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| 00:05.14 | navaismo | ?itsp |
| 00:05.20 | navaismo | ~itsp |
| 00:05.20 | 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. |
| 00:05.29 | navaismo | ~itsplist-us |
| 00:05.29 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
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| 06:35.17 | fling | how to change sip peer input and output volume? |
| 06:40.03 | *** part/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 06:45.41 | ChannelZ | there's the VOLUME() function but that means * has to be processing the audio streams |
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| 07:26.55 | fling | changed sound volume on pata, now it is loud :] |
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| 08:18.45 | eirirs | Qwell: |
| 08:18.46 | eirirs | 091501 < *** > I tried to commit suicide so I took a bunch of pills but I was upset when the pills I thought said "die" actually said "diet" |
| 08:18.49 | eirirs | 091507 < *** > lost 5 kg though |
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| 08:45.53 | zenmaster | Hi guys. :) I am trying to setup failover with two of my sip providers, and am having a issue. :0 |
| 08:50.33 | ChannelZ | "how do I tell if one is dead" ? |
| 08:51.49 | bulkorok | hi |
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| 09:06.02 | zenmaster | quit |
| 09:07.30 | Cutzmf | Hi! Have problem with FFA (licensed). Actualy tiff file is fully recieved, but transmission ends up with "FAILED HANGUP" |
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| 09:16.10 | lorsungcu | fuck |
| 09:16.12 | lorsungcu | it is late |
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| 09:33.25 | Cutzmf | Need help to review fax debug messages |
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| 09:38.42 | wdoekes | Cutzmf: I've always assumed an available tiff to mean that the fax succeeded |
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| 09:43.29 | Cutzmf | wdoekes: that's great (really), but how you identify fully transmitted tiff with partial? |
| 09:44.09 | Hrnec | Hi. Is it possible to allow RTP and SRTP from the same peer (proxy) at the same time? If yes, how? Thanks a lot. |
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| 09:45.39 | wdoekes | wait, I lied. I hangup, and from the hangup handler I check ${FAXSTATUS} for SUCCESS |
| 09:46.58 | wdoekes | you're probably getting a different faxstatus |
| 09:47.29 | wdoekes | or? (and you're using FFA, I'm using spandsp) |
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| 09:51.15 | Hrnec | Let me ask a better question. Is it possible to allow RTP and SRTP at the same time for calls when signaling comes from the same proxy (defined as a peer in sip.conf) ? Thanks. |
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| 09:57.06 | wdoekes | Hrnec: I haven't done any SRTP, but I have matched different users from the same proxy. so if it's not possible, you could make two users, an rtp and and srtp user |
| 09:58.22 | wdoekes | I don't know how you do your auth/user matching. but altering the From in the proxy should be sufficient to have asterisk match a certain user |
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| 11:38.29 | madhatt3r | helloAll |
| 11:38.43 | madhatt3r | anybody here dealt with cisco SPA501G phones? |
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| 11:40.14 | R1ck | when I Playback a file, sometimes I can't here the first 200ms of the file or something.. a Wait(4) before it doesnt fix it.. what could be wrong? the file itself is fine (gsm format) |
| 11:41.17 | wdoekes | R1ck: do you Answer() before playing? |
| 11:42.31 | R1ck | no, should I? :) |
| 11:44.40 | R1ck | hmm, now the Wait does work.. but still the same problem.. gonna double-check the file |
| 11:47.32 | kaldemar | Playback(silence/1) before the actual file may help. |
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| 11:51.24 | R1ck | the file itself has close to a second pause at the start |
| 11:55.20 | R1ck | meh, its the file :) |
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| 13:15.57 | Greenlight | I've got to the bottom of what's been causing the issue I was discussing yesterday. There appears to be a bug, in that if jitterbuffer=yes is set in ConfBridge.conf, then if channels are subsequently Bridged directly, then this causes one way audio on the bridged channels. |
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| 13:22.18 | WIMPy | Or just one of them? |
| 13:22.32 | WIMPy | Nice to hear you found it. |
| 13:23.08 | Greenlight | I know :) I was so glad when I found what caused it, and at least a temp solution, been driving me nuts for days. |
| 13:23.39 | Greenlight | I was hoping that one of the folk who know that part of the code would cast their eye over it, before I raise an issue on the JIRA |
| 13:24.29 | Greenlight | It's odd that it's one way audio tho, yea. I'd have though that it would apply the jitterbuffer to all channels in the ConfBridge. Guess the problem may lie inside the bridging code |
| 13:24.35 | WIMPy | I think opening an issue may make it more likely. |
| 13:25.32 | Ice_Strike | Hi WIMPy! |
| 13:25.52 | WIMPy | I'm currently waiting for an devstate/hint failure. Have tshark running since yesterday. Trouble is that it takes at least a day to show up. |
| 13:25.57 | WIMPy | hi Ice_Strike |
| 13:26.07 | Ice_Strike | You cool? |
| 13:26.12 | Greenlight | Those are the worst sort to track down |
| 13:26.57 | WIMPy | Just hoping there will bin anything interesting when it happens again. |
| 13:27.35 | WIMPy | And BTW: wireshark segfaults when it tries to list my interfaces... |
| 13:27.37 | Ice_Strike | WIMPy Have you ever created your own predictive dailer? |
| 13:27.51 | Ice_Strike | dialer* |
| 13:28.58 | Chainsaw | Greenlight: In fairness though, you are the king of intermittent faults. |
| 13:29.00 | *** join/#asterisk jkroon (~jkroon@dsl-244-37-92.telkomadsl.co.za) |
| 13:29.08 | Chainsaw | Afternoon jkroon. |
| 13:29.40 | Greenlight | Chainsaw: Well, yea, I guess I have the trophy for that one :) |
| 13:29.50 | WIMPy | Ice_Strike: No, I've just done some experiments towards it some years ago. |
| 13:30.18 | jkroon | hey Chainsaw Greenlight |
| 13:30.30 | Greenlight | Heya jkroon, hope you're well |
| 13:30.35 | WIMPy | Hi jkroon |
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| 13:30.44 | Greenlight | Ice_Strike: I have |
| 13:31.18 | jkroon | Greenlight, very busy but very well |
| 13:31.27 | Greenlight | Heh, I know that feeling :) |
| 13:31.38 | Greenlight | Better busy than not though, eh |
| 13:32.24 | jkroon | indeed |
| 13:33.03 | jkroon | and for once i can concentrate somewhat on non-voip stuff after ast 11. |
| 13:33.21 | jkroon | Greenlight, just wanted to double-check that all your concerns have been addressed and fixed? |
| 13:33.39 | Ice_Strike | Well I am going to create my own predictive dailer for learning purpose. What the best method to send commands from a browser to background process that run AMI. For example if agents click on Hang Up button a browsers.. |
| 13:33.47 | WIMPy | seems to remember that jkroon had some interesting issue last time, but can't remember what it was. |
| 13:34.00 | *** join/#asterisk deo (~deo@112.198.79.121) |
| 13:34.18 | jkroon | Ice_Strike, use ami directly from your server-side code that serves the browser |
| 13:34.26 | jkroon | :p |
| 13:34.28 | alami | hello i have asterisk 11.0, and when i run rasterisk i get Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 13:34.29 | jkroon | WIMPy, ISDN |
| 13:34.57 | WIMPy | jkroon: Was that the one NT is other than the others one? |
| 13:35.08 | Ice_Strike | jkroon I don't get ya. |
| 13:35.14 | Greenlight | jkroon: I'm almost finished the re-write of my code, which I'm hoping will alleviate the problems. Will almost entirely avoid Queues, Local channels, and ConfBridges. Servers are still just about stable for a day, so stil got breathing space from customers. I think the problem was the exterme number of context switches and new threads being spawned |
| 13:35.32 | Ice_Strike | Greenlight How did you do it |
| 13:35.43 | jkroon | WIMPy, yes. |
| 13:36.05 | jkroon | Ice_Strike, I use curl from php to connect to ami over http. |
| 13:36.11 | jkroon | works quite well |
| 13:36.14 | WIMPy | jkroon: And was it the NT? |
| 13:36.30 | jkroon | WIMPy, client cancelled, digium dragged a little too long looking at the issue. |
| 13:36.55 | WIMPy | bad |
| 13:37.03 | jkroon | Greenlight, sounds like you've got things under control then. |
| 13:37.07 | Greenlight | Ice_Strike: You'll need some sort of central service to speak to the AMI on behalf of the agents. |
| 13:37.15 | Greenlight | jkroon: I'm hopeful yes :) |
| 13:37.23 | Ice_Strike | Greenlight How did you do it? |
| 13:37.30 | jkroon | hopeful is good. |
| 13:37.33 | Greenlight | Ice_Strike: Blood, sweat and tears :) |
| 13:37.38 | Ice_Strike | lol |
| 13:37.39 | jkroon | rofl |
| 13:38.17 | WIMPy | Yes, don't let the frontend be the application. That will most probably hit you hard later on. |
| 13:38.27 | Greenlight | jkroon: Plan B is still to recruit help from yourself and/or Chainsaw and stick Gentoo on though. |
| 13:38.49 | jkroon | wonders how serious a difference that will really make. |
| 13:39.06 | Greenlight | Yea, that's why I'm hoping plan A works |
| 13:39.19 | jkroon | currently doing around 25k call setups/day and ast 11 works quite well, but you do really strange things |
| 13:39.23 | Greenlight | I reckon I was spawning around 50-100 new threads per second, under load |
| 13:39.43 | Greenlight | And 50k-100k context switches |
| 13:39.54 | Greenlight | I think that was the killer |
| 13:40.14 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:40.14 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:40.14 | Greenlight | And maybe even the source of the very very strange readings in "top" |
| 13:40.37 | jkroon | yea, the 50-100 new threads shouldn't take a box down, but if there is some resource leak associated with that spawning ... |
| 13:41.02 | jkroon | the load average spikes? but i did email you the clarification of how that calculation works and why it can easily spike like that |
| 13:41.28 | Greenlight | Yea indeed |
| 13:41.34 | Ice_Strike | Greenlight I was thinking something like this: Browser <--> Background Process (run in a loop,AMI Socket, Update Databases, etc ) <--> Asterisk Server |
| 13:41.53 | Ice_Strike | Send the data to a Background Process via ajax with XML or Json format |
| 13:42.00 | Greenlight | Ice_Strike: That's pretty much how we do it, yes. We have a windows service that goes inbetween |
| 13:42.08 | Ice_Strike | I dont know if there other better solution |
| 13:42.26 | WIMPy | Sounds like a good solution to me. |
| 13:42.38 | jkroon | the load average is *sampled*, so if it's sampled every 5 seconds, and it happens to sample it just after 3000 odd threads waiting on a mutex got woken up, that's 3000 tasks in the run queue, so even though they'll almost all go back to sleep within a millisecond or two they still end up chasing your load average up by a HUGE amount |
| 13:42.52 | alami | rasterisk |
| 13:42.55 | WIMPy | did it the same, jut not with a browser, but a dedicated app. |
| 13:43.28 | Ice_Strike | Do you send data to Background Process or you write the commands file in a dir and then a process keep checking the files in the dir to execute any if nessary. |
| 13:43.42 | Greenlight | Yea, but for a box doing realtime media, that could definetly be a problem. So am hoping it's all the threads and context switches that were killing it |
| 13:43.46 | Greenlight | *hoping* |
| 13:43.48 | Greenlight | praying |
| 13:43.52 | alami | i have installed asterisk but i still can't start it |
| 13:43.56 | alami | can any one help plz? |
| 13:44.12 | WIMPy | Ice_Strike: What files? |
| 13:44.21 | jkroon | Greenlight, without major redesign i fail to see how you intend to get rid of those :) |
| 13:44.46 | Greenlight | jkroon: From what I can see each Local channel runes in it's own thread |
| 13:44.57 | Ice_Strike | WIMPy like command files, in each file have like AgentID, HangUP |
| 13:45.06 | jkroon | Greenlight, that's scary, but perfectly possible. |
| 13:45.08 | Greenlight | Ice_Strike: DOn't write to files, use a database |
| 13:45.09 | Ice_Strike | a process will check if the files exist to do the action |
| 13:45.15 | alami | jkroon: can you help plz? |
| 13:45.21 | jkroon | alami, depends |
| 13:45.27 | WIMPy | Ice_Strike: To communicate from the webserver to your application? |
| 13:45.31 | alami | i have installed asterisk but i still can't start it |
| 13:46.00 | Greenlight | jkroon, Yea, like in the old code, each call placed, could have up to 3 or more Local channel pairs, as well as it's SIP channel |
| 13:46.10 | WIMPy | has been doing the file based communication, but it's definitely not the cool way to do it. |
| 13:46.11 | Ice_Strike | WIMPy No - Browser to a process background. |
| 13:46.12 | Greenlight | ANd I'm dialling 100 new calls a second sometimes |
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| 13:46.13 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:46.18 | alami | jkroon: when i type asterisk -V i get Asterisk 11.3.0-rc1 |
| 13:46.38 | jkroon | alami, what's the error you get when trying to start it? |
| 13:46.43 | WIMPy | Ice_Strike: How do you come to files there? |
| 13:46.45 | jkroon | asterisk -V will print version and exit ... |
| 13:46.51 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.19.210) |
| 13:47.11 | Ice_Strike | Or other solution is if I click on Hang Up Call button on the browser - it update to the database.. then a process keep checking any data on MySQL database. |
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| 13:47.23 | alami | jkroon: Illegal instruction |
| 13:47.39 | jkroon | alami, compile asterisk for your local machine, not something you don't have. |
| 13:47.54 | Greenlight | Ice_Strike: You will want to use ajax or similar to talk to the background service |
| 13:47.56 | WIMPy | alami: How did you install it? |
| 13:48.14 | Ice_Strike | Greenlight yea using ajax |
| 13:48.19 | alami | WIMPy:from the source |
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| 13:48.35 | Greenlight | Ice_Strike: You could even have it call a php page, which opens a socket connection to your background process |
| 13:49.03 | alami | WIMPy: why? |
| 13:49.10 | WIMPy | Sockes are the most elegant and easy thing. |
| 13:49.17 | WIMPy | alami: Fix your OS. |
| 13:49.41 | alami | WIMPy:how? |
| 13:50.27 | Ice_Strike | Greenlight Yea I could do... I could make a process to become mini web server.. so I send send data via XML or JSON |
| 13:52.25 | mjordan | alami: you most likely are building on a VM which has a virtual CPU architecture. Disable BUILD_NATIVE in menuselect and re-compile. If that doesn't work, you'll have to pass the CPU architecture type to Asterisk's build system explicitly. |
| 13:52.40 | mjordan | if you aren't building on a VM, however, disregard |
| 13:53.19 | Ice_Strike | Greenlight What did you do to recieve AMI event? |
| 13:53.36 | Ice_Strike | It update to database directly and browser read from database? |
| 13:53.42 | Greenlight | I have my service listen to and process the events |
| 13:53.54 | Greenlight | The service tracks calls, channels etc |
| 13:54.27 | WIMPy | Using browsers for interactive stuff is so horrible. |
| 13:54.28 | Ice_Strike | Yep, that is a process job. |
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| 13:58.07 | alami | @mjordan: it's not good to compile asterisk on VM? |
| 13:58.33 | Ice_Strike | Greenlight What I meant if a customer hang up a call, then a browser need to say customer have hang up a call.. What is the solution to this? I think when a process detected a hang up call via AMI event - then update to mysql database. Ajax check the database every 5 seconds? |
| 13:58.41 | Ice_Strike | or what the better solution? |
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| 13:59.59 | mjordan | alami: no, it's fine to compile Asterisk on a VM. We do it all the time. Some, however, have virtual CPU architectures that aren't interpreted by the default compiler settings. When that occurs, you have to specify the architecture you're compiling for. |
| 14:00.13 | Greenlight | Ice_Strike: It's not going to be quite as straightforward as you're making out. You need to track the channel(s) associated with the "call" and issue a hangup |
| 14:00.54 | Ice_Strike | Yes I know that |
| 14:01.40 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 14:01.44 | Ice_Strike | I meant how do you make a browser to detect a hangup event - from a database directly ... or check through socket to a process background |
| 14:01.47 | Greenlight | So, what you probably want to do is track the agent to the call on the service, and then the agent can hangup and the service will translate that request to issue an ami action |
| 14:02.04 | Greenlight | Ice_Strike: Oh, I see. |
| 14:02.14 | Greenlight | Ice_Strike: You could poll the service |
| 14:03.07 | Ice_Strike | poll the service? |
| 14:03.29 | Greenlight | javascript loop on browser polling for any new "events" |
| 14:03.50 | Greenlight | Or, if you wanna be really cool, use SignalR. |
| 14:04.09 | Greenlight | But that may tie you into .NET, which you might not want |
| 14:04.29 | Ice_Strike | javascript loop on browser every second |
| 14:04.33 | Ice_Strike | Ewww :P |
| 14:04.41 | Greenlight | Ice_Strike: Eww indeed - that's why SignalR kicks ass |
| 14:04.47 | WIMPy | Ice_Strike: Do you need it to run on a browser? |
| 14:05.23 | [TK]D-Fender | Java script connected to server. Push the message direct |
| 14:05.38 | Ice_Strike | WIMPy because I am most famililar with php and mysql |
| 14:05.45 | WIMPy | Liefe is so much easier without the browser crap. |
| 14:05.46 | [TK]D-Fender | trigger on "h", no need to "poll" anything |
| 14:06.25 | Ice_Strike | WIMPy I get ya. |
| 14:07.19 | Ice_Strike | WIMPy But browser is still good if you wanna update something quickly.. for an application you have to compile it and update all the clients PC's |
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| 14:07.54 | WIMPy | Yes, but it will make everyting so much easier and quicker. |
| 14:08.42 | WIMPy | Using a browser is really only interesting to support users on OSs you don't want to support. |
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| 14:12.48 | Ice_Strike | Yep |
| 14:13.02 | alami | mjordan:can you tell me wich file on /usr/src/asterisk-11.3.0-rc1/menuselect i will find BUILD_NATIVE |
| 14:13.13 | Ice_Strike | [TK]D-Fender what us "h" ? |
| 14:13.15 | Ice_Strike | is* |
| 14:14.31 | [TK]D-Fender | ... |
| 14:14.38 | [TK]D-Fender | Asterisk Standard Extensions <- |
| 14:14.41 | [TK]D-Fender | Dialplan basics... |
| 14:14.45 | [TK]D-Fender | ~book |
| 14:14.45 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:14.46 | mjordan | alami: run menuselect. "make menuselect" |
| 14:14.57 | mjordan | alami: you'll then find it under the section build options |
| 14:16.59 | alami | @mjordan: aha okay thanks |
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| 14:23.52 | qakhan | i m getting this error when i try to install DAHDI |
| 14:23.53 | qakhan | http://pastebin.com/VerC5rw1 |
| 14:24.19 | Greenlight | You have kernel-devel packages installed ? |
| 14:25.03 | WIMPy | Any reason you use an old version of dahdi? |
| 14:25.23 | [TK]D-Fender | Asterisk 1.4 <- |
| 14:25.33 | [TK]D-Fender | And no, he's never getting off of it |
| 14:25.58 | WIMPy | I don;t see any hope then anyway. |
| 14:26.34 | WIMPy | But even Asterisk 1.4 should work with the latest version of dahdi, shouldn't it? |
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| 14:28.43 | qakhan | [TK]D-Fender my friend listen |
| 14:29.38 | qakhan | my dev team has developed an app which is integrated with asterisk and using Manager |
| 14:30.22 | *** join/#asterisk rgsteele (~rgsteele@12.150.6.65) |
| 14:30.23 | qakhan | we are using asterisk 1.4.38 because of Agentcallbacklogin |
| 14:31.14 | [TK]D-Fender | qakhan, It is dead. |
| 14:31.20 | [TK]D-Fender | The entire branch is dead |
| 14:31.32 | [TK]D-Fender | The next THREE branches that came after it are dead |
| 14:31.32 | qakhan | i know |
| 14:31.43 | [TK]D-Fender | You do not have to use that approach to do the job. |
| 14:31.56 | qakhan | yes you are right |
| 14:32.01 | [TK]D-Fender | you are hung up on something that doesn't have be done that way. |
| 14:32.16 | [TK]D-Fender | And your explanation is an empty excuse. |
| 14:32.21 | qakhan | but i have to do it for time being |
| 14:32.25 | [TK]D-Fender | You are running yourself int a very dead end |
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| 14:32.56 | qakhan | i got you |
| 14:33.32 | qakhan | but is there any way to do it for now |
| 14:33.33 | [TK]D-Fender | Yes, and I know that in half a year you'll still be fishing for support for this unchanged |
| 14:34.13 | qakhan | man i want to move forward but my dev team dont want |
| 14:34.37 | qakhan | i am depending on them |
| 14:34.48 | WIMPy | Don't waste your time trying to push the dead end half a meter away. Make your way over to the highway. |
| 14:34.59 | Greenlight | Get a new dev team |
| 14:35.04 | qakhan | hahaha |
| 14:35.05 | WIMPy | Tell them what we tell you. |
| 14:35.17 | qakhan | ok i will tell them |
| 14:35.24 | WIMPy | Or what Greenlight said. Seems to be neccessary. |
| 14:35.55 | Greenlight | Maybe if one of them has an "accident" the others will change their minds... |
| 14:36.10 | [TK]D-Fender | and we'll STILL see you here in a half a year in the same position |
| 14:37.16 | qakhan | but plz help me for today plzzzzzzzzz |
| 14:43.19 | alami | mjordan: thanks a lot it work ;) |
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| 14:44.11 | *** join/#asterisk WindBack (~quassel@190.123.122.18) |
| 14:44.20 | WindBack | Hi |
| 14:44.44 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 14:45.20 | FlashDeluxe | hi! i got a question: I would like to forward a call to another number if the line is busy, how do i do that? |
| 14:45.41 | WindBack | Simple Question: Is insecure to have nat=no in general section and rewrite some peers with nat=yes?? This configuration can make those peers discoverables? |
| 14:46.33 | WindBack | [TK]D-Fender: Simple Question: Is insecure to have nat=no in general section and rewrite some peers with nat=yes?? This configuration can make those peers discoverables? |
| 14:46.45 | [TK]D-Fender | WindBack, Don't single people out like that.... |
| 14:46.50 | Greenlight | I raised a JIRA for the bug I found: https://issues.asterisk.org/jira/browse/ASTERISK-21144 |
| 14:46.51 | [TK]D-Fender | WindBack, Those wou'll answer will answer |
| 14:46.57 | [TK]D-Fender | who'll* |
| 14:47.39 | WindBack | [TK]D-Fender: ok, sorry. |
| 14:48.21 | qakhan | [TK]D-Fender any answer? |
| 14:48.38 | [TK]D-Fender | qakhan, UPGRADE |
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| 14:59.06 | Ice_Strike | Which is better solution? http://s18.postimage.org/kve0yjtax/background_process.png |
| 14:59.34 | Ice_Strike | To communicate - browser between processd background |
| 15:00.17 | igcewieling1 | that is like asking "when did you stop beating your wife." all answers are wrong. |
| 15:00.59 | igcewieling1 | when the user clicks hangup, make an ajax call to a script which connects to AMI and hangs up the channel. |
| 15:02.09 | Greenlight | Ice_Strike: Don't do database polling like in the first picture |
| 15:02.36 | Ice_Strike | Ok cool Greenlight |
| 15:02.40 | Greenlight | Imagine database running slow, and agents can't hangup on customers! |
| 15:02.47 | Ice_Strike | Good point |
| 15:02.53 | Greenlight | That bit needs to be as direct as possible |
| 15:03.21 | Ice_Strike | What about Hang Up message on a browser when hangup has been completed via AMI |
| 15:03.37 | Ice_Strike | Send data back to browser from process background |
| 15:03.45 | Ice_Strike | or poll from database via ajax? |
| 15:03.51 | Greenlight | Either javascript polling, or SignalR as I suggested. Or a socket connection from the browser to the service |
| 15:04.04 | Ice_Strike | javascript polling from service again |
| 15:05.15 | Ice_Strike | Greenlight Why can't use database polling for receving data? |
| 15:05.24 | Greenlight | You can, I just wouldn't recommend it |
| 15:06.09 | Greenlight | However, if you'r just learning, then it's not a huge problem, so yea, you can do it that way. |
| 15:06.31 | Greenlight | By far your biggest challenge will be tracking everything that's happening on asterisk and keeping your own records up to date |
| 15:06.43 | Ice_Strike | Yep |
| 15:07.30 | igcewieling1 | leave the process running, pass back javascript to do something in the browser when the hangup completes. |
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| 15:10.45 | igcewieling1 | we have an internal GUI for our core asterisk switch. When you save something to the user it appears immediate (you go back to the previous screen), but an ajax call happens in the background which passes the form to a scrip on the server which saves the data then spits back to the browser something like <script>alert('page saved');</script>\n which confirms to the user that it actually saved. The popup is more friendly than an ale |
| 15:11.53 | Greenlight | How do you keep a connection open from the browser to the background service? Polling, or long-running connection ? |
| 15:12.18 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 15:13.22 | Ice_Strike | Greenlight My plan are .. Send the data from Ajax to a process service using curl.. A browser recieve update information from database. |
| 15:13.40 | Ice_Strike | You recommend javascript polling from service but how you make sure you recieved right information? |
| 15:15.16 | Greenlight | BUt how do you make the browser update, when something changes in the service. At our side we use SignalR, but without that I would imagine polling to be the way |
| 15:15.53 | Greenlight | Or, does your GUI/browser only update when the user does/clicks something ? |
| 15:17.59 | Ice_Strike | Talking to me? |
| 15:18.22 | Greenlight | Ice_Strike: What do you mean "right information" ? You'd just have a queue of messages, and each time it polled it would pick them up, and deal with them client side |
| 15:19.03 | Greenlight | So, in your background process if it detected that an agents call had hungup, it would add "CallHungup" message to the queue for that agent. The next time that agents browser polled, it would receive that message. |
| 15:19.35 | Ice_Strike | Greenlight What I meant if there are mutitple agents on calls and many hangup at the same time.. you would want to recieve right information from spefic call. |
| 15:19.48 | Ice_Strike | Ahhh I get ya |
| 15:20.10 | Greenlight | Ice_Strike: Your background service would need to decided to which agent each channel and call are associated. |
| 15:20.14 | Greenlight | *decide |
| 15:21.03 | Ice_Strike | Do you know any opensource script that does the samilar thing that we are talking about. |
| 15:21.16 | Ice_Strike | and background process opensource script. |
| 15:21.42 | Greenlight | Not sure, I wrote my stuff from ground up. I beleive there are libraries out there which claim to deal with some of it, but not sure of names etc |
| 15:21.43 | qakhan | [TK]D-Fender i resolved the problem |
| 15:21.52 | Greenlight | You upgraded ? |
| 15:22.10 | [TK]D-Fender | No, he EXTENDED the problem |
| 15:22.14 | Greenlight | Oh |
| 15:22.53 | qakhan | no i used other dahdi version |
| 15:23.18 | Greenlight | Like, [TK]D-Fender said, you've extended your issues |
| 15:23.24 | Greenlight | Well, until next time :) |
| 15:24.06 | qakhan | OK let see |
| 15:24.24 | qakhan | but i will tell my dev team to use new version |
| 15:24.45 | Greenlight | mjordan: That's the component I was looking for in the list, didn't see "Core/Bridging" there. Thanks :) |
| 15:24.59 | qakhan | [TK]D-Fender which version you recommand |
| 15:25.18 | [TK]D-Fender | 11 |
| 15:27.10 | mjordan | Greenlight: np. newtonr and I were just discussing the problem a bit - jitter buffers sometimes log things when they get unhappy, but through a weird implementation quirk it is usually only seen if you enable iax2 debugging |
| 15:27.30 | mjordan | do you think you could run a test to get a DEBUG log and see what your jitter buffer spits out? |
| 15:28.06 | Greenlight | Sure |
| 15:28.25 | Greenlight | What's very odd is that it's only 1 channel, even though both have been in the confbridge |
| 15:28.29 | mjordan | it may spit out nothing, but this feels like the jitter buffer getting confused and dropping frames |
| 15:28.30 | mjordan | hm. |
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| 15:29.03 | Greenlight | It's either the Channel1 or Channel2 (can't recall off hand) in the Bridge, that doesn't have any rtp come from it |
| 15:29.07 | mjordan | This is one of those edge cases where things get funky. |
| 15:29.15 | Greenlight | Gotta love em |
| 15:29.24 | mjordan | I'd be curious what would happen if you redirected the channels into an infinite Wait() and then Bridged them |
| 15:29.49 | mjordan | but I can say I've never tried using Bridge on channels already in a bridge, so this is new ground :-) |
| 15:30.09 | Greenlight | The weird thing is that i've another service that bridges millions of channels a day (literally!) from a ConfBridge without problem, so I was looking in totally the wrong place. Turned out just happened they had jitterbuffer disabled |
| 15:30.20 | Greenlight | mjordan: I've tried that |
| 15:30.23 | mjordan | hm |
| 15:30.23 | Greenlight | ALl works ok |
| 15:30.33 | Greenlight | It's only when ConfBridge with jitterbuffer=yes is used |
| 15:31.03 | Greenlight | Basically, the reason for doing it is that we conference in a 3rd party. Once they are no longer needed, it's best to bridge the channels direct again |
| 15:31.30 | file | hum so it uses func_jitterbuffer to put a jitterbuffer on the channel... hrm |
| 15:31.57 | file | I think the timestamps would suddenly change wildly |
| 15:33.01 | Greenlight | It's *ALL* rtp traffic. For example, if you've ever used xlite, you know the 126 RTP keep alives it sends? |
| 15:33.14 | Greenlight | They spam the CLI with "NOTICES" but can be ignored. |
| 15:33.27 | Greenlight | Anyway, after the calls END I suddenly get all those coming through at once |
| 15:33.38 | Greenlight | As if they've been queued up somehow |
| 15:39.04 | mjordan | Do you know what kind of bridge the Bridge action puts the channels in? Native, local, remote? |
| 15:39.15 | Greenlight | I've tried both Native and Local |
| 15:39.21 | Greenlight | (By swapping codecs around) |
| 15:39.31 | file | a jitterbuffer on the channel won't allow that to happen |
| 15:39.54 | mjordan | it should native - local or remote would imply something went wrong |
| 15:40.06 | mjordan | and if it goes into native, the jitter buffers should be reset |
| 15:40.40 | Greenlight | Maybe I'm getting crossed wires |
| 15:40.53 | file | mjordan, generic = through the core, native = through the channel driver, if RTP is involved a native bridge will turn into a local or remote bridge |
| 15:40.58 | Greenlight | I made it use both the bridge associated with the channel tech |
| 15:41.05 | Greenlight | That i thought was native |
| 15:41.12 | mjordan | generic is what I meant :-) |
| 15:41.15 | Greenlight | I also made it use ast_bridge_generic (I think thbats the name) |
| 15:41.52 | mjordan | (I blame being sick) |
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| 15:42.28 | Greenlight | In my case the channel tech bridge, would be the SIP bridge. But since I had directmedia=no, it turned to local |
| 15:42.40 | Greenlight | Unless I'm missing something when i stepped through the code |
| 15:43.26 | Greenlight | When I set different codecs, the stars didn't "align" in that big if statement, and so, it ended up at ast_brdige_generic (iirc) |
| 15:43.32 | Greenlight | Either way, audio was one way. |
| 15:44.19 | Greenlight | Right, so IAX2 debugging |
| 15:44.33 | Greenlight | Gimme 30 misn or so, I've to call a customer quickly first |
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| 16:00.07 | Greenlight | SO, what's the command to enable IAX2 debugging |
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| 16:03.00 | navaismo | iax2 set debug on |
| 16:03.18 | Greenlight | Was afrid you were going to say that |
| 16:03.41 | Greenlight | In fact, I wonder if this is the source of the problem |
| 16:03.53 | Greenlight | I've not got iax2 enabled and compiled |
| 16:04.59 | Greenlight | Is it just the iax2 debugging that the jitterbuffer uses, or does it perhaps rely on it for other things ? |
| 16:15.29 | mjordan | Greenlight: iax2 just has a mechanism that will display debug messages from any jitter buffer. The jitter buffer works fine without i |
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| 16:16.05 | Greenlight | Ahh well, so I'm best to recompile iax2 support, to be able to enable iax2 debugging ? |
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| 16:39.55 | Micc | 360 just started sending me one number with ;rn=5092040000 attached to it. Is this a valid invite or is asterisk just not parsing it correctly? I never got this from them before. INVITE sip:5099563083;rn=5092040000@199.38.209.67:5060 SIP/2.0 |
| 16:40.16 | Micc | shouldn't the ;rn= be after the :5060? |
| 16:42.57 | WindBack | [TK]D-Fender: Simple Question: Is insecure to have nat=no in general section and rewrite some peers with nat=yes?? This configuration can make those peers discoverables? |
| 16:43.05 | WindBack | [TK]D-Fender: sorry |
| 16:43.41 | WindBack | [TK]D-Fender: I didn't want to ask you |
| 16:53.25 | [TK]D-Fender | Apparently you did |
| 16:53.28 | [TK]D-Fender | And did so twice |
| 16:53.57 | [TK]D-Fender | WindBack, NAT settings have to be what they have to be. |
| 16:56.10 | WindBack | [TK]D-Fender: If you have no desire to answer, do not answer me. But that answer does not help me at all |
| 16:58.46 | igcewieling1 | WindBack: the answer is "generally no" |
| 17:01.16 | WindBack | igcewieling1: I'm asking this because in sip.conf you can read: ONLY DEFINE NAT SETTINGS IN THE GENERAL SECTION |
| 17:01.23 | *** join/#asterisk Minotaur01 (minotaur01@d72-38-1-100.commercial1.cgocable.net) |
| 17:01.51 | igcewieling1 | WindBack: that has nothing to do with security, which is what you were asking about. |
| 17:03.31 | WindBack | igcewieling1: the sip.conf documentation says: IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from the nat setting in a peer definition, then the peer username will be discoverable |
| 17:03.31 | WindBack | <PROTECTED> |
| 17:03.31 | WindBack | ; undefined peers. |
| 17:04.37 | igcewieling1 | WindBack: then set nat=yes |
| 17:04.54 | igcewieling1 | Sorry, I was responding to what you asked, not what you mean. |
| 17:06.01 | WindBack | I think that having peers discoverables is insecure |
| 17:06.05 | igcewieling1 | what exactly they mean by "discoverable" is unclear to me. |
| 17:07.00 | file | it means that from the response you can determine if an account is configured with that name or not |
| 17:07.06 | file | so a brute force attack can focus on the user account |
| 17:07.11 | igcewieling1 | if you mean "remote attacker determine valid usernames" then set alwaysauthreject=yes and allowguest=no will solve that issue. |
| 17:07.27 | WindBack | igcewieling1: yes |
| 17:08.08 | file | well, if the response goes to a different place based on the configuration... while the packet itself may be the same between an account that exists and doesn't exist the very act of that packet going elsewhere would mean the configuration differs, and thus the account exists |
| 17:08.27 | file | information disclosure is a fun thing |
| 17:10.37 | WindBack | file: And what do you think? Having nat=no in general and having nat=yes in some peers will make asterisk answer different in a brute force attack? |
| 17:13.30 | file | yes |
| 17:13.59 | file | unless things have changed nat=no will cause the response to go to the information in the Via header, with nat=yes the response will go to the source IP address and port |
| 17:18.34 | *** join/#asterisk dfighter (~dfighter@arcemu/staff/dfighter) |
| 17:22.21 | igcewieling1 | does anyone know if this message indicates a problem? "[Feb 21 12:21:44] WARNING[28648]: chan_sip.c:20923 handle_response_invite: just did sched_add waitid(1512202) for sip_reinvite_retry for dialog 1964507-3570456098-154818@MSX2.nyigc.net in handle_response_invite |
| 17:22.21 | igcewieling1 | " |
| 17:26.22 | *** join/#asterisk arapaho (~arapaho@triton.infomaniak.ch) |
| 17:30.51 | *** join/#asterisk arapaho (~arapaho@triton.infomaniak.ch) |
| 17:32.01 | *** join/#asterisk arapaho (~arapaho@pierre.infomaniak.ch) |
| 17:32.22 | navaismo | never saw it before today ^ |
| 17:33.56 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 17:36.20 | [TK]D-Fender | Anyone here using 3CX for iPhone? I've got an iPhone 4 here an just installed it. with OS 6.2.1 (latest). Seems to work and runs in the background but I have no icon for it and can't see how I'm supposed to open a dialer, etc. |
| 17:39.16 | WindBack | file: thanks!! |
| 17:40.04 | WindBack | file: What is the disadvantage of setting in general section nat=yes when having a lot of peers that doesn't need it? |
| 17:40.37 | file | it *shouldn't* cause any issues |
| 17:40.53 | drmessano | I've not seen it cause issues |
| 17:41.09 | WindBack | file: perhaps more load for the server? |
| 17:41.45 | file | nope |
| 17:41.54 | file | it just changes where stuff goes, basically |
| 17:42.30 | file | so really if the remote device is extremely poorly implemented or doesn't support symmetric RTP (can't name any...) it won't work |
| 17:42.54 | WindBack | yes.. I understand. When setting nat=yes, Asterisk reads the IP stack whereas setting nat=no it reads from the SIP stack |
| 17:43.13 | WindBack | file: thanks a lot for your advices |
| 17:47.27 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 18:01.52 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 18:02.30 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.52.48) |
| 18:07.50 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:38c4:4980:2928:43a8) |
| 18:09.47 | igcewieling1 | navaismo: we get it ALL the time and are also having some weird calling issues. |
| 18:11.19 | file | it's a reinvite race condition |
| 18:11.27 | file | multiple reinvites happening at once |
| 18:11.39 | WIMPy | What was planned for last year has now got a definite deadline in 3 years: Switching off the PSTN. |
| 18:12.42 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-219-40.red.bezeqint.net) |
| 18:13.07 | igcewieling1 | file: Is it a problem? |
| 18:13.25 | file | no, the code just waits a period of time and then attempts to reinvite again |
| 18:13.43 | igcewieling1 | we ran out of RTP ports on one of our boxes recently --- with 5 active calls. default rtp.conf. |
| 18:14.05 | igcewieling1 | That is why I'm concerned, but it doesn't sound directly related. |
| 18:14.17 | igcewieling1 | might be nice if it was a NOTICE though. |
| 18:15.01 | ChannelZ | doesn't default rtp.conf list a range of like 10000 ports? |
| 18:15.24 | igcewieling1 | ChannelZ: correct. |
| 18:15.44 | ChannelZ | And you ran out with 5 calls? |
| 18:16.05 | igcewieling1 | thankfully calls failed over to one of our other boxes, but it was still distressing. |
| 18:16.14 | igcewieling1 | ChannelZ: correct. should not have happened. |
| 18:20.31 | ChannelZ | NAT? Is it leaving mappings in place for 3 weeks or something!? |
| 18:25.18 | ChannelZ | [TK]D-Fender: dunno about the iPhone but does it maybe integrate into the phone's own dialer like ones on Android can? |
| 18:26.35 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:34.55 | *** join/#asterisk gerhard7 (~gerhard7@82-169-24-72.ip.telfort.nl) |
| 18:36.21 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 18:41.01 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 18:51.19 | igcewieling1 | ChannelZ: almost no natted devices |
| 18:51.23 | *** join/#asterisk MrTAP (~Trev@d108-180-49-126.bchsia.telus.net) |
| 18:58.10 | ChannelZ | So is it a bug in * allocating RTP ports but not releasing them, or..? I wasn't really tracking the conversation, I think I must have been missing something way back in scrollback |
| 19:04.23 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 19:06.02 | *** join/#asterisk areski (~areski@80.174.255.57.dyn.user.ono.com) |
| 19:06.15 | [TK]D-Fender | ChannelZ, was away for a while. Restarting the phone had it come up as normal. |
| 19:06.36 | [TK]D-Fender | ChannelZ, And it works pretty decently just FYI |
| 19:13.31 | ChannelZ | rebooting fixes everything |
| 19:13.53 | leifmadsen | always does |
| 19:14.34 | WIMPy | Only if it's broken by design. |
| 19:14.59 | leifmadsen | everything is broken by design |
| 19:15.20 | WIMPy | Might be true by now. |
| 19:16.12 | *** join/#asterisk nicknam1232 (d9494764@gateway/web/freenode/ip.217.73.71.100) |
| 19:16.43 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:22.28 | [TK]D-Fender | Well this was FruitTech based.... |
| 19:24.13 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 19:24.13 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 19:26.33 | *** join/#asterisk Minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
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| 19:37.00 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 19:38.26 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 19:40.48 | danfromuk | Hi, when using a macro to allow the callee to filter calls, how can I set MACRO_RESULT=BUSY when the user hangs up? Currently, if the user hangs up, the macro simply ends and the caller is disconnected. |
| 19:42.25 | danfromuk | In short, I want to carry on running the Macro's dialplan even when the called party disconnects. |
| 19:44.10 | danfromuk | The macro is run like this: "Dial(SIP/201,,M(press1))" |
| 19:45.42 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 19:45.43 | saint_ | hi all |
| 19:45.56 | lorsungcu | sup yo |
| 19:46.17 | saint_ | can someone tell me in this simple auto attendant menu http://pastebin.com/aH3Ldpc4 if it is OK at the end to call autoattendant,mainmenu if someone presses 0 ? |
| 19:46.26 | saint_ | because when I try that, it hangs up on me |
| 19:47.37 | lorsungcu | there is no invalid destination |
| 19:48.21 | saint_ | so if someone is in the exten => 1,1 , and chose 0 , it should go back to the main menu prompt, right ? |
| 19:48.38 | lorsungcu | no |
| 19:48.43 | lorsungcu | it should hang up on you |
| 19:48.53 | saint_ | why is that ? |
| 19:48.58 | lorsungcu | oh, in 1,1? |
| 19:49.10 | saint_ | yez |
| 19:49.30 | lorsungcu | try adding "" around ${CHOICE} |
| 19:49.31 | saint_ | when I ' m in 1,1 if I press 0 , it hangs up on me instead of sending me back to the n(mainmenu) |
| 19:50.05 | *** join/#asterisk BriGuy (~BriGuy@74.115.41.6) |
| 19:50.24 | [TK]D-Fender | <danfromuk> In short, I want to carry on running the Macro's dialplan even when the called party disconnects. <- "h" at best |
| 19:50.24 | saint_ | I added something else in my dialplan: |
| 19:50.24 | saint_ | exten => 999,1,Goto(autoattendant,mainmenu) |
| 19:50.33 | saint_ | oh . |
| 19:50.34 | saint_ | hold on |
| 19:51.22 | [TK]D-Fender | saint_, Helps when you tell it a proper full destination.... |
| 19:51.44 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 19:51.44 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 19:54.57 | saint_ | I added the " around ${CHOICE} , but it still hangs up on me. |
| 19:55.16 | saint_ | [TK]D-Fender: can I have your lights on http://pastebin.com/aH3Ldpc4 |
| 19:55.39 | saint_ | [TK]D-Fender: then the caller choses menu 1 , 2 options: 5 or 0. If he presses 0 , it hangs up. |
| 19:56.44 | saint_ | i think i know where the problem is. |
| 19:56.50 | saint_ | there is no priority in my destination... |
| 19:57.11 | [TK]D-Fender | There is, and it's wrong |
| 19:57.22 | saint_ | can i put a n as a priority ? |
| 19:57.24 | [TK]D-Fender | no |
| 19:57.31 | saint_ | so what's the work around in this case ? |
| 19:57.38 | [TK]D-Fender | numbering them |
| 19:57.42 | saint_ | i need to create a specific context ? |
| 19:57.43 | [TK]D-Fender | or referring to a LABEL |
| 19:57.43 | saint_ | ouch.. |
| 19:58.13 | [TK]D-Fender | And I'm not sure what you're asking me following the pastebin |
| 19:59.53 | saint_ | [TK]D-Fender: in my pastebin, i m using a label .. |
| 20:00.29 | saint_ | autoattendant,mainmenu |
| 20:03.33 | saint_ | [TK]D-Fender: is there any trick so someone who;s in my 1,1 can press 0 and go back to autoattendan,mainmenu ? |
| 20:06.16 | [TK]D-Fender | saint_, You failed to read GOTO's instructions. |
| 20:06.24 | [TK]D-Fender | saint_, "core show application goto" |
| 20:10.39 | saint_ | [TK]D-Fender: i think i understand there is no priority in my goto. but from what you can see in my pastebin, is this the best way to do it ? |
| 20:11.01 | saint_ | or should i create context for every options of the auto attendant ? |
| 20:11.04 | [TK]D-Fender | No, what you fail to realize is there is ALWAYS a priority if you pass parameters at all. |
| 20:11.15 | [TK]D-Fender | exten => 999,1,Goto(autoattendant,mainmenu) |
| 20:11.21 | [TK]D-Fender | autoattendant = EXTENSION |
| 20:11.22 | saint_ | [TK]D-Fender: i fixed this one |
| 20:11.26 | [TK]D-Fender | mainmenu = PRIORITY |
| 20:11.36 | saint_ | goto(attendant,mainmenu,start) |
| 20:11.42 | [TK]D-Fender | what you WANTED for the values was wrong. |
| 20:12.00 | saint_ | i read goto and fixed this one |
| 20:12.20 | saint_ | my question is about the pastebin. is this the best way to do an auto attendant, or should i create a context for each menu ? |
| 20:12.21 | [TK]D-Fender | priority. exten,priority. context,exten,priority |
| 20:12.26 | *** join/#asterisk bitwize (~bitwize@c83-253-251-219.bredband.comhem.se) |
| 20:12.35 | [TK]D-Fender | saint_, that makes no sense |
| 20:12.47 | saint_ | [TK]D-Fender: what does not ? |
| 20:13.09 | danfromuk | Hi, can someone assist with this? http://pastebin.com/gvHc9Tgn I've been using Macros with the Dial command to allow the Called Parties to accept or reject an incoming call. But if a called party answers the phone, doesnt press 1 for accept and simply hangs up the phone, the macro_result doesnt get set and the incoming call gets disconnected instead of continuing to ring at the other peers. |
| 20:15.18 | danfromuk | Sorry, forgot this. http://pastebin.com/4Jwkyyq8 |
| 20:23.21 | AkkerKid | Hey all! How come when i have exten => _.#,1,DoStuff() with exten => _.,1,DoStuff() right after it and I type 12345 on the keypad (without the pound), it runs the first instead of the second? |
| 20:24.29 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-csdubmarpsmneewp) |
| 20:25.08 | [TK]D-Fender | AkkerKid, You don't get to put anything after "." |
| 20:25.25 | [TK]D-Fender | AkkerKid, "." = end of the line. There is no way to pattern up "ends with" |
| 20:25.52 | leifmadsen | AkkerKid: because the . eats everything, the # at the end means nothing |
| 20:26.36 | *** join/#asterisk teff (~teff@client-80-1-163-115.bsh-bng-011.adsl.virginmedia.net) |
| 20:27.38 | saint_ | is it legal to do this; Set(COUNTDOWN = $[${COUNTDOWN} + 1]) |
| 20:27.49 | saint_ | because it looks like my variable is not getting incremented.. |
| 20:28.07 | [TK]D-Fender | saint_, No extra spaces around the = |
| 20:28.10 | danfromuk | Set(COUNTDOWN = ${COUNTDOWN} + 1) |
| 20:28.23 | [TK]D-Fender | ^ even worse |
| 20:28.33 | [TK]D-Fender | it is an expression, it needs the $[] |
| 20:28.39 | [TK]D-Fender | no spaces around the = |
| 20:28.45 | [TK]D-Fender | Don't get creative. |
| 20:28.52 | [TK]D-Fender | Dialplan is quite fussy |
| 20:29.05 | [TK]D-Fender | It was clearly based on Systen 360 COBOL |
| 20:29.06 | danfromuk | One second. I was getting ahead of myself. Does he not need MATH ? |
| 20:29.07 | [TK]D-Fender | ;) |
| 20:29.13 | [TK]D-Fender | danfromuk, No. |
| 20:29.43 | saint_ | [TK]D-Fender: worked, thanks |
| 20:30.01 | danfromuk | Learn something new every day. |
| 20:31.28 | danfromuk | [TK]D-Fender: whats the need for the MATH(x+x) function if the above works? |
| 20:32.16 | AkkerKid | So D-Fender, in my case i need to save a variable-length variable from an IVR that may or may not include a pound at the end. If the pound is there, I must remove it. Is there any way to do this? |
| 20:35.02 | AkkerKid | Basically I need a function that returns only digits. |
| 20:38.59 | saint_ | [TK]D-Fender: this example https://wiki.asterisk.org/wiki/display/AST/Goto+Application+and+Priority+Labels shows that you can call and extension and label (the line with goto(start,monkey) |
| 20:39.57 | [TK]D-Fender | danfromuk, Not sure really.... |
| 20:40.34 | [TK]D-Fender | saint_, <[TK]D-Fender> priority. exten,priority. context,exten,priority |
| 20:40.59 | [TK]D-Fender | label is a priority |
| 20:41.11 | danfromuk | Is Tilghman Lesher ever on IRC? Whats his nick? |
| 20:41.26 | [TK]D-Fender | AkkerKid, remeber the difference is stripping in from a variable in how you use it VS what you match |
| 20:41.41 | saint_ | [TK]D-Fender: got my stuff to work... thanks again |
| 20:41.47 | [TK]D-Fender | danfromuk, tzafrif |
| 20:41.58 | [TK]D-Fender | tzafrir_laptop <-- |
| 20:42.00 | [TK]D-Fender | currently |
| 20:42.47 | danfromuk | tzafrir_laptop: are you currently online? I want to discuss an issue regarding macros which you've commented about in this https://issues.asterisk.org/jira/browse/ASTERISK-9561 |
| 20:42.52 | danfromuk | [TK]D-Fender: thanks |
| 20:44.05 | tzafrir_laptop | [TK]D-Fender, err... I'm not Tilghman Lesher |
| 20:44.51 | tzafrir_laptop | He's on-line, and in #asterisk-dev |
| 20:45.00 | danfromuk | oops. thanks for replying. |
| 20:45.41 | danfromuk | Whats his nick? I cant see which one he is. |
| 20:45.42 | mjordan | Tilghman is Corydon76-* |
| 20:46.22 | tzafrir_laptop | We need something like http://db.debian.org/ :-) |
| 20:46.37 | mjordan | tzafrir_laptop: that would be handy |
| 20:47.02 | mjordan | there's all sorts of 'mappings' of nicks to names and other aliases in the repotools script |
| 20:47.09 | mjordan | it would have been nice to have a db to query :-) |
| 20:48.36 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.52.48) |
| 20:52.21 | danfromuk | Does setting MACRO_RESULT cause the macro to exit immediately? My thought is that maybe I can set MACRO_RESULT=BUSY at the start the macro, I can update it later on if the caller presses 1. In the event that the user presses nothing, MACRO_RESULT will already be set and therefore the call will continue for the other called parties. |
| 20:53.28 | *** join/#asterisk goldkatze (~nobody@unaffiliated/goldkatze) |
| 20:53.34 | goldkatze | Hi |
| 20:54.35 | goldkatze | Is it possible to use Asterisk and mISDN to build a PBX which provides an S0-Bus for ISDN-phones? |
| 20:55.15 | goldkatze | As in, can I use mISDN like a "host" to /generate/ an S0-bus or just as yet-another-isdn-client-phone |
| 20:55.38 | *** join/#asterisk Nugget (nugget@rennsport.macnugget.org) |
| 20:58.35 | AkkerKid | <[TK]D-Fender>, So rather than try to build two lines that match for the two possibilities, i should make one line that matches it all and strip the variable in that? Makes sense. |
| 21:00.29 | *** join/#asterisk Vince-0 (~Vince-0@41-135-186-164.dsl.mweb.co.za) |
| 21:01.47 | teff | can anyone help with a problem with inbound uri calls? I am trying to call myusername@mydomain from myusername@getonsip.com, asterisk seems to be matching the username and trying to register instead of using the default context |
| 21:03.09 | [TK]D-Fender | <tzafrir_laptop> [TK]D-Fender, err... I'm not Tilghman Lesher <- sorry, think I got my names mixed up |
| 21:03.20 | [TK]D-Fender | asdgahjdghjasdgtyadn0ew6rv0w96r7nv0w9erv |
| 21:03.26 | [TK]D-Fender | blarg |
| 21:03.44 | ChannelZ | writes that down as a password |
| 21:03.56 | [TK]D-Fender | danfromuk, No |
| 21:03.58 | AkkerKid | Would this do it? Set(StrippedNumber=${IF($[${EXTEN:-1:1}=#] ?${EXTEN:0:-1}:${EXTEN})}) |
| 21:04.31 | ChannelZ | teff: you mean it's trying to match a peer called 'myusername' ? |
| 21:05.46 | AkkerKid | looks like it works. Sweet! |
| 21:05.52 | [TK]D-Fender | AkkerKid, not bad :) |
| 21:06.01 | teff | ChannelZ, ermmm :D maybe, WARNING[9789]: chan_sip.c:12886 check_auth: username mismatch, have <myusername>, digest has <getonsip_myusername> |
| 21:06.04 | AkkerKid | I'm almost good at this |
| 21:06.24 | AkkerKid | now if only i can get asterisk to connect to MSSQL |
| 21:06.26 | [TK]D-Fender | AkkerKid, The wheels are spinning decently... |
| 21:06.27 | teff | myusername is a friend (my extension) in sip.conf |
| 21:06.53 | ChannelZ | well |
| 21:07.23 | AkkerKid | FreeTDS refuses to use the password for the MSSQL server that's stored in plaintext in odbcinst.conf. |
| 21:07.28 | [TK]D-Fender | AkkerKid, I've heard reference to using FreeTDS for this |
| 21:07.42 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 21:07.43 | ChannelZ | the problem with friends is that they are both a peer and a user. And a user matches by the From: so if you're calling yourself (or someone tries to call you with the same exact name) you're just going to run into that. |
| 21:07.55 | teff | ahhh |
| 21:07.59 | AkkerKid | FreeTDS all of a sudden requires that you use kerberos for authentication or some such... |
| 21:07.59 | [TK]D-Fender | AkkerKid, IIRC you have to mention the user & pass on *'s side, not just expecting the SDSN pass-through |
| 21:08.28 | AkkerKid | D-Fender, you mean in res_odbc.conf? |
| 21:08.33 | teff | ChannelZ, so make it a user only? |
| 21:08.39 | AkkerKid | it doesn't help. :( |
| 21:09.18 | ChannelZ | no, a peer.. which matches by IP first |
| 21:09.48 | ChannelZ | If you want this remote call to be treated as 'anonymous' |
| 21:10.08 | teff | ChannelZ, nice one, got it. Thank you very much |
| 21:11.31 | ChannelZ | np |
| 21:11.35 | coreyf1513 | AkkerKid: are you sure FreeTDS caused the authentication requirement to change? That seems like something MSSQL would do. |
| 21:13.00 | AkkerKid | that may be possible. I was able to connect in the past just fine but the feature that needed that conection was depreciated |
| 21:13.09 | AkkerKid | so i left it off and didn't bother with it. |
| 21:13.52 | AkkerKid | since then, we've upgraded from MSSQL 9 to 10 |
| 21:13.58 | AkkerKid | maybe that's it |
| 21:15.12 | coreyf1513 | AkkerKid: verify you can connect to MSSQL with plain-text user/password (specifically not integrated authentication). I'm willing to bet you won't be able to do it even with the MS tools. |
| 21:16.20 | AkkerKid | this doesn't work: echo "select 1" | isql -v "asterisk-MSSQL-connector" |
| 21:16.33 | AkkerKid | this does: echo "select 1" | isql -v "asterisk-MSSQL-connector" "username" "password" |
| 21:16.57 | AkkerKid | when i pass the UID and Pass directly to isql, it works fine. |
| 21:17.11 | *** join/#asterisk italorossi (~italoross@177.43.116.186) |
| 21:17.45 | AkkerKid | but when i expect isql to get that info from odbc.ini it won't do it |
| 21:19.52 | AkkerKid | the first example errors out saying it couldn't connect with username (blank) |
| 21:20.02 | coreyf1513 | AkkerKid: is "username" an activedirectory account? |
| 21:20.42 | AkkerKid | it is not. |
| 21:21.08 | *** join/#asterisk zoelorenz (~Alfiela@tdev246-231.codetel.net.do) |
| 21:21.58 | zoelorenz | hey, a quick question, i have a TDM410 with 3 FXS ports and 1 FXO, can i connect a t1 line in a fxs port? |
| 21:22.56 | AkkerKid | nope |
| 21:22.59 | leifmadsen | lolz, no |
| 21:23.05 | AkkerKid | T1 requires a different kind of card |
| 21:23.38 | leifmadsen | that would be like a yak mating with a cat |
| 21:23.53 | AkkerKid | ...ew |
| 21:24.09 | zoelorenz | oh, i see. |
| 21:24.18 | AkkerKid | *sneeze* (Cat allergies) |
| 21:24.25 | leifmadsen | one is digital circuit, the other is analog |
| 21:24.31 | leifmadsen | they are totally different things |
| 21:24.37 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 21:26.05 | AkkerKid | OK, so when I try connecting through FreeTDS with and active directory account, it works even less. |
| 21:26.24 | AkkerKid | can't even get isql to connects when defining the user/pass in the string. |
| 21:26.57 | coreyf1513 | AkkerKid: maybe verify mssql can be used remotely (newer versions sometimes restrict to localhost only or named pipes only), make sure you can reach the mssql tcp port from asterisk. |
| 21:27.26 | AkkerKid | my isql test proves that it's not an issue on the remote side |
| 21:27.39 | *** join/#asterisk dfighter (~dfighter@arcemu/staff/dfighter) |
| 21:30.26 | AkkerKid | this should show it nicely: http://pastebin.com/8tdB3jtQ |
| 21:30.58 | _Corey_ | just logged into a production machine somewhere running 1.2.14... |
| 21:35.01 | navaismo | XD |
| 21:36.53 | coreyf1513 | AkkerKid: pastebin missing your res_odbc.conf |
| 21:39.04 | *** join/#asterisk themrrobert (cf4322ee@gateway/web/freenode/ip.207.67.34.238) |
| 21:39.13 | themrrobert | is it possible to have 2 AMI ports open? |
| 21:39.27 | WIMPy | themrrobert: Sure |
| 21:39.45 | leifmadsen | I thought AMI just listened on a single port... |
| 21:39.49 | leifmadsen | or do you mean 2 connections? |
| 21:39.51 | [TK]D-Fender | checkout time, BBIAB |
| 21:39.58 | themrrobert | @leifmadsen you read me correct |
| 21:40.09 | leifmadsen | AMI listens on a single port. |
| 21:40.16 | themrrobert | thought so, thanks :) |
| 21:40.17 | WIMPy | Unless you're talking about listeners. In that case you have to do it with iptables, but I'm not sure that makes any sense. |
| 21:40.29 | themrrobert | boss wants the impossible as always. |
| 21:40.32 | leifmadsen | astmanproxy *may* listen on multiple ports |
| 21:40.33 | themrrobert | i agree WIMPy |
| 21:40.41 | leifmadsen | I think your boss doesn't understand how services work :) |
| 21:40.56 | WIMPy | Obviousely. |
| 21:42.11 | AkkerKid | updated pastebin: http://pastebin.com/Si8YC54u |
| 21:42.28 | *** join/#asterisk UnixDev (~UnixDev@unaffiliated/unixdev) |
| 21:42.53 | UnixDev | on a dual core machine w/1 gig of ram, how many Registered endpoints can Asterisk safely have? |
| 21:43.01 | UnixDev | running 1.8.x (latest stable) |
| 21:43.18 | AkkerKid | UnixDev, a few hundred. |
| 21:43.40 | UnixDev | so like 200 or closer to 500 ? |
| 21:44.18 | AkkerKid | assuming it's only registering and not transcoding... several hundred |
| 21:45.33 | themrrobert | We're encountering an issue where we're timing out before getting confirmation of member login to a queue. isymphony is having the same issue, but our .Net error looks like his http://pastebin.com/Xv8uXrec |
| 21:45.42 | AkkerKid | I've got a quad Xeon with 104 extensions currently registered and CPU is mostly idle |
| 21:46.03 | leifmadsen | I have over 2000 on Asterisk 1.4 boxes |
| 21:46.19 | AkkerKid | well of course you would you wrote the book. |
| 21:46.23 | leifmadsen | on Asterisk 1.8 just don't use DEBUG_THREADS and DONT_OPTIMIZE or you'll get less than 600 |
| 21:53.37 | AkkerKid | @leifmadsen: Have any idea why isql refuses to use the passwords stored in odbc.ini to connect to my MSSQL server? |
| 22:00.48 | coreyf1513 | AkkerKid: do you have selinux enabled (maybe it's interfering with asterisk)? sorry that's my last idea if isql from the asterisk box works but asterisk odbc doesn't |
| 22:01.27 | leifmadsen | AkkerKid: not sure, could be your odbc.ini isn't right. At some point the field names changed |
| 22:01.37 | leifmadsen | I tend to place them in res_odbc.conf as well when testing |
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| 23:13.31 | UnixDev | AkkerKid: sometimes transcoding is needed, say 10% of regs doing active calls, and say 20% of those clients transcode, |
| 23:13.35 | UnixDev | AkkerKid: what do you estimage? |
| 23:13.39 | UnixDev | estimate* |
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| 23:19.39 | Phoebus | Just took part in a straight to the point, cool Webinar by digium. I dig them even more. |
| 23:20.31 | file | points are pointy |
| 23:20.51 | Phoebus | True true... |
| 23:21.04 | lorsungcu | Phoebus: for what |
| 23:21.19 | Phoebus | Build or buy, asterisk vs switchvox. |
| 23:21.30 | Phoebus | Pretty helpful to a newbie. |
| 23:34.23 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 23:34.23 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 23:55.48 | *** join/#asterisk docelmo (~docelmo@pool-96-235-211-100.lyncva.east.verizon.net) |
| 23:56.06 | docelmo | Say anyone know if you can point MWI messages to a specific host? |
| 23:56.46 | docelmo | For instance.. I have an asterisk box used for VM.. I want to point any MWI it creates and send it to a host where the users are registered |
| 23:56.59 | docelmo | Is this possible and if so can someone point me in a direction? |
