| 00:01.34 | LemensTS | Anyone alive |
| 00:12.14 | *** join/#asterisk coppice (~chatzilla@m180-219-200-59.smartone-vodafone.com) |
| 00:13.19 | CRCinAU_ | nope |
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| 00:51.46 | pierrelrq | hello |
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| 01:29.10 | recluze | Hi, I'm getting a REJECTED in the sip show registry. http://pastebin.com/W1tDWJmA |
| 01:31.10 | *** join/#asterisk recluze (~recluze@203.82.94.91) |
| 01:31.20 | recluze | Hi, I'm getting a REJECTED in my sip show registry |
| 01:31.34 | recluze | full log shows a 404 corresponding to this register string |
| 01:31.34 | recluze | http://pastebin.com/W1tDWJmA |
| 01:32.02 | recluze | any hints (newbie-level) for debugging this |
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| 01:51.30 | Smirker | hey hey |
| 01:52.07 | Smirker | can i confirm that if i use the default settings in sip.conf for udpbindaddress(=0.0.0.0) and allowguest(=yes), anyone can make calls through my box? |
| 01:52.18 | Smirker | as long as i'm not NAT'd ? |
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| 01:53.22 | WIMPy | yes |
| 01:54.40 | Smirker | i'd like to propose setting the default value of allowguest to no. |
| 01:55.31 | WIMPy | The idea is that you can use it without configuration. |
| 01:56.14 | WIMPy | And you certainly shouldn't put any service up to the public internet that you haven't configured with the appropriate care. |
| 01:57.45 | Smirker | ah well. it's configured now. |
| 01:57.49 | Smirker | 2,500 international calls later. |
| 01:59.30 | WIMPy | You certainly made additional mistakes when laying out your dialplan contexts then. |
| 02:00.36 | Smirker | contracted someone to do it. sigh |
| 02:01.31 | WIMPy | At least it was only international. |
| 02:02.26 | Smirker | do you mind mentioning what possible additional mistakes would have led to this? |
| 02:03.24 | WIMPy | Allowing outbound calls in the default context. Do you know the book? |
| 02:03.26 | WIMPy | ~newbook |
| 02:03.27 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 02:03.52 | WIMPy | There's a secutity sectiaon as well, IIRC. |
| 02:04.05 | Smirker | ah okay, thanks. i'll take a look at it. |
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| 02:34.42 | a_m_y | hi, may I know the exact dependencis for asterisk 1.4 and for 64bit, tnx in advance |
| 02:35.20 | a_m_y | hi, may I know the exact dependencis for asterisk 1.4 and for centos and ubuntu both 64bit, tnx in advance |
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| 02:52.29 | maxagaz | hi |
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| 03:08.21 | maxagaz | my asterisk phone call end after 30 min, has someone faced this problem already ? |
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| 05:02.06 | shamelessn00b | ~itsplist-usa |
| 05:02.12 | shamelessn00b | ~itsp |
| 05:02.13 | 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. |
| 05:02.13 | shamelessn00b | Is the bot down? |
| 05:02.20 | shamelessn00b | ~itsplist-us |
| 05:02.20 | 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 |
| 05:20.08 | *** join/#asterisk pepesmith (~pepesmith@unaffiliated/pepesmith) |
| 05:20.20 | pepesmith | i have a question |
| 05:20.44 | pepesmith | if i reload the sipconfig while there is a current call, will the current call be terminated? |
| 05:20.48 | pepesmith | good to to you all |
| 05:21.39 | CRCinAU_ | pepesmith: not unless you restart everything |
| 05:21.45 | CRCinAU_ | it only updates the config |
| 05:22.02 | pepesmith | thanks CRCinAU_ |
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| 06:58.43 | schmidts | good morning |
| 07:05.01 | ChannelZ | and goodnight |
| 07:05.02 | ChannelZ | runs off to bed |
| 07:09.16 | vfabi | Hi all, |
| 07:09.16 | vfabi | Is it possible to get channel variables such as ANSWEREDTIME and HANGUPCAUSE after caller hangup ? |
| 07:09.16 | vfabi | By default after call hangup by "caller" it go to h extension and all channel variables erased. |
| 07:09.16 | vfabi | If 'dial' F flag used - same, all variables erased. |
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| 07:36.07 | wdoekes2 | morning |
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| 07:51.45 | t3hrealadamd | Good morning, looking for some advice. New to asterisk; I have two Polycom IP 500's set up on an internal LAN talking to an external Asterisk server. Both have NAT=yes, and bindport set to different ports. However, when the phones come up and register, they show their respective extensions on the phones themselves, but on the server, they're both showing the same port (5060), and the one phone has the wrong username. St |
| 07:51.47 | t3hrealadamd | ranger still to me, one phone calls the other phone just fine, but when you call in vice versa, the phone calls itself. |
| 07:51.56 | t3hrealadamd | I can't figure out what to check; everything seems squared away. |
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| 07:54.14 | wdoekes2 | smells like a rogue ALG |
| 07:54.39 | wdoekes2 | the NAT router should choose a different port anyway, so you wouldn't need to tweak the bindport |
| 07:54.51 | wdoekes2 | but if it comes out the same, something fishy is going on |
| 07:55.10 | t3hrealadamd | even without the bindport, it's still sticking in "sip show peers" |
| 07:55.40 | wdoekes2 | do you have external ip set in the phone? don't |
| 07:55.45 | t3hrealadamd | No |
| 07:55.59 | wdoekes2 | do a 'sip set debug on' on the asterisk |
| 07:56.11 | wdoekes2 | and watch both phones register |
| 07:56.29 | wdoekes2 | if you can read it, you'll know where the problem is |
| 07:56.54 | wdoekes2 | if you have SIP support in your router, disable it |
| 07:57.04 | t3hrealadamd | no, this is all home stuff |
| 07:57.10 | wdoekes2 | (or VoIP support, or whatever it might be called) |
| 07:57.47 | t3hrealadamd | I find it strange it's still holding on to the incorrectly-configured username. |
| 07:57.55 | t3hrealadamd | [Doing the debug now.] |
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| 08:01.01 | t3hrealadamd | Offhand how long it should be before inactive peers are purged from the "sip show peers" table? |
| 08:01.39 | p3nguin | What do you mean by inactive? |
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| 08:01.53 | t3hrealadamd | If I have a phone unplugged, softphone not connected, etc. |
| 08:02.06 | wdoekes2 | I guess it depends on the Expires: header |
| 08:02.13 | p3nguin | Just because they aren't currently registered does not mean the account has been removed. The entry will stay on there forever. |
| 08:02.23 | wdoekes2 | i.e. how often your phones register |
| 08:02.48 | p3nguin | If there's a peer defined in sip.conf, it will show up on sip show peers. |
| 08:02.56 | p3nguin | regardless of its state. |
| 08:02.56 | t3hrealadamd | Got this at the bottom of the registration: Sending to 1.0.0.200 : 5060 (no NAT) |
| 08:03.09 | t3hrealadamd | That's the phones, IP, but why it's saying No NAT |
| 08:03.13 | t3hrealadamd | p3nguin: ah |
| 08:03.46 | wdoekes2 | that can't be good, if the asterisk is external to your lan |
| 08:04.02 | t3hrealadamd | it is |
| 08:04.16 | wdoekes2 | did you have nat=yes in your peer definition? |
| 08:04.20 | t3hrealadamd | Yes, I do. |
| 08:04.29 | t3hrealadamd | and, as I mentioned, the bindport |
| 08:04.39 | wdoekes2 | bindport in the asterisk? |
| 08:04.43 | t3hrealadamd | in the sip.conf |
| 08:04.55 | wdoekes2 | that's not what you want to set |
| 08:05.18 | p3nguin | What does the bindport have to do with nat? |
| 08:05.44 | t3hrealadamd | from what I could understand, it was so I could specify what NAT port the client device communicated on |
| 08:05.50 | t3hrealadamd | rather than letting it be auto-determined |
| 08:05.55 | wdoekes2 | source port for packets from the asterisk will be from bindport.. but that's a global setting, afaik |
| 08:05.56 | t3hrealadamd | perhaps I am wrong? |
| 08:06.13 | wdoekes2 | you want auto-determination |
| 08:06.26 | t3hrealadamd | bindport = Number : UDP Port to bind to (listen on). Used to be port in Asterisk v1.0.x. Default 5060. |
| 08:06.34 | wdoekes2 | unless you do manual port forwarding on your nat router |
| 08:06.36 | t3hrealadamd | It was part of my troubleshooting; I'll take it back out |
| 08:06.47 | wdoekes2 | in which case you'd be setting port, not bindport |
| 08:06.53 | p3nguin | Just leave bindport alone. |
| 08:07.18 | t3hrealadamd | Yeah, you know what, I didn't read that close enough... |
| 08:08.09 | t3hrealadamd | well, that puts me back at square one |
| 08:08.25 | wdoekes2 | no.. lower the register timeout on the phones |
| 08:08.32 | wdoekes2 | and watch them register |
| 08:08.52 | wdoekes2 | if asterisk tries to answer to a nat IP, then the nat=yes is not properly picked up |
| 08:09.03 | wdoekes2 | do a sip 'show peer <peername>' |
| 08:09.12 | wdoekes2 | s/sip '/'sip / |
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| 08:15.29 | AskewDread | Hey all, ive got a cisco iphone with the SIP stuff loaded on it, it registers sucesfully with my asterisk server but then every 10 secs it seems to reregister even though its set to 60 secs for now, and if i try make a call it starts ringing but then d/c's (im guessing because it reregisters or something or drops connection) |
| 08:16.26 | t3hrealadamd | wdoekes2: it seems that adding "defaultuser=[extension]" to the sip.conf for each physical phone did the trick |
| 08:16.39 | t3hrealadamd | wdoekes2: the phone grabbed a different NAT'd port |
| 08:18.12 | wdoekes2 | the phones should grab different NAT'd ports (or rather, the nat router should) |
| 08:18.17 | wdoekes2 | I |
| 08:19.15 | t3hrealadamd | yeah, they did |
| 08:19.20 | wdoekes2 | I'm not sure why setting defaultuser did the trick for you right now, but I'm glad it works :) |
| 08:19.24 | t3hrealadamd | they're all different now, two physical phones and softphone |
| 08:19.40 | t3hrealadamd | they're polycom IP phones; they apparently are rather quirky with asterisk |
| 08:20.10 | t3hrealadamd | in the [extension].conf in the TFTP for the phones, when setting up the registrar, I found out I had to use the extension as the username, even with a username set |
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| 08:21.22 | t3hrealadamd | wdoekes2: thanks for you help |
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| 08:33.09 | dandre | Hello, |
| 08:33.20 | dandre | These see here: http://pastebin.fr/10721 |
| 08:33.54 | dandre | on the fly queue creation issue or anything else |
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| 08:51.56 | p3nguin | dandre: When you send the call to a phone and it is busy, you want the call put on hold... and then what happens next? |
| 09:00.01 | dandre | when the phone is available I want it to ring |
| 09:00.19 | dandre | as if it were in a queue |
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| 09:13.05 | kaldemar | dandre: what exactly is complicated in creating a queue for each phone? |
| 09:13.46 | kaldemar | if you do the same thing by some other means, it will only be more complicated. |
| 09:17.25 | dandre | ok so I will create a queue for each phone |
| 09:18.24 | *** join/#asterisk no1peanut (~rudolf@h59ec0d32.stgertrud.dyn.perspektivbredband.net) |
| 09:23.37 | CRCinAU_ | p3nguin: progress on the conference :) |
| 09:24.10 | CRCinAU_ | it works with the latest chan-sccp-b trunk - however theres a bug that causes asterisk to crash when you do a 'sccp reboot <phone>' |
| 09:26.16 | CRCinAU_ | p3nguin: however: http://sourceforge.net/mailarchive/message.php?msg_id=27198110 |
| 09:28.37 | p3nguin | "sccp reboot ..." shouldn't cause anything to crash, considering it's not a valid command. |
| 09:28.45 | p3nguin | Maybe it was something else? |
| 09:28.51 | CRCinAU_ | errr yes, yes it was.... |
| 09:28.54 | p3nguin | There's reset and restart. |
| 09:28.56 | CRCinAU_ | sccp reset and sccp restart |
| 09:29.09 | p3nguin | They both have problems now? |
| 09:29.31 | CRCinAU_ | only from revision 2472 onwards |
| 09:29.48 | CRCinAU_ | its something to do with either sccp_socket.c or sccp_actions.c |
| 09:30.15 | p3nguin | I see. I'm using 2389. |
| 09:30.49 | CRCinAU_ | 2471 seems to work perfectly |
| 09:31.03 | CRCinAU_ | the developer uses 2450 in production without issue. |
| 09:31.15 | p3nguin | I might try it, but all the revs in the 2400s that I have tried all have the MWI light problem. |
| 09:31.21 | CRCinAU_ | thank got with svn you can checkout different revisions! |
| 09:31.38 | CRCinAU_ | got/god even |
| 09:32.25 | p3nguin | I'll try both that you suggested. |
| 09:33.03 | CRCinAU_ | let me know if you have an MWI issue on 2450 or 2471 |
| 09:33.05 | p3nguin | actually, no I won't. I know 2471 will have the MWI light problem. |
| 09:33.17 | p3nguin | I'll try 2450, but I expect it will have it, too. |
| 09:33.43 | CRCinAU_ | hmmmmm |
| 09:36.33 | CRCinAU_ | hmmm |
| 09:36.45 | CRCinAU_ | the MWI comes up when I leave a message |
| 09:36.54 | CRCinAU_ | but after making a call, it goes out |
| 09:37.04 | CRCinAU_ | even though theres one message thats new |
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| 09:37.38 | p3nguin | Pick up the handset and put it back down. It'll light up again. |
| 09:38.06 | p3nguin | That's not an acceptable bug for me. |
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| 09:39.58 | dimm | hello! is license from 'chan_skype.so' is only for one skype account? |
| 09:39.59 | p3nguin | Skinny Client Control Protocol (SCCP). Release: 3.0 PreRC4 - 2450 (built by '' on '2011-03-14 09:36:11 UTC') |
| 09:40.16 | p3nguin | 2450 test, coming right up! |
| 09:40.17 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:40.48 | p3nguin | MWI light bug, present. |
| 09:42.14 | p3nguin | I guess I need to test every rev between the one I know is good and the first one I have used with the bug. That way I can find out which rev it started in, and I can check the tracker to see what was changed. |
| 09:43.03 | CRCinAU_ | nods |
| 09:43.50 | CRCinAU_ | p3nguin: thats how I did it for tracking the crashes for the reset/restart |
| 09:44.21 | p3nguin | I haven't ran across that problem. When did it start and did it ever get fixed? |
| 09:44.32 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 09:45.17 | CRCinAU_ | it started with 2471 |
| 09:45.20 | CRCinAU_ | errr 2472 |
| 09:45.31 | CRCinAU_ | it hasn't been fixed as yet |
| 09:45.50 | CRCinAU_ | The good news is that there are only changes to TWO files between 2471 |
| 09:45.51 | CRCinAU_ | and 2472: |
| 09:45.51 | CRCinAU_ | U chan-sccp-b/src/sccp_socket.c |
| 09:45.51 | CRCinAU_ | U chan-sccp-b/src/sccp_actions.c |
| 09:47.05 | CRCinAU_ | I've just emailed the sccp mailing list regarding the MWI bug :) |
| 09:47.11 | p3nguin | So 2471 is good, 2472 is broken. |
| 09:47.21 | CRCinAU_ | for restart/reset, yeh |
| 09:47.23 | p3nguin | I hope you explained it well. |
| 09:47.32 | CRCinAU_ | no idea where the mwi bug is though |
| 09:47.40 | p3nguin | It does not affect the MWI on the line key, just the light. |
| 09:47.48 | CRCinAU_ | yeah |
| 09:47.56 | CRCinAU_ | same as 2471 |
| 09:48.52 | p3nguin | No MWI bug in 2400. |
| 09:48.57 | CRCinAU_ | hmmmm |
| 09:49.21 | p3nguin | I'm narrowing it down. |
| 09:49.55 | CRCinAU_ | probably sccp_mwi.c? |
| 09:50.29 | *** join/#asterisk m4xmr (~m4xmr@93-36-139-173.ip60.fastwebnet.it) |
| 09:50.35 | p3nguin | I thought it existed in 2411, but I just installed it again and I'm not seeing the bug behavior this time. |
| 09:50.54 | p3nguin | I'm trying to figure out why I went back to 2389 if 2411 was working. |
| 09:52.06 | CRCinAU_ | hmmm |
| 09:52.11 | CRCinAU_ | I'm trying to query sn to get some info |
| 09:52.43 | p3nguin | It's in 2430. |
| 09:53.21 | CRCinAU_ | as in the bug starts from 2430? |
| 09:53.34 | p3nguin | no |
| 09:53.39 | p3nguin | but it is in 2430. |
| 09:54.08 | CRCinAU_ | as in the lamp doesn't light up at all? or it goes out after a call? |
| 09:54.32 | p3nguin | I'm testing various revs, narrowing it down. I am going to find out which one it starts in, but I have to test various builds. |
| 09:54.40 | CRCinAU_ | nods |
| 09:54.42 | CRCinAU_ | will wait :) |
| 09:55.07 | p3nguin | On the rev that the bug is not present, the light is off during the call and then goes back on when I hang up the handset. |
| 09:55.17 | CRCinAU_ | sweet |
| 09:55.25 | p3nguin | ? |
| 09:55.36 | CRCinAU_ | will wait until you figure this one out :P |
| 09:55.36 | p3nguin | I'm explaining the bug. |
| 09:55.52 | CRCinAU_ | I know 2471 has the bug |
| 09:56.08 | p3nguin | And I've told you that it is in 2430. |
| 09:56.18 | p3nguin | but it's not in 2411. |
| 09:56.24 | CRCinAU_ | hmmm |
| 09:56.25 | p3nguin | So I am narrowing it. |
| 09:56.48 | CRCinAU_ | # svn diff -r2411:2471 https://chan-sccp-b.svn.sourceforge.net/svnroot/chan-sccp-b/trunk/src/sccp_mwi.c |
| 09:57.07 | CRCinAU_ | doesn't show any differences :( |
| 09:57.32 | p3nguin | Just be patient. I'm doing the work for you. |
| 09:57.36 | CRCinAU_ | :) |
| 09:58.55 | p3nguin | It doesn't seem to be in 2420. When I hang up the handset, the light is still lit. |
| 09:59.34 | *** join/#asterisk [netman] (~netman@241.Red-80-39-202.dynamicIP.rima-tde.net) |
| 10:00.06 | *** join/#asterisk manji (~manjiki@94-193-165-46.zone7.bethere.co.uk) |
| 10:05.37 | p3nguin | The MWI problem is in 2425. |
| 10:05.56 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 10:09.10 | *** join/#asterisk jkroon (~jkroon@dsl-242-2-90.telkomadsl.co.za) |
| 10:09.30 | CRCinAU_ | hmmmm |
| 10:09.36 | jkroon | hi guys, the AMI command DAHDIRestart is supposed to be an exact replica of the cli dahdi restart right? |
| 10:10.42 | jkroon | it seems that the AMI DAHDIRestart won't pick up newly configured DAHDI channels from users.conf ? |
| 10:10.49 | jkroon | is this how it's supposed to be, or a bug? |
| 10:11.54 | p3nguin | MWI bug in 2421. |
| 10:12.00 | CRCinAU_ | hmmmmm |
| 10:12.05 | p3nguin | So there you have it. It's not in 2420, but it is in 2421. |
| 10:12.15 | p3nguin | Check the tracker to see what files changed. |
| 10:12.22 | CRCinAU_ | checking via svn |
| 10:12.35 | p3nguin | I'm going to rebuild 2420 and reinstall just to be sure. |
| 10:13.16 | *** join/#asterisk nirs (~nirs@2.54.61.220) |
| 10:14.02 | jkroon | nm, my stupidity, i think. |
| 10:14.23 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
| 10:14.50 | nirs | reporting live from Asterisk Israel |
| 10:14.57 | *** join/#asterisk eelcob (~eelco@87.249.110.67) |
| 10:15.26 | nirs | bryan johns just finished his talk about the Asterisk community |
| 10:15.35 | eelcob | hi all! is there a working codec_g729a.so for asterisk 1.8.4-rc2? |
| 10:15.38 | nirs | i think this is the first time it was done in Israel |
| 10:16.10 | nirs | eelcob, there should be - although, I'm still in 1.6 production systems |
| 10:16.11 | eelcob | the download only points to 1.8.0-1.8.3, but we are using 1.8.4-rc2 due to memory leaks in previous versions |
| 10:16.43 | eelcob | (and that version does not work / start up with 1.8.4-rc2) |
| 10:18.11 | p3nguin | crcinau_: Confirmed, 2420 does not have the bug. 2421 has it, though. |
| 10:18.15 | CRCinAU_ | p3nguin: something to do with this? http://fpaste.org/8XSz/ |
| 10:19.11 | p3nguin | I have no clue. I can barely read that paste. |
| 10:19.19 | CRCinAU_ | go patch ;) |
| 10:19.50 | *** join/#asterisk sourcode (~code@ppp-58-8-201-19.revip2.asianet.co.th) |
| 10:20.04 | nirs | eelcob, I've encountred a funky issue with Asterisk 1.6.X, where the SIP channel will simply stop handling calls, all registrations will become halted and you have to reboot |
| 10:20.09 | nirs | ever encountered something like that ? |
| 10:20.43 | eelcob | nirs, no but we never have run with 1.6, only 1.2 and 1.8 |
| 10:21.03 | nirs | wow, that's a serius jump |
| 10:21.28 | nirs | that's like moving from a Mazda 3 to a jazzed up Ferrari Testarosa |
| 10:21.35 | eelcob | nirs, yeah, we tried 1.4 for a while, years ago, but that was not a great succes... |
| 10:21.36 | p3nguin | crcinau_: Five files were changed between 2420 and 2421. |
| 10:22.04 | eelcob | yeah, but up until now a very leaky ferrari in terms of memory that is |
| 10:22.47 | nirs | hmmmm... upgrading to 1.8 is an option, however, I have no idea what would be the ramifications in regards to the the software compatibility |
| 10:23.40 | p3nguin | I'll stick to my 1.4.40, thanks. |
| 10:23.57 | nirs | eel, are you familiar with the kvin g729 codec ? |
| 10:24.32 | p3nguin | crcinau_: The log message in 2421 says: monitor feature revised (incomplete), mwi indicator for recording |
| 10:24.46 | p3nguin | crcinau_: So they did mess with MWI on purpose. |
| 10:24.49 | eelcob | nirs, no, not yet, do you have experience with that? |
| 10:25.20 | nirs | well, I have several customers using that "FREE" g729 implementation on 1.6 and 1.8, and it appears to be fairly stable |
| 10:25.54 | eelcob | ok, nice, we'll check it out in our testing environment |
| 10:25.59 | nirs | technically speaking, in 2012 the patent on g729 expires, so the kvin implementation will become legal |
| 10:26.00 | nirs | http://asterisk.hosting.lv/ |
| 10:26.21 | p3nguin | crcinau_: I guess I will stay with 2420 unless I run into something that causes me to revert back to 2389 or some other version that I like better. |
| 10:26.30 | nirs | however, I don't recommend that codec for large systems, simply due to issues of Digium support if you need it |
| 10:26.32 | shapr | nirs: Rumor has it the expiration is now 2016 |
| 10:26.32 | eelcob | ok, we'll just set our clock a year in advance |
| 10:26.55 | nirs | really? i was under the impression that 2012 is the date |
| 10:27.05 | nirs | well, 4 years in our business fly by like nothing |
| 10:27.11 | shapr | But that's word from the patent cartel^Wlicensing company. |
| 10:27.42 | CRCinAU_ | p3nguin: agreed. |
| 10:27.54 | p3nguin | corydon76-home said it expires in 2014. |
| 10:28.18 | nirs | corydon, you around ? |
| 10:28.29 | Corydon76-home | Yes |
| 10:28.32 | nirs | hadn't seen you in years, but yet again, I hadn't logged on in years |
| 10:28.36 | *** join/#asterisk jetlag (~jetlag@pool-173-61-212-173.cmdnnj.east.verizon.net) |
| 10:29.13 | nirs | isn't it like 3am over your side ? |
| 10:29.26 | Corydon76-home | There are multiple patents on G.729. The last one expires in 2014 (May, I think) |
| 10:29.34 | Corydon76-home | 5:30am |
| 10:29.34 | nirs | Bryan johns needs some long sleep, I think the jet lag has gotten to him a little |
| 10:29.52 | p3nguin | You're in the same TZ as I am. |
| 10:29.53 | nirs | oh right, you on DST now |
| 10:30.07 | p3nguin | Mon Mar 14 05:30:07 CDT 2011 |
| 10:31.52 | *** part/#asterisk t3hrealadamd (~t3hrealad@pool-173-75-21-13.pitbpa.fios.verizon.net) |
| 10:32.29 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 10:33.38 | *** join/#asterisk Swannie (4f475cb5@gateway/web/freenode/ip.79.71.92.181) |
| 10:37.50 | CRCinAU_ | Mon Mar 14 21:37:49 EST 2011 |
| 10:38.09 | CRCinAU_ | hello from the future ;) |
| 10:38.24 | *** join/#asterisk Pranav_rcmas (~prp_rcmas@117.193.194.67) |
| 10:38.37 | *** join/#asterisk jetlag (~jetlag@pool-173-61-205-165.cmdnnj.east.verizon.net) |
| 10:39.00 | p3nguin | crcinau_: I noticed it takes 3 minutes 22 seconds to compile chan_sccp ON my PBX system... |
| 10:39.22 | CRCinAU_ | :\ |
| 10:39.24 | p3nguin | crcinau_: So I decided to build it on my desktop and see how long it takes... |
| 10:39.33 | p3nguin | 0m17.090s |
| 10:39.34 | p3nguin | lol |
| 10:40.52 | CRCinAU_ | real 0m9.894s |
| 10:40.53 | CRCinAU_ | user 0m8.559s |
| 10:40.53 | p3nguin | That can't be right. |
| 10:40.53 | CRCinAU_ | sys 0m1.143s |
| 10:41.11 | p3nguin | Let me rm -r my src directory and make it again. |
| 10:42.02 | p3nguin | This is more accurate: 0m28.549s |
| 10:42.05 | dimm | 'skype show users' return nothing. why it can be? |
| 10:42.08 | CRCinAU_ | :\ |
| 10:42.12 | CRCinAU_ | on what system? |
| 10:42.23 | p3nguin | So 3.5 minutes on the PBX, 0.5 minutes on my desktop. |
| 10:42.51 | CRCinAU_ | and 9 seconds on an ancient P4 3Ghz? :\ |
| 10:43.36 | p3nguin | I think your time is skewed. Delete the build directory and start over. |
| 10:44.10 | p3nguin | I'm running a quad-core @ 3.4 GHz and it takes me 28 seconds. |
| 10:44.26 | CRCinAU_ | :| |
| 10:44.34 | dimm | 'skype show user [skypeuserhere]' crashed asterisk =( |
| 10:44.36 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 10:44.49 | CRCinAU_ | dimm: what are you using for skype connectivity? |
| 10:45.12 | dimm | CRCinAU_, chan_skype |
| 10:45.18 | CRCinAU_ | dimm: url? |
| 10:45.34 | dimm | CRCinAU_, what url ? |
| 10:45.39 | shapr | dimm: sounds like you have the wrong version of SFA, check to make sure the binary you downloaded matches your version of Asterisk |
| 10:45.48 | p3nguin | I can't get it any better than 0m28.212s. That's the best time out of multiple builds. |
| 10:45.53 | CRCinAU_ | for chan_skype |
| 10:46.06 | CRCinAU_ | or is that part of 1.8.x |
| 10:46.07 | CRCinAU_ | ? |
| 10:46.07 | p3nguin | Still, that's a world of difference between my desktop and my PBX. |
| 10:46.47 | shapr | p3nguin: Are you using -j 3 or so? |
| 10:46.58 | p3nguin | Let me check make.conf. |
| 10:47.24 | p3nguin | err, makepkg.conf |
| 10:47.25 | shapr | I got Asterisk to build in thirty seconds or so with make -j 11 on an eight core system. |
| 10:47.28 | CRCinAU_ | oh wait - are you timing the configure as well? |
| 10:47.31 | *** join/#asterisk killdashnine (~matthias@82.161.138.210) |
| 10:47.32 | p3nguin | MAKEFLAGS="-j4" |
| 10:47.52 | dimm | shapr, i use asterisk-1.6.2.16.12 and sfa skypeforasterisk-1.6.2.0_1.1.2-x86_32.tar.gz |
| 10:48.01 | p3nguin | Yes, I am running makepkg, which runs ./configure, make, and then make install, all inside fakeroot. |
| 10:48.01 | shapr | dimm: That looks right to me. |
| 10:48.41 | CRCinAU_ | ahhhhh |
| 10:49.03 | p3nguin | (I only install packages in my systems) |
| 10:49.57 | CRCinAU_ | dimm: did you buy it? |
| 10:50.18 | dimm | CRCinAU_, byy license, yes |
| 10:50.38 | CRCinAU_ | bah. |
| 10:51.32 | dimm | buy |
| 10:51.49 | CRCinAU_ | I was looking at skypeforasterisk |
| 10:51.55 | CRCinAU_ | but just not worth the $$ :p |
| 10:51.57 | zkn | anyone knows what memory leak in 1.8 versions nirs was talking about ? |
| 10:52.06 | dimm | i dont' understand |
| 10:53.01 | CRCinAU_ | ?? |
| 10:55.20 | *** join/#asterisk nicola_pav (~nicola@mail2.tikalnetworks.com) |
| 10:56.44 | eelcob | zkn, we are experiencing a small memory leak with the 1.8 version |
| 10:57.19 | *** join/#asterisk garymc (~chatzilla@host81-148-15-59.in-addr.btopenworld.com) |
| 10:57.57 | eelcob | i must ask my colleague for details, but if i remembered correctly, it was somewhere in chan_sip |
| 10:58.20 | *** join/#asterisk nicola_pav (~chatzilla@mail2.tikalnetworks.com) |
| 11:00.28 | nicola_pav | hello. any hints please on how to uninstall iaxmodem? |
| 11:01.25 | dimm | CRCinAU_, how do you think, is license for sfa is only for one skype account? |
| 11:01.46 | p3nguin | nicola_pav: Use the package manager, just like when you installed it. |
| 11:02.00 | zkn | eelcob, i'm running version 1.8.3 for production, where can I read about this memory leak or could you tell me in short how/when this affects my systems? |
| 11:02.24 | dimm | CRCinAU_, i see that my license is only for 1 channel, but i want to know, am i have right to use some accounts with the same license file ? |
| 11:02.32 | *** join/#asterisk jetlag (~jetlag@pool-173-61-239-226.cmdnnj.east.verizon.net) |
| 11:03.13 | p3nguin | You should be able to use any account you want as long as you are within the limits of the channels you have licensed. |
| 11:03.20 | eelcob | zkn, the leak we are experiencing is very small, like a couple of bytes for each sip session. When compiled with DEBUG_THREADS (or something like that) however, the leakage is huge.. (so don't do that in production) |
| 11:04.06 | eelcob | the memory footprint goes up to 300MB (start 50MB) after 2 mln calls... |
| 11:04.34 | zkn | eelcob, i don't do DEBUG_THREADS, so i'm more or less safe then, i guess |
| 11:04.56 | dimm | p3nguin, ok, i use one channel :) |
| 11:04.56 | nicola_pav | p3nguin: its installed manually |
| 11:05.01 | eelcob | yes, just check the memory usage of asterisk once in a while |
| 11:05.18 | *** join/#asterisk Cadey (~Cadey@62.84.178.106) |
| 11:05.26 | zkn | eelcob, alright, cool, thanks for explaining |
| 11:05.33 | p3nguin | nicola_pav: Yes, you manually installed it, so now you can manually deinstall it. |
| 11:05.53 | *** join/#asterisk [netman] (~netman@241.Red-80-39-202.dynamicIP.rima-tde.net) |
| 11:06.54 | Cadey | Hi guys, I have a strange problem here which I was thinking was down to a Dialplan collision but its not the case as ive cleared my entier dialplan and added a singe extention with a dial to my mobile phone. The issue is, when it dials I hear two ringing noises about a second apart. Also when I change the my mobile number to a valid length but invalid number I hear the UK standard "The number |
| 11:06.54 | Cadey | you have dialed has not been recognised" message but also still hear it ringing? |
| 11:07.39 | CRCinAU_ | anyhow - time for bed. |
| 11:07.47 | CRCinAU_ | I have a plane to fly tomorrow. |
| 11:07.49 | CRCinAU_ | g'night all |
| 11:08.02 | Cadey | im now thinking the SIP phone is making the ringing sound as well as being sent a ringing tone down the line |
| 11:08.05 | Cadey | does that make sense? |
| 11:09.49 | nicola_pav | p3nguin: how? |
| 11:10.38 | p3nguin | nicola_pav: I already told you. You install software by the package manager, so deinstall it with the package manager. apt-get, yum, whatever you have. |
| 11:10.39 | dimm | where i can find something called "buisness panel" in asterisk or in free pbx ? |
| 11:11.21 | p3nguin | dimm: Asterisk doesn't have any panels of any type, including business panel. |
| 11:15.34 | nicola_pav | p3nguin: i am sorry i was not clear. it was installed using a script |
| 11:15.48 | nicola_pav | which uses make and make install |
| 11:16.10 | cobra2 | nicola_pav: then go to the source directory and 'make uninstall' |
| 11:16.47 | cobra2 | and avoid scripts unless you know what they are doing and how to reverse it |
| 11:18.01 | nicola_pav | cobra2: i will try that |
| 11:18.42 | cobra2 | nicola_pav: good luck |
| 11:18.52 | Cadey | no one experienced a double rining of a channel then? |
| 11:19.02 | Cadey | its as Aastra 57i if that helps |
| 11:19.03 | nicola_pav | cobra2: thanks. i need it |
| 11:19.57 | cobra2 | nicola_pav: if worse comes to worse, you can always to do some manual file editing, i.e. 'find / -name <file>' |
| 11:20.01 | zkn | hmm...after the memory leak "news" i went to check my system an ended up restarting it because SIP was acting up.. i was able to see the call coming in from the trunk but the call flow did not activate.. by entering "core show channeles verbose/concise" just froze my CLI |
| 11:21.53 | zkn | i'm running a Intel(R) Pentium(R) 4 CPU 3.20GHz system with 1G ram |
| 11:21.56 | nicola_pav | cobra2: and delete then one by one? |
| 11:22.53 | cobra2 | nicola_pav: if you have to, but you might as well start over from scrach if your going to have to do that. |
| 11:23.32 | zkn | htop shows 2 cores, i suppose these are threads really for P4 chip, and during the incident one of the threads was on 100% constantly |
| 11:23.58 | *** join/#asterisk jkroon (~jkroon@dsl-242-2-90.telkomadsl.co.za) |
| 11:24.11 | jkroon | are there any known RTP port leaks in @ 1.6.2.16.2 ? |
| 11:24.45 | cobra2 | nicola_pav: your trying to get rid of iaxmodem right? from sourceforge? |
| 11:25.24 | jkroon | I get to make a few thousand calls then * just goes entirely messed up due to limits on allocating more sockets, and with netstat -nulp asterisk has a LARGE number of sockets open, some with data in the Recv Q, but usually empty on both Recv and Trans Q's. |
| 11:26.55 | *** join/#asterisk cyborg-one (1000@212-178-18-177.broadband.tenet.odessa.ua) |
| 11:27.47 | zkn | hmm..."netstat -nulp asterisk" got to try that when I catch my system acting up again |
| 11:28.32 | zkn | i might have the same problem as you, jkroon |
| 11:29.14 | zkn | althoug as of this weekend I'm running 1.8.3 |
| 11:29.43 | zkn | and just a while ago ended up restarting the damned thing |
| 11:35.21 | jkroon | yea, 1.8.3 apparently has a crapload of memory leaks. |
| 11:35.51 | p3nguin | nicola_pav: Maybe this will teach you the importance of using your package manager. |
| 11:35.54 | jkroon | at 3500 completed calls/day (close on 6000 attempted) those tend to add up quickly. |
| 11:36.21 | jkroon | p3nguin, unfortunately there are too many "fixes" out there that bypasses package managers. |
| 11:37.03 | jkroon | ok, i guess my memory leak restart cron job comes back into use :( |
| 11:37.14 | zkn | :/ |
| 11:37.26 | zkn | i wouldn't want to resort to anything like that |
| 11:38.14 | zkn | this problem should them be widespread? i mean how do others solve this? |
| 11:38.15 | p3nguin | I'm starting to feel lucky I still use 1.4. |
| 11:38.17 | jkroon | i don't WANT to. |
| 11:38.38 | jkroon | but having * die on you every other day is also not an option. |
| 11:38.44 | cobra2 | p3nguin: but LFS doesn't have a package manager |
| 11:38.57 | zkn | that sounds exactly what i'm going through |
| 11:39.09 | jkroon | zkn, pvt ? |
| 11:39.10 | p3nguin | I don't have any problems with 1.4.40 that I can't live with. |
| 11:39.23 | zkn | i was on 1.6.X before... same shit all the time |
| 11:39.40 | jkroon | 1.6.2.X was fine for a LONG time. |
| 11:40.03 | p3nguin | I started using Asterisk in 1.4 branch and I've stayed on it from day 1. |
| 11:44.08 | zkn | jkroon, pvt, sure |
| 11:46.28 | zkn | i'm not even using much, just sip and iax2 channles with quelines |
| 11:46.45 | zkn | but actually to me it seems that iax2 is causing all the problems |
| 11:50.24 | p3nguin | I use IAX2 in 1.4.40 and don't have any problems. |
| 11:51.43 | zkn | aah, you might maybe know the answer to that one... when I need to restart Asterisk for the abovementioned problems, when I get to CLI again, I'm seeing constant flow of "Remote UNIX connection... Remote UNIX connection disconnected" lines |
| 11:51.52 | zkn | where do these come from ? |
| 11:52.04 | *** join/#asterisk Pranav_rcmas (~prp_rcmas@117.193.194.67) |
| 11:52.08 | *** join/#asterisk coppice (~chatzilla@m180-219-200-59.smartone-vodafone.com) |
| 11:52.21 | zkn | if I restart Asterisk again, then they disappear |
| 11:54.21 | zkn | I suppose they do no harm but are quite an annoyance, on the other hand I don't think this is normal behaviour |
| 11:54.36 | zkn | how to avoid it? |
| 11:55.09 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 11:56.06 | *** join/#asterisk jkroon (~jkroon@dsl-242-2-90.telkomadsl.co.za) |
| 11:57.51 | Cadey | hates this double ring tone issue hes got :( |
| 11:57.53 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:58.28 | *** join/#asterisk CareBear\ (peter@stuge.se) |
| 11:59.54 | schmidts | Cadey which phone do you use? Take a look if you can find something like Sticky-183 (Linksys or Cisco Phones) this solves this problem ;) |
| 12:00.11 | Cadey | I use Aastra 57i |
| 12:01.27 | p3nguin | zkn: The remote connection message is from someone or something running asterisk -r or asterisk -rx. It's often a script or cron job. |
| 12:02.16 | zkn | only thing I can think of is safe_asterisk |
| 12:02.33 | p3nguin | I wouldn't use safe_asterisk. |
| 12:02.47 | zkn | why not? |
| 12:03.57 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-136-43.dsl.stlsmo.sbcglobal.net) |
| 12:04.00 | CareBear\ | Hi all! Suddenly my asterisk starts saying "Failed to authenticate on INVITE" when I try to make output SIP calls. It came from one call to another. For some reason asterisk no longer tries to send back a response to the INVITE challenge. :\ |
| 12:04.02 | p3nguin | I think the more important question is, "Why should I use it?" |
| 12:04.08 | LemensTS | MORNING |
| 12:04.27 | CareBear\ | output = outbound :) |
| 12:05.17 | zkn | smart people writing books on Asterisk claim it is a good idea to use safe_asterisk |
| 12:05.19 | Cadey | Arrrr |
| 12:05.26 | CareBear\ | I used to be running 1.6.2.0-rc1 but just now upgraded to 1.8.1.1, which allowed one call to go out, but not much since. |
| 12:05.26 | Cadey | schmidts : thanks for that |
| 12:05.26 | p3nguin | pffft |
| 12:05.30 | zkn | :) |
| 12:05.38 | Cadey | schmidts : I belive the new firmware changed the default! |
| 12:06.43 | *** join/#asterisk awclin_ (~alinford@g0962184.demon.co.uk) |
| 12:06.55 | zkn | Asterisk can be loaded in a variety of ways. The easiest way is to start Asterisk by running |
| 12:06.56 | zkn | the binary file directly from the Linux command-line interface. If you are running a |
| 12:06.56 | zkn | system that uses the init.d scripts, you can easily start and restart Asterisk that way as |
| 12:06.56 | zkn | well. However, the preferred way of starting Asterisk is via the safe_asterisk script. |
| 12:07.31 | CareBear\ | doh. they now changed to putting the IP in their realm. |
| 12:07.43 | p3nguin | Why? Why would that be preferred? What is safe_asterisk going to do for me? |
| 12:07.44 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 12:08.31 | CareBear\ | brilliant. working again. later people! |
| 12:08.32 | *** part/#asterisk CareBear\ (peter@stuge.se) |
| 12:08.35 | leifmadsen | p3nguin: the init scripts use safe_asterisk -- it also provides the ability to specify where the core file is dropped (if a crash happens) , restarts asterisk automatically, and can email (notify) the admin a restart has occurred |
| 12:08.37 | zkn | i suppose Leif should take that one... i quote this from his book |
| 12:08.40 | zkn | haha |
| 12:08.52 | zkn | nice one |
| 12:10.11 | leifmadsen | there is also a sidebar near the end of this section: http://ofps.oreilly.com/titles/9780596517342/ch03.html#Installing_id293688 |
| 12:10.26 | *** join/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 12:11.42 | zkn | but it seems that when doing "/etc/init.d/asterisk restart" some prcess of smth is removed, which will start to login-logout of Asterisk causing these "Remote UNIX connection... Remote UNIX connection disconnected" lines to appear in the CLI |
| 12:12.03 | zkn | meant, is not removed |
| 12:13.03 | p3nguin | Maybe safe_asterisk is creating the connections that you're seeing. |
| 12:14.17 | zkn | aha |
| 12:14.19 | zkn | indeed |
| 12:14.31 | zkn | when I check for running processes |
| 12:14.54 | zkn | then after the restart i have now two /bin/sh /usr/sbin/safe_asterisk -c running |
| 12:15.10 | p3nguin | I've never had a reason to want safe_asterisk, but Leif did make a good argument as to why I should consider it. |
| 12:15.14 | zkn | one of which should have been killed |
| 12:16.14 | zkn | so, I wonder what is the occasion when this happens that "/bin/sh /usr/sbin/safe_asterisk -c " is not killed during the restart |
| 12:16.35 | leifmadsen | are you using the latest version of safe_asterisk? that has been a bug in the past |
| 12:17.33 | p3nguin | My rc script simply runs /usr/sbin/asterisk -G asterisk -U asterisk. I'll consider safe_asterisk and I will probably tinker with it later to see what it will do for me. |
| 12:17.55 | zkn | i did "make uninstall-all" to the prvious version before installing 1.8.3 |
| 12:17.58 | p3nguin | I don't worry about crashes, so I doubt I will switch to it for production purposes. |
| 12:18.10 | zkn | i suppose it removed also the safe_asterisk script |
| 12:18.15 | zkn | but i'm not 100% sure |
| 12:20.42 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 12:20.56 | zkn | yeah, the date on the scrip file is recent |
| 12:21.02 | zkn | but |
| 12:21.46 | zkn | i notice now that the script is owned by root whereas I have made asterisk itself start as asterisk in group asterisk |
| 12:22.13 | zkn | could that be the reason ? |
| 12:23.01 | zkn | as in, I should chown safe_asterisk, too ? |
| 12:23.17 | dimm | p3nguin, i have chan_skype working now :) |
| 12:23.26 | p3nguin | great |
| 12:23.46 | dimm | p3nguin, please help me, is [default] it is a context named 'default' ? |
| 12:23.47 | p3nguin | What turned out to be the problem? |
| 12:24.22 | p3nguin | If [default] is found in extensions.conf, it is a context named default. |
| 12:24.50 | dimm | p3nguin, i not find what problem was, but just put in chan_skype.conf settings to my skype account. and something another settings. I was read README from skypeforasterisk.tar.gz |
| 12:25.16 | dimm | p3nguin, now i will write dialplan and another stuff for work with skype |
| 12:25.22 | p3nguin | It's always a good idea to read the instructions when you don't know what you're doing. |
| 12:25.23 | dimm | my first dial plan :) |
| 12:25.56 | dimm | p3nguin, can i use freepbx for edit extensions.conf ? |
| 12:26.08 | p3nguin | You can, but I don't recommend it. |
| 12:27.02 | p3nguin | You won't learn anything about Asterisk and dial plan if you do... and you'll mess up the original Asterisk config files. |
| 12:27.22 | dimm | p3nguin, i really like manual edit conf file |
| 12:27.33 | p3nguin | vim is the best! |
| 12:27.43 | dimm | p3nguin, ut sys.admin before me was install freepbx, and now i must use it :( |
| 12:27.51 | dimm | p3nguin, vim 4ever :) |
| 12:28.02 | dimm | s/ut/but/ |
| 12:28.22 | p3nguin | If you know what you're doing, you can convert the FreePBX crap back to regular Asterisk configs. |
| 12:28.36 | p3nguin | It will be difficult and take some time, but it can be done. |
| 12:35.15 | zkn | yep,i had to kill the old safe_asterisk process to get rid on the "Remote UNIX connection... Remote UNIX connection disconnected" notices |
| 12:35.24 | *** join/#asterisk davlefou (~david@196.203.159.98) |
| 12:35.40 | *** join/#asterisk jkroon (~jkroon@dsl-242-2-90.telkomadsl.co.za) |
| 12:35.50 | zkn | i still don't get it though why isn't this taken care of the init.d script when I restart Asterisk |
| 12:36.04 | zkn | by |
| 12:36.06 | defswork | p3nguin, I thought freepbx wasn't obtrusive - i.e. you can rolled your own and coexist ? |
| 12:36.24 | jkroon | defswork, no, it's freepbx or the highway. |
| 12:36.36 | jkroon | and the highway it is ... :D |
| 12:36.44 | p3nguin | FreePBX is 100% intrusive. |
| 12:36.52 | defswork | jkroon, is that arecent change ? |
| 12:37.17 | defswork | on my admittedly old installs I have I could still craft my own dial plans |
| 12:39.04 | nathan7 | jkroon++ for the matrix ref |
| 12:40.37 | LemensTS | Hey p3nguin, you know whatever happend to TKD_FENDER? Haven't seen him here for a while |
| 12:41.18 | *** join/#asterisk garymc (~chatzilla@host81-148-15-59.in-addr.btopenworld.com) |
| 12:42.36 | p3nguin | lemensts: Yes, there was a little problem and he had to leave #Asterisk for a while. He's still on IRC... and you can find him in #FreePBX. |
| 12:43.20 | p3nguin | He's on right now, actually. |
| 12:45.18 | LemensTS | p3nguin: Thanks, he helped a lot of people in here that sucks. |
| 12:45.42 | p3nguin | Yep, he spent a lot of thankless hours helping people who often did not deserve help. |
| 12:46.06 | *** join/#asterisk awclin (~alinford@g0962184.demon.co.uk) |
| 12:52.33 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
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| 13:21.31 | Katty | zombies in |
| 13:22.07 | n3hxs | hands Katty a coffee. |
| 13:27.42 | *** join/#asterisk engrxyz (~yuyuyuyuu@109.204.109.126) |
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| 13:31.51 | *** join/#asterisk DanFromUK (DanFromUK@2.27.7.170) |
| 13:32.09 | DanFromUK | hi all, is anyone available to help with a realtime queue issue? |
| 13:32.33 | p3nguin | ~ask |
| 13:32.33 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:32.41 | p3nguin | danfromuk: ^^^^^^^^^ |
| 13:33.15 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 13:34.01 | DanFromUK | Im trying to setup realtime queuses, but when i join the queue, i get this |
| 13:34.12 | DanFromUK | <PROTECTED> |
| 13:34.12 | DanFromUK | [2011-03-14 13:32:26] WARNING[4679]: app_queue.c:4261 queue_exec: Unable to join queue 'queue_101' |
| 13:34.23 | DanFromUK | QUEUESTATUS is unknown. |
| 13:34.27 | DanFromUK | the cli cant find any queues |
| 13:34.34 | DanFromUK | but the DB has a queue in it |
| 13:34.41 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 13:34.56 | DanFromUK | i've set up extconfig in the same way i've done for the other realtime items |
| 13:35.40 | DanFromUK | any ideas how i can debug this? |
| 13:38.35 | *** join/#asterisk m4xmr (~m4xmr@93-36-139-173.ip60.fastwebnet.it) |
| 13:41.01 | p3nguin | Oh boy! We're getting snow again. |
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| 13:47.14 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85de.bcn.adamo.es) |
| 13:47.59 | CRCinAU_ | p3nguin: yay - I think both bugs are going to be fixed :p |
| 13:48.04 | CRCinAU_ | ... eventually ;) |
| 13:48.14 | p3nguin | They've been answered? |
| 13:48.22 | CRCinAU_ | nods |
| 13:48.28 | CRCinAU_ | mailing lists ftw. |
| 13:49.20 | CRCinAU_ | thanks for your help on the MWI bug too in narrowing it down.... |
| 13:49.49 | p3nguin | No problem. I was wanting to do it before, but didn't have the motivation to get it done. |
| 13:49.52 | CRCinAU_ | I didn't really want to do all that while my g/f was next to me... supposedly quality time type stuff ;) |
| 13:49.57 | Cadey | cant see any setting when i export them :( |
| 13:50.03 | Cadey | what have aastra done :( |
| 13:50.33 | CRCinAU_ | so hopefully, 2480 or so should be perfect ;) |
| 13:50.58 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 13:52.01 | p3nguin | I'm stuck on 2420 until I see something that says they fixed the MWI. |
| 13:52.01 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:52.01 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:52.05 | *** join/#asterisk Mhaddog__ (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 13:53.33 | CRCinAU_ | at least I had a pretty envelope fading in and out next to the button :0 |
| 13:54.10 | p3nguin | My envelopes just blink, since I have legacy equipment. |
| 13:54.20 | CRCinAU_ | :p |
| 13:55.13 | CRCinAU_ | hmmmm |
| 13:55.18 | p3nguin | When I was using a version that was buggy, I would hang up the handset, then just lift it and drop it again. That turned the light back on. |
| 13:55.39 | CRCinAU_ | if I have a line: 101@1, I can use something in the dialplan to remove the @1 correct? |
| 13:55.58 | CRCinAU_ | p3nguin: heh - mine won't come on at all atm |
| 13:56.23 | p3nguin | Why would you need to remove the @1? Just don't type it and you don't have to remove it. |
| 13:56.54 | CRCinAU_ | it gives you a way to have a shared line, but also be able to address the phones seperately |
| 13:57.01 | p3nguin | Right. |
| 13:57.06 | CRCinAU_ | dialing 1011 will get you to one phone |
| 13:57.09 | CRCinAU_ | 1012 another etc |
| 13:57.30 | CRCinAU_ | however it still recognises 101 - so both will ring |
| 13:57.34 | p3nguin | yeah? I'm not seeing your problem. |
| 13:57.52 | CRCinAU_ | well, the caller ID is coming through as 1011 instead of 101 |
| 13:58.06 | CRCinAU_ | which I guess is technically correct, but it breaks my voicemail ;) |
| 13:59.21 | p3nguin | Then change it. |
| 13:59.40 | p3nguin | It's correct until you make it incorrect. |
| 14:00.55 | p3nguin | If you don't know what you've done wrong, paste your configs and explain the scenario. |
| 14:02.30 | Cadey | Well.... |
| 14:02.46 | Cadey | I called aastra about the double ring tone and they told me to down grade the 2.6 fireware |
| 14:03.06 | Cadey | so i guess the 3.2.0.1011 has not been tested with asterisk |
| 14:04.36 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 14:06.47 | *** join/#asterisk mawhii (~mawhii@99-117-188-231.lightspeed.dybhfl.sbcglobal.net) |
| 14:06.54 | CRCinAU_ | this: exten => 999,n,Set(caller=${CUT(${CALLERID(num)},,-3)}) |
| 14:07.10 | CRCinAU_ | but it doesn't do what I expect it to do... |
| 14:07.20 | p3nguin | You don't need to do that. |
| 14:07.26 | dimm | bye guyz |
| 14:07.34 | p3nguin | So stop trying to complicate things. |
| 14:08.12 | CRCinAU_ | then how do I get the correct mailbox # for the call to voicemail? |
| 14:08.40 | *** join/#asterisk m4xmr (~m4xmr@93-36-139-173.ip60.fastwebnet.it) |
| 14:08.47 | p3nguin | Set the callerid parameter correctly for the line. |
| 14:09.42 | CRCinAU_ | I do |
| 14:09.55 | p3nguin | Then there's no problem. |
| 14:10.03 | p3nguin | And this discussion is pointless. |
| 14:10.04 | CRCinAU_ | p3nguin: http://fpaste.org/QCBj/ |
| 14:10.18 | CRCinAU_ | I don't think you quite get it... |
| 14:10.20 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 14:10.38 | CRCinAU_ | because in the phone button, I have button = line,101@1: Desk Phone |
| 14:10.46 | p3nguin | Alright, so line 101 has a cid_num equal to 101. |
| 14:11.02 | CRCinAU_ | when it tries to use $CALLERID(num) for voicemail, it gets passed 1011 |
| 14:11.10 | p3nguin | Show me. |
| 14:11.32 | CRCinAU_ | <PROTECTED> |
| 14:11.49 | p3nguin | One line isn't showing me anything. |
| 14:11.55 | Cadey | I guess people dont use aastra phones in here then :) |
| 14:12.03 | CRCinAU_ | the previous one is: -- Executing [999@101:1] Ringing("SCCP/101-00000006", "") in new stack |
| 14:12.14 | p3nguin | Another single line... |
| 14:12.22 | CRCinAU_ | what more do you want? :\ |
| 14:12.25 | p3nguin | C'mon, show me the problem. |
| 14:12.32 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:12.34 | CRCinAU_ | VoiceMailMain("SCCP/101-00000006", "s1011") |
| 14:12.38 | p3nguin | Show me the configs. Show me the entire call. |
| 14:12.39 | CRCinAU_ | note: "s1011" |
| 14:12.44 | CRCinAU_ | not "s101" |
| 14:12.46 | p3nguin | Stop pasting single lines here. |
| 14:13.04 | CRCinAU_ | so it tries to go to mailbox 1011, which doesn't exist, because its supposed to be 101 |
| 14:13.13 | p3nguin | Show me the configs in pastebin. Then show me a complete call in the pastebin. |
| 14:13.15 | CRCinAU_ | its treating the 101@1 as 1011 |
| 14:13.50 | p3nguin | It won't magially make 101 into 1011 unless you've told it to, so show me the configs and a complete call. |
| 14:14.50 | CRCinAU_ | p3nguin: http://fpaste.org/ROPb/ |
| 14:16.07 | p3nguin | Okay, that might be the call, but I still don't have the configs. |
| 14:16.08 | CRCinAU_ | whops - forgot a bit: http://fpaste.org/XjZ7/ |
| 14:17.29 | CRCinAU_ | p3nguin: thats all the bits that are involved |
| 14:18.04 | p3nguin | No it isn't. |
| 14:18.58 | CRCinAU_ | what else is there then? no point adding in extra dialplan stuff that isn't used - or other line configs that don't apply to this case.... |
| 14:19.11 | p3nguin | I guess I'm going to have to set up a shared line like you've done and try to reproduce the problem you're describing. |
| 14:19.42 | CRCinAU_ | the point is, asterisk or chan-sccp-b is interpreting 101@1 at 1011 for caller id purposes |
| 14:20.06 | p3nguin | I'll find out in a minute. |
| 14:20.38 | CRCinAU_ | technically, its probably correct, as dialing 1011 will ring the phone associated with 101@1, 1012 would ring 101@2 |
| 14:20.41 | p3nguin | If that's the problem, I'll come up with a solution. |
| 14:21.27 | CRCinAU_ | but obviously, this breaks the mailbox associated with the main 101 line |
| 14:21.34 | p3nguin | It can only ring those lines if you have dial plan to do it. exten => 1011,1,Dial(SCCP/101@1) |
| 14:21.39 | CRCinAU_ | or at least the autologin part |
| 14:22.20 | CRCinAU_ | true |
| 14:22.24 | m4xmr | I'm tring to configure the busy signal for the extensions, which are already involved in a conversation. I tried to modifiy the call-limit option in the sip.conf but .. this creates some problem with the redirects because I'm not able to create all the necessary channels ..., do you have any idea, please? |
| 14:22.50 | p3nguin | button = line, 101@1: Shared Line |
| 14:22.59 | CRCinAU_ | p3nguin: I figure just chop off the last digit of the caller ID. ;) |
| 14:23.00 | p3nguin | Is that what you've done to add the shared line? |
| 14:23.07 | CRCinAU_ | p3nguin: yeah |
| 14:24.16 | *** join/#asterisk Swannie (4f475cb5@gateway/web/freenode/ip.79.71.92.181) |
| 14:25.36 | p3nguin | http://pastebin.us/2469 |
| 14:25.45 | p3nguin | See if this is right for the line. |
| 14:26.13 | CRCinAU_ | p3nguin: looks good |
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| 14:29.38 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan.noare-1.holmedal.net) |
| 14:32.52 | *** join/#asterisk sourcode (~code@ppp-58-8-201-19.revip2.asianet.co.th) |
| 14:34.06 | p3nguin | crcinau_: Okay, I've reproduced what you've been trying to explain to me. chan_sccp is modifying the caller id number! |
| 14:34.11 | p3nguin | So... |
| 14:34.36 | p3nguin | Do you only have a shared line on the phone? |
| 14:35.27 | p3nguin | Will you sometimes call vmail from a private line, and some other times from the shared line? |
| 14:37.12 | CRCinAU_ | nah - it'll always be from that line |
| 14:37.20 | CRCinAU_ | each line has its own mailbox |
| 14:38.12 | jaytee | p3nguin, thanks to yours and [TK]D-Fender's help I've got external Polycom phones working :-) |
| 14:38.28 | p3nguin | "sccp debug channel" shows me that it modified the caller id number for the call: -- SCCP: cid_num: "1011" |
| 14:38.37 | p3nguin | but you saw that I put 101 in the conf. |
| 14:38.49 | CRCinAU_ | nods |
| 14:38.52 | CRCinAU_ | that was my point :) |
| 14:39.47 | p3nguin | jaytee: Nice! Those NAT settings (previously listed) are *critical* to get right. Without them, Asterisk just won't work well with NATted phones. |
| 14:40.50 | *** join/#asterisk coppice (~chatzilla@9.160.232.220.dyn.pacific.net.hk) |
| 14:46.17 | p3nguin | crcinau_: exten => *86,1,VoiceMailMain(${CALLERID(num):0:3}) |
| 14:46.56 | CRCinAU_ | p3nguin: that'll do it! :) |
| 14:47.05 | p3nguin | crcinau_: Will this take care of it? That should take only the first three numbers of the cid num from any line that calls *vm |
| 14:47.57 | CRCinAU_ | oh wait... I that'll also kill the other mailboxes for the external phone numbers.... |
| 14:47.58 | p3nguin | I was trying to come up with something that would work with both three-digit private lines as well as more-than-three-digit shared lines. |
| 14:48.00 | CRCinAU_ | ponders |
| 14:48.15 | p3nguin | You're going to have 10-digit voicemail mailboxes? |
| 14:48.17 | CRCinAU_ | yeah - as in something that just trims the last digit? |
| 14:48.39 | CRCinAU_ | I have, just to make life simple - as it reads the full phone number :) |
| 14:48.51 | CRCinAU_ | ie "The person at 0 3 9 etc etc" |
| 14:48.55 | p3nguin | I couldn't come up with anything that would trim the last digit; I could only think of how to capture the first three digits. |
| 14:49.32 | CRCinAU_ | maybe I'll just abandon the 101@1 idea in favour of just 101 |
| 14:49.34 | CRCinAU_ | unless.... |
| 14:49.58 | CRCinAU_ | context matters... I can have a seperate mailbox extension in the dialplan for each context |
| 14:50.02 | CRCinAU_ | ugly, but it'll work |
| 14:50.33 | Cadey | does anyone with a good knowledge of SIP and 30 mins to spare got a second to look at a list of "upgrade" to a phones firmware and take a guess as to why one of them would cause a phone to doube up on the ringing tone when you make a call? - I dont no SIP well enough to work out what it could be :)? |
| 14:50.48 | p3nguin | I guess you could do it that way. I probably wouldn't. |
| 14:51.41 | CRCinAU_ | it works. like I said, ugly, but it won't screw with other mailboxes... |
| 14:52.38 | p3nguin | Man, the snow is really coming down in HUGE flakes now. |
| 14:52.55 | CRCinAU_ | snow? :\ |
| 14:53.00 | CRCinAU_ | I've heard of that lol |
| 14:53.24 | CRCinAU_ | that being said, we have rain in the middle of our desert at the moment.... |
| 14:53.27 | p3nguin | Where are you, again? |
| 14:53.32 | CRCinAU_ | Australia |
| 14:53.55 | p3nguin | Ah, right. I knew you told me, but I didn't remember and I didn't want to pull the log. |
| 14:54.10 | CRCinAU_ | :) |
| 14:54.22 | CRCinAU_ | yeah - the world is just wierd atm.. |
| 14:54.53 | CRCinAU_ | we copped major bushfires, then major floods, then NZ cops a major earthquake, then japan gets completely screwed by an earthquake |
| 14:55.29 | CRCinAU_ | then just to make sure they got properly screwed, nature threw a massive tsunami at them :\ |
| 14:55.38 | p3nguin | Earlier when I thought it was snow, it turned out to be freezing rain and sleet. Then it was slowly turning into a light snow. Now it's huge snow and coming down rather heavily. |
| 14:55.45 | CRCinAU_ | waits for the planet to break in half |
| 14:56.55 | CRCinAU_ | this says it all: http://www.bom.gov.au/gms/IDE00035.latest.shtml |
| 14:57.01 | p3nguin | Just yesterday and the day before, we had 60+ F degrees temps. |
| 14:57.13 | p3nguin | Now we have snows. |
| 14:57.19 | CRCinAU_ | I've never seen so much cloud over the bloody desert! it's been raining there for 3-4 days |
| 14:58.04 | CRCinAU_ | heh - I was at work at the airport last summer... it was 48C in the shade one day... |
| 14:58.15 | CRCinAU_ | now this year, we've been lucky to have a week of hot weather |
| 14:58.35 | CRCinAU_ | its like the complete opposite of last year |
| 14:59.18 | CRCinAU_ | 48C = 118F |
| 14:59.37 | jaytee | toasty |
| 14:59.48 | CRCinAU_ | indeed :\ |
| 14:59.56 | coppice | that's not so hot |
| 14:59.58 | jaytee | "Yeah, but it's a dry heat!" |
| 15:00.13 | CRCinAU_ | lol |
| 15:00.22 | CRCinAU_ | ohhh man, imagine that heat at a high humidity. |
| 15:00.24 | CRCinAU_ | urgh |
| 15:00.51 | Katty | so |
| 15:00.52 | coppice | I found 55 in the desert quite pleasant... for the 10 minutes before dehydration set in |
| 15:00.59 | Katty | i need help picking out a roller derby name |
| 15:01.00 | CRCinAU_ | hahah yeah |
| 15:01.01 | Katty | suggestions? |
| 15:01.19 | CRCinAU_ | Roller Durps |
| 15:01.27 | jaytee | I'd like to visit Coober Pedy someday. It's on my bucket list. |
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| 15:02.13 | CRCinAU_ | I'm happier down south in Melbourne :P |
| 15:02.24 | CRCinAU_ | its less.... toasty ;) |
| 15:02.32 | jaytee | I'm a rockhound with a softspot for opal. |
| 15:03.03 | CRCinAU_ | lol |
| 15:03.26 | CRCinAU_ | I try to just think I'm sane ;) |
| 15:03.28 | jaytee | bought alot of rough from a couple miner/sellers there |
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| 15:07.11 | p3nguin | Is there a way in weechat to display the server configuration? equivalent to "/network list" used in irssi. |
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| 16:10.03 | Katty | derby name: Ester Risque? |
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| 16:37.41 | ClintGoudie-Nice | Heya all. What key usages & extended key usages are required for a trunk to another sips system? It seems that Server Auth and the enhanced key usages digital sig, non repudiation, key encipherment, and data encipherment are not enough. |
| 16:38.25 | ClintGoudie-Nice | asterisk cli reports Problem setting up ssl connection: error:14094413:SSL routines:SSL3_READ_BYTES:sslv3 alert unsupported certificate |
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| 16:53.21 | batfastad | Hi everyone. Got an OT question about US/Canada toll free numbers starting with 866. We're based in the UK and signed up for a call forwarding agency about 10-15 years ago who provided us with 2 of these 866 numbers which forward to our regular UK geographic tel numbers. However the original company got bought out, and bought out again, and bought out again and now the current company have no record or knowledge of these toll free n |
| 16:54.19 | batfastad | I'm looking to move them to a new provider, also UK-based, but they need the installation information. Are there any companies/trade bodies in the US I can contact to see who is currently handling calls on these numbers? |
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| 17:03.22 | DanFromUK | hi all, is there anything special i need to do to get queue to load realtime? i've added the settings into extconfig, but the debug shows that its not even searching the database for the queue details |
| 17:10.46 | Sertys | DanFromUK: do u have the realtime modules loaded? |
| 17:11.04 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 17:11.06 | Sertys | and compiled support for your db backend in *? |
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| 17:19.10 | dimm | what interesting solutions can do based on asterisk ? |
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| 17:22.26 | pabelanger | dimm: enable / disable Christmas tree lights |
| 17:28.04 | jaytee | coffee pot IVR: Option 1 = Check Coffee Level Option 2 = Brew new pot Option 3 = Ringall/Intercom/Playback(coffeeisready) |
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| 17:35.54 | paulc | batfastad: You can find out the resporg for the toll free numbers, then get in touch with them? |
| 17:36.44 | paulc | dimm: telephone chat/date line.. call recording.. sports team info line.. all sorts :-) |
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| 17:37.13 | batfastad | paulc: That sounds excellent. How do I find the resporg? |
| 17:37.38 | dimm | :-) |
| 17:39.52 | paulc | batfastad: what's one of the TF numbers? |
| 17:40.57 | batfastad | paulc: 866 849 5704 and 866 521 0099 are the numbers... hopefully with the same root provider |
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| 19:08.23 | nightwalk | Kind of curious...is anyone here doing faxing (T.30) across asterisk (via hylafax, or other means)? |
| 19:09.54 | nightwalk | There's a local company that's spending $10k for 24 port fax cards to dangle off of their PRIs, and I'm just wondering if there isn't a better (less costly) way |
| 19:14.44 | Kobaz | fax gateways are expensive |
| 19:19.56 | nightwalk | Yeah...that's kind of the point. If the data is already available in a digital format, why convert from/to analog again just to have to convert right back to digital? |
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| 19:20.59 | KNERD | nightwalk: yeah..cheaper way = email |
| 19:21.08 | nightwalk | Seems like something that could be handled easily enough in software, and without the extra DACs & ADCs |
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| 19:22.06 | nightwalk | (though my telephony knowledge is admittedly quite limited, so I may well be talking out of my a**) |
| 19:22.12 | KNERD | nightwalk: or buy a bunch of cheap fax modems to put in PCI slots |
| 19:24.26 | nightwalk | So, what you two are trying to tell me is that asterisk and/or hylafax DON'T handle T.30 faxing well? |
| 19:26.17 | nightwalk | Hmm....guess I should've said 'asterisk or asterisk+hylafax' instead |
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| 19:28.18 | _Corey_ | nightwalk: If you ask the Hylafax guys, the only absolutely reliable way of getting faxes is with something like a Brooktrout card... that said SIP faxing works OK for most people |
| 19:28.53 | _Corey_ | The IAXmodem that Hylafax uses w/Asterisk doesn't have a lot of support, so I've been seeing more people starting to use Digium's fax product |
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| 19:36.25 | errr_ | ThiIm using aastra 6755i phones with the newest 3.2 firmware. Does anyone happen to know what configuration options I need so that blind transfers work like they did pre 3.2? Now when I hit xfer <exten> HANGUP it puts the caller on speaker phone where before it would transfer them |
| 19:36.31 | errr_ | -Thi |
| 19:48.11 | nightwalk | _Corey_: I wasn't aware digium had anything comparable to sangoma's offerings. Looks like they might be far cheaper, too. Thanks for that little tidbit :) |
| 19:49.04 | _Corey_ | nightwalk: Yeah, they're not expensive |
| 19:49.26 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 19:49.43 | nightwalk | Evidently, sangoma's are still ~$10k |
| 19:51.26 | nightwalk | Looks like google thinks the TDM2400P is a comparable card, and it appears it might be no more than ~$2k fully-populated |
| 19:52.20 | _Corey_ | nightwalk: Well, the Digium fax product I'm speaking of is a software module that is licensed per concurrent channel. You need a TDM card or VoIP provider to get the calls into your box |
| 19:52.24 | drift- | asdf |
| 19:56.55 | nightwalk | _Corey_: don't see how that'd be a problem as long as the "license" isn't time-limited (software as a service -- evil), and as long as it's no more than a few k over the price of the tdm. At any rate, I can check that out for myself. Thanks again for the pointer :) |
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| 19:59.45 | p3nguin | Is Cepstral a suitable text-to-speech engine for a large scale deployment? |
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| 20:01.47 | Kobaz | sure |
| 20:03.49 | p3nguin | You think it can handle more than 100 instances for more than 100 simultaneous calls? |
| 20:04.55 | p3nguin | Will it put a lot of load on the server computer which is handling the call? |
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| 20:11.35 | paulc | Depends on the spec of the server ;-) That's a lot to be all dynamic - any way you can split the static from dynamic parts of the prompting out? |
| 20:12.55 | p3nguin | The usage of tts will be to read email messages based on a caller's dtmf selection. |
| 20:13.35 | p3nguin | If there is anything static involved, it could probably be split off of the dynamic reading of email. |
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| 20:15.55 | paulc | p3nguin: Ah, gotcha.. not much static there at all then - for the length of emails etc you're probably best off TTSing everything.. too much hassle to split the static from the variable |
| 20:16.18 | Kobaz | p3nguin: best thing to do is try it out, they have a demo version |
| 20:16.43 | p3nguin | That's sort of what I was thinking. I can try the starter kit and see what kind of load it creates. |
| 20:17.17 | Kobaz | p3nguin: if load is an issue you can have one call server and several servers that do tts |
| 20:18.03 | p3nguin | It's $79.99 for the Allison voice and 1 port license. We could test how much load it puts on a system. |
| 20:19.18 | p3nguin | Do you have a suggestion for how to have multiple tts servers available to a single call server? |
| 20:19.35 | p3nguin | Would it be done purely in dial plan? |
| 20:19.39 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:19.44 | Kobaz | i would figure your setup would be stateless |
| 20:19.58 | Kobaz | standard load balancing theory would apply |
| 20:20.06 | Kobaz | you could round robin through the tts servers |
| 20:20.19 | Kobaz | or check the load on each one, and send the request to the item with the lowest load |
| 20:20.55 | p3nguin | Are you saying that dundi would be involved? |
| 20:21.11 | Kobaz | not necessarily |
| 20:21.27 | p3nguin | What mechanism would you employ? |
| 20:21.34 | Kobaz | probably a perl agi script |
| 20:21.52 | Kobaz | that figures out what server to use, sends the rest to the server, gets the audio back, plays the audio, and exists |
| 20:21.54 | Kobaz | exits |
| 20:22.12 | Kobaz | but that all is based on high usage where you would need multiple servers |
| 20:22.20 | Kobaz | it may not be that bad, so i would try a load test first |
| 20:22.36 | Kobaz | get the demo version and load up 100 concurrent text to speaches on some lorum ipsum text |
| 20:25.16 | p3nguin | Are there scripts available for this, or would it have to be developed by someone here? |
| 20:25.46 | Kobaz | i would think it would be pretty custom |
| 20:26.29 | Kobaz | you mentioned dundi... dundi is more for like, distributing endpoints across locations... so your server can say 'I want to dial endpoint XYX where is it?' and dundi would find it |
| 20:28.30 | p3nguin | There will need to be multiple servers if a single server cannot handle all the calls. 100 seems like the guideline of concurrent calls per server computer. |
| 20:32.24 | Kobaz | if asterisk is doing minimal work you can throw way more than 100 calls on a server |
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| 20:46.43 | Aut0ExeC | anyone recommend the grandstream handytone here? :| |
| 20:47.01 | p3nguin | ~gs |
| 20:47.01 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 20:47.03 | Aut0ExeC | about to purchanse in 1 hour |
| 20:47.14 | Aut0ExeC | lol |
| 20:47.21 | Aut0ExeC | p3nguin: so linksys it is |
| 20:47.22 | Aut0ExeC | :) |
| 20:47.37 | p3nguin | You're looking to buy ATAs? |
| 20:47.38 | Aut0ExeC | cost like 10 bucks more but it'll do i guess |
| 20:47.44 | Aut0ExeC | yep but not cards |
| 20:47.48 | p3nguin | ~grandstream |
| 20:47.49 | infobot | grandstream is probably the Yugo of VoIP hardware. Run... Run away now. Though, therealcircut says that they're not that bad. |
| 20:47.49 | Aut0ExeC | since i'm using a vbox |
| 20:48.20 | Tozz_ | we have grandstream ata's. They work one you got them configured |
| 20:48.21 | Sertys | lol |
| 20:48.28 | Sertys | the yugo of voip... |
| 20:48.29 | Tozz_ | but configuring them is a pain in the ass, the interface is full of bugs |
| 20:48.41 | Sertys | hail infobot |
| 20:48.42 | Aut0ExeC | Tozz_: oh really? u think linksys is better? |
| 20:48.42 | Tozz_ | its just crap. and really, 10 Mbit ethernet ports is so 1990's |
| 20:48.55 | Tozz_ | no experience with linksys ata's |
| 20:48.58 | Aut0ExeC | k |
| 20:49.07 | Aut0ExeC | what do u use at the moment? |
| 20:49.26 | Tozz_ | grandstream ;) |
| 20:49.34 | Aut0ExeC | lol wtf.. u just said it wasnt good |
| 20:49.34 | Tozz_ | just because I had them lying around |
| 20:49.38 | Tozz_ | but I wouldnt buy them again |
| 20:49.39 | Aut0ExeC | oh ok |
| 20:49.44 | Aut0ExeC | u have the 503? |
| 20:49.54 | Tozz_ | 486 I think |
| 20:49.57 | Aut0ExeC | k |
| 20:50.03 | Aut0ExeC | 1 fxs, 1 fxo? |
| 20:50.19 | Tozz_ | eeh, just 1 port. |
| 20:50.25 | Aut0ExeC | huh |
| 20:50.26 | Aut0ExeC | ? |
| 20:50.31 | Aut0ExeC | 1 port? |
| 20:50.38 | Aut0ExeC | wth? |
| 20:50.48 | Tozz_ | yes, just 1 fxs |
| 20:50.51 | Aut0ExeC | oh ok |
| 20:50.53 | Aut0ExeC | i gatcha |
| 20:50.56 | Aut0ExeC | with 1 fxo right? |
| 20:51.07 | Aut0ExeC | for telco |
| 20:51.17 | Tozz_ | yes, but that is just a eeh 'fallback' |
| 20:51.24 | Aut0ExeC | any echo? |
| 20:51.33 | Tozz_ | not that I know of, but we use it for faxes |
| 20:51.38 | Aut0ExeC | oh ok |
| 20:51.46 | Tozz_ | never had a telephone connected to it. But I doubt faxing will work if there was an echo |
| 20:51.51 | Tozz_ | so I guess no echo ;) |
| 20:51.56 | Aut0ExeC | lol ok |
| 20:52.05 | Aut0ExeC | i'll prolly just stick with the linksys tho |
| 20:52.14 | Aut0ExeC | everyone i ask says grandsteam is terrible |
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| 20:53.23 | Tozz_ | the configuration is.. |
| 20:53.28 | p3nguin | If you want an FXO port to connect to your telco wall jack, plus an FXS port for a phone, the SPA-3102 is possibly what you're after. |
| 20:53.31 | jaytee | ~gs |
| 20:53.31 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 20:53.40 | jaytee | ~grandstream |
| 20:53.40 | infobot | somebody said grandstream was the Yugo of VoIP hardware. Run... Run away now. Though, therealcircut says that they're not that bad. |
| 20:53.45 | Aut0ExeC | p3nguin: thas what i was gonna get |
| 20:53.51 | Tozz_ | I have 1 grandstream GXP-2000 phone, the quality is terrible |
| 20:53.55 | Tozz_ | feels cheap, bad audio quality |
| 20:54.08 | Aut0ExeC | kk |
| 20:54.13 | Tozz_ | and webinterface just as terrible as the 486 ;) |
| 20:54.16 | jaytee | Polycom FTW, Linksys where cheaper makes sense |
| 20:54.19 | p3nguin | aut0exec: I think I paid about $70 for my 3102 on Amazon. |
| 20:54.32 | Tozz_ | i have a Linksys/Cisco SPA525G |
| 20:54.32 | Aut0ExeC | p3nguin: nice .... gonna buy tonight |
| 20:54.38 | Tozz_ | not too happy with that either |
| 20:55.11 | Aut0ExeC | p3nguin: hows the quality of yours? |
| 20:55.14 | Aut0ExeC | any echo? |
| 20:55.34 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 20:55.37 | jaytee | the 3102 works ok if you need an fxo port. if you need more than one fxo port I'd go with a Digium card in the server |
| 20:55.55 | Aut0ExeC | nah only 1 fxo port is what I need |
| 20:55.59 | Aut0ExeC | i only have 1 telco line |
| 20:56.02 | Aut0ExeC | :) |
| 20:56.05 | p3nguin | There is sometimes echo when using any SIP device... it's often a case of the volume setting being too high. |
| 20:56.25 | Aut0ExeC | p3nguin: ok |
| 20:57.20 | p3nguin | The 3102 I have is used every day. It works just fine for me. I have a really old firmware to make it work well, though. Don't upgrade the firmware on yours or you will probably add bugs. |
| 20:58.19 | Aut0ExeC | ok |
| 20:58.27 | Aut0ExeC | thanks for the sip |
| 20:58.31 | Aut0ExeC | :) |
| 20:59.18 | Aut0ExeC | last thinkg... when I use background in my dialplan.. I hear nothing... |
| 20:59.23 | Aut0ExeC | does a module need to be enabled? |
| 20:59.25 | Aut0ExeC | or something |
| 21:00.22 | Aut0ExeC | weird...I can play other sounds fine |
| 21:05.53 | paulc | Trying in install DAHDI on a Centos box. I get the error "You do not appear to have the sources for the 2.6.18-194.el5PAE kernel installed". I'm pretty sure they're there, either I've missed a package, or there's something with naming (like -i686 on the end of a directory name somewhere?). This is on Centos, which I don't have a lot of love for (for this exact reason!). Can anyone point me in the right direction? |
| 21:07.13 | *** join/#asterisk GTXComm (~John@72.128.62.30) |
| 21:08.10 | p3nguin | CentOS is a find distro. Any problems are most likely a result of an inexperienced administrator or an inexperienced user. |
| 21:08.32 | p3nguin | s/find/fine/ |
| 21:09.33 | paulc | shrugs |
| 21:10.02 | pabelanger | paulc: install the kernel headers |
| 21:10.19 | paulc | I'm not looking for a distro flame war - I'm just saying I have way more hassle dealing with Centos boxes than Ubuntu boxes. I'll gladly doff my hat to any experienced Centos administrator who can point me in the right direction and help me learn - humble enough? |
| 21:10.33 | Aut0ExeC | pabelanger: hey man... I just finished reading ur tut on adding google voice ..:) |
| 21:10.42 | Aut0ExeC | pabelanger: i'm going to set that up tonight |
| 21:11.04 | pabelanger | Aut0ExeC: cool |
| 21:11.04 | paulc | pabelanger: I've done yum install kernel-headers and yum install kernel-devel and also gave kernel-PAE-headers a shot but still no joy :( |
| 21:11.30 | Aut0ExeC | pabelanger: does ideasip allow free calls to local lines? |
| 21:11.42 | pabelanger | Aut0ExeC: unknown |
| 21:11.49 | Aut0ExeC | k |
| 21:12.18 | pabelanger | paulc: are you running the same kernel as the headers installed? |
| 21:12.28 | *** join/#asterisk yoda1410 (~yoda@aether.hipocoon.be) |
| 21:13.31 | p3nguin | What does uname -r show you? |
| 21:14.04 | paulc | pabelanger: I believe so. uname -r gives me "2.6.18-194.el5PAE" |
| 21:14.43 | p3nguin | Now check the packages. rpm -qa|grep kernel |
| 21:16.46 | paulc | p3nguin: http://pastebin.com/ykAC4mj8 |
| 21:17.05 | p3nguin | Wow, a pastebin. It should have been 1 or 2 lines. |
| 21:18.15 | *** join/#asterisk js (~js@2001:470:9c3f::1) |
| 21:18.26 | p3nguin | I guess you are using kernel-PAE-2.6.18-194.el5 |
| 21:18.32 | js | hi. is it possible that asterisk uses zrtp when passing a call to a phone? |
| 21:18.38 | paulc | I'm putting it all together for completeness - it was 5 lines, didn't want people getting bent out of shape for flooding |
| 21:18.58 | paulc | uname -r reports "2.6.18-194.el5PAE" |
| 21:19.17 | p3nguin | which should be provided by kernel-PAE-2.6.18-194.el5 |
| 21:20.24 | paulc | so what's with the .32 in those package names? 32 bit? |
| 21:20.43 | p3nguin | minor version numbering |
| 21:21.05 | p3nguin | 2.6.18-194 is not the same as 2.6.18-194.32.1 |
| 21:21.11 | p3nguin | BUT... |
| 21:21.35 | p3nguin | I don't know for sure if the packages by the name of 2.6.18-194.32.1 will product a uname -r output of 2.6.18-194.32.1 or 2.6.18-194 |
| 21:21.39 | pabelanger | paulc: $ ls -la /lib/modules |
| 21:21.59 | p3nguin | is only a part-time CentOS admin, who doesn't know everything about that sort of thing |
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| 21:26.35 | paulc | p3nguin: I've got both in there apparently: |
| 21:26.35 | paulc | drwxr-xr-x 6 root root 4096 Mar 14 14:00 2.6.18-194.32.1.el5PAE |
| 21:26.36 | paulc | drwxr-xr-x 6 root root 4096 Nov 2 2008 2.6.18-194.el5PAE |
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| 21:31.56 | vinhdizzo | im getting this error when compiling asterisk |
| 21:32.32 | vinhdizzo | cc1: error: say.c: Input/output error |
| 21:32.37 | vinhdizzo | any thoughts? |
| 21:33.08 | florz | buy a new hard disk |
| 21:33.34 | p3nguin | paulc: Check your grub menu.lst to see which one you are booting. |
| 21:33.39 | *** join/#asterisk jkroon (~jkroon@dsl-241-250-103.telkomadsl.co.za) |
| 21:33.50 | WIMPy | /earth is >95% full. Please delete anyone you can. |
| 21:34.16 | vinhdizzo | florz: are you referring to me? |
| 21:34.26 | florz | vinhdizzo: yes |
| 21:34.33 | p3nguin | wimpy ;) Nice one. |
| 21:34.39 | vinhdizzo | florz: it's a hard disk issue? |
| 21:34.45 | WIMPy | Old |
| 21:34.58 | florz | vinhdizzo: well, probably you should first check your kernel log |
| 21:35.13 | florz | that should tell you what's broken |
| 21:35.18 | vinhdizzo | florz: how might i ask? |
| 21:35.24 | florz | dmesg ? |
| 21:35.37 | vinhdizzo | thanks |
| 21:37.01 | Tozz_ | I'm having issues with * 1.6 where audio sometimes stops. I also see "exited non-zero on SIP/xxx-<ZOMBIE>". Could those be related and what can cause Zombie SIP channels? |
| 21:41.26 | paulc | p3nguin: Take a look at http://pastebin.com/x9visdnF - both are in there, but it seems like uname -r reports the second one? maybe? even though the default's 0 (the first one?) |
| 21:41.37 | paulc | I wonder if some updating went on |
| 21:41.42 | paulc | or if a reboot's worth it |
| 21:44.16 | GTXComm | I am trying to test sipuri inbound; however, I keep getting a loopback detected from my IP address, Forwarding to Local/john@default. ?? Thanks if you can help. |
| 21:44.32 | GTXComm | I have outbound working now. |
| 21:49.33 | vinhdizzo | are there daily builds of asterisk available? |
| 21:50.06 | *** join/#asterisk kchehab (~kchehab@89.249.210.60) |
| 21:50.11 | kchehab | hi ppl |
| 21:51.08 | kchehab | how to keep a copy of fax file ,since a user is using a GW and a fax machine is connected to |
| 21:51.22 | p3nguin | paulc: The default is 0, which is the first entry. And the first entry uses vmlinuz-2.6.18-194.32.1.el5PAE. |
| 21:51.30 | kchehab | i AM USING iax MODEM |
| 21:51.51 | paulc | p3nguin: we just don't know then if it reports as such in uname -r or the shorter version, right? |
| 21:52.39 | p3nguin | paulc: Unless you booted up using the second entry in the menu, it appears that the version number is truncated. |
| 21:53.01 | p3nguin | paulc: It's not unusual to truncate a long version number like that. |
| 21:54.23 | paulc | I could get a reboot it at some point (but it's a production machine). I have no reason to think the 2nd entry would have been chosen - it's a headless blade, so my money'd be more on truncation of minor version subnumber etc |
| 21:54.42 | paulc | which brings us back to.. where would it expect to find the source? maybe in a xxxxx.32 subdirectory? |
| 21:55.46 | p3nguin | Check the package contents of kernel-devel-2.6.18-194.32.1.el5 |
| 21:58.01 | paulc | it puts it in /usr/src/kernels/2.6.18-194.32.1.el5-i686 ... which exists in /usr/src/kernels.. |
| 21:58.30 | paulc | tehre's also a 2.6.18-194.32.1.el5-PAE-i686 which is symlinked to from 2.6.18-194.32.1.el5PAE-i686 |
| 21:59.50 | p3nguin | Can you either specify the full path to the source, or find out what path it checks for when doing automated checking for source? |
| 22:00.56 | p3nguin | If I didn't feel like doing either of those things, I might symlink 2.6.18-194.32.1.el5PAE-i686 to /usr/src/linux |
| 22:01.10 | p3nguin | and then try again. |
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| 22:03.17 | paulc | the symlink didn't work - linking to just /usr/src/linux (from /usr/src/kernels/...192.31.1.el5PAE-i686) I still get told I don't have the kernel source installed by the DAHDI "make all" |
| 22:03.41 | p3nguin | Are you supposed to run configure? |
| 22:04.16 | paulc | for DAHDI, the "make all" builds it for you.. although there must be a manual route (the classic configure/make/make install) |
| 22:04.56 | p3nguin | So "./configure" is not in the instructions for installing dahdi? |
| 22:05.14 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 22:05.50 | paulc | not if you do the 'complete' install |
| 22:06.04 | p3nguin | What other method is there? |
| 22:06.47 | paulc | there's the "install the kernel modules + userspace tools by hand" method |
| 22:06.52 | paulc | the "make all" just does both for you |
| 22:06.55 | p3nguin | oh, okay |
| 22:08.17 | *** join/#asterisk capitan__ (~A@ppp-71-140-102-61.dsl.irvnca.pacbell.net) |
| 22:09.06 | p3nguin | Let me take a look and I'll get back to you in a few minutes. |
| 22:10.09 | paulc | ok cheers - appreciate it |
| 22:10.21 | paulc | it's gotta be something stupid like a symlink or something, but it's got me beat :( |
| 22:11.05 | *** join/#asterisk ariel_ (~chatzilla@99-1-236-49.lightspeed.miamfl.sbcglobal.net) |
| 22:15.12 | p3nguin | paulc: What does ls -l /lib/modules/`uname -r`/build tell you? |
| 22:18.01 | paulc | p3nguin: I've got a symlink in red pointing to something that doesn't exist... in ../../../usr/kernels/xxxx |
| 22:18.19 | p3nguin | Can you tell me exactly what it says? |
| 22:18.25 | paulc | fix that to point to the xxx-el5-PAE directory and it seems a lot happier in terms of compiling DAHDI |
| 22:18.28 | paulc | sure, 2 secs |
| 22:18.41 | paulc | lrwxrwxrwx 1 root root 48 Nov 2 2008 /lib/modules/2.6.18-194.el5PAE/build -> ../../../usr/src/kernels/2.6.18-194.el5-PAE-i686 |
| 22:19.20 | p3nguin | Does /usr/src/kernels/2.6.18-194.el5-PAE-i686 exist? ls -ld /usr/src/kernels/2.6.18-194.el5-PAE-i686 |
| 22:20.16 | paulc | it didn't.. but I figured I could symlink it to 2.6.18-194.32.1.el5-PAE-i686? |
| 22:21.02 | p3nguin | If it does not exist, go ahead and symlink the existing kernel directory to /usr/src/kernels/2.6.18-194.el5-PAE-i686 |
| 22:21.21 | paulc | then try my DAHDI "make all" again |
| 22:21.22 | paulc | ? |
| 22:21.25 | p3nguin | yes |
| 22:21.58 | paulc | p3nguin: it works! :) |
| 22:22.04 | p3nguin | If it works, go ahead and make checkinstall (or make install if you don't build packages). |
| 22:22.37 | p3nguin | Then after you install it, delete that symlink so we don't cause a problem for the package manager later. |
| 22:24.20 | paulc | sorted :) |
| 22:24.24 | paulc | that seems to have been just the ticket |
| 22:24.29 | paulc | thanks for your help -really appreciate it |
| 22:24.39 | p3nguin | Delete that symlink. |
| 22:24.47 | paulc | done :) |
| 22:25.06 | p3nguin | I'm glad it worked out. |
| 22:25.25 | paulc | and me - thanks again |
| 22:25.37 | p3nguin | No problem. |
| 22:27.14 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
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| 22:41.10 | GTXComm | I am testing sipuri inbound on my server. Would anyone want to try to dail me? I am having trouble with loopback.. dialing myself. |
| 22:43.46 | p3nguin | gtxcomm: Give me the URI and I will. |
| 22:46.47 | p3nguin | ... or... don't, and I won't. |
| 22:46.58 | GTXComm | john@gtxcommunications.com |
| 22:47.14 | GTXComm | I tried to message you, but I don't know how. :( |
| 22:47.52 | p3nguin | Did you get my call? |
| 22:47.55 | *** join/#asterisk wonderworld (~ww@port-92-201-5-80.dynamic.qsc.de) |
| 22:48.03 | GTXComm | no |
| 22:48.08 | p3nguin | Let me try again. |
| 22:48.11 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:48.48 | GTXComm | Yes, I see it. |
| 22:49.07 | GTXComm | [2011-03-14 17:50:08] NOTICE[10512]: chan_sip.c:17171 handle_request_invite: Call from '' to extension 'john' rejected because extension not found. |
| 22:49.40 | GTXComm | :) I put the extension in; however, I must not have it configured correct still :) thanks. |
| 22:50.07 | p3nguin | I'll help you fix it if you can be patient and follow my instructions. |
| 22:50.12 | GTXComm | Ok. |
| 22:50.49 | p3nguin | In your sip.conf, under the [general] heading, there is a context paramter. What value did you put for context? E.g., context=misc-calls |
| 22:51.03 | GTXComm | default |
| 22:51.14 | GTXComm | including all of my others.. |
| 22:51.37 | p3nguin | That's certainly not preferred, so that configuration design need to be changed immediately. |
| 22:51.42 | GTXComm | Well, at the bottom, I include primary.ael and that has all of my customer contexts. |
| 22:51.49 | GTXComm | Ok |
| 22:52.36 | p3nguin | In your dial plan, under the [default] context, I would prefer that you delete EVERYTHING under the [default] context. |
| 22:53.26 | p3nguin | Leave [default], but delete any extens and includes that are under it. |
| 22:53.33 | p3nguin | or move them |
| 22:53.46 | p3nguin | move would be a better term to use, I guess. |
| 22:54.11 | GTXComm | Let me copy my config to you so you can understand how I set this up. |
| 22:54.48 | p3nguin | It's not important to me. I just want you to have a better design. Since you don't know how, I'm going to help you do it. |
| 22:55.15 | GTXComm | Wonderful, It is a live hosted PBX. |
| 22:55.29 | p3nguin | sigh |
| 22:55.41 | GTXComm | :) I have around 40 clients on it.. :) |
| 22:55.48 | p3nguin | So we've got another AWESOMELY-CONFIGURED PBX on the internet. |
| 22:55.57 | p3nguin | Just what we need. |
| 22:56.41 | p3nguin | Okay, so I get the feeling you don't give a shit about how poorly you have configured it and you don't care about my suggestions to make it better... |
| 22:56.54 | GTXComm | No I do care. |
| 22:56.59 | p3nguin | so I'll skip all of the important steps and get right to the URI calling. |
| 22:57.28 | GTXComm | I thank you. |
| 22:57.46 | p3nguin | The context defined in sip.conf general is where URI calls will end up (unless you have defined a peer which matches the caller). |
| 22:58.05 | GTXComm | ok. |
| 22:58.27 | p3nguin | So you have to have exten => john,1,..... in that context if you are going to receive a call to john@yourhost |
| 22:59.26 | p3nguin | And that exten needs to do something useful. It could Dial() a phone, or run an application that reboots the computer... it just needs to do something. |
| 22:59.39 | GTXComm | Okay, in my extensions.ael this is where my default is. Then from Default, I include primary.ael / from primary ael I have all of my customers contexts on theim including from-sipuri |
| 22:59.44 | *** join/#asterisk envisean (~envisean@unaffiliated/envisean) |
| 23:00.03 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 23:00.21 | kchehab | is it feasible to save a copy for the fax sent by sip gateway with fax machine ,through digium fax ? |
| 23:00.29 | p3nguin | And I'm sure that you include your outbound context in it, too, so that allows me to call in and have access to call back out and make you pay for the calls. |
| 23:00.44 | p3nguin | kchehab: Sure, you can save the file. |
| 23:00.49 | GTXComm | Yes. |
| 23:01.20 | p3nguin | So now I am going to call UAE on your dime. |
| 23:01.23 | GTXComm | It is from a broker / that has level 3 / global crossings and such. |
| 23:01.28 | p3nguin | Then maybe Puerto Rico. |
| 23:02.09 | kchehab | p3nguin can you tell me how |
| 23:02.21 | kchehab | p3nguinusing wireshark ? |
| 23:02.32 | GTXComm | no. |
| 23:02.39 | p3nguin | What? Why would you use wireshark to do fax? |
| 23:02.54 | p3nguin | That's the strangest thing I have heard all day. |
| 23:02.59 | GTXComm | I have the dialplan set for include international.ael |
| 23:03.07 | p3nguin | And I hear some pretty strange things. |
| 23:03.28 | p3nguin | Great! I can call China mobile numbers and you can pay for it! Sweet! |
| 23:03.34 | kchehab | p3nguin not to send a fax its to save the fax file the sent by the user through its sip gateway |
| 23:04.02 | p3nguin | kchehab: Is Asterisk involved in the transmission of the fax? |
| 23:04.03 | kchehab | p3nguin like monitor function for voice |
| 23:04.09 | GTXComm | well, if you are not authenticated with my server how can you do that? |
| 23:04.28 | kchehab | p3nguin yes i am using digium fax |
| 23:04.50 | GTXComm | <--- not arguing, just learning. |
| 23:05.24 | p3nguin | gtxcomm: When you don't know what you're doing and you allow anonymous calls to come in, and then allow those anonymous callers to make outbound international calls... you'll pay for the calls I make from your system. |
| 23:05.39 | GTXComm | ok |
| 23:05.39 | p3nguin | If you want to fix it, I'll explain where you went wrong... like I was trying to do a while ago. |
| 23:05.51 | GTXComm | Ok. |
| 23:06.02 | p3nguin | Also, why are you using ael? |
| 23:06.17 | GTXComm | because that is what I learned in. |
| 23:07.09 | GTXComm | I looked at a dial plan that made sence to me, but I can convert now. |
| 23:07.13 | p3nguin | kchehab: I would think you can save a copy of the fax when it is transmitted through the dial plan via SendFax() or whatever app sends the file. |
| 23:08.20 | p3nguin | gtxcomm: The default context is used for internal asterisk things, so I recommend leaving it either blank or with a very specific strict set of rules in it. You shouldn't be doing any call routing through the default context. |
| 23:08.55 | GTXComm | kk |
| 23:08.59 | p3nguin | gtxcomm: The general sip.conf context should not be set to default, since we are going to leave default alone for asterisk internal stuff. |
| 23:09.27 | p3nguin | gtxcomm: So we choose a more suitable context for sip.conf general, such as context=misc-calls |
| 23:10.07 | GTXComm | so, what I would do, is change my context in my extensions.ael to something more suitable. |
| 23:10.12 | GTXComm | I can do that now. |
| 23:11.18 | p3nguin | gtxcomm: Then the dial plan context [misc-calls] has in it only rules that you want ANONYMOUS callers to have access to. You don't want those ANONYMOUS callers to have access to make any calls, so you won't likely include any other files or contexts until you understand the concept we're working on. |
| 23:12.59 | kchehab | p3nguin 1st how to save the file if its transmitted through sip Gw fax machine ? |
| 23:13.01 | p3nguin | gtxcomm: In the misc-calls context in the dial plan, you'll have extensions that you want unauthenticated users to have access to. exten => john,1,Dial(SIP/johns-phone,36) exten => john,n,Hangup() |
| 23:13.13 | GTXComm | Ok // I have changed the general context to something else, and everything is working |
| 23:14.39 | p3nguin | If you have that extension in the misc-calls context, which is the context you're using for unauthenticated callers, I would be able to dial you by SIP URI john@yourhost. |
| 23:15.26 | p3nguin | If I call you by SIP URI, you won't have a peer match for me, so it will use the context you've set under general. |
| 23:17.18 | p3nguin | Do you understand everything I'm saying? |
| 23:17.36 | GTXComm | Yes, but I think I have a problem with my setup. |
| 23:17.53 | p3nguin | Yeah? What's wrong? |
| 23:18.06 | GTXComm | under my main context... I include all of my other contexts. |
| 23:18.43 | p3nguin | I haVe no idea what "main context" is, but it's very likely a BAD idea to include everything else in it. |
| 23:19.01 | GTXComm | it is like extensions.conf. |
| 23:19.08 | GTXComm | but AEL. |
| 23:20.22 | GTXComm | So, I have include inbound.ael / international.ael and my clients dialplans, like c10000.ael c100001.ael excetera. I thought I would include a sipuri.ael |
| 23:20.32 | GTXComm | this way, I can sepporate it. |
| 23:21.00 | GTXComm | I know it is probably not good; however, I probably should think about a new way ;) |
| 23:21.14 | GTXComm | or And, not however. |
| 23:22.42 | GTXComm | so, in customer c10000.ael I can include differnent things like loacal dialing... longdistance and international dialing or remove them. |
| 23:23.42 | p3nguin | That part of your design sounds okay. |
| 23:23.43 | GTXComm | In my inbound dialplan I use jump to c10000-100 and then I have the _100 => dial. |
| 23:23.51 | GTXComm | Ok. |
| 23:24.00 | p3nguin | It's the part where you include everything in the default context that bothers me. |
| 23:24.54 | GTXComm | I guess that if I change my default context, to something else, everything else falls behind it, kind of like it is defaut anyway. |
| 23:25.10 | GTXComm | And I have done that. |
| 23:26.09 | GTXComm | I would be glad to send you a complete dialplan. |
| 23:26.25 | GTXComm | example. |
| 23:27.05 | p3nguin | The design just needs review. I could do it all for you, but obviously not for free on IRC. |
| 23:27.22 | GTXComm | Ok. |
| 23:27.35 | p3nguin | It may only take a little work to make the design a lot better. |
| 23:27.46 | GTXComm | Hey, no problem. |
| 23:27.50 | GTXComm | :) |
| 23:27.51 | p3nguin | Of you could put a lot of work into it and only make it a little better. |
| 23:28.20 | p3nguin | s/Of/Or/ |
| 23:28.34 | GTXComm | :) |
| 23:29.12 | GTXComm | I would be glad to make my system better and give back to the community. |
| 23:30.02 | GTXComm | P3 let me know if you would like to review my system outside of this forum. |
| 23:30.59 | p3nguin | I would like to, but I can't get to it today. I still have at least two projects that I have to complete before I can start a new one. |
| 23:31.20 | GTXComm | okay. |
| 23:32.34 | *** part/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com) |
| 23:33.27 | Corydon76-home | infobot: forget gs |
| 23:33.27 | infobot | Corydon76-home: i forgot gs |
| 23:34.50 | GTXComm | BTW Thank you for your advice. |
| 23:36.23 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:36.49 | GTXComm | ? |
| 23:37.13 | GTXComm | ? was an accident. |
| 23:37.57 | GTXComm | p3nguin test |
| 23:38.06 | p3nguin | Yes? |
| 23:38.27 | p3nguin | I was in the middle of a yum update and I wasn't paying attention. |
| 23:38.28 | GTXComm | I wanted to know how to private message. lol so I put say. |
| 23:38.39 | GTXComm | test |
| 23:38.44 | GTXComm | doesnt work. |
| 23:38.46 | GTXComm | lol |
| 23:39.02 | GTXComm | Okay, I am going to stop chatting / noob IRCer. |
| 23:39.56 | p3nguin | Well, you use /query to open a query window or /msg to send a private message... but I don't accept private messages from people who have not told me they are going to send me private messages so I can enable it for him/her. |
| 23:40.15 | GTXComm | ahh. |
| 23:40.27 | p3nguin | I'll enable it for you so you can try it out. |
| 23:42.35 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:44.58 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
| 23:51.35 | *** join/#asterisk hairyraven (~nobody@95.72.232.167) |
| 23:55.30 | *** join/#asterisk eugeneoden (~goden@63.133.138.10) |