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| 00:56.07 | jpcansa | how can I set multiple files in periodic-announce setting? queues.conf |
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| 03:38.55 | saint_ | hi all - is there a place where I can find the default rules for asterisk, and what they mean ? |
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| 03:39.58 | saint_ | I have a WaitExten that waits for an extension, obviously.. and I am trying to enter something i did not program, just to see the behavior and work around it. when i do that, asterisk sends a message in the console: WARNING[32351][C-00000074]: pbx.c:6440 __ast_pbx_run: Invalid extension '8', but no rule 'i' or 'e' in context 'autoattendant' |
| 03:40.11 | saint_ | what are the rules i and e ? |
| 03:43.10 | ChannelZ | well i is the 'invalid' extension. I have no idea what 'e' is.. |
| 03:46.50 | saint_ | ChannelZ: is there any doc about this ? |
| 03:47.20 | ChannelZ | https://wiki.asterisk.org/wiki/display/AST/Handling+Special+Extensions |
| 03:47.41 | ChannelZ | I see some old patches about 'e' being an 'error' extension but honestly this is the first I've heard of it. |
| 03:48.44 | saint_ | ChannelZ: that is perfect. thanks a ton for the URL |
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| 03:50.51 | ChannelZ | sure |
| 03:53.19 | saint_ | and what is the best way, if I have a menu (auto attendant) and I want to keep statistic on what choice was the most used ? |
| 03:53.27 | saint_ | do you know anything about that by any chance ? |
| 03:55.11 | saint_ | I guess I can use a local database.. |
| 03:57.28 | ChannelZ | or the call logs |
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| 04:28.27 | Penguin | Extension e is basically when nothing else worked. |
| 04:29.14 | Penguin | You should always define extension i if there is a chance that an invalid extension can be entered. |
| 04:30.49 | ChannelZ | It's weird I've never seen it documented before |
| 04:31.10 | Penguin | I almost never read documentation, so I haven't either. :) |
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| 04:31.27 | Penguin | But I first saw it when I changed to asterisk 1.8. |
| 04:31.46 | Penguin | I saw the same warning saint_ saw. |
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| 09:25.25 | War_Bear | Hi Guys. I have an Asterisk 1.8.21.0 install that has been running fine for about 100 days. However with no increase in traffic we are seeing dial take in excess of 20 seconds to complete as of last night. Any one have any ideas where to start debug? |
| 09:26.14 | charkee | Hi guys, have you ever had Cisco SPA phones that would reboot hourly with "Backup IP" as the reason? |
| 09:31.54 | Chainsaw | charkee: IP address conflict? |
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| 10:06.57 | Greenlight | War_Bear: DNS issues? |
| 10:07.13 | War_Bear | its dialing @ and ip |
| 10:07.17 | War_Bear | would this still be an issue? |
| 10:07.27 | Greenlight | Yes, I have seen it be |
| 10:07.33 | War_Bear | any idea how I can correct it? |
| 10:07.35 | Greenlight | Asterisk needs dns for the oddest tbhings |
| 10:07.40 | Greenlight | Check your dns is working ^^ |
| 10:08.01 | War_Bear | ping to google.co.uk seems to resolve in normal time |
| 10:08.23 | Greenlight | Perhaps not that then |
| 10:08.33 | Greenlight | Grab a SIP trace and see where the holdup is |
| 10:08.47 | vandyk | does anyone has issues with calls being canceled on Asterisk (SIP response 486 and 487)? |
| 10:09.08 | vandyk | I'm running Asterisk 11.5.1 |
| 10:09.40 | War_Bear | We did just that. Weirdly the sip packets are being handled quickly. The issue seems to be post dial delay. When app_dial runs it take roughly 25 seconds to get to the "app_dial.c: -- Called" line |
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| 10:10.51 | Greenlight | So, it's 25 seconds *before* even the inital invite gets sent? |
| 10:11.15 | War_Bear | yes |
| 10:11.58 | Greenlight | Enable debug and see if anything jumps out as to where it might be getting held up. I still reckon DNS could be an issue. |
| 10:13.46 | War_Bear | I think you might be right |
| 10:13.50 | War_Bear | I changed DNS server |
| 10:13.50 | War_Bear | s |
| 10:13.53 | War_Bear | Calls are flowing |
| 10:16.09 | War_Bear | Greenlight: You sir are a legend |
| 10:16.58 | Greenlight | :) |
| 10:17.00 | Greenlight | Glad to help |
| 10:17.13 | Greenlight | As a bit of advice, you should look at running a dns server locally on the box |
| 10:17.45 | War_Bear | I have one one site luckly |
| 10:20.14 | Greenlight | DNS outages or problems can effect asterisk in weird and wonderful ways |
| 10:23.32 | War_Bear | Greenlight: I'd kiss you if I could |
| 10:25.09 | Greenlight | In this instance, I think distance is a wonderful thing :) |
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| 10:28.44 | War_Bear | :) |
| 10:28.49 | War_Bear | Thanks agian |
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| 10:35.04 | danbell77 | Anyone home? |
| 10:37.26 | wdoekes | ~ask |
| 10:37.27 | 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. |
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| 10:52.52 | vandyk | anyone knows if there is a timeout on Asterisk to complete a call? |
| 10:53.20 | vandyk | for example, if a call is not completed on 50 seconds its dropped |
| 10:54.49 | Greenlight | Define "completed" |
| 10:57.21 | vandyk | for example, I have a server A that receives a request for dial out, the server A send the call to a trunk on server B. the server B send the call to GSM gateway. The gateway try to complete the call dialing out |
| 10:57.55 | vandyk | this transaction has some duration |
| 10:58.36 | vandyk | I'm having an issue about 30% to 40% of all calls being cancelled in my server |
| 10:58.52 | WIMPy | There are timeouts everywhere. |
| 10:59.02 | vandyk | the GSM gateway tell me that the IP side cancels the call |
| 10:59.21 | vandyk | I can't find the reason why the call is being cancelled |
| 10:59.36 | vandyk | the only thing that came up on my mind is something about timeouts |
| 11:00.25 | vandyk | if the call is not completed (reach the final destination) in 30 seconds, or 50 seconds whatever... Asterisk cancel the call |
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| 11:02.43 | Greenlight | What timeout are you using on Dial ? |
| 11:05.38 | vandyk | the default on Asterisk |
| 11:05.43 | vandyk | I did not change |
| 11:05.50 | WIMPy | The default is unlimited. |
| 11:06.05 | Greenlight | Lets see your dialplan for a sec |
| 11:06.45 | vandyk | from which server? Server A or Server B? or both? |
| 11:06.57 | Greenlight | Both |
| 11:07.14 | WIMPy | Which one cancels the call? |
| 11:07.26 | vandyk | server B, I think |
| 11:10.30 | vandyk | http://pastebin.com/vCC81XjF |
| 11:10.40 | vandyk | this is from SErver B |
| 11:11.21 | Greenlight | I don't see a Dial ? |
| 11:11.26 | WIMPy | That's not doing any calls. |
| 11:11.28 | Greenlight | Unless I'm being blind |
| 11:11.41 | vandyk | I'll paste the rest |
| 11:12.19 | WIMPy | Could it be that you should take the whole issue to #freepbx? |
| 11:12.34 | Greenlight | Looks freepbxy |
| 11:13.58 | vandyk | http://pastebin.com/b7uFSFC8 |
| 11:14.04 | vandyk | yes it is FreePBX |
| 11:14.20 | vandyk | not, its not related with FreePBX |
| 11:14.23 | Greenlight | ~frepbx |
| 11:14.26 | WIMPy | It |
| 11:14.29 | Greenlight | ~freepbx |
| 11:14.29 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 11:14.40 | Greenlight | <vandyk> not, its not related with FreePBX <-- How so? |
| 11:14.57 | WIMPy | There's no point in us finding the issue, as we caouldn;t tell you how to fix it anyway. |
| 11:15.09 | WIMPy | You have to use freepbx to fix freepbx. |
| 11:15.12 | Greenlight | exten => s,n,Dial(${OUT_${DIAL_TRUNK}}/${OUTNUM},${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS}) |
| 11:15.23 | vandyk | I already talked with other guys that looked at SIP DEBUG |
| 11:15.28 | Greenlight | Why would you say "the asterisk default" timeout |
| 11:15.37 | vandyk | and they said it is not related with FreePBX |
| 11:15.47 | vandyk | that is the reason I'm asking here |
| 11:16.00 | WIMPy | Yes, it'd clearly TRUNK_RING_TIMER, whatever that is. |
| 11:16.06 | Greenlight | mmhmm |
| 11:16.09 | WIMPy | it's |
| 11:16.41 | Greenlight | Why not be honest, when I asked what options you were passing to Dial |
| 11:16.59 | Greenlight | And say "whatever freepbx does..." |
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| 11:17.38 | WIMPy | It's definitely not the Asterisk default. |
| 11:17.43 | Greenlight | Indeed |
| 11:18.39 | vandyk | Guys I'm sorry if I answered the wrong way. It was not my intention. Clearly you separate things. As far as I knew FreePBX just used Asterisk defaults. |
| 11:18.53 | vandyk | I was honest when I answered |
| 11:19.12 | WIMPy | I very much doubt that freepbx uses Asterisk defaults anywhere. |
| 11:19.21 | WIMPy | It's like raping Asterisk. |
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| 11:20.09 | Greenlight | A harsh analysis. But it does completely abstract you from what asterisk is actually doing and from the dialplan, and it's very difficult to support by someone who's not heavily involved |
| 11:20.51 | vandyk | Ok. |
| 11:20.57 | Greenlight | For example I've no idea what ${TRUNK_RING_TIMER} is set to, but at a guess that's where your timeout is introduced |
| 11:21.14 | Greenlight | At a guess it's on one of the configuration screens |
| 11:21.31 | WIMPy | And we have no idea how to change it, either. |
| 11:21.40 | vandyk | I looked to .conf files now, and this variable has a value of 300 seconds |
| 11:21.58 | Greenlight | Hmm |
| 11:22.14 | Greenlight | Usually freepbx stores is stuff in a DB |
| 11:22.24 | WIMPy | It's not read from a database? Only ONE global value? |
| 11:22.38 | Greenlight | Why not ask the folks who use freepbx in #freepbx |
| 11:22.49 | vandyk | yes, it does. But after you apply the changes on GUI, it will create the .conf files |
| 11:22.59 | vandyk | I'll ask there. |
| 11:23.19 | vandyk | anyway thanks for help |
| 11:23.23 | Greenlight | Post dial delay *should* not cause say your SIP timeouts to trigger |
| 11:23.41 | Greenlight | So if everything is behavoing correctly, that Dial timeout is what would cause your issue imo |
| 11:24.59 | vandyk | I'm not an expert on Asterisk or SIP. I though first that the problem was with GSM gateway. The GSM gateway support team told me that the problem was with Asterisk side. So I started researching and looking for help on this. |
| 11:26.31 | WIMPy | And now the support team thinks Asterisk is shit because you mixed it up with freepbx. |
| 11:26.34 | Greenlight | GSM gateways are generally used to call mobiles. Calling mobiles often has a larger than usual post dial delay (sometimes 10 seconds or so). The Dial timeout starts as soon as the INVITE is sent |
| 11:27.36 | WIMPy | Wou should hopefully get some progrss in between. |
| 11:30.17 | Greenlight | Yea, but if your timeout is set to say 20 seconds (5 rings so not an unusual timeout), then when calling a mobile over GSM gateway if it doesn't start to 183/180 RING/PROGRESS for 15+ seconds, then it's going to cause issues which appear like calls being canclled |
| 11:30.38 | Greenlight | If your timeout was set to even less time, then potentially calls are cancelled before they've even started to progress at all |
| 11:30.55 | WIMPy | Absolutely possible. |
| 11:31.26 | vandyk | Greenlight this timeout your talking about would be on GSM gateway or at FreePBX side? |
| 11:31.53 | WIMPy | Both. |
| 11:32.08 | Greenlight | The one I'm speaking of would be getting passed to Dial |
| 11:32.19 | WIMPy | Even the calling phone might have one. |
| 11:32.35 | Greenlight | GSM may or may not have it's own timeouts, but it sounds that you've had their people looking into it, and they're seeing calls "CANCELLED" |
| 11:32.56 | vandyk | Greenlight you're corret |
| 11:33.51 | vandyk | on SIP DEBUG (Asterisk) the SIP receive a message of CANCEL |
| 11:34.02 | Greenlight | Now, the underlying issue may be that the GSM gateway does nothing with the call for 60 seconds cause it's not working correctly, and they Asterisk cancelled the call. It's an easy way out for their techies to say "yea, your cancelling calls" |
| 11:34.11 | Greenlight | vandykq: After how long ? |
| 11:34.47 | vandyk | it depends, there is no a pattern, but usually between 30 and 50 seconds |
| 11:35.05 | Greenlight | See, that's a damn long time for a call to start ringing, or to wait for it to be picked up... |
| 11:35.16 | Greenlight | I would quite expect a cancel after that length of time... |
| 11:35.55 | WIMPy | Yes, but it should be a contant time. |
| 11:36.00 | Greenlight | It should yea |
| 11:36.04 | vandyk | I discussed with them about the amount of calls that the gateway would be able to handle |
| 11:36.04 | Greenlight | Which is odd |
| 11:36.41 | vandyk | the issue occures more frequently when I have a huge amount of calls being sent to gateways |
| 11:36.54 | vandyk | about 45 or 50 calls at same time |
| 11:36.58 | WIMPy | So it might be a sip timer that is started after some progress message that arrives at different times. |
| 11:37.04 | vandyk | I have 3 different gateways |
| 11:37.38 | WIMPy | Maybe you should ask the network operator how many calls they can handle. |
| 11:37.50 | WIMPy | How many radios do these GWs have each? |
| 11:38.08 | vandyk | 32, 16, 16 |
| 11:38.08 | danbell77 | Anyone here use 7970s with SIP? I have an issue that it does not complete a conference call (the join). |
| 11:39.46 | WIMPy | I'd check the radio logs. Maybe the calls fail due to congestion and it just doesn't tell you. |
| 11:39.52 | Greenlight | vandyk: It sounds like the gateways are introducing delays are larger call volumes |
| 11:39.55 | WIMPy | SIP doesn't have conferences. |
| 11:40.19 | WIMPy | Or at least I'm not aware of any such extension. |
| 11:40.51 | Greenlight | I think those phones do their own conferencing, from the SIP side it's just two separate calls to the phone./ |
| 11:41.29 | WIMPy | Probably, but that's local to the phone then. |
| 11:41.32 | vandyk | Greenlight: I'll take a look again on parameters of gateways |
| 11:41.45 | Greenlight | vandyk: Try increasing your dial timeout |
| 11:42.11 | vandyk | on asterisk side, right? |
| 11:42.36 | Greenlight | Yea, although I doubt you users are going to wait for that long for it to start ringing |
| 11:42.45 | Greenlight | In fact, is this automated dialling or manual dialling ? |
| 11:43.13 | vandyk | automated dialling |
| 11:43.20 | vandyk | its a call center |
| 11:43.27 | vandyk | with predictive dialing |
| 11:43.40 | Greenlight | Perhaps your dialler is giving up on the calls then |
| 11:44.02 | Greenlight | I'd imagine your dialler is set to "ring" for 20 seconds |
| 11:44.11 | vandyk | humm |
| 11:44.15 | Greenlight | So, there's your issue ... |
| 11:44.33 | vandyk | you pointed to something that I did not think yet |
| 11:44.46 | Greenlight | Remember, its the originator of the call who likely cancells it |
| 11:44.48 | vandyk | I'll ask the guys from there |
| 11:45.25 | Greenlight | But I would agree with WIMPy in that it appears to be an issue with your GSM's which is introducing the delays |
| 11:45.27 | WIMPy | >>Even the calling phone might have one. |
| 11:45.52 | Greenlight | I don't think there is a "calling phone" if he's using a predictive dialler |
| 11:46.12 | WIMPy | That's the "phone" in that case. |
| 11:46.23 | Greenlight | Yes |
| 11:47.02 | WIMPy | And 64 channels is quite a lot for a mobile network. |
| 11:47.03 | Greenlight | Remember as well, predictive diallers place a huge stress on the network by their very nature |
| 11:47.08 | Greenlight | mmhmm |
| 11:47.34 | WIMPy | You might want to ask the network operator to give you a cable, so they don;t have to set up more antennas for you. |
| 11:47.42 | *** join/#asterisk kresp0 (~kresp0@231.Red-2-136-86.dynamicIP.rima-tde.net) |
| 11:47.51 | Greenlight | Indeed.. the oddity that is GSM gateways |
| 11:48.24 | Greenlight | I'm glad at least here in UK, ITSP's are now chearper over SIP and we don't have to use GSM anymore |
| 11:48.33 | WIMPy | Are they on different networks? |
| 11:48.52 | *** join/#asterisk danjenkins (~danjenkin@62.254.236.250) |
| 11:49.36 | Greenlight | You could even be hitting capacity at your cell tower |
| 11:49.55 | WIMPy | That's what I suggested before. |
| 11:49.56 | vandyk | they are on different network |
| 11:50.09 | Greenlight | Or perhaps the line hasn't completely "cleared" and your predictive dialler is immediately placing another call |
| 11:50.09 | WIMPy | It seems rather likely. |
| 11:50.15 | vandyk | but throught IPSEC tunnel |
| 11:50.22 | Greenlight | Huh? |
| 11:50.25 | vandyk | this is the gsm gateway http://tinypic.com/r/2i7ljk1/5 |
| 11:50.27 | Greenlight | GSM through IPSEC? |
| 11:50.30 | Greenlight | what? |
| 11:50.31 | vandyk | no |
| 11:50.31 | WIMPy | What? |
| 11:50.55 | vandyk | the predictive dialer is on a network 10.50.X.x |
| 11:51.05 | vandyk | that send the calls to my server on 10.193.x.x |
| 11:51.21 | Greenlight | Oh, the network between those ? |
| 11:51.33 | vandyk | the communication between both servers is IPSEC tunnel |
| 11:51.41 | Greenlight | I don't think that's related here |
| 11:51.50 | WIMPy | no |
| 11:52.57 | WIMPy | But anyway you're using a lot of carriers. |
| 11:52.58 | Greenlight | vandyk: Under FreePBX -> Settings -> General Settings -> Ringtime Default |
| 11:54.04 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 11:54.11 | vandyk | yes I changed that to 3000 |
| 11:54.34 | Greenlight | And on the other box ? |
| 11:54.59 | Greenlight | Or is that the "dialler" one? |
| 11:55.46 | vandyk | looking to the scenario, the dialler is the box one, that will send to box 2 that will send to box 3. |
| 11:55.56 | vandyk | Actually I just have control of boxes 2 and 3 |
| 11:56.07 | vandyk | I already changed on box 3 |
| 11:56.13 | vandyk | changing now on box 2 |
| 11:56.14 | Greenlight | Check box 2 :) |
| 11:57.30 | vandyk | changed there :-) |
| 11:57.52 | Greenlight | WHat *was* it set to ? |
| 11:58.01 | vandyk | 300 seconds |
| 11:58.07 | vandyk | changed to 3000 seconds |
| 11:58.17 | Greenlight | Yea, so likely the cancellation is coming from the dialler |
| 11:58.28 | Greenlight | Which is to be expected, diallers are renowned for being patient :) |
| 11:58.39 | eirirs | lol |
| 11:58.56 | vandyk | I just sent an email to tech guys from dialler asking about if there is a timeout set there |
| 11:59.37 | Greenlight | The question is whether your dialler can actually detect when the call truely starts to "ring", or just starts its timer blinly from when the call is placed. |
| 12:00.01 | Greenlight | If it can't truly detect when ringing starts, then they can't really increase their timeout |
| 12:00.40 | Greenlight | A lot of these diallers are just rebadged vicidial in my experience |
| 12:00.53 | vandyk | I think that is vicidial |
| 12:01.25 | Greenlight | SO, either fix the underlying issue causing delays, or get a different dialler ^^ |
| 12:01.53 | WIMPy | vicidial can't detect ringing? |
| 12:02.23 | Greenlight | As far as I know it's "ringing" timer starts when the call is placed, and NOT when 180/183 is received |
| 12:02.40 | WIMPy | Sounds pretty unusable. |
| 12:03.44 | Greenlight | I know we had to make a custom patch to asterisk to fire an AMI event when media (183) started |
| 12:03.53 | Greenlight | Because by default, there isn't one |
| 12:04.26 | WIMPy | huh? |
| 12:04.37 | vandyk | I'll confirm with dialler team if is vicidial that they are using and confirm either if there is a timeout option there |
| 12:04.49 | Greenlight | YOu know in the CLI the "SIP/100-asasadas has started to make progress" type messages |
| 12:05.14 | WIMPy | newstate: ringing |
| 12:05.25 | Greenlight | Yea, that one lies :) |
| 12:05.55 | WIMPy | That is bad :-( |
| 12:05.58 | Greenlight | Or is only fired if you get a SIP 180 |
| 12:06.57 | WIMPy | That's what I'd expect. |
| 12:07.13 | Greenlight | SIP 180 isn't required to be sent, and many ITSP don't |
| 12:07.40 | WIMPy | That's not Asterisks fault. |
| 12:07.45 | Greenlight | Esepcially when calling the PSTN, a lot just go straight to 183 PROGRESS and give you media of the ringing |
| 12:08.04 | WIMPy | But it seems odd not to send it, unless the call was answered without ringing. |
| 12:08.19 | Greenlight | 183 is a very similar state |
| 12:08.31 | WIMPy | Are they using analog gateways? |
| 12:08.44 | Greenlight | And even triggers the messages in the CLI to tell you it's "proceeding" |
| 12:08.49 | Greenlight | Which was the odd bit |
| 12:09.14 | Greenlight | I would watch the CLI along side the AMI and when my phone started to ring,the CLI would say, and yet AMI was empty |
| 12:09.16 | WIMPy | But it is somethign different. |
| 12:10.33 | WIMPy | I guess the issue is that there are too many gateways out there that are just too bad to be usefull. |
| 12:11.31 | Greenlight | When the ultimate endpoint is a 100 year old mostly analogue network, there's going to be things that are open to interpertation |
| 12:12.21 | WIMPy | Yes, but analog networks should have become rare. |
| 12:12.39 | Greenlight | Not for the last mile.. |
| 12:12.55 | WIMPy | That doesn't matter. |
| 12:14.14 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 12:15.03 | vandyk | well talked with dialler support team, they will check if there is a timeout, but for the answer of the guy: hummm I see.... I'll check this |
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| 12:16.45 | Greenlight | vandyk: Grap a SIP trace of some of the problematic calls so you can see exactly what's going on and where the delays are. |
| 12:16.59 | vandyk | sure |
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| 12:45.13 | martinfletcher_ | hey all |
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| 13:20.25 | danbell77 | Surely someone uses a Cisco 7970 with SIP on Asterisk... |
| 13:24.27 | Greenlight | Ask your asterisk related question, and see :) |
| 13:28.34 | Penguin | ~TIAS |
| 13:28.34 | infobot | Try it and see! You will learn more, and probably have a better feeling of satisfaction, if you experiment with simple cases to learn a language than you will by asking yes or no questions. |
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| 13:29.41 | Greenlight | Truer words have never been spoken |
| 13:33.10 | danbell77 | My question is has anyone got conferencing to work on a Cisco 7970 with SIP? |
| 13:33.53 | Greenlight | Doesn't sound very asterisk related ^^ |
| 13:34.21 | Greenlight | Assuming your meaning the phones built in conferencing features |
| 13:34.26 | danbell77 | It is very Asterisk related. The Cisco 7960 and 7940 work...just the 7970. |
| 13:34.37 | Greenlight | So, are you using MeetMe or ConfBridge ? |
| 13:35.04 | danbell77 | I didn't preface the question with Asterisk as that was a given by the channel. ;) |
| 13:35.08 | danbell77 | Confbridge. |
| 13:35.38 | Greenlight | So the endpoint makes no difference, it just sees a SIP call. |
| 13:35.46 | Greenlight | Lets see your dialplan, and a snippet of the call in question |
| 13:35.53 | Greenlight | ANd, what's actually not working ? |
| 13:37.00 | danbell77 | They issue does revolve around the handset. Effectively it looks like the 7940s and 7960 join the two legs themselves whereas the 7970 does not appear to do the same. |
| 13:37.45 | danbell77 | I have debug and packet captures for this. |
| 13:39.19 | danbell77 | This isn't confbridge application...I'm trying to use the phones builtin conference button...which appears to be different on the 7970 |
| 13:42.28 | danbell77 | both legs complete...I have the calls working. The dialplan is fine as the same config works if I change the handset... |
| 13:43.20 | danbell77 | It appears the issue lies with the handset itself, I'm really looking for someone who uses these handsets and how (if) they have overcome the conference issue. |
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| 13:52.20 | Penguin | Does the conference button normally join a meetme or confbridge, or does it do something else? |
| 13:53.53 | *** join/#asterisk gmalsack (~gmalsack@23.30.198.161) |
| 13:55.23 | danbell77 | No...it normally just joins the legs on the endpoint itself. |
| 13:55.33 | Penguin | With chan_sccp there is a meetme button which you can program to join a specific meetme conference. I'd have to go look to see if it supports confbridge. |
| 13:55.52 | Greenlight | I thought he said he was using SIP |
| 13:56.02 | Penguin | Yep. |
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| 13:57.06 | Penguin | Does the conference button allow more than two callers plus the handset on the call? |
| 13:57.26 | danbell77 | It can...although my requirement is only two legs. |
| 13:57.32 | danbell77 | Yes..and it must be SIP. |
| 13:57.53 | Penguin | 7900 series phones and SIP... |
| 13:58.08 | Penguin | That always results in reduced features and unhappy people. |
| 13:58.16 | danbell77 | I have 7940s and 7960s working fine. |
| 13:58.34 | Greenlight | So, *only* 30% failure :) |
| 13:58.39 | Penguin | "working" and "fine" are clearly relative terms. |
| 13:59.08 | danbell77 | I can deliver virtually my entire feature set, (except some "automated" delivery of speed dials to the 7960s) |
| 13:59.28 | Penguin | I used SIP on my 7960s/7940s for a while. The available features were extremely limited. That was my reason for going to chan_sccp. |
| 13:59.44 | danbell77 | Hahaha. 33.3R% if we want to be precise. ;) |
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| 14:03.22 | dym | dronacid carries a virus! |
| 14:04.32 | danbell77 | I can do everything I need with 7940 and 7960 on SIP with Asterisk. What problems did you encounter? |
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| 14:17.47 | anonymouz666 | you can do anything at zombo com |
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| 14:22.50 | SuperNull | hey gents. |
| 14:24.46 | anonymouz666 | Super Null Pointer |
| 14:24.49 | anonymouz666 | not good |
| 14:25.02 | anonymouz666 | it will crash |
| 14:25.21 | Greenlight | Only if you dereference it |
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| 15:34.27 | Penguin | anonymouz666: Anything is possible... at Zombo com. |
| 15:34.50 | Penguin | You're the first person I ever saw make a reference to that site. I didn't even know if it was still up. |
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| 15:48.25 | Scapal | In a gosub context called by the 'queue' cmd on queue member answer, how to get the wait time for that call ? It's not in the channel variables. |
| 15:54.00 | gmalsack | read the queues.conf.sample file from the source. there is a directive you can enable which will set a channel variable on answer of the queue call. |
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| 16:02.33 | Scapal | gmalsack: thx |
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| 16:10.22 | asteriskmonkey | heya anyone know this? with Asterisk realtime and defining the db, is it possible to define a backup db incase of a primary db fail/ |
| 16:11.22 | gmalsack | if you're using res_odbc wouldn't that be a function of the odbc driver? |
| 16:11.45 | asteriskmonkey | i dont use res_odbc i use native |
| 16:12.09 | karl-s | i thought native got deprecated in 1.8? |
| 16:12.47 | navaismo | deprecated but you can compile it even in 11 |
| 16:12.55 | gmalsack | nope. still in 11 too. but rumor has it that it will be so I went with res_odbc |
| 16:13.35 | gmalsack | I don't quite get that either, it just adds a layer of resource overhead, and another point of failure.... :-( |
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| 16:14.47 | gmalsack | but I guess I can see it from a developers point of view. less code to manage. |
| 16:14.55 | karl-s | to be honest, i liked the native dialplan mysql app instead of this crazy func ODBC. Things are much more difficult with it |
| 16:15.28 | asteriskmonkey | yeah odbc seems to be an unessacry chain failure layer |
| 16:16.07 | asteriskmonkey | ive stoped using most dialplan db calls though, switch to curl and array works better |
| 16:17.06 | karl-s | i'm tempted to just have 1 function in func_odbc: writesql=${VAL1} |
| 16:17.27 | karl-s | but i dont since that would be the easy way out... |
| 16:18.17 | karl-s | I like curl but that sounds like even more overhead... |
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| 16:29.33 | Dovid | is there any way of getting the codec information on a call into a variable? i am trying to see if i can get the codec order from an invite into a call |
| 16:31.04 | navaismo | parsing the invite via sip debug maybe? |
| 16:32.05 | Dovid | navaismo: I want to get it into a variable. |
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| 16:34.57 | navaismo | yea |
| 16:35.12 | navaismo | so now how are you trying to doing that |
| 16:35.23 | navaismo | via ami, external script or ...? |
| 16:36.12 | Dovid | navaismo: Via the dial plan but it seems it can't be done. http://lists.digium.com/pipermail/asterisk-users/2012-March/271070.html |
| 16:40.24 | navaismo | hmmm do you need it to force the other client? You can force the codec on a call with the SIP_CODEC channel variable |
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| 16:58.20 | saxa | anonymouz666: thats what WIMPy said, I wanted to know which card to buy to use digitronco with my already existing * box |
| 16:58.54 | saxa | btw thanks to all for support and suggestions, today i bought a TA133 Digium card |
| 16:59.01 | saxa | the PCI express one |
| 16:59.27 | WIMPy | Any card should do. Never heard that name before. |
| 16:59.28 | saxa | hopefully this will work ok and will permit me to maintain the system configured as it is now. |
| 16:59.48 | WIMPy | But for the old stuff I'd suggest one that used dahdi. |
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| 17:00.40 | saxa | http://www.digium.com/en/products/telephony-cards/digital/single-span |
| 17:00.47 | saxa | WIMPy: its TE133 |
| 17:00.49 | saxa | sorry |
| 17:01.11 | saxa | this one should use dahdi iinw |
| 17:01.48 | WIMPy | A new design and a new name? |
| 17:01.56 | *** part/#asterisk sarobat (~saroth@24.244.29.118) |
| 17:01.58 | saxa | yes seems so |
| 17:02.14 | saxa | anyway this one is with hw echo cancellation |
| 17:02.27 | saxa | for about a 35% cheaper than was the TE122 |
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| 17:25.54 | vandyk | Greenlight: are you there? |
| 17:26.13 | vandyk | remember about the call canceled? I got a SIP trace |
| 17:26.19 | vandyk | http://pastebin.com/xGj8EZ7t |
| 17:27.51 | WIMPy | From which of the three possible points? |
| 17:28.49 | vandyk | this one from the box 2 (the one that receive the calls from dialler) |
| 17:29.44 | WcktKlwn | Is there an easy way to setup a remote SIP phone through a NAT? |
| 17:37.23 | anonymouz666 | saxa: for just a single span is safe to use SW echo can |
| 17:37.48 | anonymouz666 | i.e.: oslec |
| 17:42.12 | [TK]D-Fender | WcktKlwn: * has a few basic settings to do to account for this and that's it |
| 17:42.34 | [TK]D-Fender | WcktKlwn: There is one part to consider for the PHONE's environment, and another if * itself is behind a NAT |
| 17:45.49 | WcktKlwn | [TK]D-Fender: both devices are behind NAT's. Is there any howto's on how to set this up? |
| 17:46.05 | [TK]D-Fender | ~nat |
| 17:46.06 | infobot | i heard nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 17:48.27 | vlad_starkov | Question: is it possible to implement attended new ConfBridge member join? |
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| 18:28.15 | *** join/#asterisk redotis (~redotis@nc-76-5-176-123.dhcp.embarqhsd.net) |
| 18:28.32 | redotis | How can you set a time for a parking lot? |
| 18:30.52 | redotis | Where do you set the time for parked calls in feature.conf? How do you do it? |
| 18:39.19 | redotis | I found it. |
| 18:39.28 | redotis | Boy this room sucks. |
| 18:39.40 | redotis | Anybody know how to tell what time is used from the CLI? |
| 18:40.24 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 18:41.30 | [TK]D-Fender | What a wonderful thing to see broadcast for all to see. It's like someone is looking to be invited to leave... |
| 18:44.27 | *** join/#asterisk jpoz (~jpoz@199.108.71.40) |
| 18:47.56 | ChannelZ | features show |
| 18:48.33 | ChannelZ | parkedcalls show |
| 18:52.33 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 18:53.13 | *** join/#asterisk bkruse (~Adium@c-50-130-4-19.hsd1.al.comcast.net) |
| 18:54.07 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 18:59.49 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-tjlejqbxyxtzmfzr) |
| 18:59.49 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 19:05.09 | *** join/#asterisk Docfxit (~Console1@netblock-75-79-6-10.dslextreme.com) |
| 19:08.50 | Docfxit | I have one voice mail box that isn't responding. I can't leave a message. How can I reset the voice mail box? |
| 19:09.32 | navaismo | delete it |
| 19:13.24 | *** join/#asterisk kchehab (~kchehab@92.62.170.30) |
| 19:17.36 | *** join/#asterisk danjenkins (~danjenkin@cpc3-folk2-2-0-cust34.1-2.cable.virginmedia.com) |
| 19:18.26 | Docfxit | How can I delete it? Then re-create it? |
| 19:20.28 | [TK]D-Fender | go to the VM folder |
| 19:20.33 | [TK]D-Fender | Delete the entire folder |
| 19:20.36 | [TK]D-Fender | leave a new vm |
| 19:20.38 | [TK]D-Fender | done |
| 19:21.40 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 19:24.27 | navaismo | OMG i loved the show(conference) of Olle Johansson from now and then im a groupie |
| 19:27.22 | Docfxit | d-fender: Thank you very much. I deleted the inbox folder and the old folder and it started working. |
| 19:31.57 | Kobaz | anyone have a rfc2833 rtp debug example handy |
| 19:32.04 | Kobaz | like what's the log line look like when getting dtmf over rfc2833 |
| 19:33.05 | redotis | Dang I thought I blocked you TKD-Fender...wonder why I'm seeing your posts. |
| 19:33.12 | redotis | thanks Channelz |
| 19:34.30 | Kobaz | but fender is so wonderful |
| 19:34.32 | Kobaz | why would you block him |
| 19:35.07 | redotis | I believe he's among a handful of jerks here. |
| 19:35.07 | redotis | I can't remember if it was him. |
| 19:35.10 | Kobaz | unpossible |
| 19:35.18 | Kobaz | [TK]D-Fender: right? |
| 19:35.22 | redotis | I had a few ignored. I'm not sure if it was him. |
| 19:35.30 | redotis | yeah |
| 19:35.44 | redotis | Advice is one thing but snide remarks is another. Maybe it's not him. |
| 19:36.02 | redotis | Personally I'd rather not get help if it means you have to be treated like crap. |
| 19:36.15 | Kobaz | welcome to open source |
| 19:36.26 | [TK]D-Fender | Kobaz: He finds ways to misread all sorts of things and is bitter to anyone who doesn't hand him exactly what he wants on a silver platter. |
| 19:36.36 | Kobaz | not that I like that type of behavior.. but that's the way it is |
| 19:36.44 | [TK]D-Fender | Kobaz: including "everyone" for his waiting a few minutes without getting answered and then : |
| 19:36.46 | [TK]D-Fender | [14:39]redotisBoy this room sucks. |
| 19:36.50 | Kobaz | hehe |
| 19:36.51 | Kobaz | yeah |
| 19:36.52 | Kobaz | see |
| 19:36.56 | Kobaz | that's not a good way to get help |
| 19:37.05 | redotis | Kobaz stop sucking up |
| 19:37.10 | Kobaz | heh |
| 19:37.11 | redotis | TK is rude as hell |
| 19:37.14 | Kobaz | i'm friends with everyone |
| 19:37.17 | redotis | lol |
| 19:37.18 | Kobaz | (ask anyone) |
| 19:37.29 | [TK]D-Fender | Kobaz: He burned himself off of the 2 most active people who give out information here on his first day and got himself kicked by an op |
| 19:37.38 | Kobaz | see that's bad |
| 19:37.41 | Kobaz | tisk tisk |
| 19:37.54 | Kobaz | now.. hug and make up you too |
| 19:37.58 | redotis | D-Fender is worse. He's established a reputation as an ass that makes folks suffer for advice. |
| 19:38.01 | Kobaz | and get on with the asterisking |
| 19:38.36 | Kobaz | waits for windows to finish updating so i can shut down this noisy machine |
| 19:38.55 | redotis | TK is from the 80s school of thought that says if you know somethign someone else doesn't, you are better than them and have the right and responsibility to treat them like garbage. |
| 19:38.58 | [TK]D-Fender | Kobaz: We lost #3 while Manx is on "walkabout" for an indefinite period |
| 19:39.06 | Kobaz | mm |
| 19:39.38 | Kobaz | redotis: the big problem with doing free support, like we do here |
| 19:40.00 | Kobaz | is people don't do their homework, and come in with half completed information (or none at all) and expect help |
| 19:40.01 | redotis | Somehow he thinks that if he's "active" it makes up for bad behavior. I just wonder how many other "active" members he's turned off from this room and how bad a reputation he's giving the Digium/asterisk community. |
| 19:40.25 | redotis | If I were a member of sales at Digium, I'd hate the guy. |
| 19:40.51 | Kobaz | so there's a legitimate reason for frustration when we're here giving advice and people don't take it and expect results. |
| 19:41.11 | [TK]D-Fender | redotis here has spread nothing but bile since his arrival sporting an inflated sense of entitlement and not listeneing to even direct answers from others. |
| 19:41.15 | redotis | Some of the "leaders" of this room could definitely use a refresher course on being human. I would suggest seeing how the javascript room handles things. |
| 19:41.31 | Kobaz | see, i think you're just venting |
| 19:41.39 | SunTsu | There are no "rooms" on IRC |
| 19:41.44 | Kobaz | yeah |
| 19:41.46 | redotis | lol |
| 19:41.53 | redotis | Is this kiss up day today? |
| 19:41.58 | Kobaz | let's get on that chat room! |
| 19:42.02 | *** join/#asterisk nix8n82 (~AndChat27@24.sub-70-196-197.myvzw.com) |
| 19:42.04 | Kobaz | i don't need to kiss up to anyone |
| 19:42.10 | redotis | Sorry I don't know the terminology. |
| 19:42.13 | Kobaz | people know what i know and respect my experience |
| 19:42.14 | redotis | D-Fender is a dick. |
| 19:42.22 | elguero | nope, you are just out of line |
| 19:42.23 | redotis | plain and simple and he promotes dickishness |
| 19:42.45 | Kobaz | people might find it abrasive but he gets the job done |
| 19:42.51 | redotis | It's like an invasive virus. |
| 19:43.01 | redotis | meh..I've managed without his advice. |
| 19:43.07 | Kobaz | instaling update 6 of 9 |
| 19:43.09 | redotis | and actually when he did give advice...it sucked. |
| 19:43.15 | Kobaz | DO NOT POWER OFF YOUR MACHINE!@*!&@*(&!@!@ OMGPONIES |
| 19:43.19 | SunTsu | "Helpful, nice, for free, choose two" |
| 19:43.33 | redotis | he's not helpful...that's the thing. |
| 19:43.38 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.10) |
| 19:43.39 | Kobaz | redotis: do you have a technical problem that needs solving? |
| 19:43.49 | redotis | I figured it out |
| 19:43.50 | redotis | thanks |
| 19:44.01 | elguero | redotis: Are you here just to bash someone who volunteers his time to help people? |
| 19:44.10 | elguero | if so, please go elsewhere |
| 19:44.13 | SunTsu | redotis: I don't find your rambling over the last minutes not helpful in any way either |
| 19:44.16 | newtonr | redotis, if you can't find something more productive to talk about in here, please leave |
| 19:44.21 | Kobaz | if you collect the required data, and explain the problem thoroughly.. tk (and other people) can generally solve your problem |
| 19:44.44 | Kobaz | for example... if I have a complicated sip problem... I know tk can help |
| 19:44.54 | Kobaz | take it or leave it |
| 19:45.11 | redotis | Dude can't help without being a dick. |
| 19:45.13 | redotis | bottom line |
| 19:45.14 | redotis | so meh |
| 19:45.16 | redotis | ignore for me |
| 19:45.35 | Kobaz | your loss |
| 19:45.43 | Kobaz | okay let's move onto more productive things |
| 19:45.47 | *** join/#asterisk kresp0 (~kresp0@231.Red-2-136-86.dynamicIP.rima-tde.net) |
| 19:46.05 | redotis | Please do, it's like you guys are obsessed with your resource. |
| 19:46.07 | redotis | lmao |
| 19:46.35 | Kobaz | speaking of sip |
| 19:47.33 | Kobaz | having delay problems. need to wireshark some stuff |
| 19:47.34 | drmessano | sudo apt-get remove burr-under-saddle |
| 19:48.17 | Kobaz | and i think i need sub-second logging from asterisk verbose |
| 19:48.33 | [TK]D-Fender | Kobaz: http://ibot.rikers.org/%23asterisk/20131005.html.gz |
| 19:48.46 | [TK]D-Fender | Kobaz: 22:40.36 onwards |
| 19:48.50 | redotis | yum remove sycophants |
| 19:49.12 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 19:50.06 | Kobaz | heh |
| 19:50.08 | Kobaz | okay |
| 19:50.12 | drmessano | Thats a big word |
| 19:50.12 | Kobaz | that's what i figured |
| 19:50.20 | drmessano | Also, redhat user |
| 19:50.29 | drmessano | That explains a lot |
| 19:50.32 | Kobaz | redotis: i don't really want to get into a battle or anything, but looking at the chat log... that's not how you get help on irc |
| 19:51.03 | Kobaz | you had two separate people tell you how to get your answer |
| 19:51.05 | Kobaz | and you ignored it |
| 19:51.23 | redotis | Kobaz stop baiting. |
| 19:51.27 | drmessano | Kinda hard to build a case when the channel is logged and you're the one holding the brick |
| 19:51.38 | Kobaz | but anyway |
| 19:51.44 | drmessano | Theres a word for that |
| 19:51.58 | newtonr | redotis, if you don't like the #asterisk room or people in it, you are free to leave.. continue and it'll just end up with a ban |
| 19:52.04 | navaismo | slap drmessano |
| 19:52.11 | drmessano | slaps navaismo |
| 19:52.23 | redotis | Guys this dude's history goes way, way back and the real problem is the community doesn't stand up to him. The mods let his behavior continue because he is good. |
| 19:52.37 | Kobaz | your own history? |
| 19:52.42 | navaismo | "Also, redhat user That explains a lot" <---- XD |
| 19:52.43 | drmessano | You're here to save us? |
| 19:52.47 | redotis | You're looking at an hour's slice of time. |
| 19:53.16 | redotis | The question is do you want a decent room or one that treats people that asks questions like less-thans. |
| 19:53.26 | drmessano | We should be thanking you for pointing out the ways of [TK]D-Fender and exposing the truth? |
| 19:53.28 | [TK]D-Fender | Every can read the log right there. Nothing I've done in a long time has been anywhere near the levels claimed. |
| 19:53.53 | redotis | If you want D-Fender to be a human then, you expose his behavior. |
| 19:54.02 | [TK]D-Fender | And we can see the level of contempt an ungratefulness held right here today. |
| 19:54.04 | drmessano | THE CHANNEL IS LOGGED |
| 19:54.14 | drmessano | We don't need anyone exposing anything |
| 19:54.16 | Kobaz | hehe |
| 19:54.17 | drmessano | But thank you |
| 19:54.20 | redotis | Good, go back over the last year and you'll see what I mean. |
| 19:54.28 | redotis | There are countless examples. |
| 19:54.33 | drmessano | redotis, like we did with you and the log HE posted? |
| 19:54.35 | Kobaz | just wait, the truth will be revealed! |
| 19:54.41 | Kobaz | next murray pauvitch |
| 19:54.48 | redotis | That is fine. |
| 19:54.49 | drmessano | Because that log and todays rant makes you look like a giant douche |
| 19:55.16 | redotis | drmessano you're looking at an hours slice of time. |
| 19:55.25 | Kobaz | redotis: you're not going to win this |
| 19:55.30 | drmessano | Thats a pretty good hour |
| 19:55.33 | redotis | and neither are you |
| 19:55.36 | redotis | that's the sad part |
| 19:55.44 | drmessano | Actually, we already have won it |
| 19:55.50 | redotis | Kobaz and drmessano, you guys just like the drama. |
| 19:55.52 | drmessano | You just dont know it.. yet |
| 19:55.58 | redotis | Will redotis get kicked...that's what you're after. |
| 19:55.59 | drmessano | We LOVE the drama |
| 19:56.00 | redotis | so meh |
| 19:56.20 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 19:56.22 | Kobaz | update 9 of 9... weeee |
| 19:59.32 | Kobaz | so i guess rtp type 101 is dtmf? |
| 19:59.51 | Kobaz | never really got into the specifics |
| 20:01.01 | [TK]D-Fender | Kobaz: rfc2833 |
| 20:01.16 | Kobaz | [2013-10-17 16:00:04] Sent RTP P2P packet to 192.168.5.1:16988 (type 101, len 000004) |
| 20:01.40 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 20:09.15 | Kobaz | <PROTECTED> |
| 20:11.39 | Kobaz | pesky avaya was sending dtmf in-audio over sip |
| 20:11.40 | Kobaz | gah |
| 20:11.52 | Kobaz | er rather sip signalling in-audio over ulaw |
| 20:14.25 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 20:18.59 | *** join/#asterisk bkruse (~Adium@74.51.115.113) |
| 20:21.15 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 20:23.48 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 20:30.22 | [TK]D-Fender | And ... it's checkout time.. heading home... |
| 21:01.39 | *** join/#asterisk eirirs (eirirs@dalvik.ping.uio.no) |
| 21:03.39 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 21:04.57 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:05.38 | Kobaz | hmm |
| 21:05.46 | Kobaz | i think i need to add debugging for when rtp starts flowing |
| 21:11.12 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 21:22.34 | *** join/#asterisk jsjc (~Adium@225.Red-88-12-12.staticIP.rima-tde.net) |
| 21:24.08 | saxa | anonymouz666: ok, but the price i paid for TE133 with echocanc was the same as without it for TE122, so basically i choose this option more. |
| 21:24.23 | saxa | anonymouz666: thanks anyway for your help, and WIMPy too. |
| 21:49.39 | *** join/#asterisk danjenkins (~danjenkin@cpc3-folk2-2-0-cust34.1-2.cable.virginmedia.com) |
| 22:00.24 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.36) |
| 22:04.44 | *** join/#asterisk danjenkins (~danjenkin@cpc3-folk2-2-0-cust34.1-2.cable.virginmedia.com) |
| 22:07.36 | *** join/#asterisk felipealmeida (~user@186.212.182.57) |
| 22:17.05 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 22:26.56 | *** join/#asterisk vandyk (~vandyk@177.18.86.162) |
| 22:27.21 | vandyk | hi guys, someone here are facing issues with SIP attack? |
| 22:28.00 | *** join/#asterisk navaismo (~navaismo@189.191.237.126) |
| 22:28.07 | Chainsaw | vandyk: Why, can we pay you to stop or something? |
| 22:29.05 | vandyk | no, because my servers has been under attack since last week |
| 22:29.11 | vandyk | at least there are tentatives |
| 22:29.18 | vandyk | but as I'm using fail2ban |
| 22:29.42 | vandyk | it does not impact $$$ |
| 22:30.56 | vandyk | but I'm just curious if is someone else has been this |
| 22:31.42 | WIMPy | Not for mor than some minutes. Do you give them hints? |
| 22:32.20 | vandyk | no |
| 22:32.32 | WIMPy | Are you sure? |
| 22:32.36 | vandyk | sure |
| 22:33.00 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.202) |
| 22:33.01 | vandyk | I think that they tried to find on DNS all hosts that has voip.domain |
| 22:33.06 | WIMPy | Then ignore it. |
| 22:33.25 | WIMPy | They try every IP. |
| 22:33.41 | vandyk | its a script, because try to register a lot of extensions |
| 22:33.46 | vandyk | in sequence |
| 22:34.06 | WIMPy | sure |
| 22:36.06 | vandyk | fail2ban is doing a great job, probably I'll increase the amount of minutes to block the IPs |
| 22:37.00 | vandyk | or better, ban forever the IP |
| 22:38.13 | *** join/#asterisk saint_ (~saint@c-50-166-85-78.hsd1.nj.comcast.net) |
| 22:38.14 | saint_ | hi all |
| 22:40.52 | saint_ | I have a question about a dialplan... I have an "auto attendant" menu ( http://pastebin.com/k73zcnsU ) . |
| 22:41.03 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-tjlejqbxyxtzmfzr) |
| 22:41.15 | saint_ | If users press 1 for the first choice, there is a delay of 2 to 3 seconds before the audio is played |
| 22:41.30 | saint_ | if I replace the exten => 1 with exten => 9 for example , it plays right away |
| 22:41.45 | saint_ | is there any reason what choice "1" is delayed into playing Background files ? |
| 22:42.22 | saint_ | it's only 1 . if I replace 1 by a non used number, then it plays right away. |
| 22:47.02 | [TK]D-Fender | saint_: "dialplan show autoattendant" |
| 22:47.11 | [TK]D-Fender | saint_: and the call attempt please... |
| 22:47.33 | [TK]D-Fender | saint_: Aslo, what is the point of Wait() between your backgrounds? |
| 22:48.43 | saint_ | [TK]D-Fender: this magical command showed me that I had extension with 11 available also . that's why. i m going to get ride of that. |
| 22:49.21 | saint_ | [TK]D-Fender: thanks for this trick. regarding the wait() , without it the audio files play once after the other too quick, and since it's TTS , it does not sound natural without a second between each choice. |
| 22:50.19 | [TK]D-Fender | saint_: so a chain of TTS recordings? |
| 22:50.19 | [TK]D-Fender | (pre-made) |
| 22:51.01 | [TK]D-Fender | saint_: Better : same => n,Background(customized/200&silence/1) |
| 22:51.08 | saint_ | [TK]D-Fender: yes, because each recording changes once in a while depending on what we have coming. so I can comment recording 200 for example . |
| 22:51.30 | saint_ | [TK]D-Fender: what's this silence/1 ? |
| 22:51.39 | [TK]D-Fender | saint_: 1 second of silence |
| 22:52.22 | saint_ | [TK]D-Fender: i understand , but it's not in the options. is reads a file in the silence folder ? |
| 22:52.26 | [TK]D-Fender | saint_: same => n,Set(COUNTDOWN=${INC(COUNTDOWN)}) <- IRONY |
| 22:52.28 | [TK]D-Fender | saint_: incrementing .... a countDOWN |
| 22:52.39 | saint_ | lol |
| 22:52.41 | [TK]D-Fender | saint_: Yes, those are stock sounds |
| 22:53.04 | saint_ | damn it |
| 22:53.11 | saint_ | never paid attention to this audio folder. thanks for the trick |
| 22:53.59 | saint_ | ok. I changed the code, and I'm using "count" instead of "countdown" , you are right. |
| 22:55.59 | *** join/#asterisk danjenkins_ (~danjenkin@cpc3-folk2-2-0-cust34.1-2.cable.virginmedia.com) |
| 22:56.00 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 22:56.03 | *** part/#asterisk danjenkins_ (~danjenkin@cpc3-folk2-2-0-cust34.1-2.cable.virginmedia.com) |
| 22:57.21 | saint_ | [TK]D-Fender: works like a charm. thanks for the 2nd pair of eyes |
| 22:58.21 | [TK]D-Fender | saint_: You're welcome |
| 22:58.46 | saint_ | Is there a web based interface for faxes on asterisk ? Whether it's a 3rd party module or not ? |
| 23:00.22 | [TK]D-Fender | everything "web" is 3rd party |
| 23:00.43 | [TK]D-Fender | * is a tel;ephony toolkit. The idea of specifically doing anything at all is your job to invent in the dialplan, etc |
| 23:01.31 | saint_ | ok. i m getting back into php . i was thinking about doing an open source if it does not already exists in order to be able to send and receive faxes through a web interface |
| 23:01.43 | saint_ | that is the issue that have been dealing with lately with a couple of customers |
| 23:02.25 | saint_ | i ll see what i come up with. if it is worth it , i ll throw it out there. thanks for the help once again. |
| 23:03.49 | [TK]D-Fender | saint_: There are a few ready made OSS ones out there.... easily googl-able and a few list on the old wiki |
| 23:03.54 | [TK]D-Fender | ~wikis |
| 23:03.54 | infobot | somebody said wikis was VoIP Wiki covering FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners. For Asterisk, see ~asteriskwiki |
| 23:04.12 | saint_ | ok, i ll check it out. thanks |
| 23:04.23 | [TK]D-Fender | saint_: Is web your direct prefrence or jsut the first interface idea you came to? |
| 23:04.55 | saint_ | [TK]D-Fender: preference because it can work out of a smart phone , or computer without paying attention to the OS |
| 23:05.43 | [TK]D-Fender | saint_: On a minimalistic scale this would be easy to make. |
| 23:05.50 | [TK]D-Fender | saint_: But go look around to see if one that is "close enough" is out there already |
| 23:05.59 | saint_ | will do. |
| 23:10.12 | [TK]D-Fender | BRB, router upgrade.'.. |
| 23:18.49 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:24.05 | *** join/#asterisk danjenkins (~danjenkin@146.185.158.234) |
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| 23:29.56 | *** part/#asterisk danjenkins (~danjenkin@146.185.158.234) |
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| 23:38.10 | *** join/#asterisk jetlag (~jetlag@pool-71-168-241-76.cmdnnj.east.verizon.net) |
| 23:56.13 | *** join/#asterisk martinfletcher_ (~martinfle@87.237.70.109) |
| 23:56.28 | *** part/#asterisk j4m3s_ (~j4m3s@pdpc/supporter/active/j4m3s) |