| 00:00.30 | *** join/#asterisk friartuck (~pmccary@66.162.90.57) |
| 00:01.31 | *** part/#asterisk Tech_Travis (~Travis@mail.techglia.com) |
| 00:03.23 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 00:09.53 | doubletoker | tyvm for all your help |
| 00:10.30 | doubletoker | one problem I had was with the new config, if I have allowguest=no then it fails to authenicate the incoming calls |
| 00:11.11 | doubletoker | but it's working now, thanks |
| 00:13.18 | p3nguin_ | allowguest=no should just force unauthenticated peers to land their calls in the default context. |
| 00:21.29 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 00:29.03 | doubletoker | while I'm here, is there a free version of g729 that doesn't require a license? |
| 00:29.23 | p3nguin_ | There's an old open source project. |
| 00:29.33 | doubletoker | alright |
| 00:29.46 | p3nguin_ | I've never heard anything good about it, though. |
| 00:31.18 | bmoraca_work | lol |
| 00:31.59 | bmoraca_work | my 800 wholesale provider is originating in Alaska for $0.019/min...supposed to be like $0.30 |
| 00:32.06 | bmoraca_work | i'm not going to complain! |
| 00:32.36 | *** join/#asterisk Jhirley (~Jhirley@h69-21-54-248.ldlwvt.dsl.dynamic.tds.net) |
| 00:33.31 | paulc | It's all good until they realise the mistake.. (but will they come after you or just fix it going forward?) |
| 00:34.22 | Elvis__ | hey guys, just wanted to thank you for all your help today -- got everything worked out. You guys rock. |
| 00:35.30 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
| 00:40.09 | bmoraca_work | no idea, and it doesn't really matter. |
| 00:43.25 | [TK]D-Fender | p3nguin_: the ocde may be an open implementation, but the algorythm itself is still patented. |
| 00:43.39 | [TK]D-Fender | p3nguin_: thus not legal unless you've paid for the rights to use it |
| 00:46.04 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 00:48.36 | *** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 00:54.03 | norrec_zzz | it only requires a licence if your converting it from one format to another I believe |
| 00:54.12 | doubletoker | thank you everyone I'm out pz |
| 00:58.28 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 01:01.45 | p3nguin_ | norrec: Don't you mean "if you're converting" it? |
| 01:03.32 | norrec | You're really big on semantics arnt you p3nguin_? |
| 01:03.51 | p3nguin_ | norrec: It's not semantics, it's called language. |
| 01:04.47 | *** join/#asterisk Xetrov` (~xetrov@unaffiliated/xetrov/x-827361) |
| 01:05.00 | p3nguin_ | "your converting" means converting which belongs to you. Totally the wrong word there. |
| 01:06.58 | coppice | p3nguin_: yeah, its really pathetic that you actually *care* what people mean :-) |
| 01:07.51 | p3nguin_ | It would effectively become a gerund instead of a verb, but since it is followed by a pronoun, things just don't work out. Therefore, the sentence is a failure. |
| 01:08.56 | coppice | of all the oddball put downs, common on the internet, complaining that someone actually cares about the meaning of things has got to be one of the wackiest |
| 01:08.56 | p3nguin_ | coppice: No, what is pathetic is people that make lame remarks about my caring. |
| 01:09.33 | p3nguin_ | It's no put down. That's the difference between your intention and mine. |
| 01:09.55 | p3nguin_ | You are trying to insult me, but I'm trying to educate people so they can learn to use language correctly. |
| 01:10.05 | coppice | i believe it was intended as such |
| 01:11.07 | p3nguin_ | That's your opinion, which, in this particular scenerio, doesn't carry all that much weight with me. |
| 01:12.17 | p3nguin_ | If I wanted to insult him, I would have said something such as, "You ignorant piece of crap, learn English -- the word is YOU'RE!" |
| 01:12.19 | *** join/#asterisk Whitor (~Whitor@cpe-74-76-185-31.nycap.res.rr.com) |
| 01:12.23 | p3nguin_ | But I didn't do that, now did I? |
| 01:12.42 | coppice | you seem to be loosing the thread here |
| 01:14.45 | p3nguin_ | norrec: And by the way, I don't think you're an ignorant piece of crap; I just think you chose the wrong word, and so I informed you about it. |
| 01:17.32 | *** join/#asterisk nix8n82 (~AndChat@63.162.27.14) |
| 01:17.33 | *** part/#asterisk Xetrov` (~xetrov@unaffiliated/xetrov/x-827361) |
| 01:18.02 | *** join/#asterisk ltd_wk (~z@sixified.transact.net.au) |
| 01:28.31 | *** join/#asterisk dparker_ (~dparker_@c-67-180-135-191.hsd1.ca.comcast.net) |
| 01:31.42 | norrec | p3nguin_: i know, idc that much, i was just giving u a hard time lol |
| 01:32.02 | norrec | p3nguin_: if i bugged me i would have just told u to stfu last night haha |
| 01:33.27 | Katty | wanders in |
| 01:33.58 | eppigy | HELLO KATTY |
| 01:34.01 | Katty | hi dave. |
| 01:34.02 | *** join/#asterisk GGD (~GGD@ip72-196-241-104.dc.dc.cox.net) |
| 01:34.10 | eppigy | hi |
| 01:34.24 | Katty | howarchu |
| 01:35.29 | eppigy | oh I am well |
| 01:35.32 | Katty | :> |
| 01:37.25 | p3nguin_ | This is unusual. I just came across an OEM computer that has nvidia onboard dual vga. |
| 01:37.49 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 01:39.17 | Katty | wow |
| 01:39.19 | Katty | dual? |
| 01:39.22 | Katty | hot dog. |
| 01:40.53 | p3nguin_ | I've seen nvidia onboard before, but never duals. |
| 01:41.28 | carrar | Hot dogs? |
| 01:41.30 | carrar | Where |
| 01:42.51 | p3nguin_ | here: http://www.tmz.com/tmztv/?autoplay=true&mediaKey=ba6fa7cf-ca0a-4742-a1c3-5a5ae6607e5a |
| 01:43.31 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 01:44.46 | [TK]D-Fender | p3nguin_: I've had a few boards with it. 6100 / 6150 chipsets... |
| 01:44.51 | [TK]D-Fender | p3nguin_: like what I'm running now |
| 01:45.12 | p3nguin_ | VGA compatible controller: nVidia Corporation NV18 [GeForce4 MX - nForce GPU] (rev a3) |
| 01:45.31 | p3nguin_ | nForce2 chipset on the mainboard |
| 01:46.24 | p3nguin_ | I'm going to use it as a gateway, so the video card is going to go to waste. |
| 01:46.27 | [TK]D-Fender | p3nguin_: I've seen that one on laptops.. |
| 01:46.50 | [TK]D-Fender | p3nguin_: Still good to have something in that freak chance you'll use X on it |
| 01:47.32 | p3nguin_ | I could also run across another machine more well-suited for a gateway, and then I can repurpose that one back into a workstation again. |
| 01:53.02 | *** join/#asterisk mayfield (~mayfaild@64.39.4.132) |
| 02:15.22 | *** join/#asterisk rgsteele (~rgsteele@c-71-230-37-172.hsd1.pa.comcast.net) |
| 02:28.47 | Katty | hmm. dinner. hmm. hot dogs. hmm |
| 02:28.56 | *** join/#asterisk voipmonk (~shido6@dsl-67-204-40-42.acanac.net) |
| 02:30.26 | TJNII | wants a meatball sub from the local Italian restaurant, but doesn't want to pay for it. |
| 02:30.46 | Katty | i'm just too lazy to go out and get something |
| 02:30.55 | Katty | i already had my shower, and i doubt anyone would wanting me showing up in pjs |
| 02:32.41 | TJNII | I'll probably settle for burritos again, though I am concerned about scenting the lab tomorrow. |
| 02:32.49 | Katty | *hee* |
| 02:33.00 | Katty | sounds like a plan stan. |
| 02:38.24 | Katty | returns with chili |
| 02:43.00 | *** join/#asterisk Kumbang (~dsp@rusnas.paume.itb.ac.id) |
| 02:48.50 | *** join/#asterisk V4mpire (~gary@82.118.111.254) |
| 02:49.59 | jaskew | hi Katty |
| 02:51.01 | *** join/#asterisk Blackthorn (~blackthor@76-77-161-238.smyth.net) |
| 02:53.04 | Blackthorn | I am fairly new to working with diffent types of *nix and i have a working ubuntu and asterisk server. However, i'm setting up a centos and will proiblby move my asterisk over to it later. But one of the things it has by default is iptables. |
| 02:53.27 | Blackthorn | i somewhat remember that asterisk and iptables don't get along very well is that correct? and I should just disable it? |
| 02:54.42 | Katty | jaskew: hello. |
| 02:54.42 | p3nguin_ | blackthorn: No, that is not correct at all. |
| 02:54.42 | TJNII | Blackthorn: You don't have to disable it, but you will probably have to poke holes in it. |
| 02:54.49 | p3nguin_ | blackthorn: What doesn't get along is a misconfigured iptables and asterisk. |
| 02:55.33 | p3nguin_ | blackthorn: The admin controls it, so either learn how to make it work or ask for help after you get the OS installed and Asterisk installed/configured. |
| 02:55.44 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 02:55.58 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 02:57.18 | jaskew | Katty: How's the chili? |
| 02:57.28 | Katty | jaskew: gone (= |
| 02:58.03 | jaskew | heh - all the food talk here has got me looking for new & interesting things to eat. |
| 02:58.18 | p3nguin_ | blackthorn: iptables is very easy to handle, so you won't be left alone when it comes time to make changes. |
| 02:59.56 | *** join/#asterisk lordmortis (~lordmorti@203-206-117-94.dyn.iinet.net.au) |
| 03:00.12 | jaskew | I tried a little chinese place near my office the other day - turns out it's a vegan joint. Food wasn't bad though - very flavorful |
| 03:00.41 | jaskew | The people in there were kinda pasty tho. |
| 03:03.02 | mayfield | blackthorn, the initial creation of rulesets can be a task itself ;) |
| 03:03.18 | *** join/#asterisk rgsteele (~rgsteele@c-71-230-37-172.hsd1.pa.comcast.net) |
| 03:03.56 | TJNII | cheats and uses shorewall wrappers. |
| 03:07.57 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 03:10.39 | Blackthorn | ok thanks for thehelp |
| 03:11.44 | Blackthorn | well i'll add some things onto this server such as dns/web/dhcp and poke holes in it learn how to run it before setting up another server just for *. it's all on a xen virtual machine anyway |
| 03:13.45 | p3nguin_ | blackthorn: Remember to disable SElinux on CentOS. |
| 03:19.18 | *** join/#asterisk Tako-san (~Tako-san@p4022-ipad69osakakita.osaka.ocn.ne.jp) |
| 03:22.52 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 03:23.40 | Blackthorn | whats selinux? |
| 03:24.06 | Blackthorn | well guess i could go google it :P |
| 03:27.06 | Blackthorn | so asterisk is incompatiable with the securty that selinux provideS? |
| 03:33.01 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 03:37.14 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 03:37.23 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 03:40.43 | *** join/#asterisk geneticx (~geneticx@c-75-74-66-161.hsd1.fl.comcast.net) |
| 03:51.45 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 03:51.46 | jaskew | Blackthorn: Everything is incompatible with the security that SELinux provides. |
| 03:52.17 | voipmonk | heh |
| 03:52.21 | voipmonk | bed time |
| 03:53.26 | jaskew | Blackthorn: IIRC, after you install, open up /etc/selinux/config and set a line SELINUX=disabled |
| 03:53.33 | jaskew | I guess he missed that... |
| 03:55.34 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 03:57.58 | *** join/#asterisk Kumbang (~dsp@rusnas.paume.itb.ac.id) |
| 04:06.16 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 04:06.22 | p3nguin_ | He'll probably be back eventually. |
| 04:09.08 | *** join/#asterisk Neo31 (~Neo31@unaffiliated/neo31) |
| 04:11.17 | *** join/#asterisk jpsharp (~jsharp@c-68-56-227-73.hsd1.fl.comcast.net) |
| 04:14.55 | Katty | hi |
| 04:22.16 | doneir | hrm, i'm trying to get realtime meetme working. I'm using 1.6.1.16, setup odbc, setup extconfig.conf, res_odbc (all the files), when i try to execute the MeetMe(confnum) it just exists as non-zero - no more output to decypher what is wrong |
| 04:55.54 | *** join/#asterisk xpot-mobile (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 04:56.05 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 04:59.58 | doneir | blarg, example conf files were messing things up |
| 05:00.01 | doneir | getting more output now |
| 05:05.48 | norrec | p3nguin_: hey can you take a look at this for me and tell me if u see anything wrong http://pastebin.com/Zv5qi2Mx |
| 05:06.22 | norrec | its supposed to send a call notification via email, and it worked on my 1.4 asterisk server, but it doesnt seem to be getting to sendmail on 1.6 =/ |
| 05:08.06 | p3nguin_ | line 6 seems to be missing a ) |
| 05:08.40 | p3nguin_ | It's also missing a comma, but that's not a functional issue. |
| 05:09.24 | p3nguin_ | Oh, 6 is missing more than that. |
| 05:09.39 | p3nguin_ | You shouldn't use pico/nano for configs unless you use the -w option. |
| 05:10.44 | norrec | hm. yeah that apperently cut it off lol |
| 05:10.48 | norrec | hold on |
| 05:11.14 | p3nguin_ | alias nano='nano -w' |
| 05:12.28 | norrec | well pastebin would have cut the line anyways |
| 05:12.45 | p3nguin_ | eh, why? |
| 05:12.51 | norrec | cause it goes off the page |
| 05:13.01 | p3nguin_ | uh no |
| 05:13.07 | norrec | line 6? |
| 05:13.35 | norrec | oh wait |
| 05:13.36 | p3nguin_ | If you use -w in nano, nano will wrap words. If you copy wrapped text and then paste it, it will paste what you copied. |
| 05:13.43 | norrec | yeah |
| 05:13.50 | norrec | i got it sorry, looking at something else |
| 05:13.51 | doneir | ugh, found out wtf is going on, it's doing DB calls with the start time being the end time as well, so the meeting is ony available for one minute |
| 05:13.55 | doneir | lol |
| 05:13.58 | norrec | been another long day lol |
| 05:14.06 | norrec | anyways so where am i missing a )? |
| 05:14.23 | p3nguin_ | Disregard that... you were missing much more, since you cut off the line. |
| 05:14.26 | norrec | ah, end of line 6 |
| 05:14.30 | norrec | ah |
| 05:15.06 | p3nguin_ | hence the -w recommendation if you use nano/pico on configs. |
| 05:15.35 | p3nguin_ | Of forget those altogether and use vim. |
| 05:15.50 | norrec | yeah |
| 05:15.53 | norrec | brb |
| 05:16.28 | doneir | ah, basically you can't leave endtime blank for conferences, even though the schema for the db says it can be null |
| 05:16.37 | doneir | dohh |
| 05:17.47 | norrec | p3nguin_: http://pastebin.com/c17ETaz1 |
| 05:17.48 | norrec | ok |
| 05:17.52 | norrec | look at that |
| 05:19.05 | p3nguin_ | ${DIALED_PUBLIC_NUMBER} Where does this get its value? |
| 05:22.26 | norrec | well, its the inbound number |
| 05:24.05 | norrec | oh yeah |
| 05:24.19 | norrec | i set it as part of my inbound dialing rules |
| 05:24.40 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 05:28.45 | norrec | p3nguin_: http://pastebin.com/TggEK7nc |
| 05:29.15 | kaldemar | norrec: does the file get created? |
| 05:29.33 | kaldemar | you might lack path for echo in the command. |
| 05:29.42 | norrec | hmm |
| 05:29.48 | norrec | i didnt really think about that lol |
| 05:29.50 | norrec | let me check |
| 05:30.37 | p3nguin_ | I tried in on the command line, and it works fine, so it's a problem within asterisk. |
| 05:33.28 | kaldemar | remove the -e |
| 05:33.50 | p3nguin_ | What's the problem with the -e? |
| 05:34.18 | kaldemar | it gets echoed in the file instead of being handled as a parameter |
| 05:34.36 | p3nguin_ | hmm |
| 05:34.44 | kaldemar | so it screws up the from-header |
| 05:34.45 | p3nguin_ | Is that a bug in System()? |
| 05:36.00 | kaldemar | shouldn't be. parameters work with other commands. |
| 05:37.04 | p3nguin_ | If it is putting the -e into the file, but bash doesn't... sounds like a bug of some sort. |
| 05:37.06 | norrec | well, after i made the file |
| 05:37.12 | norrec | i started getting this "SAC1 sendmail[8776]: o1P5aTSc008776: from=root, size=0, class=0, nrcpts=0, msgid=<201002250536.o1P5aTSc008776@SAC1>, relay=root@localhost" |
| 05:37.16 | norrec | in the mail log |
| 05:37.19 | norrec | but still no email =/ |
| 05:37.38 | norrec | i also removed the -e |
| 05:37.42 | kaldemar | interesting. when you put path to the command, it handles -e as a parameter. |
| 05:37.45 | norrec | but that didnt seem to make a diff |
| 05:38.02 | kaldemar | what does the file look like? |
| 05:38.14 | norrec | what do you mean? |
| 05:38.17 | norrec | like the permissions? |
| 05:38.48 | kaldemar | the contents of the file. is it what it's supposed to be? |
| 05:39.27 | norrec | it looks like asterisk isnt setting the contents |
| 05:39.32 | norrec | but its just the msg |
| 05:39.59 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-xfkzqclarqtfnprs) |
| 05:40.46 | norrec | http://pastebin.com/avAag9cR |
| 05:40.51 | norrec | thats what it should look like |
| 05:41.02 | norrec | the "to:" was blank i didnt remove that |
| 05:41.15 | p3nguin_ | Don't feel bad, I can't get System(/bin/echo "Please see attachment"|/usr/bin/mutt ...) to work. |
| 05:42.47 | norrec | hmm, so u think its a problem with 1.6 then? |
| 05:42.58 | p3nguin_ | No idea, since I use 1.4. |
| 05:43.04 | norrec | oh |
| 05:43.12 | norrec | well it worked on 1.4 =/ |
| 05:43.52 | kaldemar | my 1.6.2 writes to a file just fine. |
| 05:44.10 | Nugget | writes muffins to a file |
| 05:44.48 | norrec | well i filled out the file myself and still no email |
| 05:44.58 | norrec | so it doesnt seem to be running the command properly |
| 05:45.22 | kaldemar | path to sendmail... |
| 05:52.15 | norrec | can i get asterisk to run the command manually from the cli so i can see the output? |
| 05:53.45 | p3nguin_ | You can try. |
| 05:54.33 | norrec | how would i do it from the cli? |
| 05:55.45 | p3nguin_ | I just ran it from my dialplan and I received the email. |
| 05:55.47 | norrec | well i got it to run by just calling the phone so it ran the macro and i got my email, so it means its a problem with the echo comman |
| 05:56.28 | p3nguin_ | You can run shell commands in CLI by putting a ! on the front of them. |
| 05:56.52 | p3nguin_ | Like !uname -a |
| 05:57.14 | norrec | ah |
| 05:57.15 | norrec | ty |
| 05:58.19 | Nugget | what happens if I run !asterisk -r from the console. :) |
| 05:58.50 | p3nguin_ | I used System(/bin/echo -e ...) and System(/usr/sbin/sendmail ...) in dialplan, and I got the email. |
| 05:59.57 | norrec | i used !/bin/echo "test" > /tmp/asterisk-notify.txt and it worked |
| 06:00.03 | norrec | but it wont do it in the dial plan =/ |
| 06:00.25 | norrec | cause it's not writing the information to the file |
| 06:00.31 | Nugget | just write an agi. |
| 06:00.35 | norrec | but it is calling sendmail |
| 06:00.50 | norrec | agi? |
| 06:01.18 | Nugget | http://lmgtfy.com/?q=asterisk+agi |
| 06:02.45 | p3nguin_ | norrec: http://pastebin.com/h0BbUvLt |
| 06:02.54 | kaldemar | no reason whatsoever to use an agi for that. i'd even prefer writing a shell script that makes the file and takes to and subject as an argument. |
| 06:04.17 | *** join/#asterisk aruntomar (~aruntomar@61.17.193.163) |
| 06:04.45 | p3nguin_ | Works fine for me right in the dialplan, so I wouldn't bother writing anything else. |
| 06:04.50 | aruntomar | wht's the best possible tool we could use to convert wav file recordings to ogg format |
| 06:05.21 | ChannelZ | In batch? sox? |
| 06:06.58 | norrec | p3nguin_ i agree, but it doesnt seem to be calling echo or not writing it to a file =/ |
| 06:07.08 | norrec | but if i do it in the cli with ! it works |
| 06:07.13 | norrec | in the dialplan it doesnt |
| 06:07.26 | norrec | but it does call sendmail |
| 06:07.52 | p3nguin_ | Why would it work here but not on yours? |
| 06:08.09 | kaldemar | norrec: are you calling echo with a path? is the path right? does echoing something else to another file work? |
| 06:08.42 | norrec | yes, yes, and no |
| 06:08.50 | norrec | but if i do it from the asterisk cli it does work |
| 06:09.14 | norrec | so !echo "test" >/tmp/asterisk-notify.txt works |
| 06:09.50 | aruntomar | ChannelZ: i want to write a script that will take all the wav files and convert it to ogg format |
| 06:09.59 | kaldemar | what exactly was a no? use pastebin. |
| 06:10.45 | p3nguin_ | And did you ever copy what I used exactly, changing only your email address, and try it? |
| 06:10.49 | norrec | i'm calling echo by its path (/bin/echo) and the path is right, and changing the file doesnt make a diff |
| 06:12.24 | kaldemar | aruntomar: for in in *.wav ; do sox $i ${i%.*}.ogg ; done |
| 06:12.58 | aruntomar | kaldemar: thanx |
| 06:13.05 | p3nguin_ | error |
| 06:13.31 | p3nguin_ | you used "for in in *.wav" but then used $i |
| 06:13.32 | kaldemar | yep. an extra n in there. |
| 06:13.46 | kaldemar | aruntomar: ^ |
| 06:15.44 | norrec | p3nguin_: hmm well that worked just fine |
| 06:17.45 | p3nguin_ | Okay, so what was the problem, again? ;) |
| 06:18.24 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 06:18.40 | norrec | well its not writing it to the file |
| 06:19.02 | p3nguin_ | How did my command work if it didn't write to the file? |
| 06:19.10 | norrec | yours did write it |
| 06:19.12 | norrec | idk mine doesnt |
| 06:19.40 | kaldemar | pastebin what you have after changes. |
| 06:20.18 | *** join/#asterisk eharris (eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 06:20.23 | norrec | heres the cli output when its run http://pastebin.com/rPJkvE1i |
| 06:21.21 | p3nguin_ | That doesn't jive. |
| 06:21.55 | hardwire | many things don't jive. |
| 06:21.58 | norrec | http://pastebin.com/5vbyXc4L |
| 06:22.15 | kaldemar | "*3 |
| 06:22.19 | norrec | thats what i have in the macro |
| 06:22.24 | norrec | everything gets filled in just fine |
| 06:22.29 | norrec | but its not putting it into the file |
| 06:23.03 | *** join/#asterisk lost_soul (~noymfb@cpe-67-241-68-202.twcny.res.rr.com) |
| 06:23.42 | norrec | =/ |
| 06:23.44 | aruntomar | sox is working fine, there are some more choices, is there a diff in using speex, or oggenc from ogg vorbis site, etc etc |
| 06:26.13 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 06:37.18 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 06:43.05 | *** part/#asterisk Elvis__ (~elvis@s69-163-39-5.in-addr.arpa.static.dsn1.net) |
| 06:48.17 | *** join/#asterisk s14ck (~s14ck@190.142.78.144) |
| 06:49.07 | s14ck | sup b14ck |
| 06:50.08 | vader-- | any of you guys using sipxecs? |
| 06:50.27 | s14ck | vader--, not yer |
| 06:50.35 | s14ck | vader--, what do you need? |
| 06:50.50 | vader-- | just wondering it looks pretty cool |
| 06:51.28 | s14ck | p0wn3d |
| 06:57.29 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 06:58.12 | s14ck | sup soman |
| 07:01.21 | soman | s14ck: gm |
| 07:02.40 | *** join/#asterisk thinko (~jdoe6alph@smaug.rackdragon.com) |
| 07:02.43 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 07:05.28 | s14ck | florz, I like you IP =D |
| 07:06.19 | *** join/#asterisk grEvenX (~even@apb91b.ip.ssc.net) |
| 07:10.25 | aruntomar | we've a redfone appliance and pri line and asterisk server, the link on redfone is red, in dahdi_tool it shows as alarm red , now how do i find whether it's the problem of pri or a connectivity issue between asterisk box and redfone |
| 07:11.54 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 07:16.34 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
| 07:20.41 | norrec | hey p3nguin_ or kaldemar, are either of u still around? |
| 07:20.56 | kaldemar | norrec: yes |
| 07:22.12 | norrec | kaldemar: http://pastebin.com/uahayR7q |
| 07:22.43 | norrec | there seems to be something wrong with the syntax but i dont see it =/ |
| 07:23.54 | kaldemar | <> needs to be escaped |
| 07:24.23 | norrec | escaped? |
| 07:24.43 | kaldemar | <xxxxxxxxxx> to \<xxxxxxxxxx\> |
| 07:24.48 | norrec | oh ok |
| 07:28.15 | norrec | kaldemar, ok sweet, that took care of the error, now i just have to figure out how to get that into the variable and i'll be good =D |
| 07:31.34 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 07:44.11 | *** join/#asterisk LnxBil (~LnxBil@p5099b332.dip0.t-ipconnect.de) |
| 07:44.42 | LnxBil | Hello everybody. I'd like to ask if voicemail over ldap is currently supported |
| 07:46.27 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 07:59.46 | norrec | kaldemar: hey thanks for the help, i got it working =) |
| 08:01.06 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 08:02.01 | kaldemar | norrec: great |
| 08:06.43 | *** join/#asterisk TheTosh (~Lynx@unaffiliated/thetosh) |
| 08:06.53 | TheTosh | Hey all |
| 08:12.03 | TheTosh | Ping? |
| 08:13.59 | *** join/#asterisk fiddur (~fiddur@192.121.104.121) |
| 08:15.27 | fiddur | I think I read that asterisks support for SRV lookups is limited, so that is just uses one of the servers found in the lookup... is that correct or is there any setting I can change? My sip trunk had a problem with one of the servers now, and my asterisk didn't try the other one even if both are in the srv... :( |
| 08:21.01 | *** join/#asterisk ltd (~z@pat.transact.net.au) |
| 08:21.43 | *** join/#asterisk bobnormal (~irc@87-194-32-179.bethere.co.uk) |
| 08:21.55 | *** join/#asterisk nix8n82 (~AndChat@63.162.27.14) |
| 08:23.09 | bobnormal | hi there, i upgraded to 1.6 and my extensions.ael has TRUNK=DAHDI/G2 but i get No channel type registered for 'DAHDI'. help debug? :) |
| 08:23.36 | bobnormal | dahdi module is loaded, card is A502 (2 port, only 1 port in use), using recent wanpipe release |
| 08:23.47 | bobnormal | whats the asterisk command to list known channeltypes |
| 08:24.18 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:25.13 | kaldemar | chan_dahdi.so doesn't seem to be loaded. |
| 08:25.34 | bobnormal | console command to manually load it? |
| 08:25.45 | kaldemar | module load chan_dahdi.so |
| 08:26.41 | bobnormal | chan_dahdi.c:10036 mkintf: Unable to open channel 1: No such device or address |
| 08:26.52 | bobnormal | dmesg shows successful wanpipe load though |
| 08:27.04 | bobnormal | and dahdi kernel module is loaded |
| 08:27.43 | bobnormal | shall i re-run wancfg_dahdi? |
| 08:28.09 | kaldemar | i'm unfamiliar with wanpipe. |
| 08:28.23 | bobnormal | ok, anyone else? |
| 08:28.31 | bobnormal | sangoma support is great but they're on canada time :P |
| 08:29.13 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:30.36 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:31.13 | *** part/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 08:33.37 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:33.37 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 08:38.06 | *** join/#asterisk Chodorenko (~chodorenk@194.186.188.113) |
| 08:41.03 | norrec | chan_sip.c:14774 handle_request_info: Unable to retrieve DTMF signal from INFO message from 85DD0F81EF2742D8FB233C8B5495FC00CA6EC8CD <-- does that error mean that the peer was attempting to use sip info for dtmf signalling? |
| 08:43.27 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 08:45.10 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-ggstyhddccvmvpna) |
| 08:48.24 | *** join/#asterisk sulex (~sulex@88-149-154-95.static.ngi.it) |
| 08:50.41 | *** join/#asterisk lenne_dk (~leif@0x573cc07b.odnxx13.dynamic.dsl.tele.dk) |
| 08:57.11 | *** part/#asterisk rossh (~ross@host217-40-110-153.in-addr.btopenworld.com) |
| 09:08.04 | aruntomar | our pri line is working fine, but the dahdi_tool shows the it as in alarm red |
| 09:08.18 | aruntomar | when i started the dahdi debug mode |
| 09:08.20 | aruntomar | Sending Set Asynchronous Balanced Mode Extended |
| 09:08.41 | aruntomar | i'm getting the msg, mentioned above |
| 09:11.38 | Amorsen | How do I get queue_log to store in a database in addition to the file? |
| 09:13.24 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 09:14.27 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 09:21.04 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.176.219.102) |
| 09:23.05 | *** join/#asterisk Polysics (~luca@host113-41-static.25-87-b.business.telecomitalia.it) |
| 09:23.07 | Polysics | hello |
| 09:23.24 | Polysics | how do i flush the realtime SIP cache? i loaded a record with a wrong field and now it won't register |
| 09:26.06 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 09:27.18 | *** join/#asterisk sergey (~sergey@ua0zeh.iks.ru) |
| 09:30.02 | kaldemar | Polysics: help sip prune realtime peer |
| 09:30.02 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 09:31.46 | Polysics | oh, prune, true. thanks! |
| 09:33.26 | *** join/#asterisk Tulga (~chatzilla@203.91.113.10) |
| 09:33.36 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 09:34.15 | Tulga | my operators using SIP phone and I'm using Dial(SIP/username) in extensions. but I need record all voices. how to do it? 2,Recording() not working |
| 09:34.28 | *** join/#asterisk clouddeveloper (~clouddeve@zeus.clouddeveloper.co.uk) |
| 09:35.26 | sergey | Tulga: MixMonitor ? |
| 09:35.29 | *** join/#asterisk dobry (~d@95.111.7.95) |
| 09:36.08 | Tulga | what is mixmonitor? |
| 09:36.24 | Tulga | ahh http://www.voip-info.org/wiki/view/MixMonitor |
| 09:36.29 | Tulga | ok let me try |
| 09:38.17 | Tulga | there is many options: Monitor, Record, ChanSpy,... what is better? |
| 09:39.04 | kaldemar | they are all for different tasks |
| 09:41.32 | kaldemar | if you want to record a call so that both participants get recorded to a single file, use MixMonitor. |
| 09:41.57 | Tulga | ok thanks. I used MixMonitor |
| 09:42.07 | Tulga | Begin MixMonitor Recording DAHDI/30-1 |
| 09:42.13 | Tulga | but where it saves files? |
| 09:42.25 | Tulga | I not found file in /var/lib/asterisk/sounds |
| 09:42.56 | kaldemar | core show application MixMonitor in cli will tell you |
| 09:43.27 | Tulga | thanks |
| 09:43.30 | Tulga | another problem |
| 09:43.37 | Tulga | I have 5 SIP operators |
| 09:43.46 | kaldemar | if it doesn't, look under /var/spool/asterisk/monitor/ |
| 09:44.23 | Tulga | I want first operator call goes to A operator, second goes to B, third goes to C etc |
| 09:44.28 | Tulga | is it possible? |
| 09:44.31 | kaldemar | yes |
| 09:44.53 | Tulga | what application I use for it? or dial |
| 09:45.05 | Tulga | thanks I found monitor file in /var/spool |
| 09:45.33 | kaldemar | you need to make an extension that does the decision and then use Dial in normal manner. |
| 09:46.54 | aruntomar | Sending Set Asynchronous Balanced Mode Extended |
| 09:47.16 | aruntomar | asterisk*CLI> pri show spans |
| 09:47.16 | aruntomar | PRI span 2/0: Provisioned, In Alarm, Down, Active |
| 09:47.23 | aruntomar | plz help |
| 09:52.15 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:53.44 | norrec | chan_sip.c:14774 handle_request_info: Unable to retrieve DTMF signal from INFO message from 85DD0F81EF2742D8FB233C8B5495FC00CA6EC8CD <-- does that error mean that the peer was attempting to use sip info for dtmf signalling? |
| 09:56.42 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 09:57.02 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 09:58.33 | Tulga | I set musiconhold(10) but it plays 3-4 seconds then hangup |
| 09:58.38 | Tulga | what is wrong? |
| 09:59.30 | *** join/#asterisk LnxBil (~LnxBil@p5099b332.dip0.t-ipconnect.de) |
| 10:00.29 | c0rnoTa | hello everyone |
| 10:01.25 | c0rnoTa | norrec: use sip debug for this peer to exam it. i think, it does. |
| 10:02.03 | c0rnoTa | Tulga: MusicOnHold(class) not MusicOnHold(duration) |
| 10:02.21 | Tulga | what is class |
| 10:02.28 | c0rnoTa | Tulga: try WaitMusicOnHold |
| 10:02.46 | Tulga | yes I tried. but it said it'll removed next versions |
| 10:03.19 | c0rnoTa | Tulga: class is like context. class describe foldre and filetypes used for playing MOH |
| 10:03.29 | LnxBil | Hi! Is it normal that if i do a 'sip show peers' on a 1.6.2.0-1 with ldap, I cannot see anything? |
| 10:03.34 | c0rnoTa | s/foldre/folder |
| 10:03.48 | Tulga | ok |
| 10:04.09 | Tulga | c0rnoTa: I'm making small call center. so agents is start point? |
| 10:04.12 | c0rnoTa | Tulga: WaitMusicOnHold is depricated but still not removed |
| 10:05.14 | c0rnoTa | Tulga: start point? what do you mean. |
| 10:05.39 | c0rnoTa | Tulga: MusicOnHold(class[,duration]) Try MusicOnHold(default, 10) |
| 10:06.58 | c0rnoTa | Guys, can anybody tell me something about "chan_dahdi.c: Requested indication 20 on channel" |
| 10:07.00 | c0rnoTa | ? |
| 10:10.51 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 10:14.58 | c0rnoTa | LnxBil: it's normal for realtime peers |
| 10:15.11 | Amorsen | Yay queue_log over odbc works to PostgreSQL |
| 10:15.37 | LnxBil | c0rnoTa: Ah okay. It the asterisk thinks, that the peer is offline: 'app_dial.c:1745 dial_exec_full: Unable to create channel of type 'SIP' |
| 10:17.39 | c0rnoTa | LnxBil: but after peer register on asterisk it should be shown in 'sip show peers' output |
| 10:21.09 | LnxBil | c0rnoTa: But it doesn't. I can call user2 from user1, it rings, i can open the connection, but still, nothing in peers while the connection is open, but 'sip show channels' shows the active connection |
| 10:22.50 | c0rnoTa | LnxBil: sip.conf has option for caching realtime peers. I don't know could it be wired or not, but you can try ;) |
| 10:23.44 | *** join/#asterisk basty (~basty@212.218.65.240) |
| 10:23.51 | c0rnoTa | LnxBil: rtcachefriends=yes |
| 10:24.22 | basty | Hi, anyone using asterisk 1.4.29 ? I have strange problems with the music on hold. As soon as I try to call out of the asterisk 1.4.29 to an external caller and this external caller puts me on hold...I hear my own moh instead of the external... |
| 10:24.57 | LnxBil | c0rnoTa: I already tried this. Now I cannot call anymore :-/ |
| 10:27.17 | LnxBil | c0rnoTa: cachefriends leads to the following ldap search problem: SRCH base="dc=XX,dc=XX" scope=2 deref=0 filter="(&(?cn=)(AstAccountHost=dynamic)) |
| 10:27.42 | LnxBil | Removing it from the config lead to callability |
| 10:27.56 | c0rnoTa | LnxBil: awesome! O_o i'm using this option in all my servers: rtcachefriends=yes, rtupdate=yes, rtautoclear=no, ignoreregexpire=yes |
| 10:28.19 | LnxBil | I'll try all the other parameters |
| 10:29.01 | c0rnoTa | but i'm talking about mysql realtime. it could be ldap problem - i don't know. But it's true, that sip show peers show realtime pees |
| 10:29.20 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 10:29.35 | LnxBil | c0rnoTa: With all these options the calls are working, but still no peers visible :-/ |
| 10:29.55 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 10:30.33 | *** join/#asterisk usam (~50ca5755@gateway/web/freenode/x-amplqwcgdlytnaqi) |
| 10:31.19 | usam | hello, any1 knows a alternative to Betamax voip provider? |
| 10:31.19 | c0rnoTa | LnxBil: no peers after re-registering? |
| 10:31.34 | *** join/#asterisk Lappy-Win7 (~chatzilla@1405ds1-svo.0.fullrate.dk) |
| 10:32.49 | LnxBil | c0rnoTa: You mean new login by re-registrering? |
| 10:33.05 | Lappy-Win7 | Is there a way to limit the incoming calls and if it is exceeded then the call will be redirected? |
| 10:33.26 | kaldemar | Lappy-Win7: yes, take a look at GROUP functions. |
| 10:34.05 | LnxBil | c0rnoTa: In the console (verbose 27) i see the login: 'Received SIP subscribe for peer without mailbox: user1' but peers still emtpy |
| 10:39.21 | TheTosh | SO can i use Asterisk to make a free voip system? |
| 10:41.06 | LnxBil | c0rnoTa: what options do you use on your sip users? e.g. quality? |
| 10:41.32 | TheTosh | SO can i use Asterisk to make a free voip system? |
| 10:41.50 | *** join/#asterisk Caplain (shayne@shayne.caplain.loves.boys.especially.bridget.silverelitez.org) |
| 10:41.51 | c0rnoTa | TheTosh: yes |
| 10:42.00 | TheTosh | c0rnoTa, how would i go about doing that |
| 10:42.41 | c0rnoTa | LnxBil: "peer without mailbox" - this message about VoiceMail box |
| 10:43.40 | c0rnoTa | LnxBil: "quality"? r u mean "qualify"? there is no quality option |
| 10:43.54 | c0rnoTa | TheTosh: read the book |
| 10:43.57 | c0rnoTa | ~thebook |
| 10:43.58 | infobot | rumour has it, thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 10:45.21 | TheTosh | ~tehbook |
| 10:46.16 | dwarken | kaldemar: where do i put the exten setgroup command?? in wich conf file? |
| 10:46.47 | dwarken | is very confused about the group functions |
| 10:49.08 | *** join/#asterisk pokoko222 (~pokoko222@62.162.180.202) |
| 10:49.12 | *** part/#asterisk pokoko222 (~pokoko222@62.162.180.202) |
| 10:50.18 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 10:51.05 | EmleyMoor | Is there any way to relay MWI (from my own * setup) through a pbxes.org account? |
| 10:51.40 | LnxBil | c0rnoTa: Yes, If qualify is set, I get the ldap (?cn=) error in ldap. I deactivated the setting. But still, no peers lists. |
| 10:52.38 | LnxBil | c0rnoTa: The mailbox setting is always 'without mailbox', but a 'voicemail show users for default' shows the mailbox |
| 10:54.59 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 10:54.59 | c0rnoTa | LnxBil: qualify option is for checking online state peer. |
| 10:55.05 | c0rnoTa | LnxBil: why it reproduce ldap error - i don't know :) |
| 10:55.38 | c0rnoTa | LnxBil: about mailbox. it could be default mailboxes from config |
| 10:56.10 | LnxBil | It seems there are a lot of issues with LDAP right now. The documentation is very short on this |
| 10:56.34 | LnxBil | only very easy examples and everythin I found on the net was almost homebrew stuff for older asterisk versions |
| 10:56.52 | *** join/#asterisk Diogo (~info@a89-152-10-8.cpe.netcabo.pt) |
| 10:57.17 | c0rnoTa | LnxBil: is there reason to use ldap for store sip peers? |
| 10:58.05 | Diogo | hi people i saw on ebay this model Authentic X100P SE FXO PCI for Digium Asterisk VoIP PBX |
| 10:58.18 | LnxBil | c0rnoTa: All our accounts are stored in LDAP. It would be very cool to use this. We also invenstigate a export from LDAP to asterisk |
| 10:58.46 | Diogo | i need to anser my call using asterisk from analogic line |
| 10:58.48 | LnxBil | c0rnoTa: Everything works perfect if i only use text files, but it is not very dynamic |
| 10:58.51 | Diogo | this product can do this? |
| 10:59.36 | tzafrir | Diogo, yes. |
| 10:59.57 | tzafrir | Though don't get too excited about the "Authentic" part :-) |
| 11:00.17 | Diogo | i know :) |
| 11:00.22 | Diogo | this product is a clone |
| 11:00.25 | Diogo | right? |
| 11:00.42 | tzafrir | Right. A modem. But if it works, who cares? |
| 11:01.18 | EmleyMoor | will need to replace his TDM card fairly soon - but needs to save for it |
| 11:01.32 | Diogo | thanks for your help tzafrir |
| 11:01.55 | Diogo | and asterisk detect automaticaly this PCI right? |
| 11:02.03 | Diogo | or i need a special drivers for this? |
| 11:02.20 | tzafrir | You need DAHDI |
| 11:02.22 | EmleyMoor | The zaptel/DAHDI modules drive it |
| 11:03.18 | Diogo | ok :D |
| 11:03.20 | EmleyMoor | is looking to get an AEX400E |
| 11:04.10 | *** join/#asterisk Polysics (~luca@host113-41-static.25-87-b.business.telecomitalia.it) |
| 11:04.12 | Polysics | hello |
| 11:04.25 | Polysics | how do i set a "custom" ringing sound? |
| 11:04.43 | EmleyMoor | Polysics: On what kind of phone? |
| 11:04.44 | Polysics | like "you are waiting to be connected to the desired person?" |
| 11:04.51 | Polysics | Zoiper Web softphones |
| 11:04.53 | EmleyMoor | Oh, right... |
| 11:05.09 | Polysics | don't even know if i phrased the question properly :-) |
| 11:05.12 | EmleyMoor | You mean a ringing tone |
| 11:05.18 | Polysics | i suppose it should be a ringing tone |
| 11:05.52 | EmleyMoor | Presumably you have queuing? |
| 11:07.03 | Polysics | not yet |
| 11:07.10 | Polysics | i just have a Dial that doesn't ring |
| 11:07.28 | Polysics | single user queues and proper queuing are next on the table :-) |
| 11:07.47 | EmleyMoor | What does the Dial connect to? |
| 11:08.09 | c0rnoTa | Polysics: you can use musiconhold instead of ring tones. |
| 11:08.18 | c0rnoTa | Polysics: on Dial app |
| 11:08.57 | Polysics | EmleyMoor, to a SIP/ user |
| 11:09.07 | c0rnoTa | Polysics: for queues it's better to use periodic message with this sound "you are waiting to be connected to the desired person" |
| 11:12.20 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 11:12.55 | Polysics | i just need to figure out how to pass Dial options to Adhearsion :-) |
| 11:15.59 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 11:17.58 | Polysics | i don't get why the CALLING party does not hear anything |
| 11:18.02 | dwarken | going nuts!! |
| 11:18.15 | Polysics | until the call is accepted, that is |
| 11:20.30 | *** join/#asterisk cgc (~chatzilla@truff.demon.co.uk) |
| 11:20.35 | basty | could somebody please give me a hand with my problem ? I am using asterisk 1.4.29 (update from 1.2.30) if "A" calls an external number (no asterisk pbx), "B" answers the call and puts "A" on hold. While "A" is on hold, his own asterisk moh is playing instead of the moh of the external pbx. |
| 11:20.38 | cgc | hi everyone |
| 11:20.41 | dwarken | Where to put the group function commands? |
| 11:28.31 | cgc | I am trying to get an openvox b100p card to work in asterisk 1.4.26.1 using misdn, have i configured something wrong? |
| 11:28.34 | cgc | http://pastebin.ca/1809446 |
| 11:30.00 | c0rnoTa | basty: what type of connection? IAX? SIP? PRI? |
| 11:30.23 | basty | c0rnoTa: sip with dahdi |
| 11:30.45 | kaldemar | dwarken: extensions.conf. the functions are used to implement dialplan logic. |
| 11:31.53 | c0rnoTa | basty: A connected to asterisk over SIP and dial external number over DAHDI. So what type of dahdi channel? PRI? analog? |
| 11:32.25 | basty | c0rnoTa: yup - its pri over a digium te121 |
| 11:32.32 | *** join/#asterisk michael-i (~michael-i@141.41.40.185) |
| 11:34.15 | c0rnoTa | basty: your telco throw you information frame, that you are now on hold, that's why A listerning your MOH. |
| 11:34.36 | c0rnoTa | basty: there must be an option in chan_dahdi.conf to ignore this frame |
| 11:34.49 | c0rnoTa | basty: try to dig in this way |
| 11:35.05 | basty | c0rnoTa: mhh..okay cool - thanks... |
| 11:35.08 | kaldemar | dwarken: it works like a counter. put Set(GROUP()=yourgroupname) for each incoming call so that you can count them and before that a GotoIf($[${GROUP_COUNT(yourgroupname)} = insert_limit_number_here]?true:false) that defines what to do with the call. |
| 11:35.19 | c0rnoTa | basty: gl |
| 11:37.35 | basty | c0rnoTa: you think it could be something with the isdn timer ? (pritimer) ? |
| 11:39.10 | norrec | chan_sip.c:14774 handle_request_info: Unable to retrieve DTMF signal from INFO message from 85DD0F81EF2742D8FB233C8B5495FC00CA6EC8CD <-- does that error mean that the peer was attempting to use sip info for dtmf signalling? |
| 11:39.32 | kaldemar | basty: discardremoteholdretrieval=yes in chan_dahdi.conf |
| 11:39.54 | basty | kaldemar: i didnt even find that one in the dahdi_chan.conf example... |
| 11:40.19 | kaldemar | basty: ah, you're using 1.4.29. |
| 11:40.21 | *** join/#asterisk candyban (~candyban@ip-83-134-89-32.dsl.scarlet.be) |
| 11:40.25 | candyban | Hi guys |
| 11:40.30 | basty | kaldemar: yep with dahdi 2.2.1 |
| 11:40.37 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 11:41.56 | kaldemar | basty: try mohinterpret=passthrough |
| 11:42.15 | c0rnoTa | basty: yes mohinterpret=passthrough |
| 11:42.17 | candyban | I'm trying to upgrade my 7940G phone from P00307020300 to P0S3-08-8-00 ... the phone is trying to get SEP<mac>.cnf.xml over and over, but it is not upgrading |
| 11:42.23 | c0rnoTa | kaldemar: right ж) |
| 11:42.25 | c0rnoTa | ;) |
| 11:42.40 | kaldemar | c0rnoTa: exactly :) |
| 11:42.44 | candyban | anyone has a good idea on how to proceed/debug/... |
| 11:42.45 | c0rnoTa | If this option is set to "passthrough", then the hold message will always be passed through as signalling instead of generating hold music locally. |
| 11:43.05 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 11:43.19 | kaldemar | discardremoteholdretrieval is not implemented in 1.4. |
| 11:43.52 | basty | kaldemar: i have tried that allready..didnt work :-/ same problem - for like 20ms i hear the actually moh...then it changes back to the local asterisk moh |
| 11:44.19 | basty | kaldemar: reload chan_dahdi should set the "mohinterpret=passthrough" - right ? |
| 11:45.35 | kaldemar | all parameters are not activated upon a reload. i'd try a restart before giving that up. |
| 11:46.23 | basty | when I call my mobile (from SIP/100) and do the hold function with my mobile I should hear the mobile hold music on SIP/100 - but as soon as I do that I see in the asterisk cli "-- Started music on hold on SIP/100-0000828d" - so I hear this dang local moh :-/ |
| 11:46.32 | basty | but okay..I will try to restart the server tonight |
| 11:47.16 | basty | is there anything else I should change in my chan_dahdi.conf (http://pastebin.com/YT8vXeD0) ? |
| 11:48.50 | norrec | candyban: if its just one phone, do a manual upgrade from the web interface |
| 11:50.00 | candyban | norrec, I think I found the issue ... typo in the MAC address (86 instead of B6) ... unfortunately it was the first phone (out of 15) I tried to uprade |
| 11:50.30 | norrec | candyban: well that would do it |
| 11:50.31 | candyban | norrec, it's doing something more now loading application ... crosses fingers |
| 11:51.32 | *** join/#asterisk Chinorro (~Chino@202.219.27.77.dynamic.mundo-r.com) |
| 11:54.07 | basty | mhh - https://issues.asterisk.org/view.php?id=13454 <- seems to be already in libpri and asterisk 1.4 ? |
| 11:55.55 | *** join/#asterisk jblack (~jblack@71.181.248.16) |
| 11:56.08 | *** join/#asterisk darkskiez_ (~dz@62-50-207-164.client.stsn.net) |
| 11:59.12 | *** join/#asterisk HenrikJott (~info@d83-183-134-141.cust.tele2.se) |
| 12:03.26 | LnxBil | c0rnoTa: Is your dialplan from mysql shown if you quere 'dialplan show'? Mine is not shown. I start to think that nothing is shown there anyway :-/ |
| 12:06.22 | *** join/#asterisk guax (~guax@unaffiliated/guaxinim) |
| 12:06.28 | guax | mnicholson, ping? |
| 12:08.55 | c0rnoTa | LnxBil: i don't use realtime dialplan. only sip peers with caching. And after sip reload, they are now shown, but some time later (REGISTER pockets, or dial to this peers) they are exist in output. |
| 12:09.10 | c0rnoTa | s/now/not/ |
| 12:10.40 | LnxBil | c0rnoTa: Okay, and this register is a regular login, e.g. with ekiga |
| 12:11.36 | c0rnoTa | LnxBil: yes. Register expire time could define re registratation time |
| 12:11.54 | c0rnoTa | s/registratation/registration/ |
| 12:15.16 | *** join/#asterisk McLazarus (~McLazarus@dogpile.mcallister.ws) |
| 12:15.33 | *** join/#asterisk `Sauron (sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 12:15.36 | *** join/#asterisk cgc (~chatzilla@truff.demon.co.uk) |
| 12:16.15 | cgc | when making an outbound call over an isdn bri line, i get the following error ' Jitterbuffer Underrun. Got 96 of expected 128', any idea what this means? |
| 12:18.40 | *** join/#asterisk aandrade (~aandrade@189.58.128.179) |
| 12:19.37 | *** join/#asterisk TommyBotten (tommy@145.185-205-91.dhcp.blixbone.net) |
| 12:21.21 | *** join/#asterisk darkskiez_ (~dz@62-50-207-164.client.stsn.net) |
| 12:23.30 | LnxBil | c0rnoTa: For the record, the LDAP-Entry has to have the AstAccountType=friend tag, then it'll be shown |
| 12:25.53 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 12:26.06 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 12:29.21 | *** join/#asterisk ickmund (~magnus@ada-bcn-fw01.adamoeurope.com) |
| 12:32.44 | c0rnoTa | LnxBil: wow. you have solved your problem with setting option AstAccountType=friend? |
| 12:34.20 | c0rnoTa | s/you have/have you/ |
| 12:38.14 | *** join/#asterisk cgc (~chatzilla@truff.demon.co.uk) |
| 12:39.30 | *** join/#asterisk hipitihop (~denis@203.132.229.187) |
| 12:40.10 | cgc | does anyone know what this means? Jitterbuffer Underrun. Got 96 of expected 128 |
| 12:41.34 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 12:42.08 | *** join/#asterisk CloudDeveloper (~CloudDeve@zeus.clouddeveloper.co.uk) |
| 12:44.44 | c0rnoTa | cgc: i think that it means that you have rtp delay less then you have set in configuration file |
| 12:44.55 | *** join/#asterisk CloudDeveloper (~CloudDeve@zeus.clouddeveloper.co.uk) |
| 12:44.57 | guax | someone have mnicholson mail? |
| 12:45.52 | tzafrir | guax, http://svn.digium.com/svn/repotools/authors |
| 12:46.35 | guax | tzafrir, thankyou |
| 12:46.49 | c0rnoTa | is there some one with advanced PRI or DECT knowledge? :) |
| 12:47.31 | c0rnoTa | i have problems with calls to DECT station made over PRI from asterisk |
| 12:47.45 | *** part/#asterisk slashtom (~tom@k-rad.co.uk) |
| 12:50.10 | hipitihop | given that I only have inbound voip, and hence one DID, is it possible for outside users to dial an internal extension directly ? |
| 12:52.55 | cgc | c0rnoTa: in the misdn.conf file? |
| 12:55.57 | c0rnoTa | cgc: if you'v got this message from "...misdn.so" then misdn.conf |
| 12:59.22 | *** join/#asterisk darkskiez_ (~dz@62-50-207-164.client.stsn.net) |
| 13:00.20 | cgc | c0rnoTa: this is what happens when i make an outbound call: http://pastebin.ca/1809543 |
| 13:06.00 | tuxx- | When i try to record a voicemail to some user, i hear the 'beep' sound, and then the voicemail application exits somehow. I thought it could be some permission error on the directory the voicemail application is trying to write to, so i chmod -R 777'd that, but the voicemail app still exits.. |
| 13:06.05 | tuxx- | ANyone have a clue how this is possible? |
| 13:06.10 | V4mpire | anyone know of cheap sip providers that offer free multiple landline numbers for canada/UK |
| 13:06.25 | HenrikJott | i have a problem that i don´t really understand... my itsp is swedish tele2, and the use 2 servers for redundancy sip-corporate1.tele2.se and sip-corporate2.tele2.se and i have no problem registering to any one of them, but the have some sort of dns-thing or router at sip-corporate.tele2.se so thats what i use to register, with srv_lookup=yes in sip.conf. Yesterday one of thier servers (sip-corporate2.tele2.se) went down but we still (via |
| 13:09.00 | LnxBil | c0rnoTa: Yes, I tried to match it to my text setup |
| 13:09.27 | LnxBil | c0rnoTa: Now, I see myself, but cannot call anymore. I get "status is 'CHANUNAVAIL'" if i try to call |
| 13:20.43 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 13:21.25 | ariel_ | Morning |
| 13:22.54 | c0rnoTa | morning for you, day for us. not so good as it could be. |
| 13:23.56 | c0rnoTa | but we still wish you a good morning |
| 13:24.01 | c0rnoTa | :-D |
| 13:24.03 | ariel_ | not so good, wow and I was wondering why it was so cold this morning. I would have hope winter would not reach us down here any more. |
| 13:24.47 | coppice | we traded in our winter for a good deal on some warmth on Tuesday |
| 13:25.33 | ariel_ | we did last week but it's back to cold. argh yesterday it was 80 and today we might not reach 60 |
| 13:25.35 | c0rnoTa | cold? heh, we have +1*C outside |
| 13:26.05 | coppice | its 24 here |
| 13:27.19 | c0rnoTa | 32F |
| 13:27.44 | LnxBil | Is there any way to set the min lagging time? My status flaps a couple of times per minute from OK to LAGGED. My ping is around 30ms |
| 13:28.05 | *** join/#asterisk michael-i_ (~michael-i@141.41.40.185) |
| 13:28.29 | c0rnoTa | LnxBil: try to increase qualify option |
| 13:28.34 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:29.20 | c0rnoTa | wow. Morning, [TK]D-Fender |
| 13:33.05 | *** join/#asterisk jpvoip (~jpvoip@201-34-141-34.fnsce704.e.brasiltelecom.net.br) |
| 13:34.04 | [TK]D-Fender | c0rnoTa: y0 |
| 13:34.47 | jpvoip | hello guys |
| 13:36.11 | c0rnoTa | [TK]D-Fender: have a question for you. About debug message. What does it mean 'chan_dahdi.c: Requested indication 20 on channel..' indication 20 - what is it? |
| 13:36.21 | c0rnoTa | jpvoip: hello! |
| 13:38.29 | LnxBil | c0rnoTa: thx |
| 13:38.40 | *** join/#asterisk Skeeter- (~Skeeter@190-141.cgocable.ca) |
| 13:39.39 | [TK]D-Fender | c0rnoTa: No idea |
| 13:47.16 | *** join/#asterisk jpvoip (~jpvoip@201-34-141-34.fnsce704.e.brasiltelecom.net.br) |
| 13:50.32 | jpvoip | Im starting the final project of my course and i want to do something whith asterisk. Im looking for ideas for what to do. Others projects of my course was about using dundi to do load balance, and about conversation security on asterisk.. Someone have ideas of projects ? |
| 13:51.21 | jpvoip | Things such developing solutions to make things easy on Asterisk, like billing.. or testing and doing bechmarks whith protocols.. |
| 13:52.18 | *** join/#asterisk etfonhomey (~etfonhome@74-143-192-74.static.insightbb.com) |
| 13:53.30 | tuxx- | When i try to record a voicemail to some user, i hear the 'beep' sound, and then the voicemail application exits somehow. I thought it could be some permission error on the directory the voicemail application is trying to write to, so i chmod -R 777'd that, but the voicemail app still exits.. Anyone have a clue about this? |
| 13:53.54 | ManxPower-work | tuxx-: pastebin the cli out put and your voicemail.conf |
| 13:54.05 | *** join/#asterisk jaytee (~jforde@unaffiliated/jaytee) |
| 13:54.16 | ManxPower-work | chmoding 777 is very, very stupid. |
| 13:54.22 | tzafrir | tuxx-, if you had to do chmod 777, you're doing something wrong |
| 13:54.39 | tuxx- | tzafrir: its just for testing... |
| 13:54.51 | tzafrir | Still. It hides the real problem |
| 13:54.54 | ManxPower-work | waits for the pastebin |
| 13:54.58 | tuxx- | http://pastebin.org/97153 <- pastebin cli |
| 13:55.50 | tuxx- | http://pastebin.org/97158 <- voicemail.conf |
| 13:56.37 | ManxPower-work | try using a more standard voicemail.conf |
| 13:56.39 | [TK]D-Fender | tuxx-: What codec is the call in? |
| 13:57.02 | tuxx- | [TK]D-Fender: ulaw |
| 13:57.08 | [TK]D-Fender | tuxx-: And you have skipped parameters in your VM config... |
| 13:57.43 | tuxx- | parameters at the voicemail boxes, or at the general section? |
| 13:59.00 | ManxPower-work | tuxx-: I'd start with the mailboxes |
| 13:59.07 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:59.37 | ManxPower-work | For one thing SET A PASSWORD. |
| 13:59.45 | tuxx- | :) |
| 13:59.50 | tzafrir | tuxx-, I don't see the actual error there. Could you please point to something specific that is wrong? |
| 14:00.04 | ManxPower-work | MAILBOX => PASSWORD,NAME,,, |
| 14:00.05 | tzafrir | You expected X to happen but Y happened |
| 14:00.25 | tuxx- | well, i call the voicemail, i want to record a voicemail for some user |
| 14:00.30 | ManxPower-work | tuxx-: your voicemail.conf is so screwed up I doubt anybody knows what will happen. Are you going to fix your config file or not? |
| 14:00.31 | tuxx- | i hear the beep tone, so i can record my message |
| 14:00.42 | tuxx- | then it just exits |
| 14:00.45 | tuxx- | ManxPower-work: on it now :P |
| 14:01.01 | ManxPower-work | tuxx-: look at the sample config file. |
| 14:01.32 | tuxx- | hm, got a meeting now. Thanks for all your advices :) |
| 14:01.39 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 14:01.47 | ManxPower-work | thanks for wasting our time. |
| 14:01.55 | tuxx- | no proble |
| 14:04.22 | *** join/#asterisk Tim_Toady (~moi@193.92.197.188.dsl.dyn.forthnet.gr) |
| 14:08.02 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 14:14.11 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:14.17 | hipitihop | What does this mean on an incoming call "WARNING[10858]: chan_sip.c:12230 check_auth: username mismatch, have <gotalk>, digest has <0944xxxx> |
| 14:14.52 | *** join/#asterisk theBruno (~ChrisBrun@32.129.3.43) |
| 14:15.36 | Katty | hi |
| 14:15.42 | eppigy | OHN HAY |
| 14:16.56 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 14:18.46 | Katty | pamples eppigy |
| 14:20.01 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:20.01 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:20.19 | *** part/#asterisk theBruno (~ChrisBrun@32.129.3.43) |
| 14:25.02 | tuxx- | figured out what the voicemail problem was, the format is now wav only, and now it can record the voicemail message correctly |
| 14:25.06 | tuxx- | yay |
| 14:25.34 | Katty | yay |
| 14:25.40 | Katty | also |
| 14:25.41 | Katty | ATTENTION |
| 14:25.44 | Katty | IT IS THURSDAY |
| 14:25.47 | Katty | that means TOMORROW |
| 14:25.49 | Katty | IS FRIDAY |
| 14:26.04 | Katty | that is all. |
| 14:26.37 | tuxx- | :) |
| 14:27.12 | Katty | petco is so overpriced. |
| 14:27.33 | Katty | they carry a marshall playpen for small animals, 8 panels (9ft^2) for 70 bucks |
| 14:27.48 | *** join/#asterisk xLP (~test@mail-out.lpcorp.fr) |
| 14:27.56 | Katty | i can buy the same playpen, except it has 11 panels, at ferretdepot.com for $56 |
| 14:29.47 | c0rnoTa | yeah, tomorrow is friday! |
| 14:30.14 | lordvadr | Finally. |
| 14:30.57 | lordvadr | And what's this about petco |
| 14:31.06 | lordvadr | am I in the wrong #asterisk? |
| 14:31.24 | Katty | nope. |
| 14:31.27 | *** join/#asterisk voipmonk (~shido6@dsl-67-204-40-42.acanac.net) |
| 14:31.40 | Katty | hi mister monk |
| 14:32.06 | ManxPower-work | lordvadr: the Squirrel Girl likes to talk about her critters. |
| 14:32.14 | Katty | and recipes. |
| 14:32.21 | lordvadr | ah |
| 14:32.21 | Katty | bmoraca_work: SPEAKING OF RECIPES |
| 14:32.33 | lordvadr | ZOMG!!!!11eleven |
| 14:33.05 | Katty | lordvadr: http://ustre.am/bEBU <- not squirrels. |
| 14:33.45 | lordvadr | I dated a girl for a while that had ferrets. a.d.d. with legs... |
| 14:34.01 | Katty | it takes a special type of person to have ferrets. |
| 14:34.12 | Katty | and yes, they very much are ADD on legs |
| 14:34.27 | Katty | then settle down a good bit after 2 years. |
| 14:35.41 | lordvadr | one of them stole my wallet, out of my pants. She had a hiding place in the box-spring. After cancelling all my credit cards, I reached up in there and foudn 4 packages of ramen, 3 of those little foam things you put between your toes to paint your toenails, and my chewed up coach wallet. |
| 14:35.51 | *** join/#asterisk [8none1] (~8none1]@c-68-52-180-102.hsd1.tn.comcast.net) |
| 14:36.32 | Katty | mhmm |
| 14:36.54 | Katty | you have to ferret proof your home |
| 14:37.48 | Katty | lordvadr: this is one of the reasons i'm looking into buying a playpen |
| 14:38.19 | Katty | lordvadr: the other reason is the dog. |
| 14:38.44 | Katty | lordvadr: not that he's ever mistreated a fuzzy. and he was raised as a pup with them, but i need to be able to do laundry and other things while they're out |
| 14:38.49 | Katty | lordvadr: without constant supervision |
| 14:38.50 | xLP | may I bother with a non ferret related (but asterisk related) question ? ;) |
| 14:38.56 | Katty | infobot: ask |
| 14:38.57 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:39.51 | Katty | lordvadr: http://ustre.am/8H5d <- this is the other broadcast |
| 14:41.08 | *** join/#asterisk mmattice (mmattice@unaffiliated/mmattice) |
| 14:41.10 | xLP | I'm encoounterting troubles with Asterisk, no clue why, I can only describe symptoms... Randomly, when you try placing a call, it fails, and some phones will fail registering. I sniffed network, while watching asterisk console and monitoring box load... Load is always around 0.04, I see packets incoming and being replied normally, and when the problem starts, I see incoming packets, but no answer, and nothing in the console, then, suddenly, after about 30 s |
| 14:41.10 | Katty | hrmm looks like the feeder is mostly empty |
| 14:41.22 | *** part/#asterisk mmattice (mmattice@unaffiliated/mmattice) |
| 14:41.37 | *** join/#asterisk mmattice (mmattice@unaffiliated/mmattice) |
| 14:44.04 | lordvadr | So I'm having a really annoying DTMF problem. I dail a call (Cisco SIP), call goes through, but then if I get into an IVR prompt, digits I press don't go through. I'd imagine it has something to do with listening for the codes in features.conf but I don't know what I did to get it not sending the dtmf. |
| 14:44.31 | Katty | i had that problem before |
| 14:44.37 | Katty | didn't have the dtmf defined |
| 14:44.39 | lordvadr | Cisco SIP -> Asterisk -> iax -> pstn |
| 14:45.05 | ManxPower-work | lordvadr: I think you are missing something there. |
| 14:45.15 | ManxPower-work | Like maybe a provider? |
| 14:45.28 | lordvadr | sorry, iax -> provider -> pstn |
| 14:45.34 | ManxPower-work | Many providers have issues translating IAX2 DTMF into SIP DTMF. I recommend you try to use SIP. |
| 14:45.53 | lordvadr | I can't say for certain what the provider is running (I believe asterisk), nor how they terminate (I believe isdn) |
| 14:46.52 | ManxPower-work | lordvadr: It is doubtful that your provider uses PRI for PSTN. Chances are they use a SIP provider. |
| 14:47.08 | *** join/#asterisk crt_devel (~crt@host9-51-static.91-82-b.business.telecomitalia.it) |
| 14:47.09 | Katty | i agree. a sip provider would be a lot cheaper for them |
| 14:47.22 | *** join/#asterisk faiz_grw (~faiz_grw@209.17.184.117) |
| 14:47.25 | crt_devel | hello chan. |
| 14:47.45 | lordvadr | The problem works the other direction as well. If I point one of my DID's into DISA, dial it, hit buttons, and I still get dialtone |
| 14:47.53 | Katty | crt_devel: hello. |
| 14:48.04 | ManxPower-work | lordvadr: We are sad for you. Now stop making excuses and try SIP |
| 14:48.23 | crt_devel | anyone can help me with a strange issue that i havce after upgrade the asterisk from 1.4.21 to 1.4.29.. |
| 14:48.42 | ManxPower-work | ~ask |
| 14:48.43 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:49.51 | lordvadr | I don't think it's purely a SIP/iax problem. Here at work, we have a toshiba phone system. I run SIP between here and my home box. If I dial from here, over sip, into DISA at my box, I still don't get DTMF at the box at my house. That's all SIP and still doesn't work. |
| 14:50.20 | lordvadr | at least its all sip for the IP part |
| 14:50.42 | xLP | did you try sniffing to see how DTMF are actually sent (if sent at all) ? |
| 14:50.59 | ManxPower-work | lordvadr: Do you think you are the first person with this issue? You are not the first. You are not the 15th, you are not even the 100th person to have this issue. Take my advice or not. |
| 14:51.27 | crt_devel | calls that going to external numbers have the ringback tone on the caller telephone , and is ok |
| 14:51.49 | crt_devel | call that coming from the external to sip clients have the ringback tone , and are ok |
| 14:52.27 | crt_devel | calls from sip to sip users internal on the asterisk don't hear the ringback, only silence. After the answer, conversation flow, voice is ok. |
| 14:52.32 | lordvadr | ManxPower: I realize that. I'd like to understand the issue before giving up and doing it differently "because someone said so". I have a hunch it has something to do with automon, or parking, or such, since that was what I experimented with last. |
| 14:53.01 | crt_devel | i make a packet capture on the caller side, and the aserisk don't send the 183 progress to the router |
| 14:53.04 | crt_devel | any idea ? |
| 14:53.26 | ManxPower-work | crt_devel: remove the "r" option to Dial and make sure you have a /etc/asterisk/indications.conf |
| 14:53.29 | lordvadr | ManxPower: and, if I run straight sip (albeit it to the pstn) the issue is still present. |
| 14:53.56 | lordvadr | xLP: how would it be sent or not. Does sip send the dtmf inband or out of band? |
| 14:53.58 | lordvadr | or both? |
| 14:54.05 | crt_devel | r option is not present. I try to add it but don't make difference. indications.conf is present and the format is ok. |
| 14:54.10 | ManxPower-work | crt_devel: This advice is only valid if you do NOT use a GUI for Asterisk. |
| 14:55.22 | crt_devel | is running on a centOS at runlevel3. |
| 14:55.22 | ManxPower-work | crt_devel: which phones? |
| 14:55.28 | crt_devel | linksys spa2102 routers, grandstream, vigor3300. all with the same issue. |
| 14:55.55 | Katty | :< |
| 14:55.59 | crt_devel | and - i try also g711 and g729 swap on codecs.. |
| 14:56.12 | crt_devel | ..and tried also progressinband=yes on sip.conf. |
| 14:56.26 | xLP | lordvadr: it all depends on your settings |
| 14:56.31 | crt_devel | 1.4.21 works fine |
| 14:56.32 | *** join/#asterisk darkskiez_ (~dz@62-50-207-164.client.stsn.net) |
| 14:56.39 | xLP | can be sent as AUDIO, RTP or another one |
| 14:56.59 | crt_devel | btw, i compiled only sources of 1.4.29, and nothing else. |
| 14:57.53 | lordvadr | "Choices are inband, rfc2833, or info". I am using the default, which is rfc2833. Lemme go read that. |
| 14:57.59 | *** join/#asterisk adadelu (~dennis@ada-bcn-fw01.adamoeurope.com) |
| 14:58.18 | V4mpire | anyone know of cheap sip providers that offer free multiple landline numbers for canada/UK |
| 14:58.34 | xLP | lordvadr: ok, I thought you had a look at it before. Usually when DTMF of some phones were not detected, it was linked to that... |
| 14:58.50 | xLP | (some provider accept only RFC, some only info, some all.....) |
| 14:59.19 | lordvadr | I dug into this a while ago. Problem cropped some months ago, and now that I can talk phone-system to phone-system, it makes me think it has nothing to do with my provider |
| 14:59.21 | adadelu | does anyone know what the Max value in queuestatus stands for? looking for a way to put up the hold-time for the person who waited the longest in a queue. |
| 14:59.39 | lordvadr | xLP: how would I sniff for inband? |
| 14:59.55 | lordvadr | either way, let me look into this for a min |
| 15:00.01 | crt_devel | a thing that i note is that on outside-to-internal calls, and in inside-to-external calls, the CLI said me that "xxxx is making progress passing to yyyy" |
| 15:00.14 | crt_devel | on internal sip calls, the CLI don't say this. |
| 15:00.32 | adadelu | V4mpire: I'm using mydivert for my Canadian did:s. Tried Les.net before but never got It up and running properly. |
| 15:01.15 | crt_devel | on sip calls, the CLI said only "201 is ringing" |
| 15:01.22 | adadelu | V4mpire: This is for my home Asterisk that is. As we are a bit multinational. I'm Swedish, and my girlfriend is Canadian. We live in Barcelona/Spain :-) |
| 15:01.38 | V4mpire | ahh kl |
| 15:02.07 | xLP | lordvadr: for inband, welll... take Wireshark with RTP player, hoping your codec is G711 then you can record is listen with RTP player, check if you hear anything... |
| 15:03.22 | V4mpire | adadelu, ahh well im after a certain area really they dont cover it |
| 15:03.28 | lordvadr | xLP: I've tried it in all three modes. I'll try the rtp player in wireshark, and do some digging on the other two modes to find out what I'm looking for. Thanks for your help. |
| 15:03.32 | lordvadr | I'm sure I'll be back |
| 15:03.59 | xLP | lordvadr: yw... however if you've already tried all 3 modes I'm not sure there's much hope in that direction... |
| 15:04.01 | adadelu | V4mpire: Ah, my Canucks are located in Ottawa, so we are lucky then. |
| 15:04.07 | hipitihop | I'm getting "chan_sip.c:19477 handle_request_invite: Failed to authenticate device "0402xxxxxx"<sip:0402xxxxxx@202.169.178.10:5060>;tag=7e58e8aa-co1349-INS001" when I try to call in to * from my mobile -> vsp -> asterisk |
| 15:04.32 | V4mpire | yea found a few canadian DID's many free but none covering the area im after lol |
| 15:04.44 | xLP | hipitihop: maybe a too restrictive "type" setting? |
| 15:06.25 | xLP | hipitihop: or something with "insecure" |
| 15:07.01 | adadelu | V4mpire: Ah, guess smaller towns can be hard? |
| 15:07.08 | leifmadsen | V4mpire: pretty picky for someone looking for free |
| 15:07.28 | leifmadsen | what area are you looking for? In Canada, inbound DIDs are typically only available in larger markets |
| 15:07.40 | hipitihop | xlp insecure=port,invite ... btw I'm on 1.6.x |
| 15:07.48 | V4mpire | leifmadsen, well if i cant get 1 for a certain area theres no point in me using it as its only pretty much for 1 person over there that only has local calls free |
| 15:08.04 | V4mpire | leifmadsen, towns in saskachewen |
| 15:08.15 | leifmadsen | V4mpire: if it's not Saskatoon it's likely to not exist |
| 15:08.21 | leifmadsen | free or otherwise |
| 15:08.37 | V4mpire | haven't managed to find any what so ever for there either |
| 15:08.44 | V4mpire | ahh well nvm |
| 15:08.46 | xLP | hipitihop: just to check, you may try insecure=very |
| 15:09.35 | adadelu | V4mpire: It seems to be hard finding DID:s over there. Guess It's somewhat because the Bell and Rogers megalomania. Here in Europe It's not the same thing really.. |
| 15:09.58 | leifmadsen | V4mpire: ya, I just looked at unlimitel.ca and there are no Saskatchewan DIDs available at all. Just Winnipeg and Calgary |
| 15:10.07 | Katty | arlkgjlakjsdlf |
| 15:10.13 | Katty | stop making me beep >.< |
| 15:10.14 | V4mpire | yea |
| 15:10.24 | leifmadsen | Katty: meep |
| 15:10.26 | V4mpire | uk ones seem quite easy to get ahold of if you look in the right place |
| 15:10.30 | Katty | THAT"S IT |
| 15:10.32 | Katty | hugs leifmadsen |
| 15:10.37 | leifmadsen | V4mpire: a LOT less area to cover in the UK :) |
| 15:10.46 | leifmadsen | Katty: I accept your hug and raise you a coffee |
| 15:11.03 | V4mpire | more area codes tho :p |
| 15:11.12 | Katty | leifmadsen: i meet your coffee and raise you an iced tea |
| 15:11.16 | adadelu | leifmadsen: Isn't Calgary in Alberta? and Winnipeg in Manitoba? |
| 15:11.26 | hipitihop | xLP, WARNING[10858]: chan_sip.c:22639 set_insecure_flags: Unknown insecure mode 'very' ... afaik no longer supported on 1.6 |
| 15:11.31 | leifmadsen | adadelu: precisely my point |
| 15:11.44 | leifmadsen | hipitihop: replace with "invite,port" |
| 15:11.48 | leifmadsen | (less quotes) |
| 15:11.55 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 15:11.59 | leifmadsen | insecure=port,invite |
| 15:12.23 | adadelu | leifmadsen: Then I was not to drunk on "La fin du monde" while reading up on my Canadian geography :-) |
| 15:12.25 | xLP | hipitihop: I have it on 1.6.... where do you see this warning? |
| 15:12.30 | leifmadsen | adadelu: lol |
| 15:12.30 | ManxPower-work | hipitihop: Looks like you did NOT read the UPGRADE*.txt files -- these files tell you the important Asterisk changes. |
| 15:12.46 | leifmadsen | 1.6 is not specific enough |
| 15:12.54 | leifmadsen | 1.6.0 vs. 1.6.1 vs. 1.6.2 are all MAJOR version changes |
| 15:13.17 | leifmadsen | the same as 1.2 vs. 1.4 is a MAJOR version change |
| 15:13.25 | V4mpire | anyone know of a cheaper uk voip provider than voiptalk.org that offer geographical numbers for free ? |
| 15:13.26 | adadelu | leifmadsen: you don't have any clue about queuestats? Trying to figure out that the Max: value stands for? |
| 15:13.35 | hipitihop | leifmadsen, Asterisk 1.6.2.0~rc2-0ubuntu1.2 |
| 15:13.35 | crt_devel | hmm - a little question - what to compile when i make an upgrade of an asterisk ? |
| 15:13.38 | leifmadsen | adadelu: sorry, never used queuestats |
| 15:13.46 | leifmadsen | hipitihop: yuck... that's crazy old |
| 15:13.53 | LnxBil | c0rnoTa: So, LDAP is mainly working now, but still some esthetic things: still 'Received SIP subscribe for peer without mailbox' and the weird ldap syntax errors in between aka 'filter="(&(?cn=))'. Maybe someone is reading this who is a master in asterisk+ldap |
| 15:14.02 | LnxBil | I'll be back tomorrow |
| 15:14.05 | xLP | V4mpire: dunno about prices, have you checked sipgate? |
| 15:14.05 | LnxBil | thx |
| 15:14.07 | hipitihop | ManxPower-work, I had insecure=port,invite ... someone suggested = very |
| 15:14.08 | leifmadsen | LnxBil: feel free to open issues! |
| 15:14.21 | leifmadsen | 'very' == 'port,invite' |
| 15:14.24 | LnxBil | leifmadsen: I'll consider :-p |
| 15:14.30 | leifmadsen | LnxBil: then I can triage it when they come in :) |
| 15:14.40 | LnxBil | okay, bye |
| 15:14.42 | leifmadsen | peas |
| 15:14.43 | ManxPower-work | hipitihop: correct. insecure=very was removed and replaced with something that is documented in UPGRADE*.txt |
| 15:14.51 | adadelu | leifmadsen: got this question from our CEO that he wanted to represent the holdtime for the longest waiting in each queue.. Holdtime just gives a 24h average or something like that.. |
| 15:15.49 | hipitihop | leifmadsen, only thing standard version available on my karmic ubuntu distro, what should I have ? |
| 15:16.06 | ManxPower-work | hipitihop: You should install from source |
| 15:17.28 | ManxPower-work | hipitihop: if you don't you'll just be wasting everyone's time. |
| 15:18.14 | Skeeter- | is there anyway to remvoe the ringing when u have a 2nd call |
| 15:18.22 | V4mpire | xLP, 0.1p more lol |
| 15:18.39 | [TK]D-Fender | Skeeter-: thats up to your PHONE |
| 15:18.59 | Skeeter- | its ez to do with soundpoint, not with spectralink |
| 15:19.16 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:19.35 | [TK]D-Fender | Skeeter-: And asking in here when you're using FreePBX for something that is, at best, outside of your control (should there even be a flag you could pass the phone) is pointless. |
| 15:20.02 | hipitihop | ManxPower-work, fair enough, I assume I can just keep my sip.conf and extensions.conf and apt-get remove the package ... will also go lookup how to build from source |
| 15:20.53 | Skeeter- | [TK]D-Fender, understood |
| 15:21.02 | ManxPower-work | hipitihop: we don't if your package removes those files or not. |
| 15:21.07 | crt_devel | ManxPower-work: i tried to check if there any r options on the Dial command, but there's no "r" anymore. |
| 15:22.02 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 15:22.30 | vader-- | any of you guys using/used sipx? |
| 15:22.41 | adadelu | lordvadr:rtpbreak is rather nice, If you want to gather alot of data, since wireshark pukes if your pcaps are too large. but maybe that's not your problem :-) |
| 15:22.41 | Katty | hi ariel_ |
| 15:22.59 | hipitihop | ManxPower-work, good point probably wont |
| 15:23.10 | ariel_ | says hello Katty , gives a hug |
| 15:23.15 | Katty | hugs ariel_ |
| 15:24.03 | adadelu | Katty: are we at the point when you start jumping ang giggling soon? :-) |
| 15:24.18 | Katty | no my caffeine isn't quite finished yet |
| 15:24.22 | Katty | i still have half of it to go |
| 15:24.27 | Katty | try back around 2PM |
| 15:25.06 | adadelu | Katty: Caffeine... 200M to starbucks, but not quite desperate enough yet. |
| 15:27.02 | Katty | aww. |
| 15:27.10 | Katty | BB is passed out with his paws up in the air |
| 15:29.02 | adadelu | Katty: wish that were the case over here too, and the weather was better. Got a nice roof terrace awaiting summer, for those kinds of activities :-) |
| 15:29.31 | Katty | yeah i can't wait until summer. |
| 15:29.36 | Katty | it's 17F this morning |
| 15:29.40 | ManxPower-work | I can't wait for spring |
| 15:29.44 | Katty | that too. |
| 15:29.57 | *** join/#asterisk youngproguru (~youngprog@74.10.229.58) |
| 15:30.06 | Katty | i can't wait until it's at least 65F so i can put the boys in a playpen outside |
| 15:30.37 | coppice | 65F sounds chilly |
| 15:30.52 | Katty | it is a bit chilly |
| 15:31.01 | Katty | but it's a lot better than 40F |
| 15:31.16 | Katty | and the boys can't be out in really hot weather anyway |
| 15:31.37 | *** join/#asterisk romaNewbie2010 (~roma0@adsl-77-187-191.mia.bellsouth.net) |
| 15:31.41 | coppice | here it only ever gets as low as 40F in the hills |
| 15:31.46 | romaNewbie2010 | hello everyone |
| 15:31.56 | adadelu | Katty: 65 is good. We get around 104-ish here during the summer.. |
| 15:32.11 | romaNewbie2010 | anyone care to help an asterisk newbie? |
| 15:32.22 | beek | hugs Katty |
| 15:32.26 | beek | ~ask |
| 15:32.26 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:32.45 | romaNewbie2010 | :) |
| 15:33.21 | romaNewbie2010 | This is my dial command: Dial(SIP/100&SIP/106&SIP/107,20) |
| 15:33.42 | romaNewbie2010 | that 20 sec pause... if no extension is registered/connected results in dead air for callers |
| 15:33.43 | beek | and this is my dial command on drugs: Dial(!$#%@%$^QWRE) |
| 15:33.43 | Katty | optimum fuzzy temperature is actually 60 to 70F |
| 15:33.47 | Katty | hugs beek |
| 15:33.50 | romaNewbie2010 | is there a way to send it right to voicemail? |
| 15:34.06 | romaNewbie2010 | like skip the 20 sec pause? |
| 15:34.08 | Katty | romaNewbie2010: those are basics covered in the book |
| 15:34.12 | Katty | infobot: thebook |
| 15:34.13 | infobot | rumour has it, thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 15:34.15 | ManxPower-work | romaNewbie2010: make sure you have a /etc/asterisk/indications.conf |
| 15:34.21 | coppice | 60F sounds more shivery than fuzzy |
| 15:34.22 | [TK]D-Fender | romWhat pause? You asked it to dial for 20 secs.. |
| 15:34.35 | [TK]D-Fender | romaNewbie2010: What pause? You asked it to dial for 20 secs.. |
| 15:34.37 | Katty | coppice: they also have a coat tho |
| 15:34.43 | ManxPower-work | I think he means "caller does not hear ringback" |
| 15:34.49 | romaNewbie2010 | yes but I dont have a sip phone connected or anything... if I take the 20 out it never goes to voicemail |
| 15:34.52 | Katty | coppice: ferrets don't have a way to cool themselves off. |
| 15:34.57 | coppice | I have a coat, but I prefer not to wear it |
| 15:35.09 | Katty | coppice: so anything above 80F is asking for disaster |
| 15:35.25 | beek | Katty: Do they like to be sprayed with water? |
| 15:35.28 | coppice | ferrets seem perfectly have in the summer |
| 15:35.41 | Katty | beek: yes, and swimming (= |
| 15:35.46 | [TK]D-Fender | romaNewbie2010: Only reason it would wait with dead air is if * thought there was something to contact.... like if you specified a specifiy host IP and didn't have qualify enabled for it to track if it goes down or not |
| 15:35.54 | [TK]D-Fender | romaNewbie2010: Fix your peer setup |
| 15:37.37 | adadelu | Katty: we had a persian kitty before. Spain was a bit to hot for him during summer, so he mostly lived under the air-conditioning between June-September |
| 15:39.03 | hipitihop | ManxPower-work, ok, biting bullet, removed and now downloading 1.6.2.4 source |
| 15:39.43 | ManxPower-work | hipitihop: I'd go with 1.6.1x, actually. 1.6.2 is so new I would not trust it without extensive testing |
| 15:40.39 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:40.39 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:41.26 | Kobaz | this is bad |
| 15:41.28 | Kobaz | this is really bad |
| 15:41.34 | [TK]D-Fender | ManxPower-work: I'm hearing that 1.6.2 is coming out more stable..... old paradigm is fading fast |
| 15:42.17 | Kobaz | latest polycom firmware/bootrom... the polycom will randomly return CANCEL when you dial it... and when it doed return CANCEL, it doesn't hang up the call on the phone side... so the polycom is still ringing, but asterisk has hung up the call |
| 15:42.25 | ManxPower-work | [TK]D-Fender: I hold a grudge for a long time. |
| 15:42.42 | ManxPower-work | Kobaz: not in MY experience it doesn't. |
| 15:42.57 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 15:43.20 | ralonso | someone recommend a good program to backup-mirror and restore of a asterisk or complete debian? |
| 15:43.27 | Kobaz | ManxPower-work: 3.2.2/4.2.1 |
| 15:43.33 | Kobaz | ralonso: rsync |
| 15:43.38 | romaNewbie2010 | Fender: the host is dynamic but I qualify was set to no for the 3 extensions |
| 15:43.44 | romaNewbie2010 | so I will try with yes right now... |
| 15:43.46 | leifmadsen | ralonso: dd |
| 15:43.47 | ManxPower-work | Kobaz: Yes, that's what we run on about 100 phones across 3 customers |
| 15:43.53 | Kobaz | hmm |
| 15:44.27 | leifmadsen | I have used 1.6.2 in a call centre for the last 4-5 months with no issues |
| 15:44.32 | leifmadsen | (or at least no major issues) |
| 15:44.37 | leifmadsen | even pre 1.6.2.0 release |
| 15:44.45 | leifmadsen | its running revision 196xxx or something |
| 15:44.51 | Kobaz | 1.6.2.0-rc's crashed on me left and right |
| 15:44.54 | leifmadsen | so things definitely stabilize a lot faster now |
| 15:45.03 | leifmadsen | I think it depends on what you're doing too (as always) |
| 15:45.04 | Kobaz | the latest 1.6.2s are pretty good though |
| 15:45.07 | Kobaz | yeap |
| 15:45.18 | leifmadsen | I'm not using ODBC or anything in that system. Straight up dialplan and single system. |
| 15:45.24 | hipitihop | ManxPower-work, [TK]D-Fender, I guess I can afford to try either at this point... just experimenting @ home and voip only, not familiar enough to know +/- |
| 15:45.34 | Kobaz | yeah |
| 15:45.42 | Kobaz | odbc was very problematic in the rc's |
| 15:45.50 | leifmadsen | for experimenting, I pretty much always recommend to use the latest branches as you might as well use the latest features |
| 15:45.53 | romaNewbie2010 | Fender: it did not work with qualify to yes |
| 15:46.00 | leifmadsen | that's too bad, because ODBC stuff has always been rock solid for me |
| 15:47.01 | Kobaz | heh not me |
| 15:47.16 | [TK]D-Fender | romaNewbie2010: pastebin your SIP configs masking only passwords, the output of "sip show peers" and the CLI output with SIP DEBUG for your call attempt |
| 15:47.17 | [TK]D-Fender | ~pb |
| 15:47.18 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 15:47.18 | Kobaz | odbc to postgres constantly leaves dangling connections, when using stuff in func_odbc |
| 15:47.19 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 15:47.21 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 15:47.42 | Kobaz | so i find that a box has maxed out it's connections and no longer can connect to the db |
| 15:47.52 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 15:47.58 | Kobaz | so i wrote a script to kill any connections older than an hour, and that fixed it |
| 15:48.14 | ManxPower-work | Heck, a health insurance company screwed me over 25 years ago and I still hold a grudge against them. |
| 15:48.20 | leifmadsen | Kobaz: ah, I used postgres back in the day, but now I use mysql with odbc |
| 15:48.39 | Kobaz | hah |
| 15:48.40 | Kobaz | why? |
| 15:49.17 | romaNewbie2010 | FENDER: I have it in database not in text |
| 15:49.26 | romaNewbie2010 | let me try to put it in readable format for u |
| 15:49.39 | Kobaz | romaNewbie2010: well export it then |
| 15:49.45 | Kobaz | oh, you are |
| 15:49.48 | Kobaz | never mind then |
| 15:49.50 | [TK]D-Fender | romaNewbie2010: No... BINARY dammit |
| 15:49.52 | *** join/#asterisk Deeewayne (~dwayne@75.76.254.162) |
| 15:49.52 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:49.55 | [TK]D-Fender | SHOW ME YOUR BITS! |
| 15:49.55 | Katty | hi Deeewayne |
| 15:50.01 | [TK]D-Fender | :p |
| 15:50.02 | Katty | Deeewayne: did you attend playtime? |
| 15:50.19 | [TK]D-Fender | wow... that was kinda bad.. even for me ;) |
| 15:50.31 | leifmadsen | Kobaz: I find mysql easier to manage. I don't have to create manual blob objects for ODBC voicemail, and mysql replication works pretty decently |
| 15:50.32 | Deeewayne | Katty, no, but I saw them waking up around dinner time yesterday |
| 15:50.33 | ManxPower-work | [TK]D-Fender: only if you buy me dinner and drinks first. |
| 15:50.37 | Katty | Deeewayne: :> |
| 15:50.48 | leifmadsen | Kobaz: although the main reason is mostly because my clients use it |
| 15:50.55 | Katty | Deeewayne: i ran all of pippins toys through the laundry...i'm gonna put them out in a laundry basket for him to restash tonight |
| 15:51.00 | Katty | Deeewayne: if you want to attend (= |
| 15:51.04 | [TK]D-Fender | ManxPower-work: And if I screwed you that night... would you still hold a grudge? ;) |
| 15:51.10 | [TK]D-Fender | is getting worse.... |
| 15:51.13 | Deeewayne | I will! |
| 15:51.18 | Katty | Deeewayne: :>>> |
| 15:51.25 | ManxPower-work | [TK]D-Fender: I guess that would depend on how good you are. 8-) |
| 15:51.40 | romaNewbie2010 | ok |
| 15:51.40 | Deeewayne | I should bookmark crittercam. I have to google it each time |
| 15:51.42 | romaNewbie2010 | here it is |
| 15:51.47 | romaNewbie2010 | idnamehostnattypeaccountcodecallgroupcall-limitcancallforwardcanreinvitecontextdefaultipdtmfmodefromuserfromdomaininsecurelanguagemailboxmd5secretdenypermitmaskmusiconholdpickupgroupqualifyregextenrestrictcidrtptimeoutrtpholdtimeoutt38pt_udptlt38pt_rtpt38pt_tcpsetvardisallowallowfullcontactportregserverregsecondsusernamedefaultuser |
| 15:51.47 | romaNewbie2010 | 301106dynamicyesfriendLAB710yesyesphonesrfc2833inviteen1060.0.0.0/0.0.0.00.0.0.0/0.0.0.0default7noyesnonoallulaw;alaw01242748831106106 |
| 15:51.47 | romaNewbie2010 | 302107dynamicyesfriendLAB710yesyesphonesrfc2833inviteen1070.0.0.0/0.0.0.00.0.0.0/0.0.0.0default7nonononoallulaw;alaw50601265575731107107 |
| 15:51.47 | romaNewbie2010 | 310100dynamicyesfriendLAB710yesyesphonesrfc2833inviteen1000.0.0.0/0.0.0.00.0.0.0/0.0.0.0default7nonononoallulaw;alaw10241265036918100100 |
| 15:51.47 | romaNewbie2010 | 324117dynamicyesfriendLAB710yesyesphonesrfc2833inviteen1170.0.0.0/0.0.0.00.0.0.0/0.0.0.0default7nonononoallulaw;alaw537641267112620117117 |
| 15:51.51 | Katty | eeek |
| 15:51.57 | [TK]D-Fender | ManxPower-work: My girlfriends all say I'm great... not always in so many words ;) |
| 15:51.57 | Katty | Deeewayne: just do crittercam to infobot |
| 15:52.00 | ManxPower-work | MY EYES!!! MY EYES!!! |
| 15:52.04 | [TK]D-Fender | roma.. PASTEBIN dammit |
| 15:52.06 | beek | my eyes are burning! |
| 15:52.06 | [TK]D-Fender | ~pb |
| 15:52.07 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 15:52.21 | adadelu | i just fax myself, and I went blind! |
| 15:52.26 | adadelu | faxed even |
| 15:52.30 | Deeewayne | Katty, yay! |
| 15:52.59 | Katty | Deeewayne: i won't get home till about 5:30CST |
| 15:53.07 | Katty | Deeewayne: so i'd guess about 6 or 6:30 CST |
| 15:53.24 | Deeewayne | ok :-) |
| 15:53.35 | romaNewbie2010 | srry about that |
| 15:53.38 | romaNewbie2010 | here: http://pastebin.com/BndsKUZr |
| 15:54.02 | [TK]D-Fender | romaNewbie2010: the rest now please... |
| 15:54.22 | crt_devel | ok. maybe a rm -rf * can solve my problem ? :) |
| 15:55.06 | [TK]D-Fender | crt_devel: you forgot the "/" :p |
| 15:55.23 | crt_devel | ah , right. Just because i am in / :P |
| 15:55.32 | edwin_quijada | Hi! |
| 15:55.49 | edwin_quijada | I am trying to use any TTS with good voices in spanish |
| 15:55.59 | edwin_quijada | Somebody knows one? |
| 15:56.19 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:56.23 | edwin_quijada | I tested Cepstarl, FreeTTS |
| 15:56.25 | edwin_quijada | Festival |
| 15:56.26 | [TK]D-Fender | romaNewbie2010: And I see all of your qualify= as "NO" there |
| 15:56.43 | edwin_quijada | but the voices are so bad |
| 15:56.57 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:57.37 | *** join/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 15:57.40 | *** part/#asterisk benngard (~benngard@213.88.138.230) |
| 15:57.47 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:58.52 | hipitihop | edwin_quijada, Festival |
| 16:01.28 | michael-i_ | Does anyone have some BRI examples for chan_dahdi.conf? The documentation does not list any and signaling types, etc have not yet been included. I'm having trouble with using both b-channels on the span and transfers on euroisdn. (asterisk 1.6.1.14) |
| 16:01.44 | hipitihop | edwin_quijada, not sure what distro you are on but I followed this thread and voices were orders of magnituted better http://ubuntuforums.org/showthread.php?t=677277 |
| 16:04.18 | *** join/#asterisk Jhirley (~Jhirley@h69-21-54-248.ldlwvt.dsl.dynamic.tds.net) |
| 16:05.21 | *** join/#asterisk darkskiez_ (~dz@62-50-207-164.client.stsn.net) |
| 16:05.51 | vader-- | any of you guys using/used sipxecs? |
| 16:06.30 | ralonso | uhm, and systemrescuecd is a good choice? |
| 16:07.08 | TheDavidFactor | ~pastebin |
| 16:07.09 | infobot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:07.45 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:07.50 | *** join/#asterisk nickaugust (~anonymous@71-33-207-229.hlrn.qwest.net) |
| 16:08.22 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 16:08.28 | wcselby | o/ |
| 16:08.39 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 16:09.21 | tzafrir | michael-i_, signalling = bri_cpe_ptmp ; for PtP |
| 16:09.30 | tzafrir | michael-i_, signalling = bri_cpe ; for PtP |
| 16:09.39 | tzafrir | michael-i_, signalling = bri_cpe_ptmp ; for PtMP |
| 16:09.48 | romaNewbie2010 | Fender? |
| 16:09.53 | tzafrir | (sorry for the typo) |
| 16:09.56 | romaNewbie2010 | the Qualify did it... my bad |
| 16:09.58 | Deeewayne | pets Katty's ferrets |
| 16:10.14 | romaNewbie2010 | BUT there is a qualify msg every 10th of a second or so |
| 16:10.23 | romaNewbie2010 | like 4 qualify msgs per extension per second |
| 16:10.47 | romaNewbie2010 | is that normal? can that be set to X seconds? |
| 16:11.11 | [TK]D-Fender | romanpastebin "sip show peer X" for each peer, and provide the rest of what I asked for |
| 16:11.18 | tzafrir | michael-i_, also, "transfers"? or are you connecting a BRI phone to it? |
| 16:12.18 | romaNewbie2010 | I am running realtime |
| 16:12.28 | romaNewbie2010 | it won't show anythin on sip show peers |
| 16:13.05 | michael-i_ | tzafrir: perhaps I'm looking for the wrong thing. After switching to a channel group, outgoing and incoming calls are failing after requesting transfer capability speech and then hangup cause 47. This is only a provider line. |
| 16:13.54 | TheDavidFactor | I've got fax for asterisk and I don't know how to use it :-( I've got a DID from Broadvox pointed at an Asterisk box, but I'm seeing this when I try to send a fax: http://pastebin.com/DJbC1kEB Can anyone help me? |
| 16:14.10 | *** join/#asterisk garymc (~chatzilla@host81-148-109-86.in-addr.btopenworld.com) |
| 16:14.20 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 16:14.36 | wcselby | TheDavidFactor - you're trying to send or receive a fax on the asterisk box? |
| 16:14.45 | TheDavidFactor | Receive |
| 16:15.50 | TheDavidFactor | I don't understand why the channel is falling through while ReceiveFAX is still executing, this may be completely unrelated, but it's the one thing I noticed |
| 16:15.55 | tzafrir | michael-i_, can you provide a PRI-level trace? pri set debug on span 1 |
| 16:16.04 | garymc | [TK]D-Fender : Hi im trying to configure a ip650 via tftp . Now I want the first 5 line buttons to be extension 200 and the sixth line button to be extension 203. Do I have to set each line in phone1.cfg, eg; reg1 to reg5 individualy then reg6 to 203. Or is there a better way of doing it? |
| 16:16.07 | wcselby | TheDavidFactor - pastebin the results of fax show stats |
| 16:16.42 | wcselby | garymc - that's how I've done it on that phone |
| 16:16.48 | [TK]D-Fender | garymc: You only set reg 1 & 2. #1 for 5 linekeys, #2 for 1 |
| 16:17.02 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:17.04 | garymc | cool |
| 16:17.07 | garymc | thanks |
| 16:17.18 | garymc | was just about to try that :P |
| 16:17.31 | TheDavidFactor | http://pastebin.com/0NWtYr5r |
| 16:18.59 | wcselby | TheDavidFactor - can you also pastebin your sip.conf entry for broadvox, removing the password / username? sorry |
| 16:19.16 | wcselby | (i'm still waking up, and it's been about a month since I've dealt with a t38 issue) |
| 16:19.38 | TheDavidFactor | wcselby, I don't have a sip entry for broadvox |
| 16:19.52 | michael-i_ | tzafrir: this is like when I used to tell my dad the mower wouldn't start....everything works now. I think these drivers have some slow reset/settle down issues. Thanks for responding at least |
| 16:19.53 | ManxPower-work | TheDavidFactor: You should |
| 16:19.59 | wcselby | TheDavidFactor - ... you have a register statement? |
| 16:20.18 | wcselby | TheDavidFactor - ... pastebin the [general] section of your sip.conf then, removing anything private |
| 16:21.14 | wcselby | TheDavidFactor - ... i want to see your udptl_t38 settings |
| 16:21.33 | michael-i_ | tzafrir: outgoing works fine, incoming does not work as expected. let me sort through the trace |
| 16:22.13 | TheDavidFactor | http://pastebin.com/YsH4vQd9 |
| 16:22.37 | TheDavidFactor | wcselby: I don't have any udptl_t38 settings, where can I find the documentation on them? |
| 16:22.50 | wcselby | TheDavidFactor - in the default sip.conf file, for starters.... |
| 16:22.59 | TheDavidFactor | ok |
| 16:22.59 | wcselby | TheDavidFactor - ... other places as well, let me find some. |
| 16:23.17 | wcselby | TheDavidFactor - which version of asterisk are you running? |
| 16:23.59 | *** join/#asterisk hipitihop (~denis@203.132.229.187) |
| 16:24.00 | TheDavidFactor | 1.6.2 trunk |
| 16:24.05 | garymc | [TK]D-Fender : Sorry to be a pain. That Worked for the line but the phone isnt picking up the time and date. all the others are? is it sip.cfg where the sntp server is set? |
| 16:24.10 | TheDavidFactor | it's just a test box |
| 16:24.17 | garymc | if so why would the ip650 not be getting the right time? |
| 16:24.20 | *** join/#asterisk farkus_ (chatzilla@cpe-72-225-212-219.nyc.res.rr.com) |
| 16:25.52 | romaNewbie2010 | ok Fender |
| 16:25.57 | romaNewbie2010 | here it is: http://pastebin.com/99ie7XY1 |
| 16:26.04 | romaNewbie2010 | thats the sip trace of the call |
| 16:28.27 | garymc | [TK]D-Fender : Sorry its all working. Super Star :) |
| 16:29.42 | wcselby | TheDavidFactor - sorry it's t38pt_udptl, not udptl_t38 |
| 16:30.03 | *** join/#asterisk Maximo (~maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 16:30.14 | wcselby | http://pastebin.com/Q7LTDdBE |
| 16:30.26 | *** join/#asterisk motu (~eotu@c-cdcee355.189-5-64736c12.cust.bredbandsbolaget.se) |
| 16:31.26 | wcselby | TheDavidFactor - also you may want to have a look at the example ReceiveFax config in the FFA Administrator Manual |
| 16:31.39 | TheDavidFactor | wcselby: thx, I have the samples so I went back and read them. I've added the t38pt_udptl setting. I'll see if that makes a difference |
| 16:32.04 | motu | Using 1.4.22, I do not receive any dtmf tones on incoming calls to my sip trunk at swedish voip provider megaphone, what should I look for? |
| 16:32.31 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 16:32.35 | wcselby | motu - the dtmfmode entry for your sip provider? |
| 16:32.45 | motu | rfc2833 |
| 16:32.51 | michael-i_ | Should / can switchtype= be used for BRI configs? |
| 16:33.38 | motu | have tried all dtmf modes |
| 16:33.43 | motu | using alaw |
| 16:33.46 | wcselby | motu - what codec? |
| 16:33.48 | wcselby | ahh |
| 16:33.59 | wcselby | check with them? |
| 16:34.04 | motu | voice works excellent |
| 16:34.11 | TheDavidFactor | wcselby: I was not aware of the FFA Administrator Manual, google found it and I am reading it, thanks! |
| 16:34.13 | romaNewbie2010 | Fender are u still here? |
| 16:34.39 | motu | shouldnt inband work with alaw if voice works well? |
| 16:36.43 | hipitihop | ManxPower-work, I did not do "make samples" incase that clobbered existing /etc/asterisk files. should tha tbe ok ? |
| 16:37.44 | ManxPower-work | hipitihop: "make samples" WILL overwrite your config files. You should, however, do a "make config" which install the scripts to start Asterisk as part of the system boot process |
| 16:37.48 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:38.14 | *** join/#asterisk RobH (~robh@2620:0:860:2:21e:c2ff:fe03:2465) |
| 16:39.17 | hipitihop | ManxPower-work, yep did that and rebooted now logged in to cli so progress, but getting some errors e.g. on outgoing call attempt I get " app_dial.c:1745 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)" |
| 16:39.33 | *** join/#asterisk xpot-mobile (~xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
| 16:39.41 | ManxPower-work | hipitihop: "sip show peers" does that peer have an ip address listed for it? |
| 16:39.47 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 16:40.40 | hipitihop | ManxPower-work, I have two peers my provider and my ATA, both show and both have IP's |
| 16:40.53 | redax | hi |
| 16:41.06 | ManxPower-work | hipitihop: pastebin your sip.conf (changing ONLY passwords) and the CLI output of a failed call. |
| 16:41.12 | [TK]D-Fender | hipitihop: Now answer the question he actually ASKED you |
| 16:42.27 | redax | please help me in Digium TE220 and HDLC. I've configured span2 as "span=2,2,0,ccs,hdb3,crc4" And "nethdlc=32-62". how can I get the hdlc0 device ? |
| 16:42.46 | hipitihop | [TK]D-Fender, sorry I thought I had, "both have IP's" |
| 16:42.57 | ManxPower-work | redax: What you are trying to do is something very few people ever do. |
| 16:43.26 | hipitihop | although originaly only one... hmm some timing issue perhaps... as subsequnet attempt now gets out. |
| 16:43.42 | QubeZ | Asterisk runs fine in VMWare ESXi environment right? I have a box with ESXi and just put latest * on it. Dual 3.0 Ghz w/ 4G of ram of which 1G is dedicated to *. I ran dahdi_test and it producing horrible numbers: < 99.800%. Any suggestions on how to optimize? |
| 16:44.38 | romaNewbie2010 | Fender did u see my binpaste? |
| 16:44.38 | ManxPower-work | QubeZ: why in the world would you think Asterisk runs fine in a VM? |
| 16:44.40 | redax | ManxPower-work: we're changing our GSM trunk provider, the old one gave PRI, the new one wants IP over HDLC E1, and SIP |
| 16:44.41 | wcselby | QubeZ - lots of controversy over running * in a VM |
| 16:44.45 | ManxPower-work | expecially when using DAHDI |
| 16:45.04 | ManxPower-work | redax: why not just get a Cisco router? |
| 16:45.08 | *** join/#asterisk nickaugust (~anonymous@71-33-207-229.hlrn.qwest.net) |
| 16:45.19 | QubeZ | ManxPower-work many people have told me, in here even, that they have managed to get it running well in VMWare ESXi |
| 16:45.25 | ManxPower-work | HDLC is the native T-1/E-1 protocol of Ciscos |
| 16:45.33 | QubeZ | i've read many success stories too, guess its not possible and quit? |
| 16:45.35 | ManxPower-work | QubeZ: how many people told you it is a bad idea? |
| 16:45.38 | wcselby | QubeZ - it will work, but it requires tweaking |
| 16:45.52 | redax | ManxPower-work: Time pressure... there's no time to buy a cisco |
| 16:45.53 | QubeZ | wcselby yup, understandable. I'm wondering what do I need to tweak. |
| 16:46.02 | wcselby | QubeZ - and no, i haven't done one in specifically vmware esxi |
| 16:46.14 | wcselby | QubeZ - you have dahdi dummy installed? |
| 16:46.15 | hipitihop | ManxPower-work, [TK]D-Fender ... dialing oout now working ... and attempted call in from my mobile shows "extension not found" so I'll see if I can sort that myself .... thanks for your help both |
| 16:46.21 | ManxPower-work | redax: I suspect you could easily have a Cisco delivered before you managed to figure out running HDLC with Zaptel/DAHDI |
| 16:47.11 | wcselby | motu - sorry my last statement may not have been clear - i was suggesting you contact your sip provider and troubleshoot with them |
| 16:47.21 | ManxPower-work | redax: I suspect that none of the 272 people here have ever used HDLC, and I suspect not more than 5 people here even know what it is. |
| 16:47.21 | redax | I though it is some kind of childplay, as some documentation talks about PPP/HDLC and dahdi/zaptel tools has an utility sethdlc :-o |
| 16:47.36 | redax | hihi |
| 16:47.51 | romaNewbie2010 | Fender did u had a chance to look at my paste??? |
| 16:48.10 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 16:48.11 | wcselby | redax - you may want to send an email to the list, lots of dahdi pros on there. a few in here, but I don't know if they're paying attention right now |
| 16:48.44 | romaNewbie2010 | Yes, No, Maybe? |
| 16:49.15 | ManxPower-work | romaNewbie2010: You understand that [TK]D-Fender has a real job that is not working here helping people, right? |
| 16:49.37 | ManxPower-work | If you take too long to do something that was asked of you chances are the person helping you will no longer be available. |
| 16:49.43 | [TK]D-Fender | romaNewbie2010: I never got a decent peer dump. |
| 16:50.14 | ManxPower-work | He, like many of us, may have just gotten tired of asking for information over and over again and just gave up on you. |
| 16:50.31 | redax | ManxPower, wcselby: ok, will try the list. either way thank you for your help. |
| 16:50.31 | QubeZ | wcselby yes dahdi_dummy |
| 16:50.46 | wcselby | QubeZ - describe your asterisk setup |
| 16:51.03 | wcselby | QubeZ - versions, what you're trying to do, etc |
| 16:51.29 | michael-i_ | tzafrir: here is my configuration and a failed incoming trace: http://pastebin.ca/1810351 The first call to 861 connects, connecting to 862 fails with busy |
| 16:51.49 | QubeZ | wcselby ESXi v4, * v1.6.2. Basically just trying to get the dahdi dummy to report good results. I haven't moved onto much else besides compiling the dahdi and * software. |
| 16:51.56 | QubeZ | basic setup thus far |
| 16:52.55 | tzafrir | michael-i_, are you sure it's 'dchan' and not 'hardhdlc'? What device do you use? |
| 16:53.17 | wcselby | QubeZ - what OS are you running for your VM? Centos 5.4? I thought esxi was just the VM container? |
| 16:53.25 | QubeZ | wcselby CentOS 5.4 final |
| 16:53.44 | wcselby | QubeZ - dahdi compiled and everything for that went fine? |
| 16:53.46 | QubeZ | all updates done running kernel 2.6.18.164-11 |
| 16:53.51 | romaNewbie2010 | ManxPower jesus... what is ur problem? |
| 16:53.57 | romaNewbie2010 | do you have ur period or something? |
| 16:54.16 | wcselby | romaNewbie2010 - wow, that's the way to get help in a free help channel, for sure! |
| 16:54.36 | michael-i_ | tzafrir: these are for sure dchan. they're custom drivers for an embedded appliance. with hardhdlc nothing works at all |
| 16:55.01 | wcselby | QubeZ - like I said, I haven't done exactly what you're trying. Your only issue so far is the dahdi_test result? |
| 16:55.06 | romaNewbie2010 | this is a waste of time anyway |
| 16:55.08 | ManxPower-work | romaNewbie2010: I wish you the BEST of luck. |
| 16:55.13 | *** part/#asterisk romaNewbie2010 (~roma0@adsl-77-187-191.mia.bellsouth.net) |
| 16:55.58 | hipitihop | does a dance both incoming and outgoing calls now work... ManxPower-work , leifmadsen & [TK]D-Fender thanks for suggesting to use the source :-) |
| 16:56.07 | QubeZ | wcselby yup, only the dahdi_test result is below 99.98% |
| 16:56.15 | QubeZ | specifically, 99.8xx |
| 16:56.55 | hipitihop | hey voipmonk |
| 16:57.09 | geneticx_wrk | Hi everyone. |
| 16:57.16 | wcselby | hmmm....that's a lot lower than my current test, but then again I've got an actual t1 card in. i'll look for a machine that has dummy and check those results |
| 16:58.26 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 17:00.17 | hipitihop | thanks to all for your help, way too late here, time for bed... till next time. |
| 17:01.56 | *** join/#asterisk Khratos (~khratos@190.166.116.107) |
| 17:02.03 | *** join/#asterisk hfb (~hfb@pool-96-247-114-78.lsanca.dsl-w.verizon.net) |
| 17:02.34 | *** join/#asterisk spenguin[work] (~penguin@122.182.0.38) |
| 17:07.02 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 17:07.48 | redax | geez. where can I find a searchable asterisk-dev or any dahdi/zaptel related mailinglist archive |
| 17:08.33 | wcselby | http://lists.digium.com/pipermail/asterisk-dev/ ? |
| 17:08.41 | wcselby | either that or google, or a combination fo the two |
| 17:09.11 | wcselby | you may want to look at the asterisk-users list too, http://lists.digium.com/pipermail/asterisk-users/ |
| 17:10.13 | wcselby | http://www.google.com/search?hl=en&client=firefox-a&hs=fR2&rls=org.mozilla%3Aen-US%3Aofficial&q=site%3Alists.digium.com+asterisk-users+dahdi+hdlc&aq=f&aqi=&aql=&oq= |
| 17:10.30 | *** join/#asterisk cguerrero (~cuauhtemo@200.79.231.94) |
| 17:12.23 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 17:14.06 | redax | wcselby: that pipermail stuff is not searchable :D |
| 17:14.23 | wcselby | redax - hence the suggestion to combine with google :) |
| 17:14.25 | redax | hehh. found an initscript at dahdi-tools... ifup-hdlc |
| 17:14.48 | redax | wcselby: or download all the gzipped months :) |
| 17:15.16 | redax | the only problem with the ifup-hdlc is there's no default configuration for the script |
| 17:15.31 | ManxPower-work | ~mailinglist |
| 17:15.32 | infobot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 17:16.08 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 17:16.47 | redax | ah site: trick, thanks ManxPower |
| 17:18.26 | *** join/#asterisk dinesh___ (~dinesh@77-58-221-165.dclient.hispeed.ch) |
| 17:19.38 | dinesh___ | hi folks, I got a quick question: Is it easy to configure Asterisk as a SIP server that registers to another SIP provider for the incoming number and that uses a few several other SIP providers for outgoing call depending on the number prefix ? |
| 17:19.58 | paulc | dinesh__: absolutely! :-) |
| 17:20.16 | dinesh___ | okay that's cool then because I plan to do that next week ;) |
| 17:21.21 | dinesh___ | i have to wait for the wireless rj-11 <-> sip adapter i just ordered |
| 17:21.57 | paulc | dinesh__: what make/model is that then? |
| 17:22.21 | redax | wait. I have the hdlc0 device :D |
| 17:22.35 | wcselby | redax - grats! |
| 17:23.41 | *** join/#asterisk [8none1] (~8none1]@ps14528.dreamhost.com) |
| 17:24.41 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 17:24.47 | ujjain | Is Asterisk better than Hylafax for faxing? |
| 17:24.57 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 17:25.08 | wcselby | ujjain - depends |
| 17:25.12 | [TK]D-Fender | ujjain: Over what? <- |
| 17:25.25 | [TK]D-Fender | ujjain: What qualifies as "good"> |
| 17:25.27 | [TK]D-Fender | ? |
| 17:25.28 | ujjain | over internet. I am looking for faxing via a webinterface. |
| 17:25.40 | wcselby | you want to fax over VoIP? |
| 17:25.42 | ujjain | no physical fax involved, via SIP protocol. |
| 17:25.42 | [TK]D-Fender | ujjain: Both can. |
| 17:25.43 | ujjain | Yes. |
| 17:26.29 | redax | how can one compare a fax sw to a pbx? |
| 17:26.41 | dinesh___ | paulc it's the SPA2102 + wireless adapter actually |
| 17:26.54 | [TK]D-Fender | redax: Like apples & oranges |
| 17:27.28 | paulc | move dinesh__: Ah gotcha - I've used a SPA-3001 with the wireless adapator before, it worked really well :-) |
| 17:30.47 | ujjain | redax: I have no idea, I am new to this VOIP business. |
| 17:31.38 | carrar | me too |
| 17:31.43 | carrar | What is this VOIP crap anyways |
| 17:31.49 | redax | you can fax with asterisk using spandsp or there's several commercial swfax solution. but if you want _JUST_ a faxserver, use hylafax or efax/mgetty or whatever :D |
| 17:31.53 | Kobaz | ughh, i wish there was a way to reconfigure a polycom without rebooting |
| 17:32.10 | Naikrovek | you're using the web interface, huh |
| 17:32.20 | coppice | ujjain: FAX over SIP is always a bit iffy. Do it as VoIP and its very unreliable. Do it as T.38 and its subject to a lot of quirky implementations. * fronting iaxmodem+hylafax is one of the more successful combinations for high volume FAXing (hundreds of lines), and there are web interfaces for HylaFAX. However, that's usually with the * connected to the PSTN |
| 17:32.20 | Kobaz | that's one of the serious drawbacks of polycoms |
| 17:32.26 | Kobaz | Naikrovek: web interface? of course not |
| 17:32.32 | Naikrovek | okay cool |
| 17:33.04 | ujjain | coppice: Thhank you! |
| 17:34.31 | *** join/#asterisk etfonhomey1 (~etfonhome@74-143-192-74.static.insightbb.com) |
| 17:34.33 | ujjain | My provider does not support T38, but v.17 and slower. |
| 17:34.45 | coppice | ujjian: and watch out for patent trolls |
| 17:35.12 | wcselby | ujjain - if you want to seriously consider fax over VoIP, find a provider that does support t38 |
| 17:35.19 | *** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 17:36.19 | *** part/#asterisk Maximo (~maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 17:36.43 | *** join/#asterisk orangepower (~jon@jonsaves.xen.prgmr.com) |
| 17:37.42 | orangepower | anyone have a polycom VVX 1500? i'm having trouble with some settings, i've tried everything |
| 17:38.21 | *** join/#asterisk aruntomar (~aruntomar@61.17.193.163) |
| 17:39.10 | redax | geeez. can't unload dahdi :/ |
| 17:39.26 | *** join/#asterisk michael-i (~michael-i@p3EE2991B.dip0.t-ipconnect.de) |
| 17:39.32 | redax | as I can't shutdown the hdlc0 interface :-( |
| 17:39.57 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 17:41.00 | orangepower | anyone have any luck with asterisk hosted solutions? |
| 17:42.28 | [TK]D-Fender | orangepower: No. Every hosting provider simultaneously went out of business |
| 17:42.43 | *** part/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 17:43.47 | Katty | awwww |
| 17:43.57 | Katty | BB's all passed out |
| 17:43.59 | jaskew | was that when the RTP time counter hit 65535? |
| 17:44.00 | Katty | ZzzZZzzzz |
| 17:44.21 | orangepower | [TK]D-Fender: lol sucky. |
| 17:44.57 | redax | grr. hdlc0: Unable to set Cisco HDLC protocol information: Invalid argument |
| 17:45.18 | orangepower | I'm at a doctors office, I setup a asterisk machine, bought some VVX 1500 polycom phones, but I can't get them working, literally spent 5 hours configuring it.. no luck.. anyone familiar with polycom gear? |
| 17:45.34 | orangepower | at this point i'd paypal to get it fixed, even remotely |
| 17:45.56 | *** join/#asterisk elchorisolo (~elsopapa@200.123.148.68) |
| 17:45.59 | elchorisolo | hello |
| 17:46.26 | elchorisolo | ? |
| 17:46.33 | jaskew | orangepower: I use LES.NET - I'm pretty sure they use asterisk on the back end. I wouldn't call them a "hosted asterisk" per se, but they seem to be pretty reliable. |
| 17:47.03 | orangepower | jaskew: that's just a SIP provider? |
| 17:47.12 | elchorisolo | i have a little problem between an asterisk and a pbx panasonic can anyone help me? |
| 17:47.44 | [TK]D-Fender | elchorisolo: Maybe if you asked a specific question |
| 17:47.50 | elchorisolo | okei |
| 17:48.01 | Katty | http://farm3.static.flickr.com/2080/2262972677_b91f8f27ca_o.jpg |
| 17:48.03 | jaskew | orangepower: yeah - basically, but you can do a bit of configuration via their web interface. You couldn;t really set up a whole PBX though. Sorry - that probaably wasn;t helpful ;) |
| 17:48.09 | elchorisolo | [TK]D-Fender i use a fxo gateway.... |
| 17:48.20 | elchorisolo | and when i dial |
| 17:48.21 | orangepower | jaskew: it's ok, thanks anyway.. |
| 17:48.30 | elchorisolo | i only listen the tone |
| 17:48.37 | orangepower | i'd love for this polycom to "just work" but i'm pretty sure i should give up on it lol |
| 17:48.45 | Naikrovek | never |
| 17:48.58 | elchorisolo | and the asterisk doesnt dial |
| 17:48.58 | Naikrovek | i don't have a vvx phone but all my polycom's just work |
| 17:49.06 | wcselby | orangepower - there was discussion recently on the mailing list about that phone - they talked about needing the latest firmware and bootroms and stuff |
| 17:49.13 | jaskew | Katty: is that "found on the web" or are you really doing that? |
| 17:49.21 | orangepower | wcselby: that was my next try |
| 17:49.22 | elchorisolo | so after a time.. my pbx give me busy tone |
| 17:49.26 | wcselby | if you look at this month's list archives you should be able to find some useful information |
| 17:49.35 | orangepower | wcselby: thanks, it's a bitch of a phone |
| 17:49.36 | *** join/#asterisk Poincare (~jefffnode@v74.ampersant.be) |
| 17:49.45 | Katty | jaskew: what do you think? |
| 17:50.12 | Katty | jaskew: would you like another? |
| 17:50.16 | jaskew | Katty: Not sure - I don;t know you well enough yet... |
| 17:50.36 | jaskew | and that answer works for both questions. |
| 17:50.42 | elchorisolo | any ideaaaaaaaa |
| 17:50.44 | elchorisolo | ? |
| 17:50.58 | Katty | jaskew: http://farm4.static.flickr.com/3251/3020681556_f8e2ef4b9a_o.jpg |
| 17:51.03 | Katty | jaskew: that's Merry as a baby |
| 17:51.49 | [TK]D-Fender | orangepower: What is your issue with it? What ar you testing it with? what aspects aren't working? |
| 17:52.11 | jaskew | Katty: is that an opossum? They look different out here. Maybe because they aren't clean... |
| 17:53.06 | [TK]D-Fender | elchorisolo: Sorry your description is too weak as to what hardware is being used, what signalling, etc |
| 17:53.19 | [TK]D-Fender | elchorisolo: Which is calling which as well |
| 17:53.21 | elchorisolo | i need some help, i have a problem between an asterisk and a pbx panasonic tx 200 , when i dial from asterisk i listen the pbx tone but nothing is dialed, so after a time my pbx give me busy tone.... |
| 17:53.41 | [TK]D-Fender | elchorisolo: Well what are you dialing on? |
| 17:53.48 | elchorisolo | my gw is Grandstream fxo gateway |
| 17:53.59 | [TK]D-Fender | elchorisolo: So what is dialing a number? |
| 17:53.59 | Katty | jaskew: http://ustre.am/bEBU <- Merry realtime |
| 17:54.06 | elchorisolo | i dial 915151515 for exmaple |
| 17:54.12 | Katty | jaskew: and no, Merry is not a possum |
| 17:54.25 | elchorisolo | and my pbx give me tone but nothing is dialed |
| 17:55.16 | [TK]D-Fender | elchorisolo: show us a failed call from * CLI |
| 17:55.18 | [TK]D-Fender | ~pb |
| 17:55.19 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 17:55.20 | [TK]D-Fender | ^^^^^^^^ |
| 17:55.34 | elchorisolo | okei |
| 17:56.24 | elchorisolo | <PROTECTED> |
| 17:56.25 | elchorisolo | <PROTECTED> |
| 17:56.25 | elchorisolo | <PROTECTED> |
| 17:56.25 | elchorisolo | <PROTECTED> |
| 17:56.25 | elchorisolo | <PROTECTED> |
| 17:56.25 | elchorisolo | <PROTECTED> |
| 17:56.39 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 17:56.41 | jaskew | Katty: Silly me - Couldn't quite make it out from that angle. |
| 17:56.44 | *** kick/#asterisk [elchorisolo!~chatzilla@216.191.106.163] by [TK]D-Fender (elchorisolo) |
| 17:56.51 | *** join/#asterisk elchorisolo (~elsopapa@200.123.148.68) |
| 17:56.54 | elchorisolo | sorry |
| 17:57.04 | [TK]D-Fender | elchorisolo: PASTEBIN. Do not flood in here again |
| 17:57.16 | elchorisolo | what is pastebin |
| 17:57.23 | [TK]D-Fender | elchorisolo: Also you are not passing your gateway a number to dial anywhere in there |
| 17:57.24 | paulc | ~pb |
| 17:57.25 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 17:57.35 | elchorisolo | ahh okei |
| 17:57.35 | [TK]D-Fender | 12:55]<[TK]D-Fender>~pb |
| 17:57.37 | [TK]D-Fender | [12:55]<infobot>[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 17:57.53 | *** join/#asterisk lost_soul (~noymfb@cpe-67-241-68-202.twcny.res.rr.com) |
| 17:58.02 | [TK]D-Fender | [12:57]<[TK]D-Fender>elchorisolo: Also you are not passing your gateway a number to dial anywhere in there |
| 17:58.13 | [TK]D-Fender | elchorisolo: [12:56]<elchorisolo> -- Executing [9151515@i-transfer:1] Dial("SIP/802-0132cea0", "SIP/gwfxo") in new stack |
| 17:58.33 | elchorisolo | http://pastebin.com/XxywaSLT |
| 17:59.05 | elchorisolo | okei.. sorry i change to make a test |
| 17:59.07 | elchorisolo | wait |
| 17:59.23 | Katty | jaskew: http://farm3.static.flickr.com/2439/3589702173_f7aac3cafd_b.jpg <- BB |
| 17:59.34 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 17:59.49 | Deeewayne | awwww.... ferret ears! |
| 17:59.51 | Katty | jaskew: http://farm4.static.flickr.com/3623/3589701741_1016922822_b.jpg <- also BB, not me. |
| 18:00.13 | Katty | BB was about 6 months old when I adopted him from the local vet. |
| 18:00.15 | elchorisolo | http://pastebin.com/etC3AN1E |
| 18:00.19 | orangepower | [TK]D-Fender: so.. it just won't dial out/connect to the local asterisk server or the sip provider directly, i can't figure out what i'm doing wrong |
| 18:00.22 | Katty | he was put up for boarding and his parents never came back for him. |
| 18:00.25 | Naikrovek | katty: http://i.imgur.com/A7lmy.jpg <-- BL |
| 18:00.47 | Katty | Naikrovek: i saw that on reddit this morning :> |
| 18:00.55 | Naikrovek | ah fellow redditor |
| 18:00.57 | *** join/#asterisk Mhaddog (~Mhaddog@173-149-111-153.pools.spcsdns.net) |
| 18:01.00 | Naikrovek | can't sneak anything past you |
| 18:01.35 | Katty | :P |
| 18:01.58 | [TK]D-Fender | oragWhere do we see SIP DEBUG of attempts to register/call to *? |
| 18:02.03 | [TK]D-Fender | orangepower: Where do we see SIP DEBUG of attempts to register/call to *? |
| 18:02.40 | elchorisolo | [TK]D-Fender http://pastebin.com/etC3AN1E this is the trace |
| 18:02.56 | Mhaddog | good afternoon |
| 18:03.05 | elchorisolo | hi Mhaddog |
| 18:03.22 | [TK]D-Fender | elchorisolo: And if you plug a phone in parallel with it you still don't hear it dial? |
| 18:05.13 | elchorisolo | [TK]D-Fender i didnt do it, but i think it will dial, because i have another providers in asterisk working.. |
| 18:05.39 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 18:05.45 | elchorisolo | [TK]D-Fender i dont have any tool to do that |
| 18:06.11 | orangepower | [TK]D-Fender: no... |
| 18:06.45 | elchorisolo | sorry for my english ... |
| 18:06.53 | orangepower | [TK]D-Fender: i'm about to do it from scratch again, i'm not using a boot server, just using the phone itself for config is that ok? |
| 18:08.03 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 18:09.09 | elchorisolo | :s |
| 18:11.35 | [TK]D-Fender | orangepower: Either should work |
| 18:14.57 | *** join/#asterisk bahjons (~robert@140.99.23.26) |
| 18:15.13 | xLP | I'm encoounterting troubles with Asterisk, no clue why, I can only describe symptoms... Randomly, when you try placing a call, it fails, and some phones will fail registering. I sniffed network, while watching asterisk console and monitoring box load... Load is always around 0.04, I see packets incoming and being replied normally, and when the problem starts, I see incoming packets, but no answer, and nothing in the console, then, suddenly, after about 30 s |
| 18:15.18 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 18:16.13 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 18:16.15 | orangepower | [TK]D-Fender: i keep getting a fast busy when i try to make a call from the polycom, what does that mean? |
| 18:16.44 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.179.173.dsl.dyn.forthnet.gr) |
| 18:17.04 | bahjons | does anyone know how to use the set variables from asterisk manager 'originate' command? |
| 18:17.26 | [TK]D-Fender | orangepower: Stop looking at the phone and start looking at * |
| 18:17.36 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:18.02 | [TK]D-Fender | bahjons: They are accessable in your dialplan whereve you dump them. |
| 18:18.15 | bahjons | or how to troubleshoot if the variables are set... I tried a NoOp(show me $var1) but it's showing blank |
| 18:18.15 | orangepower | [TK]D-Fender: xmeeting works fine in asterisk, i'll go check asterisk logs |
| 18:20.30 | orangepower | [TK]D-Fender: it's not even hitting the * machine |
| 18:20.38 | orangepower | [TK]D-Fender: nothing shows up in the log at all |
| 18:20.49 | paulc | is it registered? |
| 18:20.57 | [TK]D-Fender | orangepower: then you haven't set it up right at all. |
| 18:22.06 | orangepower | [TK]D-Fender: seems that way, i'm missing something dumb i think |
| 18:22.31 | orangepower | it tries to go to the boot server, then fails, everytime on boot, is that normal for a standalone phone? |
| 18:22.45 | [TK]D-Fender | orangepower: yes |
| 18:22.58 | [TK]D-Fender | orangepower: then it will take whatever settings its got locally |
| 18:23.25 | orangepower | ok... i can screenshot the web settings, if it helps? i have no idea why it won't even connect |
| 18:23.42 | *** join/#asterisk TimeRider (~steve@78.32.26.1) |
| 18:23.47 | [TK]D-Fender | orangepower: Neither do we.... probably because we have nothign to look at... |
| 18:24.08 | orangepower | what's the next step? |
| 18:25.36 | paulc | I'll ask again - is it registered? |
| 18:25.42 | ariel_ | setup the phone with a tftp server and put the settings in the proper files. (Sorry I have never configured a polycom via it's web). |
| 18:26.26 | orangepower | paulc: it's not hitting * at all it seems |
| 18:26.32 | [TK]D-Fender | orangepower: You'd better start showing us config screens from it or something.... |
| 18:26.35 | Naikrovek | yeah |
| 18:26.44 | Naikrovek | we need more than just "this phone sucks" |
| 18:26.48 | [TK]D-Fender | orangepower: because right now we have nothing |
| 18:27.06 | orangepower | i'm screenshotting now |
| 18:27.14 | orangepower | lot of screens, sec |
| 18:27.19 | Naikrovek | k |
| 18:27.30 | orangepower | thanks guys, not trying to be annoying |
| 18:27.38 | paulc | orangepower: So back to basics. You need to create a peer in sip.conf, then get your phone to successfully register to Asterisk (the phone icon will be filled in, not hollow/outlined). Once done, we can look at dialplan issues. |
| 18:28.09 | orangepower | paulc: i can't configure it from the webinterface? or directly on the phone? |
| 18:28.48 | Naikrovek | orangepower: it is FAR easier to do it via (t)ftp |
| 18:29.06 | Naikrovek | i have some polycom config generation scripts that do everything for me except download firmware |
| 18:29.11 | orangepower | oh |
| 18:29.17 | Naikrovek | i'm happy to share them |
| 18:29.24 | Naikrovek | pm me your email and i'll send them |
| 18:31.13 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 18:32.16 | [TK]D-Fender | Naikrovek: ... this isn't a SoundPoint series model... |
| 18:32.33 | orangepower | VVX 1500 |
| 18:32.45 | orangepower | should be the sam econfig though? |
| 18:32.53 | [TK]D-Fender | orangepower: Doubt it |
| 18:33.14 | orangepower | ... |
| 18:33.46 | [TK]D-Fender | orangepower: Confirmed... it is the same. |
| 18:33.51 | [TK]D-Fender | orangepower: So that's good. |
| 18:34.05 | [TK]D-Fender | orangepower: SCREEN SHOTS PLEASE |
| 18:34.06 | orangepower | so * is setup correctly.. i can dial out, dial extensions.. everything from xmeeting (soft sip client) on the same network |
| 18:34.12 | paulc | orangepower: I like configuring through config files, cos I'm hardcore.. but web interface works too.. just reset it to factory defaults and get a simple registration to asterisk working. |
| 18:34.31 | orangepower | yea i'm gonna reset it now |
| 18:34.39 | orangepower | i got too far down the rabbit hole |
| 18:36.08 | [TK]D-Fender | orangepower: far from |
| 18:36.10 | vader-- | any of you guys using/used sipxecs? or any thoughts on it? |
| 18:41.51 | bahjons | http://pastebin.com/C09UmzrZ |
| 18:41.51 | bahjons | Here's my php code for starting the 'Originate' command with the asterisk manager. And the extensions context it's routing to. NoOp returns the set variables as blank. Am I doing something wrong? |
| 18:43.18 | [TK]D-Fender | bahjons: its SetVar, not Variable |
| 18:43.43 | [TK]D-Fender | oops |
| 18:43.45 | [TK]D-Fender | strike that |
| 18:43.48 | [TK]D-Fender | whong opt |
| 18:43.50 | [TK]D-Fender | wrong8 |
| 18:44.02 | Corydon76-dig | typo city |
| 18:44.09 | bahjons | haha, yea... |
| 18:44.10 | *** join/#asterisk korihor (~korihor@201.210.226.98) |
| 18:45.28 | [TK]D-Fender | bahjons: well... does your call actually originate? You don't seem to ahve the extr CRLF's between operations there |
| 18:45.58 | bahjons | yea, it originates |
| 18:51.18 | BCS-Satori | I am having a problem in 1.6.2.2 with call termination (hangup) when the person on the other end hangs up the phone (on SIP Locally, External Caller On SIP, External Caller on POTS, and DUNDI phones) where the local phone gets an immediate disconnect busy tone repeating until they physically. How can I make it auto hangup the local phones. |
| 18:51.39 | bahjons | hmm, if I add the extra CRLF's it doesn't originate |
| 18:52.42 | Jhirley | holla folks, any place I can look for find good write up on AIX vs SIP trunking ? |
| 18:52.57 | *** join/#asterisk Tim_Toady (~moi@193.92.197.188.dsl.dyn.forthnet.gr) |
| 18:53.28 | *** join/#asterisk Tim_Toady (~moi@193.92.197.188.dsl.dyn.forthnet.gr) |
| 18:53.49 | *** join/#asterisk Whtsup (~sssi@WimaxUser379-63.wateen.net) |
| 18:53.51 | [TK]D-Fender | Jhirley: What are you expecting to see? |
| 18:53.58 | Whtsup | hello |
| 18:54.10 | Whtsup | how can i test voice quality in asterisk |
| 18:54.44 | BCS-Satori | err Typo... I didn't mean DUNDI I meant DAHDI. sorry |
| 18:54.52 | paulc | Whtsup: Call someone and see how good it sounds? |
| 18:54.59 | Jhirley | [TK]D-Fender: looking for PRO / CONs , for IAX and SIP trunking. |
| 18:55.53 | paulc | stifles a giggle |
| 18:55.53 | Whtsup | i have check it its okay when i call |
| 18:55.53 | Whtsup | but when my client sent the traffic |
| 18:55.53 | [TK]D-Fender | Jhirley: Only pro for IAX is if you are in a SIP hostile networking scenario, or desperately need the bandwidth savings using IAX2 Trunk Mode |
| 18:55.53 | Whtsup | calls are disconnected |
| 18:57.16 | paulc | Whtsup: Inbound or outbound? Does the log shed any light on it? (I'm not sure how this relates to voice quality?) |
| 18:57.27 | Whtsup | outbound |
| 18:57.59 | Whtsup | latency from my client switch to my asterisk server is 140ms |
| 18:58.07 | paulc | So we're talking: Your Client --> Asterisk --> Your Provider - and the call fails immediately? after a while? |
| 18:58.15 | Whtsup | yes |
| 18:58.24 | Whtsup | my acd is not good |
| 18:58.36 | paulc | Does the call get setup, proceed, and fail after a while? or immediately when they try and place a call? |
| 18:58.54 | Whtsup | call are answering |
| 18:59.06 | Whtsup | but average call duration is very low |
| 18:59.50 | *** join/#asterisk aandrade (~aandrade@189.58.128.179) |
| 19:00.04 | paulc | So the question is - why are calls ending? |
| 19:00.08 | Whtsup | yes |
| 19:00.12 | paulc | Are they hearing silence and hanging up? |
| 19:00.38 | Whtsup | issue is delay in voice |
| 19:00.42 | *** join/#asterisk [8none1] (~8none1]@ps14528.dreamhost.com) |
| 19:08.36 | *** join/#asterisk devafree (~kannan@58.68.68.26) |
| 19:09.22 | *** join/#asterisk rare1980_ (~rare1980@203.175.76.218) |
| 19:09.26 | *** join/#asterisk jpvoip (~jpvoip@201-34-141-34.fnsce704.e.brasiltelecom.net.br) |
| 19:10.08 | jpvoip | hello guys, someone has any idea of a good project using Asterisk + XMPP? Is for my course final project |
| 19:10.31 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:10.36 | devafree | hello, i am running asterisk 1.4. Client eyebeam softphones on the LAN on ulaw, periodically get 408 request timeout registration errors, this happens on over the last couple days. But aftre a couple times re-starting the eyebeam phone, it works OK again for a whileHow can I start trouble shooting |
| 19:11.00 | Corydon76-dig | jpvoip: you realize the driver is already written, right? |
| 19:12.06 | _Raptor_ | i want the user to enter an extension containing a various number of digits and wenn he presses # the dial should be executed right away. exten => _[0-9].#,1,Dial(...) waits for timeout because # matches to . as well |
| 19:12.30 | *** join/#asterisk cguerrero (~cuauhtemo@200.79.231.94) |
| 19:12.37 | _Raptor_ | so how can i express the regexp ^[0-9]+#$ in extensions? |
| 19:15.06 | jaskew | devafree: try SIP DEBUG PEER <name of peer or ip address> and then see if asterisk is seeing the registrations. Also, Eyebeam has a debug log console. you might try using both of these at the same time |
| 19:15.23 | devafree | jaskew , ok thanks , i will do |
| 19:15.53 | jaskew | devafree: hang on - my syntqax is wrong |
| 19:16.22 | jaskew | devafree: SIP SET DEBUG PEER <name or ip address> I think that is right :) |
| 19:16.52 | devafree | jaskew , ok got i t:) |
| 19:16.57 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:16.59 | jaskew | _Raptor_: what kind of phone - is it a SIP phone or POTS? |
| 19:17.33 | devafree | the prob is i got at least 50 -60 calls running all the time, i have to be able to read the CLI logging , :( |
| 19:17.38 | *** join/#asterisk jpvoip (~jpvoip@201-34-141-34.fnsce704.e.brasiltelecom.net.br) |
| 19:17.40 | [TK]D-Fender | raptYou can't do this directly. there is no "use X as a terminator" |
| 19:17.47 | jpvoip | Corydon76-dig: yes |
| 19:17.47 | [TK]D-Fender | _Raptor_: You can't do this directly. there is no "use X as a terminator" |
| 19:18.09 | _Raptor_ | [TK]D-Fender: what do you suggest? |
| 19:18.21 | jpvoip | Corydon76-dig: im looking for a use of * +XMPP... |
| 19:18.28 | _Raptor_ | jaskew: various phones, sip, iax, ... |
| 19:18.38 | *** join/#asterisk M1s3ry (~M1s3ry@76.164.165.1) |
| 19:18.45 | jaskew | yeah - I was going to direct him to the phone's own dialplan if it was a SIP phone. |
| 19:18.57 | [TK]D-Fender | _Raptor_: you'll need to do a bigger pattern with "." at the end and check that it ends with a # in the dialplan. This means you can't just 404 a dialed # as you have to accept it first then check if ti ends right |
| 19:19.16 | jaskew | _Raptor_: TKD-Fender knows more than me - I'll let him take it :) |
| 19:19.17 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 19:19.28 | _Raptor_ | jaskew: thx :-) |
| 19:19.42 | *** join/#asterisk mayfield (~mayfaild@76-250-152-224.lightspeed.snantx.sbcglobal.net) |
| 19:20.18 | _Raptor_ | [TK]D-Fender: my problem also is the timeout. i don't want to wait for the timeout but setting off the call wenn i press # |
| 19:20.18 | leifmadsen | jpvoip: I think I have a blog post written about Asterisk and XMPP at http://leifmadsen.wordpress.com |
| 19:20.29 | _Raptor_ | (thats the only reason for #) |
| 19:20.57 | jaskew | [TK]D-Fender: Doesn't it have to be set in the phone itself so the phone knows when to send the INCITE? |
| 19:21.07 | jaskew | s/INCITE/INVITE |
| 19:21.11 | leifmadsen | INCITE VIOLENCE?!?!?! |
| 19:21.27 | jaskew | why doesn't that s trick work for me? |
| 19:21.33 | [TK]D-Fender | jaskew: depends if you want to use the phone to do the limiting.... thats poor methodology if you're in a mixed environment... |
| 19:21.42 | *** join/#asterisk mayfield (~mayfaild@76-250-152-224.lightspeed.snantx.sbcglobal.net) |
| 19:22.02 | jaskew | So timeout is the way to go in those situations? |
| 19:23.14 | devafree | ok , i think i may have got a solution, not asure. Asterisk global settings has nat=yes, but the eyebeam phones have got no specific nat entry in sip-something-additional.conf (auto gen by a script from a MySQL DB app). The asterisk has 2 IP addresses , one to connect to the service provider on eth0, and the other connect to the LAN on eth1 (the LAN has the eyebeam phones). Can this possibly cause the error? |
| 19:23.35 | devafree | i get intermittent 408 request timeouts, that resolve itself after a bit |
| 19:24.23 | *** join/#asterisk V4mpire (~gary@82.118.111.254) |
| 19:24.55 | bmoraca_work | devafree: you're gonna need to try that in #freepbx . but, why not add nat=yes and try? |
| 19:25.15 | jaskew | davfree: do you have a localnet setting in sip.conf? |
| 19:25.33 | devafree | jaskew , -> yep |
| 19:26.17 | devafree | bmoraca_work , i will, i will edit sip.conf manually and see if it recurs |
| 19:26.27 | devafree | its not fpbx |
| 19:27.26 | *** join/#asterisk Tech_Travis (~tech_trav@208.179.137.131) |
| 19:30.03 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 19:30.34 | Katty | hi |
| 19:30.53 | Katty | my asterisk does not work at all |
| 19:30.55 | Katty | how to fix pls |
| 19:31.09 | Sargun | haha |
| 19:31.14 | paulc | Katty: paste your whole config file to the channel and we'll alllll have a look see |
| 19:31.14 | wcselby | lol |
| 19:31.18 | Sargun | please tell me you're joking. |
| 19:31.24 | wcselby | katty never jokes |
| 19:31.28 | Katty | NEVAR |
| 19:31.29 | jaskew | Hi Katty. your menagerie is larger than mine I'm afraid! |
| 19:31.54 | Katty | jaskew: :P |
| 19:32.02 | Katty | jaskew: yesh, i live in a zoo |
| 19:32.19 | Katty | jaskew: and it'll be bigger if i can find a house outside city limits! |
| 19:32.26 | Katty | :>>>>>>>>>>>>>>>>>>>>>>>>> |
| 19:32.46 | *** part/#asterisk M1s3ry (~M1s3ry@76.164.165.1) |
| 19:32.57 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.179.173.dsl.dyn.forthnet.gr) |
| 19:33.00 | *** join/#asterisk M1s3ry (~M1s3ry@76.164.165.1) |
| 19:33.14 | paulc | jaskew said "menagerie" and I thought "lingerie" - hmm.. twisted much? |
| 19:33.59 | Katty | weird. |
| 19:34.30 | Katty | hopefully outside city limits i will get a pair of pygmy goats and a couple chickens. |
| 19:34.32 | M1s3ry | I have a dialplan where I need to put caller in meetme that create dynamically (this works fine) |
| 19:34.32 | M1s3ry | Then I need to call another number and put them in that same meetme. It seems to stop the dial plan once it goes in Meetme. Is there a way around this without AGI script. Using the dialplan in => http://pastebin.com/NR1KWBDk |
| 19:34.34 | jaskew | paulc: they are quite different things, but who am I to judge the associations of others |
| 19:34.41 | Katty | maybe a duck or two |
| 19:35.15 | hardwire | it maeks me think margarine or menage or menengitis or Ménage à trois |
| 19:35.29 | hardwire | err.. meh |
| 19:35.30 | hardwire | anyways |
| 19:35.32 | hardwire | tired |
| 19:35.39 | jaskew | is anatidaephobic |
| 19:35.52 | [TK]D-Fender | M1s3ry: where do we see the failed call? |
| 19:36.16 | hardwire | askew is a cool name |
| 19:36.23 | M1s3ry | in the the copied CLI which will be pastebin'd shortly :) |
| 19:36.25 | hardwire | not a lot of V* in your lineage. |
| 19:36.27 | hardwire | V8 |
| 19:36.31 | [TK]D-Fender | M1s3ry: and you can't just call dial after Meetme... this is a SINGlE CALL you are processing there. You need to ORIGINATE a new call which will be technically independent of this one |
| 19:36.45 | [TK]D-Fender | M1s3ry: what you have now has no chance. |
| 19:36.53 | jaskew | heh - I have fun with it. |
| 19:37.00 | Katty | jaskew: MERRY IS AWAKE |
| 19:37.19 | paulc | I had to look the phobia up. Makes me laugh. |
| 19:37.22 | paulc | Makes me think of http://www.threadless.com/product/1281/Scare_List |
| 19:37.24 | Katty | anddd he's awake again |
| 19:37.25 | paulc | awesome t-shirt |
| 19:37.26 | Katty | err asleep |
| 19:37.35 | [TK]D-Fender | M1s3ry: Look up "call files" , use Originate() from * 1.6+, or an AMI Originate to span this secondary call PRIOR to sending this caller into the MeetMe |
| 19:37.36 | Katty | ohohwaitwait |
| 19:37.43 | Katty | jaskew: http://ustre.am/bEBU |
| 19:37.52 | Katty | jaskew: you might be too late tho |
| 19:37.57 | jaskew | http://n4.nabble.com/file/n277836/anatidaephobia.png |
| 19:39.22 | Katty | you ever wonder why some animals sleep in a pile, and others dont? |
| 19:39.24 | *** join/#asterisk DMeloUK (~DominicMe@64.129.95.226) |
| 19:39.58 | jaskew | don't knock it 'til you've tried it! |
| 19:40.17 | Katty | i don't think i'd want someone crushing a rib |
| 19:40.27 | Katty | but the boys don't seem to mind |
| 19:40.53 | jaskew | Merry crashed my firefox! |
| 19:40.58 | Katty | lol |
| 19:41.31 | mayfield | how are things in the asterisks world? |
| 19:41.32 | Katty | jaskew: http://www.ustream.tv/channel-popup/the-nut-house-bird-bath |
| 19:41.48 | Katty | jaskew: try that one instead. less going on in the page to crash your browser |
| 19:41.50 | jaskew | I shoulda closed the window with the duck. There it is again! |
| 19:41.59 | wcselby | omg |
| 19:42.05 | mayfield | most importantly the communications of remote sip endpoints and nat traversing issues with rtp traffic =P |
| 19:42.08 | wcselby | my client just sent out a warning notice about a virus |
| 19:42.14 | wcselby | in it, they included a link to the virus |
| 19:42.27 | Katty | jaskew: merrys already gone back to sleep but pippin seems to be awake. |
| 19:42.28 | wcselby | they sent it to like 3000 people |
| 19:42.37 | Katty | wcselby: oh boy :/ |
| 19:42.43 | wcselby | lol oh well... |
| 19:43.11 | wcselby | I have a feeling their support queue is about to heat up.... |
| 19:43.17 | wcselby | anyways, time to grab some lunch |
| 19:43.20 | Katty | LOL |
| 19:43.24 | Katty | good timing! get out of there! :P |
| 19:43.32 | leifmadsen | wcselby: ya, get the hell outta dodge :) |
| 19:43.45 | jaskew | Me too (luinch). OK - I see movement |
| 19:43.58 | Katty | jaskew: they got up for a snack |
| 19:44.06 | Katty | jaskew: now they're grooming |
| 19:44.18 | Katty | jaskew: merry is on th eleft |
| 19:45.22 | orangepower | SO CLOSE to getting the VVX1500 working, i have outgoing audio, just no incoming audio |
| 19:45.30 | orangepower | is that a firewall problem? |
| 19:45.50 | Katty | could be. |
| 19:45.55 | Katty | check your firewall logs. |
| 19:46.38 | paulc | orangepower: is it behind NAT, compared to your * box? |
| 19:47.15 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 19:47.15 | *** join/#asterisk aandrade (~aandrade@189.58.128.179) |
| 19:47.25 | orangepower | both are behind NAT |
| 19:47.28 | orangepower | :( |
| 19:47.36 | Katty | logs. |
| 19:47.38 | Katty | are. |
| 19:47.38 | orangepower | audio between extensions works fine |
| 19:47.39 | Katty | useful. |
| 19:47.46 | orangepower | yea |
| 19:47.56 | orangepower | just gonna open it as the dmz, see if that fixes the issue |
| 19:48.04 | orangepower | then figure it out |
| 19:48.05 | jblack | no they're not. You should give out tiny bits of info at a time. |
| 19:48.17 | paulc | character by character? |
| 19:48.24 | Katty | foooo |
| 19:48.26 | Katty | barrrr |
| 19:48.27 | Katty | eeed. |
| 19:48.31 | jblack | make it like a mystery novel. Make everyone guess as to your problem! |
| 19:48.46 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 19:48.48 | paulc | barrrr makes me think of the olympic curlers, when they shout hard! Hard! HARRRRRRD! |
| 19:51.57 | [TK]D-Fender | orangepower: READ... |
| 19:51.58 | [TK]D-Fender | ~sipnat |
| 19:51.59 | 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 |
| 19:52.01 | [TK]D-Fender | ^^^^^^^^^^^^^6 |
| 19:54.04 | *** join/#asterisk sulex (~sulex@host-78-14-173-189.cust-adsl.tiscali.it) |
| 19:55.43 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:55.51 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 19:58.03 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 19:58.37 | mayfield | damn, that is a nice link ^^ |
| 19:58.39 | Katty | sighs at clock |
| 19:58.56 | orangepower | IT WORKS omg yes |
| 19:58.58 | orangepower | finally! |
| 19:59.14 | Katty | nifty |
| 19:59.54 | orangepower | Naikrovek: thanks you sooo much for those scripts and the help.. it's perfect |
| 20:00.03 | Naikrovek | ah you're welcome |
| 20:00.06 | Naikrovek | glad i could help |
| 20:00.21 | Naikrovek | so much easier if you do it via FTP and config files |
| 20:00.34 | Naikrovek | you can store that stuff in cvs or whatever and put your feet up |
| 20:00.47 | orangepower | yea |
| 20:03.57 | xLP | I'm encoounterting troubles with Asterisk, no clue why, I can only describe symptoms... Randomly, when you try placing a call, it fails, and some phones will fail registering. I sniffed network, while watching asterisk console and monitoring box load... Load is always around 0.04, I see packets incoming and being replied normally, and when the problem starts, I see incoming packets, but no answer, and nothing in the console, then, suddenly, after about 30 s |
| 20:04.05 | *** join/#asterisk zeyui (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net) |
| 20:04.09 | zeyui | hi there |
| 20:04.16 | zeyui | i just start with asterisk |
| 20:04.30 | zeyui | i create an sip user |
| 20:04.55 | zeyui | when i try to check the user with a softphone |
| 20:05.02 | zeyui | it not sign is it normal ? |
| 20:05.09 | Katty | zeyui: read the book. |
| 20:05.11 | Katty | infobot: thebook |
| 20:05.12 | infobot | extra, extra, read all about it, thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 20:05.16 | [TK]D-Fender | xLP: why don't WE see these incoming packets and debug? You've been repeating that all day and shown us nothing. |
| 20:05.17 | mayfield | authentication issue maybe? |
| 20:05.23 | *** join/#asterisk aandrade (~aandrade@189.58.128.179) |
| 20:05.45 | mayfield | xlp are these remote end points that are experiencing these problems? |
| 20:05.49 | xLP | zeyui: you're probably French, and you probably means "ring"... and I fear it's a bit harder... you need to route the calls |
| 20:05.54 | zeyui | i got this Name/username Host Dyn Nat ACL Port Status |
| 20:05.55 | zeyui | ivan/ivan (Unspecified) D 5060 Unmonitored |
| 20:05.57 | zeyui | 1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0 offline] |
| 20:06.12 | zeyui | when i do sip show peer |
| 20:06.19 | Katty | read. |
| 20:06.21 | Katty | the. |
| 20:06.22 | Katty | book. |
| 20:06.56 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 20:07.29 | mayfield | zeyui thats one part of the puzzle.. double check the config on your soft phone. |
| 20:07.33 | xLP | [TK]D-Fender: what would you like to see? I can run a capture and wait for problems to happen... But it's just everything normal, and I just expect someone to have encountered it... else I think no change. You can consider everything is running as usual, just for 30 seconds it stops answering, and 30s after everything runs as normal |
| 20:07.53 | xLP | mayfield: they are remote, over a VPN |
| 20:08.14 | mayfield | what type of filesystem/disk configuration? |
| 20:08.18 | mayfield | on the * box |
| 20:08.33 | xLP | mayfield: ext3, no particular info on the HDD itself (must be IDE) |
| 20:08.34 | [TK]D-Fender | xLP: We have no veriosn, and no usable details at all from your side. We have nothing to help you with |
| 20:09.51 | xLP | [TK]D-Fender: Asterisk 1.6.0.6, what else would be usable data? |
| 20:10.04 | mayfield | xlp, only thing i can think of is variable latency causing the session traffic/vpn tunnel to linger then resume |
| 20:10.20 | mayfield | xlp, like fender said |
| 20:10.38 | xLP | mayfield: sorry, forgot... many endpoints are local (same switch) and encounter the same problem :( |
| 20:10.45 | [TK]D-Fender | xLP: Well first you are 18 releases off of current.... |
| 20:11.01 | [TK]D-Fender | xLP: First UPGRADE... then we'll need to see a full dump of call attempts |
| 20:11.19 | xLP | [TK]D-Fender: ok, I'll then first look at how to upgrade, thx |
| 20:11.44 | xLP | [TK]D-Fender: what do you call a full dump of call attempts? would a Wireshark capture suit? |
| 20:12.05 | mayfield | you know that spam you see in the asterisks console? |
| 20:12.17 | mayfield | up the verbosity |
| 20:12.46 | xLP | I'm at level 4 (sometimes at 28), but I read nothing output at more than 4 |
| 20:12.48 | [TK]D-Fender | xLP: * SIP DEBUG |
| 20:12.49 | mayfield | a tcpdump off the wan or whatever interface you are using for 5060/10k-20k blah blah traffic |
| 20:13.23 | mayfield | those to go hand in hand togethor ;) |
| 20:13.30 | xLP | RTP & SIP, no pb... ok well, will first try upgrading then I'll see |
| 20:13.38 | mayfield | yes |
| 20:13.41 | mayfield | def.. |
| 20:13.49 | mayfield | 1.6 *face palm* ...sad story. |
| 20:13.53 | mayfield | :P |
| 20:14.25 | xLP | all I can tell for now regarding output in asterisk console is... after 30 secs, for example, invites are processed JUST NORMALLY... |
| 20:14.46 | xLP | ok, thx to u 2... I'll go find how to update... |
| 20:14.50 | mayfield | one sec xlp. |
| 20:15.00 | xLP | mah, I stay anyway :) |
| 20:15.28 | mayfield | isn't there like a variable that you can toggle for the ttl on the sip notifications to the pbx? |
| 20:15.40 | mayfield | i wonder if your flooding the pbx with the keep alives |
| 20:15.48 | mayfield | from the remote end points |
| 20:15.52 | xLP | interesting |
| 20:15.58 | mayfield | how many end points? |
| 20:16.00 | xLP | I'll dig there |
| 20:16.06 | xLP | mmm let me check |
| 20:16.16 | leifmadsen | qualify=<time_in_ms> |
| 20:16.21 | mayfield | yea, qualify |
| 20:16.33 | xLP | (by remote, you mean any endpoint outside of the box, or outside LAN ?) |
| 20:16.43 | xLP | I disabled qualitfy as I suspected it |
| 20:16.48 | xLP | it didn't seem to help |
| 20:17.03 | mayfield | try hard setting that value to a reasonable value |
| 20:17.08 | mayfield | dunno what the default value is. |
| 20:17.23 | xLP | even if I completely disabled it? (well, I mean no endpoint is being qualified) |
| 20:17.29 | mayfield | install bwm |
| 20:17.34 | xLP | bwm? |
| 20:17.37 | mayfield | and monitor the traffic on your network interfaces |
| 20:17.45 | mayfield | on the pbx |
| 20:17.55 | mayfield | banwidth monitor |
| 20:17.58 | xLP | dunno bwm, I'm monitor thru a managed switch |
| 20:18.01 | mayfield | gogle it |
| 20:18.09 | mayfield | its a app you install on your * box |
| 20:18.12 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:18.34 | mayfield | bwm-ng |
| 20:18.49 | Naikrovek | really |
| 20:18.53 | Naikrovek | perks up |
| 20:18.54 | mayfield | if your sing centos/rehl yum install bwm-ng |
| 20:18.56 | xLP | I will, yes yes... just doesn't seem related to bandwidth... the box is not getting anything else than SIP traffic, which is low, and LAN endpoints are affected too |
| 20:19.26 | xLP | package not available, guess it's outdated |
| 20:19.30 | mayfield | lies |
| 20:19.31 | Naikrovek | yeah same |
| 20:19.33 | Naikrovek | lol |
| 20:19.40 | Naikrovek | No package bwm-ng available. |
| 20:19.41 | mayfield | update your repos :P |
| 20:19.42 | xLP | thx for your help, don't waste your time, I'll try upgrading first |
| 20:19.58 | Katty | bored. |
| 20:20.01 | Katty | BORED. |
| 20:20.04 | Katty | bored bored bored. |
| 20:20.14 | mayfield | http://sourceforge.net/projects/bwmng/ |
| 20:20.17 | Naikrovek | so do something |
| 20:20.33 | mayfield | man, i havn't been in here for a long time. |
| 20:20.39 | Katty | Naikrovek: like what |
| 20:20.40 | mayfield | fender has ben in here for years.... |
| 20:20.48 | Katty | mayfield: so have i |
| 20:20.58 | mayfield | yea i recognize the name |
| 20:21.03 | Qwell | speaking of [TK]D-Fender.. nice @, sir :D |
| 20:21.10 | Katty | QWELL |
| 20:21.13 | Katty | did your package arrive. |
| 20:21.17 | Qwell | Katty: YES! Friday. |
| 20:21.20 | Qwell | but I've been sick. |
| 20:21.20 | Katty | :> |
| 20:21.22 | Katty | :< |
| 20:21.30 | Katty | i am sorry to hear of illness. |
| 20:21.43 | Qwell | Friday for reals this time. |
| 20:21.56 | Katty | k |
| 20:22.47 | [TK]D-Fender | Qwell: Yeah, we had a flooder :) |
| 20:22.54 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 20:23.18 | [TK]D-Fender | Qwell: those are 95% of my reasons for "upgrading" :) |
| 20:23.35 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 20:26.27 | *** join/#asterisk Alagar (~Administr@122.164.34.12) |
| 20:28.35 | ariel_ | I have a strange issue which is with iax2 and using it with trunk=yes. It works, but when I do actual data anaiyes it only trunks one way. Has anyone done any testing with this. |
| 20:32.24 | *** part/#asterisk bahjons (~robert@140.99.23.26) |
| 20:33.41 | *** join/#asterisk linelevel (~ticktackt@208.65.172.155) |
| 20:36.20 | *** join/#asterisk lost_soul (~noymfb@cpe-67-241-68-202.twcny.res.rr.com) |
| 20:38.13 | linelevel | hi guys. i'm running asterisk with the trixbox GUI. In the CDR logs, I have fields for the src channel and dest channel. The records for the source channel look like (e.g.) 'SIP/603-b770ed50', and an example dest channel is 'Zap/2-1'. Can anyone explain what each of these is? (I know what SIP is, and I know that the '603' is the extension #, but that's where my knowledge ends.) |
| 20:39.32 | ariel_ | means your sip extension 603 made a call via the pstn line on channel 2 |
| 20:42.22 | orangepower | so i've tried all the firewall config's port forwarding everything, i can make calls out, but i get the "disconnected" tones when i call in |
| 20:42.40 | orangepower | asterisk has no log of the events |
| 20:44.22 | linelevel | ariel_, thanks. what is the 'b770ed50'? I know it's some 4-byte hex code, but what does it represent? I've noticed that it is sometimes the same for different extensions, but not always the same for a fixed extension. |
| 20:45.09 | Qwell | linelevel: That is a unique ID for the channel. It is essentially random. |
| 20:45.28 | Qwell | (really, for a SIP call, I believe it's the address of the pointer for the SIP pvt, but...consider it "random") |
| 20:46.05 | Deeewayne | randomizes Qwell's face |
| 20:46.30 | linelevel | Qwell, thanks. This org has multiple copper phone lines and I'm trying to determine which one this call went out on. Is that possible from this data? |
| 20:46.42 | Qwell | Deeewayne: Hey, I'm a picasso! |
| 20:47.04 | Qwell | linelevel: Zap/2 |
| 20:49.19 | linelevel | Qwell, so that means it's line 2? What is the '-1' in 'Zap/2-1'? Also, might it be the case that the line that Asterisk calls Zap/2 is mapped to Line1 on the org's phone? |
| 20:49.54 | leifmadsen | Qwell: any chance you could sign those releases for me? |
| 20:49.55 | [TK]D-Fender | LinNo, just ignore that |
| 20:50.01 | [TK]D-Fender | linelevel: No, just ignore that |
| 20:50.38 | linelevel | [TK]D-Fender, ignore what? |
| 20:51.36 | Deeewayne | leifmadsen, can you ping me after your release stuffs, please ? |
| 20:52.36 | leifmadsen | Deeewayne: ping :) |
| 20:52.52 | leifmadsen | I'm just waiting on people at this point :\ |
| 20:53.04 | wcselby | new releases coming out soon? |
| 20:53.29 | [TK]D-Fender | linelevel: ignore the "-1" |
| 20:53.44 | linelevel | [TK]D-Fender, are you telling me to ignore what Qwell said, or to ignore the '-1' part of that line? |
| 20:53.50 | linelevel | ah ok |
| 20:54.09 | linelevel | [TK]D-Fender, know about my other question: might it be the case that the line that Asterisk calls Zap/2 is mapped to Line1 on the org's phone? |
| 20:54.28 | wcselby | all it means is the call is going out on whichever line you've got plugged into your zap/2 interface |
| 20:54.28 | linelevel | and in particular, where could i find such info? |
| 20:54.33 | [TK]D-Fender | linelevel: There is no such thing as "mapping" |
| 20:54.39 | wcselby | physically plugged into the zap/2 interface |
| 20:55.33 | *** join/#asterisk norrec (~norrec@76-201-85-140.lightspeed.frokca.sbcglobal.net) |
| 20:55.35 | linelevel | so would i need to physically visit there to find that out? Currently, I'm just using the trixbox web interface remotely |
| 20:55.41 | [TK]D-Fender | linelevel: the "-1" has no functional meaning to you. there is no association between ANY 2 devices in your system. A call from any device hits the dialplan and that does whatever it does with it |
| 20:55.51 | ManxPower-work | linelevel: "mapping" is a feature of key systems, not of PBXs |
| 20:55.52 | *** join/#asterisk jameswf (~james@unaffiliated/jameswf-home) |
| 20:57.14 | jameswf | I want to save a bunch of money on a phone system by installing asterisk on this old P4 layin here but i dont wanna learn linux or how to program it can someone doo it all for me for free or cheaper |
| 20:58.04 | *** join/#asterisk jmacz (~jmacz@190.25.7.33) |
| 20:58.24 | Qwell | jameswf: Sure, just enter your credit card number here, and I'll get right on that. |
| 20:58.39 | Qwell | oh don't worry, we won't charge it. It's just for proof of...umm...something. |
| 20:59.12 | Deeewayne | proof of silliness |
| 21:00.23 | mayfield | jameswf sure |
| 21:01.18 | mayfield | jameswf but i am gonna need you to dig around in my pants pocket for the duration of the job. |
| 21:01.57 | Qwell | mayfield: ...wow. |
| 21:02.24 | *** join/#asterisk jpvoip (~jpvoip@201-34-141-34.fnsce704.e.brasiltelecom.net.br) |
| 21:03.10 | mayfield | Qwell quality work does not come cheap my friend. |
| 21:05.47 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 21:06.01 | *** join/#asterisk lost_soul (~noymfb@cpe-67-241-68-202.twcny.res.rr.com) |
| 21:08.23 | jameswf | but it does come easy :) |
| 21:08.55 | mayfield | jameswf we'll get along just fine. |
| 21:09.38 | wcselby | so, if I use Dial(DAHDI/G1/${NUMBER}, on a dahdi span that has 23 channels in the group, will it start at channel 23 in the group or at channel 1? And isn't this controllable by using DAHDI/G1 vs DAHDI/g1 ? Or do I have my letters mixed up? |
| 21:09.42 | mayfield | in all honesty.. installing linux & asterisk is not uberly complicated. just timely. =P |
| 21:09.50 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 21:10.03 | jameswf | seems all the forum post etc have revolved around It doesnt work and I dont wanna learn how to make it work, paying someone to work on FREE software that is insane..... I know how postal workers feel |
| 21:10.08 | mayfield | jameswf don't be discouraged. just prepare yourself for total isolation from humanity for a drawn out finite amount of time. |
| 21:10.29 | [TK]D-Fender | wcselby: g = ascending, G=descending |
| 21:10.33 | paulc | LOL - yeah, it's not THAT hard.. you can trade time for control. |
| 21:10.37 | wcselby | jameswf - there are some guides out there that are quite useful |
| 21:11.11 | paulc | PBX in a flash etc = drop the CD in, reboot, come back in 15 minutes done.. but at a cost of a webby front end, and not being able to totally hack the config easily/fully |
| 21:11.23 | mayfield | exactly |
| 21:11.28 | paulc | or go with a default distro install, build Asterisk from source, and tweak the sample config files |
| 21:11.35 | paulc | I prefer the latter way, cos I like being in control |
| 21:11.40 | jameswf | wcselby, it was a joke I am a well versed Asterisk Jedi :) |
| 21:11.47 | mayfield | plus those config files are all bloaty |
| 21:11.52 | mayfield | on the prebuilt installs |
| 21:12.01 | mayfield | lots of linked config files |
| 21:12.06 | wcselby | [TK]D-Fender - thanks! how do I control inbound calls on the dahdi channel? i.e I want my outbound calls to start on channel 23, my inbound to start on channel 1? So outbound calls use DAHDI/G1, where do I make the setting for inbound call control? |
| 21:12.14 | KavanS | would anyone be able to tell me why DTMF does not work with SIP provider reliably? |
| 21:12.18 | KavanS | using codec ulaw |
| 21:12.18 | [TK]D-Fender | wcselby: You don't, the telco does |
| 21:12.25 | wcselby | gotcha |
| 21:12.52 | jameswf | ~jameswf |
| 21:12.52 | infobot | jameswf loves unsolicited technical support, or http://jameswf.info |
| 21:13.09 | jameswf | infobot forget jameswf |
| 21:13.09 | infobot | jameswf: i forgot jameswf |
| 21:13.11 | wcselby | and jameswf - :) |
| 21:13.11 | KavanS | server not found |
| 21:13.14 | KavanS | lol |
| 21:13.24 | *** join/#asterisk jmacz (~jmacz@190.25.7.33) |
| 21:14.37 | mayfield | kavans, its quite simple. |
| 21:15.14 | KavanS | mayfield, ok... |
| 21:15.26 | KavanS | the problem I am having is intermittent |
| 21:15.31 | KavanS | provider tells us it's not them (of course) |
| 21:15.37 | mayfield | lies |
| 21:15.38 | KavanS | so trying to track down what the issue is on our side... |
| 21:15.48 | KavanS | so you think it is provider related? |
| 21:15.54 | mayfield | SIP/VoIP trasversion DTFM tones |
| 21:16.01 | mayfield | transversing |
| 21:16.19 | mayfield | the digitized audio degrades |
| 21:16.40 | mayfield | it is harder to detect dtfm after multiple points of encoding/decoding |
| 21:16.42 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 21:16.47 | mayfield | need for jigga watts :P |
| 21:17.00 | [TK]D-Fender | Jigga whats? |
| 21:17.09 | mayfield | need more jigga watts : |
| 21:17.16 | mayfield | power! |
| 21:17.24 | wcselby | 1.2 jiggawatts |
| 21:17.25 | [TK]D-Fender | hordes his pre-release version of res_fluxcapacitor.so |
| 21:17.28 | wcselby | all that's needed for time travel |
| 21:17.37 | mayfield | fender omg.. plz xdcc send |
| 21:17.40 | wcselby | assuming you have a flux-capacitor laying around somewhere |
| 21:17.51 | wcselby | bah |
| 21:17.58 | wcselby | [TK]D-Fender beat me too it |
| 21:17.59 | mayfield | but is this not the truth? |
| 21:18.24 | mayfield | or did i type alot of jibberish =P |
| 21:18.51 | *** join/#asterisk jmacz (~jmacz@190.25.7.33) |
| 21:20.03 | [TK]D-Fender | mayfield: http://tinyurl.com/yow2q8 |
| 21:20.23 | mayfield | i see what you did there. |
| 21:20.25 | paulc | Was it jiggawats or gigawatts |
| 21:20.36 | [TK]D-Fender | paulc: former |
| 21:20.50 | [TK]D-Fender | paulc: and has been adopted as a proper official spelling as well |
| 21:21.21 | paulc | stands corrected (I lose my inner geek sometimes - it's the stress of the day job) |
| 21:21.56 | mayfield | i was trying to explain why dtmf does not work with sip gateway providers =P |
| 21:21.59 | mayfield | i failed obviously |
| 21:23.06 | *** join/#asterisk aceking5 (~aceking5@71-94-132-102.static.mtpk.ca.charter.com) |
| 21:26.38 | TheDavidFactor | what does asterisk issue a (re)invite when it starts to try to receive a fax? |
| 21:27.00 | wcselby | TheDavidFactor - because it's transitioning from ulaw to t38 |
| 21:27.18 | [TK]D-Fender | checkout time, later all |
| 21:27.21 | TheDavidFactor | ok thanks. |
| 21:27.47 | geneticx_wrk | hello everyone. I'm installing asterisk and was wondering what the adviced or more common packages should I install under "core sound packages" and "MOH file packages" ? any advice |
| 21:31.08 | geneticx_wrk | can I install all of them? |
| 21:32.00 | Qwell | geneticx_wrk: the defaults are fine for most installs |
| 21:32.20 | Qwell | However, yes, you *can* download all of them. |
| 21:32.56 | Qwell | (in most cases, you would only ever use 1-2 though) |
| 21:32.56 | wcselby | i tend to download the .gsm, the .ulaw, and the .g729 versions |
| 21:33.01 | wcselby | of the ones I want |
| 21:33.02 | Qwell | ^^ what he said |
| 21:33.12 | *** part/#asterisk mick_laptop (~mick@clamwin/admin/mickhome) |
| 21:33.14 | Qwell | just depends on what codecs your endpoints will be using |
| 21:33.44 | wcselby | and I only download the -EN ones, because I have an english-based system |
| 21:34.42 | leifmadsen | I tend to just install .ulaw and .wav for core, moh, and extras |
| 21:35.05 | leifmadsen | but I also don't use G.729 in my network :) |
| 21:37.13 | geneticx_wrk | cool. Thanks guys |
| 21:37.29 | Naikrovek | people with bandwidth don't need g729 |
| 21:37.35 | Naikrovek | i have a flippin' T1 that i have to live with |
| 21:37.51 | Naikrovek | boss seems to think one can have unlimited calls go across a T1 |
| 21:38.25 | redax | shall I upgrade my AST 1.6.0.4 to 1.6.2.X or simply use the latest 1.6.0.X ? |
| 21:40.31 | Naikrovek | redax: 1.6.0 and 1.6.2 are different branches, just like 1.4 and 1.6.0 are different |
| 21:40.55 | *** join/#asterisk Arsenick (~y@modemcable022.82-21-96.mc.videotron.ca) |
| 21:41.55 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 21:43.18 | *** part/#asterisk mayfield (~mayfaild@76-250-152-224.lightspeed.snantx.sbcglobal.net) |
| 21:43.50 | *** join/#asterisk mayfield (~mayfield@cosmic.sized.penisinyourface.com) |
| 21:44.15 | wcselby | in fact, it's two branches apart, sort of like 1.2 - 1.6.0 |
| 21:45.39 | redax | Naikrovek: I understand they're different, but is it worth to upgrade? |
| 21:45.51 | Naikrovek | does your phone system work properly? |
| 21:45.56 | Naikrovek | does it do what you need? |
| 21:46.16 | Naikrovek | if not, does 1.6.1 or 1.6.2 offer features you do need? |
| 21:46.21 | redax | actually no :) |
| 21:46.39 | redax | there's something wrong with the Transfer |
| 21:46.42 | Katty | does anyone know how to cut a hole in a plastic storage container? |
| 21:46.50 | Naikrovek | Katty: like a tote? |
| 21:46.57 | Katty | like a rubbermaid |
| 21:46.59 | redax | call transfer, when calls coming from the SIP Trunk... |
| 21:47.08 | Katty | Naikrovek: no sharp edges |
| 21:47.16 | Naikrovek | Katty: do you have a hole saw? |
| 21:47.23 | Katty | no |
| 21:47.26 | mayfield | katty lasers? |
| 21:47.27 | Katty | no |
| 21:47.28 | Naikrovek | do you have a jigsaw? |
| 21:47.30 | Katty | no |
| 21:47.32 | Naikrovek | hm. |
| 21:47.32 | beek | Katty: I put a loop of #12 wire in a soldering gun and use that. |
| 21:47.40 | Naikrovek | yeah melting |
| 21:47.45 | Katty | i do have a soldering gun |
| 21:47.47 | mayfield | Katty then it isnt going to happen today =\ |
| 21:47.47 | Naikrovek | light sabre that rubbermade |
| 21:47.51 | Katty | k |
| 21:48.02 | Katty | ohoh |
| 21:48.07 | Katty | what about those plumbing pipes |
| 21:48.14 | mayfield | magnify glass and sun? |
| 21:48.27 | mayfield | submit magnify for glass cup? |
| 21:48.31 | beek | PVC you can cut with a hacksaw and sand smoothe. |
| 21:48.35 | mayfield | substitute |
| 21:48.44 | mayfield | autocompletefail |
| 21:49.03 | Katty | http://schroeder-family.us/jpg/rice%20box.jpg <- i'm trying to do that |
| 21:49.07 | *** join/#asterisk garymc (~chatzilla@host86-164-36-128.range86-164.btcentralplus.com) |
| 21:49.19 | Katty | i need a light sabre |
| 21:49.42 | beek | Diameter is too large. |
| 21:49.44 | mayfield | yea, def need a ls |
| 21:49.50 | wcselby | Katty - http://www.thinkgeek.com/computing/thumb-drives-storage/c12e/ |
| 21:50.13 | Katty | ha |
| 21:50.15 | Katty | that's cute |
| 21:50.27 | Katty | they should make external wireless adaptors like that too |
| 21:51.14 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:54.11 | vader-- | any of you guys using/used sipxecs? or any thoughts on it? |
| 21:54.14 | vader-- | sipx |
| 21:54.30 | *** join/#asterisk etfonhomey (~etfonhome@74-143-192-74.static.insightbb.com) |
| 21:54.34 | Naikrovek | haven't used |
| 21:54.43 | vader-- | it looks really cool |
| 21:57.45 | orangepower | i changed the outgoing CID on my trunk, but it's still staying the same as the trunk's DID |
| 22:00.07 | Naikrovek | orangepower: some providers don't let you monkey with the CID |
| 22:00.24 | Naikrovek | some mandate that outgoing CID matches your DID and fix it for you if it doesn't |
| 22:02.33 | p3nguin_ | I'd like to know how that works for people using termination only services. |
| 22:02.46 | p3nguin_ | "termination only" that is. |
| 22:03.46 | p3nguin_ | For a long time, I had just a DID with one company and used someone else for termination. I since got a DID with the other, though. |
| 22:04.18 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 22:05.20 | Corydon76-dig | Many providers will turn that off for you, if you give them a legitimate reason why you want that ability |
| 22:05.44 | Corydon76-dig | "Forwarding calls to cell phones" is a legit reason. "Punking my friends" is not. |
| 22:07.02 | p3nguin_ | Luckily, my provider doesn't ask for me to convince them. |
| 22:08.29 | Corydon76-dig | I suppose it would depend on how good their logs are. If they get a call from an irate user, they could probably trace the call directly back to you |
| 22:09.07 | Corydon76-dig | Of course, the secondary problem with faking CID is whether the irate user can identify your provider as the source |
| 22:10.17 | Corydon76-dig | Some providers require the CID to be 10 digits and not an invalid number (i.e. real area code, real exchange, etc.) |
| 22:10.53 | wcselby | exit |
| 22:10.56 | wcselby | oop |
| 22:10.59 | wcselby | oops |
| 22:11.04 | Corydon76-dig | FAIL |
| 22:11.08 | wcselby | agreed |
| 22:12.55 | [TK]D-Fender | [16:57]<orangepower>i changed the outgoing CID on my trunk, but it's still staying the same as the trunk's DID |
| 22:13.04 | [TK]D-Fender | orangepower: "tunk" pardon? |
| 22:13.09 | [TK]D-Fender | trunk* |
| 22:13.15 | *** join/#asterisk sulex (~sulex@host-78-14-173-189.cust-adsl.tiscali.it) |
| 22:13.30 | Corydon76-dig | batunkatunk |
| 22:13.43 | *** join/#asterisk ellisdee (~ellisdee@cosmic.sized.penisinyourface.com) |
| 22:13.46 | p3nguin_ | You mustn't have seen the "asterisk trunk" conversation. |
| 22:13.55 | Katty | Corydon76-dig: do what? |
| 22:14.05 | Corydon76-dig | ba-trunk-a-trunk |
| 22:14.21 | Katty | i don't get it |
| 22:14.36 | Corydon76-dig | whoosh! |
| 22:15.15 | Corydon76-dig | Katty: know what a ba-dunk-a-dunk is? |
| 22:15.35 | doneir | isn't it a big booty? |
| 22:15.47 | Katty | Corydon76-dig: no? |
| 22:15.57 | Corydon76-dig | Katty: that's why you don't get it |
| 22:16.15 | Katty | reads up |
| 22:16.32 | p3nguin_ | Asterisk trunk: http://imagebin.org/85039 |
| 22:17.06 | p3nguin_ | Asterisk trunks: http://imagebin.org/85038 |
| 22:18.02 | Katty | hehehe |
| 22:18.05 | Katty | i like that last one |
| 22:18.11 | p3nguin_ | http://www.urbandictionary.com/define.php?term=badonkadonk |
| 22:18.32 | doneir | i was right |
| 22:18.34 | doneir | ;] |
| 22:18.59 | Corydon76-dig | p3nguin_: not an Asterisk trunk: http://upload.wikimedia.org/wikipedia/commons/8/82/African_Elephant_Trunk.jpg |
| 22:19.01 | *** join/#asterisk t_j (~tj@tomjudge.vm.bytemark.co.uk) |
| 22:19.30 | t_j | anyone know if I can use a polycom SpectraLink i640 without an SVP server? |
| 22:21.16 | Corydon76-dig | t_j: I believe so, but the question is, why would you want to take a very expensive phone that will roam like that and confine it to a single base station? |
| 22:21.37 | t_j | surely it can roam without svp? |
| 22:21.51 | Corydon76-dig | Nope, it's confined to a single base station otherwise |
| 22:22.01 | t_j | humm, damn |
| 22:22.20 | t_j | guess its time to raise a PO |
| 22:23.10 | p3nguin_ | corydon76-dig: http://imagebin.org/86512 |
| 22:23.31 | Corydon76-dig | A single base station may work in a small office... but not so much in a multi-floor office building or in a warehouse |
| 22:23.51 | Corydon76-dig | p3nguin_: nice. ;-) |
| 22:24.16 | t_j | Corydon76-dig: yeah, just i cant even find where to disable the SVP to even place a test call |
| 22:24.39 | Corydon76-dig | t_j: you have a DECT base station? |
| 22:24.40 | p3nguin_ | Now if it were only sipping water from a stream, it would be an asterisk elephant sip trunk. |
| 22:24.51 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 22:24.52 | t_j | Corydon76-dig: no its a wifi phone |
| 22:25.32 | Corydon76-dig | Ah, I was under the impression that it was a DECT phone |
| 22:25.45 | t_j | nah WiFi.. |
| 22:26.00 | Corydon76-dig | Many wifi phones still won't roam. |
| 22:26.20 | Corydon76-dig | QuickPhone QA-342, last I checked (and I have one) won't roam |
| 22:26.57 | t_j | this one suposedly does |
| 22:28.08 | *** join/#asterisk rossand (~aross@dhcp-233-179.tb-classrooms.carleton.ca) |
| 22:28.14 | Kobaz | how do i increase the volume of the call waiting beep on polycom phones |
| 22:32.37 | *** join/#asterisk oldhack (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
| 22:35.07 | *** join/#asterisk Chris-NB (~chris@home.fuerstaller.com) |
| 22:35.09 | Chris-NB | hi |
| 22:35.20 | Chris-NB | anyone using q.sig with asterisk |
| 22:35.34 | Chris-NB | and qsigchannelmapping? |
| 22:36.10 | Chris-NB | I try to activate logical channel mapping, but without success |
| 22:36.45 | Chris-NB | pri show span X allways shows Logical Channel Mapping: 0 instead of Logical Channel Mapping: 1 |
| 22:36.50 | Chris-NB | anyone seen this Problem? |
| 22:39.31 | *** part/#asterisk ellisdee (~ellisdee@cosmic.sized.penisinyourface.com) |
| 22:39.54 | *** join/#asterisk ellisdee (~ellisdee@cosmic.sized.penisinyourface.com) |
| 22:43.45 | *** join/#asterisk sprite-- (~sprite@c-98-251-108-29.hsd1.ga.comcast.net) |
| 22:44.13 | sprite-- | What's the best TTS engine that is compatible with Asterisk? I'm using Cepstral right now and not happy with the quality. I do not mind paying a bit more. |
| 22:44.34 | Qwell | sprite--: Cepstral is often considered the "good but expensive" one |
| 22:45.29 | Kobaz | expensive? |
| 22:45.34 | Kobaz | cepstral is really cheap |
| 22:45.34 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.5, 1.6.1.17, 1.6.0.25 (2010/02/25), 1.4.29.1 (2010/02/18), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2, 1.6.0.4, 1.4.10 (2009/12/02), dahdi-linux 2.2.1 + dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 22:45.44 | sprite-- | yeah Cepstral is super cheap |
| 22:45.49 | Qwell | Kobaz: > $0 is considered expensive :p |
| 22:45.54 | leifmadsen | Asterisk 1.6.0.25, 1.6.1.17, and 1.6.2.5 are now available. Read the release announcement here: http://www.asterisk.org/node/49910 |
| 22:45.57 | Kobaz | leifmadsen: 1.6.0.4 ? |
| 22:46.05 | leifmadsen | eh? |
| 22:46.11 | Kobaz | look at what you topic'd |
| 22:46.11 | leifmadsen | huh... |
| 22:46.28 | leifmadsen | look at the prefix to those version numbers :) |
| 22:46.30 | leifmadsen | *-Addons |
| 22:46.31 | sbrath | can I make a hint that's one number, but different statued based on each phone? |
| 22:46.33 | Kobaz | oh |
| 22:46.38 | Kobaz | oh yeah |
| 22:46.46 | sbrath | I don't think that will work, but just checking. |
| 22:46.46 | sprite-- | http://www.acapela-group.com/text-to-speech-interactive-demo.html like this is the quality I would like. Those voices are not bad at all compared to cepstral. |
| 22:48.46 | leifmadsen | sbrath: DEVICE_STATE() function |
| 22:49.08 | Kobaz | i've had problems with the device state function |
| 22:49.16 | Kobaz | i had to write my own implementation |
| 22:49.38 | Chris-NB | noone using qsigchannelmapping=logical ? |
| 22:49.44 | Kobaz | nope |
| 22:49.46 | sprite-- | So there are no solutions that easily integrate with asterisk for better quality TTS than Cepstral? |
| 22:49.54 | Kobaz | sprite--: sure there are |
| 22:50.09 | sprite-- | Kobaz: Which ones? I would like to research them. |
| 22:50.37 | Kobaz | sprite--: any of them that can output a wav file, or something that asterisk can play |
| 22:54.34 | *** join/#asterisk pa (~paolo@unaffiliated/pa) |
| 22:56.05 | Katty | almost time to go home |
| 22:56.09 | Kobaz | i really wish there was a way to update polycom configs without rebooting |
| 22:56.11 | Kobaz | such a pain |
| 22:56.37 | *** join/#asterisk andres833 (~andres833@190.159.5.11) |
| 22:56.39 | p3nguin_ | Surely it can't take more than about 60 seconds for that to happen. |
| 22:56.46 | Kobaz | i heard a rumor that nortel was doing stuff with polycom phones.. apparently they wrote custom firmware and got them to boot in 10 seconds, |
| 22:58.12 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.15.160) |
| 23:00.07 | *** join/#asterisk JayTee52 (~jforde051@unaffiliated/jaytee) |
| 23:00.30 | Kobaz | i figured out how to raise the volume of the call waiting tone for polycom |
| 23:00.31 | Kobaz | <PROTECTED> |
| 23:00.40 | Kobaz | i increased the duration too |
| 23:00.51 | *** join/#asterisk Zettatronic (~nick.croc@99-89-192-120.lightspeed.hstntx.sbcglobal.net) |
| 23:03.58 | *** join/#asterisk clintc (~clintc@n128-227-15-193.xlate.ufl.edu) |
| 23:05.03 | *** join/#asterisk Tako-san (~Tako-san@p4022-ipad69osakakita.osaka.ocn.ne.jp) |
| 23:09.24 | [TK]D-Fender | If you need to care how long it takes a phone to boot.. you have far more serious issues |
| 23:09.57 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 23:11.59 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 23:12.39 | *** join/#asterisk clintc (~clintc@n128-227-15-193.xlate.ufl.edu) |
| 23:12.54 | *** part/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 23:19.03 | *** join/#asterisk mnt_real (~sinan@bas1-montreal43-1177754737.dsl.bell.ca) |
| 23:23.13 | ellisdee | any recommendations for a osx soft phone? |
| 23:23.18 | ellisdee | free |
| 23:24.01 | t_j | i use xlite |
| 23:28.11 | Zettatronic | xlite works great |
| 23:28.33 | Zettatronic | even through double NAT lol seen it happen... |
| 23:29.18 | *** join/#asterisk hipitihop (~denis@203.132.229.187) |
| 23:32.28 | hipitihop | I'm running 1.6.2.4 compiled from source on an ION Atom 330 and * i s using 100% cpu, someone have any ideas |
| 23:33.10 | ellisdee | what does top say? |
| 23:34.45 | *** join/#asterisk Zettatronic (~nick.croc@99-89-192-120.lightspeed.hstntx.sbcglobal.net) |
| 23:35.02 | hipitihop | ellisdee, what figures/columns do you need ? |
| 23:35.41 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 23:36.05 | hipitihop | ellisdee, 4448 root 20 0 488m 13m 7324 R 100 0.8 7:14.54 asterisk |
| 23:39.16 | ellisdee | if you are lookig at top; look at the cpu % column. find the processes with the highest cpu consumption |
| 23:40.05 | hipitihop | ellisdee, asterisk is 100% see 4th figrue from the right on above line form top |
| 23:40.28 | ellisdee | asterisk -vvvvvvc |
| 23:40.28 | Qwell | Using 100% CPU doing what? |
| 23:40.34 | ellisdee | check out the console |
| 23:40.37 | ellisdee | see whats going on |
| 23:40.45 | ellisdee | may have some wierdness in your dialplan hosing the pbx |
| 23:41.15 | *** join/#asterisk unspin (~unspin@209-207-88-129.ip.van.radiant.net) |
| 23:41.25 | hipitihop | Qwell, that is what I am trying to establish ... there are no calls in progress |
| 23:42.36 | hipitihop | ellisdee, console is quiet.... what should I run at console to see what it's doing ? |
| 23:44.09 | ellisdee | dunno man. only thing i can tell you is to read up on strace and ps attributes.. |
| 23:45.11 | *** join/#asterisk drako (~luisjose@nelug/coreteam/luisjose) |
| 23:47.25 | anthm | top -H to see which thread it is vs a gcore bt all |
| 23:49.01 | hipitihop | anthm, not sure what to look at |
| 23:49.30 | anthm | top -H shows each thread and the cpu usage for that thread so you find the one that is 100% and save the thread id |
| 23:49.38 | *** join/#asterisk lost_soul (~noymfb@cpe-67-241-68-202.twcny.res.rr.com) |
| 23:49.40 | Qwell | anthm: nice |
| 23:49.44 | anthm | then you gcore the process and run gdb on the core file and thread apply all bt |
| 23:49.54 | anthm | and match up the thread id |
| 23:50.05 | anthm | and you can tell which one it is and what code it's in |
| 23:50.13 | Qwell | yeah, seeing what the thread is doing would be very useful |
| 23:51.28 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:52.45 | ellisdee | do a strace on asterisk.. |
| 23:57.22 | hipitihop | anthm, ok have core file, can you elaborate the gdb and thread apply steps |
