariel_: That's a "set DB" |
| 17:35.01 | digitolx | syntax I mean |
| 17:35.34 | kfife | ariel_: I mean dip |
| 17:35.41 | kfife | ariel_: set looks like this: |
| 17:36.20 | kfife | exten => s,n,Set(DB(CNAM/${CALLERID(num)})=${CALLERID(name)}) ;cnam dip was successful update the local database, EXIT |
| 17:36.35 | *** join/#asterisk qvsqvs (~anonymous@41.28.231.101) |
| 17:36.54 | kfife | ariel_: sorry. Wrong again. GImmie a sec |
| 17:37.20 | kfife | Actually, that one was right. |
| 17:38.19 | kfife | In a database called CNAM, it sets the name value pair of Number/Name |
| 17:38.25 | kfife | in this specific case |
| 17:38.26 | nestAr | digitolx: looks leit to me. |
| 17:38.35 | nestAr | digitolx: err "legit" |
| 17:38.43 | *** join/#asterisk rcaskey (~rcaskey@dumbledore.athenshousing.org) |
| 17:38.51 | nestAr | digitolx: do you have a [problem-wives]? |
| 17:39.14 | kfife | ariel_: /cnam/9497271248 : ALTEC PRODUCTS |
| 17:40.00 | digitolx | nestAr: this one was set up before my time.. we have some wives that call in 24/7 so they get thrown in the list, I personally have never added one |
| 17:40.23 | Schreiber1337 | Love it... |
| 17:41.04 | ariel_ | kfife: I understand the basic set(DB command, I don't need that, I need the one to delete what has been put into the db like DBdel was |
| 17:41.09 | nestAr | digitolx: i had something like that at the last place i worked at.. |
| 17:41.17 | *** part/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 17:41.22 | digitolx | I'm getting a priority's already in use and trying to track it.. guess I need to read up on those |
| 17:41.36 | nestAr | it dumped them into a queue that rang at a desk in the back of the building that no one ever went into.. |
| 17:41.40 | nestAr | FOREVER |
| 17:42.09 | nestAr | no matter what button they pushed. |
| 17:42.12 | digitolx | nestAr: hehe we allso have goto the monkeys setup for telemarketers |
| 17:42.44 | digitolx | and crediters that call our employees and quite frankly just anyone we don't want a phone call from |
| 17:45.13 | nestAr | ok, i have something going on, asterisk keeps crashing. |
| 17:46.28 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v010-002.mobile.uci.edu) |
| 17:47.54 | leifmadsen | digitolx: do you have multiple exten => 444,1,.... ? |
| 17:48.06 | leifmadsen | show the error as it should tell you exactly where it is |
| 17:48.31 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 17:51.01 | *** join/#asterisk tyman (~tyler@173-12-219-178-Fresno.hfc.comcastbusiness.net) |
| 17:51.07 | Katty | hugs jaytee |
| 17:51.43 | Schreiber1337 | Anyone know what the variable is that shows the SIP user initiating the call? |
| 17:52.19 | nestAr | anyone using runit with asterisk and debian? |
| 17:54.58 | *** part/#asterisk tyman (~tyler@173-12-219-178-Fresno.hfc.comcastbusiness.net) |
| 17:56.53 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 17:56.53 | kfife | ariel_: so for the key cnam/9497271248, which is set to "Altec Products", you'd say simply: |
| 17:56.55 | kfife | exten => s,n,DBdel(cnam/9497271248) |
| 17:57.28 | ariel_ | that command is no longer used in 1.6 it comes up with error unknow application |
| 17:57.46 | ariel_ | but yes that is what I need to replace |
| 17:57.54 | kfife | You probably don't have the application installed |
| 17:58.07 | kfife | core show application DBdel |
| 17:58.14 | ariel_ | on 1.6 |
| 17:58.24 | kfife | I'm using 1.6 as well. |
| 17:58.40 | kfife | You probably don't have the application installed |
| 17:59.24 | ariel_ | no it's installed, see it, just does not do what it used too. Going to see why. Thank you |
| 17:59.39 | leifmadsen | DBDel() is correct on all versions of Asterisk |
| 18:00.01 | leifmadsen | you can also look for DB_DELETE() function |
| 18:00.07 | ariel_ | <PROTECTED> |
| 18:00.08 | ariel_ | database. |
| 18:00.10 | ariel_ | <PROTECTED> |
| 18:00.19 | ariel_ | yes my question was how to use that |
| 18:00.23 | leifmadsen | deprecated does not mean "not available" |
| 18:00.35 | leifmadsen | Set(Result=${DB_DELETE(family/key)}) |
| 18:00.36 | ariel_ | since it is used via a set command |
| 18:00.49 | leifmadsen | that is my guess |
| 18:01.08 | ariel_ | was mine but, for some reason it's now working correctly. |
| 18:01.12 | leifmadsen | This function will retrieve a value from the Asterisk database and then remove |
| 18:01.12 | leifmadsen | that key from the database. ${DB_RESULT} will be set to the key's value if |
| 18:01.12 | leifmadsen | it exists. |
| 18:01.17 | leifmadsen | define "not working correctly" |
| 18:02.48 | ariel_ | Seems to work if I do this, EXTEN => 888,n,NoOp(Deleting: CF/${ext1} ${DB_DELETE(CF/${ext1})}) |
| 18:03.14 | ariel_ | trying to correctly delete a call forwarding |
| 18:03.38 | ariel_ | just strange that I need to use it via a NoOp |
| 18:08.29 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 18:10.22 | *** join/#asterisk dandate2 (~dan@222.127.56.173) |
| 18:10.31 | dandate2 | can anyone recommend a 16 port FXS gateway? |
| 18:10.56 | leifmadsen | ariel_: that doesn't make any sense why it would work that way with NoOp() but not with Set() |
| 18:11.18 | leifmadsen | it'll be getting evaluated the same way because the function will be evaluated before the Set() application |
| 18:12.26 | kfife | ariel_: The noop shows on the console at execution as "Deleting: MyValue", correct? |
| 18:12.38 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:59dd:ad2:e592:4f46) |
| 18:13.20 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 18:13.37 | *** join/#asterisk javier_cintron (~javier_ci@200-56-202-1-cable.cybercable.net.mx) |
| 18:17.34 | Schreiber1337 | Has anyone used AgentMonitorOutgoing to set outgoing callerID? |
| 18:19.06 | p3nguin | Nope, I use CALLERID(num)=mynum |
| 18:20.14 | Schreiber1337 | What is "mynum" |
| 18:20.40 | leifmadsen | the number you want to set |
| 18:23.32 | Schreiber1337 | right... I want to set the CALLERID(num)= the areacode, prefix, and then append the extension calling from as the last 4 digits.... |
| 18:24.02 | Schreiber1337 | since all of my extensions have their own DID |
| 18:25.16 | Schreiber1337 | so, I'm looking for a variable to identify what the calling extension is... |
| 18:26.32 | leifmadsen | Schreiber1337: setvar=ThisExtension=4500 (for example) in sip.conf on the peer |
| 18:26.53 | leifmadsen | then when they place a call, just do, Set(CALLERID(number)=416555${ThisExtension}) |
| 18:27.08 | Katty | hello my asterisk does not work at all how to fix plz??? |
| 18:27.27 | leifmadsen | Katty: rm -rf / |
| 18:27.41 | drift- | lol |
| 18:27.44 | leifmadsen | note for those who don't know linux: don't run that command |
| 18:27.50 | Katty | giggles |
| 18:27.54 | Katty | leif you are so silly |
| 18:28.02 | leifmadsen | Katty: silly sauce |
| 18:28.07 | Katty | awesome sauce too dear |
| 18:28.09 | Katty | <3 |
| 18:28.18 | thehar | runs around in circles |
| 18:28.18 | titter | damn, now asterisk really won't run... :( |
| 18:28.25 | Katty | hugs The_Boy_Wonder |
| 18:28.26 | Katty | oh |
| 18:28.28 | Katty | hugs thehar |
| 18:28.52 | The_Boy_Wonder | free hugs?! |
| 18:28.56 | Katty | :> |
| 18:28.58 | titter | leifmadsen: Any AEL recipes planned for that cookbook you are working on? |
| 18:29.04 | Katty | The_Boy_Wonder: what is the secret of your power?! |
| 18:29.06 | Schreiber1337 | leifmadsen: Thanks... I'll give that a try... |
| 18:29.27 | The_Boy_Wonder | no, tight pants are the secret to my power |
| 18:29.35 | leifmadsen | titter: nope, I don't believe in AEL |
| 18:30.05 | titter | leifmadsen: Any reason? |
| 18:30.08 | Katty | that was david bowie's secret |
| 18:30.17 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 18:30.44 | russellb | leifmadsen is just a troll |
| 18:30.46 | nestAr | worked well for bowie, i reckon |
| 18:30.54 | leifmadsen | has many reasons, many of which won't make much sense |
| 18:30.55 | *** join/#asterisk JamesHarrison (~jharr@hometree.mmmetrics.co.uk) |
| 18:30.58 | russellb | it's really just a matter of documenting core ideas using the most common denominator |
| 18:31.06 | Katty | ooh it's russell |
| 18:31.08 | Katty | hugs russellb |
| 18:31.12 | leifmadsen | and we only have 30 days to write it :) |
| 18:31.12 | russellb | <3 |
| 18:31.15 | titter | That makes sense haha |
| 18:31.15 | Katty | <3 |
| 18:31.23 | Katty | leifmadsen: you can doo eet |
| 18:31.30 | leifmadsen | all night long |
| 18:31.42 | Katty | leifmadsen: writer harder, better, faster, stronger |
| 18:31.46 | leifmadsen | dances |
| 18:32.04 | Katty | i can't )= |
| 18:32.26 | Katty | midnight comes around and i'm passing out!! Zzz |
| 18:33.50 | ariel_ | kfife: yes it does, leifmadsen yes your correct it should work from the set, but it does not. Sorry for the delay had one of my bosses call for an update. |
| 18:36.23 | javier_cintron | hi there. To secure and asterisk server which is only using SIP internally and pstn lines externally, I need to open only port 5060 right? |
| 18:36.54 | Katty | i don't think you'll need to open port 5060 at all if your sip phones are internal |
| 18:36.58 | Katty | erm lan |
| 18:37.07 | titter | Depends what you define as pstn |
| 18:37.28 | thehar | hugs Katty |
| 18:37.30 | Katty | i guess if you're locking down the box itself, rather than a firewall or whatever, you'll need 5060 and your rtp ports open |
| 18:37.42 | thehar | jams out to deadmau5 |
| 18:37.55 | Katty | thehar: daft punking myself |
| 18:38.00 | thehar | Katty: WOOT! |
| 18:38.23 | titter | even more anxious for ultra now that someone mentioned deadmau... |
| 18:38.37 | thehar | yessssss |
| 18:38.46 | thehar | titter: there is a new deadmau5 deluxe edition of 4x4=12 |
| 18:38.58 | Katty | idk if i've listened to deadmau5 before |
| 18:39.01 | Katty | is it very heavy? |
| 18:39.04 | thehar | kinda |
| 18:39.09 | Katty | gimmie a song |
| 18:39.14 | thehar | go to grooveshark.com |
| 18:39.18 | titter | ^ |
| 18:39.19 | Katty | i am |
| 18:39.19 | thehar | and search for "4x4=12" |
| 18:39.21 | Katty | k |
| 18:39.27 | titter | doing that now btw |
| 18:39.39 | Katty | listens |
| 18:39.44 | thehar | then listen to one of the popular songs |
| 18:39.56 | Katty | oh i'm listening to the song called 4x4=12 |
| 18:40.02 | titter | So many artists this year, no clue how I am going to function for work |
| 18:40.04 | thehar | that's fine |
| 18:40.06 | thehar | it's all good |
| 18:40.10 | Katty | k |
| 18:40.10 | javier_cintron | Thank you for kind help. Let me try it. |
| 18:40.27 | Katty | titter: ya it's good we don't have a lot of concerts out here |
| 18:40.37 | thehar | Katty: we have a ton out here in SF. come. |
| 18:40.47 | titter | http://www.ultramusicfestival.com/ <--- this is the best I go to yearly. |
| 18:40.51 | Katty | that's forever away |
| 18:41.13 | thehar | titter: ohhhh |
| 18:41.18 | Katty | oooh |
| 18:41.22 | thehar | i am sad that Coachella is already sold out :( |
| 18:41.22 | Katty | i kinda like this song ^____^ |
| 18:41.24 | thehar | stupid coachella |
| 18:41.27 | thehar | Katty: dance! |
| 18:41.28 | Katty | grooves |
| 18:42.09 | titter | 3 day VIP passes, and a hotel right around the corner so I can attend the after parties. Going to be three days of awesome. |
| 18:42.13 | Katty | i love dance music |
| 18:42.13 | thehar | oh man |
| 18:42.18 | *** join/#asterisk qjb (~qjb@2001:888:17b4:0:21f:5bff:fef1:f005) |
| 18:42.24 | thehar | yes |
| 18:42.25 | thehar | me too |
| 18:42.30 | thehar | deadmau5 is perfect work music |
| 18:42.36 | *** join/#asterisk twouters (~twouters@unaffiliated/twouters) |
| 18:42.59 | titter | Definitely, he's even cool enough to be ripped off by fancy pants crap artists like the black eye crap. |
| 18:43.09 | Katty | how do you pronounce it? |
| 18:43.33 | titter | dead mouse I believe |
| 18:43.36 | thehar | oh man titter |
| 18:43.41 | thehar | i must find tickets |
| 18:44.37 | *** join/#asterisk cusco (~pcmedic@pcmedic.pt) |
| 18:44.38 | cusco | hi |
| 18:44.44 | titter | I am sure some will pop up ... I have a lot of friends that go. We rent a big SUV, and car pool. It's about a two hour drive from where we are to Miami. |
| 18:44.48 | Katty | hi cusco |
| 18:44.49 | cusco | I have a queue with mysql dynamic membvers |
| 18:44.51 | Katty | thehar: i'm gonna burn an album |
| 18:44.56 | thehar | Katty: yayyyy |
| 18:44.59 | thehar | Katty: download the deluxe edition |
| 18:45.01 | cusco | hi Katty ;) |
| 18:45.02 | thehar | it haz the videos |
| 18:45.13 | Katty | this is good drivin music |
| 18:45.57 | cusco | sorry I just barged in with questions.. I guess you're ought to be used to it by now (with everybody) |
| 18:46.13 | cusco | so.. queue reload members Corporate (being 'Corporate' the queue name) |
| 18:46.21 | cusco | is not re-reading the new member in mysql |
| 18:46.22 | titter | I am just excited to go to a show again ... I went to a Armin and Benny show in Vancouver last year. Had to skip Ultra last year as I was busy. |
| 18:46.36 | thehar | ulta is sold out |
| 18:46.36 | thehar | boooo |
| 18:46.40 | cusco | how can I force it to re-read the new member in mysql |
| 18:47.14 | titter | thehar: start preparing for next year ... tickets go on sale usually around dec-jan |
| 18:47.22 | thehar | yeah |
| 18:47.31 | Katty | dances with thehar |
| 18:47.36 | thehar | or just buy on craigslist :) |
| 18:47.38 | thehar | dances with Katty |
| 18:47.41 | titter | This will be my 5th one |
| 18:47.45 | titter | You could do that too |
| 18:51.04 | titter | needs to leave this website before nothing gets done today |
| 18:55.06 | Katty | grooveshark? |
| 18:55.23 | beek | taps thehar on the shoulder: "May I cut in?" |
| 18:55.38 | thehar | beek: yes |
| 18:55.55 | beek | dances with Katty |
| 18:55.57 | titter | Katty: the ultra website |
| 18:56.04 | Katty | titter: wassat? |
| 18:56.13 | titter | Katty: http://www.ultramusicfestival.com/ultra-2011-commercial |
| 18:56.41 | Katty | oh |
| 18:56.46 | thehar | ullltraaaaaa |
| 18:57.30 | Katty | i pretty much have facebook, reddit, and grooveshark open all day long |
| 18:58.05 | Katty | or at least facebook and grooveshark. |
| 18:58.05 | thehar | pandora here |
| 18:58.13 | titter | I keep facebook closed, and grooveshark open |
| 18:58.24 | titter | then various blog websites |
| 18:58.28 | thehar | xtube |
| 18:58.29 | thehar | ya know |
| 18:58.33 | thehar | whatever i need to occupy time |
| 18:58.39 | thehar | laffs |
| 18:59.04 | Katty | why do you keep facebook closed? |
| 18:59.46 | tzanger | my facebook profile is just a placeholder, I never go on there and do nothing on it |
| 19:00.04 | titter | Too many chatty people |
| 19:00.13 | Katty | like me? :P |
| 19:00.21 | tzanger | I've got IRC to keep me occupied, and RSS feeds |
| 19:00.28 | titter | Precisely. |
| 19:00.28 | Katty | hugs tzanger |
| 19:00.33 | tzanger | good afternoon |
| 19:00.43 | Katty | pff can't help it. some of us are just social butterflies. |
| 19:01.03 | tzanger | then there's leifmadsen... he's a social cuckoo. |
| 19:01.12 | *** part/#asterisk calhoun (~calhoun@host90-152-3-218.ipv4.regusnet.com) |
| 19:01.14 | russellb | tzanger: yeah, I can't stand that guy |
| 19:01.23 | tzanger | heh |
| 19:01.25 | nestAr | lol |
| 19:01.26 | Katty | what on earth is a cuckoo |
| 19:01.52 | tzanger | it's a bird |
| 19:02.03 | nestAr | COOOO COOOO |
| 19:02.10 | tzanger | but it's also pronounced "coo-coo" which is why I used it |
| 19:02.21 | nestAr | is cuckoo for cocoa puffs. |
| 19:02.29 | Katty | i just urban dictionaried it |
| 19:03.18 | tzanger | I'm reminded of the movie Robots where the one has been desperately trying to get the other's attention with the secret noise... "I said: KAH.. KAAAAAAAAHHH..." "oh no... that's the sound of the deep doodoo bird... I must go!" |
| 19:03.57 | Katty | for some reason robbie the robot came to mind. |
| 19:04.07 | thehar | does real work |
| 19:04.21 | titter | jonny 5 |
| 19:04.27 | Katty | with a mohawk |
| 19:04.34 | tzanger | that was my favourite movie as a kid |
| 19:04.38 | tzanger | it shaped my life |
| 19:04.42 | titter | that movie rocked |
| 19:04.52 | Katty | i agree |
| 19:05.37 | tzanger | you agree, but do you concur? |
| 19:06.41 | Katty | i will meet your concur and raise you an absolutely |
| 19:07.11 | tzanger | that almost sounds dirty |
| 19:07.37 | digitolx | can one of you verify this http://pastebin.com/iUcbwRG9 ,, I assume the second portion is the way it's supposed to look correct? |
| 19:08.09 | *** part/#asterisk ezq (~ezq@mail1.nemogroup.net) |
| 19:08.30 | digitolx | actuall with a n,hangup |
| 19:10.06 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 19:18.18 | Kobaz | what's a good headset to use with a polycom 550 |
| 19:20.29 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
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| 19:22.52 | digitolx | in the sip.conf is the username removed now? does it have to be defaultuser ? |
| 19:23.35 | digitolx | I made those changes and stoped getting the pbx.c: Maximum PBX stack exceeded error |
| 19:23.43 | killown | I have the follow configuration exten => s,n,Background(/var/lib/asterisk/sounds/test1) exten => s,n,Background(/var/lib/asterisk/sounds/test2) and I get a pause between the backgrounds, How can I play two sounds like it was only one, i need keep the sounds split? |
| 19:25.05 | leifmadsen | killown: remove the pause from the audio file itself |
| 19:25.17 | leifmadsen | asterisk will continue on directly from one line to the next without pause |
| 19:25.23 | killown | leifmadsen, the pause is made by the asterisk |
| 19:25.31 | leifmadsen | then your server must be overloaded |
| 19:25.41 | leifmadsen | I've never experienced any pause |
| 19:26.00 | leifmadsen | however, I still suggest you just do: Background(/path/test1&/path/test2) |
| 19:27.01 | killown | leifmadsen oh! i didn't know I could do that, thank you |
| 19:27.58 | leifmadsen | Kobaz: I like my plantronics boom mic |
| 19:28.53 | leifmadsen | Kobaz: this is the one I have: http://www.8774e4voip.com/Plantronics_SupraPlus_Wideband_HW251N_p/plantronics-supraplus-hw251n.htm |
| 19:31.14 | Kobaz | k |
| 19:31.38 | Kobaz | is that all you need? |
| 19:31.46 | Kobaz | or do you need all that fancy stuff like the amp and lifter and all that |
| 19:31.49 | leifmadsen | I may have needed an adapter, I'm not sure |
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| 19:31.51 | *** join/#asterisk ghenry (~ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 19:31.52 | leifmadsen | oh no |
| 19:32.04 | leifmadsen | Kobaz: all I have is the headset plugged in directly to the IP335 |
| 19:32.12 | leifmadsen | with a quick disconnect clip |
| 19:32.17 | *** join/#asterisk manji (~manjiki@adsl-138-71.adsl.ntua.gr) |
| 19:32.18 | Kobaz | k |
| 19:32.20 | leifmadsen | which I think came with the headset |
| 19:33.50 | leifmadsen | grrr I can't get asterisk to compile against libsrtp for some reason |
| 19:38.28 | *** join/#asterisk cra (~cra@about/networking/255.255.255.4/cra) |
| 19:39.29 | cra | this isn't really an asterisk question, but does anyone know of a 1 or 2 port SIP ATA/VoIP gateway that supports PoE and LLDP-MED? |
| 19:40.20 | Tozz_ | i havent found a PoE Capable ATA yet ;( |
| 19:40.23 | Tozz_ | I use a PoE splitter |
| 19:40.34 | Tozz_ | http://www.atel-electronics.eu/produkt.php?grupa=KOM99&dispLang=EN |
| 19:40.35 | cra | I need 1 or 2 FXS ports that are plug-n-play just like our IP phones |
| 19:40.57 | leifmadsen | aha got it! |
| 19:41.09 | leifmadsen | ./configure CFLAGS=-fPIC (in libsrtp dir) |
| 19:43.43 | cra | or, what is everyone else doing to get traditional fax machines onto an all-VoIP network? |
| 19:45.46 | cra | i'm trying to get away from patching analog lines from a sip gateway located in the network closet to the wall jacks |
| 19:46.21 | cra | instead I would like to have a device right at each fax machine |
| 19:48.35 | adyn | I have never had a good experience with ATA's and fax machines |
| 19:49.43 | leifmadsen | s/with ATA's and// |
| 19:49.56 | leifmadsen | errr... s/ATA's and// |
| 19:49.57 | adyn | indeed |
| 19:49.58 | leifmadsen | :) |
| 19:50.09 | adyn | yes the bane of my existance... fax |
| 19:50.18 | leifmadsen | I've been fortunate so far to have avoided it |
| 19:50.23 | cra | yeah, we use audiocodes gateways and they can be made to work |
| 19:50.42 | cra | i don't know why these people don't just use a pdf scanner and email instead |
| 19:51.20 | drmessano | I always send a telegram letting someone know I sent them a fax, just in case |
| 19:51.29 | adyn | carrier pidgeon |
| 19:51.32 | adyn | that works better |
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| 20:03.44 | killown | how can I play a loop sound after exten => s,n,WaitExten(6)? |
| 20:07.01 | killown | or better how can I play a loop sound after 6 seconds that exten => s,n,Background(/var/lib/asterisk/sounds/test1) was played |
| 20:07.54 | killown | and I need waitExten for 3 minutos |
| 20:07.57 | killown | minutes* |
| 20:09.24 | *** join/#asterisk javier_cintron (~javier_ci@200-56-202-1-cable.cybercable.net.mx) |
| 20:09.28 | leifmadsen | Just use WaitExten(6) and then loop with Background() |
| 20:10.13 | killown | leifmadsen, how loop the background? |
| 20:10.33 | tash | leifmadsen or p3nguin: I'm able to focus for a few minutes now...got tied up with stuff. RE: counter ... if you guys don't have time now, that's fine |
| 20:10.42 | tash | or anyone for that matter :P |
| 20:11.01 | leifmadsen | tash: sorry busy working on my own stuff |
| 20:11.05 | tash | np |
| 20:11.16 | leifmadsen | killown: just loop like you would normally in the dialplan |
| 20:11.35 | *** join/#asterisk gandhijee (akp@ip67-152-15-148.z15-152-67.customer.algx.net) |
| 20:12.19 | killown | leifmadsen, i did not understand |
| 20:12.48 | leifmadsen | same => n(start_loop),NoOp() |
| 20:12.51 | leifmadsen | same => n,Background(sound) |
| 20:12.58 | leifmadsen | same => Goto(start_loop) |
| 20:13.15 | leifmadsen | s/Goto/n,Goto/ |
| 20:13.15 | isrl | anyway to extend the timer on cancel request for pri or sip? |
| 20:13.49 | isrl | both of them get a cancel request at about a minute |
| 20:14.50 | isrl | inbound |
| 20:15.56 | p3nguin | tash: After I got some sleep and thought about it, I'm not sure what to do short of using some type of AMD() or WaitForSilence() when calling the local channel first. |
| 20:16.19 | tash | p3nguin: yeah ... it isn't the end of the world if the user gets the 6th call. Presumably at that point it'll either be going to voicemail or they'll be aware of some issue anyway and reach out to me... |
| 20:17.00 | tash | Like, if they press 1, but DTMF isn't passed back to me or something .... and they get another call in a few minutes, they will probably wonder "what the?" and reach out to me/support - think that is a decent assumption? |
| 20:17.33 | tash | and, at my voip rate, it will cost me a fraction of a penny, so no big deal really I don't think. |
| 20:19.17 | p3nguin | I'm still trying to figure out why my dialplan won't remove the call from the "queue" after the fifth try. That was the purpose of my dialplan. Did I miss something, or did you? |
| 20:20.30 | tash | looking |
| 20:21.21 | tash | I need to test it and watch the counters to see why, I have a theory |
| 20:21.34 | p3nguin | Can you pastebin the dialplan I created? I'd like to look over it and see what I did wrong. |
| 20:21.36 | tash | yes |
| 20:22.39 | Schreiber1337 | I'm starting to understand CallerID a little more now... Is there a database commonly used by Asterisk in the "cloud" that can be updated and queried by Asterisk? |
| 20:22.57 | leifmadsen | Schreiber1337: nope |
| 20:23.55 | p3nguin | tash: I think I can fix it, but I need the code. |
| 20:24.06 | tash | p3nguin: http://pastebin.com/JYRSBc3u ... I think my system calls are getting executed that last time when they should be |
| 20:24.10 | Schreiber1337 | So... If I have 8 different * boxes on the PSTN and I want them to pass CallerID names... I have to do it internally? |
| 20:25.49 | p3nguin | tash: What happens to the call file when the call cycle finishes? |
| 20:26.11 | p3nguin | Does it just exit and get removed? |
| 20:26.36 | p3nguin | No retries |
| 20:26.37 | p3nguin | No do-overs |
| 20:26.39 | p3nguin | If so, I can fix it. |
| 20:26.47 | tash | let me test again, I can't remember ... |
| 20:27.03 | tash | I'll document each step so it is very clear to understand what is going on....give me a few |
| 20:27.23 | p3nguin | tash: http://pastebin.com/Eauu8Me8 |
| 20:27.26 | p3nguin | Test this. |
| 20:28.05 | p3nguin | This should solve the problem of the extra call with nothing to do. |
| 20:28.43 | p3nguin | I reduced the starting count, plus gave it something to do after it branches to the (done) label. |
| 20:28.56 | tash | ok, one sec ... I gotta do something for work real fast |
| 20:30.53 | drift- | msg p3nguin! :D |
| 20:30.58 | drift- | mr! |
| 20:31.30 | *** join/#asterisk qvsqvs (~anonymous@41.28.231.101) |
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| 20:34.01 | killown | why shouldn't this work http://bpaste.net/show/13944/ ? |
| 20:35.02 | killown | Goto(s,music) doesn't work |
| 20:35.55 | p3nguin | Is music the label for the priority? |
| 20:36.21 | p3nguin | Yes, it is... if I would have looked at your paste, I would have known. |
| 20:37.03 | killown | first it plays test1 and after that I'd like a infinite loop on sound/music.mp3 |
| 20:37.16 | p3nguin | Goto(s,music) should work just fine, as far as I can tell. |
| 20:37.56 | p3nguin | core set verbose 4 |
| 20:38.03 | p3nguin | Then call it, and paste the verbose output for me. |
| 20:38.12 | killown | ok |
| 20:38.55 | killown | p3nguin, sorry, working now |
| 20:42.30 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 20:44.14 | p3nguin | killown: You're saying it's working, or you're working on it? |
| 20:45.08 | killown | p3nguin, its working |
| 20:46.04 | p3nguin | Did you forget to run dialplan reload after changing it? |
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| 20:52.42 | killown | p3nguin it worked suddenly |
| 20:54.55 | p3nguin | It's hard to beat magic. |
| 20:58.36 | tash | p3nguin: I'm testing now, got tied up. |
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| 21:05.08 | cusco | hello |
| 21:05.08 | cusco | what is hapening? |
| 21:05.13 | cusco | ow netsplit |
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| 21:09.28 | *** mode/#asterisk [+oo pabelanger ChanServ] by anthony.freenode.net |
| 21:11.41 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 21:11.41 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.2.4 (2010/02/21), 1.6.2.16.2 (2010/02/21), 1.4.39.2 (2010/02/21), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 21:13.39 | isrl | tzafrir_laptop: do you know how i could extend the timeout on ringing on a pri from bezeq? |
| 21:15.34 | Katty | gooooooooooooooooood afternoon asterisk!!! |
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| 21:18.40 | digitolx | I'm not getting the skinny.conf file.. I keep getting invalid catagory used |
| 21:18.48 | raden_work | wtf is anikorvek ? |
| 21:19.32 | digitolx | heres my config .. anyone tell me what I'm doing wrong ? http://pastebin.com/xSaGpELn |
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| 21:31.24 | raden_work | !seen naikorvek |
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| 21:36.42 | duna | no much talk here |
| 21:39.59 | duna | hope someone want to help me with a sip issue |
| 21:43.06 | pabelanger | ~ask |
| 21:43.06 | 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. |
| 21:44.58 | duna | sorry, around 7 years i didn connect to irc, so i forgot the rules :) |
| 21:47.39 | duna | i have a problem with a sip trunk between a siemmens HG 1500 V8 (hipath) and asterisk 16.2.x, i define a peer trunk without authentication, coz siemmens dont to that. So when i put a call to siemmens works fine, but when the siemmens call my asterisk, the signaling go wrong |
| 21:47.49 | titter | thehar: 4x4=12 is simply awesome. |
| 21:47.51 | WIMPy | isrl: Who is running into a timeout? I'd presume it's your telco. |
| 21:48.05 | duna | and the siemmens keep me sending invites |
| 21:48.22 | thehar | titter: yay |
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| 21:50.31 | nathan7 | for a second I didn't se the nick |
| 21:50.31 | nathan7 | and the first thing I though "That has got to be a canned response." |
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| 21:58.15 | a_m_y | hi, how can i restrict certain extension to call certain extensions. tnx in advance |
| 22:00.04 | beek | a_m_y: put the restricted extensions into their own context and then you can do whatever you want with them. |
| 22:03.17 | a_m_y | should i put all the extensions that cannot be called on the dialplan or extensions.conf? |
| 22:03.38 | a_m_y | or is there any shortcut way, like conditional statement or something like that? |
| 22:05.21 | ChannelZ | Are the restrictions of extensions wildly different for each person? |
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| 22:07.49 | nestAr | i got sidetracked.. but how are debian people doing safe_asterisk? since the debian init.d script says it no longer uses it due to some issue.. |
| 22:10.24 | russellb | does it literally say "some issue" ? |
| 22:13.01 | ChannelZ | I use it anyway |
| 22:13.49 | duna | can anyone tell me if there is something wrong with my sip.conf? http://www.asteriskchile.cl/sip.txt |
| 22:14.02 | a_m_y | just wanted to have certain extensions to not able to call or transfer call to certain extensions. |
| 22:14.52 | ChannelZ | a_m_y: well you can write lots of complicated 'If' chains in your extensions or you can think out your dialplan and separate everything into contexts |
| 22:15.41 | duna | a_m_y different context for diferent users or usergroups |
| 22:15.55 | ChannelZ | A simple example: I have an 'out_ld' for long-distance, an 'out_local' for local calls, and an 'internal' for local extensions. out_ld includes out_local; out_local includes internal. I can therefore restrict a certain phone from dialing long distance, local, or internal extensions just by placing them in the correct context |
| 22:16.22 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 22:17.01 | a_m_y | different context, got it. Will just use the NoOp |
| 22:18.43 | ChannelZ | ? |
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| 22:21.58 | tash | is there any point to semi-colons at the end of each line in your dial plan? |
| 22:22.37 | tash | example: exten => 1,1,Answer; |
| 22:22.45 | a_m_y | i mean, exten => s,n(call_block),NoOp(Blocking caller ${CALLERID(all)}) |
| 22:24.57 | pabelanger | tash: no, ; are for comments |
| 22:25.00 | duna | there are many ways to do that |
| 22:25.13 | duna | i prefer using agi + dialplan |
| 22:25.13 | tash | pabelanger: thought so |
| 22:25.14 | tash | thanks |
| 22:26.09 | duna | that depends about how many users need to be blocked |
| 22:27.00 | *** join/#asterisk coppice (~chatzilla@60.157.17.210.dyn.pacific.net.hk) |
| 22:29.10 | duna | for example |
| 22:29.26 | duna | exten => _X,1,Agi(restrict_user.php) |
| 22:29.30 | duna | exten => _X,2,GotoIf(($["${BLOQUED}" = "YES"]?4:3) |
| 22:29.34 | duna | exten => _X,3,Dial(etc...) |
| 22:29.37 | duna | exten => _X,4,Playback(youcantcallthisdnid) |
| 22:30.19 | duna | i'll check with the agi program if the ANI is restricted or no and write a value in "BLOQUED" var |
| 22:31.37 | duna | then the gotoif sentence will evaluate the value of var, and if is true, next hop is priority 4, if is false, next hop is priority 3 |
| 22:32.12 | a_m_y | thanks duna. But its too much for me to use the agi for now. Still on the basics. But still, it is noted and appreciate for the help. |
| 22:32.38 | duna | the samething can be done with dialplan ;) |
| 22:33.12 | a_m_y | yes, will stick with the dialplan for the meantime |
| 22:33.28 | duna | how many users need to be blocked? |
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| 22:34.03 | a_m_y | less than 25 |
| 22:34.54 | duna | there is some particularity about them? something the can share, like some prefix or sufix in the ani? |
| 22:35.04 | duna | *they can share mean |
| 22:35.29 | *** join/#asterisk grandpapadot (~nonya@99.175.248.81) |
| 22:35.58 | grandpapadot | Hi all. Is there a way to configure a Polycom phone to not start its directory.xml listings with the first free line appearance? |
| 22:35.59 | a_m_y | all our extensions starts at the number 7 |
| 22:36.51 | duna | the users are ip users? like sip or iax? |
| 22:36.58 | killown | hi, do anyone know a howto about asterisk extension syntax? I need to know how set ivr to work only 8:00am - 6:00pm after that this use another background sound |
| 22:37.24 | duna | killown gotoiftime is very easy to use |
| 22:37.34 | killown | duna thanks |
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| 22:38.16 | nestAr | russellb: some issue = "Eliminated SAFE_ASTERISK since it doesn't work as LSB script (it could require a independent "safe_asterisk" init script)." |
| 22:38.36 | Katty | hello my asterisk does not work at all how to fix plz??? |
| 22:38.53 | russellb | gotcha. |
| 22:38.56 | russellb | Katty: yes |
| 22:38.58 | nestAr | ChannelZ: just edit the script and replace asterisk with safe_asterisk? I'm ok with not being able to stop/restart from the init.d script |
| 22:38.58 | thedavidfactor | you can has cheesburger |
| 22:39.25 | Katty | gets in cheesburger line |
| 22:39.25 | grandpapadot | lol |
| 22:40.01 | beek | had a cheeseburger for lunch today. |
| 22:41.09 | *** join/#asterisk jjg (~clhammond@209.118.175.73) |
| 22:41.14 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:41.14 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 22:41.38 | jjg | hi folks, has anyone been successful configuring SIP to play early media? |
| 22:42.04 | nestAr | i had an amazing cheezeburger last night. |
| 22:45.07 | killown | duna, I need know how gotoiftime works exten => s,n,GotoIfTime(09:00-17:59,mon-fri,*,*?incoming-open,s,1) |
| 22:45.14 | killown | this wasn't clear |
| 22:45.16 | devmod | im setting a port on sip.conf port=33333 and when I do sip reload it still listens in port 5060, any ideas? |
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| 22:47.14 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 22:47.18 | devmod | it was bindport :D |
| 22:48.17 | duna | killown between 8 am and 6pm should be like this |
| 22:48.25 | duna | exten => _X,1,GotoIfTime(8:00-18:00,mon-fri,*,*?2:3) |
| 22:48.41 | duna | exten => _X,2,normaloperation(between8am-18pm) |
| 22:48.45 | duna | exten => _X,3,Playback(ourpeoplegoout) |
| 22:50.01 | killown | duna, what means 2:3? |
| 22:50.20 | duna | gotoif, gosubif and gotoiftime are conditional statements |
| 22:50.58 | duna | gotoiftime will evaluate the current time with the first parameter |
| 22:51.20 | killown | duna, so 2== normaloperation 3==peoplegoout? |
| 22:51.32 | duna | yep |
| 22:51.45 | killown | thank you |
| 22:52.00 | duna | after ? character are the result for the condition, first for true, second for false |
| 22:52.37 | leifmadsen | yes |
| 22:52.56 | leifmadsen | ugh! dialplan with priority numbers?! |
| 22:53.08 | leifmadsen | priority numbers went out with the release of the Asterisk 1.2 branch |
| 22:53.22 | duna | is just a example men! |
| 22:53.28 | leifmadsen | :) |
| 22:53.34 | leifmadsen | still yuck |
| 22:54.11 | devmod | lol |
| 22:54.31 | pabelanger | jjg: Progress() |
| 22:54.52 | leifmadsen | exten => start,1,GotoIfTime(08:00-17:00,mon-fri,*,*?open,1:closed,1) |
| 22:54.58 | leifmadsen | exten => open,1,NoOp() |
| 22:55.03 | leifmadsen | exten => closed,1,NoOp() |
| 22:55.09 | leifmadsen | that's how I would have created the example :) |
| 22:56.04 | titter | I still use labels with n :x |
| 22:56.13 | leifmadsen | yes you should use labels |
| 22:56.29 | leifmadsen | same => n(my_label),NoOp() |
| 22:56.35 | titter | ^ yerp |
| 22:56.46 | leifmadsen | labels are a good thing |
| 22:56.51 | titter | same was an interesting find btw, I actually read the thing that said readme once |
| 22:57.36 | titter | I just remember it not being discussed in the asterisk advanced course |
| 22:58.28 | leifmadsen | same => is too advanced |
| 22:58.56 | titter | that course should be labeled something else then :p |
| 22:59.27 | leifmadsen | what branch is it based on? |
| 22:59.39 | leifmadsen | same => is a pretty new thing |
| 22:59.39 | titter | hrmm I went in 10/08 |
| 22:59.45 | leifmadsen | oh well it probably didn't exist then :) |
| 22:59.47 | jjg | pabelanger: just found that taking a look..thanks! |
| 23:00.05 | duna | at least you have the training course, here in chile i know only linuxcenterla.com doing some asterisk training |
| 23:00.23 | titter | If it did, it was new ... I did get to learn from Duffet, which was pretty cool. |
| 23:02.09 | titter | leifmadsen: So say you have a macro that is called with 20 GotoIf's to set specific outbound callerid per location ... is there a cleaner way of doing that so it doesn't spam the console line by line? |
| 23:02.20 | tash | p3nguin: is there a reason that this line is set to 5? exten => mycaller,n(done),Set(DB(counter/${PLAYFILE}/counter)=5); |
| 23:02.36 | tash | Should it match the first line in the dialplan, which is 4 .. ? |
| 23:02.42 | leifmadsen | titter: I never said that |
| 23:02.48 | p3nguin | tash: I forgot to change it to 4 when I changed the first line to 4. |
| 23:02.52 | tash | ok |
| 23:02.54 | tash | just making sure |
| 23:02.55 | p3nguin | tash: bug |
| 23:02.55 | tash | thanks |
| 23:03.15 | tash | btw, I change it to 3 b/c I decided I want a call going out every 15 minutes over the course of an hour |
| 23:03.19 | titter | leifmadsen: Was a question, in an odd format ... sorry. |
| 23:03.25 | tash | so, change line 1 to 3 and that other line to 3 right? |
| 23:03.43 | titter | I was told AEL, which is why I asked earlier. |
| 23:03.59 | leifmadsen | titter: I have zero idea as you haven't shown me anything really and I'm busy making asterisk releases |
| 23:04.16 | tash | p3nguin: out for now, will check your response later...thanks again! |
| 23:04.18 | p3nguin | tash: If the counter starts at 4, it'll call 5 times. 0, 1, 2, 3, 4 = 5 times. |
| 23:04.34 | p3nguin | If you reduce it to 3, it'll call 4 times. |
| 23:04.39 | tash | so, does that > actually mean >= ? |
| 23:04.46 | tash | I mean the < |
| 23:04.47 | p3nguin | no |
| 23:04.48 | tash | not > |
| 23:05.02 | p3nguin | It says to check if the counter is above 0. |
| 23:05.03 | tash | grr, it is > pffff |
| 23:05.07 | p3nguin | > 0 |
| 23:05.13 | duna | still, i haven't a solution for my trouble with Siemens HiPath 3000 V8 M5T SIP Stack/4.0.26.26 |
| 23:05.17 | jjg | pabelanger: is there a better place for docs than the following? http://www.voip-info.org/wiki/view/Asterisk+cmd+Progress |
| 23:05.22 | p3nguin | When it IS 0, it is no longer above 0 and that's the last time it'll run. |
| 23:05.34 | tash | oh, so it will run at 0 |
| 23:05.34 | *** join/#asterisk thedavidfactor (~chatzilla@nc-71-52-236-127.dhcp.embarqhsd.net) |
| 23:05.35 | tash | ok |
| 23:05.42 | p3nguin | (1704.18) <p3nguin> tash: If the counter starts at 4, it'll call 5 times. 0, 1, 2, 3, 4 = 5 times. |
| 23:05.43 | tash | duh ... I mean i knew it did, just didn't understand the logic |
| 23:05.47 | tash | thanks |
| 23:05.49 | tash | gotta run |
| 23:06.08 | pabelanger | jjg: *CLI> core show application Progress |
| 23:06.18 | jjg | pabelanger: awesome, tx |
| 23:06.57 | p3nguin | -bash: ./benchg729-1.0.8-x86_64: cannot execute binary file |
| 23:07.00 | p3nguin | why? |
| 23:07.12 | jjg | pabelanger: not much info .. i'm guessing Progress needs to be used in conjunction with playback so the media file can be specified? |
| 23:07.20 | Tozz_ | p3nguin: r u running 64 bit OS? |
| 23:07.46 | pabelanger | jjg: Yes. |
| 23:07.52 | pabelanger | exten => s,1,Progress() |
| 23:07.54 | p3nguin | I was able to run the register file without problems, so I expect benchg729 should run, too. |
| 23:08.05 | pabelanger | exten => s,n,Playback(tt-monkeys) |
| 23:08.17 | jjg | pabelanger: just got it to playback .. thanks a lot!! |
| 23:08.33 | pabelanger | np |
| 23:08.40 | p3nguin | tozz_: Apparently not. I thought it was, but I guess not. Even though I downloaded register from x86_64, it's a 32-bit binary. |
| 23:09.20 | Tozz_ | what does your uname -a say? |
| 23:09.43 | Tozz_ | or 'uname -m' |
| 23:09.56 | p3nguin | it says I should have checked the platform before I assumed it was 64-bits. |
| 23:10.02 | Tozz_ | :) |
| 23:10.32 | p3nguin | I guess I got this system confused with another one or something. |
| 23:11.11 | p3nguin | Someone needs to fix the README for g.729. |
| 23:11.18 | p3nguin | The benchg729 utility may be downloaded from: http://downloads.digium.com/pub/telephony/codec_g729/benchg729 |
| 23:11.28 | p3nguin | curl -O http://downloads.digium.com/pub/telephony/codec_g729/benchg729 |
| 23:11.32 | p3nguin | file benchg729 |
| 23:11.32 | p3nguin | benchg729: HTML document text |
| 23:11.39 | p3nguin | It's not at that location. |
| 23:12.38 | duna | wget http://downloads.digium.com/pub/telephony/codec_g729/benchg729/x86-64/benchg729-1.0.8-x86_64 |
| 23:12.38 | p3nguin | Well, nevermind. It's okay. I'm just irritated right now. |
| 23:12.45 | *** join/#asterisk marienz (~marienz@freenode/staff/marienz) |
| 23:12.47 | p3nguin | Disregard what I said. |
| 23:14.17 | duna | ok, i hate sip...so i leave for resting, tomorrow i'll hope for new fresh thoughts |
| 23:19.32 | *** join/#asterisk tvc123 (~tcameron@host-148.pl1071220.fiber.net) |
| 23:20.18 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 23:20.51 | tvc123 | dumb question about asterisk startup on ubuntu ... my init.d scrips works fine I have sym linked it to the rc#.d run levels but it still won't boot |
| 23:20.54 | tvc123 | any ideas |
| 23:21.12 | tvc123 | err I should say it still won't start on boot |
| 23:22.10 | p3nguin | Did you try using the update-rc.d command to tell it when to startup? |
| 23:22.29 | tvc123 | p3nguin: I did that as well |
| 23:22.40 | tvc123 | when direct sym linking didn't seem to work |
| 23:22.45 | p3nguin | I think that's how all startup stuff is supposed to be controlled. |
| 23:23.09 | p3nguin | But if you run /etc/init.d/asterisk start it works fine? |
| 23:23.11 | tvc123 | p3nguin: yeah although its just creating the sym links for you |
| 23:23.17 | p3nguin | oh, okay. |
| 23:23.23 | tvc123 | p3nguin: yeah works great if I run it directly |
| 23:23.33 | p3nguin | is not a Debian look-alike fan |
| 23:23.38 | tvc123 | and I have made sure the execute bit is set |
| 23:23.57 | p3nguin | That's pretty weird, then. It sounds like you know what you're doing. |
| 23:24.16 | p3nguin | Maybe someone will pipe in shortly. |
| 23:24.38 | tvc123 | yeah I'm not sure if it is trying to execute to early? but I have it set as S90 in init.d |
| 23:24.42 | leifmadsen | odd, I've never had to do anything beyong a 'make config' and perhaps an update-rc.d |
| 23:24.53 | leifmadsen | to the syslog! :) |
| 23:24.59 | tvc123 | looked there |
| 23:25.19 | tvc123 | I'm not seeing anything other than the messages that execute when I force it to start up |
| 23:27.58 | p3nguin | If I have installed my g729 license and codec, and g729 show licenses says "...3 licensed channels..." does that mean I'll only be able to have three individual call legs using g729 at any given time? |
| 23:28.48 | tvc123 | as a side question to that I have added /usr/sbin/dahdi_cfg into my init script is there a better place to put that |
| 23:29.05 | p3nguin | For 3 2-legged phone calls using g729 both from the phone and back out to the ITSP, I'd need a total of 6 licenses? |
| 23:29.08 | tvc123 | (although it won't startup on boot with or without that addition) |
| 23:31.46 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 23:32.52 | jjg | all, i'd like to use an ipkall.com DID to forward to my asterisk install .. do I just point the "username" to the extension in extensions.conf, or do I need to create a fake sip account to answer? |
| 23:33.54 | p3nguin | jjg: I can help you with this issue... |
| 23:34.50 | jjg | p3nguin: cool, tx .. it looks like I need to create a sip.conf entry that matches my extensions.conf |
| 23:36.14 | p3nguin | jjg: You'll need to create a SIP peer entry for ipkall if you're going to use SIP. http://pastebin.com/f8yhxAdt |
| 23:36.23 | p3nguin | jjg: Copy/paste |
| 23:36.30 | thedavidfactor | p3nguin, if I remember correctly g729 license are only needed for transcoding into or out of g729 |
| 23:36.41 | jjg | :q |
| 23:36.54 | thedavidfactor | so if you're simply connecting g729 endpoint together you don't need any licenses |
| 23:37.11 | p3nguin | thedavidfactor: It shows there is an encoder/decoder for each channel, but I'm just not sure how many call legs can be handled by just three licenses. |
| 23:37.14 | tvc123 | ok I'll take that back after removing the dahdi_cfg from my init.d scripts it is working again ... where should I be running this? |
| 23:37.14 | jjg | p3nguin: just pasted in the sip.conf peer entries |
| 23:37.26 | p3nguin | jjg: both of them, I hope. |
| 23:37.32 | jjg | p3nguin: yep |
| 23:37.54 | p3nguin | jjg: It's configured to send calls into the from-ipkall context inside extensions.conf. |
| 23:38.12 | p3nguin | jjg: Did you already sign up at ipkall and get your phone number? |
| 23:38.17 | jjg | p3nguin: y |
| 23:38.40 | jjg | p3nguin: so create a new context in extensions.conf now? |
| 23:39.03 | p3nguin | jjg: Yes, call it [from-ipkall] for easy identification. |
| 23:39.09 | p3nguin | jjg: Here's how I like to handle it: Login on ipkall. |
| 23:39.09 | p3nguin | |
| 23:39.31 | jjg | p3nguin: i'm in |
| 23:40.22 | p3nguin | jjg: Here's how I like to handle it: Login on ipkall. Mark the SIP button. In the "SIP phone number" box, type in your new IPKALL phone number. In the "SIP proxy" box, put your DNS name of IP address of your computer. Put in your email address and a password. SUBMIT. |
| 23:40.41 | jjg | p3nguin: ok, done |
| 23:41.11 | p3nguin | jjg: Now you can create a new extension in your [from-ipkall] context for your new phone number. |
| 23:41.35 | p3nguin | jjg: Do you want it to ring your SIP phone when a call comes in? |
| 23:41.38 | jjg | p3nguin: if I just use 's' instead of an extension, it will just drop into that path, right? |
| 23:41.47 | p3nguin | jjg: I wouldn't do that. |
| 23:41.48 | jjg | p3nguin: actually just wanting to drop into a menu basically |
| 23:42.08 | jjg | p3nguin: ok |
| 23:42.09 | p3nguin | jjg: It makes a lot more sense to use a real extension number, which matches your actual phone number. |
| 23:42.31 | jjg | p3nguin: ok .. can I still put a menu in front so that I don't dial my sip phone? |
| 23:42.35 | p3nguin | jjg: If you need to send it to a macro or IVR, use extension s there. |
| 23:42.58 | jjg | ok, so I'll create an extension in from-ipkall |
| 23:43.15 | jjg | ok, got it |
| 23:43.22 | p3nguin | exten => 4259980123,1,Goto(IVR,s,1); for example |
| 23:43.34 | jjg | p3nguin: ok |
| 23:43.53 | jjg | p3nguin: so i've created the extension in from-ipkall context |
| 23:44.04 | p3nguin | It just makes sense to me to use the phone number as the extension. |
| 23:44.23 | jjg | p3nguin: i see what you are saying now |
| 23:44.25 | jjg | good point |
| 23:44.30 | jjg | use the ipkall # |
| 23:44.47 | p3nguin | jjg: Save your changes for sip.conf and extensions.conf if you haven't already. Then run "sip reload" and "dialplan reload" in your Asterisk CLI. Make a call to your new number. |
| 23:47.56 | *** join/#asterisk babilen (~babilen@unaffiliated/babilen) |
| 23:48.33 | nestAr | digitmaps make me sad. |
| 23:50.24 | jjg | p3nguin: think there are some ports blocked .. looking into it |
| 23:50.59 | p3nguin | jjg: You need ports 5060 and 10000-20000, all UDP. |
| 23:51.16 | jjg | p3nguin: tx |
| 23:51.59 | tvc123 | ok not sure if this was the "right" way to resolve the issue but adding wctdm24xxp to /etc/dahdi/modules seems to fix things |
| 23:52.31 | tvc123 | I think that module was being loaded to late and that was causing the problems |
| 23:56.54 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |