| 00:02.06 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 00:02.12 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 00:02.32 | *** join/#asterisk momelod (~smelo@bas1-toronto08-1279487786.dsl.bell.ca) |
| 00:02.44 | momelod | greetings channel |
| 00:03.14 | *** join/#asterisk pawz (~pawz@ppp118-208-157-82.lns20.bne1.internode.on.net) |
| 00:04.15 | momelod | trying out agi but not getting anywhere.. can anyone tell me why my script doesn't seem to run properly: http://pastebin.com/M24zTa5E |
| 00:04.40 | momelod | basically im recording a call, but would like to delete in side of the conversation.. |
| 00:04.59 | momelod | my agi script is bash and no matter what i try, it seems to ignore the 'rm' command |
| 00:07.42 | *** join/#asterisk Deeewayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 00:07.42 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 00:08.03 | jaytee | momelod, try rm -f /tmp/$1-out.wav |
| 00:08.49 | momelod | hrmm.. i should have mentioned it works when i run it interactivly |
| 00:08.56 | momelod | and i dont have an alias for rm |
| 00:10.19 | momelod | but worth a try |
| 00:12.18 | momelod | jaytee, did u see my script in pastebin? |
| 00:12.20 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 00:12.26 | jaytee | yes |
| 00:12.49 | momelod | when i added the -f it passed the if statement and played the file.. but it didnt actually delete my file |
| 00:14.18 | momelod | so very strange.. |
| 00:15.42 | Tobarja | permissions perhaps? |
| 00:17.33 | momelod | all owned by asterisk:asterisk |
| 00:19.01 | momelod | i think the argument i pass to the script gets wiped out by `while read -e ARG && [ "${ARG}" ] ; do :; done` |
| 00:19.14 | h1kz0r | i'm sure it's something simple but I'm trying to route a DID call to another sip provider based on ANI and i'm getting some error that the extention does not exist |
| 00:19.17 | h1kz0r | http://pastebin.com/Z6UnxeNV |
| 00:20.16 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:20.48 | *** join/#asterisk Netgeeks (~chris@173.11.68.155) |
| 00:25.00 | jaytee | h1kz0r, you shouldn't use DISA for that, DISA is to dial into your switch and get dialtone to make another outbound call, not reroute calls. |
| 00:25.29 | h1kz0r | yes that's exactly what i'm trying to do |
| 00:31.10 | jaytee | do you ever get dialtone? |
| 00:31.13 | h1kz0r | dialin, * detects my ani and give me a tone, dial a number and it should go through another sip provider |
| 00:31.23 | h1kz0r | no ig ot this error |
| 00:31.40 | h1kz0r | [Apr 7 20:14:30] NOTICE[3336]: chan_sip.c:18225 handle_request_invite: Call from '' to extension '585492xxxx' rejected because extension not found. |
| 00:31.57 | carrar | ~book |
| 00:31.58 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 00:32.15 | h1kz0r | http://pastebin.com/PPmBJVPg |
| 00:33.22 | [TK]D-Fender | h1kz0r: its not looking in the CONTEXT you think it is. |
| 00:33.30 | [TK]D-Fender | h1go to * CLI and do "sip set debu on" |
| 00:33.45 | [TK]D-Fender | h1kz0r: and pastebin the complete attempt. |
| 00:33.47 | [TK]D-Fender | h1go to * CLI and do "sip set debug on" |
| 00:33.55 | h1kz0r | k done |
| 00:35.00 | h1kz0r | let me try again and see what's happening, thanks again TK |
| 00:36.08 | *** join/#asterisk diegomad (mad@186.81.136.65) |
| 00:44.58 | h1kz0r | [TK]D-Fender: http://pastebin.com/0Ty61YJw |
| 00:45.21 | h1kz0r | the 646 number is mine and 585 is the DID |
| 00:47.02 | h1kz0r | btw * answers but tells me that the number is not in use |
| 00:48.12 | spenguin[work] | hrm ive been trying to get this sangoma card working |
| 00:48.15 | spenguin[work] | its being a pita |
| 00:48.26 | spenguin[work] | followed all the steps right |
| 00:48.31 | spenguin[work] | but yet I get |
| 00:48.32 | spenguin[work] | wanpipe1 | AFT TE1 | N/A | Disconnected | |
| 00:48.43 | spenguin[work] | while dahdi shows |
| 00:48.44 | spenguin[work] | wanpipe1 card 0 OK 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) |
| 00:50.26 | spenguin[work] | what could be wrong? |
| 00:51.12 | [TK]D-Fender | h1kz0r: No matching peer for '646573xxxx' from '64.154.41.150:5060' <--- didn't match your peer for this |
| 00:51.19 | *** part/#asterisk hluesea (~hluesea@88.247.127.66) |
| 00:51.49 | [TK]D-Fender | h1kz0r: h1kz0r Looking for 585492xxxx in default (domain 68.44.81.246) <-- it's looking in [default] because thats where [general] told it to (or you didn't tell it anything at all |
| 00:51.55 | [TK]D-Fender | h1kz0r: fix your peer to match. |
| 00:52.17 | p3nguin | Ah, this again? |
| 00:52.21 | [TK]D-Fender | is heading out |
| 00:52.27 | h1kz0r | lol ye |
| 00:52.34 | h1kz0r | thank man |
| 00:53.01 | h1kz0r | how can i tell it to use the incoming context? include? |
| 00:53.25 | p3nguin | Match a peer, and make sure that peer's context is set correctly. |
| 00:53.44 | p3nguin | You have to match the peer, or it uses the context in the general section. |
| 00:54.02 | h1kz0r | here is the config.. http://pastebin.com/PPmBJVPg |
| 00:54.13 | h1kz0r | I'll change the ani and test again |
| 00:54.20 | h1kz0r | but the did is correct |
| 00:55.43 | p3nguin | Is your call matching a peer when it comes in? |
| 00:56.36 | h1kz0r | apparently not |
| 00:56.54 | h1kz0r | this is the log from the first attempt http://pastebin.com/0Ty61YJw |
| 00:57.33 | h1kz0r | looks like something simple like tk mentioned, i'm using the wrong context or something |
| 00:58.10 | p3nguin | No user '646573xxxx' in SIP users list |
| 00:58.23 | p3nguin | No matching peer for '646573xxxx' |
| 01:00.00 | h1kz0r | and it says Looking for 585492xxxx in default (domain 68.44.81.246) but i have the extension defined in another context |
| 01:02.17 | p3nguin | Do you have a sip.conf? |
| 01:02.32 | p3nguin | If so... Have I seen it yet? |
| 01:02.40 | h1kz0r | yes the sip provider is defined there |
| 01:02.45 | h1kz0r | let me paste it |
| 01:03.42 | h1kz0r | actually i'm using the default template and just added the provider i'm using in the dialout extension |
| 01:03.59 | p3nguin | Default template? What's that? |
| 01:04.26 | h1kz0r | the sip.conf that was there when i just installed it |
| 01:04.27 | p3nguin | Asterisk does not come with any default working config. |
| 01:05.08 | h1kz0r | let me strip out the commented line to get a meaningful view |
| 01:06.31 | *** join/#asterisk chendy (~chatzilla@204.152.211.137) |
| 01:07.42 | h1kz0r | this is my sip.conf |
| 01:07.42 | h1kz0r | http://pastebin.com/wKWe1t95 |
| 01:09.52 | h1kz0r | is the context=default what I'm looking for here? |
| 01:10.16 | p3nguin | Your peer named mysipprovider does not have a context. |
| 01:11.33 | h1kz0r | my context here should be the incomingcalls? |
| 01:11.36 | p3nguin | To get a call to go into the incoming_calls context, you must first match that peer definition, which should have context=incoming_calls in it. |
| 01:11.56 | p3nguin | You haven't met the first criterion, though. |
| 01:12.07 | h1kz0r | i have it there but it's commented out :\ |
| 01:12.24 | p3nguin | Until you match the peer, it's useless anyway. |
| 01:13.56 | h1kz0r | you mean a user for my callerid? |
| 01:14.06 | p3nguin | no |
| 01:15.11 | h1kz0r | should i change the context=default in sip.conf or leave it alone? |
| 01:15.12 | p3nguin | The call that comes in from the ITSP doesn't match the peer definition. |
| 01:15.15 | p3nguin | leave it. |
| 01:15.45 | h1kz0r | should I create another peer for the DID sip provider? |
| 01:15.54 | h1kz0r | this peer there is or the outgoing guy |
| 01:15.58 | h1kz0r | for* |
| 01:16.37 | p3nguin | You're trying to call inbound, by dialing the DID number from the PSTN (or cell phone, which is still on the PSTN). |
| 01:16.46 | h1kz0r | yes |
| 01:17.23 | p3nguin | Does the provider have a unique name, or do you desire to continue calling it mysipprovider? |
| 01:17.47 | h1kz0r | i have them called properly in the file now |
| 01:17.56 | p3nguin | I'm not able to see it. |
| 01:18.05 | h1kz0r | i only have the outgoing one defined but i'm creating another entry for the DID one |
| 01:18.23 | p3nguin | Have you read The Book yet? |
| 01:18.23 | p3nguin | ~book |
| 01:18.24 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 01:18.50 | h1kz0r | ye i did |
| 01:19.35 | h1kz0r | the terminology is a bit confusing to me, having worked on other systems with same names |
| 01:19.50 | p3nguin | This is my last attempt at getting some useful information to work with. If you're going to keep hiding the pertinent pieces, there's nothing more I can do. |
| 01:20.31 | carrar | You can fly there and fix it |
| 01:20.54 | p3nguin | He would put xxxx's on the airliner's name. |
| 01:20.59 | carrar | haha |
| 01:21.05 | *** part/#asterisk QbY (~kelvin@c-24-126-145-123.hsd1.ga.comcast.net) |
| 01:21.30 | h1kz0r | those are the user/pass and my cell phone number and i can't see why u need them |
| 01:22.26 | h1kz0r | and the DID/cell phone i'm leaving half of it so u can identify |
| 01:22.26 | p3nguin | Flight xxx, going toward xxxx, arrives at xx. |
| 01:22.26 | spenguin[work] | whats a loopback cable I can make to test my sangoma E1 card? |
| 01:22.26 | h1kz0r | user/pass/host are all fine and they're registered |
| 01:22.57 | p3nguin | I can see that you clearly know more about this than I do, so I'll leave you to wrap up this little issue and get to making calls right away. |
| 01:23.11 | carrar | crosspins 1/4 and 2/5 |
| 01:23.45 | spenguin[work] | carrar: any images I can look at? |
| 01:23.55 | carrar | yes |
| 01:24.09 | carrar | This on one end |
| 01:24.09 | carrar | http://www.osburn.com/~osburn/images/rj45-straight.png |
| 01:24.16 | carrar | this on the other |
| 01:24.16 | carrar | http://www.osburn.com/~osburn/images/rj45-t1crossover.png |
| 01:25.06 | carrar | only pings 1,2,4,5 are used with T1's |
| 01:25.08 | carrar | pins |
| 01:25.32 | spenguin[work] | carrar: but hows this used to test the card? |
| 01:25.46 | carrar | it puts a hard loop on the T1 interface |
| 01:25.55 | carrar | err |
| 01:25.58 | carrar | sorry |
| 01:26.03 | carrar | don't maek the cable |
| 01:26.07 | carrar | Just make a connector |
| 01:26.10 | spenguin[work] | carrar: the card has just one interface |
| 01:26.20 | carrar | with the wire comign out of pine 1 goign back into pin 4 |
| 01:26.29 | carrar | and 2 comign out and going back into pin 5 |
| 01:26.30 | p3nguin | You just need to use a connector and a pair of wires from a piece of cable. |
| 01:26.37 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 01:26.42 | spenguin[work] | hrm |
| 01:26.43 | carrar | forget making the cable |
| 01:26.54 | p3nguin | That's going to test your loop. |
| 01:27.05 | carrar | thats making a hard loop |
| 01:27.23 | spenguin[work] | 1-4, 2-5 ? |
| 01:27.29 | p3nguin | If you do that on a card in the computer, does it work the same as testing the local loop via smart jack? |
| 01:27.35 | carrar | but you did say loopback cable |
| 01:27.39 | carrar | so the images apply |
| 01:27.53 | carrar | straight on one end, crossover for T1 on the other |
| 01:28.03 | spenguin[work] | ok well I meant, I wanted to test the card |
| 01:28.15 | carrar | So a loopback 'plug' |
| 01:28.22 | spenguin[work] | k |
| 01:28.29 | ChannelZ | sounds dirty |
| 01:28.43 | carrar | plug it in and it makes a hard loopback, just use a RT45 and cross pins 1 with 4 and 2 with 5 |
| 01:28.54 | carrar | RT=RJ |
| 01:29.06 | *** join/#asterisk g-ram (~gsaathoff@cpe-66-66-4-11.rochester.res.rr.com) |
| 01:29.08 | carrar | Dirty as you know is the most fun |
| 01:29.10 | g-ram | evening all |
| 01:29.23 | g-ram | anyone here familiar with common carrier laws in the US? |
| 01:29.44 | g-ram | I think I've found a telco that's specifically discriminating against voip calls that terminate on their network |
| 01:29.45 | carrar | mind being more specific |
| 01:30.06 | carrar | Did you see the FCC ruling? |
| 01:30.09 | g-ram | long distance calls, calls to cell phones, international calls -- all are fine |
| 01:30.11 | g-ram | yes I saw that |
| 01:30.17 | carrar | they can not enforce it at this time |
| 01:30.26 | g-ram | they can't enforce common carrier at all? |
| 01:30.29 | g-ram | I read the ruling |
| 01:30.36 | carrar | I'm not a lawyer |
| 01:30.40 | carrar | but I can play one |
| 01:30.41 | g-ram | are you saying that verizon can stop sending calls to AT&T customers? |
| 01:31.01 | carrar | they could probably |
| 01:31.07 | carrar | would be bad for biz |
| 01:31.22 | g-ram | hmmm |
| 01:31.25 | g-ram | yes it would |
| 01:31.30 | p3nguin | extremely |
| 01:31.49 | g-ram | but for a small rural carrier (that has a monopoly) to screw up inbound voip calls from small providers to their customers |
| 01:31.53 | g-ram | would actually be good for business |
| 01:32.03 | g-ram | it would convince local customers that voip is unreliable |
| 01:32.21 | carrar | don't make it obvious it's a voip call |
| 01:32.24 | g-ram | interestingly inbound calls (to our system) from that carrier are just fine |
| 01:32.27 | carrar | use a differnet port or tunnel it |
| 01:32.31 | g-ram | how can I possibly do that? |
| 01:32.44 | g-ram | I'm already sending sip traffic to a provider that interconnects with the pstn |
| 01:33.08 | ChannelZ | How would the carrier even know the call was coming from VoIP? You're saying they're looking up lines owned by VoIP companies and blacklisting them? |
| 01:33.18 | g-ram | well |
| 01:33.23 | g-ram | they have a monopoly here |
| 01:33.30 | carrar | thats a good game BTW |
| 01:33.44 | carrar | people still play board games |
| 01:34.00 | g-ram | ha |
| 01:34.15 | ChannelZ | I like Sequence |
| 01:34.40 | h1kz0r | I did create the other peer and give meaningful names, now i'm getting another error... |
| 01:34.43 | h1kz0r | http://pastebin.com/6nmwt2EC |
| 01:36.19 | spenguin[work] | so I need a rj45 plug for the loopback plug? |
| 01:36.30 | spenguin[work] | RT45? |
| 01:36.40 | carrar | yes |
| 01:36.43 | carrar | RJ45 |
| 01:36.47 | spenguin[work] | ok |
| 01:36.54 | h1kz0r | [Apr 7 21:25:26] WARNING[21935]: chan_sip.c:11319 check_auth: username mismatch, have <ipcom_incoming>, digest has <585492xxxx> but the 585 number is the DID for ipcom_incoming in the config |
| 01:37.48 | carrar | RJ45c i think to be picky, but a rj45 is fine |
| 01:37.54 | p3nguin | h1kz0r: You're using IPcomms for the DID? |
| 01:38.01 | h1kz0r | yes |
| 01:39.31 | *** join/#asterisk OrNix (~ornix@178.49.0.149) |
| 01:39.37 | p3nguin | h1kz0r: http://pastebin.com/00K00aNw |
| 01:39.42 | p3nguin | This is right off my system. |
| 01:40.34 | h1kz0r | no user/secret stuff here? |
| 01:40.54 | p3nguin | There isn't any. |
| 01:41.01 | p3nguin | They send calls to you. That's all. |
| 01:41.24 | p3nguin | If you are using the .100 system, it is static and provisioned in their web interface. |
| 01:42.31 | p3nguin | If you didn't try to hide their name and IP address from me earlier, we would have already covered this. |
| 01:42.32 | *** join/#asterisk hluesea (~hluesea@88.247.127.66) |
| 01:42.49 | *** part/#asterisk hluesea (~hluesea@88.247.127.66) |
| 01:43.15 | h1kz0r | i didn't have it earlier, just added it when u told me about the peer thing |
| 01:43.29 | h1kz0r | i only had that one and i named it something generic, my bad |
| 01:44.54 | h1kz0r | now i'm back to first error |
| 01:45.00 | h1kz0r | Looking for 5854928868 in default (domain 68.44.81.246) and no peer thing |
| 01:45.00 | *** join/#asterisk tessier_ (~treed@kernel-panic/copilotco) |
| 01:46.05 | p3nguin | Did you save and run sip reload? |
| 01:46.16 | h1kz0r | yes |
| 01:46.22 | h1kz0r | i'm using the host they gave me instea to test now |
| 01:46.38 | p3nguin | is confused |
| 01:46.51 | p3nguin | Are you still talking about an incoming call from IPcomms? |
| 01:49.10 | h1kz0r | yes i just changed the IP you have with the hostname they gave me for the peer configuration and the incoming is now UP :) |
| 01:49.35 | h1kz0r | i detected my ANI, i put the pin, gave me a tone and i dialed... then it failed |
| 01:49.36 | h1kz0r | http://pastebin.com/WWJ1jvqS |
| 01:50.47 | p3nguin | You're doing outbound via IPcomms, too? |
| 01:51.05 | h1kz0r | no it's the other guy' |
| 01:51.10 | p3nguin | That's what I thought. |
| 01:51.10 | h1kz0r | ipc_outgoing |
| 01:51.23 | p3nguin | Show me the peer for it. |
| 01:51.28 | p3nguin | peer definition |
| 01:51.55 | h1kz0r | http://pastebin.com/6nmwt2EC |
| 01:54.59 | *** join/#asterisk TJ^ (~tjq@host86-142-46-156.range86-142.btcentralplus.com) |
| 01:55.14 | p3nguin | What codec are you supposed to use with them? |
| 01:55.20 | TJ^ | anyone know a good SIP to skype? |
| 01:55.27 | TJ^ | free maybe? |
| 01:55.51 | h1kz0r | 723 i guess |
| 01:57.27 | p3nguin | http://pastebin.com/iw0hKgv3 |
| 01:57.34 | p3nguin | Try with this and give me a debug. |
| 02:00.57 | tvaughn | what plays .ul?? |
| 02:01.46 | p3nguin | What's a .ul? |
| 02:01.53 | p3nguin | underwriter's laboratory? |
| 02:01.55 | carrar | ulaw? |
| 02:01.58 | p3nguin | UL listed. |
| 02:02.03 | tvaughn | ulaw |
| 02:02.05 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 02:02.23 | p3nguin | What does the file command say about the file? |
| 02:03.43 | h1kz0r | http://pastebin.com/0MBt0v4j |
| 02:04.39 | h1kz0r | i did verify my SPA was working with 723 |
| 02:04.57 | p3nguin | The ITSP doesn't agree on it, and that's not a sip debug that you gave me. |
| 02:05.04 | tvaughn | data |
| 02:05.36 | p3nguin | I would try playing the file with all the media players I have available: mplayer, vlc, sox, et cetera. |
| 02:05.59 | carrar | file can't always accurately detect a ulaw audio file as such |
| 02:06.08 | p3nguin | yeah |
| 02:06.23 | p3nguin | I would try to play it. It'll either work or it won't. |
| 02:06.37 | carrar | go into your sound directory: file `ls -1 *.ulaw` |
| 02:07.48 | p3nguin | wide range of results there. |
| 02:10.55 | h1kz0r | btw p3nguin this is the provider i told you about, which does not allow me to make outbound calls without registering so i have a register line for it |
| 02:11.24 | p3nguin | That's fine... but the problem so far is that you didn't agree on the codec. |
| 02:12.04 | p3nguin | If you would have given me the sip debug like I asked, I would know what codecs they are offering. |
| 02:12.51 | p3nguin | The codec negotiation is a problem right now. |
| 02:13.38 | p3nguin | I bet they allow ulaw, so if you aren't going to show me the debug, change the allow=g723 to allow=ulaw and try again. |
| 02:16.43 | h1kz0r | i used to get similar error when not registered but the register line should be enough right? |
| 02:17.45 | *** join/#asterisk hluesea (~hluesea@88.247.127.66) |
| 02:18.32 | tvaughn | I've tried a couple players :( |
| 02:19.39 | h1kz0r | i tried fromuser and fromdomain as per some forum user posting it but got something similar |
| 02:22.07 | p3nguin | I recommend knowing what each does before expecting it to magically work. |
| 02:22.25 | p3nguin | You're not going to show the debug, are you? |
| 02:25.43 | h1kz0r | i sent it to you as a pm since my numbers were all there |
| 02:25.48 | *** part/#asterisk hluesea (~hluesea@88.247.127.66) |
| 02:27.16 | carrar | Sounds HOT++ |
| 02:27.40 | p3nguin | You shouldn't assume that I receive PMs. |
| 02:28.06 | carrar | <PROTECTED> |
| 02:28.09 | carrar | oh crap |
| 02:37.47 | *** join/#asterisk Slugs_ (~yeah@173-8-52-45-Jacksonville.hfc.comcastbusiness.net) |
| 02:39.19 | *** join/#asterisk zomenox (~brock@254120.firewall.winthrop.edu) |
| 02:43.13 | p3nguin | waits longer |
| 02:45.42 | h1kz0r | if ur talking to me, i'm not pasting it here, sorry |
| 02:50.59 | *** join/#asterisk hluesea (~hluesea@88.247.127.66) |
| 02:51.32 | hluesea | hello how can i monitor the calls in realtime ? |
| 02:52.59 | p3nguin | hluesea: ChanSpy() or ExtenSpy() |
| 02:53.45 | *** join/#asterisk wdbl (~not@ool-44c0668f.dyn.optonline.net) |
| 02:54.21 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 02:54.27 | hluesea | ohh sorry for my poor english, i guess i ask the question wrongly. Actually i want to show realtime total calls in system a web page. Have a way to do this ? |
| 02:54.58 | *** join/#asterisk Teevo (~steve@CPE-58-168-128-26.lnse3.lon.bigpond.net.au) |
| 02:55.03 | Teevo | Hello everyone. |
| 02:56.38 | ChannelZ | aloha |
| 02:57.47 | Teevo | I clearly have no idea what I am doing here. |
| 02:58.08 | Teevo | I've just setup asterisk and extensions using the AsteriskNow install. |
| 02:58.20 | Teevo | Got the extentions working without any trouble. |
| 02:58.27 | Teevo | Can't get the trunks to work. |
| 02:58.44 | Teevo | They arn't registering for some reason, and I don't even know where to start looking. |
| 02:59.03 | p3nguin | I think I know where to have you look... |
| 02:59.10 | p3nguin | #asterisknow |
| 02:59.35 | Teevo | Yeah, only there's no one active in that channel, at least not for the last 4 hours. |
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| 03:06.35 | p3nguin | If you want help here, it's going to be help on asterisk without asterisknow/freepbx. |
| 03:12.27 | Teevo | Well I don't even know how to get into the asterisk console, but it would probably help if I did. |
| 03:15.30 | p3nguin | From the system command line, run "asterisk -r" and it should load. |
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| 05:20.36 | sawgood | ChannelZ: are you here by chance? |
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| 05:59.57 | ChannelZ | yeah |
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| 06:15.31 | ChannelZ | debates playing some Little Big Planet |
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| 07:17.08 | sergee | There a lot of software echo cancelers in DAHDI, i've tested mg2, sec and kb1 - and it seems that CPU load is the same for each algorithm - is that correct? or am i missing something? |
| 07:20.26 | sergee | shouldn't CPU load differ for sec, kb1 and mg2? or are they all equaly CPU-intensive? |
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| 07:21.45 | kaldemar | you just seemed to answer your own question before even asking it. :) |
| 07:23.00 | sergee | kaldemar: in whic part? :) "am i missing something" or "they all are equaly CPU-intensive" ? :) |
| 07:24.24 | kaldemar | in the part where you told you have observed them to have the same load on your CPU. |
| 07:24.44 | ChannelZ | The proof is in the pudding, as they say |
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| 07:28.53 | sawgood | Is there any way to set for good yoru verbose level in Asterisk, so you can simply do asterisk -r (instead of asterisk -rvvvvvvv each time) |
| 07:29.14 | sawgood | Also, is there a max verbose level for Asterisk? |
| 07:29.14 | sergee | kaldemar: the strangest thing, is that CPU load with hardware EC is equal to CPU load with those software EC in my tests |
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| 07:32.38 | sergee | here are my tests: http://img511.imageshack.us/img511/5404/graphimage.png |
| 07:33.05 | sergee | from left to right: hardware, mg2, kb1, sec |
| 07:33.19 | kaldemar | sawgood: do a wrapper script for the command if you're too lazy to add a few v's. last time i checked, the largest verbosity level was 5, but that was a long time ago. go for 10 and you'll be sure to get all verbosity. |
| 07:34.19 | ChannelZ | sawgood: you can put commands in cli.conf that will set options in the host process |
| 07:34.27 | sergee | This load is caused by 120 calls through TE412P listening MOH |
| 07:34.43 | sawgood | thank you! |
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| 07:55.51 | WIMPy | sergee: Interesting stuff. Are you sure hw-ec is actually working? |
| 07:58.55 | sergee | WIMPy: no i don't know how to check it, the only thing about hw-ec is a string in dmesg "VPM450: echo cancellation for 128 channels" |
| 07:59.20 | sergee | WIMPy: oh, and there is one more string there "VPM450: Present and operational servicing 4 span(s)" |
| 07:59.41 | WIMPy | Ok, that's more than I got. |
| 08:00.06 | WIMPy | Would be nice if the internals were a little more transparent. |
| 08:01.02 | sergee | yeah |
| 08:01.48 | ChannelZ | I'm not sure how the system functions if you have HWEC, it's possible you can't force it to software if it's present? |
| 08:06.23 | niekie | AIEEEEEEEEEEEEUH%U!HU!!&(Y$Y^!*$^!( |
| 08:06.29 | niekie | I spent hours recording voiceprompts. |
| 08:06.35 | niekie | Now they want a female voice >:( |
| 08:06.39 | WIMPy | That's the impression I get from reading, yes. |
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| 08:09.55 | niekie | is pissed off now. :\ |
| 08:10.01 | ChannelZ | niekie: pitch it up! |
| 08:10.28 | ChannelZ | Or tell them a lesbian read it |
| 08:10.39 | ChannelZ | a butch lesbian |
| 08:11.03 | sawgood | when you say to 'add' -rvvvvvvvvv to the cli.conf file (I did not see that file listed in /etc/asterisk |
| 08:11.17 | ChannelZ | Well I didn't say that |
| 08:11.28 | ChannelZ | you'd add something more like "core set verbose 5" |
| 08:11.47 | sawgood | will that remain for every session in the future? |
| 08:11.53 | ChannelZ | If you don't have an example file (FreePBX strikes again) it should look something like |
| 08:11.56 | ChannelZ | [startup_commands] |
| 08:12.09 | ChannelZ | core set verbose 5 = yes |
| 08:12.21 | sawgood | is there a max verbose level (I just did 12 and it took) |
| 08:13.00 | sawgood | I exited out, and went back in ... and the verbose level was 'saved' |
| 08:13.02 | sawgood | thank you |
| 08:13.14 | sawgood | Verbosity is at least 12 |
| 08:13.22 | sawgood | probably no higher then 10 though ... |
| 08:13.29 | sawgood | asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 08:13.32 | ChannelZ | The max is 2147483647 |
| 08:13.33 | sawgood | wonder what that would do? |
| 08:13.56 | ChannelZ | but I'm not sure what the practicality of what is output is. Probably between 1 and 10 |
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| 08:25.04 | niekie | ChannelZ: I'm not sure if enough v's would fit on the command line for that :P |
| 08:25.55 | ChannelZ | heh no but you can 'core set verbose' it on the console. A cursory browse through the source seems to show 10 being the highest verbosity anything gets logged at |
| 08:41.16 | talntid | niekie, use a program to modify the pitch :P |
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| 08:45.35 | kruemeltee | hello all together :-) |
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| 08:48.02 | kruemeltee | how am I able to quit the "*CLI>" Console of * without terminating asterisk himself? |
| 08:48.42 | ChannelZ | asterisk has to have been run in 'detatched' or forked mode |
| 08:49.04 | sergee | kruemeltee: use wraper to launch asterisk (safe_asterisk) |
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| 08:49.23 | kruemeltee | okay ... thanx a lot ... :-) |
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| 08:50.33 | ChannelZ | or don't run it as 'asterisk -f' initially which causes it not to fork |
| 08:51.28 | kruemeltee | so i just have to start * with -f --- after that I'm able to connect to * with -r ... if I want to get to the * console ... and how to quit then? Just simple "quit"? |
| 08:51.49 | kruemeltee | oh ... sorry, not with "-f" ;-) |
| 08:52.22 | kruemeltee | currently I'm starting * with "asterisk -vvvvvc" ... if it's running I reconnect with -r ... |
| 08:52.49 | ChannelZ | no, NOT start with -f |
| 08:52.58 | kruemeltee | yeah ... I recognized :-) |
| 08:53.20 | ChannelZ | so if you are doing 'asterisk -r', when you 'quit' you aren't stopping the main asterisk process |
| 08:53.37 | kruemeltee | okay ... that's what I need :-) |
| 08:53.42 | sergee | kruemeltee: start asterisk with safe_asterisk |
| 08:53.52 | sergee | kruemeltee: that is the proper way |
| 08:54.00 | kruemeltee | whats the difference between asterisk and asterisk-safe? |
| 08:54.09 | ChannelZ | no it's not, it's just a different way. safe_asterisk does other things |
| 08:54.39 | sergee | kruemeltee: safe_asterisk is a shell script which forks to background and returns console to you |
| 08:54.46 | ChannelZ | safe_asterisk runs asterisk and if it dies, runs it again. This may or may not be desireable. |
| 08:54.55 | sergee | also safe_asterisk restart asterisk in case of unexpected failures |
| 08:55.26 | ChannelZ | But, safe_asterisk also has the added benefit of giving asterisk a console to attach to and will make your colors work when you 'asterisk -r' which is of interest to some people |
| 08:56.06 | niekie | talntid: no, they want someone else to do it. :\ |
| 08:56.10 | kruemeltee | but the colored console is great, isn't it? ... I always work with this one ;-) |
| 08:57.16 | ChannelZ | yes it's nice |
| 08:58.36 | sergee | ChannelZ: yes it is :) default init scripts prefer to use safe_asterisk over asterisk binary |
| 08:58.44 | ChannelZ | not anymore |
| 08:59.07 | ChannelZ | (unless they've switched back in 1.6.2) |
| 09:00.05 | sergee | ChannelZ: you are wrong http://svnview.digium.com/svn/asterisk/trunk/contrib/init.d/rc.redhat.asterisk?view=markup |
| 09:00.34 | ChannelZ | ah that's RH. They switched Debian some time ago |
| 09:01.44 | ChannelZ | looks like most of the others are still using it, but the debian one they changed to run asterisk saying that there really aught to be a separate init script for 'safe'. *shrug* |
| 09:02.16 | ChannelZ | I'm off to bed, have fun all |
| 09:02.31 | kruemeltee | bye bye ChannelZ |
| 09:03.22 | sergee | ChannelZ: yes you are right, Debian's init script was changed back in 2007 |
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| 09:13.05 | aceio | ChannelZ: thank for last night |
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| 09:13.31 | aceio | so far the phone are working correctly |
| 09:14.35 | aceio | the only thing that i changed was to remove encapsulation dot1q on the switch ports |
| 09:14.58 | aceio | and so for its behaving |
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| 10:26.20 | Gido-E | for everybody who wants to see a sneak preview of the new kernel traffic shaper possibilities |
| 10:26.22 | Gido-E | http://www.syn-3.nl/index.php/nl/home-mainmenu-1/nieuws/353-traffic-shaper-video |
| 10:26.57 | Gido-E | the gui in the movie is dutch, but it wil give you some idea. Most important is the traffic view per shaping rule |
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| 11:40.17 | carrar | moo |
| 11:42.46 | Gido-E | BOE! |
| 11:49.57 | sergee | a-moo-con |
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| 11:54.29 | carrar | BSOE |
| 11:54.41 | carrar | BSoE |
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| 12:10.29 | carrar | Morning! |
| 12:10.38 | carrar | Let the crazieness begin! |
| 12:15.14 | [TK]D-Fender | carrar: It... never stopped :( |
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| 12:31.48 | carrar | heh |
| 12:31.50 | carrar | thats true |
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| 12:34.19 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:35.10 | ariel_ | Morning |
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| 12:36.09 | carrar | Evening |
| 12:36.25 | Gido-E | afternoon |
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| 12:37.01 | carrar | Konbanwa actually |
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| 12:53.07 | *** join/#asterisk Skeeter- (skeeter@190-141.cgocable.ca) |
| 12:58.36 | Skeeter- | Anyone knows about Polycom's Device Certificates?? |
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| 13:01.12 | carrar | Skeeter, you mean the first link that comes up in google? |
| 13:01.45 | carrar | http://tinyurl.com/ybpyh6q |
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| 13:09.54 | Skeeter- | I did |
| 13:10.00 | Skeeter- | i read the lil pdf |
| 13:10.23 | Skeeter- | it shows you how to check if you have the cerfificate or not |
| 13:10.37 | carrar | and what it's for |
| 13:12.21 | Skeeter- | i didnt really understand anything |
| 13:12.34 | Skeeter- | like most of model dont require/need/can have it |
| 13:12.49 | Skeeter- | theres not how-to install it |
| 13:12.51 | carrar | Sounds like you don't need to worry about it then if you don't understand it |
| 13:13.09 | Skeeter- | carrar, i wish it would work like that |
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| 13:14.29 | ariel_ | Skeeter-: this is for the mini web on some of the polycom phones |
| 13:14.51 | carrar | and for using the TLS |
| 13:14.55 | ariel_ | it's so your able to go to sites requiring them. |
| 13:15.41 | ariel_ | I have not found any use for them on the phones |
| 13:15.55 | carrar | required if you plan to use TLS |
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| 13:27.36 | Skeeter- | ariel_, morning btw |
| 13:28.00 | ariel_ | morning |
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| 13:32.16 | maisa | anyone awake? |
| 13:32.28 | Gido-E | yep awake. |
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| 13:32.45 | maisa | can i use asterisk with panasonic kx-nt136 ip phones? |
| 13:33.04 | maisa | i have a fresh install but i'm not sure i can use it propperly |
| 13:33.25 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 13:33.48 | Gido-E | maisa, does it do sip/rtp? |
| 13:34.24 | maisa | well.... it asks for a pbx server? :) |
| 13:34.29 | Gido-E | :-) |
| 13:34.55 | *** join/#asterisk PhoenixMage (~Phoenix@49.71.96.58.static.exetel.com.au) |
| 13:35.09 | PhoenixMage | Hi |
| 13:36.06 | PhoenixMage | I have just compiled asterisk from source but when I dd chan_sip.so (its failing to load) I am getting a huge list of undefined symbols such as "/usr/lib/asterisk/modules/chan_sip.so: undefined reference to `ast_rtp_alloc_size'" anyone know what I have done wrong? |
| 13:36.27 | maisa | wait a second |
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| 13:37.03 | *** join/#asterisk djMax (~chatzilla@66.92.91.132) |
| 13:37.44 | kaldemar | PhoenixMage: in case of a dependency issue it would fail to build. what does it say when it fails to load? |
| 13:38.31 | djMax | So we have a Voice T1 into Asterisk with about 200 DIDs, but relatively low volume. We want to switch to an IP-based provider to stop paying $600 for phone service. How does one do this and maintain the DID blocks? |
| 13:38.35 | maisa | Gido-E: http://www.palebeck.co.uk/Products/Phones/IP-Phones/KXNT-136-IP.aspx does that help? |
| 13:39.26 | PhoenixMage | kaldemar: It doesnt give me any reason, just says "Module 'chan_sip.so' could not be loaded. |
| 13:39.27 | PhoenixMage | Unable to load module chan_sip.so" |
| 13:39.29 | mcateer | djMax: Local Number Portability |
| 13:39.52 | djMax | a lot of the trunk providers seem to bill "a lot" per DID, whereas we pay like $5 for the whole block now. |
| 13:40.07 | [TK]D-Fender | maisa: http://www.ciao.co.uk/Productinformation/Panasonic_KX_NT_136__6502290 <- NO |
| 13:40.08 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 13:40.58 | PhoenixMage | kaldemar: I have completely mangled all of the config files, maybe I have done something there, but its not complaining about anything when I start asterisk |
| 13:41.07 | [TK]D-Fender | maisa: Useless |
| 13:41.17 | djMax | So I'm wondering if perhaps I'm looking in the wrong place - that the trunk providers don't handle situations like this |
| 13:41.47 | [TK]D-Fender | djMax: Its hard enough to port a few random numbers from carrier to carrier. entire blocks is decidedly harder |
| 13:41.52 | kaldemar | PhoenixMage: pastebin your sip.conf so someone can take a look at it. |
| 13:43.29 | maisa | [TK]D-Fender: the document or the phone? |
| 13:43.30 | sysreq | hi! quick question; is there a way to select which items to compile/install without using make menuselect's ncurses menu? it seems like macos x's latest update broke ncurses.. |
| 13:43.32 | maisa | :) |
| 13:44.19 | [TK]D-Fender | maisa: Your phone speaks proprietary protocols which * does not. They cannot be used together |
| 13:45.01 | coppice | if its a panasonic it probably speaks japanese |
| 13:45.18 | Gido-E | maisa, at this moment with that info, i think it is not compatible with sip (asterisk). They have special phones with sip in the name, like yours have ip |
| 13:45.28 | PhoenixMage | http://pastebin.com/WZ94nX1q |
| 13:45.37 | maisa | too bad |
| 13:45.47 | djMax | is there an "IRC approved" business VoIP provider in the Northeast US? |
| 13:45.51 | pentanol | hello, perhaps some one knows a clinet for sip which be able to visualize existen voice while a call? i.e. sinusoidal graph or something... |
| 13:45.54 | maisa | i have like a million of those but nothing to connect them |
| 13:46.30 | Gido-E | maisa, sometimes it is possible to install an other firmware. So it is compatible with sip (asterisk). |
| 13:46.31 | *** join/#asterisk P424D0X (~DL7RAY@2001:470:1f0b:449:215:f2ff:fef4:a244) |
| 13:46.45 | *** join/#asterisk AlHafoudh (~ahmed@195.46.69.4) |
| 13:46.48 | AlHafoudh | hi all |
| 13:46.49 | [TK]D-Fender | Gido-E: Not on a Panasonic phone. |
| 13:47.00 | AlHafoudh | where can I find h323 channel module not ooh323 ? |
| 13:47.01 | maisa | Gido-E: any idea where to find instructions? |
| 13:47.16 | Gido-E | maisa: Thu 08-Apr-10 15:46 < [TK]D-Fender> Gido-E: Not on a Panasonic phone. |
| 13:47.21 | maisa | :( |
| 13:47.43 | maisa | and then i have this 2n voiceblue lite, can i use that with asterisk? |
| 13:47.46 | Gido-E | maisa, ok. All ip phones? |
| 13:47.56 | maisa | Gido-E: yes, all the same kind |
| 13:48.11 | Gido-E | voiceblue lite is a device? |
| 13:48.15 | maisa | yes |
| 13:48.25 | Gido-E | url or something? |
| 13:48.37 | maisa | http://www.2n.cz/en/products/gsm-gateways/voip/voiceblue-lite/ |
| 13:48.40 | P424D0X | I can't receive incoming calls, but outgoing calls run.. In the debug mode i can see a Msg: "No application '' for extension" |
| 13:48.48 | maisa | some kind of gsm gateway |
| 13:48.51 | [TK]D-Fender | maisa: Let me summarize this a bit simpler : They are dead-ends meant only to work with THEIR PBX core. Nothing else. Not adaptable. Sell them, or trash them. |
| 13:49.12 | maisa | okay, i will burn them tomorrow |
| 13:49.27 | djMax | Broadvox spends all this money on SEM and then can't even keep their site up. Yeesh. |
| 13:49.35 | maisa | i just got them for free and i tought i could use them somehow |
| 13:49.43 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 13:50.03 | [TK]D-Fender | maisa: The VoiceBlue unit should work fine (assuming its a decent product) |
| 13:50.06 | PhoenixMage | kaldemar: sorry http://pastebin.com/WZ94nX1q |
| 13:50.17 | [TK]D-Fender | maisa: Now you know why you got them for free.... |
| 13:50.23 | [TK]D-Fender | maisa: You should ask for your money back :p |
| 13:50.30 | maisa | [TK]D-Fender: now i know :) |
| 13:51.03 | [TK]D-Fender | P424D0X: APSETBIN the complete call attempt from * CLI along with your dialplan |
| 13:51.05 | [TK]D-Fender | ~pb |
| 13:51.06 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 13:51.07 | [TK]D-Fender | ^^^^^^^^^ |
| 13:51.08 | maisa | does cisco phones work with asterisk? |
| 13:51.10 | Gido-E | maisa, that is a kewl device. |
| 13:51.24 | Gido-E | maisa, for sisco phones are firmware patches |
| 13:51.27 | *** join/#asterisk Akiraaa (~Akira@92.81.200.223) |
| 13:51.28 | [TK]D-Fender | maisa: Yes, but stop aiming for companies that don't want to interconnect properly |
| 13:51.36 | [TK]D-Fender | maisa: Where are you located, and what are yuor needs? |
| 13:51.59 | [TK]D-Fender | Finland it seems... |
| 13:52.03 | maisa | finland, and a few working phones mostly for in-house calls |
| 13:52.10 | Gido-E | [TK]D-Fender yep, downloading a firmware from Cisco site, is like hell! |
| 13:52.19 | [TK]D-Fender | maisa: Just go for Linksys SPA series then |
| 13:52.32 | maisa | are those ready to go with asterisk? |
| 13:52.41 | [TK]D-Fender | maisa: Yuo probably aren't looking to muck around with PoE, etc and have basic needs. SPA-941's or similar should do fine |
| 13:52.46 | [TK]D-Fender | maisa: Yes |
| 13:53.07 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 13:53.23 | [TK]D-Fender | Gido-E: They cost more, there are legal and technical issues in getting them converted. Why fight? |
| 13:53.27 | maisa | the less buttons, more better |
| 13:53.42 | [TK]D-Fender | maisa: Then get ATA's and use analog phones. |
| 13:53.43 | maisa | but the voiceblue should be fine with asterisk |
| 13:53.55 | Gido-E | [TK]D-Fender i got a bunch for free :-) |
| 13:54.06 | [TK]D-Fender | maisa: You could get * to use an actual cellphone via bluetooth if you've got one that is compatible |
| 13:54.18 | [TK]D-Fender | Gido-E: Go ask for your money back :p |
| 13:54.31 | maisa | house has only fiber and copper ethernet left, no more phone lines |
| 13:54.42 | anonymouz666 | [TK]D-Fender: out of curiosity... are you using any 1.6.X.X in production? |
| 13:54.51 | maisa | cells would be fine, but staff is changing and they cost something to use |
| 13:55.00 | [TK]D-Fender | anonymouz666: I've got a customer or two, plus my home system |
| 13:55.48 | maisa | thank you guys, i'm sure i'll come back for more.... but now, to the phone store! :) |
| 13:55.51 | djMax | do people like VoicePulse? (or is there an unwritten rule not to recommend trunk providers?) |
| 13:55.53 | [TK]D-Fender | maisa: No.. the voiceBlue is a GSM gateway... that uses a SIM. I'm saying there are several phones you can use as a gateway directly and not have to buy that probably expensive device to do the job |
| 13:56.13 | maisa | [TK]D-Fender: but that one was allso free ;) |
| 13:56.28 | [TK]D-Fender | djMax: They are "OK". They used to be better, but have fallen in opinion over the past year or so |
| 13:56.36 | [TK]D-Fender | maisa: Oh... well then go try it |
| 13:57.25 | *** join/#asterisk ks3 (~ks3@74.203.195.1) |
| 13:57.30 | djMax | $11/DID/month, that is insane-o. |
| 13:57.34 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:57.52 | *** join/#asterisk smooth_penguin (~smoove@59.95.19.116) |
| 13:58.09 | Gido-E | sow! |
| 13:59.06 | *** join/#asterisk freckle (~jon@84.45.168.57) |
| 13:59.50 | ks3 | Is there a way to break SQL statements in func_odbc.conf into multiple lines? I've got several large queries that are getting really difficult to follow... |
| 13:59.51 | [TK]D-Fender | djMax: Oh? What do they give you for that exactly? |
| 14:00.46 | carrar | Just a DID |
| 14:01.07 | byteme` | djMax: I'm using Vitelity |
| 14:01.13 | djMax | I'm confused about their terms, but it appears that every inbound number is $11, not including the regular calling costs. So for 200 numbers we'd be at $2200 per month. Compared to $5 |
| 14:01.17 | *** join/#asterisk freckle (~jon@84.45.168.57) |
| 14:01.35 | [TK]D-Fender | djMax: Link please. |
| 14:01.50 | Gido-E | djMax i can imagtion, that is right |
| 14:01.56 | Gido-E | djMax i cant imagtion, that is right |
| 14:03.29 | PhoenixMage | no one else able to offer why I cant load the sip module? |
| 14:03.42 | [TK]D-Fender | dj "Phone numbers are $11/month and include free incoming minutes. " <- this is ONE plan there |
| 14:03.44 | djMax | http://www.voicepulse.com/plans/default.aspx?plan=lnkPlan1 SMB SPI Trunk plan |
| 14:03.50 | PhoenixMage | is it dependent on any other modules other then chan_local? |
| 14:04.16 | pentanol | I;d not believe, nobody knows any client with sinusoidal voice viewer? |
| 14:04.43 | djMax | yeah, see that now. Still, $11/DID feels like somebody is fleecing somebody. |
| 14:04.56 | [TK]D-Fender | djMax: What are your actual call volumes like? In? Out? |
| 14:04.57 | byteme` | djMax: what provider do you use ? |
| 14:05.24 | [TK]D-Fender | pentanol: Try #oscilloscope |
| 14:05.29 | djMax | Very low, maybe 50 calls per day. 6 people. We have the DIDs for other SMBs we represent. We use XO now. |
| 14:05.50 | djMax | I could probably dump most of the DIDs, but it's certainly nice to have a block |
| 14:07.00 | [TK]D-Fender | djMax: VP does not appear to have the right product for you |
| 14:07.01 | pentanol | heh, oscilloscope itn's what I looking for |
| 14:07.19 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:07.19 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:07.27 | [TK]D-Fender | djMax: http://voip.ms/ |
| 14:07.32 | djMax | yeah. Looking at Vitelity now too, better but still would be $200/mo for the DIDs. Just confusing why XO is willing to give them out like pez |
| 14:08.02 | pentanol | perhaps it's a way to use another asterisks as client call and recording |
| 14:08.42 | [TK]D-Fender | pentanol: I don't see where you are going with this as it is |
| 14:09.01 | [TK]D-Fender | djMax: Sounds like you have a good dea already. |
| 14:09.05 | [TK]D-Fender | dal* |
| 14:09.13 | djMax | except for the $600 for the T1. |
| 14:09.30 | [TK]D-Fender | djMax: that includes DID's, inbound, and some amount of outbound. |
| 14:09.43 | [TK]D-Fender | djMax: Might not be so bad. |
| 14:09.43 | djMax | which I already pay for my data T (that's another boondongle - $500/mo for T1 speed because Comcast isn't available in the building) |
| 14:09.46 | [TK]D-Fender | djMax: depending. |
| 14:10.07 | djMax | yeah, it's true. It all seems to point to dumping the block of DIDs and just porting the ones we need. But that just seems odd. |
| 14:11.26 | djMax | In 2005, it didn't seem insane to spend $1200 per month for voice and data services for a 6 person company. In 2010 I cry at all the money we flushed down the toilet in the past two years. :) |
| 14:12.05 | [TK]D-Fender | djMax: How many DID's do you really feel you need? |
| 14:12.12 | djMax | probably 10. |
| 14:12.29 | [TK]D-Fender | djMax: Night & day. at this point voip.ms is looking really good |
| 14:12.48 | djMax | yeah, I'm talking with their live support now. |
| 14:13.01 | [TK]D-Fender | djMax: Ok, go shop around |
| 14:13.16 | djMax | voip.ms is "well liked"? |
| 14:14.19 | [TK]D-Fender | djMax: They are on the higher end this past yaer |
| 14:14.26 | [TK]D-Fender | year* |
| 14:14.27 | *** join/#asterisk moy (~chatzilla@74.12.128.42) |
| 14:14.32 | djMax | sounds like it moves like the weather. :) |
| 14:15.01 | [TK]D-Fender | djMax: Virginia : if you don't like the weather...wait 10 minutes |
| 14:21.56 | djMax | yeah, holy smokes we pay $1.60 per month for 100 DIDs |
| 14:22.19 | *** part/#asterisk PhoenixMage (~Phoenix@49.71.96.58.static.exetel.com.au) |
| 14:22.19 | P424D0X | [TK]D-Fender: http://pastebin.com/hvLDtQvh |
| 14:22.59 | [TK]D-Fender | djMax: 1.60$ ea? |
| 14:23.43 | djMax | no, total. |
| 14:24.00 | djMax | But we pay $740 for the rest of the crap. |
| 14:24.14 | mcateer | For metered DIDs the best I was able to find about a year and a half ago was $0.75 each. |
| 14:24.21 | [TK]D-Fender | P424D0X: exten => 30,2 Voicemail(30) <---- missing comma |
| 14:24.47 | mcateer | I ended up going with Vitelity who is charging me $1.00 per DID per month. |
| 14:25.18 | djMax | that disparity is somehow broken. Perhaps it's because XO is making the margin up in the T1, but something is out of whack. |
| 14:25.27 | mcateer | I think to get a really good deal on DIDs you have to go with a LEC/CLEC, but then they are charging you much more for all of the features. |
| 14:26.12 | P424D0X | [TK]D-Fender: Ah!! Many thanks.. I'll probe it.. |
| 14:29.24 | djMax | is there such a thing as an IP based LEC/CLEC? |
| 14:29.41 | P424D0X | [TK]D-Fender: Now it goes perfectly!! Thank you very much!! |
| 14:29.41 | djMax | meh, hunting through our cdr, we really only use 20. |
| 14:35.10 | [TK]D-Fender | P424D0X: You're welcome |
| 14:35.20 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
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| 14:40.06 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:40.18 | Skeeter- | i wish russellb could help me with presence over IAX |
| 14:45.44 | *** join/#asterisk suprstar (~suprstar1@216.54.131.253) |
| 14:50.18 | Corydon76-dig | Given that Russell is on vacation, he's not helping anybody but his wife today |
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| 14:52.03 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:52.42 | siriusil | I'm looking for a switch/firewall to go into data center. anyone has recommendation? |
| 14:54.30 | *** join/#asterisk Knightfal (~j@mailer.1callres.com) |
| 14:54.43 | suprstar | for firewalls, I use linux servers with iptables and a buncha nic's for my network of 5000-ish users |
| 14:56.01 | suprstar | solid and cheap to build |
| 14:56.09 | leifmadsen | could use something like pfSense |
| 14:56.26 | Gido-E | suprstar do you prefere a certan network card? |
| 14:56.26 | leifmadsen | Corydon76-dig: we all need a vacation every once in a while |
| 14:56.35 | leifmadsen | I like Intel cards |
| 14:56.36 | *** join/#asterisk thehar (thehar@thehar.xmission.com) |
| 14:57.20 | suprstar | whatever 4-port gigabit cards I have handy, linux supports everything I've come across |
| 14:58.07 | *** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68) |
| 14:58.07 | suprstar | ya intel is good, i have some intelpro 1000's |
| 14:58.38 | *** join/#asterisk Fs0L (~Fs0L@136.223.19.47) |
| 14:58.41 | leifmadsen | that's what I like too |
| 15:00.19 | *** join/#asterisk mutax_ (~mutax@erde.kf.TU-Berlin.DE) |
| 15:03.45 | Gido-E | i prefer those intel cards to |
| 15:04.29 | *** join/#asterisk boch (~fran@200.61.191.9) |
| 15:04.42 | boch | which is the most mature h323 channel for using with asterisk ? |
| 15:06.09 | Knightfal | I have a feature set up in the applicationmap context wifeature => 88,self/callee,Macro,wi-fi when a call comes in and is answered the agent can use it but if the agent dials a call out it will not work I have tried using 88,both,Macro,wi-fi and had the same luck. |
| 15:06.40 | *** join/#asterisk suprstar (~suprstar@216.56.88.4) |
| 15:06.45 | Knightfal | DO I neet to set a var some other place to make it work on an outgoing call |
| 15:07.15 | mutax_ | Hi, I have a question on Hardware support. Does anybody know if a "X-Tensions XC-525 USB ISDN" adaptor will work with asterisk? |
| 15:07.35 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
| 15:07.55 | WIMPy | mutax_: I thik, it's the one I also have. Do you have USB-ID? |
| 15:08.14 | *** join/#asterisk sgimeno (~chatzilla@83.46.36.105) |
| 15:08.17 | mutax_ | WIMPy: not ye, I am planning to buy it |
| 15:08.46 | mutax_ | WIMPy: they are cheap so I think I'll give it a try. Was it 'easy' to get your running? |
| 15:08.59 | mutax_ | I have a picture: http://www.amazon.de/X-Tensions-XC-525-USB-ISDN-Adapter/dp/B000WGN7SO |
| 15:09.58 | WIMPy | mutax: Yes, that one is identical to the billion thing, except for the USB-ID. |
| 15:10.22 | mutax | ah, great :) |
| 15:10.34 | WIMPy | I had to add the ID to the hfcsusb kernel module. No it works fine. |
| 15:10.48 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 15:10.57 | mutax | *click* thanks. I'll give it a try. Changing the kernel is not a problem :o) great! |
| 15:11.03 | WIMPy | Tried to submit a patch, but never got a reply, so probably you also have to duplicate that entry yourself. |
| 15:11.13 | *** join/#asterisk Fubard (~brawr@173.10.235.205) |
| 15:11.45 | mutax | WIMPy: I'll kick my personal kernel developer to include it upstream :D |
| 15:12.07 | WIMPy | Haven't tried the old mISDN, but it works with mISDN2 and even with i4l. |
| 15:12.10 | Fubard | any asterisk/polycom cons/seminars/events coming up that any one can recommend? |
| 15:12.24 | WIMPy | Let me find that patch... |
| 15:12.28 | Naikrovek | Fubard: i think yearly there is Astricon |
| 15:12.33 | Naikrovek | but i don't know when it is this year |
| 15:12.41 | Naikrovek | polycom usually has a heavy presence, as i understan dit |
| 15:13.04 | Fubard | and its in atlanta. not sure if mr. bossman will pay for a plane ticket |
| 15:13.04 | Naikrovek | there's a website that will tell you all that you wanna know |
| 15:13.26 | Fubard | well the astricon is i think |
| 15:13.46 | Naikrovek | Fubard: as far as i know, it's astricon or bust. unless maybe you can find an asterisk users group or something, but honestly I think this channel is your best bet outside of astricon |
| 15:13.57 | mutax | WIMPy: I'm planning a silly setup as first contact with asterisk. To analogue lines coming off a cable-modem put into an analog to ISDN converter, then ISDN into asterisk. two external sip providers coming in there too and two wireless voip phones at home.... the analog-to-isdn part is just silly -.- |
| 15:15.01 | Fubard | Naikrovek: schveet, tyvm for info. maybe i'll luck out and astricon wont be in ATL this year |
| 15:15.37 | Naikrovek | Fubard: according to the website it's in maryland this year |
| 15:15.49 | Naikrovek | Fubard: http://www.astricon.net/ |
| 15:15.54 | Fubard | Naikrovek: just saw that, and dag nabbit lol |
| 15:16.00 | Polysics | hello |
| 15:16.09 | Fubard | Naikrovek: its like they're taunting me |
| 15:16.13 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 15:16.13 | Polysics | any ideas on why MEMBERINTERFACE is not being set on a queue i am invoking in AGI? |
| 15:16.21 | Polysics | i don't think AGI is the problem |
| 15:16.30 | Polysics | the queue does have the proper settings, afaik |
| 15:16.51 | WIMPy | mutax: From analogue to ISDN sounds "interesting". |
| 15:17.29 | Polysics | is tehre a way i can check from the command line to see if my queue is configured properly? |
| 15:17.52 | mutax | WIMPy: *cough* yes.... very ;) http://www.stollmann.de/en/telecommunications//ta-p2i.html |
| 15:18.10 | WIMPy | http://voice.yeti.dk/hfc-usb.patch |
| 15:18.12 | mutax | WIMPy: the interesting part is the price. only like 40 euros new |
| 15:18.18 | Polysics | setinterfacevar in the DB is "yes", let's hope it's not a problem with realtime |
| 15:19.03 | Fubard | Naikrovek: found some sweet stuff - InfoComm 2010 in May @ NV - Cisco Live 2010 in June @ NV |
| 15:19.20 | Naikrovek | i take it you're in NV then |
| 15:19.28 | Fubard | CA |
| 15:19.29 | WIMPy | mutax: I think it's tose things Kabel Deutschland sold to their customers that didn't like the fact that the cable modems only had analogue ports. |
| 15:19.37 | *** join/#asterisk parag7732 (~PARAG@mail2.eastnets.com) |
| 15:19.45 | Naikrovek | i'm in illinois. there ain't ever jack squat near here |
| 15:19.46 | mutax | WIMPy: exactly :) I am a KD customer |
| 15:19.47 | parag7732 | I have 4 failover betamax gateway.....When call jumps to failover gateway, if gateway has money it works fine, but if gateway dosn't have money it dosn't jump to next gateway and instead of caller hear this message "You don't have money, please buy credit"...can i avoid this message and it should jump to another gateway without showing any payment issue ? |
| 15:19.53 | Fubard | i got some friends in henderson though. might be able to knock out on a couch for the weekend. |
| 15:19.54 | Naikrovek | unless it's microsoft doing something small in chicago |
| 15:19.55 | mutax | WIMPy: but bought mine cheaper at ebay |
| 15:20.10 | WIMPy | mutax: Ah, I've read they're pretty shitty. |
| 15:20.54 | mutax | WIMPy: I don't know yet. The main problem is the gap between analogue and isdn signaling like call forwarding, etc |
| 15:21.22 | WIMPy | mutax: It's easier if you change to the scheme with a FritzBox and nick the VOIP account from it's configuration. |
| 15:22.02 | mutax | WIMPy: yes, only KD does not give me a Fritz box If I dont pay 99 Euros per month to get HD-TV and stuff |
| 15:22.02 | WIMPy | ... or accountS |
| 15:22.40 | Fubard | Naikrovek: The American Organization of Nurse Executives 2010 in IN, starts tomorrow. doesn't 'sound' telecom related BUT its on the polycom website so there might be something promising there! |
| 15:22.48 | WIMPy | mutax: Bad luck. I also tried then and I'm very happy, I'm no longer their customer. |
| 15:22.53 | mutax | WIMPy: and although the cable modems do Voip on private IPs from the 10/8 net I will never be able to connect to their voip gateways. |
| 15:22.59 | Naikrovek | Fubard: nurse executives? |
| 15:23.10 | coppice | Fubard: hot nurses? |
| 15:23.11 | Fubard | Naikrovek: its the closest i could find |
| 15:23.24 | mutax | WIMPy: I the normal phone line would give me more than 2 MBits I would change back, too. |
| 15:23.32 | Fubard | Naikrovek: lots in TX if you're down for a drive |
| 15:23.32 | *** join/#asterisk Tulga (~chatzilla@203.91.114.3) |
| 15:23.42 | Tulga | what GUI best for call center solution? |
| 15:23.48 | WIMPy | mutax: The new accounts ar on real IPs. But I have no idea if you can connect to them using a login from an old account. |
| 15:23.50 | mutax | WIMPy: but paying for 6 MBit at german telekom while getting only 2 is ... |
| 15:23.51 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 15:24.41 | Fubard | coppice: not likely, and they're business owners so even if they are hot they probably got an unmanagable personality |
| 15:24.49 | mutax | WIMPy: I am going to test sipgate the next year. If the quality is okay and averything works I think on migrating there. |
| 15:24.49 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 15:25.07 | WIMPy | mutax: Their phone service didn't work at all for me. Huge delays (several seconds), echos and periods of silence. And the internet speed was sometimes lower than ISDN. |
| 15:25.12 | mutax | WIMPy: and KD pisses me off, their customer support is none existent. |
| 15:25.38 | mutax | WIMPy: phone and internet now works. after opening 8 tickets with customer support but never getting any feedback |
| 15:25.53 | florz | plus, they are very fond of censorship ... |
| 15:25.54 | mutax | WIMPy: they upgraded the backbone here |
| 15:26.00 | WIMPy | mutax: Sipgate seems ok. Just that you can't have more than three numbers in a sensible way, even when paying. |
| 15:26.20 | mutax | WIMPy: they have trunking on the business-accounts |
| 15:27.27 | WIMPy | mutax: Some time I started to open a ticket after almost every phone call. I alwas ot replies like: We know there a problems and are working on ith with high pressure. |
| 15:27.53 | WIMPy | That pressure must habe been very low by normal standards as it didn;t change in 7 Months. |
| 15:31.52 | leifmadsen | DarthPointer: why are you trying to start a DCC chat with me? |
| 15:32.13 | [TK]D-Fender | leifmadsen: ASL? |
| 15:32.31 | leifmadsen | I only know a little bit of American Sign Language |
| 15:33.17 | suprstar | lol :D |
| 15:33.45 | Chainsaw | <PROTECTED> |
| 15:33.45 | Chainsaw | <PROTECTED> |
| 15:33.45 | Chainsaw | [2010-04-08 16:22:34] WARNING[22699]: enum.c:518 parse_naptr: NAPTR Regex match failed. Reason: No match |
| 15:33.53 | Chainsaw | Is there a reason why it lies? There is such a record. |
| 15:34.11 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 15:34.38 | mutax | WIMPy: thanks for your help. I just ordered the adaptor... cu... |
| 15:34.40 | *** part/#asterisk mutax (~mutax@erde.kf.TU-Berlin.DE) |
| 15:34.47 | *** part/#asterisk siriusil (~Sirius@87.113.86.107.plusnet.pte-ag2.dyn.plus.net) |
| 15:35.09 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 15:35.37 | *** join/#asterisk moy (~chatzilla@74.12.128.42) |
| 15:35.39 | DarthPointer | leif, you around? |
| 15:39.28 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 15:47.54 | krion | anyone using st2022 from thomson ? |
| 15:48.01 | krion | i've a really weird thing... |
| 15:48.17 | krion | when i call the voicemail using the button dedicated for |
| 15:48.29 | *** join/#asterisk n3llyb0y (~quassel@64.236.163.23) |
| 15:48.36 | krion | i don't see RTP DTMF EVENT on the server |
| 15:49.14 | krion | if i use the number related to the voicemail (here *888), i see rtp dtmf event on the server... |
| 15:52.06 | Chainsaw | http://www.pastebin.ca/1858980 <- Do I see the ENUM resolver blowing up here? If so why? (Full number not scrubbed, you can run your own analysis) |
| 15:52.26 | n3llyb0y | Hello I need some troubleshooting tips. I have a working incoming/outbound configuration using sipgate.co.uk and a couple of local softphones. I am trying to enable an IVR menu on the inbound calls using Background() - the IVR menu works, the target line is called but then there is no audio either way. I have the console output in front of me but I'm not sure what I'm looking for - cheers |
| 15:53.12 | [TK]D-Fender | n3llyb0y: Your server behind NAT by any chance? |
| 15:53.40 | Fubard | n3llyb0y: can you get the sound files to ever play? are you certain they're sampled correctly? |
| 15:53.50 | n3llyb0y | yes but when I do not use Background() - ie. I just connect the incoming to one of my local phones it all works wonderfully |
| 15:54.17 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 15:55.05 | [TK]D-Fender | n3llyb0y: Read up : |
| 15:55.06 | [TK]D-Fender | ~sipnat |
| 15:55.07 | 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 |
| 15:56.20 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 15:56.43 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 15:56.49 | n3llyb0y | Thanks but maybe I'm not making myself clear. My setup works until I introduce a menu via Background(). For all intents and purposes the NAT is working okay. regular incoming and outgoing calls work. |
| 15:57.56 | [TK]D-Fender | n3llyb0y: the isue would be how * sits in the middle which would spell out a NAT issue normally |
| 15:57.58 | bmoraca_work | n3llyb0y: try waiting 2 seconds before playing the IVR prompt |
| 15:59.07 | Fubard | n3llyb0y: maybe the script is still set to listen? i'm no expert on IVR's but it sounds to me like the IVR isn't quitting properly, after an extension is dialed |
| 15:59.54 | n3llyb0y | Fubard - hmmm, is there a way I can find out? |
| 16:00.24 | Fubard | n3llyb0y: @ * CLI, try sip set debug and see whats happening |
| 16:00.46 | Fubard | n3llyb0y: should give some feedback on whats been interpretted/actions taken |
| 16:01.10 | Fubard | n3llyb0y: other than that, I dont know how to debug IVR |
| 16:01.34 | n3llyb0y | Fair enough. Thanks people :) I will have another play |
| 16:01.37 | bmoraca_work | n3llyb0y: <bmoraca_work> n3llyb0y: try waiting 2 seconds before playing the IVR prompt <----------------------------- do this and if it doesn't work, try answering the channel before the prompt plays |
| 16:04.15 | *** join/#asterisk kazaa_lite_ (~eddie@78-86-126-14.zone2.bethere.co.uk) |
| 16:07.00 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 16:08.09 | kazaa_lite_ | hi all... which video codecs are supported by asterisk? |
| 16:08.19 | kruemeltee | may anybody help me on a serious problem ... I don't hear the first cycle of musiconhold while waiting within the queue ... after some time a nice voice is telling me, that all agents are currently busy and after that musiconhold starts again and this time I hear the sound ... |
| 16:08.34 | kazaa_lite_ | i could see http://www.voip-info.org/wiki/view/Asterisk+video but it mainly talks about asterisk 1.4 |
| 16:11.52 | Fubard | kruemeltee: add a delay to the first cycle of musiconhold |
| 16:12.09 | Fubard | kruemeltee: add about 2 seconds to what the delay already is |
| 16:12.37 | Fubard | i feel like i'm plagerizing bmoraca_work lol |
| 16:12.42 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:12.42 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:12.46 | kruemeltee | okay ... I'll give it at try :-) |
| 16:13.07 | p3nguin | How would you add 2 seconds delay in moh? |
| 16:13.20 | bmoraca_work | Fubard: MOH is generally a different issue |
| 16:13.23 | Fubard | in extensions.conf |
| 16:13.34 | p3nguin | I don't think so. |
| 16:13.44 | bmoraca_work | kruemeltee: how is your MOH handled? is it streaming or is it file-based? |
| 16:13.45 | Fubard | wouldnt you just Wait() it? |
| 16:13.47 | p3nguin | When the queue is entered, the queue handles the moh. |
| 16:13.56 | kruemeltee | file based, bmoraca_work |
| 16:14.12 | kruemeltee | wait doesn't help ... |
| 16:14.16 | p3nguin | of course not. |
| 16:14.16 | bmoraca_work | kruemeltee: i would check the first file that plays and make sure it's valid and that you have the appropriate codec translations installed |
| 16:15.13 | kruemeltee | it's only one file in there ... at this time ... and the CLI says it's starting the musiconhold with the right class ... after that nice voice the same line comes up and I hear the sound |
| 16:15.39 | p3nguin | Is the channel in state "Up" before you enter the queue? |
| 16:16.00 | kruemeltee | how do I check this? |
| 16:16.02 | p3nguin | or at the time you enter it, actually. |
| 16:16.15 | kruemeltee | better to ask, how do I do that? |
| 16:16.47 | p3nguin | During the time when you think the moh should be playing but is silent, check the CLI with "core show channels" |
| 16:17.00 | kruemeltee | okay ... just one moment ... |
| 16:18.16 | kruemeltee | as I try to dial from inside the outside number there are 4 active Channels ... and 2 active calls (one outgoing and one incoming) |
| 16:18.53 | kruemeltee | 3 channels are up ... and one channel is ringing (the agent's line) |
| 16:19.13 | p3nguin | okay |
| 16:19.18 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
| 16:21.14 | *** join/#asterisk sbrath (~sbrath@unaffiliated/sbrath) |
| 16:21.48 | kruemeltee | the intresting thing is, that after hearing that voice the file is playing correctly ... but there are no differences between the outputs of both ... "start playing musiconhold ... stop playing musiconhold ... Playing 'queue-periodic-announce' ... and so on |
| 16:22.00 | sbrath | So am I correct in guessing that to make a TAPI windows app aware that a call is being recieved that I have to add some code to the stdexten to "signal" that the call is coming in? |
| 16:22.59 | p3nguin | I'm wondering if there is a chance of a queue or moh bug. |
| 16:23.24 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 16:23.30 | kruemeltee | I don't know ... but ... lets try another phone ... a real ISDN phone ... wait a moment :-) |
| 16:24.06 | kruemeltee | no ... it's the same thing ... |
| 16:24.17 | Chainsaw | Very well, the potential ENUM bug is in a report now; https://issues.asterisk.org/view.php?id=17150 |
| 16:24.58 | p3nguin | As part of troubleshooting, I would try another asterisk version within the same branch. |
| 16:25.58 | kruemeltee | hmm ... difficult to handle ... I just have one machine ... ;-) Maybe I'll start with the playback manually ... and let start musiconhold after that ... |
| 16:26.35 | p3nguin | I don't see why it's difficult to handle. |
| 16:26.47 | p3nguin | And having just one machine makes it that much easier. |
| 16:27.09 | p3nguin | Build a new version within the same branch, install it, restart asterisk, test again. |
| 16:27.30 | kruemeltee | starting the playback manually during entering the queue works ... |
| 16:27.43 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:27.56 | kruemeltee | but as there is a new version of asterisk (I'm still using 1.4) ... that's one possibility |
| 16:28.05 | p3nguin | Which version are you using? |
| 16:28.46 | kruemeltee | 1.4.21.2 |
| 16:28.59 | p3nguin | That's almost an antique. |
| 16:29.08 | anonymouz666 | this version works fine |
| 16:29.19 | kruemeltee | yeah? ... it works fine here ... :-) |
| 16:29.19 | anonymouz666 | for me at least |
| 16:29.37 | kruemeltee | for me too, at least dropping this little failure :-) |
| 16:30.46 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 16:32.37 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 16:33.59 | Katty | today sucks. |
| 16:34.49 | anonymouz666 | why |
| 16:35.09 | Katty | had a bad dream about a person last night. |
| 16:35.16 | Katty | it's just thrown off my entire day |
| 16:38.01 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 16:38.44 | anonymouz666 | omg |
| 16:40.52 | Knightfal | When using ChannelRedirect is there a way to send the channel that was not redirected to a context |
| 16:42.08 | *** join/#asterisk jpcansa (~jpbenavid@201.196.95.200) |
| 16:42.17 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 16:46.45 | *** join/#asterisk nagios1111 (~xx@adsl196-154-128-217-196.adsl196-13.iam.net.ma) |
| 16:47.26 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 16:50.17 | nagios1111 | hello, can any one help me to use asterisk for sending notification nagios to a softphone like x-lite |
| 16:56.33 | SuPrSluG | nagios1111, you need to create a call file. there are examples out there. |
| 16:57.34 | SuPrSluG | nagios1111, usually the call file is in a script that nagios calls |
| 16:58.21 | *** join/#asterisk hfb (~hfb@pool-98-112-219-90.lsanca.dsl-w.verizon.net) |
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| 17:00.32 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 17:00.54 | nagios1111 | nagios generate notification as message text that send it to asterisk for generate call file |
| 17:01.20 | nagios1111 | but i don't know how to do this |
| 17:02.26 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 17:04.23 | SuPrSluG | nagios1111, here's a good example http://www.it-slav.net/blogs/2008/10/31/use-aterisk-to-call-a-number-and-read-the-errormessage-from-op5monitornagios/ |
| 17:05.01 | SuPrSluG | nagios1111, that should get you started in the right direction |
| 17:05.22 | *** join/#asterisk githogori (githogori@51.sub-75-210-124.myvzw.com) |
| 17:07.20 | *** join/#asterisk ruyo (~psantos@bl6-125-105.dsl.telepac.pt) |
| 17:08.19 | nagios1111 | thank u much SuPrSluG |
| 17:08.29 | nagios1111 | i will try it |
| 17:10.52 | ruyo | What can be the reason for my dahdi line always answering every outbound call? |
| 17:11.44 | ruyo | Like "DAHDI/1-1 answered SIP/blabla" |
| 17:12.14 | *** join/#asterisk Skeeter- (skeeter@c216.218.2-65.clta.globetrotter.net) |
| 17:12.19 | ruyo | It answers pretty much at the same time it starts ringing in the other side. |
| 17:13.41 | tzafrir_laptop | ruyo, it's analog? |
| 17:14.04 | SuPrSluG | nagios1111, np |
| 17:14.44 | tzafrir_laptop | ruyo, basically it is because it has no better way to tell when the call was actually answred. So it answers right away |
| 17:15.23 | *** join/#asterisk Alric (~nbowyer@anvault.com) |
| 17:16.44 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 17:17.14 | ruyo | Yes, it's analog. |
| 17:17.45 | ruyo | Is that a telco problem or a configuration problem? |
| 17:18.22 | [TK]D-Fender | ruyo: Configuration. |
| 17:19.11 | ruyo | I tried changing answeronpolarityswitch and hanguponpolarityswitch, no effect. |
| 17:19.18 | ruyo | What else could it be? |
| 17:19.27 | [TK]D-Fender | ruyo: 2 choices: A: Continue as is and have normal calls. B: Enable "callprogress=yes" for your channels and risk having * hang up on them randomly when it thinks it has died and hasn't |
| 17:20.24 | ruyo | Thank you, I'll try that. |
| 17:21.26 | jpcansa | hello, how can i identify a 3G phone connecting to my * box, so i can add it to iptables exceptions?? |
| 17:22.10 | leifmadsen | at least with my provider the 3G phones IP address changes quite often |
| 17:22.31 | jpcansa | mine too, it changes on every connection |
| 17:22.46 | bent_screwdriver | jpcansa: does the mac? |
| 17:23.56 | jpcansa | bent_screwdriver, i dont think it has mac, does it? i think it uses the imei number |
| 17:24.01 | Qwell | How would a 3G phone connect to Asterisk, exactly? |
| 17:24.09 | idespinner | is there any way to do something similar to #include ${variable_file_name} ? |
| 17:24.15 | idespinner | in extensions.conf |
| 17:24.20 | bent_screwdriver | jpcansa: i would think it would if it is coming through the inet. |
| 17:24.22 | Qwell | idespinner: not with a variable, no |
| 17:24.26 | jpcansa | @Qwell, i installed a sip softphone on it |
| 17:24.39 | Qwell | jpcansa: so then it has a mac |
| 17:25.32 | bent_screwdriver | jpcansa: if the mac doesn't change....iptables -A INPUT -m mac --mac-source 00:00:00:00:00:00 -j ACCEPT |
| 17:25.51 | p3nguin | If the device isn't on the same network segment, the MAC address is going to be useless. |
| 17:25.57 | leifmadsen | I didn't know layer 2 addressing information was passed up through layer 3 |
| 17:25.58 | idespinner | I dont think that would work, the source mac changes as it hits different routers |
| 17:26.03 | idespinner | I agree with p3nguin |
| 17:26.07 | leifmadsen | p3nguin: :) |
| 17:26.42 | jpcansa | mac address will not work |
| 17:26.50 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 17:26.58 | Slugs_ | jpcansa, what softphone did you intall on iphone? |
| 17:27.07 | bent_screwdriver | jpcansa: don't know then.... |
| 17:27.58 | *** join/#asterisk n3hxs (~HAMming@63.68.135.4) |
| 17:28.23 | ruyo | [TK]D-Fender, with callprogress=yes it doesn't answer at all. :< Keeps ringing even after I answer it. |
| 17:28.58 | idespinner | jpcansa, you're probably going to have to open up an entire range since your 3g connection is likley giving you a very dynamic IP.... |
| 17:29.02 | [TK]D-Fender | ruyo: you'll also have to ensure that you have the right indications zone set and not to use answeronpolarityswitch and hanguponpolarityswitch inappropriately |
| 17:29.08 | jpcansa | Slugs_: it´s Windows Mobile, i installed this one: http://www.nch.com.au/talk/index.html |
| 17:29.47 | jpcansa | It works great if i stop iptables |
| 17:30.21 | idespinner | jpcansa, stop ip tables, connect, do a 'sip show peers', get your ip address and open it up |
| 17:30.26 | p3nguin | You could allow the entire block(s) owned by AT&T for use with the iPhones. |
| 17:31.02 | jpcansa | idespinner, i think i will have to open the IP address range of my country or something like that |
| 17:31.14 | p3nguin | They probably only have a few in your area, actually. |
| 17:31.26 | p3nguin | Oh, you aren't in the US? |
| 17:31.35 | jpcansa | nop, no US |
| 17:31.56 | ruyo | The zone is set, I'll play arund with answer/hangup polarities. |
| 17:32.20 | sbrath | If I change the Locale from en_US.UTF8 to en_US.LATIN1 will I need to recompile/restart asterisk? |
| 17:32.24 | p3nguin | Do like idespinner was saying, then lookup the block where that address came from and allow it in iptables. |
| 17:33.19 | jpcansa | p3nguin, how do i add a block of addresses? |
| 17:34.05 | p3nguin | 1.2.3.4/16, I guess |
| 17:34.23 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 17:34.40 | p3nguin | Once you find the block, use the actual numbers though. :) |
| 17:35.06 | jdoe | does anyone know if realtime hints work? I found a bug report from 2006 that was closed as a feature request, but I can't find any further documentation anywhere. |
| 17:35.07 | jpcansa | yeah |
| 17:35.22 | jpcansa | Iwill try that, thanks guys |
| 17:36.31 | [TK]D-Fender | jdoe: They don't |
| 17:37.05 | jdoe | [TK]D-Fender: doh. |
| 17:37.55 | jdoe | dynamic hints refuse to work for me with these phones (polycom ip500s, yes, I know they're old) and realtime extensions etc would be more convenient. |
| 17:38.05 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 17:38.14 | leifmadsen | realtime hints don't exist |
| 17:38.27 | leifmadsen | that issue was closed as a feature request because no one provided code to enable that functionality |
| 17:39.30 | jdoe | I saw, but as near as I can tell it was never opened for a bounty either on voip-info, that's why I was asking. |
| 17:39.36 | [TK]D-Fender | jdoe: Dynamic hints? Pardon? |
| 17:39.40 | jdoe | time to look into how painful it would be to implement I guess. |
| 17:40.19 | jdoe | [TK]D-Fender: voip-info, rightly or wrongly, implies that "exten => _XXX,hint,SIP/${EXTEN}" should work in 1.6.1+ |
| 17:40.44 | [TK]D-Fender | jdoe: Show me it failing |
| 17:43.00 | jdoe | [TK]D-Fender: core show hints never lists anything (other than a _XXX line) and even if that was lying it complete fails for queueing, asterisk will happily send calls to phones that are already inuse. |
| 17:43.31 | [TK]D-Fender | jdoe: Again, show us. |
| 17:43.48 | jdoe | sure, what in particular do you want to see? |
| 17:43.50 | leifmadsen | which version of Asterisk? |
| 17:43.59 | jdoe | 1.6.2.something, one sec I'll grab the exact version. |
| 17:44.22 | jdoe | 1.6.2.6 |
| 17:44.34 | leifmadsen | show the dialplan, and the console output |
| 17:44.47 | leifmadsen | core show hints (or sip show hints, I can't remmeber which it is) |
| 17:45.31 | jdoe | http://pastebin.com/RzUEQDjd is core show hints, one second on dialplan... |
| 17:46.14 | leifmadsen | I'm not sure if the output of that should show anything different... |
| 17:46.19 | leifmadsen | Corydon76-dig: might know |
| 17:46.51 | *** join/#asterisk dennisG (~dennisG@2002:541e:88d0:0:213:2ff:fe56:e380) |
| 17:47.11 | [TK]D-Fender | jdoe: Still don't see the problem.... |
| 17:47.42 | [TK]D-Fender | jdoe: Show me it fail |
| 17:48.17 | hardwire | giggles |
| 17:48.18 | jdoe | http://pastebin.com/UgivtHHh is extensions.conf ... it's nothing exciting, pretty much just a mangling of leifmadsen's queue guide. |
| 17:48.39 | ruyo | [TK]D-Fender, besides answeronpolarityswitch and zone indications, you see any other configuration that can influence the answer detection? |
| 17:49.07 | [TK]D-Fender | ruyo: signaling type |
| 17:49.33 | Corydon76-dig | jdoe: you won't get anything until something actually subscribes to the hint |
| 17:49.43 | [TK]D-Fender | [13:49]<Corydon76-dig>jdoe: you won't get anything until something actually subscribes to the hint <- |
| 17:49.48 | ruyo | Even if it works without callprogress it could be it? |
| 17:49.49 | [TK]D-Fender | jdoe: Show me it FAIL |
| 17:49.58 | Corydon76-dig | Then the hint is duplicated to match the new extension |
| 17:50.12 | [TK]D-Fender | ruyo: lack of callprogress means * doesn't ahve to look for a sign that it ahs been answered |
| 17:50.25 | Naikrovek | jdoe: you need to follow [TK]D-Fender's very plain suggestion to show the failure if you want help fixing it |
| 17:50.46 | Corydon76-dig | It had to be done that way, because devicestate is stateful, so there needs to be a container to hold each distinct extension |
| 17:51.59 | jdoe | Naikrovek: I'm not new here, I understand how this works. |
| 17:52.06 | Naikrovek | just sayin' |
| 17:52.14 | [TK]D-Fender | no seein' |
| 17:52.16 | jdoe | [TK]D-Fender: I guess I'm just crazy, I can't reproduce it today. I must have changed something and not reloaded the config. Sorry for the noise. |
| 17:53.33 | Corydon76-dig | jdoe: again, a phone needs to SUBSCRIBE to the hint to make it show anything more |
| 17:54.11 | Corydon76-dig | jdoe: You've shown me the pattern, but that tells me that none of your phones have SUBSCRIBED to see the status of another |
| 17:55.46 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 17:57.04 | jdoe | Corydon76-dig: do I even want the phones to care about the status of another? I thought the purpose of the hints wrt queues was to let asterisk know what the phones were up to and whether it was a good idea to send them a call. |
| 17:57.23 | [TK]D-Fender | jdoe: No |
| 17:57.29 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 17:57.42 | p3nguin | app_queue should be able to do that all by itself. |
| 17:57.51 | p3nguin | Hints are for BLF. |
| 17:57.52 | [TK]D-Fender | jdoe: Queue's look at device states. Hints look at device states. Queues do not look at hints as some sort of intermediatry |
| 17:58.27 | Corydon76-dig | [TK]D-Fender: correction. Hints look at extension states |
| 17:58.46 | Corydon76-dig | device state != extension state |
| 17:58.54 | [TK]D-Fender | Corydon76-dig: Perhaps more abstract than that... |
| 17:59.12 | [TK]D-Fender | Corydon76-dig: Yeah, I boiled it down a little quick to this particular case. |
| 17:59.21 | [TK]D-Fender | Corydon76-dig: But yeah, we know this |
| 17:59.27 | p3nguin | If extension 1234 is a hint for SIP/1234... wouldn't it check the device state? |
| 17:59.44 | [TK]D-Fender | p3nguin: in the case that it references a device. |
| 17:59.57 | [TK]D-Fender | p3nguin: you can monitor parking lots, custom devices, etc |
| 18:00.31 | jdoe | ah, super, then I guess I don't even need to worry about this at all. |
| 18:00.34 | Corydon76-dig | p3nguin: queues only reference devices, not extensions |
| 18:00.47 | jdoe | thanks guys, helpful as always :) |
| 18:00.59 | Corydon76-dig | p3nguin: You'd need the Local channel to turn an extension into a device |
| 18:01.18 | Corydon76-dig | and even then, for this to work at bootup, you need a very particular load order |
| 18:01.47 | Corydon76-dig | Both the dialplan and the Local channel need to load before the queue application |
| 18:02.08 | p3nguin | corydon76-dig: Your statement that hints look at extension states seemed a bit inaccurate since his hint references a SIP channel. That was my reason for asking. |
| 18:04.06 | p3nguin | I'm just going by what he used in his dialplan rather than making up hypotheticals. |
| 18:04.13 | Corydon76-dig | p3nguin: You have it backwards. Extension states look at hints |
| 18:04.32 | p3nguin | Just basing it on what you said and his dialplan, nothing more. |
| 18:04.33 | *** join/#asterisk blaines (~blaines@c-98-213-119-125.hsd1.il.comcast.net) |
| 18:05.28 | Corydon76-dig | Right, but he'd need a Local channel for the hint to be involved at all |
| 18:07.12 | p3nguin | For what case? |
| 18:07.29 | Corydon76-dig | For a Queue |
| 18:07.29 | p3nguin | I thought he wanted to hint the state of the SIP devices. |
| 18:07.40 | *** join/#asterisk alexfranci (~shiva@host191-58-dynamic.2-79-r.retail.telecomitalia.it) |
| 18:08.09 | Corydon76-dig | p3nguin: for a Queue |
| 18:08.18 | [TK]D-Fender | And queue's don't look at hints at all |
| 18:08.32 | Corydon76-dig | Right, not unless he used a Local channel |
| 18:08.34 | p3nguin | I think that's the root of the confusion. |
| 18:08.42 | alexfranci | hello!when i call an estension that does not exists, asterisk makes busy tone. is there a way to dial a default extension, if the former isn't found? |
| 18:09.18 | Corydon76-dig | alexfranci: _X. is a default extension. |
| 18:09.30 | alexfranci | ok, i will try now |
| 18:11.13 | idespinner | alexfranci, also, 'i' is for invalid extension |
| 18:11.30 | Corydon76-dig | idespinner: yes, but that won't work for origination dialling |
| 18:11.41 | Corydon76-dig | 'i' is only for IVRs |
| 18:12.00 | idespinner | good to know |
| 18:12.01 | [TK]D-Fender | Corydon76-dig: And ZAP/DAHDI channels IIRC |
| 18:12.12 | [TK]D-Fender | Corydon76-dig: as they don't 404 |
| 18:12.15 | Corydon76-dig | [TK]D-Fender: shouldn't |
| 18:12.20 | alexfranci | neither 'i' or '_X.' does not work |
| 18:12.23 | Corydon76-dig | [TK]D-Fender: they should just Congestion |
| 18:12.40 | [TK]D-Fender | Corydon76-dig: I believe they will, but only if "i" doesn't exist. |
| 18:12.43 | Corydon76-dig | alexfranci: then you're putting it in the wrong context |
| 18:13.04 | Corydon76-dig | [TK]D-Fender: I'm telling you chan_dahdi should not be looking for the 'i' extension ever |
| 18:13.19 | Corydon76-dig | Only thing that should be looking for i is the pbx loop |
| 18:13.30 | [TK]D-Fender | Corydon76-dig: I could be mistaken, but I'd like to confirm it... |
| 18:14.02 | Corydon76-dig | [TK]D-Fender: Confirmed. chan_dahdi pays no attention to "i" |
| 18:14.10 | anonymouz666 | heh |
| 18:14.21 | [TK]D-Fender | Corydon76-dig: Ok, works for me :) |
| 18:14.27 | Corydon76-dig | Took me a 2 second grep |
| 18:14.33 | Corydon76-dig | [TK]D-Fender: for origination? |
| 18:14.55 | [TK]D-Fender | Corydon76-dig: Took me an unrelated sip of coffee :) |
| 18:14.59 | Corydon76-dig | I'm not talking about once the call is already established |
| 18:15.12 | [TK]D-Fender | Corydon76-dig: FXS -> call to dialplan |
| 18:15.17 | [TK]D-Fender | Corydon76-dig: Not IVR. |
| 18:15.19 | Corydon76-dig | For an analog line, that means the 's' extension doesn't exist |
| 18:15.32 | *** join/#asterisk DMeloUK (~DominicMe@64.129.93.210) |
| 18:15.46 | Corydon76-dig | FXS is a different story entirely, as it's already in the pbx loop |
| 18:16.04 | Corydon76-dig | Basically, FXS starts out as a NULL extension |
| 18:16.50 | [TK]D-Fender | Corydon76-dig: that is the one I'm looking to confirm. Analog FXS |
| 18:16.55 | Corydon76-dig | FXO and PRI lines should not do anything other than Congestion if the extension doesn't exist, even if an 'i' extension exists |
| 18:17.24 | [TK]D-Fender | Corydon76-dig: this was what I had a potentially misplaced recollection that it would trigger "i" first on an invalid call from dialtone |
| 18:17.26 | Corydon76-dig | [TK]D-Fender: that's not a trunk, though |
| 18:17.33 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:17.41 | [TK]D-Fender | Corydon76-dig: I did say FXS... |
| 18:18.02 | Corydon76-dig | [13:12:01] <[TK]D-Fender> Corydon76-dig: And ZAP/DAHDI channels IIRC |
| 18:18.13 | Corydon76-dig | [TK]D-Fender: only later did you specify FXS |
| 18:18.43 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 18:19.00 | [TK]D-Fender | Corydon76-dig: So.... you admit that I said it? ;) |
| 18:19.20 | Corydon76-dig | [TK]D-Fender: and immediately after you said it, I confirmed that FXS is a different story |
| 18:19.47 | [TK]D-Fender | Corydon76-dig: \o/ |
| 18:20.24 | Corydon76-dig | [TK]D-Fender: but it's not DAHDI that is looking for the "i" extension. As I said, FXS channels start immediately in the pbx loop |
| 18:20.41 | Corydon76-dig | and only the pbx loop looks for the "i" extension |
| 18:21.05 | [TK]D-Fender | Corydon76-dig: OK.... I don't know the core bits, just making sure I'm guaging the end outcome properly. |
| 18:21.31 | *** join/#asterisk korihor (~korihor@190.205.251.97) |
| 18:21.45 | Corydon76-dig | Try this: set up your FXS line to immediate=yes, no 's' extension, but an 'i' extension, and tell me if it works. That should confirm for you that DAHDI doesn't look for the "i" extension. |
| 18:21.46 | *** part/#asterisk DMeloUK (~DominicMe@64.129.93.210) |
| 18:22.17 | anonymouz666 | FXS sucks |
| 18:23.12 | *** join/#asterisk simond (~simon@syria.uc.org) |
| 18:23.59 | simond | Anyone know why my RTP SSRC would be constantly changing in a call? |
| 18:24.41 | simond | My ITSP has suggested that might be why I'm getting intermittend problems with DTMF. |
| 18:25.58 | *** join/#asterisk jsjc (~chatzilla@219-90-149-184.ip.adam.com.au) |
| 18:26.24 | jsjc | i have a few questions.. I wonder how to set default language in asterisk.conf because it is not language= neither defaultlang=..... |
| 18:27.00 | jsjc | as well I just installed fax for asterisk the one from digium.Does that install the app sendfax receivefax? I am running 1.6.2 |
| 18:30.29 | Fubard | other than sip reload and dialplan reload, are there any other means of refreshing/reloading any other settings? (short of restarting asterisk daemon) |
| 18:31.07 | anonymouz666 | simond: in a simple call? the SSRC from source is changing? |
| 18:32.10 | *** join/#asterisk jshriver (~jshriver@cblmdm24-53-177-197.buckeyecom.net) |
| 18:32.15 | jshriver | greetings |
| 18:32.30 | jshriver | Can someone explain the differences from 1.4 and 1.6? Would there be significant .conf changes in upgrading? |
| 18:32.43 | jshriver | or could I reuse extensions.conf sip.conf etc frmo a 1.4 system? |
| 18:32.46 | leifmadsen | jshriver: see UPGRADE.txt and CHANGES |
| 18:33.06 | jshriver | ok will go download the source then |
| 18:33.14 | leifmadsen | their purpose is to explain differences between 1.4 and 1.6.0, 1.6.1, and 1.6.2 (all different branches) |
| 18:33.20 | leifmadsen | ~asterisk-versions |
| 18:33.21 | infobot | well, asterisk-versions is http://www.asterisk.org/asterisk-versions |
| 18:33.26 | leifmadsen | ~asterisk-versioning |
| 18:33.55 | jshriver | Is one version recommended over another? |
| 18:34.01 | simond | anonymouz666: yeah, between DTMF digit presses, the SSRC is changign. |
| 18:34.18 | leifmadsen | jshriver: 1.6.2 is recommended because 1.6.0 and 1.6.1 will no long receive bug fix support as of May of this year |
| 18:34.27 | jshriver | I mean between 1.4 and 1.6 |
| 18:34.30 | leifmadsen | infobot: asterisk-versioning is http://blogs.asterisk.org/2009/06/24/about-the-new-asterisk-versioning-method/ |
| 18:34.31 | infobot | okay, leifmadsen |
| 18:34.36 | leifmadsen | there is no such thing as 1.6 |
| 18:34.48 | jshriver | talking about branches not specific versions |
| 18:34.52 | leifmadsen | so am I |
| 18:35.01 | leifmadsen | ~asterisk-versioning |
| 18:35.02 | infobot | [asterisk-versioning] http://blogs.asterisk.org/2009/06/24/about-the-new-asterisk-versioning-method/ |
| 18:35.08 | leifmadsen | see that link |
| 18:35.22 | leifmadsen | for an explanation about the differences between 1.6.x branches |
| 18:35.31 | jshriver | ok would you recommend 1.6.2 over 1.4.30 |
| 18:35.44 | jshriver | reading |
| 18:36.57 | leifmadsen | jshriver: depends what you want to do -- if you need the features in 1.6.2.6 vs 1.4.30, then yes |
| 18:37.15 | leifmadsen | maintenance support for 1.4 vs 1.6.2 will end about the same time, so that isn't really a consideration |
| 18:37.28 | anonymouz666 | I am almost using the 1.6.2.X in production |
| 18:37.32 | jshriver | ok might stick with 1.4 since that what I'm use to |
| 18:37.34 | *** part/#asterisk lynxsys (~lynxsys@213-152-36-251.dsl.eclipse.net.uk) |
| 18:37.47 | leifmadsen | I've been using 1.6.2 in production before 1.6.2.0 :) |
| 18:37.48 | simond | so instead of incrementing the 1, another dot was added? |
| 18:37.50 | jshriver | second question, anyone recommend a TDM card+FXO chip besides Digium |
| 18:38.11 | leifmadsen | simond: the purpose originally was that there were going to be more feature releases in a shorter period of time |
| 18:38.30 | leifmadsen | simond: this is all explained in several blog posts at http://blogs.asterisk.org |
| 18:38.31 | anonymouz666 | leifmadsen: there are some nice features that I want to start using. But that UDP issue about not releasing the ports scared me |
| 18:38.39 | simond | tl;dr |
| 18:38.45 | simond | well, I skimmed |
| 18:38.58 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 18:39.38 | *** join/#asterisk boch (~fran@200.61.191.9) |
| 18:39.56 | simond | but my real question is about why the ssrc is changing in the midst of a call *swallows some focusin* |
| 18:42.12 | lirakis | i have a did provider who's calls always die at my pbx with " chan_sip.c:13971 handle_request_invite: Failed to authenticate user" |
| 18:42.21 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 18:42.26 | lirakis | i have their ip in ... ive tried insecure=very on their context |
| 18:42.46 | lirakis | does anyone have experience with this happening and/or know what the issue might be? |
| 18:44.29 | jshriver | anyone recommend a good tdm+ fxo chip card not by Digium? |
| 18:44.58 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 18:45.32 | ChannelZ | *crickets* |
| 18:45.39 | jshriver | lol |
| 18:46.17 | ChannelZ | Maybe the 'Top Quality Happytimes Voice Card' from China on eBay someone found the other day |
| 18:46.37 | TheDavidFactor | jshriver are you asking for a card that has both digital on analog ports on the same card? |
| 18:46.50 | TheDavidFactor | s/on/and/ |
| 18:46.57 | boch | is it possible for a FXO card not to mangle incoming calls at all? let other phones answer the calls |
| 18:47.07 | BCS-Satori | How can I make asterisk get the CID(num) from the SIP-URI rather then From Field. My Carrier is passing a + (plus sign) on one my numbers in the from which gets passed to the phone; they said to use the SIP-URI header for caller-id information instead. |
| 18:47.18 | leifmadsen | boch: add a Wait(x) before you answer the call |
| 18:47.57 | jshriver | just analog |
| 18:48.04 | boch | leifmadsen, what happens if another phones picks up before Wait finishes, does this dialplan continues executing? |
| 18:48.13 | lirakis | jshriver, so .. you want fxs and fxo ? |
| 18:48.14 | leifmadsen | boch: although not 100% sure if after you answer the call that the dialplan will exit |
| 18:48.31 | TheDavidFactor | jshriver, I've had a good history with Sangoma as have others |
| 18:48.33 | leifmadsen | boch: there is no signalling (other than voltage changes on the line) with analog, so it is very difficult to do stuff like that |
| 18:48.46 | jshriver | hrm good question not sure what the diff is in fxs vs fxo.. whatever those little red chips are that digium makes or equivalant 100M? |
| 18:48.52 | leifmadsen | its been several years since I played around with analog anything |
| 18:48.54 | lirakis | jshriver, what TheDavidFactor said |
| 18:48.58 | leifmadsen | ~fxo |
| 18:48.59 | infobot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 18:49.00 | leifmadsen | ~fxs |
| 18:49.01 | infobot | i guess fxs is foreign exchange station - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 18:49.03 | lirakis | jshriver, go visit sangoma |
| 18:49.15 | jshriver | ok will look them up ty. |
| 18:50.24 | jshriver | if I can't get this 3rd RMA set of parts working I might buy some from Sangoma, appreciate the info |
| 18:51.23 | *** join/#asterisk imcdona (imcdona@173.160.189.69) |
| 18:51.42 | *** join/#asterisk dajhorn (~dajhorn@adsl-75-17-124-26.dsl.rcsntx.sbcglobal.net) |
| 18:52.00 | BCS-Satori | mmm I love Sangoma |
| 18:52.28 | bmoraca_work | ~channelvariables |
| 18:52.33 | bmoraca_work | no bot on that one? |
| 18:52.34 | bmoraca_work | hmmm |
| 18:53.55 | Qwell | ~channel variables |
| 18:55.30 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:59.22 | bmoraca_work | BCS-Satori: i was going to link to it, but check your doc/tex/channelvariables.tex file. that's where you'll find what you need |
| 18:59.29 | *** part/#asterisk jshriver (~jshriver@cblmdm24-53-177-197.buckeyecom.net) |
| 18:59.57 | sbrath | jdoe: regarding dynamic hints, be careful, I've reported a deadlock issue with them, and had to back them out. In the 1.6.2.x tree |
| 19:01.15 | sbrath | jdoe: and the show hints will show you the _XXXX as a hint, but thats just a side-effect, what should happen is when the phone tries to subscribe, it will cause asterisk to create the hint. |
| 19:01.57 | BCS-Satori | bmoraca_work: thanks |
| 19:03.24 | jdoe | sbrath: I'm not using BLF or anything else (I can see) that needs them. I just left it in because the queue guide implied they were required. |
| 19:03.28 | jdoe | ... so I think I'm safe. |
| 19:05.26 | *** join/#asterisk jshriver (~jshriver@cblmdm24-53-177-197.buckeyecom.net) |
| 19:05.57 | jshriver | !dahdi-tools |
| 19:06.18 | jshriver | ~dahdi-tools |
| 19:06.25 | jshriver | !topic |
| 19:06.26 | jaytee | ~botsnack |
| 19:06.26 | infobot | :), jaytee |
| 19:06.34 | jshriver | ~dahdi-tools |
| 19:06.35 | jshriver | hmr |
| 19:06.39 | Skeeter- | ~jaytee |
| 19:06.40 | infobot | it has been said that jaytee is likes to play with the animals in the zoo |
| 19:06.42 | jshriver | lol anyone know where to download dahdi-tools |
| 19:06.48 | Skeeter- | it works |
| 19:06.59 | Skeeter- | jshriver, are you serious? |
| 19:06.59 | jshriver | ~jshriver |
| 19:07.09 | jshriver | yup, digging around asterisk.org site now |
| 19:07.13 | Skeeter- | ~Qwell |
| 19:07.14 | infobot | i heard qwell is a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. |
| 19:07.20 | Skeeter- | this 1 is amazing |
| 19:07.37 | Skeeter- | jshriver, 1st asterisk install? |
| 19:07.54 | jshriver | installed asterisk and dahdi, but dahdi is now saying to install dahdi-tools |
| 19:08.01 | Skeeter- | oh |
| 19:08.07 | Skeeter- | get the whole package |
| 19:08.12 | Skeeter- | linx+tools |
| 19:08.13 | jshriver | so guessing it's a seperate package. |
| 19:08.18 | Skeeter- | it can be |
| 19:08.36 | Skeeter- | u got the BOTH the LINUX and the TOOLS package |
| 19:08.40 | jshriver | This is what I just compiled: dahdi-linux-2.2.1.1.tar.gz from asterisk.org |
| 19:08.53 | [TK]D-Fender | jshriver: http://www.asterisk.org/downloads |
| 19:09.03 | [TK]D-Fender | jshriver: Look 1 link to the lest |
| 19:09.06 | [TK]D-Fender | left* |
| 19:09.18 | Skeeter- | which distro are you using? |
| 19:09.39 | jshriver | ty that did it :) |
| 19:09.42 | jshriver | CentOS |
| 19:10.05 | Skeeter- | do you have any FXO card |
| 19:10.31 | jshriver | yup 2 TDM410's one with hardware echo cancelation, other w/o and 6 FXO cards |
| 19:10.43 | Skeeter- | ok |
| 19:10.49 | jshriver | *crossing fingers* this is the 3rd RMA batch |
| 19:10.52 | Skeeter- | i nvr played with those |
| 19:11.22 | Skeeter- | give a try to sangoma if you got the oppotunity and money |
| 19:11.53 | Skeeter- | and i think they are less expensive, need a comfirmation on that |
| 19:11.53 | jshriver | aye, really frustrated with digium. will spare details but they royally screwed me |
| 19:12.13 | jshriver | eh it's not about money I just want something that works and will last more than 30minutes lol |
| 19:12.32 | jshriver | but if I have to buy a whole new set again would be nice to save a few bucks lol |
| 19:12.40 | Skeeter- | i used ovre 20 sangoma card over the last 6 months, no problems |
| 19:13.10 | jshriver | nice :) |
| 19:13.20 | jshriver | bbiab need to pop in the cards so make config can work |
| 19:14.06 | Skeeter- | dont get freepbx if you can, it will messed up your whole life this time |
| 19:15.56 | Skeeter- | to anyone using sangoma cards |
| 19:16.02 | *** join/#asterisk Wildy (~simba@89.222.134.42) |
| 19:16.42 | Skeeter- | you need to modify dahdi-base.o in the dahdi-linux in order to make it work if your wanrouter(sangoma card service) restart |
| 19:16.44 | paulc | I'm not, but a buddy of mine swears by Sangoma cards - real big fan. And I respect him and his stuff a lot.. so that's a thumbs up from me |
| 19:17.09 | Skeeter- | you need to comment( add //) at the line 4177 |
| 19:17.26 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 19:17.30 | Skeeter- | paulc, your friend probably encounter that |
| 19:20.25 | *** join/#asterisk talntid (~t@173.160.174.126) |
| 19:20.42 | talntid | Whats the possiblity of making it so if I dial a certian extension, i can restaert asterisk? :) |
| 19:21.51 | Naikrovek | why would you need to do that? |
| 19:21.59 | lirakis | .. hmm my provider changed the signaling source of the did's ... lame |
| 19:22.05 | Naikrovek | you can do that, I think. you can run anything in the dialplan |
| 19:22.08 | talntid | because for some reason, every once in a while, my zaptel stuff stops working... |
| 19:22.18 | talntid | and restarting asterisk fixes it immediately |
| 19:22.32 | Naikrovek | i think there's a system() application or something equivalent |
| 19:22.33 | talntid | but i am not always at my computer |
| 19:28.28 | Skeeter- | talntid, what card are you using |
| 19:28.35 | talntid | sangoma card |
| 19:28.37 | Skeeter- | umm |
| 19:28.46 | Skeeter- | asterisk version? |
| 19:28.54 | Skeeter- | zaptel dahdi vervion? |
| 19:29.08 | talntid | SAN-A101D Sangoma A101D |
| 19:29.22 | talntid | Asterisk 1.4.17~dfsg-2ubuntu1, |
| 19:29.32 | Naikrovek | ubuntu package? |
| 19:29.35 | Naikrovek | or compiled |
| 19:29.39 | talntid | package |
| 19:29.40 | Skeeter- | seems a package |
| 19:29.42 | Naikrovek | (just curious, not accusing) |
| 19:29.43 | Naikrovek | ok |
| 19:29.48 | *** join/#asterisk blaines (~blaines@c-98-213-119-125.hsd1.il.comcast.net) |
| 19:29.57 | Skeeter- | Naikrovek, we might have found the problem |
| 19:30.09 | talntid | how do i find the zaptel dahdi version? |
| 19:30.28 | Skeeter- | in the CLI |
| 19:30.32 | imcdona | Does anyone know of a good click-to-call asterisk app for windows? I want to input a number, hit dial, and have the call dialed on my desk phone. |
| 19:30.33 | Skeeter- | -> dahdi version |
| 19:30.36 | Skeeter- | sth like that |
| 19:30.54 | Skeeter- | imcdona, let me know if you find one |
| 19:30.59 | talntid | No such command 'dahdi version' (type 'help' for help) |
| 19:31.01 | jshriver | core dahdi version |
| 19:31.03 | jshriver | think skeeter is right |
| 19:31.17 | Naikrovek | imcdona: i can do that with openfire and spark (XMPP server and client) |
| 19:31.18 | talntid | negative. neither works |
| 19:31.23 | Naikrovek | but it's probably not the best way to go about it |
| 19:31.42 | Skeeter- | dahdi show version |
| 19:31.47 | Skeeter- | that works for me |
| 19:32.14 | talntid | not for me :) |
| 19:32.44 | Skeeter- | zaptel show version |
| 19:32.58 | Skeeter- | i dont know anything about asterisk 1.4, someone help me here |
| 19:33.09 | *** join/#asterisk riddlebox (~rid@75-128-170-26.static.stls.mo.charter.com) |
| 19:33.29 | [TK]D-Fender | Skeeter-: DAHDI works fine with 1.4 |
| 19:33.29 | Skeeter- | talntid, type dahdi, or zaptel, then SPAM the TAB button to see possibilities |
| 19:33.35 | riddlebox | hey all, isnt there a cd you can download from asterisk.org, with just asterisk and centos? not asterisknow? |
| 19:33.44 | ChannelZ | help zap |
| 19:33.47 | [TK]D-Fender | riddlebox: `no. |
| 19:33.54 | ChannelZ | (I don't remember was it called just zap or zaptel?) |
| 19:33.58 | Naikrovek | zap |
| 19:34.02 | Skeeter- | riddlebox, we can help you to compile it tho |
| 19:34.04 | Naikrovek | but there's no way I can see to show a versino |
| 19:34.09 | jshriver | hrm after install asterisk it's not registering as a service, or do I have a /etc/init.d/asterisk how do you get those |
| 19:34.11 | talntid | http://pastebin.com/eGjF2wae |
| 19:34.12 | riddlebox | [TK]D-Fender, I guess I installed centos, then installed the yum repos then |
| 19:34.19 | talntid | dahdi doesn't show anything when spamming tab |
| 19:34.40 | riddlebox | Skeeter-, I can compile it, have done it plenty of times, I just wanted a quick download here at the office to do some testing |
| 19:34.54 | [TK]D-Fender | riddlebox: You are GUESSING what you did? We're you there when you did it? I know I'm a first-hand witness to all of MY actions... |
| 19:35.10 | Naikrovek | maybe he was stoned |
| 19:35.29 | Naikrovek | that was a joke |
| 19:35.29 | riddlebox | [TK]D-Fender, I havent touched my main box since I installed it, but I play with many test machines |
| 19:35.31 | Skeeter- | riddlebox, what i do, i install a base debian(use centos) then i untar a BLANK/FRESH install of asterisk after, the whole process takes about 15 minutes |
| 19:35.39 | talntid | does it made any difference that I am using wanpipe? |
| 19:35.52 | Naikrovek | Skeeter-: debian != centos |
| 19:35.55 | *** join/#asterisk Tech_Travis (~tech_trav@208.179.137.131) |
| 19:36.12 | Skeeter- | Naikrovek, really |
| 19:36.12 | riddlebox | Skeeter-, the debian/ubuntu packages seem stale |
| 19:36.27 | Skeeter- | where did i mention that debian = centos |
| 19:36.39 | Skeeter- | riddlebox, what do you mean by stale? |
| 19:36.47 | Naikrovek | Skeeter-: "<Skeeter-> riddlebox, what i do, i install a base debian(use centos)" |
| 19:37.08 | Skeeter- | Naikrovek, i do that with debian, he mentionned that he is using centos |
| 19:37.19 | Skeeter- | fair enough? |
| 19:37.24 | talntid | i understood what you were saying |
| 19:37.28 | riddlebox | Skeeter-, I am wanting to try out the yum repos, and see if they work out well |
| 19:37.28 | Naikrovek | i still don't |
| 19:37.45 | talntid | Naikrovek, skeeter uses debian, the other guy uses centos... |
| 19:37.46 | riddlebox | plus I havent gotten under [TK]D-Fender's skin haha |
| 19:37.52 | riddlebox | in a while |
| 19:37.53 | Naikrovek | Skeeter-: whatever, i'm not picking on you, just looked like you thought centos was a debian distro, that's all |
| 19:37.56 | Skeeter- | riddlebox, i wouldnt recommend using yum package to install asterisk or dahdi |
| 19:38.05 | talntid | Naikrovek, skeeter was saying that he uses debian, but the other guy should use centos, and just install from source |
| 19:38.06 | Skeeter- | Naikrovek, i wasnt |
| 19:38.15 | Skeeter- | Naikrovek, trying my best everyday :) |
| 19:38.39 | talntid | anyhow, back to dahdi :P |
| 19:38.59 | Skeeter- | calm down guys, simple miss understood, Naikrovek wants to clarify just to make sure eveyrone's on the same line |
| 19:39.16 | talntid | i'm calm |
| 19:39.16 | Skeeter- | wanpipe = driver for Sangoma card |
| 19:39.28 | Skeeter- | wanrouter = deamon/service |
| 19:39.37 | *** join/#asterisk waa (~waa@balrog.credipar.com.br) |
| 19:40.27 | jshriver | Where do you get the /etc/init.d/asterisk file from? didn't install from plain source |
| 19:40.45 | [TK]D-Fender | Skeeter-: http://www.despair.com/fail24x30pri.html |
| 19:40.55 | talntid | hmm, Skeeter-...... check this out: root@pbxin:/proc/zaptel# find / -name 'dahdi' |
| 19:40.55 | talntid | root@pbxin:/proc/zaptel# |
| 19:41.09 | talntid | maybe i don't use dahdi? (i didn't set it up, so i don't know) |
| 19:41.23 | Skeeter- | [TK]D-Fender, wow, hahhaha , that made my day |
| 19:41.38 | [TK]D-Fender | talntid: dahdi_cfg -vvvv |
| 19:41.50 | [TK]D-Fender | talntid: or : ztcfg -vvvv |
| 19:42.03 | [TK]D-Fender | talntid: Version & presence confirmed |
| 19:42.06 | Skeeter- | im too new for zaptel, i nvr used it |
| 19:43.13 | talntid | http://pastebin.com/kgmxmvRN |
| 19:43.43 | [TK]D-Fender | talntid: Clear enough? |
| 19:43.48 | Skeeter- | Naikrovek, how does openfire works? |
| 19:44.07 | talntid | [TK]D-Fender, I don't know what information they were looking for to help with the issue :) |
| 19:44.08 | Naikrovek | Skeeter-: openfire is a XMPP/jabber server. it has a plugin to integrate with asterisk. |
| 19:44.18 | talntid | i was only supplying the info so maybe they could assist |
| 19:44.51 | [TK]D-Fender | talntid: ... you asked how to get the version. There you go. |
| 19:44.54 | Skeeter- | talntid, dont worry, u get used to [TK]D-Fender's methods |
| 19:45.01 | Skeeter- | hasnt |
| 19:45.06 | Skeeter- | yet |
| 19:45.44 | talntid | all i know, is he knows a ton more than myself :P |
| 19:46.26 | [TK]D-Fender | "Hi, I don't know what I want" <- Madame X will read your palm now. |
| 19:46.37 | talntid | hah :) |
| 19:46.40 | jshriver | Anyone know where or what package provies the asterisk init.d file |
| 19:46.51 | [TK]D-Fender | jshriver: We don't do packages ehre |
| 19:47.05 | [TK]D-Fender | jshriver: One comes in the asterisk core tarball however |
| 19:47.15 | jshriver | ok will look inside it, I installed from source |
| 19:47.18 | TheDavidFactor | jshriver, if you're building from source do a "make config" |
| 19:47.31 | jshriver | oh that did it ty |
| 19:48.19 | talntid | [TK]D-Fender, Quick Q: If I wanted to restart asterisk, after dialing an extension, what should I look up to accomodate this? |
| 19:49.06 | Skeeter- | TheDavidFactor, i always used "make samples" wat does make config do |
| 19:49.13 | [TK]D-Fender | talntid: You were handed that answer half an hour ago... |
| 19:49.22 | talntid | system() |
| 19:49.45 | [TK]D-Fender | talntid: head of the class with you.... |
| 19:49.52 | Skeeter- | talntid, System(asterisk -rx"core stop now") |
| 19:49.55 | TheDavidFactor | Skeeter-: it creates the init.d script so you can start and stop asterisk with the service command on RH/CentOS |
| 19:49.57 | Skeeter- | wait |
| 19:50.00 | talntid | I was wondering if there was an internal command though. |
| 19:50.01 | Naikrovek | Skeeter-: restart |
| 19:50.03 | [TK]D-Fender | Skeeter-: NOT smart.... |
| 19:50.11 | Skeeter- | Naikrovek, you are right |
| 19:50.17 | Skeeter- | thats why i said wait |
| 19:50.19 | [TK]D-Fender | talntid: Nope |
| 19:50.19 | talntid | self(restart) or something. |
| 19:50.22 | Naikrovek | talntid: no |
| 19:50.25 | talntid | ok, that's what iw as curious about :) |
| 19:51.23 | Skeeter- | well using "make config" from TheDavidFactor suggestion, i would System(/etc/init.d/asterisk restart) |
| 19:52.03 | riddlebox | Skeeter-, can I msg you? |
| 19:52.22 | Skeeter- | riddlebox, i dont accpet childhood pr0n |
| 19:52.30 | riddlebox | dont worry |
| 19:53.01 | Skeeter- | i suggest you ask ur questin here, im probably the last person that will answer your questin correctly |
| 19:53.08 | talntid | lol |
| 19:53.15 | Skeeter- | im just a smart a$$ |
| 19:53.16 | talntid | second to last. |
| 19:53.25 | [TK]D-Fender | Skeeter-: But probably the first to answer it incorrectly ;) |
| 19:53.36 | [TK]D-Fender | SPEED KILLS |
| 19:53.40 | Skeeter- | [TK]D-Fender, true that |
| 19:53.48 | *** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 19:53.55 | [TK]D-Fender | (or more directly your body's overreaction to the sudden stop at the end) |
| 19:54.08 | riddlebox | well I didnt want to start an argument...but why wouldnt you want to use Digiums Yum repos? |
| 19:54.16 | Skeeter- | i like answer incorrectly(altho its not on prupuse) at least i can get corrected |
| 19:54.52 | [TK]D-Fender | riddlebox: New versions fix old bugs and introduce new ones. Do you want a perfectly functioning server upgraded without your prior testing behind your back? |
| 19:55.06 | Skeeter- | I always been told NEVER to use yum install, or apt-get install asterisk packages |
| 19:55.21 | talntid | When I was in the military, I was off with a team who had to diffuse an IED (improvised explosive device) ... they had "bomb squad" on the radio, and one of my teammates was at the bomb. he sent a picture to the squad who quickly responded to cut the wire on the left side... he cut it, right in time to hear "no wait! second from the right!" |
| 19:55.36 | Naikrovek | ooh |
| 19:55.42 | talntid | luckily, the bomb didn't go off. but he did need a change of underwear. |
| 19:56.12 | TheDavidFactor | tarballs FTW! |
| 19:56.15 | Naikrovek | picture message bomb squad |
| 19:56.17 | TheDavidFactor | ;-) |
| 19:56.21 | riddlebox | that is true, but generally how often do you upgrade a system once it is in? I have customers who has asterisk systems not able to get onto their internet, it is just a voice network |
| 19:56.22 | Skeeter- | +1 |
| 19:56.29 | Naikrovek | I was EOD & AMMO in the Air Force |
| 19:56.33 | Naikrovek | is trained in explosives |
| 19:56.36 | Naikrovek | was, rather |
| 19:56.38 | talntid | nice =) |
| 19:56.39 | Naikrovek | it was a long time ago |
| 19:56.40 | talntid | 2m0x2 here |
| 19:56.44 | talntid | ICBM Maintenance |
| 19:57.08 | ChannelZ | I'm trained in explosive... diarrhea |
| 19:57.15 | talntid | that's deadly. |
| 19:57.22 | Skeeter- | wonders is defusing a bomb with in Glock 17 would be fore effective. Wouldnt need to change his underwear |
| 19:57.24 | TheDavidFactor | I didn't want to know that |
| 19:57.34 | Naikrovek | so many innocent underwear, killed in their prime |
| 19:57.43 | Skeeter- | lol |
| 19:57.45 | *** join/#asterisk KuArZo (~kuarzo@kuarzo-4-pt.tunnel.tserv4.nyc4.ipv6.he.net) |
| 19:57.49 | talntid | Skeeter- G17.. blah :P |
| 19:57.59 | Skeeter- | talntid, 1st gun |
| 19:58.07 | ChannelZ | Combat Tupperware! |
| 19:58.23 | talntid | HK USP .40 = yum :) (my carry piece) |
| 19:58.28 | ChannelZ | But, hurray for having anything |
| 19:58.44 | talntid | haha ChannelZ.. ;) |
| 19:59.05 | Skeeter- | talntid, im looking for a .45 or 50 CAL but its hard to get one in Canada where u can get ammo for it |
| 19:59.06 | ChannelZ | carries a Kimber Eclipse .45 |
| 19:59.21 | riddlebox | alright back to reading Digiums licenses agreements, got a customer who wants to put in a Switchvox system, want to make sure I get everything right for them |
| 19:59.23 | Skeeter- | carry |
| 19:59.28 | Skeeter- | we cant carry here |
| 19:59.46 | ChannelZ | I didn't even think you could get handguns there without selling your balls to someone in the govt |
| 19:59.49 | talntid | Skeeter-, why .50? |
| 19:59.51 | Naikrovek | riddlebox: we're not really switchvox experts here. perhaps #switchvox ? |
| 20:00.00 | Skeeter- | talntid, Desert Eagle, the blast |
| 20:00.01 | talntid | the deagle is 1000% crap. unreliable |
| 20:00.08 | riddlebox | Naikrovek I never asked a question it was a statement |
| 20:00.10 | Skeeter- | talntid, that im afraid off |
| 20:00.22 | Naikrovek | riddlebox: ah i misread you |
| 20:00.23 | talntid | it's OK for the fun factor, but never trust it to go off. |
| 20:00.24 | Skeeter- | talntid, i looked for a Smith Wesson too |
| 20:00.42 | Skeeter- | Revolver series |
| 20:00.49 | talntid | pick up a nice 1911 or USP .45 :) |
| 20:00.50 | riddlebox | Naikrovek I am not to excited about it, but their IT guys want it |
| 20:01.02 | talntid | ChannelZ has the right idea with Kimber. but they are expensive |
| 20:01.09 | Naikrovek | riddlebox: yeah i question the switchvox decision, but i don't know a lot about it |
| 20:01.34 | talntid | then again, USP's are expensive too. but my life is worth it. :) |
| 20:01.38 | ChannelZ | Is it Taurus whose making some 1911s for reasonable prices |
| 20:01.45 | talntid | yes, i have one |
| 20:01.47 | Skeeter- | talntid, price isnt a problem |
| 20:01.57 | Skeeter- | talntid, have you seen the Dan Welson |
| 20:01.59 | talntid | very very very comparable to the kimber, actually, the taurus PT1911 |
| 20:02.11 | Naikrovek | was NOT expecting gun talk in here |
| 20:02.15 | ChannelZ | Yeah I've heard they're pretty nice |
| 20:02.28 | ChannelZ | Great way to get into a 1911 for <$1000 :) |
| 20:02.31 | talntid | yeah, i have one.. ;) |
| 20:02.36 | [TK]D-Fender | Springfield Armory XD .45 <- |
| 20:02.37 | talntid | uh, try <$700! |
| 20:02.43 | talntid | very good choice, [TK]D-Fender |
| 20:02.46 | Skeeter- | i paid my glock 17 700$ |
| 20:02.49 | [TK]D-Fender | 1911... 6-round clip.. sad |
| 20:02.51 | Skeeter- | but thats in Canada |
| 20:02.52 | ChannelZ | I thought they were even cheaper than that |
| 20:02.54 | talntid | is it XDM or old XD? |
| 20:03.12 | [TK]D-Fender | talntid: Don't own.. jsut stating my likely choice. |
| 20:03.16 | talntid | ah, i see :) |
| 20:03.17 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 20:03.30 | talntid | they are great. |
| 20:03.34 | ChannelZ | There are 8 round 1911 mags |
| 20:03.38 | *** join/#asterisk voxter (~voxter@S010600090f53ea17.vc.shawcable.net) |
| 20:03.43 | Alric | Glock 19 or M&P9 |
| 20:03.54 | Skeeter- | Canada laws limit us to 10 rounds mag |
| 20:04.00 | voxter | Is there a way to find out which release version a bugfix made it into, if the bugfix was submitted to trunk? |
| 20:04.02 | talntid | 9mm is not impressive. |
| 20:04.30 | talntid | i have watched people get up from 9mm hits. multiple hits. over in the sandbox |
| 20:04.32 | [TK]D-Fender | talntid: .45 is nice big caliber, better stopping power, still cheap to reload, not as nasty on recoil as .357/.44. .40/10mm is too nominal a change for increased cost. Then again, depends if you intend to practice regularly with it |
| 20:04.33 | Skeeter- | talntid, i need/want a .45 revolver |
| 20:04.36 | ChannelZ | voxter: You can figure out a certain amount by the Changelog rXXXXXX numbers and trying to determine if something in the bug database is newer or older than the release |
| 20:04.49 | [TK]D-Fender | ske.45 ....REVOLVER? |
| 20:04.57 | Alric | Shot placement will always trump caliber. |
| 20:05.12 | ChannelZ | "Location, location, location..." |
| 20:05.13 | [TK]D-Fender | Skeeter-: Only revolvers worth considering is .44mag or .454 Cassel |
| 20:05.24 | Alric | talntid: .45 if you have to use FMJ. 9mm if you have no restrictions. |
| 20:05.28 | Skeeter- | http://images.google.ca/images?um=1&hl=en&client=firefox-a&rls=org.mozilla%3Aen-US%3Aofficial&tbs=isch%3A1&sa=1&q=dan+wesson&aq=f&aqi=g10&aql=&oq=&gs_rfai=&start=0 |
| 20:05.32 | talntid | [TK]D-Fender, I use .40 in my USP, and it's because of the cost of ammo. .40 is about 30% cheaper than .45 where I am |
| 20:05.50 | Skeeter- | [TK]D-Fender, you base that on what? |
| 20:06.07 | talntid | Alric, I have no restrictions, but I still prefer .40. I want them to stay down. ;) |
| 20:06.21 | [TK]D-Fender | talntid: My personal odd-ball choice : CZ 52. W .32 Tokarev it punches through class-3 kevlar like rice-paper |
| 20:06.24 | voxter | ChannelZ: rXXXXX, XXXXX being the bug ID? |
| 20:06.36 | [TK]D-Fender | Skeeter-: I base that on JOULES. |
| 20:06.44 | Skeeter- | ok... |
| 20:06.53 | ChannelZ | voxter: well no in the bug you'll see fixes and things checked in with the revision number |
| 20:06.54 | talntid | never used one, [TK]D-Fender... |
| 20:07.04 | voxter | ChannelZ: oh. right. duh. thanks |
| 20:07.06 | [TK]D-Fender | talntid: increadibly inexpensive |
| 20:07.15 | Alric | I can't believe I just got sucked into a caliber debate. |
| 20:07.16 | talntid | a few of my weapons: http://spokanespeed.net/media/guns.jpeg |
| 20:07.28 | talntid | Alric, worse, it's in a geek channel :P |
| 20:07.41 | Skeeter- | talntid, WHAT!!!! |
| 20:07.46 | Skeeter- | talntid, where do you live |
| 20:07.50 | talntid | spokane, WA |
| 20:07.52 | Skeeter- | talntid, are they all real |
| 20:07.58 | talntid | uh, yeah. |
| 20:08.00 | talntid | lol. |
| 20:08.09 | Alric | I thought SBRs were illegal in WA. |
| 20:08.11 | Naikrovek | why do you need all those guns |
| 20:08.16 | talntid | thats an old picture, i have more now. |
| 20:08.18 | [TK]D-Fender | talntid: Whats the bull-pup model? |
| 20:08.19 | Naikrovek | you got like 12 arms or something |
| 20:08.27 | talntid | Alric, $5 tax stamp |
| 20:08.30 | ChannelZ | I need an AR next |
| 20:08.34 | [TK]D-Fender | And AR-15's? Ew... |
| 20:08.34 | talntid | the silencers, $200 tax stamp each |
| 20:08.44 | talntid | Naikrovek, because it's a hobby. ;) |
| 20:08.46 | Alric | I understand NFA. |
| 20:08.50 | ChannelZ | talntid: geekswithguns.com :) |
| 20:08.51 | talntid | [TK]D-Fender, Fn FS2000 |
| 20:08.57 | riddlebox | ha I just bought me a H&R single shot, 12 guage slug gun with the scope |
| 20:09.34 | talntid | that AR with the suppressor, is a AR-10, .308 with silencer. sounds like a pellet gun. |
| 20:09.42 | voxter | ChannelZ: hmm, doesnt seem to be revision stamps in changelog |
| 20:10.11 | Naikrovek | i don't get hobbies that involve collecting unnecessary things, but whatever. have fun. |
| 20:10.13 | talntid | Alric, hey, sorry, i read your thing wrong. SBR's ARE illegal in WA |
| 20:10.25 | talntid | but AOW (short shotty) isn't |
| 20:10.26 | Skeeter- | riddlebox, "12 guage slug gun with the scope" Welcome to Zombieland. |
| 20:10.30 | talntid | nothing else there is SBR |
| 20:10.36 | Alric | talntid: Yeah, I realized there was no stock on that short barrel AR. |
| 20:10.38 | [TK]D-Fender | talntid: .308 ... much better... I wouldn't want a .223 |
| 20:10.55 | talntid | Alric, that's a Keltex PLR-16 |
| 20:11.01 | talntid | keltec |
| 20:11.12 | Alric | meh. Thats what I get for cursory looks. |
| 20:11.12 | talntid | [TK]D-Fender, the green AR is a .223 :) so is the FS2000 |
| 20:11.13 | voxter | ahh/ got it. |
| 20:11.18 | riddlebox | Skeeter-, shot it a few times already seems to be very nice, 3inch barrel on it too |
| 20:11.20 | [TK]D-Fender | talntid: I know... EW! |
| 20:11.27 | talntid | no way ;) |
| 20:11.30 | ChannelZ | voxter: not always which is why it sucks - but sometimes you'll see things like "2010-03-03 21:25 +0000 [r250611]" |
| 20:11.33 | talntid | suppressed FS2000 is very fun |
| 20:11.35 | [TK]D-Fender | talntid: Great alternate : Ruger Mini-30 |
| 20:11.50 | voxter | ChannelZ: searched google for the revision number and found it buried in a revision summary |
| 20:12.12 | Alric | 5.56mm is fine |
| 20:12.26 | ChannelZ | yeah I wish they'd always put a definitive revision number in the log for a released build so you could tell |
| 20:12.43 | ChannelZ | maybe there's some better way to figure it out but I dunno. I am not an SVN MASTAR |
| 20:12.49 | talntid | [TK]D-Fender, yeah.. :) |
| 20:13.01 | *** join/#asterisk op3r (~op3r@114.108.202.165) |
| 20:13.09 | ChannelZ | They will randomly mention specific issue #s in Changelog sometimes |
| 20:13.14 | op3r | hello does anyone know how to hangup the waitforsilence? |
| 20:13.36 | op3r | im getting flooded with this [Apr 8 13:13:21] -- Got 0ms silence< 2000ms required |
| 20:13.36 | op3r | [Apr 8 13:13:21] -- Got 0ms silence< 2000ms required |
| 20:13.43 | p3nguin | ~pb |
| 20:13.44 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 20:13.45 | ChannelZ | scampers off to find some lunch before the wrath |
| 20:13.55 | op3r | oooopps sorry |
| 20:14.10 | talntid | [TK]D-Fender, mini 30 is on the list of to-buy... very accurate |
| 20:15.08 | [TK]D-Fender | talntid: Large stock & magazine options as well |
| 20:15.12 | talntid | yes |
| 20:16.41 | talntid | bbiab |
| 20:17.19 | t[a]lntid | btw, before i go... |
| 20:17.28 | t[a]lntid | http://spokanespeed.net/media/hkcrossbreed.jpg <---- most comfortable carry setup. ;) |
| 20:17.55 | boch | op3r, WaitForSilence(silencerequired[|iterations][|timeout]): use timeout to set the max wait for silence |
| 20:18.05 | Alric | <3 Crossbreed |
| 20:18.34 | t[a]lntid | that thing, has been all over iraq with me. |
| 20:18.42 | t[a]lntid | absolutely awesome holster. |
| 20:19.03 | t[a]lntid | ok, gone |
| 20:19.38 | Fubard | GunOIP |
| 20:19.46 | Fubard | i'd pay good money for that |
| 20:20.15 | Skeeter- | Fubard, its called a Hitman |
| 20:20.50 | Fubard | Skeeter-: lol, evil sauce |
| 20:21.12 | Skeeter- | comments must never be taken seriously |
| 20:21.27 | Skeeter- | unless state |
| 20:25.02 | Fubard | ~which versioning |
| 20:25.20 | Fubard | anyone recall the bot call for branching info? |
| 20:25.35 | Fubard | ~whichversion |
| 20:25.42 | voxter | Can anyone think of an issue between asterisk 1.2 DTMF and asterisk 1.4 DTMF using RFC2833? Ive already enabled rfc2833compensate=yes but that doesnt seem to have completely fixed it. |
| 20:25.50 | Chainsaw | ~canbotcomeoutandplay |
| 20:26.04 | Skeeter- | ~Chainsaw |
| 20:26.05 | infobot | well, chainsaw is "<action> fires up her Stihl and lops off a random appendage from skeeter-" |
| 20:26.26 | Skeeter- | not funny |
| 20:26.31 | Fubard | lol |
| 20:26.42 | Skeeter- | heres ur evil sauce |
| 20:28.02 | [TK]D-Fender | checkout time, later all |
| 20:29.08 | Fubard | found it: ~asterisk-versioning and ~asterisk-versions |
| 20:30.12 | *** join/#asterisk randomuser (~bob@97-121-199-167.blng.qwest.net) |
| 20:30.50 | randomuser | What does asterisk require from my communications provider? will standalone DSL suffice? |
| 20:31.08 | p3nguin | randomuser: That depends on your usage. |
| 20:31.43 | p3nguin | randomuser: If your want to make a couple VoIP calls, a moderate speed DSL is probably okay. |
| 20:32.15 | Fubard | randomuser: http://www.connect802.com/voip_bandwidth.php |
| 20:32.22 | randomuser | p3nguin: learning, for the most part, but I do have a specific function in mind: dialing out to an automated system and punching in some numbers |
| 20:32.38 | p3nguin | randomuser: If you already have DSL, use it. |
| 20:32.53 | patrb | I have an AGI script that I'm using for logging to a log file as a call progresses though the context the users proggression gets logged. Is there a way I can detect if the user ends the call in a dial plan and then pass it to my agi script? |
| 20:33.16 | randomuser | p3nguin: 'it' as in asterisk? |
| 20:33.49 | Fubard | randomuser: p3nguin meant 'use the DSL' by 'it' |
| 20:34.14 | patrb | err short question: is there a way to detect if a user ends the call within a dialplan? |
| 20:34.52 | Fubard | patrb: if you sip set debug on, you'll see when a call is hungup |
| 20:35.07 | Fubard | patrb: or do you mean to use that as an event in the dialplan? |
| 20:35.26 | patrb | Fubard: yes, exactly...I need to pass it to log that event in a custom log file |
| 20:35.58 | p3nguin | randomuser: Even the slowest DSL available should be able to handle one ulaw/alaw VoIP phone call. |
| 20:36.24 | WIMPy | p3nguin: No |
| 20:36.31 | randomuser | i'm reading that response as 'use your internet connection to make the call' - something that i understood asterisk to make possible |
| 20:36.47 | p3nguin | wimpy: Why not? |
| 20:37.02 | p3nguin | wimpy: Maybe you have slower DSL than we have here. Who knows? |
| 20:37.36 | WIMPy | p3nguin: In rural areas we often still have a max of 384/64 or even worse. |
| 20:38.07 | Skeeter- | WIMPy, +1 |
| 20:38.08 | Fubard | i'm not certain on this, but i think you can use a 56k modem to make analog calls if you get all macguyver on it and are really good with programming |
| 20:38.22 | WIMPy | The slowest official product sold will be enough, but not neccessarily what you really get. |
| 20:38.33 | p3nguin | Our slowest DSL (not outside the city) is 768k/384k. |
| 20:38.44 | Skeeter- | p3nguin, WATH!! |
| 20:38.51 | randomuser | mine is 7M... |
| 20:38.53 | Skeeter- | p3nguin, thats above our best |
| 20:39.00 | WIMPy | Fubard: You can do it on a 2400bps modem, but don't ask what it sounds like. |
| 20:39.17 | Skeeter- | at work i got 1800k/50k |
| 20:39.25 | p3nguin | Outside the city on those rural telcos, they will provide slower speeds... like 512k/128k. |
| 20:39.26 | Fubard | patrb: i'm still trying to find out the event name but its probably HangUp() |
| 20:39.49 | patrb | Fubard: thanks, I use HangUp() to actually hang up the channel on my end |
| 20:40.00 | patrb | Fubard: is there a way to use that to detect if the user hung up? |
| 20:40.08 | Fubard | WIMPy: my sense of humor is very very dry, i was referring to actually making a normal POTS phone call :) |
| 20:40.13 | voxter | Interesting. SIP (phone) -> SIP (Asterisk 1.2) -> SIP (Asterisk 1.4) -> SIP (ITSP) all using RFC2833, DTMF is not picked up at the ITSP (but debug shows it is being picked up at ast 1.4 box) . Doing IAX between 1.2 -> 1.4 box, DTMF works fine. Very confused. |
| 20:40.31 | p3nguin | Our best residential DSL is 6m/768k. |
| 20:40.43 | WIMPy | Fubard: What does that have to do with bitrates then? |
| 20:40.48 | Skeeter- | 6mbs real |
| 20:40.58 | Skeeter- | or 800kbs real speed p3nguin |
| 20:41.10 | p3nguin | does not parse |
| 20:41.39 | p3nguin | If they say the offer 6Mbit service, you'll see close to 6Mbit rates. |
| 20:44.15 | *** join/#asterisk spenguin[work] (~penguin@122.182.0.38) |
| 20:45.08 | randomuser | well, i'll give it a try. thanks. |
| 20:45.23 | patrb | Skeeter: I think you mean 6Mbps = 800KBps...but i think its actually higher than that |
| 20:45.38 | p3nguin | Now on cable, we have lots more speed available. The best residential service available is the Ultra60 package, which is 60 Mbps down and upstream is 5 Mbps. |
| 20:46.10 | jshriver | anyone know how to fix this problem: |
| 20:46.12 | jshriver | chan_dahdi.c:1776 dahdi_enable_ec: Unable to enable echo cancellation on |
| 20:46.34 | jshriver | and when I call out all I can hear is myself, and the other end is very very silent |
| 20:46.58 | patrb | jshriver: try editing chan_dahdi.conf: echocancel=yes to no |
| 20:47.16 | jshriver | ok let me check, it's odd cause one of the cards has hardware echo the other doesnt |
| 20:47.27 | *** part/#asterisk randomuser (~bob@97-121-199-167.blng.qwest.net) |
| 20:47.33 | patrb | jshriver: two cards, one box? |
| 20:47.42 | jshriver | yup |
| 20:47.53 | jshriver | 1 card has 4 FXO, the other has 2 for six total lines |
| 20:47.54 | patrb | jshriver: out of my element :) still a dahdi newb myself |
| 20:48.01 | jshriver | hehe dont feel bad |
| 20:48.41 | patrb | jshriver: if the call is quiet, you can change the rx or txgains |
| 20:48.46 | patrb | in chan_dahdi.conf |
| 20:49.09 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:50.18 | patrb | So..anyone? Need to detect if a user ends a call and log it to a custom log file, anyway to do this in the dialplan? |
| 20:50.41 | Skeeter- | mbit and mbyt confuse me |
| 20:51.10 | patrb | skeeter: just remember big B is bigger |
| 20:52.23 | [TK]D-Fender | Skeeter- is definitely a few bits short of a byte... |
| 20:52.29 | Fubard | Skeeter-: most people forget to put a capitol B when talking about a byte so its very hard to trust what someone types. on my workstation the download manager says "Bitrate: xxx MBps" |
| 20:52.50 | patrb | Fubard: good point |
| 20:53.01 | [TK]D-Fender | Fubard: patrb And you both read him wrong. |
| 20:53.20 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 20:53.28 | patrb | Please clarify |
| 20:53.31 | [TK]D-Fender | bit VS byte, not "m" VS "M" |
| 20:53.47 | [TK]D-Fender | [16:50]<Skeeter->mbit and mbyt confuse me <- |
| 20:53.49 | Skeeter- | here we got again |
| 20:53.59 | patrb | never mentioned m vs M |
| 20:54.20 | Fubard | i think we're in the twilight zone patrb |
| 20:54.26 | patrb | :) |
| 20:54.32 | [TK]D-Fender | patrb: Is Byte bigger than byte? |
| 20:54.37 | patrb | yes |
| 20:54.38 | patrb | err |
| 20:54.41 | [TK]D-Fender | :D |
| 20:54.42 | patrb | byte is bigger than bit |
| 20:54.43 | [TK]D-Fender | pwned |
| 20:54.54 | patrb | hardly, misread |
| 20:54.58 | patrb | calm down son |
| 20:55.04 | Fubard | MB is bigger than Mb |
| 20:55.09 | Skeeter- | still pwned |
| 20:55.42 | patrb | damn..cant do look of disapproval here |
| 20:55.54 | ChannelZ | :| |
| 20:55.59 | patrb | ?_? |
| 21:00.44 | Skeeter- | think that speakeasy.net/speedtest is a good tool |
| 21:00.45 | Skeeter- | They advertise in Byte cuz its bigger |
| 21:00.46 | patrb | uhh..they advertise bit because it sounds bigger |
| 21:00.46 | Skeeter- | wow |
| 21:00.46 | Skeeter- | i dotn get it |
| 21:00.46 | Chainsaw | Skeeter-: If you're in the US, sure. |
| 21:00.47 | Skeeter- | Canada |
| 21:00.47 | Fubard | if i'm an average consumer and i see 1MB or 8Mb, i'd assume the larger number is better, though they're the exact same |
| 21:00.50 | Skeeter- | i know that the difference is about 1/8 |
| 21:00.52 | Skeeter- | or 8/1 |
| 21:00.53 | Skeeter- | damn i hate late afternoon, brains start shaking |
| 21:00.53 | Skeeter- | probably the Cooldown of the Reb Bull |
| 21:00.54 | Fubard | shake the screen while reading, to compensate for the shaking brain. might make it easier to understand? XD |
| 21:00.55 | Skeeter- | shakes his screen. Effectiveness: from 0 to none. |
| 21:00.56 | Fubard | you understand it, you just dont trust your gut. play with the math some, til you trust what you know |
| 21:00.56 | Skeeter- | im out, peace out guys |
| 21:00.56 | Fubard | cya skeeter |
| 21:00.56 | patrb | later skeeter |
| 21:00.56 | Skeeter- | wow |
| 21:01.03 | Fubard | you have powers you dont yet understand skeeter ... make another net split happen! |
| 21:01.04 | Skeeter- | i just timeout 5guys |
| 21:01.18 | patrb | thats what she said |
| 21:02.30 | Fubard | if she didn't have tourette's, i'd be a little more shocked. |
| 21:02.48 | patrb | hehe |
| 21:02.49 | Fubard | i'm always hearing about what she's said in the past. that woman has a mouth on her. |
| 21:04.03 | Fubard | patrb, i propose we begin a campaign. the anti-'thats what she said' campaign. everytime we feel inclined to say 'thats what she said' i move that we say 'well shes never said that before' in its place |
| 21:04.13 | Fubard | we'll be heros. visionaries! |
| 21:04.21 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 21:04.24 | Fubard | hell, we might even figure out how to spell visionairies* |
| 21:04.31 | Katty | hi |
| 21:04.48 | Fubard | katty, does linux make run? |
| 21:04.53 | Fubard | please help |
| 21:06.47 | patrb | hey katty |
| 21:07.06 | Katty | jaytee: you around? |
| 21:07.39 | patrb | Fubard: lol, i like your plan |
| 21:07.43 | patrb | fubard: im on board |
| 21:08.00 | Fubard | patrb: still not finding anything on detecting call hangups. tbh its a bit over my head, i'm still learning. |
| 21:08.26 | patrb | Fubard: thanks...just posted on the forums http://forums.digium.com/viewtopic.php?f=1&t=73687&p=143567&sid=fb885b8260a13bf1070944d1f6ca1205#p143567 |
| 21:08.40 | patrb | Fubard: hopefully I'll get some ideas soon |
| 21:10.07 | jshriver | Any ideas on what would cause static and low audio quality on a specific port/FXO when it's not the phone line itself? |
| 21:10.26 | [TK]D-Fender | Fubard: "heroes" |
| 21:10.27 | jshriver | and doesnt seem to understand DTMF very well |
| 21:10.59 | Chainsaw | jshriver: "Understanding" DTMF requires good audio quality, so that's a symptom, not a cause. |
| 21:11.08 | jshriver | ok |
| 21:11.09 | Chainsaw | jshriver: Where is this FXO port located? On a PCI card? |
| 21:11.12 | jshriver | aye |
| 21:11.14 | jshriver | on a TDM410 |
| 21:11.22 | Chainsaw | jshriver: Is the molex power stable? |
| 21:11.27 | jshriver | I used the same line on all 6 ports, and 2 of the 4 are really staticy |
| 21:11.40 | jshriver | yeah |
| 21:11.44 | Chainsaw | jshriver: i.e. not coming from some evil splitter of doom. |
| 21:12.00 | jshriver | hrm let me check just to be sure..brb |
| 21:12.10 | [TK]D-Fender | TDM410P doesn't need the molex for FXO use |
| 21:13.06 | Chainsaw | [TK]D-Fender: But I'm sure the hardware echo canceller would like it. |
| 21:13.15 | Fubard | [TK]D-Fender: thank you, you are my heroe. |
| 21:15.58 | jshriver | could havnig molex connected when using FXO's fry it? |
| 21:16.42 | voxter | I seem to have found a DTMF bug with RFC2833 between asterisk 1.2 and 1.4 on the SIP channel. Anyone have much experience with this before i submit it to the tracker? |
| 21:16.58 | Qwell | jshriver: no, the power from the molex is supplied to the FXS modules |
| 21:17.18 | *** join/#asterisk aztyiop (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net) |
| 21:17.23 | aztyiop | hello |
| 21:17.27 | *** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 21:17.55 | aztyiop | i need kind of billing management system for asterisk |
| 21:18.13 | aztyiop | which one can advice me ? |
| 21:18.18 | *** join/#asterisk b14ck (~comradeb1@cpe-76-95-129-196.socal.res.rr.com) |
| 21:18.58 | leifmadsen | aztyiop: some people use a2billing. Last night I just met guys from www.dthsoftware.com that do billing as well |
| 21:19.49 | jshriver | so I don't need to connect the molex if using FXO chips? |
| 21:20.02 | jshriver | just wanted to make sure, going to pull these two bad ones and try with 2 new ones. |
| 21:20.23 | jshriver | keep in mind, this is coming from a card just sent to me from digium, who spent 2 days with a special team and said they were good |
| 21:20.44 | aztyiop | sound not interesting leifmadsen |
| 21:20.54 | aztyiop | sound like not interesting leifmadsen |
| 21:21.00 | aztyiop | that one |
| 21:21.02 | leifmadsen | I still don't understand what that means |
| 21:21.16 | aztyiop | that mean what it mean |
| 21:21.24 | spenguin[work] | heh |
| 21:21.32 | leifmadsen | uh ok :) |
| 21:21.43 | aztyiop | first it is not an opensource |
| 21:21.47 | aztyiop | one |
| 21:21.53 | aztyiop | i can't see the demo |
| 21:21.57 | leifmadsen | oooooooh so you want to make money and not pay anything for it. Gotcha. |
| 21:22.04 | aztyiop | there is no screenshot about that product |
| 21:22.22 | aztyiop | only all incovenience to hate that product |
| 21:22.24 | aztyiop | lol |
| 21:22.25 | leifmadsen | I'm sure calling them or emailing them would get you a free demo. Most software businesses do that kind of thing. |
| 21:22.34 | leifmadsen | I was just throwing out examples |
| 21:22.36 | leifmadsen | a2billing is free |
| 21:22.41 | leifmadsen | go that route if you wish |
| 21:23.02 | spenguin[work] | a2billing seems nice |
| 21:24.13 | aztyiop | i allready got a2bill |
| 21:24.45 | aztyiop | i need some kind of web based billing management thing for callshop |
| 21:25.09 | leifmadsen | moves on |
| 21:25.27 | aztyiop | and i think a2bill don't give that functionality |
| 21:25.43 | leifmadsen | you get what you pay for |
| 21:26.21 | Fubard | leif suggested dthsoftware. they seem to have a webportal. and you'll find pictures on their website which i'm sure will excite your corneas |
| 21:27.13 | Fubard | Harvard studies have shown, that if you want your corneas excited, you'll just have to pay for it |
| 21:29.06 | aztyiop | what is the cost of that one ? |
| 21:29.08 | p3nguin | Now if they would just teach punctuation. |
| 21:29.11 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 21:29.47 | Fubard | I tend to, over use commas, on occassion. |
| 21:30.58 | *** join/#asterisk Ad-Hoc (~nimbus@194.219.186.16.dsl.dyn.forthnet.gr) |
| 21:31.03 | p3nguin | While you were in the line of fire on that, you were not the specific target of my frustrations. |
| 21:31.08 | aztyiop | is there anyone which is free ? |
| 21:31.15 | Fubard | :) |
| 21:31.29 | aztyiop | free for ever |
| 21:31.46 | p3nguin | free freE frEE fREE FREE!!!!! |
| 21:31.47 | Alric | The nature of billing systems doesn't usually lend themselves to the word "free" |
| 21:32.45 | aztyiop | who's that philosoph ? |
| 21:34.26 | *** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net) |
| 21:36.37 | Alric | Its just been my experience that complete software used to bill people usually comes with a price tag. |
| 21:38.13 | p3nguin | I think I would rather let someone else do billing and I'll just make sure the systems work. |
| 21:39.04 | *** join/#asterisk idespinner (~idespinne@cpe-76-93-115-243.socal.res.rr.com) |
| 21:39.27 | Alric | I hear you there. |
| 21:40.08 | hardwire | on IRC shouldn't we say "I read you there." |
| 21:40.16 | hardwire | or "I'm readin' you, dawg" |
| 21:40.24 | Slugs_ | lol |
| 21:41.13 | Slugs_ | I'm seeing what your puttin in front of my face |
| 21:41.21 | jdoe | ... another polycom question... I've got MWI working, for the most part, and that's good. The only problem is that if I reboot a phone with voicemail waiting, when it comes back it never gets the MWI unless I login to voicemail or leave a new vm... anyone know why that would be? |
| 21:44.51 | *** join/#asterisk Slugs_ (~yeah@173-8-52-45-Jacksonville.hfc.comcastbusiness.net) |
| 21:45.08 | *** join/#asterisk ido- (~wtf@94.159.203.234) |
| 21:45.47 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 21:46.09 | ido- | how do I add to my dialplan that when i dial 9#3, it'll pass to the trunk #3 ?, my problem is that it won't pass the # sign |
| 21:46.42 | *** join/#asterisk b14ck (~comradeb1@cpe-76-95-129-196.socal.res.rr.com) |
| 21:46.49 | idespinner | ido-, what kinda phone? # is usually a reserved digit |
| 21:47.01 | idespinner | you maybe better off using * |
| 21:47.40 | ido- | hrm. the outgoing line is connected to another pbx, which extentions #1, #2, #3 |
| 21:47.45 | ido- | and its not up to me to change that |
| 21:48.20 | [TK]D-Fender | ido-: What won't pass it? |
| 21:48.31 | [TK]D-Fender | ido-: Asterisk will dial out whatever you tell it to |
| 21:48.33 | ido- | hrm, I also tried using *, and it didn't pass that either. |
| 21:48.48 | ido- | my pbx won't pass the #,* to the outgoing trunk |
| 21:48.53 | [TK]D-Fender | ido-: "it"? Please assign proper names to these impoersonal nouns... |
| 21:49.05 | ido- | I'm using pbxes.org |
| 21:49.18 | [TK]D-Fender | ido-: Sorry, this is #asterisk |
| 21:50.09 | ido- | oh, i think i know why |
| 21:51.24 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 21:55.09 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-101-155-51.cinci.res.rr.com) |
| 21:55.19 | jdoe | ... oh, adding subscribe fixed it. Wonder why every single asterisk + polycom + mwi guide ignores subscribe. |
| 21:56.17 | Fubard | jdoe: was just looking at what i have setup. you referring to <mwi msg.mwi.1.subscribe="> ? |
| 21:58.31 | jdoe | yeah. |
| 21:58.50 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 21:58.55 | jdoe | with mwi.1.subscribe blank, mwi works fine except right after the phone is rebooted. If I check voicemail or leave a new one, then the light comes on. |
| 21:59.31 | jdoe | with mwi.1.subscribe as the phone's extension, mwi works as expected (if I leave a message before the phone is turned on, mwi is on when it finishes booting) |
| 21:59.58 | Fubard | aaaaaah, i see. |
| 22:00.40 | jdoe | hmm. Or not. |
| 22:01.24 | jdoe | yeah, mwi only comes on after I check voicemail explicitly. That's not cool. |
| 22:03.20 | Fubard | lemme paste what i'm using, one second |
| 22:04.20 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 22:04.33 | *** part/#asterisk ido- (~wtf@94.159.203.234) |
| 22:05.01 | Fubard | http://pastebin.com/MymSeZcp |
| 22:07.08 | jdoe | which phone? |
| 22:07.09 | Fubard | on a polycom 501, with bootROM 4.1.4; sip 3.1.6; mwi retains notification on phone reboot |
| 22:07.20 | jdoe | hrm. |
| 22:07.42 | jdoe | this is an ip500 with whatever the current software is for it. |
| 22:07.43 | Fubard | might be different from your model but its worth a shot if there's something there you dont have in your .conf |
| 22:07.51 | jdoe | no that's more-or-less the same. |
| 22:08.15 | Fubard | hmmm ... i'm out of options. lets riot in the streets? |
| 22:08.36 | Fubard | hey, for shits and giggles, reset your local settings |
| 22:09.52 | Fubard | to me the phone is more of a mysterious beast than a logic based device. sometimes when i reset local settings, things that i'm trying to change keep set |
| 22:10.29 | jdoe | this firmware has a bug that causes it to format and redownload everything every time, I don't think that's the problem :) |
| 22:11.34 | Fubard | well then, for this one instance thats not a bug. its a pre-emptive troubleshooting feature. |
| 22:12.11 | Fubard | sorry i cant be more help |
| 22:15.23 | jdoe | no worries. |
| 22:15.30 | jdoe | cost of working with ancient phones :) |
| 22:16.13 | jdoe | that's really strange though, it only ever lights up *after* logging in to voicemail. |
| 22:17.25 | Alric | anyone know offhand what the different fields in meetme list <conference> concise are? |
| 22:20.54 | *** join/#asterisk warmbowski (~warmbowsk@97.65.109.26) |
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| 22:22.13 | spenguin[work] | does it make sense to Set(CALLERID) on an outbound call? |
| 22:22.32 | spenguin[work] | or can I set caller id to what I want on an outbound call |
| 22:26.04 | Wildy | umm... i've a strange issue with CDRs - when I use cdr-csv, the timestamps are in UTC. When I use cdr_mysql, it writes calldate as localtime. Why it's like that? |
| 22:26.07 | hardwire | you *may* and *can* set the outbound caller id if your tech supports it. |
| 22:27.03 | spenguin[work] | but the user on the other end may not necessarily see the caller id we've set, right? |
| 22:31.20 | warmbowski | Can anyone explain the difference between the following app_voicemail error messages? They all have the same text, but different error numbers. And they show up under different circumstances. . . |
| 22:32.49 | warmbowski | ERROR[3332], ERROR[3289], and ERROR[3347] all containing the text 'app_voicemail.c: Couldn't find mailbox ### in context default' |
| 22:33.13 | Corydon76-dig | Those are thread IDs, not error numbers |
| 22:33.42 | warmbowski | ah, I see |
| 22:34.01 | Corydon76-dig | The point of putting the thread ID in the log is so you can track a progression of messages for a single thread after the fact |
| 22:37.13 | *** part/#asterisk mcateer (~jmcateer@75.76.254.162) |
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| 22:48.01 | spenguin[work] | if I have multiple DID numbers, would it be possible for me to ensure a call goes out from a specific DID number? |
| 22:49.44 | spenguin[work] | I can make sure the call made goes through the specific voip provider, but setting the callerid should be enough? |
| 22:50.53 | *** join/#asterisk jshriver (~jshriver@cblmdm24-53-177-197.buckeyecom.net) |
| 22:51.01 | jshriver | hiya again, wanted to thank everyone here for their help, and for listening to me whine the past several weeks lol. Finally have the asterisk system up and running and seems stable so far. |
| 22:51.25 | jshriver | think I even found out what was causing the hardware to fault :) |
| 22:57.47 | ChannelZ | Squirrels? |
| 22:58.17 | ChannelZ | spenguin[work]: I suppose that depends on your VoIP provider - if they let you fudge your own CallerID |
| 22:58.25 | jshriver | lol |
| 22:58.28 | jshriver | bad power supply |
| 22:58.36 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 22:58.37 | ChannelZ | Caused by squirrels |
| 22:58.44 | talntid | darn squirrels. |
| 22:58.46 | jshriver | I was connecting the two molex connectors to the cards, even though I was only using FXO chips. |
| 22:58.49 | talntid | 10/22 > squirrel |
| 22:59.10 | jshriver | Put equipment in a new computer with a good PSU and everything works fine. |
| 22:59.11 | ChannelZ | Airsoft > squirrel |
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| 23:02.58 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 23:13.25 | *** join/#asterisk infobot (ibot@rikers.org) |
| 23:13.25 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.6, 1.6.1.18, 1.6.0.26 (2010/03/12), 1.4.30 (2010/03/12), *-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 |
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| 23:23.53 | *** part/#asterisk warmbowski (~warmbowsk@97.65.109.26) |
| 23:32.21 | jsjc | as well I just installed fax for asterisk the one from digium.Does that install the app sendfax receivefax? I am running 1.6.2 |
| 23:32.49 | jsjc | I am looking to set a simple fax2email email2fax |
| 23:37.52 | *** join/#asterisk diegomad (mad@186.81.147.35) |
| 23:47.39 | ChannelZ | yes |
| 23:47.47 | ChannelZ | To do email you'll have to integrate that yourself |
| 23:52.18 | Alric | SendFax and ReceiveFax are included if you have spandsp installed. Those applications operate with TIFF files. You'll have to come up with a way to convert emails to TIFF files. |
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