| 00:04.21 | *** join/#asterisk wwalker (~wwalker@208.92.232.27) |
| 00:05.45 | wwalker | If my provider is using IP for authentication , and he's got 24 peers he may connect to me from, do I have to create 24 peer stanzas in sip.conf? |
| 00:05.48 | *** join/#asterisk AlfE_ (~quassel@83-215-36-12.bruck.dyn.salzburg-online.at) |
| 00:08.14 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 00:08.42 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:08.42 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:13.05 | doogienz | possibly a host=dynamic with a permit mask |
| 00:14.12 | doogienz | And just don't have a username or secret in the config |
| 00:16.33 | WIMPy | also has 36 peers for one of his ITSPs. |
| 00:38.09 | *** join/#asterisk SaRSAeOL (~sarsaeol@66-113-93-170.rev.ibsinc.com) |
| 00:46.29 | *** join/#asterisk Johnny- (~John@c-68-81-19-251.hsd1.pa.comcast.net) |
| 00:50.08 | p3nguin | wwalker: http://pastebin.com/EaqQC2s1 That's the easy way. |
| 00:50.36 | dmz | hey, just updated an old asterisk box and need to change queue/agent away from agentcallbacklogin; anyone have a good url to look into? |
| 00:54.16 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
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| 01:56.04 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 02:12.36 | wwalker | p3nguin: that pastebin doesn't seem to exist |
| 02:13.49 | p3nguin | wwalker: It did when I pasted it. |
| 02:14.49 | p3nguin | wwalker: Here it is again: http://pastebin.com/v2UBxVsn |
| 02:15.55 | wwalker | p3nguin: thank you. what asterisk version started supporting that templating syntax? |
| 02:16.48 | p3nguin | At least 1.4. Maybe earlier, but I started in 1.4, so I don't know for sure. |
| 02:20.19 | wwalker | started at pre 1.0 where the answer to every question was "try cvs update" :) Then I took a few years off of voip. |
| 02:20.29 | wwalker | time to do it again. |
| 02:28.39 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 02:32.40 | *** join/#asterisk koffel (~whome@c-68-83-254-134.hsd1.pa.comcast.net) |
| 02:32.46 | koffel | hello |
| 02:34.16 | *** join/#asterisk SaRSAeOL (~sarsaeol@66-113-93-170.rev.ibsinc.com) |
| 02:35.31 | *** join/#asterisk AlexR1 (~AlexR1@mrshel7.lnk.telstra.net) |
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| 02:45.57 | sunfone | \join #openvpn-as |
| 02:46.13 | AlexR1 | Really?.... Lol. |
| 02:51.05 | sunfone | bummer typo ;) |
| 02:54.18 | koffel | has anyone used asterisk -gui? |
| 02:55.20 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 03:34.35 | *** join/#asterisk gajini (~root@61.12.17.171) |
| 03:48.06 | dmz | argh, i forgot 1.6 didn't use | anymore; sigh |
| 03:52.50 | p3nguin | Not to mention all the 1.6 branches are EOL. |
| 04:14.03 | *** join/#asterisk The_Ball (~The_Ball@122.150.108.38) |
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| 05:31.59 | din3sh | mrning ppl |
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| 06:05.28 | *** join/#asterisk lkthomas (~lkthomas@59.152.236.158) |
| 06:05.41 | lkthomas | hey guys |
| 06:05.49 | lkthomas | we are suffer from strange issue |
| 06:06.11 | lkthomas | when we use traditional fax machine to fax, asterisk have dial tone without problem |
| 06:06.40 | lkthomas | but when we use PRI line to dial, the dial tone does not appear and drop within 1 second |
| 06:06.45 | lkthomas | hylafax showing no carrier |
| 06:10.42 | *** join/#asterisk zro (~zro@wikimedia/zro) |
| 06:11.05 | zro | ha! I made it start! I installed it and made it start without an obscure error! |
| 06:11.32 | zro | ... alhtough now I feel dumb that it took me like 3 times. wtf. I mut ave fat fingered something along the way. |
| 06:12.00 | *** join/#asterisk xpot-mobile (~james@mb30436d0.tmodns.net) |
| 06:18.17 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:18.20 | schmidts | good morning |
| 06:18.48 | *** join/#asterisk RogerH (545c6276@gateway/web/freenode/ip.84.92.98.118) |
| 06:22.42 | lkthomas | schmidts: any idea ? |
| 06:22.54 | schmidts | lkthomas 42 |
| 06:23.13 | schmidts | i dont have any idea what you are talking about ;) |
| 06:25.57 | *** join/#asterisk oej (~olle@office.ipvision.dk) |
| 06:27.34 | lkthomas | oh shit |
| 06:27.35 | lkthomas | sorry |
| 06:27.36 | lkthomas | you miss that |
| 06:28.45 | lkthomas | we are using T1 line, and when fax coming from analog fax machine, fax could receive, but the source is using PRI to fax out, I receive no carrier problem with the hylafax, and source fax system admin saying that the dial tone is strange as well |
| 06:29.13 | lkthomas | , I was able to capture audio and you can barely hear the CED tone <--- that's what the fax server admin saying |
| 06:30.57 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 06:31.38 | *** join/#asterisk jsjc (~Adium@6.Red-83-59-183.dynamicIP.rima-tde.net) |
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| 06:40.24 | *** join/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 06:47.57 | lkthomas | schmidts: any idea ? |
| 06:59.39 | zro | when i enable autoload for modules, asterisk -c ends and says "....Illegal instruction". I can't seem to interpret -vvv to understand what module is causeing this. how should i figure it out? |
| 07:02.21 | schmidts | lkthomas have you tried to record (monitor) both sides of the call? from the hylafax side and also outgoing over the t1 line? if you monitor the call and didnt split it up, you could playback both sides at the same time and just listen if you hear any difference |
| 07:05.08 | schmidts | zro try strace |
| 07:05.29 | *** join/#asterisk s[x]_ (~sx]@ppp59-167-154-113.static.internode.on.net) |
| 07:05.32 | lkthomas | I don't know how to record down the signal tone |
| 07:05.46 | schmidts | lkthomas use the monitor application in asterisk |
| 07:05.52 | lkthomas | which one |
| 07:06.01 | schmidts | which asterisk version ;) |
| 07:06.20 | lkthomas | 1.4.x |
| 07:06.35 | lkthomas | 1.4.24 |
| 07:06.36 | schmidts | i dont know much about 1.4 but imho the normal Monitor app should work |
| 07:13.28 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
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| 07:36.27 | *** join/#asterisk din3sh (~din3sh@41.212.203.169) |
| 07:38.23 | zro | so i tried loading asterisk, and then writing a script to load each module in turn and give me some output if it wouldn't load and then added noload to modules.conf for each, but im still getting crash when i autoload modules, any ideas? |
| 07:38.27 | zro | http://sprunge.us/WZAa?sh |
| 07:39.19 | schmidts | zro what OS do you use? if its some kind of debian please just use apt-get install strace and start asterisk with it like strace asterisk -cvvv |
| 07:39.28 | schmidts | then you can see at which modul it will fail to load |
| 07:39.47 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 07:44.12 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 07:45.18 | zro | thanks, schmidts! that sounds like a great idea... although now i gotta figure out how to read this output... |
| 07:46.10 | schmidts | search for print thats what you normally see on your bash when starting asterisk |
| 07:50.42 | zro | im not following you, schmidts, what am I looking for here? |
| 07:52.13 | *** join/#asterisk jks (~jks@4810ds1-ns.0.fullrate.dk) |
| 07:56.53 | zro | using a slightly modified version of that hackey script i posted earlier, i came up with these: http://sprunge.us/fgPi which appears to stop the crashing |
| 08:05.06 | *** join/#asterisk shamelessn00b (~bwahahaha@182.177.34.251) |
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| 08:31.35 | *** join/#asterisk ilj (~ilj@sourcemage/grimoire/apprentice/ilj) |
| 08:34.21 | ilj | hi, I have this box with * v.1.6.2.19, and I'm trying to figure out the location in filesystem path of AGI scripts used in extensions.conf. It uses FASTAGI_HOST=localhost variable in [globals] section of extensions.conf, and the scripts are called like this agi://${FASTAGI_HOST}/newCall.agi. The ... |
| 08:34.26 | ilj | ... problem is I failed to find this newCall.agi anywhere on the filesystem. Any pointers to help me out? |
| 08:39.07 | kaldemar | ilj: see where TCP localhost:4573 leads to |
| 08:40.16 | ilj | says it's localhost |
| 08:40.17 | ilj | # telnet localhost 4573 |
| 08:40.17 | ilj | Trying 127.0.0.1... |
| 08:40.17 | ilj | Connected to localhost.localdomain (127.0.0.1) |
| 08:40.17 | Nugget | telnet is eeeeeeevil! |
| 08:40.26 | ilj | ahaha |
| 08:40.30 | ilj | you knew! |
| 08:40.45 | ilj | okay how else I can establish that? :) |
| 08:42.16 | *** join/#asterisk bulkorok (~bulkorok@217.110.197.225) |
| 08:43.27 | kaldemar | ilj: the port might be redirected somewhere else. |
| 08:45.55 | *** join/#asterisk shamelessn00b (~bwahahaha@182.177.70.242) |
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| 09:05.52 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 09:09.07 | din3sh | telnet is no evil |
| 09:21.17 | *** join/#asterisk shamelessn00b (~bwahahaha@182.177.34.251) |
| 09:23.15 | *** join/#asterisk shamelessn00b (~bwahahaha@182.177.119.140) |
| 09:26.52 | shamelessn00b | can dial() be non-blocking, I want A party to call B party and bridge them and then both should be able to hear whatever that is going in the dialplan |
| 09:27.08 | shamelessn00b | dial() or confbridge() |
| 09:27.17 | shamelessn00b | can they be non-blocking |
| 09:27.52 | shamelessn00b | there is 'G' parameter in the dial app, but it doesn't bridge the parties if used |
| 09:35.13 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
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| 09:49.49 | *** join/#asterisk krotos (~d3v1l@host158-27-dynamic.8-87-r.retail.telecomitalia.it) |
| 09:49.54 | krotos | hi all |
| 09:52.11 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 09:52.53 | zro | anyone use teliax? |
| 09:53.17 | krotos | no.. |
| 09:55.10 | krotos | i'have a strange behavior in asterisk 1.8.11 with callerpres. In asterisk 1.8.8 if i use callerpres(name-pres) and num-pres =prohib on the other side the From header is something like From: <sip:asterisk@192.168.1.123>. |
| 09:55.37 | krotos | on the new asterisk 1.8.11 if i use prohib, the From header is From: <sip:Anonymous@anonymous.invalid> |
| 10:01.08 | *** join/#asterisk catphish (~charlie@2001:9d8:2005:11:2597:8e2d:435:3621) |
| 10:04.49 | *** join/#asterisk chris-NB (~chris@fw.commpany.at) |
| 10:07.52 | catphish | is any configuration required for asterisk to pass through T38 SIP to SIP (locally bridged)? |
| 10:08.39 | krotos | catphish: try t38pt_udptl directive |
| 10:08.44 | krotos | in your sip.conf |
| 10:08.56 | krotos | you can set it to "yes", and "yes,redundancy" |
| 10:09.10 | catphish | i have it set to yes right now |
| 10:09.51 | catphish | is there any way if i can tell if a call is being passed through correctly as T38 |
| 10:10.15 | *** join/#asterisk Lantizia (~lantizia@cpc25-stok15-2-0-cust110.1-4.cable.virginmedia.com) |
| 10:11.03 | catphish | i have an asterisk server which is able to receive faxes fine, however i have a user who has connected another asterisk server, and faxes are failing to be received there after passing through my host |
| 10:11.06 | Lantizia | Does anyone know if there is a list of what is said (i.e. the text) for the core and extra sounds packages? |
| 10:11.09 | catphish | just trying to work out where the problem is |
| 10:14.11 | krotos | catphish: try to debug udptl stream and wathc the configuration (sip.conf) for the other * machine connected to your |
| 10:16.46 | catphish | the configuration of both asterisks are identical, the only difference is that one passes through the other, whereas the first is directly connected upstream |
| 10:18.34 | catphish | is there a way to enable additional debug output on it |
| 10:18.43 | catphish | i'm afraid i don't understand the process very well |
| 10:20.28 | krotos | udptl set debug on |
| 10:21.01 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 10:26.14 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:cf3:a1af:777a:1d19) |
| 10:27.09 | *** join/#asterisk scubes13 (~scubes13@cpe-024-168-253-067.sc.res.rr.com) |
| 10:27.54 | zro | I'm at a point where I can run asterisk -c and it give me a CLI, but hten when I call the number, it crashes and prints "Illegal instruction". I'm not sure how to trouble shoot from here... any pointers? |
| 10:31.23 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 10:34.37 | zro | and i cant for the life of me get a more robust error |
| 10:41.35 | *** join/#asterisk raubvogel (~raubvogel@wifi.shop.monetra.com) |
| 10:48.46 | catphish | krotos: thanks, i'll try that |
| 10:54.30 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 10:56.45 | Lantizia | with a callout file... I know something like this... Channel: SIP/trunk/01234000000 Extension: 200 will call the number and once answered try and connect them to 200 |
| 10:57.07 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 10:57.08 | Lantizia | but how can I get it to dial out on 01234000000 like it already does - and connect the call to another external number? and not an extension? |
| 10:59.17 | kaldemar | Lantizia: be more specific on "like it already does". |
| 10:59.41 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 11:01.30 | kaldemar | you can only connect the given channel to an extension or an application. if you want it to dial somewhere connect it to an extension that does so or use "Application: Dial" and "Data: <dial_args>". |
| 11:01.39 | zro | this is the end of my "$ strace asterisk -c" if this helps anyone help me: http://sprunge.us/VVeG |
| 11:02.30 | kaldemar | zro: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 11:05.09 | zro | thanks kaldemar, reading now |
| 11:05.41 | Lantizia | ah cool thanks kaldemar |
| 11:10.15 | krotos | anyone use mysql and odbc / res_odbc? |
| 11:10.33 | krotos | i've got a problem with timeout |
| 11:11.00 | krotos | and every time asterisk have to reconnect to mysq..and the call slow it's flow |
| 11:15.54 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 11:18.04 | cusco | hi |
| 11:18.31 | Chainsaw | Hello. |
| 11:20.50 | cusco | I have a question... |
| 11:21.04 | WIMPy | BFG would be better. |
| 11:21.04 | bulkorok | ~ask |
| 11:21.05 | 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. |
| 11:21.05 | cusco | when a peer goes in Queue() |
| 11:21.27 | cusco | he may press some dtmf digit and get to that exten in the dialplan |
| 11:21.52 | cusco | how do I make it so, to only allow that peer to go there after X seconds ? |
| 11:22.22 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 11:22.36 | cusco | do I need to Queue(,timeout) and Queue() again ?= |
| 11:22.47 | bulkorok | does the peer need to be in the queue immediatly? just put a wait(5) for five seconds before Queue() |
| 11:24.14 | cusco | yes he needs |
| 11:24.22 | bulkorok | why? |
| 11:24.25 | cusco | and I want certain option to be onlya vailable after a minute |
| 11:24.41 | cusco | what do you mean by why? |
| 11:25.08 | bulkorok | for what do you use queue? just announcing? |
| 11:25.20 | zro | is this helpful to help me? http://sprunge.us/UKce |
| 11:25.33 | cusco | bulkorok: to make the call ring on queue members |
| 11:26.19 | cusco | (isn't that what queues are for?) |
| 11:26.25 | bulkorok | and after one minute the caller needs to be able to make the dtmf-thing... I see... (for sure) |
| 11:27.10 | cusco | I was trying to avoid calling Queue() twice |
| 11:27.27 | WIMPy | zro: Looks like you used a binary that isn't compatible to the system you're using it on. |
| 11:28.50 | bulkorok | cusco: do you use option K ? isn't it for parking?! |
| 11:29.22 | zro | ya, i think it /might/ be that I'm trying to get asterisk working on debian 6 inside openVZ, but the kernel of the host is 2.6.27 |
| 11:29.27 | zro | is that not gunna work? |
| 11:29.37 | cusco | no, we're not using parking |
| 11:30.20 | bulkorok | however... I think you have to use Queue() twice... |
| 11:30.49 | WIMPy | zro: illegal instruction is not a kernel thing, it's a CPU (emulation) thing. |
| 11:31.06 | WIMPy | Like using i686 binaries on a celeron. |
| 11:31.37 | bulkorok | zro: take a real machine. VM is hell for any voice software... |
| 11:35.29 | zro | bulkorok: im doing this on a tiny little VPS provider. I'm just trying to make a little call in thing where you hit a number and get a recording. |
| 11:36.12 | *** join/#asterisk fornax (~fornax@85.183.53.64) |
| 11:37.07 | *** join/#asterisk asr33 (~asr33@dsl-173-206-171-179.tor.primus.ca) |
| 11:37.31 | fornax | Hi, what is the best way to start / stop an external service using an asterisk extension? I would like start start /etinit.d/hostapd when I dial *11 and /etc/init.d/hostapd when I dial *10. |
| 11:38.17 | bulkorok | fornax: CLI> core show application system |
| 11:38.57 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 11:40.51 | fornax | I have System(/etc/init.d/hostapd start); in my ael but this does not seem to work, but I think i should continue debugging first. I just thought that there might be a better way or some wrapper explicitly for system scripts... |
| 11:42.33 | bulkorok | fornax: does the script has execute-rights? |
| 11:42.40 | bulkorok | mmh... do... |
| 11:43.30 | fornax | bulborok: this is the other problem, I have to give the script rights to execute it from asterisk |
| 11:43.55 | bulkorok | fornax: then start asterisk with the same rights |
| 11:44.09 | bulkorok | well with the same user... |
| 11:44.38 | fornax | bulborok: absolutely no good idea for a hardened system. I think I will have to write some wrapper for it |
| 11:45.22 | bulkorok | then turn around and make the script executable for the asterisk-user |
| 11:45.32 | fornax | bulkorok: Do you maybe have an idea how to read the status of the service and give it back as voice prompt? |
| 11:46.22 | blitzrage | fornax: the best way is really to use AGI(), but you could also use SHELL() which can return data |
| 11:46.33 | blitzrage | System() can only execute the script, not return anything to Asterisk |
| 11:47.25 | bulkorok | fornax: look for asterisk text to speech... |
| 11:48.10 | bulkorok | fornax: write a script that reads the status, and then executes an originate over ami |
| 11:48.13 | fornax | blitzrage: Okay, I thought that I have to do it this way. I will have a further look in the evening |
| 11:48.59 | fornax | bulkorok: Reading the status is the problem, tts is not even required, I simply have to output yes or no, related to the state |
| 11:49.12 | fornax | I will try to solve it ion the evening, thank you for your help. cu |
| 11:50.11 | blitzrage | basically he just needs to return 0 or 1 and then use the built in prompts (for anyone else who cares) |
| 11:51.24 | bulkorok | mmh... I have to |
| 11:51.34 | bulkorok | go deeper in this idea :) |
| 11:55.29 | bulkorok | can someone tell me what this means after ReceiveFax()?: Unexpected DCN after requested retransmission |
| 11:56.05 | blitzrage | if you figure out what DCN is the rest of the line is pretty self explanatory |
| 11:56.50 | *** join/#asterisk Bullmoose (~Bullmoose@75-174-78-5.bois.qwest.net) |
| 11:57.40 | bulkorok | mmh... seems to be i the t.38 protocoll... |
| 11:57.44 | bulkorok | in the |
| 11:59.51 | blitzrage | that would make sense as it appears you're attempting to use ReceiveFax9) |
| 12:00.53 | *** join/#asterisk hehol (~hehol@217.9.101.222) |
| 12:01.27 | bulkorok | DCN is a T30 signal message and means "Disconnect"... well.. someone plug off the cord :) |
| 12:01.31 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
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| 12:54.24 | Kobaz | hmmm |
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| 12:55.30 | din3sh | mmmh |
| 12:55.52 | bulkorok | hmmhmm |
| 12:57.36 | WIMPy | mooo |
| 12:58.43 | Kobaz | PickupChan doesn't write any success/failure variables |
| 13:01.04 | din3sh | booo |
| 13:10.57 | catphish | is there anything (apart from t38pt_udptl=yes) that could cause t38 passthrough to work or not? for example reinvite or codec settings? |
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| 13:46.00 | Kobaz | yay |
| 13:46.03 | Kobaz | fun |
| 13:46.11 | Kobaz | iax2 reload caused a deadlock in res_config pgsql |
| 13:47.50 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 13:49.34 | mjordan | Kobaz: looks like it can happen from chan_sip as well - ASTERISK-19767 |
| 13:53.03 | Kobaz | i think the solution is to make a separate pg connection for iax/sip/etc reload |
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| 14:33.19 | shamelessn00b | hi guys, can the dial and confbridge app be made non-blocking so they don't block dialplan execution once called? |
| 14:34.06 | _Corey_ | shamelessn00b: What are you trying to achieve? Dialplan execution will continue after they terminate... |
| 14:35.19 | shamelessn00b | _Corey_: I want to get channel audio seperately for each party while they are bridged |
| 14:36.00 | blitzrage | shamelessn00b: don't cross post between channels, and #asterisk-dev is not level 2 support |
| 14:36.10 | _Corey_ | shamelessn00b: Sorry, I don't understand what you mean |
| 14:36.26 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
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| 14:36.55 | [TK]D-Fender | shamelessn00b, Every app is blocking <- |
| 14:36.57 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 14:37.18 | [TK]D-Fender | shamelessn00b, You don't issue a regular Dial() and get to continue on while being in a call either |
| 14:37.23 | shamelessn00b | apps like mixmonitor etc don't block the dialplan |
| 14:37.29 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 14:38.02 | blitzrage | ok... |
| 14:38.07 | blitzrage | define what you're really trying to do |
| 14:38.15 | blitzrage | because what you said doesn't make sense |
| 14:38.15 | _Corey_ | shamelessn00b: Well, even though it's not, Monitor is more like setting a variable in the scheme of things |
| 14:38.31 | shamelessn00b | what about app_jack |
| 14:38.56 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 14:39.10 | shamelessn00b | they provide you ports to write to the channel and read from the channel (audio frames) while the dialplan executes |
| 14:39.22 | blitzrage | what are you really trying to do? |
| 14:39.26 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 14:39.31 | shamelessn00b | I want something similar but on a somewhat larger scale :) |
| 14:39.47 | [TK]D-Fender | shamelessn00b, <blitzrage> what are you really trying to do? <- |
| 14:39.53 | blitzrage | I still don't understand what you're looking to happen and why |
| 14:39.56 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 14:39.57 | blitzrage | elaborate |
| 14:40.01 | _Corey_ | shamelessn00b: uh, ok... you still haven't explained a problem |
| 14:40.20 | [TK]D-Fender | Well.. he has explained an aspect of what he wants... without explaining why |
| 14:40.26 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 14:40.33 | *** mode/#asterisk [+b *!D-Boy@*] by blitzrage |
| 14:40.47 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 14:40.51 | blitzrage | huh... |
| 14:40.52 | shamelessn00b | I'm working on an in-call translation application |
| 14:41.05 | shamelessn00b | and want to support multi-user conferencing |
| 14:41.25 | _Corey_ | shamelessn00b: ok, so multi-user conferencing is already there |
| 14:41.31 | blitzrage | Qwell: ping |
| 14:41.37 | shamelessn00b | _Corey_: I know |
| 14:41.46 | shamelessn00b | but lets say they all speak different langauges |
| 14:42.13 | _Corey_ | shamelessn00b: Maybe if you explain the call flow it will make more sense |
| 14:42.25 | shamelessn00b | I want to sit in the middle, get each individual's speech, translate it into the remaining people's langauges play it back to them |
| 14:42.26 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 14:42.45 | *** mode/#asterisk [+b *!*D-Boy@*unaffiliated/cain] by blitzrage |
| 14:42.57 | *** kick/#asterisk [D-Boy!~Leif@asterisk/documenteur-extraordinaire/blitzrage] by blitzrage (Msg an op when you fix your connection) |
| 14:42.57 | Chainsaw | shamelessn00b: That assumes reliable machine translation and reliable speech recognition. |
| 14:43.01 | _Corey_ | so you want to have a translator in the middle of a call between two parties who don't speak to each other directly? |
| 14:43.22 | _Corey_ | shamelessn00b: Please don't send private messages |
| 14:43.26 | Chainsaw | shamelessn00b: A fascinating idea, but I'm not sure current technology is ready for that. |
| 14:43.35 | shamelessn00b | I already have it working |
| 14:43.36 | shamelessn00b | for 2 users |
| 14:43.39 | shamelessn00b | interprephone |
| 14:44.17 | shamelessn00b | its near real time, takes about 2-3 secs to say back translated speech |
| 14:44.44 | shamelessn00b | right now I'm using the manager interface and ahi |
| 14:45.09 | blitzrage | I think it really comes down to a clever setup of a conference bridge with listen/speaking flags |
| 14:45.37 | blitzrage | all 4 participants in a conference bridge with the people setup to listen to certain people |
| 14:45.41 | blitzrage | and speaking to certain channels |
| 14:46.04 | shamelessn00b | problem is, I want them all to hear the original speech |
| 14:46.11 | blitzrage | ok |
| 14:46.21 | shamelessn00b | and then selective people to hear the translated/synthesized version |
| 14:46.22 | _Corey_ | then it's even easier... |
| 14:46.24 | blitzrage | like I said... clever usage of conferences |
| 14:46.56 | shamelessn00b | no, I mean, lets say there's an english speaker, a french and an italian |
| 14:47.00 | blitzrage | if the original speakers are in the conference, then just have the translators joined into the conference, hearing that same speech, and only talk to a specific person |
| 14:47.16 | shamelessn00b | italian says something, I recognize and translate/synthesize it in english and french |
| 14:47.17 | blitzrage | almost trivial |
| 14:47.25 | blitzrage | right... |
| 14:47.25 | shamelessn00b | they are all in the same conference |
| 14:47.29 | blitzrage | exactly |
| 14:47.43 | blitzrage | just use marked users and such |
| 14:48.09 | shamelessn00b | now the person who speaks english would hear the originally spoken italian |
| 14:48.20 | shamelessn00b | and synthesized translated english |
| 14:48.28 | blitzrage | yep they sure would |
| 14:48.36 | shamelessn00b | and the french would hear italian and synthesized french |
| 14:48.48 | blitzrage | look at conferences and marking users and listen-only, and speak-only flags |
| 14:49.11 | blitzrage | you may need to setup a couple conferences that are joined together, not sure the exact logistics, but it should all be possible |
| 14:49.12 | shamelessn00b | but I can't mark two users simultaneously |
| 14:49.19 | blitzrage | use multiple conferences |
| 14:49.22 | blitzrage | then join them together |
| 14:49.30 | shamelessn00b | :o |
| 14:49.58 | blitzrage | this really has nothing to do with your original request of continuing in the dialplan once bridged (which isn't possible) |
| 14:50.07 | blitzrage | you gotta be clever with Asterisk sometimes |
| 14:50.11 | blitzrage | good luck! |
| 14:50.35 | shamelessn00b | blitzrage: it is possible, if I write my own module, which is why I also posted in asterisk-dev |
| 14:50.46 | shamelessn00b | which someone was objecting to on this channel... |
| 14:50.52 | blitzrage | which was me |
| 14:50.59 | blitzrage | you hadn't asked a development question |
| 14:51.34 | shamelessn00b | can dial or confbridge be implemented in a thread |
| 14:51.48 | file | it's not possible without fundamentally changing core aspects of Asterisk to have it continue in the dialplan once bridged, even in your own module |
| 14:52.02 | shamelessn00b | so it returns immediately? |
| 14:53.02 | Lantizia | Does anyone else here think using individual OpenVPN connections for each remote SIP user of a phone system is a tad overkill when the only thing on the end of the VPN is the phone system - and SRTP/ZRTP are perfectly fine. |
| 14:53.21 | blitzrage | uh ya :) |
| 14:53.30 | blitzrage | that is a pretty significant amount of overhead I would think |
| 14:53.32 | Lantizia | http://sysadminman.net/blog/2012/product-launch-vpnpbx-asterisk-vps-openvpn-3963 |
| 14:53.50 | Lantizia | My boss thinks this product is great - I just wrote a huge rant back at him :P |
| 14:54.02 | _Corey_ | talk about re-inventing the wheel |
| 14:54.18 | blitzrage | I guess it's useful if you have devices that cna't speak SIP TLS and SRTP |
| 14:54.26 | _Corey_ | I guess if you have phone s that don't do srtp... |
| 14:54.29 | blitzrage | I'm picturing an ATA or something |
| 14:54.34 | _Corey_ | lol yeah |
| 14:54.58 | blitzrage | I mean I can see the uses... but... ya |
| 14:55.19 | blitzrage | yo dawg, I heard you like encryption so we put encryption around your encryption |
| 14:55.23 | catphish | my fax issues were caused by not allowing UDP 4xxx on my firewall, had no idea UDPTL used a separate range :) |
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| 14:57.54 | shamelessn00b | blitzrage: the main issue with confbridge would be streaming audio over to the recognition engine |
| 14:58.03 | blitzrage | sure |
| 14:58.44 | shamelessn00b | how can I overcome that? |
| 14:59.35 | shamelessn00b | playing back would be simple using local channels |
| 15:05.25 | [TK]D-Fender | shamelessn00b, So far I still don't see what the dialplan will be doing in the background for this... |
| 15:05.33 | [TK]D-Fender | shamelessn00b, As for getting the streams, just Originate another channel to chanspy into the caller's channel for your hook. |
| 15:08.01 | *** mode/#asterisk [-bbbb *!*D-Boy@*unaffiliated/cain gavimobile!*@* *!*@46.30.120.1 *!*admin@*.37.96.58.static.exetel.com.au] by blitzrage |
| 15:08.01 | *** mode/#asterisk [-bbbb *!*@200-171-49-211.dsl.telesp.net.br *!*scud@69.73.19.* *!*cast@99.246.137.* me|ong!*@*] by blitzrage |
| 15:08.01 | *** mode/#asterisk [-bbbb *!*@adsl-68-20-35-251.dsl.chcgil.ameritech.net *!*roy@cFDB1BF51.dhcp.bluecom.no *!~diego@190.23.77.213 *!*@v58.sgsvr.com] by blitzrage |
| 15:08.01 | *** mode/#asterisk [-bbbb *!*manjiki@*.cern.ch *!*@*i-evolve.net *!*@8.224.32.0/23 *!*@ti0013a380-dhcp0802.bb.online.no] by blitzrage |
| 15:08.01 | *** mode/#asterisk [-bbb *!*@gateway/web/freenode/ip.72.237.213.* *!*@pool-98-118-168-221.bflony.fios.verizon.net *!*@*98.118.168.221] by blitzrage |
| 15:08.08 | blitzrage | fuckin' xchat |
| 15:08.17 | *** mode/#asterisk [-b *!D-Boy@*] by blitzrage |
| 15:08.18 | Kobaz | yeap |
| 15:08.26 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 15:08.40 | *** mode/#asterisk [+b gavimobile!*@*] by blitzrage |
| 15:08.46 | *** mode/#asterisk [+b *!*@46.30.120.1] by blitzrage |
| 15:08.52 | *** mode/#asterisk [+b *!*admin@*.37.96.58.static.exetel.com.au] by blitzrage |
| 15:08.57 | *** mode/#asterisk [+b *!*@200-171-49-211.dsl.telesp.net.br] by blitzrage |
| 15:09.03 | *** mode/#asterisk [+b *!*scud@69.73.19.*] by blitzrage |
| 15:09.09 | *** mode/#asterisk [+b *!*cast@99.246.137.*] by blitzrage |
| 15:09.15 | *** mode/#asterisk [+b me|ong!*@*] by blitzrage |
| 15:09.21 | *** mode/#asterisk [+b *!*@adsl-68-20-35-251.dsl.chcgil.ameritech.net] by blitzrage |
| 15:09.27 | *** mode/#asterisk [+b *!*roy@cFDB1BF51.dhcp.bluecom.no] by blitzrage |
| 15:09.39 | *** mode/#asterisk [+b *!~diego@190.23.77.213] by blitzrage |
| 15:09.39 | Qwell | blitzrage: what have you done?! |
| 15:09.39 | p3nguin | Didn't take long for d-boy to show up. |
| 15:09.40 | *** mode/#asterisk [+b *!*@v58.sgsvr.com] by blitzrage |
| 15:09.46 | *** mode/#asterisk [+b *!*manjiki@*.cern.ch] by blitzrage |
| 15:09.51 | *** mode/#asterisk [+b *!*@*i-evolve.net] by blitzrage |
| 15:09.57 | D-Boy | lol p3nguin :p i wasn't banned for the same reason i guess :p |
| 15:09.59 | *** mode/#asterisk [+b *!*@8.224.32.0/23] by blitzrage |
| 15:10.02 | D-Boy | (connection flood) |
| 15:10.12 | *** mode/#asterisk [+b *!*@ti0013a380-dhcp0802.bb.online.no] by blitzrage |
| 15:10.18 | *** mode/#asterisk [+b *!*@gateway/web/freenode/ip.72.237.213.*] by blitzrage |
| 15:10.24 | *** mode/#asterisk [+b *!*@pool-98-118-168-221.bflony.fios.verizon.net] by blitzrage |
| 15:10.30 | *** mode/#asterisk [+b *!*@*98.118.168.221] by blitzrage |
| 15:10.34 | blitzrage | p3nguin: I unbanned him on purpose |
| 15:10.39 | p3nguin | oh |
| 15:10.39 | blitzrage | apparently fixed his connection |
| 15:10.46 | blitzrage | and then I unbanned everyone too :) |
| 15:10.59 | blitzrage | banned restored |
| 15:11.32 | pabelanger | blitzrage: lol |
| 15:11.43 | blitzrage | pabelanger: don't make me ban you |
| 15:17.09 | kaldemar | now that was interesting. :) |
| 15:21.48 | *** join/#asterisk sp00kz (~ilubj00@unaffiliated/sp00kz) |
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| 15:27.00 | p3nguin | Forgot to ban nighty, I guess. |
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| 15:28.56 | p3nguin | sets mode +b *!*@static-68-179-124-161.ptr.terago.net |
| 15:29.51 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
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| 15:30.43 | verywiseman | how can i show sip channels in asterisk 1.10 ? |
| 15:30.52 | Kobaz | sip show channels |
| 15:32.44 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 15:32.52 | verywiseman | Kobaz, in asterisk 1.10 there is no sip show` |
| 15:33.08 | cusco | hey |
| 15:33.31 | p3nguin | There's no asterisk 1.10 anyway. |
| 15:33.41 | p3nguin | core show version <------ |
| 15:34.32 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 15:34.35 | Kobaz | verywiseman: module load chan_sip.so |
| 15:34.50 | cusco | so... can a queue caller press 8 to go to extension 8 in dialplan (that would make him leav queue) |
| 15:34.53 | cusco | ? |
| 15:34.57 | verywiseman | Kobaz, p3nguin , sorry 10.3.1 |
| 15:35.08 | p3nguin | cusco: If you configure the queue and the dial plan as such, yes. |
| 15:35.24 | cusco | p3nguin: what options in the queue to allow that? |
| 15:35.32 | [TK]D-Fender | cusco, context= |
| 15:35.32 | [TK]D-Fender | ^ |
| 15:35.49 | p3nguin | cusco: Set the context in the queue. |
| 15:36.00 | cusco | well ok I actually noticed that setting or unsetting the context would do that.. i was hoping for a different answer I guess |
| 15:36.17 | cusco | I would like for a member to be able to go to extension 8, ONLY after 1 minute in queue |
| 15:36.37 | p3nguin | Change them to a different queue after a minute. |
| 15:36.48 | cusco | to a different queue.. that is the problem |
| 15:36.55 | cusco | can't I have them on the same queue? |
| 15:37.08 | Kobaz | this is why i wrote my own app_queue |
| 15:37.19 | cusco | is it? |
| 15:37.19 | p3nguin | If your queue is called "queue-1", make the one with the ability to exit called something like "queue-1-out". |
| 15:37.38 | cusco | hm.. |
| 15:37.39 | Kobaz | p3nguin: the problem then is line hopping |
| 15:37.59 | cusco | well I could set queue prio or queue weight |
| 15:38.06 | cusco | but it is still a problem |
| 15:38.16 | p3nguin | How is that a problem? If the person sits in the main q for a minute, they are then shifted to the other q in the same order. |
| 15:38.28 | Kobaz | yeah |
| 15:38.32 | Kobaz | but |
| 15:38.37 | p3nguin | Or... |
| 15:38.44 | Kobaz | if you're ringing the same group of agents |
| 15:39.20 | p3nguin | Run the same q time after time, separated by a GoSub() which has the exit extension in it. |
| 15:39.30 | Kobaz | there's no guarantee that callers in queue 2 will get answered first |
| 15:40.15 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-pojjvrbmqmzpbhmk) |
| 15:40.57 | keiths_ | Im with Kobaz, it can be done yes...but i does not seem logical... |
| 15:40.57 | p3nguin | They shouldn't be answered first. |
| 15:42.01 | cusco | well second queue should be answer first |
| 15:42.19 | p3nguin | Maybe I missed something. |
| 15:42.29 | p3nguin | I thought calls should be answered in the order they are received. |
| 15:42.33 | Kobaz | yeah |
| 15:42.36 | Kobaz | within the same qu9eue |
| 15:42.46 | Kobaz | yeah you can put in a weight and whatnot but it's still not guaranteed |
| 15:42.47 | *** join/#asterisk asr33 (~asr33@dsl-173-206-171-179.tor.primus.ca) |
| 15:43.23 | cusco | yes, its not |
| 15:43.31 | cusco | i was facing same drama |
| 15:43.54 | p3nguin | I totally do not understand why the second queue should get answered first, so I'm out. |
| 15:44.08 | keiths_ | might as well make a IVR trivia game out of it. most correct answers, gets top priority in the queue |
| 15:44.24 | cusco | because once a member has been shifted to the second queue, he is in total wating for a longer period? |
| 15:44.43 | cusco | keiths_: heh, that sounds fair but is not legal |
| 15:47.52 | shamelessn00b | [TK]D-Fendershamelessn00b, As for getting the streams, just Originate another channel to chanspy into the caller's channel for your hook. ??? I need to be able to stream it to my recognition engine, right now I'm using EAGI to do that, hence why I need the application to be non-blocking so I can call EAGI after the channels get bridged |
| 15:48.07 | [TK]D-Fender | cusco, It's probably be easier to pick apart their place in line and on "8" check if they waited 1 minute, otheriwse put them back into the same position (which is possible) |
| 15:48.46 | [TK]D-Fender | shamelessn00b, and I just gave you a way to do that. |
| 15:49.39 | *** join/#asterisk timahvo1 (~rogue@41.81.229.100) |
| 15:50.06 | blitzrage | shamelessn00b: can't you trigger it via M() or U() ? |
| 15:51.19 | shamelessn00b | I wonder how asterisk would cope, with so many conferences and channels, for a 3 user conference, I'll have to set up 3 conferences and join them and 3 channels spying on each user. |
| 15:51.38 | blitzrage | with ConfBridge() it should be pretty much a non-issue |
| 15:51.53 | blitzrage | you'll just have to test and scale as appropriate based on testing |
| 15:53.02 | [TK]D-Fender | blitzrage, he wants the caller to land in the confbridge and have anotehr EAGI still have access to the stream "bridged in". |
| 15:53.08 | cusco | [TK]D-Fender: is that a parameter of Queue() ? |
| 15:53.34 | shamelessn00b | [TK]D-Fender: thanks, I understand what you mean now :) |
| 15:53.37 | [TK]D-Fender | cusco, OrderlyQ does this... go look... |
| 15:54.01 | cusco | ow.. |
| 16:01.22 | *** join/#asterisk timahvo1 (~rogue@41.81.229.100) |
| 16:02.20 | *** join/#asterisk timahvo1 (~rogue@41.81.229.100) |
| 16:05.59 | *** join/#asterisk jkroon (~jkroon@dsl-244-28-232.telkomadsl.co.za) |
| 16:07.29 | Greenlight | Heya all. I've recently moved about 30 or so of my extensions over from SIP to IAX (some trunked), and noticed that my CPU load has increased dramatically (seeing around 50% total usage now, when it was around 10% before). Is IAX really resource hungry compared to SIP ? |
| 16:08.23 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-cmwusitzrpmizexu) |
| 16:10.28 | verywiseman | how can i show sip devices that connect to asterisk 1.8? |
| 16:10.38 | Greenlight | sip show peers |
| 16:10.45 | Greenlight | At the CLI |
| 16:12.32 | verywiseman | Greenlight, "No such command 'sip show peers' (type 'core show help sip show' for other possible commands)" |
| 16:13.38 | Greenlight | module show like sip |
| 16:13.42 | Greenlight | (CHeck you have sip loaded) |
| 16:15.59 | [TK]D-Fender | verywiseman, if you don't have the command... then chan_sip isn't even laoded |
| 16:16.13 | *** join/#asterisk raubvogel (~raubvogel@shop.monetra.com) |
| 16:17.00 | Greenlight | Anyone else getting high resouce usage with IAX compared to SIP. I'm trying to tell if this is the cause, or of its because I've also added a few more users too |
| 16:18.13 | *** join/#asterisk timahvo1 (~rogue@197.179.27.16) |
| 16:18.18 | WIMPy | Greenlight: Or a different configuration? |
| 16:19.42 | Greenlight | I've not changed much apart from IAX instead of SIP for a remote office (so that I can trunk it and squeeze more calls in over limited bandwidth) |
| 16:20.05 | Greenlight | I'm using a number of local channels too - around 450 at peak load |
| 16:21.33 | *** join/#asterisk rossand (~aross@CPE009400809a9c-CM78cd8ed45eb5.cpe.net.cable.rogers.com) |
| 16:21.57 | WIMPy | Are you sip and IAX configurations different? |
| 16:22.06 | WIMPy | codecs? qualify? jitterbuffer? |
| 16:22.57 | Greenlight | Both were G729, and IAX has jitterbuffer off (seems to cause issues when I turn it on) |
| 16:23.10 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 16:23.14 | Greenlight | It's an IAX trunk instead of individual SIP registrations |
| 16:25.25 | voxter | Strange, I have a queue that has timeout=0 yet agents are still getting auto-paused. any idea why that might happen? |
| 16:25.53 | [TK]D-Fender | voxter, Why do yo have no timeout? |
| 16:26.01 | [TK]D-Fender | voxter, that is the time to dial agents for.... |
| 16:28.06 | voxter | [TK]D-Fender: the agent timeout is actually used to determine how long before the system determines an agent has become unavailable and thus pauses them. I have set the max wait time on the queue() itself |
| 16:28.25 | voxter | although, maybe you have a point here, i guess it will only scan for available agents once in this case. |
| 16:28.38 | Greenlight | Could there be issues with G729 translation? I see that "core show translation" doesn't work in 10 ? |
| 16:29.28 | voxter | [TK]D-Fender, All I'm looking to do is have it so that * never decides to pause/remove an agent simply because they didnt answer a call after <x> timeframe |
| 16:29.36 | mjordan | Greenlight: what do you mean core show translation doesn't work in 10? |
| 16:29.44 | coppice | G.729 translation? Like G.七二九? |
| 16:30.32 | Greenlight | mjordan: Like it says 15000 for alaw -> ulaw and also 15000 for alaw -> g729 which is clearly silly |
| 16:31.12 | mjordan | Greenlight: so, first of all, that does not mean that the command itself isn't work. Its a reporting mechanism, and its reporting what it believes to be the translation. |
| 16:31.21 | [TK]D-Fender | voxter, timeout is how long between ringing agents. autopause is an option based on the fact of them not answering. |
| 16:31.43 | Greenlight | mjordan: Ok, so the command itself works okay, but reports junk |
| 16:31.44 | mjordan | Greenlight: second, I'd have to look at a DEBUG log to find out what the translation paths were reported as when the table was built. On my machine, I have ulaw->alaw at 9150, and vice versa. |
| 16:32.12 | Greenlight | What do you have for a codec that would take more CPU time like say GSM/g729 |
| 16:32.15 | mjordan | Greenlight: it isn't junk if that is what the time (in microseconds) is for that translation path on your machine, which we haven't verified. |
| 16:32.31 | Greenlight | All my 10 servers have the same 'feature' |
| 16:32.40 | Greenlight | 1.8 used to report correct values |
| 16:33.20 | mjordan | Greenlight: it depends on the translation path. And this is what I'm attempting to explain - you don't know that the values are incorrect. The media format architecture was completely redone in Asterisk 10. The values may be correct. |
| 16:33.57 | mjordan | The question is, why is the translation time increased from Asterisk 1.8 to Asterisk 10, and without looking at a specific example, I can't tell you that :-) |
| 16:34.03 | Greenlight | You may be right, I just presumed it was a bug, since didnt think that the overhead for alaw -> ulaw could be same as alaw -> g729 |
| 16:35.16 | *** join/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 16:35.20 | Greenlight | I did ask someone on here when 10 first came out, and they agreed it was a bug |
| 16:36.05 | mjordan | Greenlight: what is the translation path from ulaw to alaw? |
| 16:36.11 | mjordan | core show translation paths ulaw |
| 16:37.18 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 16:37.41 | Greenlight | Actualyl looking at it, alaw-> ulaw is a little less at 9150 |
| 16:37.50 | Greenlight | It's most the rest that are 15000 |
| 16:38.24 | mjordan | okay, that's similar to what I have. If you look at the paths, many of the translations first require the codecs to go through slin. |
| 16:38.34 | Greenlight | Alaw -> ulaw and back used to be a tiny fraction of the cost alaw -> g729 with 1.8 |
| 16:38.52 | Greenlight | Ahh ok |
| 16:39.03 | Greenlight | So with 1.8 it was direct, now its alaw -> slin -> ulaw |
| 16:39.19 | Greenlight | So itsnot that G729 is less or quicker, just the other translaations are a little slower? |
| 16:39.25 | mjordan | Greenlight: not necessarily, the paths would show you that. |
| 16:39.34 | mjordan | ulaw -> alaw is direct on my instance. |
| 16:39.43 | mjordan | ulaw -> gsm is ulaw -> slin -> gsm. |
| 16:39.56 | Greenlight | How can you see that? |
| 16:40.03 | mjordan | (11:36:11 AM) mjordan: core show translation paths ulaw |
| 16:40.10 | Greenlight | Ah yea |
| 16:40.33 | *** join/#asterisk beetlej00z (~IceChat9@c-76-127-107-179.hsd1.ca.comcast.net) |
| 16:40.37 | Greenlight | So yea, ulaw to alaw is direct here too |
| 16:40.43 | *** join/#asterisk timahvo1 (~rogue@197.179.27.16) |
| 16:41.00 | Greenlight | But alaw -> slin is 6000. Same as g729 -> slin. That can't be correct, can it ? |
| 16:41.06 | Greenlight | What do you have on your side? |
| 16:41.44 | mjordan | It is, but that isn't surprising. Under the hood, slin is used for a lot of things, so I'd expect translations to/from it to be quick. |
| 16:42.15 | mjordan | Greenlight: keep in mind that the media format rearchitecture greatly expanded the types of codecs that Asterisk can handle, including native support (and transcoding) to wide band audio codecs. There's no such things as a free lunch, however, which means transcoding times have - if nothing else - changed between 1.8 and 10. |
| 16:42.35 | mjordan | That isn't to say they're better or worse, but the paths things take is certainly different. |
| 16:43.16 | Greenlight | Yea, I understand that, just always expected a higher overhead if using g729 |
| 16:43.38 | coppice | Greenlight: do you have a transcoding card? |
| 16:43.44 | Greenlight | Nope |
| 16:44.22 | Greenlight | Transcoding cards will no longer have a place if the new translation results are correct |
| 16:44.23 | coppice | then if alaw -> slin is any more than 1/100th of g729 -> slin something is broken |
| 16:44.25 | *** join/#asterisk irule (~irule@187.139.6.189) |
| 16:44.35 | Greenlight | Thats what I thought |
| 16:44.43 | Greenlight | But mjordan is disagreeing :( |
| 16:45.06 | coppice | well, not everyone knows what they are talking about? :-) |
| 16:45.33 | beetlej00z | hola |
| 16:45.40 | *** join/#asterisk Carlos_PHX1_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 16:45.43 | mjordan | Greenlight: no, I'm not. What I'm disagreeing with is the statement that "core show translations" is broken. Without seeing the translation paths, how those paths are built, and what the logs are showing, I can't say what is wrong. But something that is dumping info to the CLI is probably not the culprit. |
| 16:46.23 | Greenlight | Yea, maybe we got crossed wires. I was just saying that the results displayed by that command were incorrect |
| 16:46.36 | Greenlight | As g729 should be waaaay more CPU intensive than other codecs |
| 16:47.16 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-ykqszknogfvhvmba) |
| 16:47.18 | Greenlight | At present it's very difficult to see which g729 codec binary is best to use for the current processor |
| 16:47.28 | Greenlight | Or what the increase in CPU usage is |
| 16:47.57 | Greenlight | Anyways - I'm due on site about 10 minutes ago, so gotta go. Thanks for your help, take care! |
| 16:48.02 | coppice | it shouldn't be more intensive that all the other codecs. G.729 would probably be, but what you have is G.729A. alaw->slin is trial, though |
| 16:48.32 | coppice | s/trial/trivial |
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| 16:54.30 | *** join/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
| 16:55.20 | nny | quick Q, what's the method by which asterisk checks the /var/lib/sounds/ dir for files? Can I make a custom folder there (or is there a way to have asterisk check a custom folder?). This would be in addition to the normal folder |
| 16:56.29 | Kobaz | df |
| 16:56.47 | Kobaz | Playback(custom/foo/bar) |
| 16:57.04 | nny | perfect thanks |
| 17:00.19 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 17:01.26 | *** join/#asterisk danfromuk (~IceChat77@2.26.253.199) |
| 17:01.29 | bipul | hi |
| 17:01.36 | bipul | Any one around |
| 17:02.15 | bipul | Well i have something in my brain which i would like to know from you ? |
| 17:02.42 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-yndkebgqklvywead) |
| 17:03.08 | beetlej00z | I can't get my polycom soundpoint 310 to resister to my asterisk 1.4 server. I know I've got all the correct network settings because I can register my softphone no problem. Any polycom peeps on that can give me some pointers? |
| 17:03.11 | bipul | Is it Possible to trace out the conversation of two mobile phone with asterisk server |
| 17:04.35 | WIMPy | bipul: The answer is probably no, but what do you mean by "trace out"? |
| 17:04.51 | blitzrage | beetlej00z: show the configuration so people can help |
| 17:05.06 | bipul | WIMPy, To hear there conversation |
| 17:05.22 | bipul | What they are talking ! |
| 17:05.23 | blitzrage | if the mobile phones place their calls through Asterisk... sure |
| 17:05.40 | WIMPy | bipul: what blitzrage said. |
| 17:05.40 | blitzrage | but that is not likely going to happen |
| 17:05.50 | blitzrage | if you're trying to do a man-in-the-middle-attack then basically, no |
| 17:06.11 | beetlej00z | blitzrage: the phone config or the .conf settings? |
| 17:06.14 | bipul | blitzrage, but is it there any such option to trace some one mobile with my asterisk server |
| 17:06.29 | blitzrage | beetlej00z: likely both |
| 17:06.34 | blitzrage | bipul: asked and answered |
| 17:07.16 | bipul | WIMPy, He said it's not possible unless Mobile Operator is using Asterisk |
| 17:07.42 | WIMPy | bipul: It has been dome with osmocom, but it depends on the networs security settings. |
| 17:08.01 | bipul | osmocom ? |
| 17:08.01 | WIMPy | Asterisk can only record calls that are routed through it. |
| 17:08.42 | bipul | How about GSM mobiles WIMPy |
| 17:09.30 | WIMPy | http://osmocom.org/ |
| 17:11.13 | bipul | WIMPy, can you Guide me :) i have 0 skills about osmocom ? |
| 17:12.25 | WIMPy | Nope. What you want to do requires a lot of reading and a little bit of coding. |
| 17:13.12 | cusco | lol |
| 17:16.16 | *** join/#asterisk sunfone (~jeff@c-98-220-218-74.hsd1.in.comcast.net) |
| 17:16.58 | nny | what Menu Option is the application MYSQL listed under in meake menuconfig? |
| 17:17.01 | nny | make* |
| 17:17.20 | Qwell | nny: Addons, but you really really really really should not use that. |
| 17:17.24 | Qwell | Use func_odbc instead. |
| 17:17.30 | nny | Qwell: it |
| 17:17.34 | nny | Qwell: it's for one query |
| 17:17.42 | Qwell | So use func_odbc |
| 17:18.21 | nny | w/e |
| 17:19.10 | nny | curious as to why it's deprecated |
| 17:19.17 | Qwell | Because it's garbage. |
| 17:19.27 | nny | so.. no fixing it eh? |
| 17:19.38 | nny | i mean it's garbage doesn't sound very technical |
| 17:19.42 | Qwell | There's no point. There are far better solutions. |
| 17:19.47 | nny | odbc seems convoluted |
| 17:20.34 | bobb_WU | hello to the room |
| 17:23.23 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 17:23.24 | nny | can't seem to find any discussions relating to MYSQL's removal |
| 17:23.38 | bipul | is there any channel for osmocom ? |
| 17:23.55 | *** join/#asterisk din3sh (din3sh@41.136.241.105) |
| 17:24.04 | [TK]D-Fender | bipul, Go check their page for what support options they have |
| 17:24.14 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 17:25.07 | nny | buwhahaha |
| 17:25.17 | nny | oh nm |
| 17:25.31 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:25.35 | nny | dammit, you'd think application removal would have more background to it than "cause we said so" |
| 17:27.07 | Nugget | cheers mysql removal :) |
| 17:27.25 | din3sh | mysql removal from where? |
| 17:27.30 | Nugget | good software development requires a ruthless willingness to deprecate and excise code that doesn't make sensee |
| 17:27.41 | nny | why doesn't MYSQL make sense? |
| 17:27.54 | Nugget | the native db support in asterisk doesn't make sense alongside odbc support. |
| 17:28.06 | nny | it doesn't? |
| 17:28.10 | Nugget | no |
| 17:28.14 | nny | make sense to who? |
| 17:28.16 | asr33 | gets cold sweats thinking about deprecation |
| 17:28.29 | Nugget | to anyone who has to maintain asterisk codebase or cares about asterisk long-term stability |
| 17:28.39 | [TK]D-Fender | MySQL isn't as bad as say .... telnet ... |
| 17:28.45 | nny | i really really wanna spend an hour setting up an odbc connection every time i write a dialplan to call one query froma local database |
| 17:28.47 | [TK]D-Fender | :/ |
| 17:29.17 | nny | it makes sense in that there's less parts involved, so to speak, but meh, arguing semantics, i'll use it for now |
| 17:29.37 | [TK]D-Fender | nny, You could always shell-out, AGI, etc |
| 17:29.40 | bluregard | why does it take an hour to set up an odbc connection I wonder? |
| 17:29.56 | nny | bluregard: it was hyperbiole, i figured one person would question the time frame |
| 17:29.58 | nny | derp! |
| 17:30.03 | nny | [TK]D-Fender: aye |
| 17:30.55 | bluregard | I just don't see what the big deal is using odbc... I've used it in several dialplans and its not that complicated |
| 17:31.23 | nny | bluregard: it's an extra step to connect to a database that a lot of applications and languages connect to natively |
| 17:31.49 | bluregard | fair enough |
| 17:32.14 | nny | back to banging keys, i'll be a rebel and use it for now |
| 17:33.36 | bluregard | are the Park() and ParkAndAnnounce() apps for building a parking feature in the dialplan rather than using features.conf? |
| 17:33.56 | bluregard | or can they be used that way? |
| 17:34.31 | _Corey_ | bluregard: They're used in conjunction to whatever you configure in features.conf |
| 17:36.44 | bluregard | somewhere in the docs I read you only need to include the parking lot context and not the actual parking extension. |
| 17:37.38 | bluregard | which works fine, but I want to be able to access the various channel vars and invoke other apps when a call is parked. |
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| 17:39.08 | _Corey_ | bluregard: You can use your own parking extension and just call the Park() app |
| 17:39.19 | _Corey_ | (after whatever else you wish to do) |
| 17:39.52 | bluregard | _Corey_, along side having parking configured in features.conf? |
| 17:40.24 | _Corey_ | sure, just change the default "parkext" to something else |
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| 17:59.31 | nny | if i am using ChannelRedirect for something but want to retain a channel variable from the pre-redirection (the extension that originally dialed the application) what's the best way to do that? |
| 18:00.23 | nny | if I set "somecar" to "CALLERID(num)" before i redirect, will variable persist the change or is it wiped when the channel changes? |
| 18:00.39 | blitzrage | channel variables are associated with the channel |
| 18:00.44 | nny | of course |
| 18:00.53 | [TK]D-Fender | nny, The channel is just being pointed to other dialplan, it shouldn't change in any other way I would think... |
| 18:00.56 | blitzrage | if the channel changes, and you need values to remain, you need to make the variable transitive |
| 18:01.04 | blitzrage | Set(_myVariable=value) |
| 18:01.11 | nny | yeah that's what I was asking, thanks! |
| 18:01.16 | blitzrage | single underscore == transitive to next channel |
| 18:01.25 | blitzrage | double underscore == infinitely transitive |
| 18:01.34 | nny | nice |
| 18:01.40 | nny | that's good to know |
| 18:01.53 | p3nguin | Good old variable inheritance. |
| 18:02.11 | bipul | p3nguin, Hi |
| 18:02.23 | bipul | How are you Robert |
| 18:02.32 | nny | blitzrage: by next channel is there a chance someone else could dial during that process and have thatc hannel inherit vs the one created by ChannelRedirect? |
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| 18:02.39 | p3nguin | Fair to middling, I reckon. |
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| 18:03.46 | p3nguin | One underscore allows inheritance to one new level, not just one single creation. |
| 18:04.16 | nny | sorry don't follow the "new level" term. |
| 18:04.24 | nny | can you elaborate? |
| 18:04.29 | p3nguin | Think of it like a tree. |
| 18:04.40 | p3nguin | The channel in question is the trunk. |
| 18:04.48 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 18:04.53 | blitzrage | nny: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#AdditionalConfig_id243954 |
| 18:04.54 | p3nguin | A new branch comes off that trunk. That's a new channel. |
| 18:05.08 | p3nguin | One underscore allows that new channel to inherit the variable. |
| 18:05.24 | p3nguin | Now when that new channel branches to more new channels, they don't get the variable. |
| 18:05.39 | p3nguin | Unless you have used two underscores originally, then they do. |
| 18:06.34 | nny | thanks |
| 18:06.39 | p3nguin | But another branch can still come from the trunk, and one underscore would allow that new channel (which stems directly from the original) to inherit the variable. |
| 18:06.51 | nny | gotcha, that visual makes sense |
| 18:09.10 | p3nguin | Now if you have a limitation where the "trunk" channel can't have more than one branch from it at any given time, then you'd have to wait for the previous branch to die, wither, and fall off. :) |
| 18:09.39 | p3nguin | Hopefully it doesn't land on your roof. |
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| 18:39.16 | bluregard | when using ParkAndAnnounce() and announcing to a Local channel in order to make use of ${PARKEDAT}, is it possible to make a channel var from the channel that called ParkAndAnnounce available to the local channel? |
| 18:41.17 | bluregard | I need access to both ${CHANNELL(peerip) for the device that parked the call and ${PARKEDAT} |
| 18:45.26 | bluregard | nevermind |
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| 18:50.27 | irule | hi Im getting permission denied on port 5038 from remote host, local is ok, and freepbx thinks asterisk is not even running, after installing freepoh for debian, how may I fis this? |
| 18:50.31 | *** join/#asterisk Wazzmann02 (~chatzilla@50-78-28-109-static.hfc.comcastbusiness.net) |
| 18:50.50 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-123-Miami.FL.hfc.comcastbusiness.net) |
| 18:51.30 | Wazzmann02 | Good Afternoon everybody! Is anyone handy that wouldn't mind answering a question or two? I should be brief. |
| 18:51.46 | WIMPy | ~ask |
| 18:51.46 | 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. |
| 18:52.11 | Wazzmann02 | Thank you for that! |
| 18:52.48 | Wazzmann02 | I would like to connect FreePBX to an Executone legacy PBX to transfer calls without the call leaving over the PRI (attached only to the Executone). |
| 18:53.10 | Wazzmann02 | CentOS5.5, Asterisk 1.8, FreePBX 2.9 |
| 18:54.21 | Wazzmann02 | Is that possible? Does it require a hardware gateway of some kind? |
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| 18:57.09 | *** part/#asterisk twanny796 (~twanny@85.232.219.136) |
| 18:57.09 | Wazzmann02 | I already have two asterisk boxes able to handoff calls between them using mutual IAX2 trunks, and I'm looking for a way to interface one of the asterisk boxes with the legacy pbx for similar behavior |
| 18:57.25 | pabelanger | get a TDM card |
| 18:58.53 | Wazzmann02 | can the TDM card be external, USB? The asterisk server in question runs in a VM. |
| 19:01.04 | blitzrage | I have a question about Aastra device provisioning for anyone who knows more than me. I know with the Polycoms you can define a registration appearance and tell it the number of lineKeys for it to show up on. Does the Aastra (57i in this case) also have the same time of thing, or is it always 1 registration per line key? |
| 19:01.13 | *** join/#asterisk SaRSAeOL (~sarsaeol@66-113-93-170.rev.ibsinc.com) |
| 19:01.20 | blitzrage | Wazzmann02: if you use something like a xorcom device sure |
| 19:03.55 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 19:04.11 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-123-Miami.FL.hfc.comcastbusiness.net) |
| 19:04.42 | p3nguin | irule: lsof -i :5038 |
| 19:06.33 | generalhan | blitzrage: not sure exactly if this is what youre talking about, but this is how i setup my 480i config files to get multiple apperences on the phone: http://pastebin.com/BDd9CyAC |
| 19:06.54 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 19:07.01 | blitzrage | generalhan: oh ok -- and each of those don't register separately right? |
| 19:07.17 | generalhan | you mean the 2 lines with the same ? |
| 19:07.21 | blitzrage | in re: to line1 and line2 |
| 19:07.23 | blitzrage | ya |
| 19:07.33 | blitzrage | like they don't step on each other? |
| 19:07.34 | generalhan | i only see the registration hit the server once for each different line |
| 19:07.50 | blitzrage | right, but in the case of your config line1 and line2 would step on each other |
| 19:07.57 | blitzrage | if they each register individually |
| 19:07.59 | generalhan | so when the phone comes on i see 1 registration for 7001 7100 and 7101 |
| 19:08.23 | blitzrage | and if 7001 line1 is busy, it rolls to line2? |
| 19:08.33 | *** join/#asterisk Deeewayne (~dwayne@c-76-29-230-45.hsd1.al.comcast.net) |
| 19:08.33 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:08.39 | voxter | Hey, if asterisk manager fails to load, is it possible to attempt an unload/load of the AMI module? I cant find what module it is a part of however. |
| 19:08.41 | generalhan | for the phones that i dont have a call limit set for, yes |
| 19:09.04 | blitzrage | voxter: manager.c is core component I think |
| 19:09.18 | blitzrage | voxter: manager reload is about the best you can hope for |
| 19:09.21 | voxter | blitzrage: rats. so if it didnt bind at startup, i have to restart the process? |
| 19:09.25 | voxter | manager command unknown. |
| 19:09.27 | voxter | :( |
| 19:09.34 | blitzrage | what version? |
| 19:09.46 | voxter | i lied. that worked. |
| 19:09.50 | blitzrage | heh |
| 19:09.52 | voxter | i must have typoed the first time. |
| 19:09.52 | voxter | :D |
| 19:12.06 | Wazzmann02 | Does the Xorcom external TDM device allow multiple calls to flow through a single port? What is the physical connection to the other PBX? Does it require a PRI-->In interface? |
| 19:12.48 | tzafrir_laptop | Wazzmann02, it's like any TDM device |
| 19:13.11 | Wazzmann02 | Sorry, I'm a PSTN noob |
| 19:13.31 | tzafrir_laptop | Connected through USB instead of PCI, but basically the same, apart from that |
| 19:13.44 | Wazzmann02 | I'm really a server admin breaking into the linux/pbx world. The only reason I know what a PRI is, is from seeing the phonebills |
| 19:14.10 | tzafrir_laptop | You have a PRI connection to the PSTN or an analog one? |
| 19:14.19 | Wazzmann02 | PRI on legacy pbx |
| 19:15.03 | blitzrage | Wazzmann02: assuming North America, a PRI can contain up to 23 voice channels |
| 19:15.08 | blitzrage | per interface |
| 19:15.18 | blitzrage | In Europe its 30 |
| 19:15.27 | Wazzmann02 | that's how I understand it. We have up to 23 lines into our PBX, sorry yes I'm in North America |
| 19:15.40 | blitzrage | so 1 port == 23 simultaneous calls |
| 19:15.57 | Wazzmann02 | but as for the TDM interface between the legacy pbx and asterisk |
| 19:16.04 | blitzrage | to answer your original question, yes, the Xorcom handles all the channels per port |
| 19:16.12 | p3nguin | I'm pretty ignorant to telco interfaces, too... PRI is a digital circuit? |
| 19:16.17 | blitzrage | legacy PBX --> xorcom --> asterisk |
| 19:16.19 | [TK]D-Fender | ~pri |
| 19:16.19 | infobot | it has been said that pri is [~pri] Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, R1T1,R2T1,R4T1, etc. |
| 19:16.57 | blitzrage | legacy pbx <-- T1 cable --> xorcom <-- usb --> asterisk |
| 19:17.59 | generalhan | i have a what-would-you-do question ... i am getting an MPLS circuit put in for my main and remote office, and i am trying to decide whether to put an asterisk box at each location (hooked to a PRI) and connect them via IAX, or if i should set the phones at the remote office to connect to my asterisk box here over the 4.5M MPLS. |
| 19:18.06 | Wazzmann02 | just to confirm the whole thing: PSTN provider <--> pri over T1 <--> legacy pbx <--> RJ45 cable <--> xorcom <--> ..... |
| 19:18.40 | tzafrir_laptop | ..... Asterisk (on the computer) |
| 19:18.48 | blitzrage | Wazzmann02: no, T1 from legacypbx --> xorcom |
| 19:19.03 | generalhan | If i use something like a Xorcom unit at the main location, then for less money i can have a reduntant asterisk solution, but im concered about call quality... |
| 19:19.10 | Wazzmann02 | great so the only hardware to purchase is the xorcom, and use a plain cat6 with rj45s between the legacy and the xorcom |
| 19:19.27 | blitzrage | Wazzmann02: pri from provider via T1 --> legacy pbx --> T1 --> xorcom --> usb --> asterisk |
| 19:19.29 | *** join/#asterisk aberrios_ (aberrios@cpc7-bolt8-0-0-cust258.10-3.cable.virginmedia.com) |
| 19:20.08 | tzafrir_laptop | Wazzmann02, "rj45" can be for T1 or for Ethernet |
| 19:20.18 | *** join/#asterisk eZz (~ez@195.114.6.134) |
| 19:20.25 | Wazzmann02 | the second T1 is where I'm getting confused, is that just a plain cable? or do i need some special cards, or worse yet to connect it back to the carrier's T1 |
| 19:20.26 | tzafrir_laptop | anyway, what do you need the legacy PBX for? |
| 19:20.27 | eZz | hello |
| 19:20.52 | pabelanger | tzafrir_laptop: keep existing phones? |
| 19:20.59 | Wazzmann02 | the legacy pbx is the existing infrastructure at the main office. full office full of digital handsets, voicemail, etc |
| 19:21.13 | eZz | why asterisk is trying to make a re-invite if directmedia is turned off globaly ? |
| 19:21.28 | Wazzmann02 | the asterisk box is to adapt the calls and send it over the WAN to our satelite office, which is entirely asterisk-powered |
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| 19:22.20 | tzafrir_laptop | If you have two T1 ports on the Xorcom device, you can make the Asterisk box a gateway |
| 19:22.24 | Wazzmann02 | satellite office is long distance call for main office, so least cost routing over wan would be a big plus, in addition to advanced ACD features of the asterisk system |
| 19:22.59 | tzafrir_laptop | (or if the legacy pbx can talk voip, one port may do the trick as well) |
| 19:23.27 | Wazzmann02 | Executone legacy box is from the 80s |
| 19:23.48 | p3nguin | When someone says they have a digital T1, does that imply PRI? |
| 19:24.12 | Wazzmann02 | I'd also like to say you guys are great. so much valuable know-how, and willing to share it with noobs daily. |
| 19:24.22 | [TK]D-Fender | ~redundant |
| 19:24.22 | infobot | i heard redundant is found at the Department of Redundancy Department |
| 19:24.43 | Wazzmann02 | ~joke |
| 19:24.43 | infobot | A bhuddist monk goes to a hot dog vendor and says "Make me one with everything!" |
| 19:24.59 | *** join/#asterisk wonderworld (~ww@dsdf-4db52e0c.pool.mediaWays.net) |
| 19:29.28 | Wazzmann02 | If I want to keep from disturbing the legacy pbx, can I make it the gateway for the asterisk box? (assuming it has a 2nd PRI port) |
| 19:29.38 | Wazzmann02 | err 2nd T1 port |
| 19:30.17 | *** join/#asterisk timahvo1 (~rogue@41.81.2.227) |
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| 19:31.30 | generalhan | anyone have any wisdom to pass down, in regards to my main office <-> remote office setup question ? |
| 19:31.32 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 19:32.06 | WIMPy | Wazzmann02: You can do it either way. |
| 19:32.14 | blitzrage | generalhan: it depends on if you expect the phones at one of the offices to continue working when the link breaks |
| 19:32.26 | blitzrage | not if.... when. |
| 19:33.04 | blitzrage | if you expect them to continue working, then I suggest you place an asterisk system at each location and let them live independently but set it up to allow devices to call between boxes over the MPLS link |
| 19:33.25 | blitzrage | if they are in different locations physically (different rate centres) you can also allow calling out the remote box for cheaper rates |
| 19:33.51 | generalhan | that last part is not a concern. the offices are 1 mile apart from each other |
| 19:33.55 | blitzrage | i.e. Toronto and Philadelphia rate centres... allow Toronto to traverse the MPLS network and dial out the Philly PRI |
| 19:34.11 | Wazzmann02 | I'm a complete pbx noob and I was able to setup the 2nd server with IAX2 trunks between the two with about 4-6 hours of googling and testing |
| 19:34.41 | Wazzmann02 | I could send some pictures of configs of the IAX2 trunks if it would help (using FreePBX gui only) |
| 19:34.46 | generalhan | though i have never had an asterisk box break on me before, the same concern has to be weighed on when/if the box itself breaks |
| 19:35.18 | Wazzmann02 | It looked to me like DUNDi was designed to resolve server and WANlink redundancy |
| 19:35.39 | generalhan | where if i have a redundant setup at one location, i am covered if that happens ... so i guess i need to weigh, which is more likely to happen |
| 19:35.54 | Wazzmann02 | 4 servers! |
| 19:35.59 | Wazzmann02 | linux is free :) |
| 19:36.04 | generalhan | hardware is not |
| 19:36.10 | Wazzmann02 | VMs though |
| 19:36.34 | Wazzmann02 | ohh i see what you mean, the extra TDM cards |
| 19:36.57 | blitzrage | pretty much... redundancy costs money, so you have ot figure out your level of comfort |
| 19:37.17 | blitzrage | should technically have a hot standby at each location and then multiple locations |
| 19:39.16 | generalhan | the situation is this: i am pulling all servers out of the remote office when the MPLS gets put in, to centralize everything. And i am replacing the existing Avaya system, so that means 2 servers 2 Xorcom units, and 60 phones. so if i was to do a redundant setup in both offices i would need to put 2 servers and a Xorcom in each location (after i just removed all servers from the remote site). |
| 19:39.16 | generalhan | that could seriously hurt my chances of getting this passed the CFO |
| 19:39.57 | blitzrage | then be sure you document what happens when the MPLS link dies |
| 19:40.02 | blitzrage | and whether that is acceptable |
| 19:41.46 | generalhan | that is why i presented this as a what-would-you-do question ... so let me put it this way... say you could only get aproved for 2 servers, would you put 1 in each location and assume your MPLS dies, or put both servers in one location and assume that your hardware will die ? |
| 19:43.46 | Wazzmann02 | At least if you run the Asterisk in a VM on your host at the central site, you could quickly restore and move the xorcom to a new host |
| 19:43.50 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:43.50 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:44.05 | Wazzmann02 | assuming you have another host at the main site that could take over that light load |
| 19:44.23 | *** join/#asterisk oej (~olle@95.209.113.40.bredband.tre.se) |
| 19:44.33 | Wazzmann02 | that nearly eliminates the downtime from failing hardware at the main site |
| 19:45.57 | Wazzmann02 | that being said I could offer that we've had MPLS from AT&T for a year or two now, and have never had the links go down except for complete power failure at one of the locations |
| 19:46.24 | Wazzmann02 | It has been more reliable than our old T1 WAN links used to be (also from AT&T) |
| 19:46.53 | p3nguin | Ask the provider for their statistics to see how much they go down. |
| 19:47.28 | Wazzmann02 | btw. anyone know how much these usb xorcom boxes cost? |
| 19:48.18 | generalhan | i have used them before, on an MPLS between ARizona and California, had it up for 10 years with only 1 outage, when a stupid construction crew working on the street cut through the pipe. |
| 19:48.57 | generalhan | Wazzmann02: depends on which one you get, which port setup, and if you want the appliance or just the hardware |
| 19:48.59 | p3nguin | I think I would rely more on the MPLS and have the spare server at the central location. |
| 19:50.26 | generalhan | knowing that the remote site will be connecting to the main site for all hosted applications/services, and internet traffic, would you be concerned about bandwidth for 30 SIP users |
| 19:50.57 | eZz | why does asterisk is trying to make a re-invite if directmedia is turned off globaly ? |
| 19:51.11 | Wazzmann02 | for example let's say I want the external, usb connected, with single T1 port to accept the legacy pbx being the gateway. |
| 19:51.36 | p3nguin | Assuming one call per person at all times, you're looking at a continuous 2.7 Mbits of traffic for calls. |
| 19:51.45 | p3nguin | using ulaw |
| 19:52.02 | p3nguin | (unless I did the math wrong) |
| 19:54.55 | Wazzmann02 | p3nguin: wow that's more than I expected |
| 19:55.41 | Wazzmann02 | so .09 mbps per concurrent call on ulaw |
| 19:55.44 | p3nguin | Someone should double check my estimation. |
| 19:57.14 | irule | sip show peers No such command 'sip show peers' (type 'core show help sip' for other possible commands) is this normal? :s |
| 19:57.59 | Wazzmann02 | wow all the xorcom USB boxes I can find seem to have ~24 ports on them. Do they make one that's simple and only requires connectivity to a single T1 interface? |
| 19:58.33 | tzafrir_laptop | Wazzmann02, what do you mean? |
| 19:59.08 | p3nguin | Oh, I wonder if I cut my estimate short. There would be 86 Kbps usage per direction for a single call, I guess. |
| 19:59.11 | Wazzmann02 | I'm looking for a simple Asterisk -- USB -- <appliance> -- T1 -- LegacyPBX unit |
| 19:59.38 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:59.51 | Wazzmann02 | looks like 5.03 mbps for 30 concurrent two way convos |
| 19:59.58 | tzafrir_laptop | you can have up to 4 T1 ports in that unit (and some extra analog ports, but you don't seem to need those) |
| 20:00.13 | generalhan | Wazzmann02: the one that i have used before had 2 PRI/BRI ports and 4 FXS ports. |
| 20:00.14 | p3nguin | If we figure on the more conservative side, you might get away with 157 Kbps per call. |
| 20:00.51 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-123-Miami.FL.hfc.comcastbusiness.net) |
| 20:00.56 | Wazzmann02 | generalhan: how much did that cost you? |
| 20:01.00 | generalhan | p3nguin: is that right ? that seems high ... i havent looked in a long time, but i though that the bandwidth used was close to 64k total |
| 20:01.12 | generalhan | Wazzmann02: i dont remember. |
| 20:01.15 | p3nguin | Figure in overhead. |
| 20:01.27 | [TK]D-Fender | Wazzmann02, between 2 * boxes you'd run IAX2 trunk mode and drop that to around 1.564kbps/direction |
| 20:01.30 | [TK]D-Fender | Wazzmann02, @ G.711 |
| 20:01.35 | generalhan | Wazzmann02: just pull up your favorite VoIP online store and look it up !! |
| 20:01.42 | [TK]D-Fender | (maxxed out) |
| 20:02.10 | Wazzmann02 | yeah I love the iax2 trunks and have that working over my WAN, I was helping with the calcs for generalhan |
| 20:02.27 | irule | my sip module is not loaded! is there a solution to this? |
| 20:03.12 | p3nguin | 4.5 Mbps without trunking for 30 ulaw calls. |
| 20:03.20 | p3nguin | conservatively |
| 20:03.24 | generalhan | [TK]D-Fender: i know that IAX is key for that stuff ... im just trying to figure out what the load would be if i had 30 concurrent calls from my remote site to my asterisk server at my main site |
| 20:03.28 | [TK]D-Fender | irule, Given that you also seem to be failing to load manager.conf I'd say you royally screwed up your entire install. |
| 20:03.41 | [TK]D-Fender | irule, And FreePBX is not supported here, nor is freedoh |
| 20:03.42 | p3nguin | (1503.11) <p3nguin> 4.5 Mbps without trunking for 30 ulaw calls. <---------- |
| 20:04.13 | generalhan | :) |
| 20:04.49 | [TK]D-Fender | generalhan, #calls x CodecBW + 20kbps ~ |
| 20:05.55 | generalhan | [TK]D-Fender: and the ulaw codec is about 64k right ? |
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| 20:06.04 | [TK]D-Fender | generalhan, yes |
| 20:06.33 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 20:06.51 | generalhan | so about 2.5 of my 4.5M pipe just for phones ... that might be too tough to handle. |
| 20:06.58 | p3nguin | If you follow that as a strict measurement, you're at 3.79 Mbps, |
| 20:07.01 | p3nguin | total. |
| 20:08.47 | generalhan | p3nguin: why is that not even close to what im coming up with? (30 x (64 + 20))/(1024) = ~ 2.5 |
| 20:09.14 | [TK]D-Fender | generalhan, Because that is NOT what I said |
| 20:09.23 | [TK]D-Fender | <[TK]D-Fender> generalhan, #calls x CodecBW + 20kbps ~ |
| 20:09.52 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 20:10.06 | [TK]D-Fender | pummels generalhan with a book titled "The Joy Of The Order Of Operations" |
| 20:10.22 | p3nguin | If you have 60 call legs (two legs by 30 calls), times the 64k bandwidth, you're at 3.75 Mbps. |
| 20:10.34 | p3nguin | That doesn't include the 20k overhead. |
| 20:11.37 | p3nguin | Your calc is half, because you measured for only one leg of audio.` |
| 20:11.50 | generalhan | [TK]D-Fender: i figured you left the ( ) out for time sake! lol. |
| 20:12.00 | generalhan | but i didnt realize that was 1 way either! |
| 20:12.01 | *** join/#asterisk timahvo1 (~rogue@41.80.170.234) |
| 20:12.41 | p3nguin | I'm going back to my 4.5 Mbps rating to be safe. |
| 20:12.44 | p3nguin | :/ |
| 20:13.33 | _Corey_ | p3nguin: Unless you're looping the calls, you're doubling your calculation unnecessarily |
| 20:13.33 | generalhan | lol, doesnt really matter ... i was not ok with 2.5M usage, so already at that point, i have to put a server over there and use IAX |
| 20:13.51 | p3nguin | _corey_: Then iftop is lying to me. |
| 20:14.00 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 20:14.17 | SuperNull | Hey guys, is there a way to keep track of call quality in ANY version of asterisk ? |
| 20:14.25 | _Corey_ | p3nguin: If I download something at 80kbps I do not need 1060kbps |
| 20:14.29 | SuperNull | MOS or Jitter/Loss kind of stuff |
| 20:14.36 | _Corey_ | er 160kbps |
| 20:15.20 | p3nguin | It shows, for one single ulaw call, 78.1 Kbps, plus signalling, for a total of 157 Kbps. This is per call. |
| 20:16.05 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-123-Miami.FL.hfc.comcastbusiness.net) |
| 20:16.07 | _Corey_ | I can't speak to the tool in question, but it's wrong. You need a 78.1kbps synchronous connection |
| 20:16.27 | _Corey_ | 78.1k up/ 78.1k down |
| 20:16.29 | leifmadsen | SuperNull: there is some RTCP stuff in many versions of Asterisk |
| 20:16.54 | _Corey_ | p3nguin: so, with a 1.54mbps T1 we can do ~18 calls w/g.711 |
| 20:17.02 | p3nguin | That's what I'm saying 78.1 Kbps per direction, plus signaling, for a total of 157 Kbps. |
| 20:17.16 | _Corey_ | well, circuits aren't measured that way |
| 20:17.43 | [TK]D-Fender | The only time I've seen any aggregate up&down traffic is for BILLING. So not "how big is the pipe", but rather "how much total flowed through it". |
| 20:17.45 | p3nguin | Then I have no idea how you expect to describe the bandwidth used for calls. |
| 20:17.56 | leifmadsen | 80/80 |
| 20:17.59 | Wazzmann02 | ahhh i understand it now |
| 20:18.01 | [TK]D-Fender | Which when you get down to it is never what anyone in here is asking for. |
| 20:18.09 | [TK]D-Fender | So using that math is rather pointless |
| 20:18.13 | p3nguin | OH! I see what your issue is... |
| 20:18.18 | leifmadsen | my internet connection is 1500/512 for example |
| 20:18.32 | leifmadsen | I can handle a different amount of calls in each direction on that circuit |
| 20:18.35 | p3nguin | I see the issue. |
| 20:19.00 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-123-Miami.FL.hfc.comcastbusiness.net) |
| 20:19.01 | p3nguin | I'm describing total requirement, where you're expecting it to be designated per up and down. |
| 20:19.08 | _Corey_ | yeah, you obviously need enough bandwidth to transmit your calls as well as receive them... but we usually just think of one leg for purposes of ordering bandwidth |
| 20:19.09 | leifmadsen | of course... |
| 20:19.28 | p3nguin | I was describing TOTAL, hence my usage of "total" in several places. |
| 20:19.35 | [TK]D-Fender | p3nguin, a Verizon Accounting position jsut opened up for you! |
| 20:19.52 | p3nguin | I'll have to learn their new math. :) |
| 20:20.08 | [TK]D-Fender | p3nguin, As opposed to your new new math :p |
| 20:20.21 | p3nguin | Nah, I was specific in what I was measuring. |
| 20:20.23 | [TK]D-Fender | 3NEW! |
| 20:20.33 | _Corey_ | you think that's bad... i spent a half hour on the phone the other day with Amazon over my EC2 bandwidth bill... ;) |
| 20:20.45 | [TK]D-Fender | p3nguin, Yes, and your UOM is not the one people care about here :p |
| 20:20.47 | p3nguin | I was just describing it incorrectly for THIS forum. |
| 20:20.50 | generalhan | yea, this got somewhat crazy! lol. i was just hoping to end up with a round about number to see if a 4.5M connection would be enough to handle my 30 concurrent users, plus their data |
| 20:21.18 | _Corey_ | generalhan: It depends on their data usage, but sure |
| 20:21.28 | [TK]D-Fender | generalhan, Yes <- |
| 20:21.29 | p3nguin | You'll use up half of it for your phones along. |
| 20:21.30 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-123-Miami.FL.hfc.comcastbusiness.net) |
| 20:21.31 | _Corey_ | generalhan: You could also go g.729 and cut that down |
| 20:21.40 | Wazzmann02 | I didn't realize phones registering accross the wan without a local IAX server would use so much morebandwidth, it seems like it's 10x or more |
| 20:21.42 | p3nguin | s/along/alone/ |
| 20:21.45 | [TK]D-Fender | Yes, switching codecs would be a huge savings |
| 20:22.03 | _Corey_ | you'd be around 1mbps for 30 concurrent w/g.729 |
| 20:22.18 | leifmadsen | the purpose of IAX2 trunking is applicable directly to what you're discussing |
| 20:22.19 | Wazzmann02 | is there any reason at all NOT to switch from ulaw to g729? |
| 20:22.29 | p3nguin | quality change |
| 20:22.40 | leifmadsen | it was developed for this very reason (to reduce bandwidth to allow extra calls over the same T1 circuit) |
| 20:22.43 | Wazzmann02 | which codec is the highest quality? |
| 20:22.53 | leifmadsen | Wazzmann02: price |
| 20:22.54 | _Corey_ | cost of codec (pretty trivial), perceived quality (hard to tell), transcoding cpu cost |
| 20:22.56 | Wazzmann02 | (that asterisk easily supports) |
| 20:22.58 | leifmadsen | 1 channel == $10 |
| 20:23.06 | leifmadsen | what _Corey_ said |
| 20:23.16 | SuperNull | leifmadsen does RTCP require special compiling options or is that default (1.4+) ? |
| 20:23.23 | leifmadsen | default |
| 20:23.27 | leifmadsen | just need to access it |
| 20:23.30 | leifmadsen | and do something with it |
| 20:23.31 | _Corey_ | Wazzmann02: g.722 is the most widely supported quality coded, though 10 supports some nicer codecs |
| 20:23.31 | Wazzmann02 | so you have to buy g729 for your asterisk server? 1 copy per phone? or per concurrent call? |
| 20:23.32 | p3nguin | For usage to the PSTN, ulaw/alaw is the best quality. |
| 20:23.37 | leifmadsen | accuracy is questionable |
| 20:23.39 | generalhan | G.711: Raw number of calls per megabit: 15, GSM: Raw number of calls per megabit: 68, G.729: Raw number of calls per megabit: 103. this is what i was looking at from a voip-info.org page |
| 20:23.46 | leifmadsen | Wazzmann02: per concurrent call |
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| 20:24.09 | Wazzmann02 | heh @ 10$, that's only 230$ one-time for a full pri |
| 20:24.30 | generalhan | oh and it clearly states IAX2 Bandwidth Study at the top ! |
| 20:24.37 | _Corey_ | generalhan: That's wrong |
| 20:24.42 | Wazzmann02 | does ulaw have a g### value? |
| 20:24.58 | [TK]D-Fender | Wazzmann02, Where the PSTN is concerned, G.711 |
| 20:25.31 | Wazzmann02 | thanks :) you guys are truly amazing |
| 20:26.10 | Wazzmann02 | asterisk is such an incredible platform to give away freely to the public |
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| 20:27.16 | irule | what is nicer, 1.8 or 10.3 version asterisk? |
| 20:27.41 | p3nguin | What is your definition of nice? |
| 20:28.15 | [TK]D-Fender | checkout time, BBL |
| 20:30.29 | irule | p3nguin I just dont get the difference, but the sudden multi-number jump |
| 20:30.42 | p3nguin | ~asterisk10 |
| 20:30.42 | infobot | Asterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ |
| 20:31.07 | watchy | free taco coupon with install? |
| 20:31.17 | leifmadsen | irule: basically... 10 == 1.10 but s/1.// |
| 20:31.26 | leifmadsen | Asterisk 1.10 |
| 20:31.29 | leifmadsen | s/1.// |
| 20:31.34 | leifmadsen | magic |
| 20:31.47 | pabelanger | No, asterisk 2 |
| 20:32.02 | leifmadsen | eyes the ban button |
| 20:32.03 | _Corey_ | lol |
| 20:32.10 | leifmadsen | :) |
| 20:32.32 | SuperNull | sip show channelstats is empty as heck, does it not like SQL based peers ? |
| 20:32.56 | p3nguin | irule: To summarize... if there will never be a version 2, why continue with the 1. prefix? Drop the 1. and make the next major version 10 instead of 1.10. |
| 20:33.44 | leifmadsen | SuperNull: probably not unless they are cached |
| 20:33.49 | leifmadsen | and maybe not even then |
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| 20:34.04 | SuperNull | well that just made it worthless. I think we need to get off asterisk for what we are using it for .. |
| 20:34.14 | SuperNull | more problems than solutions it seems. |
| 20:34.24 | p3nguin | There are other products waiting for you. |
| 20:34.36 | SuperNull | yeah i know .. :( |
| 20:35.00 | SuperNull | but these dumbasses developed software exlusively for Asterisk and Linksys SPA series adapters. its retarded. |
| 20:35.18 | SuperNull | no winning this. |
| 20:36.38 | SuperNull | Now we have problems due to literally 8000+ lines running on asterisk with no real quality control.. its no good. |
| 20:37.04 | _Corey_ | 8000 channels on one machine? |
| 20:37.29 | SuperNull | lol no |
| 20:37.40 | SuperNull | that would be very good. |
| 20:38.11 | _Corey_ | um yeah |
| 20:38.17 | _Corey_ | :) |
| 20:38.18 | SuperNull | probably 500ish to 1000 per machine not in calls but registered. |
| 20:38.20 | p3nguin | How could I get a tone to play into the call when I barge in using ChanSpy? I need to basically announce my presence with a tone right before my audio goes live into the call. |
| 20:38.55 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 20:39.01 | SuperNull | i have a feeling freeswitch is going to be the answer. |
| 20:39.16 | _Corey_ | SuperNull: I missed what your problem is |
| 20:39.49 | Wazzmann02 | Thanks all, checkout time for me too |
| 20:40.32 | *** join/#asterisk jsjc (~Adium@6.Red-83-59-183.dynamicIP.rima-tde.net) |
| 20:41.21 | SuperNull | _Corey_ i am ultimately trying to get some kind of quality control .. (lost packets/jitter) so we can keep track of call quality.. per call. |
| 20:42.04 | SuperNull | i think i recall seeing a custom CDR SQL insert that included call quality.. hm. |
| 20:42.14 | watchy | anyone playing with asterisk and ipv6? |
| 20:42.27 | SuperNull | someone should be if they are not lol |
| 20:42.50 | watchy | i was thinking of trying 100% ipv6 here on our new rollout |
| 20:44.07 | SuperNull | seems interesting.. |
| 20:44.18 | _Corey_ | SuperNull: I know solutions are out there from Empirix and Wildpackets and whatnot to do this on a network level... I doubt the boxes themselves would be aware enough to reliably report on this |
| 20:49.40 | *** join/#asterisk timahvo1 (~rogue@197.178.81.116) |
| 20:51.15 | SuperNull | looks like someone ghetto rigged something enough to be useful ;) |
| 20:52.02 | *** join/#asterisk timahvo1 (~rogue@197.178.81.116) |
| 20:52.11 | SuperNull | http://www.voip-info.org/wiki/view/Asterisk+RTCP |
| 20:53.23 | SuperNull | voip-info has saved me on many things.. hope this site doesnt go down heh. they have some of the more obscure asterisk things. |
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| 21:24.03 | SuperNull | success, i killed any form of real communication in an IRC channel. |
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| 21:27.46 | watchy | yes you did |
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| 21:32.01 | pabelanger | SuperNull: I suspect shitty on all things asterisk might have done it too |
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| 21:33.55 | pabelanger | s/shitty/shitting |
| 21:35.27 | p3nguin | So close. |
| 21:35.43 | p3nguin | s/close/far away/ |
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| 23:38.35 | kresp0 | hi all |
| 23:38.57 | WIMPy | lo one |
| 23:40.42 | kresp0 | I'm trying to get DUNDi to work, but when I do "dundi query 90:fb:a6:4c:7b:8c@priv" I get: [Apr 25 01:28:57] NOTICE[30307]: pbx_dundi.c:1320 update_key: No such key 'kalawela' for creating RSA encrypted shared key for '90:fb:a6:4c:7b:8c'! |
| 23:41.13 | kresp0 | but the key is on /var/lib/asterisk/keys/kalawela.pub and is readable by everyone |
| 23:42.12 | kresp0 | Why asterisk can't find it? |
| 23:42.40 | WIMPy | Turn up verbose and debug and try a 'module reload pbx_dundi'. That might give you something usefull. |
| 23:42.57 | kresp0 | thanks WIMPy, I'll try right now |
| 23:44.28 | kresp0 | WIMPy: sorry for the noob question, turn up verbose and debug where? |
| 23:44.43 | leifmadsen | console |
| 23:44.49 | leifmadsen | *CLI> core set verbose 5 |
| 23:44.54 | leifmadsen | *CLI> core set debug 10 |
| 23:44.57 | WIMPy | in *cli 'core set verbose 9' and same with debug. |
| 23:45.22 | kresp0 | thanks leifmadsen & WIMPy |
| 23:45.24 | leifmadsen | kresp0: what is your configuration for the peers? |
| 23:45.50 | leifmadsen | sounds like the peer kalawa isn't setup to pass the right key to the peer, or the peer isn't setup to receive the correct key |
| 23:45.58 | leifmadsen | you have to be very careful when configuring dundi |
| 23:46.02 | kresp0 | leifmadsen, I'm following this howto: http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP+with+no+passwords |
| 23:46.07 | leifmadsen | inkey and outkey can be tricky |
| 23:46.15 | leifmadsen | likely very out of date |
| 23:46.23 | leifmadsen | try the configurations I wrote at asteriskdocs.org |
| 23:47.12 | kresp0 | thanks again leifmadsen, I'll do so after debuggin a bit this error |
| 23:47.55 | WIMPy | BTW: Are the IDs that are supposed to be MAC addresses limited in some form? |
| 23:48.40 | WIMPy | Seeing a warning about a MAC isn;t very useful to me. My brains ARP and DNS cache are too non eistant. |
| 23:48.46 | leifmadsen | WIMPy: ya they need to be in the form of a MAC address |
| 23:48.50 | WIMPy | s/eis/exis/ |
| 23:48.53 | leifmadsen | xx:xx:xx:xx:xx:xx |
| 23:49.02 | WIMPy | bad luck. |
| 23:49.06 | leifmadsen | indeed |
| 23:49.13 | leifmadsen | it's bothered me for a bit but never bothered changing it |
| 23:49.37 | leifmadsen | not sure if the protocol has a limitation in the space available for transporting device ID |
| 23:49.55 | leifmadsen | MAC address was a good idea, but also a very bad idea :) |
| 23:50.13 | leifmadsen | what I probably need to do is add something like I did to SIP so that there is at least a description field you can set in the configuration file |
| 23:50.19 | WIMPy | Or add a name= option. |
| 23:50.27 | leifmadsen | that way when you do: dundi show peers -- it shows something reasonable |
| 23:50.31 | WIMPy | Yes, that' would work. |
| 23:50.42 | leifmadsen | Asterisk 10 is useful in that regard :) |
| 23:50.59 | leifmadsen | pretty proud of that feature ;) |
| 23:51.18 | leifmadsen | I've backported it a few times because it's so useful when looking at mac addresses when you use them for device names/identifiers |
| 23:51.40 | WIMPy | Soo many options for good features... |
| 23:52.26 | leifmadsen | heh |
| 23:52.44 | leifmadsen | at least we have the option of adding them |
| 23:53.07 | WIMPy | So many possibilities, so little time :-( |
| 23:53.14 | leifmadsen | indeed |
| 23:53.22 | leifmadsen | since changing companies I've been doing mostly PHP :( |
| 23:53.35 | leifmadsen | at least it's PHP generating configuration and changes for an Asterisk backend |
| 23:53.50 | leifmadsen | we're about to roll out our first 1.8 system next week |
| 23:53.50 | WIMPy | I also thought about a dundi<>enum gateway after the enum switch has sadly been removed. |
| 23:54.11 | leifmadsen | I don't use switches... |
| 23:54.21 | WIMPy | managed to stay clear from php luckily. |
| 23:54.26 | leifmadsen | actually that's not entirely true as I've inherited a system that uses them |
| 23:54.48 | WIMPy | They are the most useful and powerful thing. |
| 23:54.52 | leifmadsen | I kind of like php, but that's because I have the most experience with it |
| 23:55.50 | WIMPy | Probably PHP isn't really bad, It's jsut 99% of the ppl using it that give it a reputation I don't want to get anyway near. |
| 23:56.10 | leifmadsen | you can read a lot about people who hate PHP |
| 23:56.17 | leifmadsen | been learning ruby lately since I've been doing a lot of devops stuff |
| 23:56.20 | leifmadsen | chef and puppet |
| 23:58.17 | WIMPy | BTW: If someone finds that dundi<>enum thing interesting, go ahead *eg* |
| 23:58.46 | leifmadsen | :) |
| 23:58.47 | leifmadsen | <G> |
| 23:59.06 | leifmadsen | where were you pulling enum data from?> |
| 23:59.27 | kresp0 | enum = dns based = centralized authority system |
| 23:59.31 | WIMPy | dns? |