| 00:00.17 | Rahail | i read more and more on loadbalancing every one diffrent example how loadbalance trunk |
| 00:01.32 | Rahail | any one else can give me live example please loadblance in 2-5 sip trunk/peer |
| 00:03.42 | iprouteth0_ | yeaay... more fun with SIP-TLS and sRTP today.... |
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| 00:05.40 | lvlinux | anybody familiar with Adtran TA6xx devices and Asterisk?? (i've got a TA608) |
| 00:16.12 | Rahail | lvlinux do you have any idea bout how to round robin sip trunk |
| 00:17.08 | WIMPy | Use a global counter. |
| 00:17.26 | Rahail | WIMPy can you show me example |
| 00:17.34 | Rahail | i read on global counter |
| 00:17.38 | Rahail | couldnt figer out |
| 00:18.09 | WIMPy | Did you read the book? |
| 00:18.14 | Rahail | still reading |
| 00:18.16 | Rahail | and trying |
| 00:18.42 | WIMPy | Good. That should answer your questions. |
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| 00:19.04 | Rahail | i am reading and i think i am understaning |
| 00:19.05 | Rahail | wrong |
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| 00:21.24 | Rahail | WIMPy can you just show me one example please |
| 00:21.28 | Rahail | from your knowledge |
| 00:23.46 | *** join/#asterisk navaismo (~navaismo@189.241.17.58) |
| 00:25.12 | navaismo | : Hi, first of all a Pray for Boston |
| 00:26.59 | navaismo | now a question: an E1->PRI ISDN with Frame Structure as per ITU-T (CCITT) G.704, with CodeHDB3 translate to system.conf as span= 1,1,0,ccs,hdb3 right? |
| 00:27.17 | WIMPy | Rahail: Do you have any programming or scripting experience? |
| 00:27.25 | Rahail | nope |
| 00:27.30 | Rahail | just copy paste and reading |
| 00:28.00 | WIMPy | That might be a bit of an issue. |
| 00:29.29 | WIMPy | navaismo: Looks good. Might have crc4 as well. |
| 00:29.33 | Rahail | yeap thats for usre |
| 00:29.34 | Rahail | sure |
| 00:30.23 | navaismo | thanks WIMPy, I still the issue with that E1 can't take up with asterisk, using crc4 or not |
| 00:31.03 | WIMPy | What was your situation? |
| 00:31.52 | lvlinux | Rahail: no sorry I've never done it before---have you looked at the Asterisk Definitive guide? I've found all the examples in there to be explained very well. |
| 00:34.17 | navaismo | WIMPy, the leds in digium card are red when i connect the E1, |
| 00:34.43 | WIMPy | And what about LEDs on the NT? |
| 00:37.40 | navaismo | When I connect the pbx all off, when I connect the old pbx all on in both sides(old pbx and modem) |
| 00:39.06 | WIMPy | What kind of card? |
| 00:39.17 | Rahail | WIMPy do you think you can just create that conf for me and i can copy paste it |
| 00:39.33 | Rahail | turnk name gw1 gw2 gw3 gw4 |
| 00:39.35 | Rahail | like this |
| 00:39.37 | WIMPy | Could somethig have a non standard pin-out or a configurabel pin-out? |
| 00:42.09 | navaismo | WIMPy, its a Wildcard TE420 (5th Gen) |
| 00:44.02 | lvlinux | Rahail: check out this link http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#ACD_id285928 |
| 00:44.47 | WIMPy | navaismo: Ok, so that'ss always TE. Is the card correctly jumpered to E1? |
| 00:45.08 | WIMPy | Have you tried to connect the old PBX to the card? |
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| 00:48.03 | WIMPy | Rahail: Looks liike the examples use more advanced stuff like group counts. |
| 00:48.16 | WIMPy | Which is probably a good thing to do. |
| 00:48.52 | maxus2 | Hi People, I have an issue with asterisk 11 certifed, it appears on centos 6 running on xen. IAX requests and not processed but i can see them queue in the Recv-Q in netstat. If i upgrade to asterisk 11.3 eveything works fine. any ideas why the certifed would be broken? |
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| 00:49.25 | Rahail | WIMPy can you give me one please so i can compare this with the doc and see why i didnt understant that |
| 00:49.28 | Rahail | if you can |
| 00:49.53 | WIMPy | It's the good old question to auntie Google. |
| 00:50.32 | Rahail | google have enouh info which not getting in my head |
| 00:50.58 | WIMPy | Noone has said using Asterisk would be easy. |
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| 00:55.12 | navaismo | WIMPy, yes is set to E1 and I also overrided by directive in the modprobe directory to e1, no luck |
| 00:55.22 | navaismo | and not tried to connect the old pbx to asterisk |
| 00:57.24 | WIMPy | If you don't get the interface up at either end. There might be some stange pin out perhaps. Is that old equipment? |
| 00:58.33 | navaismo | the asterisk its a new server |
| 00:58.50 | WIMPy | Sure. And the NT? |
| 00:59.17 | WIMPy | Or is it a simple NT or some strange gateway setup? |
| 00:59.37 | navaismo | well not sure how old is it, the brand its pandatel fiber to e1 |
| 01:01.40 | lvlinux | Rahail: http://www.orderlyq.com/asteriskqueues.html |
| 01:01.52 | lvlinux | Rahail: http://astrecipes.net/?n=42 |
| 01:02.02 | lvlinux | Rahail: both have example code |
| 01:05.44 | WIMPy | navaismo: Oh, something with two fibers? |
| 01:09.41 | navaismo | hmmm not sure, 2 cables are two fibers? This is a really new equipment to me, i used to set a balun to the bnc acable then the rj45 to pbx |
| 01:10.21 | WIMPy | So you have the version with BNC connectors? |
| 01:11.14 | WIMPy | But what does "used to" mean? You still need that for the card (if the line befor is right). |
| 01:11.28 | navaismo | no, no, that is the i used to work now i have that modem with rj45 outputs |
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| 01:11.46 | navaismo | used--> worked in the past |
| 01:11.50 | navaismo | sorry |
| 01:11.59 | WIMPy | Ok, so it's a new line as well? |
| 01:12.23 | navaismo | yes |
| 01:12.33 | WIMPy | And now you have a modem with G.703 on an 8p8c modular plug? |
| 01:13.47 | WIMPy | And the old PBX works with the new modem, but the card doesn't? |
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| 01:15.46 | navaismo | right right |
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| 01:19.34 | WIMPy | Do you have a loopback plug? |
| 01:20.56 | navaismo | yep, tried on the asterisk and turn the led green, and also i can see in the cli the pri_cpe configuration warning, tried on the equipment and the lights turn green |
| 01:21.57 | WIMPy | Hmm. Interesting. |
| 01:23.16 | WIMPy | One might think there's some electrical issue. |
| 01:25.38 | navaismo | well at this point idk what is the issue, and the telco doesnt provides much help so basically im shooting in the dark |
| 01:26.00 | navaismo | but again thanks for the help |
| 01:26.34 | WIMPy | I think I'd try to do a crossover cable and connect the old pbx to the card to make sure it really works. |
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| 01:28.07 | navaismo | ok tomorrow ill do that |
| 01:28.45 | WIMPy | has to go to sleep now as well. |
| 01:29.12 | navaismo | GN |
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| 01:39.38 | lvlinux | anybody familiar with Adtran TA6xx devices and Asterisk?? (i've got a TA608) |
| 01:45.00 | igcewieling | WE use the 9xx TAs |
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| 02:17.40 | iprouteth0_ | could anyone clue me in on why my srtp and sip-tls calls are failing if I provided a call example? |
| 02:21.48 | kuruption | unfortunately i've tried to get sip-tls and srtp running myself w/o success |
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| 02:28.34 | iprouteth0_ | YO! |
| 02:28.47 | iprouteth0_ | Got it workin biznitches!!!! YEAAHH! |
| 02:29.45 | leifmadsen | srtp was pretty easy to do with polycom phones |
| 02:29.47 | iprouteth0_ | somewhere in there works it was trying to redirect it back to the standard SIPS port of 5061 instead of the nonstandard port I bound it to.... weird, but I suppose since I'm not transmitting things in clear text.... |
| 02:29.57 | iprouteth0_ | Still don't like using a standard port. |
| 02:30.10 | leifmadsen | a simple trace would find that port |
| 02:30.16 | iprouteth0_ | I am using CSipSimple clients on android. I'll be working on a brief tutorial of my setup soon |
| 02:30.35 | iprouteth0_ | its a matter of low hanging fruit |
| 02:31.15 | iprouteth0_ | very happy to get it working. I wonder if stun perhaps was causing it, as I do have stun enabled |
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| 05:57.52 | iprouteth0 | loving that I've got sip-tls working and srtp working.... still wish I could run it on a non-standard port |
| 05:58.56 | ChannelZ | that's more likely up to all your devices and firewall configurations |
| 06:00.16 | izbushka | iprouteth0, " I'll be working on a brief tutorial of my setup soon" This will be nice |
| 06:00.52 | iprouteth0 | tutorial is nearly done. Not even sure I'm using the correct keys for the client, but I've tested |
| 06:01.40 | iprouteth0 | it makes calls. I ran sipdump against a non-encrypted register pcap and sipcrack with wordlist and correct password. sipdump grabbed the login creds just fine |
| 06:01.58 | iprouteth0 | it did not find any login information for the recorded registration on the TLS connection |
| 06:02.02 | iprouteth0 | pcap |
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| 06:25.19 | iprouteth0 | izbushka: http://iprouteth0.blogspot.com/2013/04/csipsimple-srtp-and-sip-tls-with.html |
| 06:25.42 | izbushka | thanks |
| 06:26.54 | iprouteth0 | I'd love for other people to experiment with attemps of TLS bound to a nonstandard port |
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| 06:33.19 | _omer | I have a question, Is it possible to create a sip user in freepbx using AMI + database commands? Actually I want to create sip user in freepbx through my .php file not from freepbx gui..... please guide me. thanks |
| 06:35.18 | kaldemar | #freepbx |
| 06:35.56 | _omer | ok.thanks |
| 06:36.23 | _omer | how about creating a sip peer in asterisk through AMI ? .... |
| 06:38.30 | *** join/#asterisk Rahail (~Rahail@67.214.121.163) |
| 06:38.34 | Rahail | HI there |
| 06:38.47 | Rahail | question again.. letsay I want limit per outgoing call to x minute |
| 06:38.51 | Rahail | how can i do that |
| 06:39.06 | kaldemar | _omer: there's no command for that. |
| 06:39.57 | kaldemar | Rahail: "core show application Dial" <-- see L(), or "core show function TIMEOUT" |
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| 06:43.41 | Rahail | ~pastebin |
| 06:43.41 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 06:44.18 | Rahail | kaldemar |
| 06:44.22 | Rahail | can u se if thi is valid |
| 06:44.23 | Rahail | http://bin.cakephp.org/view/1838394438 |
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| 06:46.20 | bulkorok | first line is wrong |
| 06:46.30 | Rahail | ops yes i missed |
| 06:46.31 | Rahail | , |
| 06:46.34 | bulkorok | exten => _x.,1,Set(TIMEOUT(absolute)=3600) |
| 06:46.41 | bulkorok | and missing space after exten! |
| 06:47.02 | bulkorok | and _x must be _X |
| 06:47.13 | Rahail | cool guys thanx |
| 06:47.37 | bulkorok | second row needs a _X too... |
| 06:48.29 | _omer | kaldemar: ok but what are the commands like "database put xxxxx" ?? can I update any sip peer using database commands? I have never worked on database commands so I am not sure what kind of access these commands have. |
| 06:48.40 | Rahail | http://bin.cakephp.org/view/1838394438 |
| 06:49.54 | Rahail | it worked |
| 06:52.29 | Rahail | have any one did round robin on trunking |
| 06:56.21 | Rahail | i am massing with this group function since 4pm |
| 06:56.26 | Rahail | still couldnt figerout |
| 06:58.39 | kaldemar | _omer: they are commands to handle astdb. and no. |
| 06:59.19 | Rahail | kaldemar have you ever mass with the round robin trunk/peer |
| 07:00.15 | kaldemar | Rahail: is the case that you have more than one trunk and want to balance load between them? |
| 07:00.23 | Rahail | yes |
| 07:01.55 | kaldemar | use a global variable to hold the one that was used last and then a bunch of IF() or GotoIf() to decide which one to use. |
| 07:02.05 | Rahail | which way can i achive this |
| 07:02.12 | Rahail | do yo uhave any live example i can use |
| 07:02.30 | kaldemar | no. |
| 07:02.44 | Rahail | do u think you can make me one :) |
| 07:03.00 | kaldemar | no. |
| 07:03.09 | bulkorok | Rahail: you should really try it by yourself |
| 07:03.20 | Rahail | honestly i been trying |
| 07:03.23 | Rahail | since 4pm |
| 07:03.25 | bulkorok | it's the best way to "learn" how asterisk-dialplan works... |
| 07:03.36 | Rahail | i am nowhere near |
| 07:03.43 | bulkorok | so... show us what you got and maybe we can help |
| 07:03.53 | kaldemar | i can give you pointers if you have questions about it but you usually get paid for doing all the work for others. |
| 07:04.11 | Rahail | kaldemar i can paypal u |
| 07:04.16 | Rahail | how much u will want |
| 07:04.23 | Rahail | :) |
| 07:05.02 | kaldemar | the price tag woule correlate with my willingness to do it. |
| 07:05.20 | kaldemar | just pastebin what you have tried. |
| 07:05.29 | Rahail | i premych lost all what i try |
| 07:05.35 | Rahail | pretymuch. |
| 07:05.46 | Rahail | whatever dont work i just see other people example and try that |
| 07:05.52 | Rahail | then dont work move one other one ... |
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| 07:07.01 | Rahail | i am using asterisk 1.8.2 |
| 07:10.08 | Rahail | kaldemar do you think you can change your mind :) |
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| 07:11.16 | kaldemar | Rahail: no. i really have other stuff to do. |
| 07:13.11 | Rahail | Please i cant find any one else.. every one is busy.. |
| 07:13.47 | bulkorok | Rahail: why not using Queue() !? |
| 07:14.19 | Rahail | I do not mind.. aslong it work.. i have no idea how to do with queue |
| 07:14.58 | bulkorok | check queues.conf how to "build" a queue with static members and call the queue in the dialplan with Queue(yourqueue) |
| 07:15.25 | bulkorok | there are many options how the queuemembers can be called... |
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| 07:17.18 | polysics | hello |
| 07:17.31 | polysics | when using EXEC over AMi, what is the proper multiple argument separator? |
| 07:17.37 | Rahail | bulkorok i can use this toward my providers |
| 07:17.45 | polysics | eg. |
| 07:18.00 | Rahail | USER A call goes to queue and then it goes to provider 1 |
| 07:18.03 | polysics | Command: EXEC MyApp "arg1" "arg2" |
| 07:18.09 | Rahail | USER b call goes to queue and then it goes to provider 2 |
| 07:18.15 | polysics | or is it a comma? or both? |
| 07:18.51 | kaldemar | polysics: you mean agi, right? |
| 07:19.03 | bulkorok | Rahail: this is how it goes... check the strategy option in queues.conf |
| 07:19.09 | polysics | kaldemar: using the AMI EXEC command |
| 07:19.33 | polysics | that falls under the AsyncAGI umbrella, I think |
| 07:20.20 | bulkorok | Rahail: wait... do you want to call different targets via your 2 provider?! |
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| 07:21.45 | bulkorok | I don't know how to handle that via Queues... I suppose you must make this with dialplan-magic |
| 07:22.17 | kaldemar | polysics: my AMI does not have an EXEC command. |
| 07:23.31 | kaldemar | polysics: so is it Action: AGI and Command: exec ? |
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| 07:24.44 | polysics | kaldemar: yes, sorry |
| 07:26.38 | kaldemar | polysics: options are separated by commas. |
| 07:26.59 | polysics | so my above example would be |
| 07:27.32 | polysics | Command: EXEC MyApp "arg1\"has quotes\"here","arg2" |
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| 07:35.04 | kaldemar | polysics: you don't need to put everything in quotes. it will more likely cause you problems. |
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| 07:52.55 | polysics | waht if an argument has quotes, speaces or commas in it? |
| 07:53.03 | polysics | ie. SSML fed to a TTS engine |
| 07:53.47 | kaldemar | you'd need to escape them somehow. |
| 07:54.05 | kaldemar | i'd try a backslash first. |
| 07:54.44 | polysics | I have also seen single quotes for the XML |
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| 09:27.10 | chesty | hey, what would cause woomera to send/receive 100kbs on the lo interface? I'm not sure I even need woomera |
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| 09:53.30 | din3sh | hello all |
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| 10:12.02 | tacpilot | have installed asterisknow 11 plus updated the system. have followed the instructions on the wiki to get working with google voice. |
| 10:14.16 | tacpilot | however i do not seem to have a connection. is there something missing from the asterisknow install .. do i need to compile it my self ? |
| 10:14.30 | *** join/#asterisk Greenlight (~email@cpc1-dund9-0-0-cust142.16-4.cable.virginmedia.com) |
| 10:15.11 | tacpilot | error message when trying to dial out shows the attempt but errors with "unable to create channel type "Motif" |
| 10:15.50 | tacpilot | (cause 66 - channel not implemented) |
| 10:16.33 | tacpilot | my guess is because asterisk is failing to properly register with google voice so not channel is created? |
| 10:18.11 | tacpilot | I am working with just the asterisk install and did not install the freepbx front end |
| 10:18.38 | tacpilot | what am I missing ?? |
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| 10:57.03 | disposable | does asterisk (1.2.40) open() /var/log/asterisk/messages and keep it opened for the entire time it is running or does it just open, append line, close it on every event? |
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| 10:59.05 | WIMPy | Might be hard to find somone who knows that and is still alive. |
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| 11:12.16 | disposable | WIMPy: i really doubt the behaviour has changed with the more recent releases. |
| 11:13.08 | Greenlight | It would seem rather inefficient if it opened and closed it for each event, I would imagine it keeps it open. |
| 11:13.38 | Greenlight | What problem are you trying to solve? |
| 11:17.20 | WIMPy | It is very inefficient, but I think it's what it does. |
| 11:18.06 | Greenlight | Wow... what's the logic behind that ? |
| 11:19.51 | WIMPy | The good thing is that you can just move the logs away without telling anyone. |
| 11:21.54 | Greenlight | Close/reopen every minute would make more sense wouldn't it |
| 11:22.06 | Greenlight | Or every 10k entries, something like that |
| 11:22.22 | WIMPy | doesn't see any sense in doing so. |
| 11:22.48 | Greenlight | So you can move logs away, like you said |
| 11:22.51 | WIMPy | Eiter you close it immediately or you keep it open. |
| 11:23.07 | Greenlight | Mind, I'm writing about 50Gb a day of log files, so it can't be *that* inefficient |
| 11:23.20 | WIMPy | You have to do that with the applications help. |
| 11:23.53 | WIMPy | Sounds horrible. |
| 11:24.20 | Greenlight | This is with Asterisk I'm meaning, as it currently does it |
| 11:24.51 | WIMPy | Do you ever read them? |
| 11:25.10 | Greenlight | Yea, all 50Gb, it's my nightly reading before bed |
| 11:25.40 | Greenlight | Joking aside, I'm an ongoing issue on JIRA that needed full debug logging |
| 11:25.46 | Greenlight | *I've |
| 11:26.12 | Greenlight | Been stable for almost 2 weeks now though, so I'm hopeful. |
| 11:26.27 | WIMPy | Debug is evil. |
| 11:26.30 | Greenlight | But worst situaiton is when it crashes and I've not got any logging to help |
| 11:26.55 | Greenlight | grep cried out in pain last time i had to parse it |
| 11:27.15 | WIMPy | I once regularly crashed an Asterisk by enabling debug logging. The harddisc couldn't cope. |
| 11:27.26 | file | the oldest I have is 1.8 but it keeps the file(s) open until reload or shutdown |
| 11:27.53 | Greenlight | Makes sense |
| 11:28.27 | Greenlight | Yea, we've a decent raid controller behind it, but a normal disk might struggle i'd imagine |
| 11:28.46 | file | there's a rotate command too |
| 11:28.50 | Greenlight | "logger rotate" |
| 11:30.07 | Greenlight | Oh, file, you know that issue you helped me with a few weeks back with the deadlock ? |
| 11:30.15 | file | yes |
| 11:30.25 | Greenlight | Suspect we found the route cause (ASTERISK-21356) |
| 11:30.26 | LieutPants | [ASTERISK-21356] [Status: Waiting for Feedback] [Assigned: greenlightcrm] Segfault during bridge channel proxy inspection in a masquerade caused by an AMI Redirect of two channels - https://issues.asterisk.org/jira/browse/ASTERISK-21356 |
| 11:30.34 | Greenlight | ooo that's new |
| 11:30.48 | file | nifty |
| 11:31.07 | Greenlight | But i've not had the guts to remove the "/n" from the Local channels yet :) |
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| 12:58.21 | leifmadsen | Greenlight: ya, I added the jiraircbot back to the channels since it had been missing so long |
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| 13:24.16 | Katty | morning |
| 13:24.17 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 13:24.35 | Greenlight | Afternoon :) |
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| 13:26.33 | *** join/#asterisk AkkerKid (~AkkerKid@23.31.20.201) |
| 13:27.05 | AkkerKid | hey all! is there any way to get past the 79 character extension limit? |
| 13:27.30 | AkkerKid | i need to initiate calls from the command line and pass a bunch of variables |
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| 13:44.00 | kaldemar | AkkerKid: maybe you need to change the method you initiate calls. if you use AMI originate, you can define channel variables in the command. |
| 13:45.10 | [TK]D-Fender | AkkerKid, Call-files also support this |
| 13:45.53 | *** join/#asterisk Free99 (~Free99@ool-4350dd5c.dyn.optonline.net) |
| 13:50.06 | AkkerKid | ultimately i need to put alot of extra info into the cdr table |
| 13:50.18 | AkkerKid | i was doing it by putting it into the dial string |
| 13:50.25 | AkkerKid | but i've hit that limit |
| 13:50.41 | [TK]D-Fender | AkkerKid, So set channel variables directly |
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| 13:54.50 | kaldemar | AkkerKid: is that the only purpose of your call? to add data to CDR? |
| 13:55.55 | AkkerKid | i'm initiating calls from a timed script to let people know there products are ready for pickup at our store. |
| 13:56.23 | [TK]D-Fender | AkkerKid, So just set the vars in whatever you're initiating the calls via |
| 13:56.30 | AkkerKid | i need to assign tracking information to those calls when they're initiated so i can remove them from the queue when they're successfully anwsered |
| 13:57.26 | AkkerKid | i also add other info to the cdr like the customer numbers and tye of call |
| 13:57.56 | AkkerKid | type* |
| 13:58.30 | AkkerKid | all of this is done by a bash script run by cron |
| 13:59.08 | [TK]D-Fender | AkkerKid, Just pass them as variables. |
| 13:59.10 | AkkerKid | it pulls qualified records out of a database and makes the call. once a call shows answered in the CDR, i remove it from the queue. |
| 13:59.27 | AkkerKid | would i be better off with call files or ami? |
| 13:59.53 | [TK]D-Fender | AkkerKid, Either works just fine |
| 14:02.12 | AkkerKid | alrighty... ill try and figure it out |
| 14:02.14 | AkkerKid | thanks |
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| 14:38.11 | *** join/#asterisk Sjors (~sgielen@foo.kassala.de) |
| 14:38.14 | Sjors | Hi all! |
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| 14:40.33 | Sjors | My Asterisk server is supposed to register to a SIP trunk, but something is going wrong there |
| 14:41.06 | Sjors | is there a way to ask Asterisk to do an immediate re-register, so I can see the surrounding debug information and try again with different settings? |
| 14:41.35 | [TK]D-Fender | Sjors, "sip reload" |
| 14:41.52 | Sjors | hmm |
| 14:42.08 | Sjors | I have sip debugging enabled, but when I do "sip reload", I see no outgoing or incoming SIP packets --- well, that might be what's wrong ;-) |
| 14:42.48 | [TK]D-Fender | Sjors, use "database del [the key]" for your registry entry first and then do the reload |
| 14:42.57 | [TK]D-Fender | Sjors, "database show" to view them |
| 14:43.13 | Sjors | allright, let's see |
| 14:43.26 | AkkerKid | it seems my asterisk doesn't want to use callfiles. |
| 14:43.37 | AkkerKid | they just sit there and nothing gets processed. |
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| 14:44.03 | AkkerKid | is there something i need to turn on |
| 14:48.13 | bombev | what does it mean that message: [2013-04-16 17:01:45] WARNING[14078]: translate.c:256 ast_translator_build_path: No translator path from alaw to unknown |
| 14:49.02 | Qwell | bombev: means you're doing something funky |
| 14:49.22 | bombev | for example? |
| 14:51.16 | Sjors | [TK]D-Fender: nice, it's working, thanks a lot! |
| 14:52.05 | [TK]D-Fender | Sjors, You're welcome. |
| 14:52.20 | [TK]D-Fender | AkkerKid, pbx_spool.so needs to be loaded. |
| 14:52.34 | [TK]D-Fender | AkkerKid, And you need to be putting them in the right place in the right way. |
| 14:54.07 | *** join/#asterisk henny22 (~henny22@host217-37-52-86.in-addr.btopenworld.com) |
| 14:55.59 | henny22 | Hi all i need some help with Dahdi settings , i just installed the Freepbx module for DAhdi and it has over ritten the existing settings , i need help setting them back for a 4 port Digium - B4XXP |
| 14:57.41 | Qwell | henny22: just copy it from your backup |
| 14:57.49 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:58.48 | igcewieling | henny22: welcome to FreePBX |
| 14:58.50 | henny22 | BackUp........ arrr ..... |
| 14:59.10 | igcewieling | henny22: the freepbx module did not backup the files it modified? |
| 14:59.31 | henny22 | dono, where would it put them |
| 14:59.37 | igcewieling | henny22: in /etc/asterisk |
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| 15:08.03 | igcewieling | henny22: in any case try running "dahdi_genconf" |
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| 15:13.58 | henny22 | can you take me threw running dahdi_genconf |
| 15:14.34 | Free99 | just out of curiosity, why is it that most sample dialplans I've seen have a Wait of one second after an Answer()? |
| 15:15.57 | Greenlight | Free99: I ususally put a Wait(1) in to give the call a moment to establish fully |
| 15:16.13 | Greenlight | For isntance when I'm about to play an IVR menu greeting |
| 15:16.37 | Greenlight | If you don't do the Wait() sometimes you'll "chop" off the first second of the message you're playing |
| 15:16.49 | Free99 | ah, was just about to ask lol |
| 15:17.08 | [TK]D-Fender | Far better to Playback(silence/1) than to just wait. Playback() forces RTP to go up |
| 15:17.10 | Free99 | ok, makes sense. Thanks GreenLight |
| 15:18.05 | Free99 | [TK]: I'll try that |
| 15:18.10 | Free99 | thanks for the tip |
| 15:20.35 | henny22 | igcewieling can you plese talk me threw uesing dahdi_genconfig |
| 15:23.16 | henny22 | or any one else |
| 15:23.28 | igcewieling | henny22: send me a system with a card which I can screw up and I'll be happy to. |
| 15:23.44 | henny22 | :) |
| 15:23.58 | igcewieling | otherwise try man dahdi_genconf |
| 15:24.29 | Greenlight | Or even Google... theres like a bazzilion howtos out there |
| 15:26.16 | slav3_kitten | looks pissed as he waits for a phone call |
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| 15:28.49 | Free99 | does this seem like a reasonable route? I'm setting calls by guests to require some kind of authentication (a pin), but I'd like to allow these same callers to have calls just go through if they've successfully registered with a softphone |
| 15:28.49 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.29) |
| 15:29.30 | igcewieling | If they are registered then they are not guests. |
| 15:29.46 | ruben231 | hi guys anyone help, i setup asterisk on amazon cloud all working but when i call i cant here any audio just ring once after complete silence even the call is answered |
| 15:30.16 | igcewieling | ruben231: sounds like a NAT or directmedia issue. |
| 15:30.39 | gboudreau | hi. my voip provider, voip.ms, did a maintenance on the server I was connected to last night, and it disconnected me. Asterisk didn't reconnect after the maintenance. I had to manually issue a 'reload' command in the asterisk command line to make it re-register. how can I configure asterisk to re-register when needed ? My sip.conf uses this: register => username:password@montreal.voip.ms:5060/5145551212 and sip |
| 15:30.40 | gboudreau | <PROTECTED> |
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| 15:31.22 | WIMPy | henny22: Just run it. Or copy one of the many examples google gives you. |
| 15:31.54 | [TK]D-Fender | gboudreau, "sip show peers has nothing to do with your registration |
| 15:32.03 | [TK]D-Fender | gboudreau, that is "sip show registry" |
| 15:32.31 | [TK]D-Fender | gboudreau, And it should re-register automatically at the frequency they request or if you set a different one under [general] |
| 15:34.13 | gboudreau | during their maintenance, I had this in my log: chan_sip.c: -- Registration for 'USERNAME@montreal.voip.ms' timed out, trying again (Attempt #6) |
| 15:34.23 | gboudreau | then: Forbidden - wrong password on authentication for REGISTER for 'username' to 'montreal.voip.ms' |
| 15:34.31 | gboudreau | which I guess was a result of their maintenance |
| 15:34.55 | gboudreau | and nothing after that, until I issued the reload command |
| 15:35.35 | gboudreau | I guess the 'wrong password' error stopped the retries |
| 15:35.56 | WIMPy | gboudreau: Indeed |
| 15:36.22 | *** part/#asterisk polysics (~Adium@host116-70-dynamic.31-79-r.retail.telecomitalia.it) |
| 15:36.35 | *** part/#asterisk ruben231 (~OpenDial@112.198.90.29) |
| 15:36.41 | gboudreau | any way I can tell asterisk to retry, or should I monitor for that kind of errors using an external script? |
| 15:37.37 | WIMPy | gboudreau: the situation has been discussed before, but so far I think you still have to do it yourself. |
| 15:38.47 | gboudreau | ok. would this be the best way to confirm that I'm registered: if [ `asterisk -rx "sip show registry" | grep voip.ms | awk '{print $5}'` != 'Registered' ]; then echo "Not registered"; fi |
| 15:38.51 | Free99 | Why has DeadAGI become deprecated? |
| 15:39.36 | WIMPy | gboudreau: looks ok. Using AMI would be a lot more elegant. |
| 15:40.09 | gboudreau | ok. googling for that. |
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| 15:45.23 | Rico29 | does anybody here use Siemens gigaset C610IP phones ? |
| 15:47.31 | igcewieling | gboudreau: Asterisk will re-register automatically. Did the IP address of montreal.voip.ms change? |
| 15:47.48 | igcewieling | Free99: because the functionality was merged with AGI() |
| 15:48.21 | igcewieling | Asterisk will NOT automatically update the IP associated with a hostname unless you enable dnsmgr |
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| 15:49.30 | *** join/#asterisk HolTech (~IceChat77@pool-74-107-104-2.bltmmd.fios.verizon.net) |
| 15:51.05 | HolTech | Looking into my first hardphone purchase to use with *, anyone have anything against the Digium D40 ? |
| 15:51.18 | lvlinux | price lol. |
| 15:51.55 | igcewieling | HolTech: The main issue with Digium phones is that they are fairly new and not many people use them as compared to phones like Polycom. So you'll not get a lot of support/help from the Asterisk community. |
| 15:52.40 | HolTech | igcewieling: Do the polycomm's offer the ability to make javascript / other app menus to integrate with * also ? |
| 15:52.46 | Free99 | so I set "allowguest=yes" but a client I'm trying to connect keeps getting rejected |
| 15:52.50 | igcewieling | HolTech: no. |
| 15:53.09 | igcewieling | Free99: you know what to do. |
| 15:53.29 | Free99 | I ran "reload" and "sip reload" |
| 15:53.53 | HolTech | how about a free softphone then... any of them allow good customization ? |
| 15:53.57 | igcewieling | HolTech: polycom phones support XHTML but not Javascript. The VVX series may support JS, but not the soundpoint IP ones. |
| 15:54.10 | igcewieling | Free99: I meant pastebin the CLI output |
| 15:54.15 | Free99 | gotcha |
| 15:54.17 | [TK]D-Fender | <Free99> so I set "allowguest=yes" but a client I'm trying to connect keeps getting rejected <- you should probably look at that... |
| 15:54.38 | lvlinux | HolTech---what exactly are you wanting to be able to do? |
| 15:55.12 | Free99 | igcewieling: http://pastebin.com/nzEfxqQ9 |
| 15:55.34 | igcewieling | HolTech: Doing custom phone apps is what I call a Blue Bells feature. You get all excited about the feature, think it is awesome, but eventually you realize it's not going to work how you want and you go home disapointed. |
| 15:55.47 | HolTech | Ideally integrate with a database, and build CRM type options, like flashing up a customer's info based on the incoming line... |
| 15:56.05 | Free99 | [tk] it's looking for a username that doesn't exist.. but I thought that's what guest access allowed for? |
| 15:56.14 | igcewieling | Free99: You can't have a phone register and expect it to use guest |
| 15:56.35 | igcewieling | stop the phone from registering and you will likely start matching guest. |
| 15:56.37 | HolTech | lol, sounds like what I want to avoid... Im still new to * and don't want to drop a few hundred bucks on a phone that wont do much... so thinking about a cheaper solution while i learn and develop |
| 15:56.39 | Free99 | so what does "guest" actually mean? |
| 15:56.39 | Katty | so. |
| 15:56.44 | Katty | AirVenture. Oshkosh. |
| 15:56.47 | Katty | who's going? |
| 15:56.55 | Katty | i know some of you fly. |
| 15:56.56 | igcewieling | Free99: guest = UNAUTHENTICATED |
| 15:56.59 | Free99 | katty: skydiving? |
| 15:57.10 | Katty | not exactly ;) |
| 15:57.47 | Free99 | katty: oh snap, that's pretty awesome |
| 15:58.01 | lvlinux | HolTech: get on eBay. Pickup a Polycom 600 for $50 or so. |
| 15:58.23 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 15:58.32 | lvlinux | or several different phones (cheap). Compare them and see what you like. |
| 15:59.08 | lvlinux | then when you have a good feel for everything you will know what features you really want/need. |
| 15:59.35 | *** join/#asterisk sawgood (~sawgood@23-24-214-61-static.hfc.comcastbusiness.net) |
| 15:59.37 | lvlinux | If you get the 600 you can experiment with your database nightmare, I mean stuff. |
| 15:59.43 | lvlinux | ;-) |
| 15:59.58 | igcewieling | Free99: guest calls are so you can allow random phones from random people on the internet to call you. This is a VERY dangerous feature and will result in fraud if you are not careful |
| 16:00.55 | Free99 | igcewieling: I'm aware. Authentication is processed on the phone line.. a2billing |
| 16:00.58 | HolTech | lvlinux: thanks, nightmare sounds about right ... but nothing ventured nothing gained right. I watched some vids on * homepage about digium phones needed to be freely registered before working with *, is this correct , and for all phones ? |
| 16:01.03 | Free99 | still going to filter extensions and such though |
| 16:01.16 | igcewieling | HolTech: http://supportdocs.polycom.com/PolycomService/support/global/documents/support/setup_maintenance/products/voice/Developers_Guide_UCS_4_0_1.pdf |
| 16:01.45 | [TK]D-Fender | <Free99> [tk] it's looking for a username that doesn't exist.. but I thought that's what guest access allowed for? <- that allows guests to CALL. |
| 16:01.47 | lvlinux | HolTech: to be honest I've never had a Digium phone, so I don't know much about them. |
| 16:01.48 | *** join/#asterisk lodac (~lodac-h3@71-83-12-142.static.aldl.mi.charter.com) |
| 16:01.51 | [TK]D-Fender | Free99, 9REGISTRATIOn is another matter. |
| 16:02.15 | [TK]D-Fender | Free99, There is another option to allow that to happen.... |
| 16:02.25 | igcewieling | HolTech: pay SPECIAL attention because many features described in the guide only work on the newer VVX phones. |
| 16:02.47 | lvlinux | HolTech: but if you get the Polycom 600, it does have browsing features and all that stuff, so you should be able to experiment quite a bit with what you are wanting to do. |
| 16:03.20 | lvlinux | like igcewieling says you may not have EVERY feature, but you've only spent $50 or so. |
| 16:03.28 | [TK]D-Fender | HolTech, IP600 is very old, but does have a very modest XHTML micro-browser which is good for interesting minor stuff |
| 16:03.46 | lvlinux | yes |
| 16:03.48 | igcewieling | in my experience the soundpoint IP phones support for "web applications" REALLY REALLY sucks, but I'm not a web developer. |
| 16:03.58 | Free99 | [TK]: is the option in sip.conf? |
| 16:04.09 | HolTech | Thanks for the info guys, exactly the help I needed. |
| 16:04.20 | igcewieling | HolTech: VoIP is expensive. Accept it. |
| 16:04.40 | [TK]D-Fender | Free99, Yes |
| 16:04.46 | lvlinux | Wellllll, depends... |
| 16:05.08 | HolTech | igcewieling: definitely, more expensive when you buy the wrong gear first and then buy the right gear second... |
| 16:05.19 | Weezey | Expensive hardware and time consuming to learn, but monthly costs are awesome and you can make it profitable. |
| 16:05.21 | igcewieling | HolTech: that will happen anyway 8-| |
| 16:07.02 | *** part/#asterisk gboudreau (~gboudreau@irc.pommepause.com) |
| 16:07.45 | *** join/#asterisk vlad_sta_ (~vlad_star@87.117.139.75) |
| 16:10.23 | *** join/#asterisk vlad_sta_ (~vlad_star@91.206.59.134) |
| 16:18.23 | Free99 | hey [tk]d-fender: is it alwaysauthreject? Based on the description, it doesn't seem like that's right |
| 16:19.02 | [TK]D-Fender | Free99, autocreatepeer <------------ |
| 16:20.41 | Free99 | that's in sip.conf/ |
| 16:20.44 | Free99 | ? |
| 16:24.12 | Free99 | [tk]: * documentation says that's in asterisk 11 and 12 |
| 16:24.55 | [TK]D-Fender | Free99, And in branches LONG before those |
| 16:26.20 | Free99 | while on the topic, [tk] is there a good reason to upgrade from 1.8? |
| 16:27.19 | [TK]D-Fender | Free99, Have you looked what they added? |
| 16:27.34 | [TK]D-Fender | Free99, That's 2 entire branches newer.... |
| 16:28.01 | Free99 | I see some neat features, but I dunno if I need any. Basic phone system, you know? I like the glare avoidance though |
| 16:28.43 | [TK]D-Fender | Free99, T.38 termination is kinda big. |
| 16:29.46 | Free99 | hmm. Doesn't T.38 refer to faxing? |
| 16:29.52 | Free99 | lemme google it :) |
| 16:30.33 | Free99 | oh neat. So in 1.8 it was only a patch, but integrated into 10? |
| 16:31.47 | *** join/#asterisk krotos (~root@devapp.as48500.net) |
| 16:31.50 | krotos | Hi all guy |
| 16:32.15 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 16:33.58 | krotos | on asterisk 1.8X or major there is support for ZRTP or crypted audio? |
| 16:34.19 | Qwell | SRTP |
| 16:34.45 | Qwell | https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial |
| 16:35.18 | [TK]D-Fender | Free99, and more. |
| 16:35.29 | krotos | Qwell: thankyou :) |
| 16:38.16 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 16:38.26 | *** part/#asterisk fafaflofly (~ircap@static-173-87-78-218.sdsl01.roch.ny.frontiernet.net) |
| 16:39.30 | *** join/#asterisk sjs205 (~sjs205@host81-151-44-189.range81-151.btcentralplus.com) |
| 16:41.08 | Free99 | [tk] I'll have to look at that in the near future. For right now though, I cannot find an option regarding "autocreatepeer" |
| 16:43.16 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
| 16:44.38 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
| 16:48.44 | [TK]D-Fender | Free99, it's there |
| 16:49.40 | [TK]D-Fender | Free99, Set to "yes" under [general] and go look |
| 16:49.56 | Free99 | grep -i autocreatepeer /etc/asterisk/* |
| 16:49.59 | Free99 | returns nothing |
| 16:51.37 | mjordan | Unfortunately, it's there. |
| 16:52.09 | *** join/#asterisk linocisco (~linocisco@203.81.72.87) |
| 16:52.40 | linocisco | hi all, I have problem registering cisco 7942G with my asterisk. it is updated into sip already |
| 16:52.50 | mjordan | it isn't in sip.conf.default however. It is, however, a boolean option available - set autocreatepeer=yes and chan_sip will start making peers for you auto-magically. Hilarity shall ensue. |
| 16:53.03 | lvlinux | linocisco: what is it doing/not doing? |
| 16:54.30 | linocisco | lvlinux, it is saying registering for long time |
| 16:55.12 | lvlinux | linocisco: what is asterisk showing is happening? |
| 16:55.26 | linocisco | lvlinux, how can I check? |
| 16:56.10 | [TK]D-Fender | linocisco, "sip set debug on" <- |
| 16:56.27 | lvlinux | run "asterisk -rvvvvvvvvvvvvvvvvvvvv" and see what happens when you power up the phone |
| 16:57.09 | [TK]D-Fender | linocisco, "sip set debug on" <- |
| 16:57.27 | linocisco | [TK]D-Fender, i did and waiting to see msg |
| 16:57.51 | [TK]D-Fender | linocisco, And if you've restarted your phone and see nothing, you either misconfigured your phone, or your have a routing/firewall issue |
| 16:58.29 | linocisco | [TK]D-Fender, i have other ATA phones working. 3CX softphone working. why only this? |
| 16:58.51 | lvlinux | linocisco: hardphones are different sometimes. |
| 16:59.30 | lvlinux | linocisco: do you have "qualify=no" in your sip.conf with the phone entry? |
| 16:59.38 | *** join/#asterisk alhemicar (~alhemicar@178.77.23.208) |
| 16:59.41 | *** join/#asterisk vlad_starkov (~vlad_star@87.117.139.75) |
| 16:59.57 | lvlinux | linocisco: some Cisco phones seem to not like being "qualified" :-) |
| 17:00.01 | linocisco | lvlinux, i did qualify=yes |
| 17:00.22 | lvlinux | linocisco: try changing it to no and see if that makes a difference. |
| 17:00.40 | lvlinux | linocisco: are you still not seeing anything in your console? |
| 17:01.21 | linocisco | lvlinux, i did sip reload to affect changes |
| 17:01.37 | lvlinux | ok good. |
| 17:01.50 | [TK]D-Fender | linocisco, And if you've restarted your phone and see nothing, you either misconfigured your phone, or your have a routing/firewall issue |
| 17:01.58 | lvlinux | your phone has an appropriate ip? |
| 17:02.18 | linocisco | lvlinux, yes. through DHCP from router |
| 17:02.25 | lvlinux | can you ping it? |
| 17:02.42 | linocisco | lvlinux, ubuntuserver*CLI> sip set debug on |
| 17:02.43 | linocisco | SIP Debugging enabled |
| 17:02.43 | linocisco | <--- SIP read from UDP:192.168.1.10:62517 ---> |
| 17:02.43 | linocisco | <-------------> |
| 17:02.43 | linocisco | Really destroying SIP dialog '4fa8cb325f1f54d310aec15266b51e7e@192.168.1.8:5060' Method: NOTIFY |
| 17:02.43 | linocisco | <--- SIP read from UDP:192.168.1.10:62517 ---> |
| 17:02.44 | linocisco | <-------------> |
| 17:02.51 | *** join/#asterisk DennisG (~DennisG@82-170-131-186.ip.telfort.nl) |
| 17:04.22 | lvlinux | linocisco: that's it? |
| 17:06.09 | linocisco | lvlinux, what do u mean? |
| 17:06.13 | [TK]D-Fender | linocisco, PASTEBIN, never flood in here. |
| 17:06.30 | [TK]D-Fender | linocisco, If you don't see traffic from your phone, then it is as I have already described it to you |
| 17:06.34 | [TK]D-Fender | ~pb |
| 17:06.34 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:10.51 | lvlinux | then post the link to your pastebin here |
| 17:11.37 | *** join/#asterisk vlad_starkov (~vlad_star@87.117.139.75) |
| 17:12.21 | linocisco | ok |
| 17:12.38 | lvlinux | linocisco: when you have sip debug on and you power up your phone, you should see several SIP messages in your console after the phone boots. |
| 17:14.25 | linocisco | lvlinux, yes |
| 17:15.15 | lvlinux | ok well pastebin all those messages |
| 17:16.39 | linocisco | lvlinux, i removed all ATA and soft phones |
| 17:16.55 | linocisco | lvlinux, to pin point only cisco phone message |
| 17:17.20 | *** join/#asterisk MalMen (~MalMen@bl6-157-138.dsl.telepac.pt) |
| 17:17.26 | MalMen | hi |
| 17:18.09 | lvlinux | linocisco: good |
| 17:18.16 | lvlinux | MalMen: hi |
| 17:18.25 | MalMen | :) |
| 17:20.55 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 17:21.35 | lvlinux | linocisco: so you are seeing messages from the phone? or no? |
| 17:22.33 | linocisco | lvlinux, no more msg |
| 17:23.09 | linocisco | lvlinux, i will turn off and on again |
| 17:23.09 | lvlinux | ok. did you configure your phone on the phone itself or did you use a config file and the phone is downloading it? |
| 17:23.42 | *** join/#asterisk ujjain2 (~ujjain@unaffiliated/ujjain) |
| 17:24.45 | linocisco | lvlinux, i used config file using TFTP |
| 17:25.10 | lvlinux | ok pastebin your config file and let me take a look at it |
| 17:25.38 | lvlinux | 192.168.1.8 is your * server? |
| 17:25.48 | *** part/#asterisk HolTech (~IceChat77@pool-74-107-104-2.bltmmd.fios.verizon.net) |
| 17:27.18 | linocisco | lvlinux, ok |
| 17:29.22 | *** part/#asterisk alhemicar (~alhemicar@178.77.23.208) |
| 17:29.55 | linocisco | lvlinux, http://pastebin.com/BDC0xend |
| 17:30.29 | lvlinux | k |
| 17:31.10 | *** join/#asterisk navaismo (~navaismo@189.191.250.167) |
| 17:33.37 | lvlinux | linocisco: i thought you unplugged your ATA? |
| 17:34.10 | linocisco | lvlinux, yes. it is nothing to do with cisco phone we are trying to pinpoint |
| 17:34.44 | sawgood | I still use ATA's (yeah)! |
| 17:36.16 | lvlinux | linocisco: uhh, well your pastebin shows multiple messages from 192.168.1.102, which is a Grandstream HT702? |
| 17:36.39 | lvlinux | what is the IP of your cisco phone? |
| 17:38.00 | linocisco | 192.168.1.100 |
| 17:38.59 | lvlinux | ok ---there are no messages from that IP that I see in your pastebin. Did you pastebin the phone config file for me? |
| 17:41.17 | *** join/#asterisk DennisG (~DennisG@82-170-131-186.ip.telfort.nl) |
| 17:42.35 | linocisco | lvlinux, xml file or conf file? |
| 17:44.59 | linocisco | lvlinux, http://pastebin.com/XvUT6pxK |
| 17:46.57 | lvlinux | ok |
| 17:47.10 | linocisco | lvlinux, http://pastebin.com/AmqraDxq |
| 17:47.47 | lvlinux | are you able to ping the phone from the * server? |
| 17:48.11 | linocisco | lvlinux, yes. it was |
| 17:49.12 | lvlinux | ?? |
| 17:49.35 | lvlinux | oh well |
| 17:51.01 | Free99 | hmm. I'm trying to allow people onto my system to authenticate and make calls, but I'd like people who have successfully registered as an actual user to be placed into a different context |
| 17:51.10 | Free99 | that doesn't seem to be happening for some reason.. |
| 17:52.37 | *** join/#asterisk iprouteth0 (~james@unaffiliated/iprouteth0) |
| 17:54.51 | Free99 | is that expected behavior if "autocreatepeer" is enabled? |
| 17:56.57 | jagster` | If i want to know what happens to a transferred call in cdr |
| 17:57.02 | jagster` | what should i be grepping for? |
| 17:58.18 | *** join/#asterisk darkdrgn2k (~darkdrgn2@209.90.253.66) |
| 17:58.20 | darkdrgn2k | Hi all |
| 17:58.59 | darkdrgn2k | Can some one take a look at this SIP trace for me and give me a clue why its not working? |
| 17:58.59 | darkdrgn2k | http://pastebin.com/RkGKEdmg |
| 17:59.12 | darkdrgn2k | seems t.38 is suggested but rejected. |
| 18:00.19 | *** join/#asterisk camerin (hoax@newelite2.bshellz.net) |
| 18:02.39 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 18:03.04 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 18:26.39 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.139) |
| 18:29.59 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.139) |
| 18:40.41 | *** join/#asterisk jkroon (~jkroon@kerberos.uls.co.za) |
| 18:42.08 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 18:43.22 | *** join/#asterisk [Outcast] (~anonymous@50-200-130-22-static.hfc.comcastbusiness.net) |
| 18:43.56 | [Outcast] | is there away to get SDP information via Dialplan variables? |
| 18:44.13 | jkroon | hi all, if i'd like to tamper with res_rtp_asterisk.c to produce more usable RTCP reports, amongst other ideas, the feature idea from https://issues.asterisk.org/jira/browse/ASTERISK-20754 - but also splitting the sender and recipient AMI events into two (this immediately allows for multiple RCs which is not currently possible) |
| 18:44.14 | LieutPants | [ASTERISK-20754] [Status: Open] rtp_engine's RTCP{Sent,Received} events should contain the Channel name - https://issues.asterisk.org/jira/browse/ASTERISK-20754 |
| 18:45.04 | jkroon | also, I got this from rtcp debug output, which worries me a *lot*: |
| 18:45.06 | jkroon | res_rtp_asterisk.c: Internal RTCP NTP clock skew detected: lsr=180646927, now=180678976, dlsr=32113 (0:490ms), diff=64 |
| 18:45.30 | jkroon | essentially the reported values results in a <0 round-trip time, which is not possibe. |
| 18:46.18 | jkroon | or the remote ends's clock is running fast? |
| 18:51.46 | jkroon | but a clock running fast by 64ms over 32s seems a tad steep... |
| 18:52.37 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 18:53.03 | jkroon | granted it's only .2 % too fast, but still. |
| 18:55.12 | *** join/#asterisk bretolius (~boot@ks393582.kimsufi.com) |
| 18:56.06 | *** join/#asterisk _Corey_ (~corey@64.215.11.114) |
| 19:00.46 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 19:01.22 | *** join/#asterisk MarKsaitis (~MarKsaiti@2.217.210.210) |
| 19:08.13 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 19:12.52 | Free99 | I'm trying to ensure that autocreatepeers won't allow people to use numbers I've specifically created accounts for. Any suggestions? |
| 19:13.14 | Qwell | Don't use autocreatepeers |
| 19:13.24 | Qwell | seriously - just don't |
| 19:14.19 | Free99 | Qwell: issue is that my boss needs me to allow people to make calls through the system anonymously at first, they are authenticated on the line |
| 19:14.33 | Qwell | Incoming calls have nothing at all to do with registration. |
| 19:14.47 | Qwell | guest=yes (which is still silly, but manageable) |
| 19:14.48 | Free99 | well, outgoing |
| 19:15.07 | Qwell | You need anonymous people to be able to be called by Asterisk? |
| 19:15.19 | Free99 | no, just make calls |
| 19:15.22 | Qwell | Incoming calls have nothing at all to do with registration. |
| 19:15.47 | Free99 | qwell: incoming as in from the PSTN to *? |
| 19:15.58 | Qwell | Incoming from the anonymous locations. |
| 19:16.57 | Qwell | allowguest=yes, set a context for those guest users in [general]. Problem solved. |
| 19:19.38 | [TK]D-Fender | Free99, It shouldn't |
| 19:19.55 | [TK]D-Fender | Qwell, thing is they are registering which will fail. Not just placing calls. |
| 19:20.12 | Qwell | [TK]D-Fender: Why is some random person on the Internet going to register to his system? |
| 19:20.18 | Qwell | What if they register as 8500? |
| 19:20.31 | [TK]D-Fender | Qwell, He is attempting to fly before even leving the delivery room... running OpenSIPS w/ * with a shared DB, etc.... |
| 19:20.36 | [TK]D-Fender | Qwell, It's a ClusterFuck |
| 19:20.38 | Qwell | It's a horrible idea. Don't do it. |
| 19:22.01 | Free99 | look: I don't like this setup either. I'm expected to make client setup as simple as possible, e.g. allow people to simply type my SIP server address into their device, then authenticate over the wire... apparently at the cost of any semblance of stability or security |
| 19:22.31 | Free99 | i'm not even supposed to enable TLS on this at all, because it would negligibly increase bandwidth consumption |
| 19:22.37 | Qwell | It's not my job on the line. Have fun! |
| 19:22.51 | drmessano | Why do you even need to create a peer |
| 19:23.18 | mjordan | Free99: I would look for a solution that doesn't rely on autocreatepeer. Regardless of what the use case may be, you are opening up a world of pain with that setting. (The lack of documentation on it is really a gift) |
| 19:23.19 | drmessano | Oh I get it |
| 19:23.22 | Free99 | well then let me ask like this: what's a good way to allow people to just connect to the system then place a call? |
| 19:23.32 | Qwell | allowguest=yes, ffs |
| 19:23.46 | Free99 | mjordan: I had a feeling as much. I read the chapter on securit in the asterisk book by o'reilly |
| 19:24.02 | drmessano | If you want people to be able to connect to a system and place calls, without authentication.. Try a Trixbox ISO |
| 19:24.05 | drmessano | Gah, burn |
| 19:24.11 | Free99 | qwell: I tried that, but it didn't work |
| 19:24.11 | mjordan | oh snap |
| 19:24.13 | Qwell | drmessano: If you can find one. |
| 19:24.23 | Qwell | Free99: Then you did it wrong. What "didn't work"? |
| 19:25.28 | Free99 | Qwell: specifically the registration part. If the device didn't have to register, then my problem would be "solved"... but from what I understand of *'s architecture, I need to register to be able to do stuff |
| 19:25.38 | Free99 | hence the stupid autocreatepeers hack |
| 19:25.45 | drmessano | uh nope |
| 19:25.47 | Qwell | That's precisely the opposite of what I've been telling you for the last 10 minutes. |
| 19:25.57 | *** join/#asterisk derjanni (~derjanni@ip-178-202-27-28.unitymediagroup.de) |
| 19:26.03 | derjanni | good evening everybody! |
| 19:26.29 | Qwell | ignores and moves on |
| 19:26.35 | derjanni | I have setup asterisk on my Raspberry Pi and it is working. However I only get http 404 on the mini http server. What is the config login URL acutally with Asterisk 1.8? |
| 19:26.37 | Free99 | qwell: my bad man, I'm trying |
| 19:26.56 | derjanni | is using Asterisk 1.8.13.1~dfsg-3 |
| 19:26.57 | Qwell | derjanni: Asterisk has no configuration UI. |
| 19:27.16 | derjanni | qwell: what is the http server for then? |
| 19:27.24 | drmessano | http |
| 19:27.25 | Qwell | You'd have to ask the Raspberry Pi folks. |
| 19:27.37 | derjanni | ok. Can u recommend a good book on the pi? |
| 19:27.37 | Qwell | oh, that http server |
| 19:27.41 | derjanni | sry asterisk? |
| 19:27.44 | Qwell | ~book |
| 19:27.44 | 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 |
| 19:27.50 | derjanni | ok tanks |
| 19:28.03 | derjanni | but Asterisk 1.8 came with the "mini http" server running @ 8088 |
| 19:28.06 | drmessano | The http server is just that.. a minimal http server implementation |
| 19:28.06 | Qwell | it's for lots of things. Other software would need to actually take advantage of it. |
| 19:28.20 | [TK]D-Fender | <Qwell> allowguest=yes, ffs <- again it's the mere fact that it's trying to Register. allowguest only allows call through, but if Reg's don't get accepted then the client will never send/get calls |
| 19:28.57 | [TK]D-Fender | derjanni, * doesn't HAVE a web interface by default. |
| 19:29.02 | derjanni | ok thanks |
| 19:29.07 | Qwell | [TK]D-Fender: if I wanted to dial bob@bobspizza.com, I don't have to register. |
| 19:29.09 | [TK]D-Fender | derjanni, You're looking for something you don't have. |
| 19:29.14 | Qwell | He is bob. |
| 19:29.18 | [TK]D-Fender | Qwell, Yes, but his end IS trying to register |
| 19:29.28 | derjanni | last time I wrote a dialplan for sip is 5yrs ago |
| 19:29.30 | [TK]D-Fender | Qwell, And his end won't dial until it has |
| 19:29.34 | drmessano | CLIENT issue |
| 19:29.38 | [TK]D-Fender | derjanni, Catch-up time... |
| 19:29.45 | derjanni | if Im too fat + lazy to read the big guide - what tutorial can u recommend for dial plans? |
| 19:30.11 | mjordan | uhm |
| 19:30.12 | drmessano | http://www.trixbox.org |
| 19:30.16 | drmessano | sorry |
| 19:30.41 | mjordan | I'm trying to think of why you would deploy something as a registrar when it has no knowledge of what devices are registering with it |
| 19:30.57 | Free99 | mjordan: because in some instances, my boss is.. silly. |
| 19:31.12 | Free99 | like I said, he wants it to be "Easy to setup" |
| 19:31.18 | mjordan | easy to exploit |
| 19:31.18 | drmessano | Why wouldnt you just put in a transparent proxy |
| 19:31.23 | [TK]D-Fender | derjanni, same book... |
| 19:31.33 | [TK]D-Fender | derjanni, Dialplan hasn't really changed much |
| 19:31.36 | igcewieling | Free99: Asterisk is not easy to set up. |
| 19:31.43 | derjanni | will stop configuring * now and go for the book first #rtfm |
| 19:31.49 | mjordan | I'd punt on Asterisk being the registrar in that case and have it just act as an app server |
| 19:31.50 | Free99 | I'm not going to pass the buck here, though. |
| 19:31.57 | [TK]D-Fender | derjanni, Alternative is to flash that thing and install a GUI'd up version like IncrediblePi |
| 19:32.05 | drmessano | If you want RANDOM efftards to load up a SIP client, punch in your server address, and make calls.. why wouldn't you set up a simple proxy so they can throw calls at it all day long |
| 19:32.11 | igcewieling | Free99: don't worry too much, once your system is exploited and you get a huge phone bill your boss will change his mind. |
| 19:32.19 | Qwell | igcewieling: unlikely |
| 19:32.21 | derjanni | Time to take a * refresher now. |
| 19:32.27 | derjanni | Thank you all and have a nice weekend! |
| 19:32.28 | Qwell | igcewieling: It's never the fault of somebody like. |
| 19:32.29 | Free99 | that's what I tried originally, using openSIPS, had a lot of trouble doing that |
| 19:32.32 | derjanni | sry afternooon |
| 19:32.46 | Qwell | like that* |
| 19:33.20 | Free99 | igcewieling: I just don't want it to happen. So is it possible for a device to place calls without registering? |
| 19:33.33 | Free99 | that's device dependent I'm assuming |
| 19:33.56 | navaismo | Free99, yes its possible using allowguest=yes |
| 19:33.57 | igcewieling | Free99: Registration ONLY is related to OUTGOING calls FROM Asterisk TO the CLIENT. It has nothing whatsoever to do with incoming calls. |
| 19:34.09 | igcewieling | once you understand that your life will have less missery |
| 19:34.22 | Free99 | (facepalm) oh. |
| 19:34.46 | Qwell | igcewieling: That's been explained to him about 5 times now. |
| 19:34.53 | drmessano | Free99: Use the repro SIP proxy. The config is simple. People will be proxying calls before they know it |
| 19:35.08 | igcewieling | Qwell: and his lack of understanding is why I think he'll manage to configure a system easily exploited. |
| 19:35.22 | Qwell | You're preaching to the choir, man... |
| 19:35.43 | *** join/#asterisk RyanTG (~Thunderbi@mailer.americassuppliers.com) |
| 19:35.50 | drmessano | It sounds like the goal IS for it to be exploited lol |
| 19:36.14 | drmessano | [15:22] <Free99> look: I don't like this setup either. I'm expected to make client setup as simple as possible, e.g. allow people to simply type my SIP server address into their device, then authenticate over the wire... apparently at the cost of any semblance of stability or security |
| 19:36.29 | drmessano | Type in the address, make a call |
| 19:37.04 | Free99 | drmessano: thing is, I have an AGI application on the wire that asks for people to authenticate |
| 19:37.19 | Free99 | it doesn't let the call go through until that's done |
| 19:37.21 | drmessano | Where was that detail 20 mins ago |
| 19:37.44 | Free99 | actually, [tk]d-fender remembered it and mentioned it iirc |
| 19:37.57 | drmessano | [TK]D-Fender is not Qwell or igcewieling or drmessano |
| 19:37.59 | Free99 | when he was mentioning my trying to fly before leaving the delivery room |
| 19:38.10 | [TK]D-Fender | I don't recall anything about AGI.... |
| 19:38.15 | drmessano | lol... |
| 19:38.40 | Free99 | oh, nope, he didn't say that. I thought he mentioned a2billing |
| 19:38.43 | drmessano | throws more corn in the popper |
| 19:38.45 | Free99 | my bad |
| 19:39.23 | igcewieling | this is all pointless. |
| 19:39.25 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 19:39.55 | drmessano | Basically it sounds like they want people to make calls throw the server, create a peer in the process, give them some free calls, then beat them up into paying for service |
| 19:40.01 | drmessano | through* |
| 19:40.02 | [TK]D-Fender | The Tri-fuckta has come to light... |
| 19:40.29 | drmessano | Sounds illegal |
| 19:40.31 | Free99 | nah, you can't make a call at all unless you have an account on the system |
| 19:41.19 | Qwell | So, you're saying these people already have accounts? Why are you even talking about anonymous calls then? |
| 19:42.29 | Free99 | Qwell: I said as much to the jefe. "The devices should be able to just enter our server in and thats all the configuration required" |
| 19:42.51 | Free99 | meanwhile, a2billing already generates sip peers and secrets automatically |
| 19:43.05 | Free99 | (shrug) I don't even know. |
| 19:43.11 | drmessano | What devices ? |
| 19:43.12 | Qwell | Are these people on the Internet? |
| 19:43.17 | drmessano | Anything SIP? |
| 19:43.18 | [TK]D-Fender | Free99, and if that's the case.... then why are YOU having to have them created automatically just to match? |
| 19:44.03 | Free99 | Qwell: yes, they are on ships around the world, using our satellite internet system. drmessano: all SIP. [tk] I don't get what you mean |
| 19:44.14 | drmessano | Thats not what I asked |
| 19:44.37 | drmessano | Are the devices a SPECIFIC device or is this ANY SIP DEVICE, as in their CHOICE |
| 19:45.01 | drmessano | Because provisioning a random SIP device doesn't make sense |
| 19:45.02 | Free99 | drmessano: I think its going to be specific, but I don't know right now |
| 19:45.24 | drmessano | Are YOU selling them the device? |
| 19:46.18 | Free99 | drmessano: no idea. I guess I'll find out. For now we've been using linksys pap2t and these weird chinese voip phones |
| 19:46.36 | drmessano | Are you providing THOSE? |
| 19:46.52 | Free99 | let's assume yes. why do you ask? |
| 19:47.01 | drmessano | Why cant you just provision them? |
| 19:47.47 | [TK]D-Fender | Free99, It's a question of "are you in control of the endpoints so you don't have to deal with nonsense" |
| 19:48.21 | drmessano | Why is the customer going to access the UI on a device, punch in some crazy credentials that make no sense to them, and connect to some magical non-existant autoprovisioning system when you can just control the devices yourself |
| 19:49.02 | drmessano | Even if you ship them with no account set up, you can ship them pointing to the provisioning server for later configuration |
| 19:50.34 | drmessano | Relying on something within the confines of SIP peer configuration to provision a device is silly. Robust, proven provisioning methods exist |
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| 19:50.57 | Free99 | drmessano: I have no response to that. I'm not sure, but apparently we're going to just ship these phones along with our antenna systems and let the clients do the configuration or something |
| 19:50.57 | Free99 | not to mention on ships, people move around a lot |
| 19:50.58 | Free99 | so it doesn't make sense to tie an account to a stationary piece of hardware |
| 19:50.59 | Free99 | right? |
| 19:51.18 | drmessano | You're sending them the phones |
| 19:51.48 | drmessano | YOU control the configuration if they are provisioned remotely |
| 19:51.54 | drmessano | If something changes, push a new config |
| 19:52.37 | *** join/#asterisk alhemicar (~alhemicar@178.77.27.136) |
| 19:52.42 | drmessano | I still fail to see why the client even needs to touch the configuration |
| 19:56.05 | Free99 | That makes sense. So enlighten me, please: I have these phones which I provision centrally. Thing is, the people using the phones move around. Each has their own account. I don't need to worry about calls going *to* the phones, only from. |
| 19:57.17 | drmessano | Sounds like a calling card setup.. why wouldn't they key-in something before the call is connected? |
| 19:57.33 | Free99 | it potentially sounds like what you're saying is that I should make a "non-privileged" account for the hardware, which elevates its user once they authenticate |
| 19:57.43 | Free99 | drmessano: that's essentially what it is |
| 19:57.53 | drmessano | This has nothing to do with the account |
| 19:58.21 | drmessano | or even having an account |
| 19:58.54 | drmessano | You could use allowguest=yes and run the call through A2Billing and done |
| 19:59.21 | Free99 | drmessano: provided I can disable registration on the phones right? |
| 19:59.41 | drmessano | Instead of having a PSTN number that is dialed the phones would make a SIP call to the Asterisk box and the process would progress from there |
| 19:59.44 | drmessano | yes |
| 20:00.04 | drmessano | I know A2B has this functionality already |
| 20:00.12 | Free99 | that's been the goal this whole time |
| 20:00.40 | drmessano | So you provision each phone to make an outbound call to the server when the handset is taken off-hook |
| 20:01.05 | drmessano | Then they have to authenticate by keying in calling card info, and they can then make a call |
| 20:01.44 | Free99 | bingo. Got all the other stuff working, except for the registration issue |
| 20:01.59 | drmessano | Which isn't an issue |
| 20:02.02 | drmessano | Soooo |
| 20:02.27 | Free99 | well, just out of curiosity, what if I can't disable the stupid phone's registration thing? |
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| 20:02.54 | drmessano | I haven't seen a device that doesn't have a register yes/no setting |
| 20:03.02 | drmessano | Not every device registers |
| 20:03.09 | drmessano | Let me rephrase |
| 20:03.30 | drmessano | Not every use case requires registration. Its optional |
| 20:04.15 | drmessano | I can tell you right now the PAP2T will make a call without registration, when the handset is lifted up, to a SIP address of your choosing |
| 20:04.19 | drmessano | Because I do it |
| 20:05.21 | Free99 | yeah, I just got that working right now, (S0<:123>) |
| 20:05.31 | Free99 | but this chinese phone... lol |
| 20:05.36 | drmessano | yealink? |
| 20:05.46 | Free99 | planet |
| 20:05.52 | drmessano | I have no clue |
| 20:08.36 | Free99 | http://www.planet.com.tw/en/product/product.php?id=47882 <-- avoide |
| 20:08.38 | Free99 | *avoid |
| 20:09.15 | drmessano | Oh, its wireless |
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| 20:09.37 | Free99 | it's pretty cool, but some things are weird about it |
| 20:09.55 | Free99 | I found an option titled "anonymous calls" but it still tried to register |
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| 20:13.14 | Captain_Proton | ? I having problem with ATT flex One thing I was going to try is have asterisk control the timing. how would I accomplish this? |
| 20:15.21 | Captain_Proton | does signalling still apply ? |
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| 20:22.52 | [TK]D-Fender | Captain_Proton, CPE never controls timing in big-telco scenarios |
| 20:23.01 | [TK]D-Fender | Captain_Proton, They are master 99.999999999999999999% of the time |
| 20:23.21 | [TK]D-Fender | Captain_Proton, And they're firing the last tech who got it backwards today :) |
| 20:25.22 | Captain_Proton | [TK]D-Fender: Yep that the way it is set now. ATT made the suggestion that if asterisk is capable that I take control of time to see if it help with the echo problem we are having. |
| 20:27.16 | themrrobert | So there is another server making AMI calls, is it possible for AMI commands to cause temporary locks that stop call traffic? |
| 20:27.28 | drmessano | Echo? Not hardly |
| 20:29.39 | Free99 | hey drmessano, what's a good way to provision the linksys pap2? |
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| 20:44.40 | *** part/#asterisk InternetApe (~intenetap@ip68-99-245-46.ph.ph.cox.net) |
| 20:46.06 | drmessano | Free99: HTTP via XML file |
| 20:46.29 | AkkerKid | when i create a callfile for asterisk that includes the line "Setvar: CDR(column)=data" that data doesn't get into that column. Do I need quotes on something? |
| 20:47.55 | themrrobert | this problem seems different, no "Exceptionally long voice queue length errors", and it was a longer length (9 seconds) and it was slient without a spike (this makes me thing network, idk about you) http://pastebin.com/LwvtmTag |
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| 21:02.36 | Free99 | thanks drmessano. |
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| 21:35.27 | blackvipe | hey question |
| 21:35.38 | blackvipe | Cisco 7960g phone on asterisk |
| 21:35.53 | ruben231 | hi guys anyone cna suggest a voicemail interface to access for an asterisk install..? |
| 21:36.03 | lvlinux | blackvipe: yes? |
| 21:36.09 | ruben231 | any web interfaces to manage voicemail |
| 21:36.12 | blackvipe | I am tring to get the logo to work |
| 21:36.23 | blackvipe | but there is talk about a file called SIPDefault.cnt |
| 21:36.30 | blackvipe | but isn't that SIPDefault.cnf? |
| 21:36.36 | lvlinux | yes |
| 21:36.39 | blackvipe | oh ok |
| 21:36.50 | blackvipe | tring to figure out why it is not showing |
| 21:37.00 | blackvipe | it's 90x50 |
| 21:37.07 | blackvipe | hmmm |
| 21:37.16 | lvlinux | try w a known working file first |
| 21:37.29 | blackvipe | yea thats true |
| 21:37.33 | blackvipe | hey one other question |
| 21:37.43 | blackvipe | ya ever worked with the Cisco ATA-187? |
| 21:37.59 | lvlinux | no just the phones |
| 21:38.01 | blackvipe | I am so close to getting that to work with asterisk is not funny |
| 21:38.05 | blackvipe | I can recieve calls |
| 21:38.11 | blackvipe | but when I go to send a call |
| 21:38.27 | blackvipe | it is not letting me dial all the numbers enter just the first one |
| 21:38.31 | blackvipe | kinda weird |
| 21:38.51 | lvlinux | does it put the call through with the one number? |
| 21:38.58 | lvlinux | or error? |
| 21:39.17 | blackvipe | yep |
| 21:39.26 | lvlinux | yep which? |
| 21:39.35 | blackvipe | puts the call threw |
| 21:39.41 | blackvipe | I can see it sending the number |
| 21:40.00 | blackvipe | it's a dialplan issue in the aTA*.cnf file |
| 21:40.02 | lvlinux | is it SIP or SCCP? |
| 21:40.10 | blackvipe | I just don't know the xml var |
| 21:40.12 | blackvipe | sip |
| 21:40.31 | lvlinux | pastebin ur cnf file |
| 21:42.38 | igcewieling | blackvipe: you have an issue with the DIALPLAN on the ATA |
| 21:42.49 | blackvipe | yep |
| 21:42.55 | lvlinux | that's what he said |
| 21:42.59 | blackvipe | it's not in asterisk thought |
| 21:43.06 | blackvipe | it's in the ATA itself |
| 21:43.11 | igcewieling | correct |
| 21:44.48 | navaismo | ruben231, nope not me, only tried the freepbx ari and so far is good but needs freepbx as far i now. You can build one simple using php |
| 21:45.09 | blackvipe | http://pastebin.com/FgcUv17k# |
| 21:45.31 | blackvipe | there is very little information on that ATA |
| 21:45.45 | blackvipe | unlike the ATA-188 and ATA-186 |
| 21:46.01 | ruben231 | navaismo: yeah dont ahve knowledge with php, i trying my luck fro so made by community already somehow |
| 21:47.58 | blackvipe | I have done a tcpdump to see what files it was searching for |
| 21:48.01 | blackvipe | and there are only 2 |
| 21:48.14 | blackvipe | the ATA*.cnf is the main one |
| 21:48.34 | blackvipe | the other is XMLDefault.cnf |
| 21:48.55 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-bqppefxqlyuzaxhi) |
| 21:49.21 | blackvipe | it totally ignores downloading dailplan.xml |
| 21:49.26 | blackvipe | or cnf |
| 21:49.45 | lvlinux | ok what tftp server are you using? |
| 21:49.51 | blackvipe | linux |
| 21:49.54 | blackvipe | centos |
| 21:50.13 | blackvipe | I also turned up the loging |
| 21:50.16 | blackvipe | only 2 times |
| 21:50.34 | lvlinux | ok check your logs and see exactly which named files the device is asking for. |
| 21:51.06 | blackvipe | ATAD4D748405385.cnf.xml |
| 21:51.12 | blackvipe | CTLSEPD74840538501.tlv |
| 21:51.18 | lvlinux | and then go out and buy an ata from a different brand lol |
| 21:51.19 | blackvipe | and the tlv is a secure file |
| 21:51.24 | blackvipe | hahahah |
| 21:51.34 | lvlinux | that's all it tried to download? |
| 21:51.39 | blackvipe | yep |
| 21:51.45 | lvlinux | hmm, that's odd |
| 21:52.04 | blackvipe | it's a odd ATA what can I say hahahha |
| 21:52.36 | blackvipe | opps and XMLDefault.cnf.xml |
| 21:52.48 | blackvipe | so it takes those 3 files |
| 21:53.14 | lvlinux | k |
| 21:53.42 | blackvipe | I have another ATA coming in but just currous to get this one working |
| 21:53.53 | blackvipe | if not thats ok |
| 21:54.18 | lvlinux | does it have a web interface or tftp only? |
| 21:54.24 | blackvipe | but really need to get my hands on the actual file from CUCM to see the dial plan |
| 21:54.45 | blackvipe | yep but everything is configured threw tftp |
| 21:54.47 | blackvipe | not web |
| 21:55.29 | lvlinux | so it's just a status page then. |
| 21:55.36 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:55.46 | blackvipe | correct with some details |
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| 21:58.13 | navaismo | ruben231, If I recall there was an option called webvmail when you run make command and that create a web page or so but not sure a long long time ago I used that |
| 21:58.45 | blackvipe | its kinda cool I even got this far with it |
| 21:58.46 | blackvipe | hahahh |
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| 22:30.42 | danfromuk | Hi, Regarding the READ command, is it possible to reduce the timeout time once the first digit has been entered? IE it waits 20 seconds for the first digit and then waits a maximum of 3 seconds for subsequent digits ? |
| 22:31.15 | navaismo | maybe editing the source code |
| 22:32.16 | WIMPy | core show function TIMEOUT |
| 22:34.29 | danfromuk | So setting TIMEOUT using this function will only be used once the first digit has been entered and not before? |
| 22:35.28 | WIMPy | I have no idea if the first digit is special as in WaitExten. |
| 22:36.13 | danfromuk | Ok. I'll give it a try. Does TIMEOUT also apply to the READ app? |
| 22:36.21 | danfromuk | Or jsut readexten? |
| 22:36.54 | WIMPy | I think the digit version would apply. But so far the defaults have been ok for me. |
| 22:40.29 | danfromuk | TIMEOUT(digits) didnt affect the READ app |
| 22:40.48 | [TK]D-Fender | danfromuk: Make an IVR for it then |
| 22:41.40 | danfromuk | The client wants to have single digit menu options and Dial By Extension on the same prompt. |
| 22:42.40 | WIMPy | Sounds like the ideal application for WaitExten. |
| 22:42.41 | [TK]D-Fender | danfromuk: Then I don't see why you'd use Read at all |
| 22:43.18 | danfromuk | [TK]D-Fender: Old code which used Read. |
| 22:43.31 | [TK]D-Fender | danfromuk: Replace with new code that doesn't :) |
| 22:44.48 | danfromuk | So waitexten uses the seconds parameter initially, and then TIMEOUT(digits) after the first digit's been entered? |
| 22:45.33 | [TK]D-Fender | danfromuk: just use TIMOUT for both |
| 22:45.38 | [TK]D-Fender | digit / response |
| 22:46.09 | danfromuk | Ok, thanks. Thats perfect. |
| 22:47.45 | WIMPy | The parameter is the timeout until the first digit is entered. |
| 22:49.54 | danfromuk | What happens to the timeouts if there are two extensions that could match the digits entered? eg. if the user enters a '2' and extensions '2' and '201' exist. |
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| 22:55.59 | WIMPy | You have to continue entering digits or wait for the timeout. |
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| 22:57.40 | danfromuk | Ok, thanks. |
| 22:58.04 | WIMPy | Just the normal dialplan behaviour. |
| 23:05.54 | danfromuk | Thanks for your help. I'll give it a try tomorrow. |
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