00:00.10 | yo-mama | Meaw: if you HAVE to use g729 and you are REALLY cheezy, buy one license, get it working and then use cracked for the other channels |
00:00.32 | pmhaddad | Meaw, i would just avoid g729 - i see no real reason why you need it |
00:00.44 | pmhaddad | unless you know something about your setup i don't |
00:00.45 | yo-mama | Meaw: how new is your install? |
00:00.53 | pmhaddad | which is totally possible |
00:00.58 | yo-mama | Meaw: how good are you with Linux? |
00:00.59 | [TK]D-Fender | yo-mama: Doesn't work |
00:01.55 | yo-mama | [TK]D-Fender: You used to be able to do that trick! I have a few of them scattered throughout Cali! |
00:01.58 | yo-mama | :P |
00:02.37 | Meaw | yo-mama, im not the one who did the whole setup, we hired a guy to do it but im trying to finalize it.. we have a heavy traffic on the E1 thats why im trying to get this done at this time, after few hours i should unplug the E1 and put it on the original server |
00:02.39 | yo-mama | Meaw: Just reinstall asterisk! |
00:02.57 | pmhaddad | i already suggested that |
00:04.00 | yo-mama | pmhaddad: all hes has to do is backup his /etc/asterisk/ folder and ./configure && make clean && make && make install |
00:04.44 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
00:06.16 | yo-mama | Hey, who here is am sms guru? |
00:15.07 | cvnet | if incoming call from DID is SIP which hits the asterisk box, and your outbound provider is h323 would asterisk translate it by itself? |
00:17.40 | [TK]D-Fender | cvnet: Just like EVERY call. 2 channels bridged. Not Raw-Cat Science you know. Never called from a SIP phone and out a DAHDI/ZAP channel before? |
00:24.32 | nkohh | i've never even built dahdi =( |
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00:45.28 | Meaw | allright, i have recompiled asterisk |
00:45.38 | Meaw | now im getting this error when i try to load module g729 |
00:45.40 | Meaw | Error loading module 'codec_g729-ast16-gcc4-glibc-core2.so': /usr/lib/asterisk/modules/codec_g729-ast16-gcc4-glibc-core2.so: cannot restore segment prot after reloc: Permission denied |
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01:58.09 | kc8pxy | [TK]D-Fender: "raw-cat science"??? |
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02:17.46 | [TK]D-Fender | kc8pxy: hukt on fonix werkt 4 u |
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04:47.06 | joelsolanki | Hi all |
04:47.18 | joelsolanki | i have installed asterisk 1.6 for testing purpose |
04:47.51 | joelsolanki | I need to test the g729 codec stuff. can anybody provide me link for free g729 codec which will work with asterisk 1.6.0 |
04:47.52 | joelsolanki | ? |
04:50.06 | drmessano | joelsolanki: Doubt it.. Digium handles the licensing for legal use of G729, and using anything else is more or less "warez". |
04:52.40 | joelsolanki | yes i will buy licenses once this setup works |
04:52.52 | florz | that would very much depend on the jurisdiction you are in, I suppose |
04:52.53 | joelsolanki | i use asterisk 1.4 and have licenes for it. |
04:53.14 | joelsolanki | but i cant move the licenses until i test it sucessfully on asterisk 1.6 |
04:53.20 | trnzmeta | http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ |
04:53.22 | joelsolanki | therefore i need g729 for testing purpose. |
04:53.24 | joelsolanki | ok |
04:54.42 | drmessano | Nice |
04:54.57 | joelsolanki | http://asterisk.hosting.lv/#bin |
04:55.06 | trnzmeta | yeah that's the one |
04:55.09 | joelsolanki | i actually downloaded g729 from above location |
04:55.22 | trnzmeta | same same but different |
04:55.22 | joelsolanki | i have asterisk 1.6.0 and core2duo server |
04:55.39 | joelsolanki | so which codec i should download ? |
04:55.49 | joelsolanki | i see alot |
04:56.02 | joelsolanki | codec_g729-ast16-icc-glibc-core2.so ? |
04:56.59 | joelsolanki | trnzmeta ? |
04:57.26 | trnzmeta | just read the info |
04:57.33 | trnzmeta | cat /proc/cpuinfo |
04:57.48 | trnzmeta | and see which flags your processor supports and match equiv |
04:59.12 | joelsolanki | i see following |
04:59.13 | joelsolanki | <PROTECTED> |
05:00.20 | *** mode/#asterisk [+b trnzmeta!n=bleh@secure27.lnk.telstra.net] by Qwell |
05:00.31 | *** kick/#asterisk [trnzmeta!i=north@pdpc/sponsor/digium/Qwell] by Qwell (google: contributory infringement.) |
05:01.12 | joelsolanki | oh trnzmeta was kicked ? why ? |
05:01.28 | drmessano | google: contributory infringement. |
05:02.09 | joelsolanki | ok |
05:02.34 | Qwell | joelsolanki: you are also lucky you're still here. |
05:02.52 | Qwell | You *cannot* use those codecs. Period. |
05:03.03 | ltd_wk | joelsolanki: g.729 is patent encumbered, Digium sells licenses commercially. The above codec is infringing on that patent. You do the math. |
05:03.18 | joelsolanki | I see. |
05:03.33 | drmessano | tried to warn earlier |
05:03.42 | joelsolanki | qwell: so how can i test g729 ? |
05:03.49 | Qwell | Buy a license |
05:03.53 | joelsolanki | it will work in pass thru mode if codecs is not installed ? |
05:03.54 | *** join/#asterisk jbjuly (n=joelbrya@203.177.143.137) |
05:03.58 | Qwell | yes |
05:03.59 | florz | you people are aware that patents are a terrirorial matter? |
05:04.09 | drmessano | Same way you can test Vista or AutoCAD? |
05:04.10 | Qwell | florz: most countries respect these patents |
05:04.39 | Qwell | Anyways, *I* and *this channel* will not take part. |
05:05.11 | joelsolanki | well i tried the calls without installing g729 to use as pass thru mode. but it didnt worked. the case was i wanted transcoding from g711 to g729 |
05:05.23 | Qwell | That isn't pass through... |
05:05.24 | joelsolanki | i think pass thru g729 will not transcode right ? |
05:05.34 | joelsolanki | yes got it. |
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05:05.37 | Qwell | that is what pass through means |
05:05.55 | *** mode/#asterisk [-b trnzmeta!n=bleh@secure27.lnk.telstra.net] by Qwell |
05:06.17 | *** mode/#asterisk [+b trnzmeta!n=bleh@secure27.lnk.telstra.net] by Qwell |
05:06.22 | *** mode/#asterisk [+b *!*n=bleh@*.guard.com.au] by Qwell |
05:06.22 | *** kick/#asterisk [supa_disko!i=north@pdpc/sponsor/digium/Qwell] by Qwell (ban evasion) |
05:06.25 | joelsolanki | means what ever codec the client has it will be passed. if customer is using linksys ata with g729 and we dont have codec installed on asterisk then asterisk will just pass g729 codec right ? |
05:06.29 | Qwell | I was going to remove it.. |
05:07.21 | joelsolanki | qwell: is my understanding correct above ? |
05:07.27 | Qwell | joelsolanki: unless you need to do things with the audio, yes |
05:07.34 | joelsolanki | hmm ok |
05:07.46 | *** join/#asterisk Techdeck (i=Techdeck@77.125.43.202) |
05:07.51 | Techdeck | hey guys |
05:07.53 | ltd_wk | joel: It's only $10 USD per concurrent call channel. |
05:08.04 | Techdeck | can you guys help me with G.729? |
05:08.08 | joelsolanki | it was nice if digium had provided 1 g729 license for free. just for testing prupose :) |
05:08.20 | Qwell | joelsolanki: tell the patent owners. |
05:08.26 | joelsolanki | hehe : |
05:08.27 | joelsolanki | :) |
05:08.40 | Techdeck | I'm getting a permission denied error when trying to load the .so file |
05:08.44 | KyleK | can I get some basic info on g729? I don't get why its a bfd that its patented, or that people want to use it |
05:09.00 | Qwell | KyleK: the size/quality are "good" |
05:09.05 | Techdeck | were you guys just talking about it?? |
05:09.37 | ltd_wk | I don't agree that the quality is good, but the size certainly is :> |
05:09.58 | Techdeck | anyone? |
05:10.02 | Qwell | ltd_wk: sure, that's fair. I was saying the ratio though |
05:10.10 | Qwell | are there better? absolutely |
05:10.43 | Techdeck | are you ignoring me? |
05:10.51 | Techdeck | is it because I'm not a female? |
05:10.53 | Qwell | Techdeck: no.. permission denied doing what? |
05:11.02 | Techdeck | Qwell, loading the module |
05:11.06 | Qwell | how? |
05:11.13 | KyleK | Techdeck: permission denied usually means cant read the file or its the wrong file |
05:11.13 | Techdeck | module load .. |
05:11.24 | Techdeck | the file is wrong, and I chmod'd it to 777 |
05:11.27 | Techdeck | err, right* |
05:11.30 | Qwell | check selinux stuff |
05:11.37 | Techdeck | oh, good idea |
05:11.39 | Techdeck | let's see |
05:11.51 | Qwell | otherwise, I would suggest calling Digium support |
05:12.18 | Techdeck | I did.. |
05:12.46 | Techdeck | they told me "You are doing something wrong then" |
05:13.57 | Techdeck | why thank you for that valuable information |
05:14.06 | Techdeck | I hate tech support |
05:14.40 | Techdeck | hmm nope, selinux seems disabled |
05:14.57 | Techdeck | any other ideas? |
05:16.06 | jbjuly | just compiled and installed the latest beta 1.6.2 package, but I keep getting Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exit?) -- and it exist. |
05:16.15 | Qwell | Techdeck: Do you have other modules in that dir? |
05:16.28 | Techdeck | yeah, tons |
05:16.42 | Qwell | pastebin the the console output |
05:16.56 | Techdeck | gimme one sec, I think I found something |
05:17.33 | Techdeck | nvm, I had a typo, my bad |
05:17.34 | Techdeck | thanks! |
05:18.20 | jbjuly | just compiled and installed the latest beta 1.6.2 package, but I keep getting Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
05:18.33 | nkohh | is asterisk running? |
05:18.39 | nkohh | ps aux | grep astersisk |
05:18.49 | jbjuly | no |
05:18.57 | nkohh | well then that would be a fairly severe problem |
05:19.02 | nkohh | what're you typing? |
05:19.05 | Techdeck | maybe you should grep asterisk then :P |
05:19.05 | nkohh | what command |
05:19.19 | nkohh | Techdeck: you meanGNU grep doesn't figure out what i want to search for, too?!?! |
05:19.20 | jbjuly | asterisk -rvvvv |
05:19.24 | nkohh | jbjuly: type asterisk first |
05:19.25 | nkohh | just asterisk |
05:19.30 | nkohh | and then press the return button (enter) |
05:19.37 | nkohh | then try the -rvvvv again |
05:19.48 | jbjuly | same problem |
05:19.51 | Techdeck | nkohh, they have it as a feature request |
05:19.52 | Techdeck | :) |
05:20.10 | nkohh | jbjuly: did it output anything when you typed asterisk? |
05:20.42 | jbjuly | none |
05:20.55 | nkohh | well that is very interesting |
05:21.08 | Qwell | jbjuly: When you installed, did you clean out the modules dir like you're supposed to? |
05:22.56 | jbjuly | It's a fresh 1.6.2.0 beta1 compile and install |
05:23.09 | Qwell | and you've never had Asterisk on this system? |
05:23.16 | jbjuly | yes |
05:24.20 | nkohh | jbjuly: sounds like you might be suffering from bidirectional checksum rejection |
05:24.32 | nkohh | do you have any anti static wrist bands? |
05:24.37 | nkohh | you're gonna need three or four |
05:24.53 | jbjuly | i did restarted several times, changing /var/*/asterisk owner/groups/permissions, still the same problem |
05:26.18 | Qwell | asterisk -c |
05:26.22 | Qwell | put the entire output on pastebin.com |
05:32.59 | jbjuly | http://pastebin.archlinux.fr/347023 |
05:33.46 | Qwell | add a few v's |
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05:36.18 | jbjuly | http://pastebin.archlinux.fr/347024 |
05:36.58 | Qwell | So, I'm going to ask again without the possibility of a negated answer. |
05:37.06 | Qwell | Have you had Asterisk installed on that system before this? |
05:39.00 | seanbright | pickle |
05:39.20 | nkohh | Qwell: no, he hasn't had it installed before this. he installed it himself. |
05:39.29 | jbjuly | no, i had asterisk 1.6.0 but uninstalled it |
05:39.56 | Qwell | Okay, and did you clean out the modules when you installed, like you're supposed to? |
05:40.16 | jbjuly | how do I clean the modules? |
05:40.54 | Qwell | You know the big warning message that gets output when the `make install` is finished? |
05:40.54 | Qwell | that |
05:41.34 | jbjuly | oldmodcheck: |
05:42.27 | seanbright | rm /usr/lib/asterisk/modules/*.so |
05:42.29 | seanbright | gmake install |
05:42.37 | seanbright | who do i have to screw around here to get you to try that? |
05:42.43 | seanbright | screws himself |
05:49.28 | lanning | remember, "Righty, tighty." :) |
05:50.43 | seanbright | way ahead of you |
05:50.46 | seanbright | but for serious |
05:51.24 | drmessano | LEFT LOOSEY |
05:51.32 | drmessano | Err shit |
05:51.35 | drmessano | LEFTY LOOSEY |
05:51.57 | drmessano | Two wrongs dont make a right, but three left's do |
05:56.36 | jbjuly | THANKS It works! |
05:57.08 | seanbright | Qwell: nice work |
05:57.35 | Qwell | doesn't that warning say something like "YOU REALLY SHOULDN'T IGNORE THIS"? :( |
05:57.50 | seanbright | Qwell: people run stop signs all the time |
05:58.35 | seanbright | someone buy me a rubik's cube |
05:58.42 | seanbright | annnnnnd go |
05:58.50 | Qwell | 3x3? |
05:58.58 | timgws | 9x9 |
05:59.07 | timgws | make it something that is more sudoku-ish |
05:59.08 | timgws | :D |
05:59.10 | seanbright | 3x3 yes |
05:59.12 | seanbright | the classic |
05:59.13 | timgws | hi guys |
05:59.22 | Qwell | rubix dodecahedron? |
05:59.29 | seanbright | bite your tongue |
05:59.59 | timgws | http://twistypuzzles.com/forum/download/file.php?id=3713&sid=b8af5db7c60d41c95ffe97d036b46f97 |
06:00.04 | timgws | http://twistypuzzles.com/forum/download/file.php?id=3714&sid=b8af5db7c60d41c95ffe97d036b46f97 |
06:00.09 | timgws | ^ pictures of the 9x9 :D |
06:01.03 | seanbright | that's pretty hot |
06:01.16 | timgws | http://twistypuzzles.com/forum/viewtopic.php?t=6315 xD |
06:02.07 | seanbright | sure. ruin it for me. |
06:04.08 | KyleK | argh too many tabs |
06:04.48 | seanbright | you gotta order something if you want a tab |
06:05.55 | KyleK | i mean in google chrome |
06:06.23 | seanbright | i mean in BttF |
06:06.34 | seanbright | is tired of people not getting his movie references |
06:06.37 | seanbright | goes to sleep |
06:06.47 | KyleK | ah, well at 36 tabs i lose the icons |
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06:07.11 | KyleK | eh i cant even remember the correct pronounciation of 1.21 jiggawatts |
06:07.36 | KyleK | apparently its one point two one, not one point twenty-one |
06:08.16 | Qwell | KyleK: it's the latter |
06:08.39 | KyleK | oh so I was right |
06:09.42 | Qwell | and it's gigawatts |
06:10.25 | KyleK | well hes got a bit of a jay sound in it |
06:10.43 | Qwell | doesn't mean it's spelled that way :p |
06:11.02 | Qwell | http://www.moviewavs.com/0085412111/WAVS/Movies/Back_To_The_Future/greatscott.wav |
06:11.33 | Qwell | bed |
06:11.44 | KyleK | good night |
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06:14.33 | KyleK | are the asterisk sounds available in g729? |
06:14.46 | MaliutaLap | I have a set somewhere |
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06:14.58 | MaliutaLap | or you can always transcode them |
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07:04.44 | salzh | can ooh323 support video now? |
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07:09.00 | henk | moin |
07:10.23 | henk | i have a phone number from a sip provider (let's assume it's 12345), is it possible to have different extensions with that number, let's say 12345-1 and 12345-2? i tried dialling another digit after my own number, but the call is never seen on my asterisk. |
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07:45.59 | tamiel | Hello, sometimes after restarting asterisk, I have some pri channels stuck with "PRI Flags: Resetting". Any idea ? |
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08:13.29 | joobie | sup ladies |
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08:43.14 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
08:45.17 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
08:47.36 | tzafrir_laptop | tamiel, what versions of asterisk? libpri? |
08:49.54 | tamiel | tzafrir_laptop: asterisk 1.4.24.1 and last libpri 1.4.9 |
08:50.55 | tamiel | tzafrir_laptop: and I have this warning : chan_dahdi.c: Unable to specify channel 1: Device or resource busy |
08:52.03 | *** join/#asterisk defswork (n=andy@mx2.3gcomms.co.uk) |
08:59.23 | KyleK | what are people using to transcode files to g729? |
09:01.03 | *** join/#asterisk amaache (n=maa@80.249.75.230) |
09:01.15 | *** join/#asterisk andrebarbosa (n=andrebar@212.13.49.67) |
09:04.09 | KyleK | nm "file convert" |
09:10.30 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
09:12.40 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
09:12.57 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
09:17.01 | amaache | How can i know if my PSTN phone line is numeric |
09:21.04 | KyleK | ? |
09:26.03 | *** join/#asterisk techie (n=techie@adsl-76-214-5-228.dsl.lsan03.sbcglobal.net) |
09:27.12 | amaache | i have to connect my * TrixBox with a old analogic hicom PBX |
09:27.15 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
09:27.33 | amaache | i have TDM4000 |
09:27.42 | amaache | i have TDM400 |
09:30.41 | amaache | can i connect * to PBX hicom with E1/mic |
09:31.11 | joseph__ | guys any one worked before with app_dial.c |
09:36.02 | *** join/#asterisk ITguru (n=ITGuru@webfax.impactteachers.com) |
09:38.16 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
09:38.27 | *** join/#asterisk helloritesh (n=hellorit@116.197.178.83) |
09:39.09 | helloritesh | any recommendation for a SIP trunk provider for US and International calls. I need to setup immediately so online activation is required? |
09:40.30 | helloritesh | Also, I have outbound calls through Manager's interface and I wanted to know if there is a way (simialr to Dial Command) to specify multiple carriers (trunks) and the first one to answer gets the call? |
09:43.45 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
10:07.36 | viraptor | how safe is it to call dial_exec() from the chan_sip.c ? |
10:08.09 | viraptor | (dial being a module and all that stuff) |
10:11.05 | amaache | Hi,can i connect * with mic/siemens Hicom? |
10:12.06 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
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10:21.08 | *** join/#asterisk huye (n=huye@soho2.i-xanadu.com) |
10:32.38 | Jimbo12 | hi - has anyone in here sucessfully managed to connect an Asterisk 1.6.1.0 box with Microsoft OCS 2007 R2? |
10:46.34 | *** join/#asterisk torrikft (n=afraguas@m85-94-191-59.andorpac.ad) |
10:47.17 | torrikft | morning all, im trying to integrate OCS2007 R2 with asterisk 1.6.9 has anybody experience with this setup? |
10:50.47 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
10:51.56 | *** join/#asterisk bobsaccamano (i=cb7e888e@gateway/web/ajax/mibbit.com/x-30197eca311e1d9f) |
10:52.26 | bobsaccamano | hi..is there an option for enabling/disabling call forwarding in asterisk? |
10:53.22 | bobsaccamano | I am hooking it up with a client device that has call forwarding support..so i want to separate server side and client side call forwarding |
10:55.45 | *** join/#asterisk helloritesh (n=hellorit@116.197.178.83) |
10:56.08 | helloritesh | Guys, any recommendation for SIP trunk provider? I need to route some calls asap |
10:56.09 | bobsaccamano | anybody there? |
11:00.34 | *** join/#asterisk proxium (n=proxium@196.203.51.238) |
11:02.12 | helloritesh | . |
11:02.24 | proxium | Hi, everyone, Finally I can use my Vicidial to make manual oubound call and my previous problem was the conference error with Meetme and the time synchronisation in Vicidial 2.0.5 |
11:03.48 | proxium | Now with thos issue Fixed I receive this error in CLI when I do predective Outbound Call: [May 6 11:58:47] ERROR[3085]: chan_sip.c:15919 sipsock_read: We could NOT get the channel lock for SIP/tofreepbx1-0a54b700! |
11:03.48 | proxium | [May 6 11:58:47] ERROR[3085]: chan_sip.c:15920 sipsock_read: SIP transaction failed: 74bd46ba1a032e7243c6bd9d3694e4fd@192.168.1.89 |
11:06.24 | proxium | Any idea about this ? |
11:09.03 | *** join/#asterisk Great_Anta_Baka (n=tensai@196.33.159.83) |
11:30.12 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
11:41.41 | *** join/#asterisk gpuk3 (n=me@ARennes-205-1-6-216.w80-14.abo.wanadoo.fr) |
11:44.29 | gpuk3 | hi all. We're running asterisk1.6.0.6 and have a pure voip setup (i.e. no Digium hardware or FXS/FXO cards). We have a Digium TDM411B PCI card arriving tomorrow and I wanted to know if there is a way I can check that our build of asterisk will support it without a re-compile? |
11:45.31 | gpuk3 | Presumably the kernel will need the correct modules to talk to the card? |
11:46.23 | beek | gpuk3: You'll need to compile DAHDI, then recompile Asterisk. |
11:47.25 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
11:49.19 | gpuk3 | beek: right. From memory, I think we compiled asterisk with zaptel rather than dahdi |
11:50.09 | gpuk3 | yep, just checked my notes - zaptel 1.4 |
11:50.26 | andrebarbosa | anyone using hpec with dahdi? |
11:50.28 | gpuk3 | but i didn't record what zaptel modules etc. we selected at compile time |
11:51.00 | beek | gpuk3: You're using what version of Asterisk |
11:51.02 | beek | ? |
11:51.10 | gpuk3 | 1.6.1.0 |
11:51.17 | gpuk3 | we elected to stick with zaptel |
11:51.19 | beek | Then zaptel is a total waste of time. |
11:51.21 | gpuk3 | rather than move to dahdi |
11:51.27 | gpuk3 | cos at the time |
11:51.31 | gpuk3 | all we needed was the zt_dummy |
11:51.33 | gpuk3 | timer |
11:51.45 | beek | 1.6 requires DAHDI. |
11:51.53 | gpuk3 | we hadn't planned to use any hardware |
11:52.07 | beek | But now that you do, 1.6 requires DAHDI |
11:52.08 | gpuk3 | ok... looks like i need to get dahdi up and running then |
11:52.18 | gpuk3 | thanks for the heads up |
11:52.22 | beek | It's not all that different from zaptel. |
11:52.30 | gpuk3 | good :) |
11:52.38 | gpuk3 | hopefully wont be too much of a mission |
11:52.45 | beek | It's virtually replace zap_xxxx with dahdi_xxxx |
11:52.59 | gpuk3 | ahh kk |
11:53.09 | gpuk3 | goes off to do it |
11:53.11 | beek | Slightly different configuration files. |
11:53.16 | gpuk3 | thanks again |
11:54.38 | joseph__ | <PROTECTED> |
11:55.06 | *** join/#asterisk stope (n=nobody@69.60.247.142) |
11:55.12 | joseph__ | please inform me if you have an idea on how to ,or just if you know a hint |
11:56.09 | bobsaccamano | hi..i need some pointers on how to write a dialplan for call forwarding in asterisk |
11:58.04 | beek | bobsaccamano: http://tinyurl.com/c9bnrd |
11:59.43 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
12:02.26 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
12:04.31 | bobsaccamano | is there an asterisk function for detecting a busy line in sip? |
12:09.06 | stope | you could check the ${DIALSTATUS} |
12:13.39 | *** join/#asterisk yang (i=yang@CAcert/Assurer/freenode.sponsor.yang) |
12:16.45 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
12:16.45 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
12:17.15 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
12:18.10 | beek | morning [TK]D-Fender |
12:18.26 | bobsaccamano | stope: would DEVICE_STATE function work? |
12:18.46 | [TK]D-Fender | beek: I feel like shit-on-a-stick |
12:19.05 | beek | [TK]D-Fender: Did you have a wild evening or are you ill? |
12:19.56 | [TK]D-Fender | beek: ill. Day 2. Some kind of flu. Barely runny nose, headache, moderate congetion and loss of energy |
12:20.26 | stope | bobsaccamano: http://pastebin.ca/1414157 |
12:20.34 | beek | [TK]D-Fender: You should be in bed getting rest. |
12:21.35 | *** join/#asterisk salzh (n=Administ@122.144.138.49) |
12:22.21 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
12:22.24 | bobsaccamano | stope: thanks! |
12:23.29 | *** join/#asterisk simprix (n=simprix@cosmas.supportdept.com) |
12:33.39 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
12:35.40 | [TK]D-Fender | stope: you've nested an expression unnecessarily. remove the outer $[] |
12:38.51 | *** join/#asterisk jtodd (n=jtodd@88.128.82.57) |
12:38.51 | *** mode/#asterisk [+o jtodd] by ChanServ |
12:40.06 | stope | yes, yes, correct, tx :) |
12:40.33 | stope | I think that snippet was grabbed from another one that had an 'or' in it |
12:43.47 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-a7d297cbeae3e137) |
12:45.05 | leifmadsen | yep... all you're doing is always returning true |
12:45.14 | leifmadsen | since it's just checking if there is data, and since there is data, it is true |
12:45.23 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
12:48.00 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
12:52.33 | leifmadsen | I've got "joinempty=yes" and "leavewhenempty=no", but callers are still falling out of the queue when agents are paused. Am I missing an option? |
12:54.53 | [TK]D-Fender | leifmadsen: Whats the exit var say? |
12:56.37 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
12:57.28 | leifmadsen | [TK]D-Fender: CANCEL |
12:58.12 | leifmadsen | nevermind, that was the agent |
12:58.20 | leifmadsen | I have to add a line of debugging |
12:58.52 | leifmadsen | oh I see the problem I think anyways |
13:02.59 | leifmadsen | yep, I had an IF() function in the Queue() call that was putting in a timeout value from a global variable if no timeout value was not returned from the database. |
13:03.16 | leifmadsen | another IF() function to check for this particular queue and to return null fixed it right up :) |
13:06.54 | [TK]D-Fender | leifmadsen: NA NA NA NA NA! |
13:06.55 | [TK]D-Fender | leifmadsen: Leif is Leif! |
13:07.25 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
13:07.42 | leifmadsen | [TK]D-Fender: I don't want to meet your mom!! |
13:07.51 | [TK]D-Fender | leifmadsen: I just want |
13:07.52 | *** join/#asterisk juanIMP (n=Juancho@200.71.41.22) |
13:13.42 | *** part/#asterisk Sam2002gs (n=Sam2002g@h-213.61.105.202.host.de.colt.net) |
13:13.45 | *** join/#asterisk Sam2002gs (n=Sam2002g@h-213.61.105.202.host.de.colt.net) |
13:16.14 | [TK]D-Fender | leifmadsen: ... tease :p |
13:23.42 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
13:26.31 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.118.128) |
13:27.13 | Katty | hummmm |
13:28.32 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
13:28.36 | [TK]D-Fender | Katty: Mew. |
13:30.38 | *** join/#asterisk bgmarete (n=marebri_@196.201.210.130) |
13:32.02 | defswork | Hey did I update here with my problem the other week ? Turned out the Echo Cancellation chip was dead on my Sangoma A101 |
13:32.17 | jaytee | wow |
13:32.34 | defswork | turn off echo cancellation - all worked |
13:32.40 | beek | morning jaytee |
13:32.44 | defswork | new card in now - works properly |
13:32.48 | *** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130) |
13:32.52 | jaytee | morning beek |
13:33.02 | defswork | Stressful though when on site and nothing works :) |
13:33.29 | *** join/#asterisk tobias (n=tobias@user-0ce2hp1.cable.mindspring.com) |
13:37.37 | leifmadsen | [TK]D-Fender: ! ! ! |
13:37.45 | leifmadsen | tease, lol |
13:38.07 | leifmadsen | sorry, I don't get notified unless I'm looking at the window |
13:38.22 | stope | I'm trying to get SLA working on 1.4.23, am I wasting my time with it? |
13:38.35 | leifmadsen | can't say I've ever used it |
13:38.57 | stope | it works with SIP trunks for outgoing calls, but incoming is where I'm having issues and my customer 'wants it' :( |
13:39.04 | *** join/#asterisk awk_r (n=awk_r@nat/digium/x-3c3a7eb32bf3ffe7) |
13:39.18 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
13:40.25 | bgmarete | Hello guys. Does SugarCRM support Asterisk through any free component? Anyone with experience on this here? |
13:40.26 | [TK]D-Fender | stope: Only works for "lines", not "phones", and requires a lof of rpesence enabled speed-dials. |
13:40.36 | [TK]D-Fender | stope: Maybe some day * will support real SIP-B |
13:40.51 | stope | ok, maybe there's a better approach, I need to place a call on hold and pick it up at another extension... should I just use call parking instead? (hosted pbx, many contexts..) |
13:41.38 | [TK]D-Fender | stope: Yup |
13:41.38 | stope | k, tx |
13:42.25 | *** join/#asterisk wilsonj (n=jeremy@unaffiliated/dethstar) |
13:45.34 | proxium | Hi again, I don't receive any feedback, so I post again (may be [TK]D-Fender has an idea) |
13:45.50 | proxium | I receive this error in CLI when I do predective Outbound Call: |
13:45.57 | proxium | [May 6 11:58:47] ERROR[3085]: chan_sip.c:15919 sipsock_read: We could NOT get the channel lock for SIP/tofreepbx1-0a54b700! |
13:46.07 | proxium | [May 6 11:58:47] ERROR[3085]: chan_sip.c:15920 sipsock_read: SIP transaction failed: 74bd46ba1a032e7243c6bd9d3694e4fd@192.168.1.89 |
13:46.09 | [TK]D-Fender | proxium: What ver of *? |
13:46.12 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
13:46.19 | *** join/#asterisk tobias (n=tobias@user-0ce2hp1.cable.mindspring.com) |
13:46.25 | proxium | 1.4.22 |
13:50.31 | [TK]D-Fender | proxium: Well thats alreadya few versions behind |
13:50.46 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
13:51.40 | proxium | [TK]D-Fender: plz can you explain ? |
13:52.04 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
13:52.30 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
13:52.31 | stope | version 1.4.24.1 > 1.4.22 |
13:52.45 | [TK]D-Fender | proxium: What's to explain? We're up to 1.4.24.1 |
13:52.59 | [TK]D-Fender | proxium: Sorry, I don't teach mathematics. |
13:53.33 | stope | 1.6.1.0 ready for production use? |
13:54.47 | proxium | Ok, I don't need mathematics courses but I don't speak English well that's all |
13:56.29 | beek | proxium: Any time you get strange errors and are not at the newest version of Asterisk you're going to want to upgrade first. |
13:58.27 | proxium | beek: ok, I'll try to upgrade but I should be sure about that error to be (version dependant) not a bad configuration or something else. |
13:58.31 | [TK]D-Fender | stope: 2 words : bleeding edge |
13:58.49 | stope | aye |
13:58.53 | [TK]D-Fender | stope: I would always wait about 1-2 decimal increments for the big bugs to come out |
13:59.10 | stope | 1.4.24 it is .... :) |
13:59.13 | [TK]D-Fender | stope: Or 2 months without any announcements |
13:59.25 | [TK]D-Fender | stope: 1.6.0 is viable. |
13:59.29 | beek | stope: 1.6.0.9 has been rock-solid for me. |
13:59.43 | [TK]D-Fender | stope: remember that 1.6.1 is a whole other branch with the new release cycle |
14:00.06 | stope | hmm, maybe I'll try 1.6.0.9 if it's been a good soldier so far |
14:00.21 | stope | TLS would be good |
14:00.33 | beek | stope: I've had zero issues with the 1.6.0.x branch. |
14:00.34 | *** join/#asterisk el_-- (n=el@asterisk.net.informatik.tu-muenchen.de) |
14:00.42 | stope | k, tx |
14:01.53 | el_-- | Hi I'm using trixbox and I need some help with dialpatterns... The PBX should change the numbers that the international prefix is allways used...e.g. I dial 03 0301234455 it should dial 004930123... |
14:02.12 | el_-- | the first 03 should choose the outbound route |
14:03.37 | *** join/#asterisk mleino (n=mle@140-120.adsl.lpoy.dnainternet.fi) |
14:03.59 | [TK]D-Fender | ~trixbox |
14:04.00 | infobot | rumour has it, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
14:04.04 | [TK]D-Fender | elNot supported here |
14:04.07 | jerlique | I have a registration, which is set to infineltey regsiter ever 20 seconds, yet it stops try after a while. is this a bug> |
14:05.09 | *** join/#asterisk desdesdesdes (i=desdesde@196.211.34.3) |
14:05.28 | desdesdesdes | hi there what does jitterbuffer=yes do? |
14:05.35 | nkohh | jerlique: regardless of whether or not there is something preventing you from reregistering every 20 seconds, it is a fantastically bad idea to register that frequently. |
14:05.57 | nkohh | desdesdesdes: http://www.voiptroubleshooter.com/problems/jitterbuffer.html |
14:06.23 | el_-- | how would a dialpattern look like for asterisk? withoud freepbx? |
14:07.22 | jerlique | nkohh: The main switch sets the minimum to 180, but asterisk is set low, so that if it d/c from the switch, it will reconnect quickly |
14:07.39 | [TK]D-Fender | el_--: depends on a lot of things |
14:07.51 | desdesdesdes | thx nkohh |
14:07.55 | [TK]D-Fender | el_--: and we cannot help you witht he way you have to code this in FreePBX. |
14:08.12 | [TK]D-Fender | el_--: It is not supported here. Please use their support channels for this |
14:08.16 | el_-- | ok thanks alot anyway |
14:12.23 | mleino | Hi, I have a problem in fresh Asterisk + FreePBX installation (Asterisk + addons 1.6.1.0, dahdi 2.1.0.4, dahdi-tools 2.1.0.2, libpri 1.4.9, freepbx 2.5.1). I have set up a conference via freepbx, and when I try to join to conf, first user goes in nicely after entering pin, the second one doesn't, if enter correct pin, nothing happens, line stays muted. And when I disconnect both users from conf, asterisk goes to 99% of cpu, but no errors ca |
14:12.57 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
14:13.58 | *** join/#asterisk oej (n=olle@93.82.216.154) |
14:15.36 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
14:16.34 | plundra | Hmm, isn't there any way to do a iftime in regular ael? :-[ |
14:17.37 | [TK]D-Fender | plundra: Its jsut a function, I don't see why not |
14:17.44 | [TK]D-Fender | plundra: And what is "regular" AEL? |
14:17.44 | plundra | Regular as in not AEL2 :) |
14:18.00 | [TK]D-Fender | plundra: Still on 1.2? |
14:18.25 | plundra | [TK]D-Fender: Asterisk? No 1.6 |
14:18.45 | [TK]D-Fender | plundra: Last I checked there was no AEL as of 1.4 |
14:18.50 | [TK]D-Fender | AEL1* |
14:19.55 | plundra | Hmm, ok? :) I've just assumed that AEL2 always was written with the 2. |
14:23.42 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
14:24.10 | plundra | Gah! :-D It _does_ work. Had to use the |-syntax for the timespec. |
14:24.16 | plundra | [TK]D-Fender: Thanks 8-) |
14:24.53 | desdesdesdes | will jitterbuffer=yes and forcejitterbuffer=yes help in solving delays in speech from fxs extension to iax2 trunk? |
14:26.00 | [TK]D-Fender | desdesdesdes: Doesn't solve latency, only helps cover up PL & mis-ordering better |
14:26.46 | anonymouz666 | and could increase delay |
14:28.20 | desdesdesdes | do you add this line on the iax.conf or zapata.conf or both? |
14:29.12 | desdesdesdes | or clearer question can u use the jitterbuffer in iax.conf |
14:29.17 | jaytee | i've heard of triple-des encryption but never quadruple-des encryption |
14:30.47 | [TK]D-Fender | desdesdesdes: .... jitter is an IP problem. do the math,,, |
14:31.32 | jaytee | nuthin goes into nuthin nuthin times, carry the nuthin.....hmmm, I got nuthin |
14:37.09 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
14:37.52 | jasonwoot | does anyone have an isymphony update mirror address? |
14:39.28 | *** join/#asterisk rbd (n=rbd@rrcs-96-10-27-206.se.biz.rr.com) |
14:39.57 | rbd | hi guys, in a macro is it possible to modify the values of ARG1, ARG2, etc. .... for example: exten => s,n,Set(ARG1=${IF($[ "${CALLERID(name)}" = "ConfExpansion" ]?ltqj:${ARG1}) |
14:40.52 | nkohh | I don't have two phones in my home office to test this, but does Asterisk play music-on-hold if you put a conference on hold? |
14:41.07 | nkohh | I'm not sure if it makes any effort to' differentiate' between the two... |
14:41.11 | nkohh | oops, 'differentiate' |
14:48.11 | *** join/#asterisk therealcircut (i=circut@smoke.dope.org) |
14:48.15 | therealcircut | hey all |
14:48.35 | nkohh | hello |
14:48.45 | therealcircut | i have a bunch of polycom 601's that im trying to set the line keys to dial extensions |
14:48.53 | jaytee | what a funny URL |
14:49.10 | therealcircut | ext, line key1 == my extension, line key2 == dial 701, line key3 == dial 6000 |
14:49.11 | therealcircut | etc.. |
14:49.41 | therealcircut | is this possbile? I have the latest firmware / bootloader and have been combing the documentation but i cant seem to find the documentation |
14:50.25 | therealcircut | to accomplish this... |
14:50.33 | [TK]D-Fender | rbd: Should be able to |
14:50.54 | rbd | ok thanks |
14:51.31 | [TK]D-Fender | therealcircut: set the # of line-keys on your reg to fewer than you have, and the remainder spill over from your phone's directory in SD order |
14:52.05 | [TK]D-Fender | therealcircut: thats what that magical button labeled "Directories" is for ;) |
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14:52.50 | therealcircut | [TK]D-Fender: nice, i was looking for it in the registry stuff |
14:52.54 | therealcircut | will give the directory a go |
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14:58.20 | eppigy | never drinking again |
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14:58.33 | Techdeck | until tomorrow |
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14:58.38 | leifmadsen | btw: I really hate how X-Lite modifies the audio volume when it starts and shuts down |
14:58.50 | leifmadsen | really annoying when you're listening to music with headphones and it turns the volume all the way up |
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14:59.17 | eppigy | i keep my volume all the way up |
14:59.27 | eppigy | i get angry because xlite lowers my volume |
14:59.58 | leifmadsen | volume all the way up is too loud for these headphones, and taking it from nearly all the way down to all the way up hurts my ears |
15:00.04 | leifmadsen | it shouldn't modify the volume at all! |
15:00.11 | nkohh | whenever my coffee grinder is near my computer the screens flicker :( |
15:00.19 | leifmadsen | I don't drink coffee |
15:00.22 | nkohh | you're missing out. |
15:00.26 | leifmadsen | nah |
15:00.29 | nkohh | yah. |
15:00.36 | BlargMaN00 | i drink entirely too much coffee... |
15:00.36 | leifmadsen | if I drank coffee, I'd get addicted to it |
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15:00.43 | Qwell | and? |
15:00.53 | leifmadsen | Qwell: I don't need any more vices |
15:00.54 | BlargMaN00 | i drink about 3-4 pots a day |
15:00.58 | leifmadsen | that'd be me |
15:01.01 | Qwell | but it's coffee |
15:01.09 | leifmadsen | and not good for your heart |
15:01.13 | nkohh | BlargMaN00: me too! one in the morning before i go to my office, then two at work. sometimes one before bed |
15:01.17 | leifmadsen | when you drink that much |
15:01.26 | [TK]D-Fender | gets very cranky when he gets too much blood in his caffeine stream |
15:01.38 | nkohh | yeah, that's why you've got to smoke cannabis. it offsets the jitter ;) |
15:01.43 | leifmadsen | I like being able to wake up and function without coffee |
15:01.43 | BlargMaN00 | [TK]D-Fender: i'm right there with you... |
15:01.49 | eppigy | coffee has lots of antioxidants |
15:01.54 | eppigy | you cannot drink too much |
15:02.03 | leifmadsen | you can always have too much of anything |
15:02.08 | eppigy | lies |
15:02.13 | eppigy | and deceit |
15:02.21 | Qwell | leifmadsen: I'm gonna help you with your addiction to water |
15:02.28 | leifmadsen | Qwell: you can die from too much water! |
15:02.36 | nkohh | water intoxication is serious business |
15:02.38 | Qwell | that's why I'm going to help! |
15:02.39 | leifmadsen | totally |
15:02.49 | stope | just like in the hold your wii contest a few years ago |
15:02.56 | leifmadsen | yep |
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15:03.22 | nkohh | we had a guy die here. these people were supposed to not go to the bathroom for as long as possible and whoever won won a bedroom set or something |
15:03.25 | nkohh | the last guy died |
15:03.27 | nkohh | so nobody got the set. |
15:03.38 | nkohh | 100% true story. |
15:03.39 | BlargMaN00 | that sucks |
15:03.43 | eppigy | lol |
15:03.48 | eppigy | for a damn bedroom set |
15:03.50 | nkohh | BlargMaN00: no, darwin at work |
15:04.00 | leifmadsen | lady died in a contest in N.A. too |
15:04.07 | BlargMaN00 | or karma... which ever you prefer... |
15:04.11 | nkohh | eppigy: it probably had a tv or some such thing... it was quite a while back, I don't remember for sure. it was on local TV and radio and everything.. was a big spectacle... |
15:05.00 | eppigy | better have been a 72'' money printing tv |
15:05.07 | nkohh | lol |
15:06.00 | leifmadsen | that'd be sweet |
15:07.10 | therealcircut | we had a similar incident here in chicago |
15:07.19 | therealcircut | where kids would drink shitloads of water |
15:07.31 | therealcircut | n effectively saturate their bodies / brains |
15:09.38 | jjshoe | busy day here I see ;) |
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15:10.01 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
15:10.13 | BlargMaN00 | i love it when we all get so much work done... 8)~ |
15:10.47 | therealcircut | i got a question for ya |
15:10.49 | jaytee | chkdsk reports 53% complete but that nasty clicking noise is getting on my nerves |
15:11.26 | therealcircut | so on our asterisk server, we have an extension which effectively does: MusicOnHold(default), ParkAndAnnounce() x 4, then hangs up |
15:11.55 | therealcircut | it works, but the problem is when the next ParkAndAnnounce() call is hit, the music on hold, stops , then restarts |
15:12.11 | therealcircut | were using asterisk 1.4.24.1 |
15:12.19 | BlargMaN00 | therealcircuit: that sounds like a pointless extension... |
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15:12.53 | BlargMaN00 | therealcircuit: why would you park a call 4 times?? |
15:13.30 | therealcircut | well the calls come in to our recptionist, then she transfers to 5XX, where XX is the dest EXT |
15:13.46 | therealcircut | its here that they hit that ParkAndAnnounce() routine |
15:14.11 | therealcircut | and it runs 4 times because there are 4 people that it gets announced to |
15:14.46 | therealcircut | im fairly new to asterisk ;/ so i apologize. |
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15:20.08 | therealcircut | [TK]D-Fender: you the man |
15:20.24 | therealcircut | BlargMaN00: so i should Park() the call, then ring those groups? |
15:21.01 | BlargMaN00 | therealcircuit: in theory (i have never tried it) you should be able to use one ParkAndAnnounce() to dial all four extensions... |
15:21.21 | therealcircut | BlargMaN00: yea that was the inital attempt |
15:21.35 | therealcircut | but it didnt work, was complaing about resources being unknown |
15:21.50 | therealcircut | when i used: Local/1@office&Local/2@office.... |
15:22.11 | BlargMaN00 | try creating an extension that just dials those four extensions, and use that in the ParkAndAnnounce() app |
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15:25.53 | [TK]D-Fender | therealcircut: How do you really want to announce it? |
15:26.36 | therealcircut | well when the receptionist gets a call, the person says ' i want to talk to blah..' so she hits xfer, then that persons extension |
15:26.55 | therealcircut | the thing is, it needs to announce it to 4 people, myself, the overhead speaker, and 2 other people |
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15:27.30 | therealcircut | ammend that xfer thing, she hits transfer, then 5XX, where XX is the persons extension |
15:28.29 | therealcircut | so idealy, i would like the person to be put into a parking lot, then ring the people it needs to |
15:28.42 | therealcircut | and if the call isnt answered goto soem default vm |
15:29.11 | *** join/#asterisk salzh (n=Administ@122.144.138.49) |
15:30.18 | salzh | hi, can i set sip calls in Asterisk to work in passthrough mode? |
15:31.09 | [TK]D-Fender | therealcircut: if it hits a VM thats because thats what the exten you are transfering it to does. |
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15:31.24 | [TK]D-Fender | therealcircut: Yuo need to make sure that you send it somewhere that will not do this. |
15:31.55 | [TK]D-Fender | therealcircut: What you should do is call a local channel taht will PAGE them with auto-answer. |
15:32.24 | [TK]D-Fender | salzh: "canreinvite=yes". And only works where NAT is not involved. |
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15:32.38 | [TK]D-Fender | therealcircut: "core show application page" |
15:33.24 | [TK]D-Fender | therealcircut: And set the SIP header to have your Polycom's auto-answer on speakerphone. You will clearly have to prepare this in your phone's provisioning. |
15:34.23 | therealcircut | yea, i have a paging macro setup |
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15:47.28 | el_-- | hi... in which file do I have to place SipAddHeader |
15:47.29 | el_-- | ? |
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15:47.54 | [TK]D-Fender | el_--: Extensions.conf |
15:48.10 | [TK]D-Fender | el_--: Which of course... gets blown away by your GUI |
15:48.43 | deadpigeon | i know this is a voip channel, but out of curiousity anyone here familar with adtran TA1500 or perhaps a CACS Access Navigator GR303 concentrator? |
15:52.04 | el_-- | ok lets assum I do not have the gui problem |
15:52.31 | el_-- | i want to have the following for all extensions specified to SIP calls initiated using Dial(). Non-standard SIP headers should be preceded with an X- as in X-Asterisk-Accountcode:. |
15:52.35 | el_-- | Should be used with caution as different SIP devices expect different headers and respond differently to them. May produce unexpected behavior. |
15:52.38 | el_-- | Returns 0. |
15:52.41 | el_-- | exten => 123,1,SIPAddHeader(X-Asterisk-Account: ${CDR(accountcode)}) |
15:52.43 | el_-- | exten => 123,n,ups |
15:52.46 | el_-- | no |
15:52.50 | el_-- | this: SIPAddHeader(Remote-Party-ID: <sip:${MYNUM}@ipadresse>\;party=calling\;screen=yes\;privacy=full) |
15:53.14 | el_-- | how would I put that into extensions.conf... or do I have to add that for all extensions separately? |
15:55.48 | andrebarbosa | anyone know what asterisk version is used on switchvox 4.0? |
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16:00.53 | ghento | Hi all. I'm trying to wrap my head around the proper way to bridge two calls. I am doing two outbound calls, and the goal is to bridge the calls to allow them to converse. I know I can call out to one (callerA), and then use Dial() to call CallerB, however should I be using Transfer() instead to just connect the two calls and then not have asterisk involved? If I do the latter, would callerA be paying any long-di |
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16:11.28 | orn | I am trying to get the Diversion SIP header from a SIP message, but unable because Asterisk says it only applies to SIP channels. Call comes from PSTN, goes to SIP server, SIP server adds diversion, sends call back to PSTN |
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16:11.48 | orn | even though there is no active SIP channel, there still is a SIP message with the diversion header |
16:12.14 | orn | http://pastebin.com/m7594ed71 |
16:13.29 | orn | The reason I need it is that when there is more than one diversion, asterisk uses the oldest one to use as the RDNIS, instead of the newest one. |
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16:16.12 | jerlique | Why would * stop trying to regiser a sip registration entry. Its in state "No Authentication" |
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17:58.57 | [TK]D-Fender | Linuturk: And saying "same configs" doesn't mean they are GOOD for a 650 |
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18:02.35 | Linuturk | lol |
18:02.37 | Linuturk | jeeze |
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18:02.48 | Linuturk | firmware 1.5 vs 3 in current release |
18:02.49 | Linuturk | no wonder |
18:03.50 | [TK]D-Fender | Linuturk: SMRT |
18:04.07 | [TK]D-Fender | goes to pout some more leaded gasoline into his VW Jetta TDI |
18:04.11 | [TK]D-Fender | put* |
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18:27.02 | leifmadsen | [TK]D-Fender: lol |
18:27.14 | leifmadsen | I suggest diesel in a gas engine |
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18:28.56 | empiric | guys i have Dlink dvg 3004S FXO gateway i have make 4 sip users and configure in Dlink all 4 lines are registered |
18:29.06 | empiric | how i cal on PSTN line? |
18:29.10 | empiric | any idea? |
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18:30.26 | seb- | [TK]D-Fender: is a default * install secure? if i just add sip stuff w/ a password is it secure? there isn't 100 other services open by default? |
18:30.46 | bmoraca | pour + put = pout? |
18:31.18 | seb- | [TK]D-Fender: i tested appconference (MeetMe replacement)....i didn't get an error...it was quiet...perhaps you drop into a conf room and sit there until others arrive? are you familiar w/ appconf? |
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18:33.17 | fun330 | hey i want to beable to display realtime call duration on a webpage what would be the best way to do that? |
18:33.35 | fun330 | does anyone do that currently |
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18:35.01 | [TK]D-Fender | seb-: nope |
18:35.33 | [TK]D-Fender | seb-: disable channel drivers you don't need like H323, IAX2, MGCP, Skinny, etc |
18:35.43 | seb- | [TK]D-Fender: nope to which? |
18:35.50 | seb- | [TK]D-Fender: ah |
18:35.57 | seb- | [TK]D-Fender: how do that? |
18:35.58 | [TK]D-Fender | seb-: Firewall AMI port. That just about covers it |
18:36.10 | [TK]D-Fender | seb-: and nope = never used app_conference |
18:36.15 | seb- | [TK]D-Fender: ok |
18:36.30 | seb- | [TK]D-Fender: how disable channel drivers? |
18:37.00 | [TK]D-Fender | seb-: noload => chan_mgcp.so |
18:37.19 | [TK]D-Fender | seb-: in modules.conf. Repeat for other unneeded modules |
18:37.29 | fun330 | why don't you want to load the channel drivers? |
18:37.39 | fun330 | i only use sip should i disable the otheres? |
18:37.47 | empiric | guys any idea? |
18:37.50 | seb- | fun330: i think he thinks it is a security problem |
18:37.57 | seb- | fun330: i only use sip too |
18:38.27 | seb- | fun330: [TK]D-Fender should know...he lives this stuff |
18:38.42 | fun330 | okay |
18:38.46 | fun330 | even with a firewall? |
18:39.01 | seb- | fun330: ask him |
18:40.44 | fun330 | TK D fender: even if i have a firewall should i disable those channel drivers? |
18:42.19 | *** join/#asterisk paulius (n=paulius@unaffiliated/paulius) |
18:42.33 | paulius | What's the best POTS to SIP converter out there? |
18:43.04 | Nugget | The one where you cancel your analog lines and buy a PRI. |
18:43.14 | paulius | Funny. |
18:43.22 | paulius | I actually have that, but I'll have a second POTS line. |
18:43.52 | henk | i have an account and one phone number from a sip provider. is it possible to 'expand' that number? so it's not just 12345, but also 12345-1 and 12345-2 with different extensions? |
18:44.19 | paulius | henk: Really doubt it. What you're looking for are extensions. |
18:44.26 | BlargMaN00 | paulius: what are you trying to accomplish?? just a POTS to SIP gateway?? or what? |
18:44.31 | [TK]D-Fender | fun330: If someone breaks into your system with a weak account they can then attack * locally |
18:44.44 | [TK]D-Fender | fun330: Security is a PROCESS, not a "solution" |
18:44.48 | paulius | BlargMaN00: Just a way to make and receive POTS calls on my asterisk gateway. |
18:44.50 | seb- | [TK]D-Fender: dude! modules.conf doesn't load jack by default..the only non-comment, non-noloads |
18:45.00 | seb- | [TK]D-Fender: are [modules] |
18:45.00 | seb- | autoload=yes |
18:45.00 | seb- | load => res_musiconhold.so |
18:45.06 | seb- | [TK]D-Fender: that's pretty skimpy |
18:45.07 | [TK]D-Fender | AUTOLOAD <- |
18:45.16 | seb- | [TK]D-Fender: is autoload bad? |
18:45.17 | paulius | Nugget: You know these cheap unlocked Linksys SIP gateways that they have on ebay... That isn't what I'm looking for is it? Those are just to implement SIP on local POTS? |
18:45.26 | BlargMaN00 | paulius: your best bet, is prolly gonna be to put an analog card into your * box... how many POTS lines are we talkin?? |
18:45.30 | [TK]D-Fender | seb-: so selective EXCLUDE things you know you don't want |
18:45.34 | paulius | BlargMaN00: One. |
18:45.44 | paulius | BlargMaN00: Home setup, nothing fancy. |
18:45.47 | seb- | [TK]D-Fender: not sure what that means |
18:45.50 | Nugget | yeah, those linksys pap boxes are a good solution. |
18:45.51 | jerlique | Why would * stop trying to regiser a sip registration entry. Its in state "No Authentication" |
18:45.55 | henk | paulius: what do you mean with "i'm looking for extensions"? |
18:46.10 | seb- | [TK]D-Fender: can i turn off autoload? will i then need to load => sip.so explicitly? |
18:46.12 | BlargMaN00 | paulius: gotcha... hold on a sec... lemme find a link... |
18:46.14 | [TK]D-Fender | [14:36]<[TK]D-Fender>seb-: noload => chan_mgcp.so <-- I gave you a sample. |
18:46.17 | Nugget | I would advise against a card, an external FXO is going to be a lot less hassle |
18:47.04 | seb- | [TK]D-Fender: oh i think i misunderstood |
18:47.06 | paulius | Nugget: Well actually I can't even have a card. Plan is to run this on a Mac mini |
18:47.16 | seb- | [TK]D-Fender: if you don't specify a module in modules.conf it still gets loaded |
18:47.29 | seb- | [TK]D-Fender: you need to explicitly do a noload on what you don't use |
18:47.53 | seb- | [TK]D-Fender: is there a way to invert that so you don't load anything by default? is that what autoload=no would do? |
18:48.52 | paulius | So guys, you're saying that something like this should work: http://cgi.ebay.ca/Linksys-PAP2-NA-2-ports-Sip-Voip-gateway-ATA-UNLOCKED_W0QQitemZ370197821545QQcmdZViewItemQQptZAU_Mobile_Phones?hash=item370197821545&_trksid=p3286.c0.m14&_trkparms=72%3A1215%7C66%3A2%7C65%3A12%7C39%3A1%7C240%3A1318%7C301%3A0%7C293%3A1%7C294%3A50 |
18:48.57 | paulius | (sorry for long link) |
18:48.58 | [TK]D-Fender | seb-: Based on whats in the rest of you files... load NOTHING. |
18:49.21 | [TK]D-Fender | paulius: That is for 2 PHONES, not LINES. Do not mistake this |
18:49.26 | paulius | Ah gotcha. |
18:49.32 | Nugget | no, the PAP2 is an FXS device. It pretends to be a POTS line so you can plug analog devices in to your asterisk server. |
18:49.35 | Nugget | you need an FXO device |
18:49.42 | paulius | Nugget: Gotcha. |
18:49.51 | paulius | But do they have any sort of 2 in 1 devices? |
18:49.56 | Nugget | yes |
18:50.05 | paulius | My long-term plan is to obviously have the POTS phones go through the gateway. |
18:50.08 | [TK]D-Fender | paulius: Linksys SPA-3102 = 1 FXS, 1 FXO |
18:50.12 | paulius | Any model recommendations? |
18:50.13 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
18:50.29 | Nugget | http://www.ipphone-warehouse.com/ProductDetails.asp?ProductCode=spa3102 |
18:50.36 | paulius | ah thanks |
18:50.46 | paulius | I'm in Canada though, so it'll be finding one here. |
18:51.12 | [TK]D-Fender | paulius: Describe your expected connectivity requirements |
18:51.21 | paulius | Meaning? |
18:51.35 | [TK]D-Fender | paulius: How many lines, how many analog phones, etc |
18:51.42 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
18:51.49 | paulius | One line, have 2 phones on the POTS network here. |
18:52.07 | paulius | Will have one vo-ip Cisco IP phone. |
18:52.21 | paulius | I want the asterisk to connect to the POTS and also another SIP provider. |
18:52.35 | paulius | I'll want the analog phones to use the local POTS and be able to chose on the IP Phone |
18:52.54 | [TK]D-Fender | paulius: jsut FYI you won't really be able to use call-waiting on that line, and telco VM will kind suck for not having an indicator of use |
18:53.15 | paulius | [TK]D-Fender: I don't have call waiting on my pots line anyways |
18:53.19 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
18:53.32 | [TK]D-Fender | paulius: thent he SPA should do fine |
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18:56.52 | paulius | [TK]D-Fender: Do you know any good places to buy it from? |
18:57.14 | [TK]D-Fender | paulius: http://www.canadianvoipstore.com/home.php |
18:57.22 | paulius | heh wow |
18:57.28 | paulius | And how reliable is that adapter? |
18:57.43 | [TK]D-Fender | paulius: http://www.voipdepot.ca/ |
18:57.45 | paulius | You have to realize that I'm a guy who hates cheap consumer hardware. I'm running a proper Cisco router and switch at home. |
18:57.51 | [TK]D-Fender | paulius: Its OK normally |
18:58.02 | paulius | What does normally mean? |
18:58.10 | [TK]D-Fender | paulius: And unless you already own that Cisco IP phone.... DON'T |
18:58.12 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
18:58.25 | paulius | I'm a sucker for Cisco devices. |
18:58.35 | paulius | Original plan was to use CUCM with it. |
18:58.36 | [TK]D-Fender | paulius: Means if your line is especially shitty you might have quality issues on top from EC freak-out |
18:58.56 | Nugget | the cisco phones look sexy and are sure to impress geek visitors. |
18:58.58 | Nugget | but they suck to use. |
18:59.25 | paulius | Well I'll see about that. |
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18:59.34 | paulius | Suck and Cisco are words that I've never seen in the same sentence. |
18:59.35 | Nugget | in excrutiating detail |
18:59.48 | beek | paulius: Get over your Cisco fetish and buy a Polycom if you want a great VoIP phone. |
18:59.49 | paulius | And no, I'm not talking about their consumer Linksys stuff |
18:59.54 | Nugget | I have 30 cisco phones. Take my advice -- buy a polycom. |
19:00.18 | Nugget | cisco phones are really only practical if you're running callmanager and pay for support |
19:00.24 | KyleK | hehehe "nobody ever got fired for choosing cisco" |
19:00.35 | [TK]D-Fender | paulius: their Linksys VoIP gear is BETTER <- |
19:00.38 | Nugget | trying to make them talk to asterisk is a sisyphean task |
19:00.41 | [TK]D-Fender | paulius: Polycom > All |
19:01.07 | Qwell | KyleK: really? because I know a guy... |
19:01.12 | [TK]D-Fender | lol |
19:07.02 | *** join/#asterisk sHoZaIb (n=sHoZaIb@64.55.144.23) |
19:08.31 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
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19:09.01 | *** join/#asterisk sHoZaIb (n=sHoZaIb@91.72.213.44) |
19:10.22 | cp5 | hey guys, ulaw is the best format to use (least translation cost) for calls over a US PRI, right? |
19:10.42 | *** join/#asterisk Vec (n=Vec@78-86-163-97.zone2.bethere.co.uk) |
19:10.50 | [TK]D-Fender | cp5: Yes |
19:11.07 | cp5 | [TK]D-Fender: cool, i assume it's really determined by the channel driver you're using, right? |
19:11.17 | jaytee | so I push the Cisco boulder up the hill but just as I get to the top I always lose my footing and the boulder rolls back down. At least I ain't chained to a Grandstream rock while vultures pick out my liver. |
19:11.24 | cp5 | in which case is slin or slin16 ever used? just as an intermediate? |
19:12.14 | [TK]D-Fender | cp5: Every channel has its own parameters. |
19:13.01 | cp5 | i see, cool |
19:15.47 | Katty | sneaks a hug to jaytee :> |
19:16.17 | Katty | hugs on eppigy too :> |
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19:17.33 | eppigy | :D |
19:17.40 | sHoZaIb | hello there, when I call out from my sip device through asterisk, call gets establish and also got voice response from other end but for few seconds |
19:17.42 | eppigy | herro |
19:18.06 | cp5 | [TK]D-Fender: do you see any reason to use the .sln16 audio files or have them around? are they sometimes used as an intermediary for transcoding? |
19:18.59 | eppigy | Katty: how are you? :] |
19:20.04 | paulius | Yeah you guys are gonna kill me but I'm about to buy another Cisco IP Phone |
19:20.05 | paulius | higes |
19:20.07 | paulius | *hides |
19:21.57 | beek | Whatever floats your boat paulius. You have a knack for ignoring advice from individuals who have already been there, done that. |
19:22.30 | beek | I guess that's one of the joys of being a teenager. |
19:22.30 | KyleK | does chan_sip.so not load if i screw up sip.conf? |
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19:23.23 | KyleK | beek: if hes got a stack of them one or two more doesn't make much of a difference |
19:24.17 | beek | KyleK: It was Nugget who had the stack of them. |
19:24.46 | [TK]D-Fender | cp5: Not worth thinking about |
19:24.49 | sHoZaIb | hello there, when I call out from my sip device through asterisk, call gets establish and also got voice response from other end but for few seconds after that no voice from both side and call remain connect |
19:25.06 | [TK]D-Fender | paulius: just sad... |
19:25.51 | [TK]D-Fender | KyleK: Indeed, screw up your SIP.CONF bad enough and it won't load at all |
19:26.37 | cp5 | [TK]D-Fender: alrighty, thanks |
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19:37.11 | *** mode/#asterisk [+o Cresl1n_] by ChanServ |
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19:50.47 | paulius | What was that phone brand that you were so fond of, [TK]D-Fender? |
19:51.00 | [TK]D-Fender | Polycom |
19:51.07 | paulius | Any particular model? |
19:51.30 | [TK]D-Fender | paulius: All are good, which one I would suggest depends on the needs |
19:52.56 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
19:53.24 | Anth8708 | paulius: we're using the soundpoint 330s for analog replacements and the 560s to replace nortel 2616s |
19:53.28 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
19:54.24 | *** join/#asterisk voxter (n=voxter@190.10.13.241) |
19:54.34 | [TK]D-Fender | 560? Who needs GB phones :) |
19:56.01 | mmlj4 | all things being equal, should I answer a call and fork off to an AGI process, or handle them via extensions.conf only? is or can AGI be robust enough to handle a call from beginning to end, as opposed to native *? |
19:56.23 | paulius | Hmm. |
19:56.30 | paulius | Well but doesn't Polycom make Cisco's phones? |
19:57.50 | [TK]D-Fender | paulius: Only their speakerphones. |
19:58.10 | paulius | Well but they're much more expensive than the Cisco phones though |
19:58.15 | [TK]D-Fender | paulius: And its not the HARDWARE that sucks, its Cisco fIRMWARE and licensing BS |
19:58.25 | paulius | [TK]D-Fender: Yeah I know that. |
19:58.31 | paulius | I don't agree with their corporate policies either. |
19:58.39 | paulius | But I like the fact that their hardware works. |
19:58.52 | [TK]D-Fender | paulius: Well their SIP stack sucks. Licensing sucks, etc |
19:59.17 | [TK]D-Fender | paulius: yes, well lof of hardware works. theirs jsut takes a complete 2nd rate approach to SIP |
19:59.48 | [TK]D-Fender | paulius: standards compliance isn't compatible with "vendor lock-in" |
20:01.12 | Nugget | the cisco sip firmware is neglected and poorly maintained |
20:01.29 | Nugget | each new release is just as likely to introduce new bugs as it is to clear up old ones |
20:01.32 | mmlj4 | Anth8708: you do know that asterisk can talk to those nortel phones via chan_unistim or somesuch, right? |
20:01.32 | paulius | It still works, doesn't it. |
20:01.41 | Nugget | barely |
20:01.48 | paulius | good enough! |
20:02.06 | Nugget | plus you need a support agreement that nobody will actually sell you in order to get access to the firmware |
20:02.23 | Anth8708 | mmlj4: that's what I've been told (the voip phones, like i2004), but I've tried and not been able to get it functioning |
20:02.42 | mmlj4 | it worked for me, first time out, using * 1.6 |
20:02.49 | Anth8708 | mmlj4: got registered and can receive and make calls from an i2004, but no audio |
20:03.02 | Nugget | and, to top if off, the sip firmware is effectively undocumented, so you'll have a really fun time figuring out how to configure them. |
20:03.29 | KyleK | is that so people become cisco voip phone certified? |
20:04.20 | Nugget | no, it's because cisco don't give a flying muffin about people who don't buy callmanager. |
20:04.30 | Anth8708 | [TK]D-Fender: who needs GB phones? me:). We have a bare minimum wiring infrastructure, so as I upgrade phones to voip, I'm passing lan through the phones, just splitting out the traffic over vlans |
20:04.35 | *** join/#asterisk Ast001 (n=uros@cable-89-216-155-28.dynamic.sbb.rs) |
20:05.02 | Ast001 | Hello I've found following error in my /var/log/asterisk/messages |
20:05.03 | Ast001 | Failed to open /dev/zap/transcode: No such file or directory |
20:05.17 | Nugget | everything that the community knows about configuring cisco phones for asterisk is simply by virtue of trying to deconstruct/trace from a callmanager installation |
20:05.24 | Ast001 | and there is /dev/zap on system I wonder what is that error connected too ? |
20:06.07 | KyleK | Ast001: so /dev/zap is a device? that means /dev/zap isn't a dir for a /dev/zap/transcode file :-/ |
20:06.12 | [TK]D-Fender | Anth8708: that addas 60$ per phone assuming you even wanted the 5XX series base. You saying a cable drop would cost you more? Keeping in mind you're also losing flexibility as well |
20:06.22 | Ast001 | no I meant /dev/zap/transcode |
20:06.29 | Ast001 | I have dir /dev/zap |
20:06.55 | Ast001 | lsmod | grep zaptel gave me only wctdm which I need for my digium card. Do I need zttranscode ? |
20:07.01 | *** join/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
20:07.24 | Ast001 | Should I manualy load zttranscode to avoid the problem ? |
20:08.01 | KyleK | i personally have no clue, give it a shot? |
20:08.07 | *** join/#asterisk BreezBl0k (n=BreezBl0@5acd71a1.bb.sky.com) |
20:08.28 | manxpower | zttranscode is only for use with the Digium transcoder card, IIRC |
20:08.57 | BreezBl0k | any one got a asterisk box behind pfsense? |
20:09.11 | Nugget | BreezBl0k: I do, but not behind nat if that's where you were headed. |
20:09.28 | BreezBl0k | afraid so Nugget :( |
20:09.33 | Nugget | sorry :) |
20:09.40 | *** join/#asterisk neurosys (n=vinix@sheltercorp.net) |
20:09.42 | BreezBl0k | thanks any way |
20:10.39 | mmlj4 | Ast001: if you don't HAVE a transcode device, you don't need to load the module for it |
20:10.43 | *** part/#asterisk SparFux (n=raoul@e182023188.adsl.alicedsl.de) |
20:10.54 | manxpower | forward port 5060/UDP and 10000/UDP - 20000/UDP (or whatever the range is in rtp.conf) on your NAT router. Then set localnet= and externip= in sip.conf. Presto! Done! |
20:10.56 | Ast001 | ok |
20:11.23 | Ast001 | but why is this happening ? codec_dahdi.c: Failed to open /dev/zap/transcode: No such file or directory |
20:11.42 | Ast001 | I can not record calls using digium licenced g729 codec |
20:11.58 | BreezBl0k | Nugget any guide you used to configure pfsense without NAT for asterisk ive got a couple IP's free but idealy id like to set it up behind NAT but having no luck |
20:12.01 | Ast001 | I bought 2 licences. |
20:12.27 | manxpower | Ast001: How many g729 channels does Asterisk show that you have available. |
20:12.27 | Qwell | Ast001: codec_dahdi isn't for the g729 module |
20:12.48 | Ast001 | show g729 ? |
20:12.53 | manxpower | recording will take up 1 license, |
20:12.59 | manxpower | Ast001: depends on your asterisk version. |
20:13.04 | Ast001 | 1/0 encoders/decoders of 2 licensed channels are currently in use |
20:13.25 | Ast001 | Asterisk 1.4.21.2 |
20:13.30 | Qwell | Ast001: ignore that error unless you have transcoding *hardware*. |
20:13.47 | Ast001 | ok but I can't ignore that I can't record calls I need that |
20:14.06 | Ast001 | Recording is working fine from LAN with alaw but from WAN (g729 is not) |
20:14.19 | *** join/#asterisk UQlev (n=yuriy@91.184.221.31) |
20:14.19 | mmlj4 | hey manxpower |
20:14.55 | Anth8708 | [TK]D-Fender: it's $50/phone (academic pricing:)) and that's definitely less than a new drop to the majority of my locations because of construction (cinder block, so everything is surface mount, which means changing raceway since the idiot before me used junk), not to mention new patch panels + labor (I don't have time to do it) |
20:15.28 | [TK]D-Fender | Anth8708: Question of balance and long-term value |
20:15.29 | manxpower | Ast001: I don't see any error messages |
20:15.37 | Ast001 | [May 6 17:32:14] WARNING[7682] codec_g729a.c: out of G.729 decoder licenses |
20:15.46 | Ast001 | how is that possible ? |
20:15.48 | [TK]D-Fender | Anth8708: And of course... Gigabit |
20:15.50 | mmlj4 | manxpower: done anything with AGI? |
20:16.00 | manxpower | Ast001: and right after that message show your g729 channels and see what it says |
20:16.03 | manxpower | mmlj4: yes |
20:16.26 | Anth8708 | [TK]D-Fender: I know. Having gig to the desktop just feels like future proofing for the execs. I will likely only have 10-15 of those and 250 of the 330s. |
20:16.33 | Ast001 | [May 6 17:32:14] WARNING[7682] translate.c: g729tolin did not update samples 0 |
20:16.50 | Ast001 | then repeat and etc.... milions of times |
20:17.07 | Anth8708 | quick question guys, i'm googling this and checking wikis, but |
20:17.08 | manxpower | Ast001: that doesn't look much like the number of encoders/decoders you have in use. |
20:17.35 | Anth8708 | since we have activity right now, does * 1.6 support either shared line appearances or bridged line appearances? |
20:17.48 | mmlj4 | manxpower: ok.. in your opinion, can AGI be robust enough to handle a call by itself, as opposed to straight extensions.conf routing? |
20:17.50 | Anth8708 | just a yes or no will suffice, i can find the documentation eventually |
20:17.53 | [TK]D-Fender | Anth8708: No. |
20:17.57 | manxpower | mmlj4: yes |
20:18.09 | mmlj4 | cool |
20:18.18 | Ast001 | I dunno what's happening it said 1/0 encoders/decoders of 2 licensed channels are currently in use |
20:18.31 | manxpower | I recommend FastAGI rather than AGI |
20:18.42 | Anth8708 | [TK]D-Fender: recommendations on how to handle secretary situations? They need to be able to answer the exec's line and see if they are on the phone. |
20:18.54 | mmlj4 | I'm looking at perl's Asterisk::AGI |
20:19.14 | manxpower | Anth8708: You can do that. |
20:19.21 | [TK]D-Fender | Anth8708: Seeing if they are on the phone is basic Presence and your 560 can do that. Answering calls going to them can be managed differenly |
20:19.31 | Ast001 | I also have this http://pastebin.com/m12b6ea7e |
20:19.44 | [TK]D-Fender | Anth8708: You can either setup another reg that will be rung at the same time, or use a call-pickup feature |
20:20.03 | Ast001 | you said I need 2 g729 licences for 1 concurent call which is recorded and I bought second licence |
20:20.53 | Anth8708 | [TK]D-Fender: gotcha. I'll look up presence, thanks. |
20:22.09 | Anth8708 | one more question and then i'll try to just answer. anyone successfully using softkeys on polycom phones? I've been trying to figure it out and it looks simple in the xml, but I'm stuck. I never get anything but the default keys, even though I do have preceeding='1' and my custom xml loads before ( to the left) of the standard sip.conf |
20:22.39 | manxpower | Ast001: you should contact Digium then. You bought the license, you get support for it. |
20:22.50 | Ast001 | I made a ticket |
20:22.57 | manxpower | Ast001: then why are you here? |
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20:23.08 | *** join/#asterisk juanIMP (n=Juancho@200.71.41.22) |
20:23.19 | [TK]D-Fender | Anth8708: what are you trying to do? |
20:23.20 | Ast001 | They need few days to answer and I need system to work as soon as possible |
20:23.40 | manxpower | Ast001: best of luck with that. |
20:25.03 | Anth8708 | [TK]D-Fender: The 560 doesn't show the "callers" key like the 330 does and I'd like it to. I'd also like the 560 to show the Dir on the idle screen. Also, when you have a VM, I'd like to have a key to call the vm number. |
20:25.50 | Anth8708 | [TK]D-Fender: I think I understand the xml decently enough, it's fairly plain, but all I can get are what appear to be the "default" softkeys |
20:25.53 | [TK]D-Fender | Anth8708: No need, thats what the arrow keys are for. IP 330 "Callers" key is vastly inferior <- |
20:26.08 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
20:26.18 | Anth8708 | [TK]D-Fender: wow. didn't even know that works |
20:26.21 | Anth8708 | *worked |
20:26.43 | Anth8708 | [TK]D-Fender: Thanks. |
20:26.54 | [TK]D-Fender | up= dir, Left = answered, down = missed, Right = dialed |
20:28.10 | Nugget | up, down, left, up, power = punch caller in the face |
20:28.25 | *** join/#asterisk jpcansa (n=jpbenavi@201.201.20.90) |
20:29.02 | [TK]D-Fender | FUDOKEN!!!!! |
20:29.07 | [TK]D-Fender | checkout time, later all |
20:29.31 | Nugget | cheers |
20:29.42 | jpcansa | does any body knows what the "requested special control" line means here: http://pastebin.com/ |
20:30.10 | manxpower | jpcansa: try giving us the correct URL |
20:30.22 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) |
20:30.25 | thehar | russellb: ? |
20:30.26 | jpcansa | http://pastebin.com/d4c1f50dc |
20:30.28 | jpcansa | sorry |
20:31.03 | jpcansa | 8th line |
20:31.34 | *** part/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
20:32.08 | pmhaddad-work | anyone know an easy way to raise outbound call priority? |
20:32.27 | pmhaddad-work | like i want 911 calls to be sent out at a higher priority level than anything else |
20:33.05 | pmhaddad-work | should i make the 911 stuff priorty 1 and everything else a higher number? |
20:34.51 | stope | Nugget - I like the 'punch caller in the face' part.... Nugget++ |
20:35.09 | *** join/#asterisk Led-Hed (n=Led-Hed@66-189-167-116.dhcp.trlk.ca.charter.com) |
20:36.31 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
20:40.44 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
20:41.31 | Led-Hed | Do IP Phones need to be MultiLine to transfer calls to other IP Phones, or can Asterisk handle that? |
20:42.10 | *** join/#asterisk mclugh (n=mpearson@67.214.244.42) |
20:42.39 | *** part/#asterisk mclugh (n=mpearson@67.214.244.42) |
20:43.47 | Nugget | multiline not necessary |
20:43.50 | pmhaddad-work | Led-Hed, no, asterisk handles that |
20:44.27 | Led-Hed | pmhaddad-work, ok so what benefit does a multiline IP phone have |
20:44.49 | kb3ien | whats the minimum connection time for a fax of 1 page. i'd like to consider any calls shorter than that as broken? |
20:44.56 | Led-Hed | sorry, I'm new to asterisk and VIOP in general |
20:44.59 | pmhaddad-work | you can be talking to multiple exts concurrently basically |
20:45.18 | *** join/#asterisk macros73 (n=cs_@dsl093-063-232.pit1.dsl.speakeasy.net) |
20:45.28 | pmhaddad-work | i mean ya, if you wanted to be on the phone using it on a call and take another call and transfer that to another ext then you need a multiline phone |
20:45.41 | Led-Hed | pmhaddad-work, what do I need multiline Phone to put someone on hold? |
20:46.03 | pmhaddad-work | only if you want to be on another line talking to someone else at the same time |
20:46.19 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
20:46.23 | Led-Hed | ok, I need multiLine then . Thanks |
20:46.28 | pmhaddad-work | k :) |
20:49.57 | *** join/#asterisk _Sam-- (n=sam@unaffiliated/sam--/x-573746) |
20:50.13 | _Sam-- | hey what happens if when a caller who is blacklisted this way calls? database put blacklist 9408989740 1 |
20:50.24 | _Sam-- | they just get a busy signal, or what happens? |
20:50.51 | jaytee | nothing, unless you build a lookup to the blacklist in the incoming call handling |
20:51.03 | _Sam-- | i see. |
20:51.14 | jaytee | there's some stuff on the WIKI about it |
20:51.35 | _Sam-- | do those database puts stay after restarting asterisk? |
20:51.50 | jaytee | yes, they should |
20:51.50 | Led-Hed | ok, just to clarify, If while talking on the phone, I receive another incoming call and I want to place the currnet call on hold and answer the incoming call I need a MultiLine Phone. (Sorry, just dont want to buy the wrong phones) |
20:52.03 | jaytee | Led-Hed, yep |
20:52.08 | Led-Hed | perfect. Thanks |
20:52.24 | Led-Hed | any phone Recommendations? |
20:52.32 | pmhaddad-work | Led-Hed, polycom |
20:52.57 | pmhaddad-work | 320, 330, and up should all do what you need |
20:53.12 | paulius | Cisco IP Phones, lol |
20:53.15 | pmhaddad-work | lol |
20:53.24 | _Sam-- | thanks, J. |
20:53.25 | Led-Hed | why the LOL? |
20:53.33 | Led-Hed | are Cisco Phones crap? |
20:53.35 | paulius | Because people here hate Cisco. |
20:53.38 | pmhaddad-work | because its a major PITA with asterisk |
20:53.39 | pmhaddad-work | and that |
20:53.40 | Led-Hed | ahh |
20:53.41 | paulius | Because Cisco is proprietary. |
20:53.49 | paulius | Meh, I'm about to use a Cisco phone with Asterisk. |
20:53.53 | paulius | I like Cisco's hardware. |
20:54.07 | Led-Hed | I dont like them either. And from what I saw, the Phones are built by Linksys. |
20:54.41 | paulius | Cisco bought Linksys and the Linksys branding is their consumer products. |
20:54.46 | *** join/#asterisk delta_16 (n=delta_16@84.26.9.136) |
20:54.56 | delta_16 | hey guy's |
20:55.04 | delta_16 | go a couple of questions |
20:55.05 | Qwell | hi delta_16 is. |
20:55.24 | beek | "Meh, I'm about to use a Cisco phone with Asterisk." -- let us know how that works out for you paulius |
20:55.32 | Corydon76-dig | Odd, I had a defense contractor INSIST on Cisco phones. Six months in, they replaced almost every phone with Polycom |
20:55.36 | paulius | beek: Hehe I will. |
20:55.46 | beek | Just don't slit your wrists. |
20:55.48 | delta_16 | im looking for a way to get german phone numbers ... any idea how i can do that ? |
20:56.22 | Led-Hed | when the phones have 2 Ethernet Ports, do you connect them In-Line between the wall and the PC? |
20:56.30 | beek | Led-Hed: Cisco phones are just fine -- if you're using Cisco call manager. If you're using Asterisk they're a royal PITA. Just get Polycom. |
20:56.36 | beek | Led-Hed: ys |
20:56.38 | beek | yes |
20:56.45 | Corydon76-dig | paulius: the only phones they didn't replace where executives, where looks were more important than functionality |
20:56.45 | _Sam-- | jaytee : basically all i need to do is add the lookupblacklist function to my dialplan? |
20:57.11 | jaytee | _Sam--, basically |
20:57.15 | Led-Hed | and is it reliable to run them "In-Line" or is it likely to cause networking issues? |
20:57.59 | _Sam-- | thanks, again. |
20:58.12 | delta_16 | i have a registerd adress in germany but i would like to get a phonenumber. |
20:58.20 | *** join/#asterisk xpot (n=james@70.91.210.233) |
20:58.22 | beek | Led-Hed: they give priority to the voice packets. They'll be fine, but they're 100M (unless you get the newer 1G models) |
20:58.46 | delta_16 | my phone line in germany is not activate couse i would like to you my asterisk server |
20:58.48 | Led-Hed | beek, only need 100M |
20:58.50 | Led-Hed | thanks |
20:59.13 | *** join/#asterisk telecos (n=sergio@42.166.219.87.dynamic.jazztel.es) |
20:59.41 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
20:59.44 | jaytee | Led-Hed, I have a Polycom 330 on my desk, it has 2 lines and both are setup as a single "extension" so if someone dials my number Line 1 rings, if I'm on Line 1 and someone else dials my number Line 2 rings and I can say "Please hold" to the person on Line1 and then simply press Line2 and the phone will put Line 1 on hold automatically. |
21:00.17 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
21:00.18 | Led-Hed | jaytee, perfect. Thats exactly what I'm looking for |
21:00.56 | Led-Hed | jaytee, and how would you transfer a call to another person/phone in the office. (From Reception to Sales)? |
21:00.57 | jaytee | programming the phone that way is an option, it can also be programmed to be 2 different lines. |
21:01.30 | Led-Hed | jaytee, with 2 different PHone Numbers? |
21:02.13 | jaytee | Led-Hed, when on a call there are 3 "softkeys" on the display, one of them is End Call, the next is Transfer and the last Conf. |
21:02.32 | jaytee | and yes the phone can be setup to handle TWO seperate numbers. |
21:02.50 | [TK]D-Fender | jaytee: ... kinda lazy answer :) |
21:03.14 | [TK]D-Fender | jaytee: 2 REGISTRATIONS. "Numbers" is kinda generic |
21:03.30 | Led-Hed | are there any steps I should take when configuring my router for VOIP/Asterisk? |
21:03.46 | KyleK | forward some ports for sip and rtp? |
21:03.51 | [TK]D-Fender | jaytee: What you look like dialing out direct from the phone, then through * are 2 different matters.. and inbound... well.. we know that story too :) |
21:03.57 | jaytee | [TK]D-Fender, that's much more specific, thanks! and where were you two minutes ago? :-) |
21:04.05 | *** part/#asterisk joseph__ (i=CK@93.185.225.225) |
21:04.07 | [TK]D-Fender | jaytee: In transit home :) |
21:04.13 | [TK]D-Fender | ~sipnat |
21:04.14 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
21:04.19 | [TK]D-Fender | Led-Hed: ^^^^^^^^^ |
21:04.43 | Led-Hed | [TK]D-Fender, thanks |
21:04.48 | bmoraca | Led-Hed: depends on which router you're talking about and which direction the phone connections are going and what kind of router you have. |
21:05.06 | bmoraca | Led-Hed: the question you asked is kinda like "is there anything I should watch out for when driving?" |
21:05.06 | jaytee | ok, it's quittin time for me too, be back later |
21:05.17 | Led-Hed | bmoraca, pfSense or Smoothwall |
21:05.27 | Led-Hed | (FreeBSD or Linux) |
21:06.01 | Led-Hed | bmoraca, ya I know it was kinda an open ended question |
21:06.09 | bmoraca | uhg. i'll refrain in saying my opinion of PC-based routers right now. not in the mood for that right now. |
21:06.09 | jpcansa | does any body knows what the "requested special control" line means here: http://pastebin.com/d4c1f50dc |
21:09.57 | *** join/#asterisk pmhaddad (n=pmhaddad@24-247-42-42.dhcp.mrqt.mi.charter.com) |
21:10.06 | *** join/#asterisk Meaw (n=dino@213.244.81.144) |
21:11.16 | bmoraca | does anyone have experience using Adtran TA900 series IADs with Asterisk? |
21:18.25 | *** join/#asterisk crevetor (n=crevetor@bureau.ubity.com) |
21:19.04 | mmlj4 | I want to place a call, and instead of getting hung up on my end when the remote end of the call terminates, be given a dialtone as if I'd hung up and started to place another call... how, please? |
21:21.09 | [TK]D-Fender | mmlj4: "core show application dial" <- |
21:22.35 | mmlj4 | danke |
21:23.11 | Led-Hed | I'm looking for VOIP Providers, a lot of them offer "Auto Attendant" features, will this interfere with Asterisk or can they typicaly be turned off? |
21:24.45 | [TK]D-Fender | Led-Hed: that implies they are offering you hosted-pbx plans |
21:25.09 | [TK]D-Fender | Led-Hed: And it depends what you buy and who you buy it from. You want simple termination & origination |
21:25.28 | Led-Hed | [TK]D-Fender, ok, maybe I'm looking for the wrong service. |
21:26.09 | Led-Hed | [TK]D-Fender, any recommendations for VOIP Providers? |
21:26.10 | [TK]D-Fender | Led-Hed: Evidently |
21:26.16 | [TK]D-Fender | ~itsplist-us |
21:26.17 | infobot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
21:26.28 | Led-Hed | great! thanks |
21:26.57 | pmhaddad | nufone is dead i thought |
21:27.12 | pmhaddad | bandwidth.com is good, i use broadvoice a lot too |
21:27.55 | _Sam-- | im having a problem with this line thinking every call is in the database blacklist.... any help appreciated. exten => 877294XXXX,1,GotoIf($[{DB_EXISTS(blacklist/${CALLERID(num)})]?blacklisted,s,1) |
21:38.40 | [TK]D-Fender | _Sam--: Count your braces |
21:38.48 | _Sam-- | i added one, but it still didnt fix it. |
21:38.53 | _Sam-- | right before the ] |
21:39.02 | [TK]D-Fender | _Sam--: And the only thing I see is a clearly broken line |
21:39.05 | *** join/#asterisk ber_ (i=brad@66.94.69.34) |
21:39.20 | _Sam-- | [TK]D-Fender: im not surprised. that is why im here asking for expert help! |
21:39.20 | *** part/#asterisk Cresl1n_ (n=matt@asterisk/libpri-and-libss7-expert/Cresl1n) |
21:39.21 | [TK]D-Fender | _Sam--: IO more ways than one. |
21:39.23 | [TK]D-Fender | In* |
21:39.37 | ber_ | hi, I am trying to run an IVR after the calling party has hung up for the called party to complete |
21:39.39 | [TK]D-Fender | _Sam--: Also missing "_" |
21:39.44 | ber_ | i see this command in the dial g: When the called party hangs up, exit to execute more commands in the current context. |
21:39.49 | generalhan | anyone in here use Queuemetrics ? I seem to be having issues with the real-time data, and i cant find anything about what steps i need to take to make sure that i get agent logon/logoff information -- anyone have reference material they can direct me to ? |
21:39.50 | ber_ | i want the exact opposite |
21:39.54 | _ShrikE | Anyone here using asterisk with virtual iron? |
21:40.03 | ber_ | does anyone know if that is possible without running a conference? |
21:41.11 | BreezBl0k | any SIP behind NAT gurus here? im having one way audio problems with Elastix but with that same configuration Trixbox works but i desperatly dont want to use use Trixbox you can see my sip.conf here: http://pastebin.com/d44dbca66 |
21:42.27 | jameswf | took 45min for my free kfc :( |
21:42.28 | _Sam-- | [TK]D-Fender : would this work: GotoIf($[{DB_EXISTS(blacklist/${CALLERID(num)})} = 1 ] |
21:42.47 | pmhaddad | jameswf, oooh i should get that for dinner tonight |
21:42.55 | _Sam-- | er...sorry bad formatting, again. |
21:43.26 | [TK]D-Fender | _Sam--: Doesn't need to be an expression |
21:44.07 | *** join/#asterisk BadHAL (n=nn@66.194.174.11) |
21:44.14 | _Sam-- | k. trying to figure out where the _ would go...i only put XXXX to block the number for IRC |
21:44.16 | jameswf | this is an interesting question http://trixbox.org/forums/trixbox-forums/help/play-beep-every-15-seconds-while-call-progress |
21:44.47 | KyleK | BreezBl0k: so the ports specified in rtp.conf are forwarded to the asterisk box? |
21:44.52 | jameswf | maybe use a whisper page... |
21:44.56 | pmhaddad | jameswf, i had an issue very similar a while ago, was my fxo card |
21:45.11 | pmhaddad | oh |
21:45.15 | pmhaddad | i totally misread that |
21:45.49 | BreezBl0k | <KyleK> works fine for trixbox but if i setup Elastix with same settings oneway audio |
21:46.18 | BreezBl0k | <KyleK> yeh RTP ports and SIP port forwarded |
21:49.05 | _Sam-- | [TK]D-Fender : thanks for your patience. i got it. |
21:49.05 | _Sam-- | thanks again. |
21:50.11 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
21:50.11 | *** mode/#asterisk [+o putnopvut] by ChanServ |
21:51.11 | [TK]D-Fender | jameswf: "core show application dial" <--- |
21:51.23 | KyleK | BreezBl0k: the RTP stuff works by saying, hey send data to 1.1.1.1 port 1 so check the sip packets on the side that gets no audio? |
21:52.04 | KyleK | BreezBl0k: also I turned on canreinvite=no |
21:52.19 | [TK]D-Fender | ^^^^^^^ |
21:52.25 | [TK]D-Fender | Canreinvite is the problem |
21:52.38 | jameswf | google found it Asterisk+cmd+Monitor ... putting it in trixbox another story.. |
21:52.45 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
21:53.02 | BreezBl0k | 2secs |
21:53.21 | ber_ | does anyone know how to keep a called party on the line and have them navigate an IVR once the calling party hangs up? |
21:54.24 | BreezBl0k | :( canreinvite=no has not fixed it |
21:54.49 | KyleK | check the conf for other canreinvite lines? |
21:56.10 | *** join/#asterisk DarkRift (n=dark@65.92.166.246) |
21:56.30 | [TK]D-Fender | BreezBl0k: Confirm that you are directed to the proper internal IP, and repeat EXACTLY which ports are forwarded |
21:57.25 | BreezBl0k | <[TK]D-Fender> i made it the same IP as the trixbox that was working and switched the trixbox off |
21:58.03 | BreezBl0k | <[TK]D-Fender> so the port forwards were deffinetly working before and ive checked them a few times |
21:58.22 | BreezBl0k | i think im doooomed :( |
21:58.23 | [TK]D-Fender | BreezBl0k: pastebin CLI output for a failed call with SIP DEBUG enabled |
21:58.34 | BreezBl0k | rgr |
21:59.21 | [TK]D-Fender | BreezBl0k: Also, any ITSP peers should have "nat=no" <- |
21:59.34 | [TK]D-Fender | BreezBl0k: that usually kills things as well |
22:00.10 | BreezBl0k | <[TK]D-Fender> ITSP would sipgate for example? |
22:00.49 | [TK]D-Fender | BreezBl0k: Yes |
22:03.09 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
22:03.56 | BreezBl0k | <[TK]D-Fender> http://pastebin.com/d73f20b9b debug output here |
22:04.58 | [TK]D-Fender | BreezBl0k: I do not see a CALL ATTEMPT anywhere in there |
22:05.37 | *** join/#asterisk colinm_ (n=colinm@97-124-108-180.phnx.qwest.net) |
22:08.40 | BreezBl0k | <[TK]D-Fender> sorry ill redo it http://pastebin.com/d3f72f3fb |
22:10.43 | [TK]D-Fender | BreezBl0k: New configs please |
22:11.21 | BreezBl0k | which ones? |
22:11.34 | *** join/#asterisk patrick-- (n=patrick@eos.openroot.de) |
22:11.40 | patrick-- | Hey all im having a Problem |
22:11.50 | patrick-- | even though pbx_spool is loaded call files are not processed |
22:12.02 | patrick-- | when copied into /var/spool/asterisk/outgoing |
22:12.06 | patrick-- | permissions are correct |
22:12.11 | patrick-- | call file syntax is correct |
22:12.15 | patrick-- | no output on the CLI |
22:12.53 | BreezBl0k | <[TK]D-Fender> what do you want me to do |
22:13.18 | [TK]D-Fender | BreezBl0k: SIP clearly |
22:13.53 | hardwire | SIP Fresca. |
22:14.00 | hardwire | mmm.. now I want a soda. |
22:14.11 | beek | hardwire: Do they still make that stuff? |
22:14.21 | hardwire | you bet your sweet patookie they do. |
22:14.26 | hardwire | haven't you seen the amazing fresca girls? |
22:15.09 | hardwire | what is the voip users conference? |
22:15.30 | beek | #voup-users-conference |
22:15.32 | *** join/#asterisk delta_16 (n=delta_16@84.26.9.136) |
22:15.47 | BreezBl0k | <[TK]D-Fender> http://pastebin.com/d1e73efcb |
22:16.59 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
22:17.26 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
22:19.24 | *** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net) |
22:20.11 | [TK]D-Fender | BreezBl0k: "canvite=no" <-- spelling is horribly off. "canreinvite=no" |
22:20.51 | Qwell | [TK]D-Fender: chan_sip doesn't parse the middle few chars |
22:21.15 | BreezBl0k | <[TK]D-Fender> good shout |
22:21.34 | [TK]D-Fender | Qwell: Yeah... they're silent... like the "p" in swimming |
22:21.58 | Led-Hed | do you think Voice mail and logging for 4 users would destroy a CF card in a short period of time? |
22:22.36 | *** part/#asterisk delta_16 (n=delta_16@84.26.9.136) |
22:22.38 | Qwell | Led-Hed: no, just don't use a journaled FS |
22:22.39 | Ziaeon | my meetme stopped working on one of my asterisk boxes. Randomly. I put in the pin and it says that is an invalid pin number and hangs up on me (which is a different error than if i intentionally mess up the pin, where it doesnt hang up on me). Still works on my other asterisk boxes. What the feck. |
22:22.44 | BreezBl0k | <[TK]D-Fender> reloaded sip still no audio :( |
22:23.11 | Led-Hed | Qwell, thanks |
22:23.35 | [TK]D-Fender | BreezBl0k: I do not see good configs, including your ITSP specific peer..... |
22:23.53 | Micc | Anyone have an idea why when I call a ulaw phone from my polycom on g722 it sounds all garbled? Its fine when I dialout, but when I call one of our customers and go through their queue, I'm g722 while in their IVR system, then when they answer it sounds like mr roboto. |
22:24.32 | denon | Micc: is their queue in India? |
22:24.33 | Micc | let me clarify. dialout through an ITSP. |
22:24.44 | Micc | denon, no, just down the street. |
22:24.54 | Micc | They are on our asterisk server, only 45ms away. |
22:24.56 | denon | http://instantrimshot.com |
22:25.14 | denon | erm, nevermind :) |
22:25.28 | BreezBl0k | <[TK]D-Fender> :( they are the settings other users are using with sipgate and they have been working fine on trixbox thats the bit i cant get my head round |
22:26.19 | [TK]D-Fender | BreezBl0k: Says absolutely nothing to me. |
22:28.19 | *** join/#asterisk tamiel (n=tamiel@ip-139.net-81-220-93.rev.numericable.fr) |
22:29.18 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
22:31.00 | BreezBl0k | <[TK]D-Fender> here are my ITSP settings |
22:31.05 | BreezBl0k | <[TK]D-Fender> http://pastebin.com/d567c8532 |
22:32.44 | [TK]D-Fender | BreezBl0k: .. and [general] |
22:33.12 | generalhan | i need a little agent help ... i need to define my agents in extensions.conf for the AddQueueMember, typically i would do that as AddQueueMember(Queue1|SIP/7001@internal) but this script wants it in the format - AddQueueMember(Queue1|Local/AGENT_NUMBER@internal). so how do i define my agent 1050 in [internal] to make this work properlly? any suggestions ? |
22:33.58 | BreezBl0k | <[TK]D-Fender> http://pastebin.com/d85d8b70 |
22:34.56 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
22:35.07 | [TK]D-Fender | generalhan: thats DIALPLAN. |
22:35.48 | generalhan | [TK]D-Fender: yes. i dont understand how to set this up properlly in the dialplan. |
22:35.55 | [TK]D-Fender | BreezBl0k: [GENERAL] < should be lower case, and please confirm exactly what you have forwarded |
22:36.29 | [TK]D-Fender | generalhan: "exten => 1234,1,NoOp(FFS do something)" |
22:41.49 | generalhan | [TK]D-Fender: lol. i am VERY familiar with how to work MOST parts of my dialplan just fine.... in my agents context i used to say something like exten => 1050,1,Dial(SIP/${AGENT_SIP}) which was filled by a DB entry that i create when the agent logs in. however, now that im using Queuemetrics, the logon works differently, so i cant use my DB method anymore. so i dont know how to define my agents in my 'DIALPLAN' so that i can sync it up wi |
22:42.39 | [TK]D-Fender | generalhan: Well where DO you have a mapping of an agent to a device? |
22:42.57 | *** join/#asterisk fatnasty1 (n=chatzill@cpe-72-190-76-209.tx.res.rr.com) |
22:43.44 | generalhan | exten => 20,3,AgentCallBackLogin(${AGENTCODE}||${AGENT_EXT}@extension-dial) |
22:43.58 | *** join/#asterisk fatnasty1 (n=chatzill@cpe-72-190-76-209.tx.res.rr.com) |
22:44.02 | fatnasty1 | I am using the meetme appication, when a participant enters the wrong digits for a a conference, the meetme app plays audio advising to re-enter the meeting info, then it hangs up on them without letting them enter anything. anyone seen this before? |
22:44.17 | generalhan | [TK]D-Fender: its a seperat extensions.conf file that is '#include'ed along with my actual extensions.conf file |
22:45.06 | [TK]D-Fender | generalhan: please show something useful... |
22:45.58 | generalhan | [TK]D-Fender: haha. i showed you EXACTLY what you asked for. no? |
22:46.58 | [TK]D-Fender | generalhan: Where do I see your extensions.conf? |
22:46.58 | generalhan | i am mapping AGENT_CODE to AGENT_EXT@extension-dial |
22:47.46 | generalhan | [TK]D-Fender: the mapping itself doesnt take place in extensions.conf ... anymore |
22:47.47 | [TK]D-Fender | generalhan: You are adding a LOCAL channel to your AQM. You should be adding and AGENT channel instead |
22:47.53 | [TK]D-Fender | an* |
22:47.57 | generalhan | hmm |
22:48.13 | generalhan | i am just using the dialplan as provided to me by Queuemetrics |
22:48.27 | BreezBl0k | <[TK]D-Fender> TCP/UDP 5004 - 5082 and TCP/UDP 10001 - 20000 to 172.16.0.69 |
22:48.43 | generalhan | running that command that i just posted makes 'show agents' report my agent as being logged in to the extension i had supplied |
22:49.01 | [TK]D-Fender | generalhan: And they told you to use AQM like that exactly as well as ACBLI? |
22:49.28 | [TK]D-Fender | generalhan: because your AQM has absolutely nothing to dow ith that from what you've shown |
22:49.38 | generalhan | [TK]D-Fender: they didnt tell me anything. i click 'logon' on the web interface, and that is the dialplan peice that is run |
22:50.13 | [TK]D-Fender | generalhan: well so far those 2 pieces don't add up |
22:50.26 | generalhan | let me peice this all this together in a pastebin ... |
22:52.08 | *** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) |
22:54.03 | Led-Hed | will a 500mhz CPU support 4 users? |
22:54.35 | denon | Led-Hed: that's a pretty broad question .. |
22:54.47 | Led-Hed | I know. |
22:54.49 | denon | 500mhz doing transcoding? sip to pstn? |
22:54.59 | denon | either way, you can get something much faster for $20 or out of a dumpster .. |
22:55.17 | Led-Hed | I was hoping to install Asterisk on an Alix2, which uses a 500mhz Geode CPU. |
22:55.56 | denon | those boards have been tested with 15 sip calls (max) |
22:56.13 | denon | "(Nov 07) I have successfully compiled and installed Asterisk on an Alix board (AMD Geode 500 Mhz + 256 Mb RAM) on top of Voyage linux (a Debian variant)" |
22:56.28 | denon | "I wondered how much it could be loaded, so I tested it with pbx-test: I could place up to 15 simultaneous SIP calls before it got no more responsive." |
22:57.02 | Led-Hed | denon, great! |
22:57.11 | Led-Hed | so 4 users should be no problem |
22:57.20 | denon | I'm guessing you're safe with 4 ulaw to ulaw users |
22:57.57 | Led-Hed | sorry I dont know what a ulaw user is |
22:58.06 | generalhan | [TK]D-Fender: http://pastebin.com/d709fcf3c |
22:58.11 | mmlj4 | those boards are light on RAM |
22:58.13 | Led-Hed | <--- = Total n00b |
22:58.17 | denon | erm, you probably need to do a bit more reading before you start |
22:58.30 | Led-Hed | denon, I know. |
22:58.43 | Led-Hed | I'm just trying to see if its possible. |
22:58.59 | denon | probably is -- not wise necessarily, but probably possible |
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23:00.13 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
23:01.28 | Led-Hed | denon, will IAX or SIP make a significant difference on low powered CPU's? |
23:01.52 | denon | well, I assume your phones will be sip right? |
23:02.03 | denon | and likely the itsp or whatever you're using will allow sip |
23:02.10 | Led-Hed | not sure. was looking at the Polycom 320 |
23:02.14 | denon | does it seem wise to want to get in the middle of that media? :) |
23:02.19 | denon | yes, sip |
23:02.40 | beek | Led-Hed: get the 330 -- only $20 more and it has a built-in switch. |
23:02.45 | [TK]D-Fender | generalhan: their sample if thats what this is, is broken |
23:02.50 | denon | do a little reading before you get started, these questions are a little like asking if the sky is blue .. :) |
23:02.59 | Led-Hed | beek, ok thanks |
23:03.21 | generalhan | [TK]D-Fender: how so ? |
23:04.45 | [TK]D-Fender | generalhan: Using agents and AgentCallbackLogin, you have to tel AQM to CALL and AGENT. not a LOCAL CHANNEL |
23:05.12 | generalhan | so if i just change local/ to agent/ it would fix this issue ? |
23:07.42 | [TK]D-Fender | generalhan: well it would sure help in using the agents you are loggin in |
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23:14.46 | Led-Hed | denon, I guess an Intel Atom 330 would be a better choice. |
23:14.59 | denon | atom's a nice cpu |
23:15.01 | *** join/#asterisk paulius (n=paulius@unaffiliated/paulius) |
23:15.04 | paulius | Would anyone be able to recommend me a good vo-ip unit which has an FXO and FXS port? People recommended the Linksys SPA3102 but it's also a router. I want something basic, stable, and that works well. I've had bad experience with consumer class devices in the past. |
23:15.19 | Led-Hed | the Alix have a 9 sec PDD which is a long time for a business to wait for the phone to ring |
23:16.17 | beek | paulius: The SPA3102 is works well. I've used one for over a year and have had zero issues. |
23:16.34 | beek | You can spend some more $$$ and get AudioCodes devices. |
23:16.49 | paulius | http://www.voipdepot.ca/index.php?main_page=product_info&cPath=1&products_id=116 |
23:16.52 | paulius | Voicedepo has that one. |
23:16.57 | paulius | I wonder if it's better than the linksys |
23:17.06 | KyleK | paulius: you dont have to use an spa3102 as a router |
23:17.14 | paulius | KyleK: Well duh, I know that. |
23:17.29 | paulius | But I don't like the fact that it has it. It makes the device more mediocre. |
23:17.35 | paulius | Rather than being good at one thing. |
23:17.35 | beek | I've never used the Grandstream device. Their phones aren't well thought of. |
23:18.01 | paulius | beek: So there isn't any even linksys device which just has fxs/fxo and no router? |
23:18.13 | KyleK | yea, i game in here and asked about the same HT-503 device and someone was all grandsuck |
23:18.24 | paulius | hehe |
23:18.55 | beek | I can't figure out what the big deal is. The thing works extremely well. Who gives a shit if it happens to have a feature that you don't want to use? |
23:19.25 | KyleK | paulius: the router wont crash the rest of the unit if its not routing btw |
23:19.46 | beek | It's all done with software. If you don't run the routines, they won't hurt anything. |
23:19.57 | paulius | KyleK: I know, but it does mean that Linksys used even cheaper components because they had to pay up to put in a processor which is capable of doing more than just voice. |
23:20.38 | KyleK | i think the version without router functions would use the same parts |
23:20.41 | beek | paulius: So instead of taking all of the positive recommendations you're using that kind of rhetoric? |
23:20.44 | beek | WTF? |
23:20.54 | KyleK | beek: thats kinda normal for here isn't it? |
23:20.59 | paulius | I'm probably getting it, but I know what consumer gear means. |
23:21.07 | paulius | Anyhow, it's only about $100. |
23:21.18 | paulius | I've talked with a few people and a Cisco solution would cost roughly half a grand. |
23:21.22 | paulius | For FXS and FXO. |
23:21.24 | beek | KyleK: this is slightly different. |
23:21.27 | paulius | So we'll see. |
23:21.44 | beek | paulius: It would cost $500 and not give you a damned thing more for your application. |
23:22.58 | KyleK | paulius: are you west coast canada? |
23:23.04 | paulius | KyleK: East. |
23:23.17 | paulius | I'm probably ordering from SwiftGamers |
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23:24.16 | KyleK | I actually went to a store to buy mine since im close to ncix |
23:24.23 | paulius | ah cool |
23:24.41 | paulius | oh bOOYAH |
23:24.44 | paulius | thanks for reminding |
23:24.48 | paulius | local computer store has it |
23:25.23 | KyleK | http://ncix.com/products/?sku=21919&vpn=SPA3102-NA&manufacture=CISCO 78.54cad |
23:27.42 | bmoraca | 78.54 CAD? that's like $15 in real money, right? :P |
23:29.49 | paulius | ... |
23:29.53 | paulius | that's like $70 USD |
23:29.54 | KyleK | i wish |
23:30.02 | paulius | It's $80 at local store. |
23:30.05 | bmoraca | i was joking |
23:30.10 | paulius | Plus tax. But same as $70 plus $10 shipping. |
23:30.15 | paulius | Plus instant gratification |
23:30.20 | KyleK | hehe |
23:30.32 | KyleK | bmoraca: bison bucks aren't real money btw |
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23:41.18 | KyleK | can an agi see what codec a call is? |
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23:42.27 | jameswf | American Idol upset by way of #asterisk... oh yeah! |
23:42.49 | paulius | ?? |
23:44.14 | generalhan | jameswf: lol, i have thought about doing that a million times with any of those shows |
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