| 00:01.40 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 00:10.07 | *** join/#asterisk root52 (~root52@wsip-98-175-224-146.cl.ri.cox.net) |
| 00:14.56 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 00:28.27 | *** join/#asterisk citywok (~kvirc@67-134-194-33.dia.static.qwest.net) |
| 00:35.29 | root52 | Asterisk 1.8 MySQL (built in not odbc) Adding a Custom filed. Set(CDR(colum_name)=${my_var}) Added column to database. Still my column is not added to the query as seen in the debug log. What am I missing? Thanks!! |
| 00:40.38 | sol326 | Arggghh my telephone company doesn't keep track of my local calls |
| 00:41.11 | sol326 | have no clue how much I'm using so should stick to a monthly amount but I need like 5 lines |
| 00:41.33 | citywok | why don't you keep track then? |
| 00:41.42 | sol326 | http://www.broadvoice.com/rateplans_unlimited_busi.html |
| 00:41.57 | sol326 | because they don't charge me for them |
| 00:42.13 | sol326 | only track my LD because they charge me for it |
| 00:42.37 | citywok | lol, so you're annoyed they don't track it, but you don't track it because they don't charge you, but won't track it because they dont charge you? |
| 00:43.22 | sol326 | no I just started researching doing VOIP so i wanted to know how many minutes I use during busy season |
| 00:43.35 | root52 | hmmm worked just fine. After a restart of asterisk. Odd since I made the change and reloaded the dialplan as well as cdr_mysql.so |
| 00:43.41 | sol326 | and now I can't figure it out to know how to compare to my current plan |
| 00:44.17 | sol326 | or each other for that matter |
| 00:44.43 | sol326 | gotta go through another year but I'm not going to wait for that ;) |
| 00:45.26 | sol326 | <----CPA... sort of hate having to the blood... |
| 00:45.28 | sol326 | <PROTECTED> |
| 00:46.06 | sol326 | ....having the blood.. |
| 00:46.19 | citywok | sol326: if you are a CPA you should be able to take a month of your current call count on local, take the % of local to LD calls, and then estimate the number of local calls during peak season based on the LD calls during peak season :P |
| 00:46.23 | sol326 | gotta compare every angle every six months :D |
| 00:46.39 | sol326 | lol |
| 00:46.42 | citywok | i imagine you can guess within a pretty good margin of error |
| 00:46.56 | citywok | assuming month to month the % is pretty consistent :P |
| 00:47.12 | sol326 | they don't keep track of number either not just minutes :P |
| 00:47.17 | sol326 | of course not... |
| 00:47.22 | sol326 | tax season remember :D |
| 00:47.35 | sol326 | 75% of volume in 3 months |
| 00:47.36 | citywok | Yea, why don't you kee ptrack for a 30 day period? |
| 00:47.49 | sol326 | um today is?? |
| 00:47.58 | citywok | 12/8/2010 -- is this a trick question? |
| 00:48.07 | sol326 | quietest part of year to most busy in 60 days no time :D |
| 00:48.09 | *** join/#asterisk root52 (~root52@wsip-98-175-224-146.cl.ri.cox.net) |
| 00:48.26 | citywok | as long as the ratio of LD to local calls stays the same, it doesn't matter how quiet it is :p |
| 00:48.48 | citywok | you can go back to last january's LD numbers and figure out how many local calls you made +/- a couple percent |
| 00:48.52 | sol326 | well maybe and maybe not :D |
| 00:49.51 | sol326 | I'll just still with a monthly plan for now and then revisit plans in 6 months once i've got some tracking under my belt.. |
| 00:50.08 | sol326 | and then I can do some nitty gritty comparing :D |
| 00:50.36 | sol326 | easier that way and I still know I'm saving at least half on my bill |
| 00:50.41 | citywok | how many minutes/month do you do? 100k? |
| 00:50.48 | sol326 | more like 60% |
| 00:51.16 | sol326 | lol 400.... |
| 00:51.27 | sol326 | but local calls are much higher volume |
| 00:52.00 | citywok | 5 lines on 400k minutes? what? lol |
| 00:52.19 | sol326 | I got a lot of factoring in because of how I'll use the voip vs how I'm currently doing things and expanding with opening a few new companies |
| 00:52.34 | sol326 | I told you the volume is mostly local :P |
| 00:52.55 | sol326 | faxes.... consulting... etc. |
| 00:53.11 | citywok | i have 60 trunks for 180,00 minutes :P |
| 00:53.20 | sol326 | lmao |
| 00:53.27 | citywok | but don't care about local vs LD, i just send it all to the same trunk :P |
| 00:53.29 | sol326 | well one day... |
| 00:53.43 | citywok | wait do you have 400 minutes, or 400,000? |
| 00:53.57 | sol326 | no u r correct at first |
| 00:55.24 | citywok | how many concurrent calls do you have? |
| 00:56.37 | sol326 | mostly 2 or 3 but 2 mostly.. but during tax season all 4 employees can be on the phone pretty easy |
| 00:56.41 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 00:57.04 | sol326 | one one is in a processing room |
| 00:57.08 | citywok | lol so 400, not 400k. big difference :P |
| 00:57.11 | sol326 | one phone |
| 00:57.37 | citywok | i was saying 400k not 400 :P so you had me confused. |
| 00:57.39 | sol326 | that's y I lol and ..... |
| 00:57.55 | sol326 | no K |
| 00:57.56 | sol326 | :D |
| 00:58.16 | citywok | how much does each trunk cost you? |
| 00:59.26 | sol326 | ??? sorry I'm a newbie still... as in trunk u mean number?? |
| 01:01.01 | citywok | i'm guessing you don't use asterisk do you? |
| 01:01.14 | sol326 | I'm going to.... :D |
| 01:01.28 | sol326 | downloaded it.. going to install it when I get my routers |
| 01:01.49 | citywok | ah, in that case just install it, find an itsp you like and pay the 1c/m per call and call it good :) |
| 01:01.54 | citywok | ~itsp |
| 01:01.54 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 01:02.49 | sol326 | yeah I got that lesson from fender earlier :D |
| 01:03.08 | sol326 | but I'm still warry about the 1c/m |
| 01:03.22 | sol326 | especially during this time of year |
| 01:03.27 | citywok | worst cse you rack up a $50 phone bill. i'm sure it's still cheaper than what you arleady pay :P |
| 01:03.32 | *** join/#asterisk RobertLaptop (~RobertLap@pool-173-69-205-54.bltmmd.fios.verizon.net) |
| 01:04.34 | *** join/#asterisk root52 (~root52@wsip-98-175-224-146.cl.ri.cox.net) |
| 01:04.38 | sol326 | yeah yeah but I'm a cpa remember :D if I can beat it by anothery 10 y not :D |
| 01:04.55 | citywok | 40hrs/week* 4employees * 60 minutes/hr = 9,600 working minutes. at 100% utilization the worst you could do is $100/week :P |
| 01:05.15 | sol326 | sort of gives me a rest assuredness!?!?!?! |
| 01:05.16 | citywok | you're not going to beat 1c/m |
| 01:05.33 | sol326 | wasn't thinking I could /m |
| 01:05.52 | citywok | so why are you wary of paying the best psosible rate? what am i missing? |
| 01:06.00 | sol326 | um that's them not me.... the owner... |
| 01:06.04 | sol326 | :D |
| 01:06.15 | sol326 | I can be on the phone almost all day at times |
| 01:06.53 | sol326 | because $100/wk is WAY more than I'm currently paying... |
| 01:06.54 | citywok | if they are all local calls then get 4 POTS lines from your local carrier and an FXS card and run them through * |
| 01:07.19 | citywok | pay your telco the $30/pots line and call it good. |
| 01:07.58 | sol326 | oh blah you're not helping wok your only arguing |
| 01:08.39 | sol326 | My only thing is to find a reasonable ITSP which I think I'll give one a try to get me started |
| 01:10.30 | citywok | i'm not arguing, i'm giving you your options |
| 01:10.39 | citywok | 1) use an itsp, 2) use pots lines from your telco w/ an fxs card in your *box |
| 01:11.03 | citywok | 1 = 1c/m, 2 = $30/line + free local calling & 3-10c/m for LD |
| 01:11.35 | citywok | i've had good luck with flowroute, i hear voicepulse has gotten better than they used to be. |
| 01:11.39 | sol326 | from my telco?? |
| 01:11.55 | sol326 | no can't say it's that cheap... |
| 01:12.13 | citywok | you pay more than 10c/m for LD? dear god that hurts my brain. |
| 01:12.45 | sol326 | no $30/m |
| 01:12.57 | sol326 | http://www.broadvoice.com/rateplans_unlimited_busi.html |
| 01:13.15 | sol326 | this looks like it's 30/m and only $2/m extra each line |
| 01:13.24 | sol326 | so about $40 a month for 5 lines |
| 01:13.32 | sol326 | with taxes about $50 maybe |
| 01:13.40 | sol326 | and long distance |
| 01:13.48 | sol326 | being 1c/m |
| 01:14.04 | sol326 | but this is unlimited us free it looks like |
| 01:14.18 | citywok | that's $30 for one concurrent phone call, and it will have a soft-limit. it's not truly unlimited. |
| 01:14.38 | NightMonkey | Howdy. Anyone have a link to a decent overview of new features of Asterisk 1.8? |
| 01:15.13 | citywok | http://www.asterisk.org/node/51444 |
| 01:16.53 | NightMonkey | citywok: Thanks. I guess I'm looking for a more depthful article, but not the ChangeLog depth. :) |
| 01:17.23 | citywok | yup |
| 01:17.37 | citywok | try the * wiki |
| 01:19.45 | sol326 | is a DID equal to having another Number?? |
| 01:20.44 | sol326 | I have a 4 line phone that I use with 3 NUMBERS and want a 4th.. with a fax... |
| 01:21.19 | sol326 | number.... or is it ok as just a line for the fax |
| 01:21.35 | sol326 | um sorry dumb question... |
| 01:21.44 | sol326 | nm the last one |
| 01:21.59 | sol326 | just need clarification on DID |
| 01:23.43 | Leddy | direct inward dial |
| 01:34.28 | *** part/#asterisk root52 (~root52@wsip-98-175-224-146.cl.ri.cox.net) |
| 01:44.24 | *** join/#asterisk Nwab (~Schnitzel@unaffiliated/benwa) |
| 01:51.16 | sol326 | asterick forwards calls if both simultaneous lines are used on a DID?? |
| 01:51.24 | sol326 | or ISP does that? |
| 01:53.16 | jdoe | I would imagine you'd get a busy signal or something. |
| 01:53.54 | sol326 | oh I think I'm getting all this now.... |
| 02:19.26 | *** join/#asterisk shmaltz (~chatzilla@mail.dmaven.com) |
| 02:20.50 | shmaltz | hi everyone |
| 02:22.28 | ChannelZ | hi |
| 02:26.10 | shmaltz | ChannelZ, is ur name everyone :P |
| 02:36.33 | ChannelZ | My middle name, yes. |
| 02:37.12 | ChannelZ | And MY keyboard has an 'any' key too, so suck it! |
| 02:37.18 | ChannelZ | :D |
| 02:39.51 | shmaltz | i cant find my 'any' key, is your soft as in suckable? |
| 02:44.07 | ChannelZ | I will keep it clean and not respond to that. |
| 02:46.24 | shmaltz | but you just did respond |
| 02:47.00 | *** join/#asterisk [T]ank (~ckwall@c-76-27-9-24.hsd1.ut.comcast.net) |
| 02:50.13 | [T]ank | i was running asterisk 1.6.1.1. I removed /usr/lib/asterisk/modules/* then install 1.6.2.15. audio was working fine before i did this. now i get no audio at all. I can see the call going on the cli output but i hear nothing. |
| 02:50.21 | [T]ank | what should i check? |
| 02:50.25 | [T]ank | i have not made any other changes. |
| 02:50.47 | [T]ank | same config files, and same iptables |
| 02:55.00 | [T]ank | im getting no errors on the CLI. as far as I can <see> everything is operating. |
| 02:55.31 | [T]ank | When i call inbound, i hear messages, but if the call is answered there is no audio. When i call outbound i do not hear ringing or any audio or anything. |
| 02:56.51 | [T]ank | Wow... this channel is dead tonight |
| 02:56.56 | [T]ank | is anyone alive? |
| 02:57.07 | [T]ank | Marco! |
| 03:00.22 | *** part/#asterisk [T]ank (~ckwall@c-76-27-9-24.hsd1.ut.comcast.net) |
| 03:00.39 | *** join/#asterisk [T]ank (~ckwall@c-76-27-9-24.hsd1.ut.comcast.net) |
| 03:00.51 | [T]ank | Trying again... anyone here? |
| 03:02.10 | oDesk | Why wouldn't any SIP device register into my local server if the internet connection into my server is down..but the all network connections works, i'm on dynamic dns |
| 03:06.11 | [T]ank | oDesk: I can only speculate... you have not shared any info at all |
| 03:06.31 | [T]ank | error messages? was it once working and now is not? sip.conf? |
| 03:06.57 | *** join/#asterisk corretico (~corretico@201.201.44.82) |
| 03:07.23 | oDesk | [T]ank: no it doesn't work before while the internet connection is down |
| 03:07.37 | [T]ank | but it works when the connection is up? |
| 03:07.46 | *** join/#asterisk sshock (~sshock@sshock.net) |
| 03:07.58 | oDesk | [T]ank: yes, everything is ok when it's up |
| 03:08.15 | [T]ank | well... once again, i can only speculate until you share your configs and error output. |
| 03:09.01 | oDesk | [T]ank: well, no errors, but i'll post my config |
| 03:09.13 | [T]ank | if you have no errors.. how do you know its failing? |
| 03:09.38 | [T]ank | are you sure your sip devices are not pointed to your server via the public internet? |
| 03:10.03 | [T]ank | if you point using the ip address of the server and are on the same network, you will get reponses if your iptables is set up correctly. |
| 03:10.55 | [T]ank | anyone here have experience with loosing audio after upgrading versions? made no changes to configs or the server. |
| 03:11.36 | [T]ank | was working on version 1.6.1.1. Then i removed the modules and installed 1.6.2.15 and now i have no outbound audio at all, and on inbound I get audio, like voicemail greetings, but on connected calls, there is no audio going either way. |
| 03:13.04 | oDesk | [T]ank: i know its failing because devices does no register my mobile, wifi sip, softphone, etc. and i'm pointing all to the local ip 192.168.1.x |
| 03:14.02 | [T]ank | what version of asterisk? version of linux? do you have a firewall? are you getting any output on the cli when you turn on your device and register it? What is your verbose set to? |
| 03:14.21 | oDesk | [T]ank: centos 5 asterisk 1.6.2.13 |
| 03:15.02 | [T]ank | what is your verbose set to? |
| 03:15.09 | oDesk | [T]ank: well, no output at all into the cli with -rvvvvvvvvvvvvvvvd |
| 03:15.33 | [T]ank | just go to the cli and type set verbose 99 |
| 03:15.46 | [T]ank | then show the output of sip show peers |
| 03:16.05 | oDesk | [T]ank: i'll have to cut the internet to test |
| 03:16.16 | [T]ank | cya |
| 03:16.25 | [T]ank | is there really no one else here? |
| 03:16.31 | [T]ank | this is sad. usually so busy |
| 03:17.20 | oDesk | [T]ank: i think you know the audio problem always nat configuration |
| 03:17.54 | oDesk | [T]ank: try sip show settings and check your remote ip is set correctly |
| 03:18.14 | [T]ank | .... but, the only change was going from a working version of 1.6.1.1, removing /usr/lib/asterisk/modules/*, then installing 1.6.2.15, then no audio |
| 03:18.44 | [T]ank | no other changes to config, server, firewall, or anything like that... simply a new version of asterisk |
| 03:18.59 | *** join/#asterisk plut0 (~cory@cpe-74-76-182-29.nycap.res.rr.com) |
| 03:20.19 | oDesk | [T]ank: have you checked the codecs also sip show settings and check the codecs selected are working |
| 03:20.42 | plut0 | anyone able to get fax receive working on 1.8 for sip? its not working for me |
| 03:25.28 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 03:25.34 | ChannelZ | [T]ank: Is it one-way audio or none at all? |
| 03:27.40 | [T]ank | well... its working now... i rolled back to 1.6.1.1 :-( I will have to go over everything with a fine toothed comb and see what is different. |
| 03:28.42 | jaytee | sometimes even a fine toothed comb is of no help. specially if you're bald :-) |
| 03:29.50 | jaytee | I definitely want the t-shirt that says "Science.....it works, bitches!!!" |
| 03:31.10 | coppice | I've also seen "Science - its what religion wants to be when it grows up" |
| 03:31.23 | jaytee | lol |
| 03:31.54 | jaytee | I also like the one that says "Heisenberg slept here.....or somewhere nearby." |
| 03:39.56 | plut0 | anyone know how to get fax working in 1.8? |
| 03:46.55 | *** join/#asterisk slidesinger-lt (~jtatum@c-68-44-99-163.hsd1.nj.comcast.net) |
| 03:52.23 | *** join/#asterisk DrCron (rszasz@saxonco.com) |
| 04:02.18 | [T]ank | when i see: [Dec 8 20:48:19] NOTICE[1252]: chan_sip.c:18456 handle_response_peerpoke: Peer 'X' is now Reachable. (91ms / 2000ms), does that mean it was unreachable? Or is this simply a checkin? |
| 04:04.09 | p3nguin | I think that means it used to be either unreachable or reachable with a qualify time of more than 2000ms. Do you see a line before it that said it was unreachable? |
| 04:06.53 | [T]ank | no... thats what made me wonder. |
| 04:08.46 | *** join/#asterisk cmnky (debian-tor@gateway/tor-sasl/cmnky) |
| 04:09.06 | p3nguin | I've seen that spontaneous message without being preceded by a "lagged" or "unreachable" message, but I never really paid much attention to it. |
| 04:09.20 | *** join/#asterisk mindCrime_ (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 04:09.22 | [T]ank | yeah... me too |
| 04:09.25 | [T]ank | just wondered |
| 04:09.38 | p3nguin | Good question, though. |
| 04:12.34 | *** part/#asterisk plut0 (~cory@cpe-74-76-182-29.nycap.res.rr.com) |
| 04:21.45 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 04:26.31 | RobertLaptop | More likely means some where in your network the connection is getting maxed out causing slow downs. |
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| 04:58.02 | [T]ank | is there a compelling reason to move from 1.6 to 1.8? |
| 04:58.14 | [T]ank | anything i cant live without? |
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| 05:33.17 | joeyjones | i'm thinking of buying a Linksys SPA941, anyone used it before? |
| 05:34.11 | *** join/#asterisk sol326 (~studyofli@c-71-56-135-99.hsd1.wa.comcast.net) |
| 05:34.37 | sol326 | any special hard drive config for asterisk?/ |
| 05:37.06 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 05:39.16 | sol326 | how nice of asterisk to put it in a nice little bundle with centos for us so we don't have to yum, or apt-get itit.. :D |
| 05:43.08 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
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| 06:00.23 | ChannelZ | yes because it's soo hard to do that |
| 06:01.34 | ChannelZ | joeyjones: I have a mess of 922s which are of the same ilk |
| 06:02.29 | cmnky | has a SPA2102 |
| 06:02.47 | cmnky | with the worlds oldest phone plugged into it ;) |
| 06:03.17 | p3nguin | I'm pretty sure it won't even power the oldest phone. |
| 06:03.28 | cmnky | "Watson ... Watson ... come in here damnit!" |
| 06:04.48 | sol326 | asterick is causing some feekback on my network... |
| 06:04.51 | cmnky | p3nguin, theres no date on it ... but it says GTE |
| 06:05.18 | p3nguin | sol326: Get rid of it and install Asterisk instead. |
| 06:05.20 | sol326 | aaah I think I know why... |
| 06:05.31 | sol326 | asterisk ist what I meant |
| 06:05.34 | sol326 | I know why |
| 06:05.58 | cmnky | sol326, pro tip ... use * so you don't have to take s**t from the pros ;) |
| 06:06.12 | sol326 | lmao |
| 06:06.13 | sol326 | k |
| 06:07.03 | sol326 | argg I just wanna turn it off and change the nic now.... blah... |
| 06:09.59 | sol326 | why does * use 127.0.0.1 for it's gui?? |
| 06:10.16 | sol326 | is there some way to change that... |
| 06:10.29 | sol326 | I think that's why I have the flicker |
| 06:11.06 | ChannelZ | * doesn't have a GUI |
| 06:11.34 | ChannelZ | But FreePBX, et al use whatever interface you config them to use |
| 06:12.40 | joeyjones | ChannelZ: the 922s are solid? |
| 06:14.18 | ChannelZ | I've liked mine. Had the PoE die on one but it's still chugging along on a wall wart |
| 06:15.07 | ChannelZ | I wish they'd wake up and let us put blind transfer on the first page of buttons but I'll live. |
| 06:17.27 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
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| 06:29.37 | joeyjones | ChannelZ: any ideas for a cheaper 2-line SIP phone? |
| 06:30.11 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 06:30.27 | jdoe | uhhh |
| 06:30.36 | jdoe | you can find polycom ip500s on ebay for like $20/per |
| 06:30.56 | jdoe | those are 3 line, but they're EOL'd for years now. |
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| 07:40.42 | verywiseman | i want to know asterisk market share in USA and UK pls |
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| 07:51.12 | jsjc | anyway to start asterisk with all queue members paused? |
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| 08:19.42 | joeyjones | jdoe: i'd like to get something that could be deployed, not dug out of the trash |
| 08:20.28 | kaldemar | ~phones |
| 08:20.28 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever. |
| 08:25.19 | joeyjones | lol @ grandstream |
| 08:26.01 | joeyjones | there's an ad on the local usedeverywhere site (like craigslist) for a grandstream office system for cheap, stating that they never got it working |
| 08:28.36 | jsjc | exten => *50,1,Macro(add-agent,Local/DAHDI2@queuemembers/n,4) [add-agent] ... exten => s,2,AddQueueMember(myqueue,${ARG1},${ARG2}) why is not passing onwoards the ARG2 in this case number 4 so the penalty in the que qill be 4? If I change ${ARG2} for a number in dialplan it will get the penalty but when it comes in the macro arguments does not add any penalty. (any reason?) |
| 08:31.12 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
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| 08:37.14 | jdoe | joeyjones: you said cheap, I gave you cheap :P |
| 08:38.02 | kaldemar | jsjc: show a failing call. |
| 08:42.33 | kaldemar | jsjc: and show your real dialplan... |
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| 08:56.00 | jsjc | kaldemar I will post it to pastebin |
| 08:56.57 | jsjc | haha found it! ;) |
| 09:00.29 | kaldemar | was it the context name or something else? |
| 09:00.53 | jsjc | I had two lines one to add agent another to remove agent and the priority had it on remove agent.... |
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| 09:01.17 | jsjc | kind of stupid but when you look all the lines get confused... When I was pasting the needed things found it! |
| 09:01.18 | jsjc | heheh |
| 09:01.29 | jsjc | sometimes everything gets lost into a bit of code |
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| 09:10.06 | joeyjones | lol jdoe, truie |
| 09:10.09 | joeyjones | *true |
| 09:10.50 | joeyjones | i meant more affordable than cheap :p |
| 09:11.22 | joeyjones | something thyat's worth the money, has decent features, but isn't some crazy has-everything break the bank model |
| 09:12.02 | jsjc | wow now I got to work the add agent but got into a new issue... when on a current call and another call comes in instead of waiting on the que til first call has finished kind of a beep noise appears and connection with the first caller dissapears... |
| 09:12.44 | jsjc | could be the call waiting on DAHDI? that should be disabled? |
| 09:14.04 | kaldemar | what is the queue member? |
| 09:14.48 | kaldemar | an analog phone? |
| 09:15.01 | jsjc | yes |
| 09:15.31 | jsjc | I think most likely it will be it, just better disable call waiting and let them wait on the queue (that i thing queue was designed for that.) |
| 09:16.45 | kaldemar | you need to make the member busy from the queue's point of view. |
| 09:18.36 | kaldemar | set callwaiting=no for the channels. |
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| 09:51.18 | jsjc | I am having some issues with SIP registration. I have a DSL with dinamic IP address and when I am connected to SIP server and the DSL connection reloads with new IP, the SIP server does no know who I am til I re-register again, is there anyway I could have something to reregister ASAP after new IP has been given? |
| 09:53.52 | freckle | jsjc: if your IP is changing so often it causes a problem I would say get a better DSl |
| 09:54.26 | jsjc | feckle even if does not change often just once a day the time that it changes the asterisk drops out, and not calls can be received... |
| 09:54.54 | freckle | jsjc: how often are you re-registering? |
| 09:58.27 | heffer | jsjc: are you talking of Asterisk as a SIP registrar or as a client? |
| 09:58.52 | golikwid|mac | anyone know how to connect to the sql db in asterisk from a remote server |
| 09:59.07 | golikwid|mac | ive been searching and cant seem to find anything |
| 10:00.55 | kaldemar | the sql db in asterisk? |
| 10:01.21 | golikwid|mac | yes |
| 10:01.33 | kaldemar | do you mean astdb? |
| 10:01.44 | golikwid|mac | asteriskcdrdb |
| 10:02.12 | kaldemar | that must be a separate database, not a part of asterisk. |
| 10:02.40 | kaldemar | make your database accept connections from outside the box itself. |
| 10:03.11 | golikwid|mac | maybe its from freepbx? |
| 10:03.11 | golikwid|mac | i tried that and am really afraid that my tinkering at some point is just going to break it |
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| 10:03.22 | kaldemar | ask in #freepbx if that's what you're using. |
| 10:03.42 | kaldemar | there is no sql db in asterisk. |
| 10:04.05 | golikwid|mac | ok thanks |
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| 10:25.41 | joachim_- | Hi! Now i upgraded Asterisk 1.6.2.14 to 1.8.0, replacing app_fax with res_fax & res_fax_spandsp.. |
| 10:25.44 | joachim_- | http://forums.digium.com/viewtopic.php?f=1&t=76401&sid=294c613bb0dd114ddf51dda4a4a23d39 |
| 10:26.34 | joachim_- | why do I get::: Audio FAX not allowed on channel 'SIP/XXXXXXXXXX-00000002' and T.38 negotiation failed; aborting. |
| 10:27.49 | WIMPy | No idea about fax, but there is 1.8.1 since yesterday fixing several things. |
| 10:29.01 | ectospasm | joachim_-: I don't know about spandsp, but generally you don't want an audio fax with T.38 |
| 10:29.24 | joachim_- | WIMPy: hmm.. thanks for tip. will look into that |
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| 10:31.01 | joachim_- | ectospasm: I use a Grandstream HT 486 ATA between analog fax and asterisk.. could that be the problem? I have the Send DTMF option set to: In Audio |
| 10:31.37 | joachim_- | ectospasm: I can choose between Send DTMF: in-audio , via RTP (RFC2833) or: via SIP INFO |
| 10:31.56 | ectospasm | joachim_-: yeah, afaik using an ATA that isn't specifically designed for faxing may cause issues. And dtmfmode isn't relevant afaict |
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| 10:38.46 | m_tadeu | hi...I need a fxo with 4 lines...but it has to be pci express...what do you guys recomed? |
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| 10:56.11 | Siblor | Hello, I'm beginner to asterisk and need help about asterisk development with Java. Who could I talk to? |
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| 11:08.55 | ectospasm | Siblor: I'm not a Java expert by any stretch, but do you have a specific question? |
| 11:09.20 | ectospasm | m_tadeu: AEX410, from Digium (disclaimer, I work for them) |
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| 11:15.15 | m_tadeu | ectospasm: cool :) |
| 11:15.59 | m_tadeu | so, explain me something....the AEX410 is only the pcie board, right? then I have to get the fxo modules, is this correct? |
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| 11:30.31 | joeyjones | ~phones |
| 11:30.31 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever. |
| 11:30.36 | joeyjones | ~triggers |
| 11:30.47 | joeyjones | ~providers |
| 11:30.47 | infobot | providers is probably http://www.voipreview.org/service.all2.aspx?Country=1&Area_Code=0&CallingArea=0&provider=0&serviceType=1&Adv=1&Features=43 |
| 11:31.56 | joeyjones | ~trunks |
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| 11:41.28 | ectospasm | m_tadeu: typically you purchase modules with the card, but that's not a hard rule since they're removable |
| 11:41.54 | jpmcallister | Hi. Does anyone know when or if will digium release the rpm packages for the jabber and gtalk addons? |
| 11:42.38 | ectospasm | jpmcallister: off the top of my head, Qwell or russellb may be best suited to answer that |
| 11:43.55 | ectospasm | jpmcallister: I'd imagine if they're awake, they're just waking up |
| 11:48.09 | jpmcallister | ectospasm: tank you I try asking later ! |
| 11:50.18 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 11:50.56 | eduzimrs | i`ve tried to find "chan_sip.c" in * 1.6 but i couldnt, the name changed in this version ? |
| 11:53.54 | ectospasm | eduzimrs: should be in the channels subdir |
| 11:54.25 | ectospasm | eduzimrs: Iunno, I don't have 1.6.2 source handy |
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| 11:54.54 | WIMPy | ectospasm: He has been told so several times yesterday already. |
| 12:00.26 | eduzimrs | WIMPy but i dont understando sorry |
| 12:01.09 | eduzimrs | i couldnt find searching for "chan_sip.c" nothing matches ! |
| 12:03.15 | eduzimrs | im persisting in but im not expert in * , somethings to me not appears clear |
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| 12:09.13 | ectospasm | eduzimrs: /usr/src/asterisk-version/channels/chan_sip.c |
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| 12:18.30 | eduzimrs | 1.6 there is no this path |
| 12:19.41 | eduzimrs | i run find / -iname "*sip*" -exec ls {} \; command and nothing goes back! |
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| 12:20.09 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 12:33.44 | rethus | someone here has experiance with ant-phone on linux? |
| 12:34.01 | rethus | i couldn't start it from bash to direct calling a number |
| 12:34.35 | rethus | ant-phone -d -r -c <test-number> |
| 12:35.04 | eduzimrs | WIMPy did u find? |
| 12:40.02 | WIMPy | sure |
| 12:42.06 | eduzimrs | WIMPy in 1.6 ? |
| 12:42.35 | WIMPy | In all versions I ever had. |
| 12:43.00 | eduzimrs | so i dont know anymore, |
| 12:43.05 | eduzimrs | i didnt |
| 12:44.06 | eduzimrs | my /usr/src there is no asterisk dir |
| 12:44.34 | eduzimrs | i just found chan_sip.so |
| 12:44.50 | WIMPy | Then you should start by downloading the Asterisk Sources. |
| 12:45.00 | *** join/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda) |
| 12:45.21 | mbrevda | anyone know what this means? WARNING[19967] app_exec.c: Could not find application! (Macro( |
| 12:45.31 | *** join/#asterisk coppice (~chatzilla@96.157.17.210.dyn.pacific.net.hk) |
| 12:46.01 | mbrevda | (Yes, I know it means that it cant find the Macro application. But why not?) |
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| 14:06.35 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 14:06.35 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.1 (2010/12/08), 1.6.2.15 (2010/12/08), 1.4.38 (2010/12/08), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
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| 14:14.04 | kaldemar | Nivex: i just said the host is up. i don't know about their SIP service. |
| 14:14.29 | Nivex | kaldemar: I understand. I too was able to ping. |
| 14:14.48 | Nivex | I was just hoping there was another sipphone user in here to compare notes with. |
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| 14:43.14 | gregd | guys, what does take precedence - outbound route/dial plan or an internal extension? The context is that I transfer a call... lets say to number 1234. The 1234 fits outbound route/dial plan as well as internal extension... where will it go? |
| 14:45.08 | SuPrSluG | the first context to match |
| 14:45.31 | gregd | fair enough |
| 14:45.42 | *** part/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda) |
| 14:46.58 | gregd | ok, so I'd like to include a specific context only when transferring call that originates from outside (via inbound route).. is it possible? |
| 14:50.59 | SuPrSluG | why not just have a unique extension for that scenario? |
| 14:52.12 | gregd | SuPrSluG: I do have it currently and it works fine, but it is kind of confusing for the user, sometimes it need to dial special extension (when transferring), other time - not |
| 14:52.37 | gregd | so i just wanted to simplify this |
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| 14:54.00 | gregd | but, actually, we could simplify the case but having (including) a special context when a call originates from outside (via inbound route).... (does not matter if in transfer).... |
| 14:54.02 | gregd | is it possible? |
| 14:54.23 | WIMPy | Extensions within a context are match on a most-specific basis. |
| 14:54.38 | WIMPy | Included contexts are searched in the configured order. |
| 14:54.49 | gregd | WIMPy: heh.. how sure are you? ;) |
| 14:55.02 | gregd | cause now i have two theories.. not sure which to trust ;) |
| 14:55.05 | WIMPy | very |
| 14:55.07 | gregd | ok |
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| 14:56.45 | spenguin[work] | hey, I currently have a PRI working fine on my asterisk box, Im expecting a lot more inccoming calls in the next few months |
| 14:57.01 | riddlebox | does video on asterisk only work when both ends are sending video? |
| 14:57.09 | spenguin[work] | hence the telco is able to extend my current e1 line onto another line |
| 14:57.34 | spenguin[work] | the current digium card Ive got in the box is a dual port |
| 14:57.41 | spenguin[work] | and I have both spans configured |
| 14:57.47 | spenguin[work] | to a total of 60 channels |
| 14:57.57 | *** join/#asterisk Sipster_ (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 14:58.19 | spenguin[work] | should this work out fine, I just need to be sure before we dive into this |
| 14:58.35 | WIMPy | spenguin[work]: What? |
| 14:59.06 | WIMPy | You have two options: Two seperate lines or both together as a trunkgroup. See NFAS. |
| 14:59.09 | spenguin[work] | WIMPy: the telco would be extending the e1 channels by terminating another line |
| 14:59.32 | spenguin[work] | so a trunkgroup is what im doing |
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| 15:12.11 | _LoneCrow | If you sip dial someone and it gives you a busy signal and the asterisk console tells you is it circuit busy .. that means the other side didnt answer or gave busy signal right? |
| 15:12.48 | _LoneCrow | I am dialing another asterisk machine and its really odd.. I can use any extension from 200s and up and it'll dial them fine but if I use any extensions in the 100's I get circuit busy |
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| 15:13.13 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 15:20.50 | mbrevda | where are the asterisk sound packages checked in to? |
| 15:22.09 | leifmadsen | mbrevda: what do you mean? |
| 15:22.14 | leifmadsen | you can get them from menuselect |
| 15:22.29 | mbrevda | leifmadsen: hey m8 |
| 15:22.42 | leifmadsen | or here: http://downloads.asterisk.org/pub/telephony/sounds/ |
| 15:22.45 | mbrevda | I remember seeing a file list of |
| 15:22.56 | mbrevda | s/of/with all the text |
| 15:26.19 | malcolmd | each tarball in that directory should contain a .txt file with the file list and the "words" spoken for each prompt |
| 15:26.37 | malcolmd | e.g. core-sounds-en.txt is a part of asterisk-core-sounds-en-alaw-current.tar.gz |
| 15:27.24 | mbrevda | malcolmd: thanks |
| 15:27.28 | malcolmd | np |
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| 15:28.17 | mbrevda | am I missing something or is sounds-extra hsoted elsewhere? |
| 15:28.46 | SuPrSluG | _LoneCrow: check your phones dialplan |
| 15:29.22 | malcolmd | same location, e.g. asterisk-extra-sounds-en-g722-current.tar.gz |
| 15:29.29 | Tech_Travis | does anyone have suggestions on sip phones that will be used outside the firewall? |
| 15:29.49 | mbrevda | malcolmd: must be blind toady |
| 15:30.05 | malcolmd | mbrevda: no worries :) |
| 15:31.00 | *** join/#asterisk ganeshix (~ele@cpe-24-29-44-192.nycap.res.rr.com) |
| 15:31.42 | ganeshix | i just installed asterisknow 1.7.1 |
| 15:31.58 | ganeshix | i did the yum update part |
| 15:32.16 | ganeshix | now i can't log into the web gui |
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| 15:33.26 | *** join/#asterisk mr_steve (~steve@unaffiliated/mr-steve/x-2535065) |
| 15:33.32 | ganeshix | tried all combinations of {admin,root,freepbx}X{admin,fpbx,"root-install-password"} |
| 15:34.03 | ganeshix | any hint? |
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| 15:35.06 | oneseventeen | I'm trying to pass a variable from a dialplan to PHP, then a different variable from PHP to the dialplan... are there any working examples of this for version 1.6.2? |
| 15:35.17 | *** join/#asterisk garymc (~chatzilla@host81-148-106-225.in-addr.btopenworld.com) |
| 15:35.28 | *** part/#asterisk doolph (doolph@unaffiliated/doolph) |
| 15:35.44 | oneseventeen | (I've found tons of non-working examples all over voip-info.org and people's blogs... but apparently syntax has change since those entries) |
| 15:44.17 | *** join/#asterisk gandhijee (akp@ip67-152-15-148.z15-152-67.customer.algx.net) |
| 15:48.27 | _LoneCrow | Suprsplug - I'm actually using trixbox but you mean the phone I'm trying to dial? I'm just doing a sip/xxxx@server.com custom extension. I can make a new extension say 111 and it will not dial the sip dialed extention say is 199 - so I'll dial 199 from 111 and no go. If i only change the extension from 111 to 222 it WILL dial 199 no problems.. |
| 15:49.08 | _LoneCrow | if I dial from 111 it says circut busy but any other number above 200 it will dial them. It has to be on the other end.. not my dial plan |
| 15:49.47 | oneseventeen | ~book |
| 15:49.47 | 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, or http://ofps.oreilly.com/ |
| 15:56.34 | *** join/#asterisk F|ReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:04.59 | justdave | so in my ongoing quest to get the asterisk 1.8.x SRPM to actually build on RHEL5, I've discovered there is actually a minimum version requirement on speex which is not satisfied by the available packages in RHEL5 |
| 16:05.24 | justdave | (no big deal in my book, I can build my own with a more-recent version, and I've submitted a patch to the spec file to have it require the correct minimum version) |
| 16:05.39 | justdave | however, it's still failing to detect freetds as well |
| 16:05.50 | justdave | and I can't figure out what it's looking for that makes it fail to find it. |
| 16:06.14 | justdave | the available freetds package is also old, but I built the most-recent upstream version of freetds and installed that and it still doesn't find it |
| 16:11.48 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 16:12.46 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 16:13.16 | *** join/#asterisk SCounter (~SC@AReims-552-1-135-40.w82-127.abo.wanadoo.fr) |
| 16:13.59 | SCounter | Hi, i have a problem with asterisk 1.6 and ippi.fr service sip |
| 16:14.27 | SCounter | i not receive call in xlite with this server asterisk |
| 16:14.45 | SCounter | my file configuration is http://pastebin.com/pLkNsqqF |
| 16:15.23 | SCounter | sorry for little english because i'm french |
| 16:19.33 | ChannelZ | What does the console say when you try to call? |
| 16:20.20 | ChannelZ | My guess is the call isn't being sent to 's' and you're getting an error, but there could be any number of things wrong. |
| 16:20.27 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 16:23.41 | SCounter | ChannelZ, i received mesage asterisk from console : Verbosity is at least 29 |
| 16:23.42 | SCounter | <PROTECTED> |
| 16:23.50 | eduzimrs | anyone knows about: DEBUG[2162] chan_dahdi.c: Dunno what to do with event 27 on channel 34 ?? |
| 16:24.15 | SCounter | when call for number ippi |
| 16:26.11 | eduzimrs | ?? |
| 16:26.52 | ChannelZ | SCounter: turn up verbose a little - core set verbose 3 |
| 16:27.45 | SCounter | ChannelZ, please send command unix for verbose 3 asterisk ? |
| 16:28.14 | justdave | ok, so I confirmed that the issue is the RPMs for RHEL install the freetds headers in /usr/include/freetds/ |
| 16:28.29 | riddlebox | I was just looking at switchvox, and pricing, do you have to buy subscriptions every year for every phone? |
| 16:28.31 | justdave | passing --with-tds=/usr/include/freetds fails to find them however |
| 16:29.06 | tvc123 | riddlebox: you just need to pay a renewal fee |
| 16:29.09 | justdave | if I manually edit configure to look for "freetds/sybdb.h" instead of "sybdb.h" then it does find it |
| 16:29.18 | justdave | but I don't think that'll work |
| 16:29.28 | tvc123 | riddlebox: you may want to check out #switchvox for more info |
| 16:29.31 | justdave | because I'm sure the source elsewhere is only looking for sybdb.h |
| 16:29.46 | riddlebox | tvc123: thanks, I will check it out |
| 16:29.58 | SCounter | ChannelZ, |
| 16:29.58 | SCounter | Verbosity was 29 and is now 61 |
| 16:29.59 | SCounter | <PROTECTED> |
| 16:30.05 | WIMPy | justdave: Did you try with only --with-tds=/usr/include then? |
| 16:30.17 | WIMPy | fell for such thing a few times in the past. |
| 16:30.48 | justdave | --with-tds=/usr/include |
| 16:30.48 | justdave | checking sybdb.h usability... no |
| 16:30.49 | justdave | checking sybdb.h presence... no |
| 16:31.01 | justdave | and before that... |
| 16:31.02 | justdave | checking for dbinit in -lsybdb... yes |
| 16:31.11 | justdave | so it's finding the library and not finding the headers |
| 16:31.42 | justdave | # rpm -ql freetds-devel | grep sybdb.h |
| 16:31.42 | justdave | /usr/include/freetds/sybdb.h |
| 16:32.10 | WIMPy | I think I would better understand it if it was the other way round. |
| 16:32.41 | WIMPy | Have you tried only =/usr ? |
| 16:33.23 | ChannelZ | SCounter: and you get no other output except the 'Using SIP RTP CoS mark 5' ?? |
| 16:33.38 | justdave | with it only as /usr it doesn't find it either |
| 16:34.01 | SCounter | ChannelZ, yes |
| 16:34.09 | SCounter | only mesage Using SIP RTP CoS mark 5 |
| 16:34.13 | ChannelZ | SCounter: Actually your incoming calls might be matching the 'ippi_outgoing' peer because of the same host=ippi.fr |
| 16:34.46 | ChannelZ | SCounter: In any case probably need to see a SIP debug next. "sip set debug on" in the console and make a call, then use pastebin.com or one of the others |
| 16:35.36 | ChannelZ | I have to leave for work but I'm sure someone will help before I can get back |
| 16:37.08 | justdave | ohooooo |
| 16:38.00 | justdave | configure wants it to be in a directory named include under the path you pass |
| 16:38.07 | justdave | and never attempts to look directly in that path |
| 16:38.39 | SCounter | ChannelZ, sip set debug on |
| 16:38.42 | SCounter | oups |
| 16:38.52 | SCounter | ChannelZ, http://pastebin.com/gSe5V63Q |
| 16:39.33 | ChannelZ | There are two problems |
| 16:39.58 | ChannelZ | As I suspected on line 43: "Found peer 'ippi_outgoing' for '33613393349' from 213.215.45.230:5060" |
| 16:41.00 | ChannelZ | And it's being asked to authorize, but isn't bothering to try. Probably why you have 'insecure=port,invite' for your [incoming] peer, but since that peer is not being matched... |
| 16:41.45 | ChannelZ | have to go, BBL |
| 16:42.34 | SCounter | i commented type=peer in ippi_outgoing |
| 16:42.38 | SCounter | i fonctionnaly |
| 16:44.47 | SCounter | problem with call outgoing too |
| 16:50.29 | SCounter | [Dec 9 17:49:59] WARNING[21319]: chan_sip.c:5340 create_addr: No such host: ippi_outgoing |
| 16:50.38 | SCounter | [Dec 9 17:49:59] WARNING[21319]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 16:50.48 | SCounter | <PROTECTED> |
| 16:50.49 | SCounter | <PROTECTED> |
| 16:51.10 | SCounter | ChannelZ, ? |
| 16:52.00 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
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| 16:57.50 | SCounter | ChannelZ, thank i fonctionnely now |
| 16:57.50 | SCounter | :p |
| 17:09.06 | SCounter | ChannelZ, i find for make a config musiconhold waiting calling |
| 17:13.05 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 17:13.58 | justdave | ok, managed to make it with with an 'export CPPFLAGS="-I/usr/include/freetds"' before running ./configure |
| 17:15.28 | *** join/#asterisk nevermind_quack (~nevermind@66.232.70.40) |
| 17:20.35 | *** part/#asterisk Spirits-Sight (~Spirits-S@209-6-146-132.c3-0.bkl-ubr1.sbo-bkl.ma.cable.rcn.com) |
| 17:21.23 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 17:25.06 | justdave | hmm, well, that made configure not complain about tds being missing if I did --with-tds anyway |
| 17:25.15 | justdave | it still didn't build it |
| 17:25.23 | justdave | tries it manually again to see what menuselect complains about |
| 17:28.55 | *** join/#asterisk Tim_Toady (~moi@79.103.38.73.dsl.dyn.forthnet.gr) |
| 17:31.09 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 17:32.18 | justdave | hah, amusing. configure finds it with CPPFLAGS, and menuselect says it's enabled, but the actual source compile doesn't find it. :( |
| 17:36.11 | justdave | ok, the source compile needs it passed in via ASTCFLAGS apparently |
| 17:36.45 | justdave | although it looks like the makefile is supposed to honor whatever you passed to configure, but in practice that doesn't seem to be working |
| 17:38.27 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 17:38.47 | WIMPy | Maybe as parameter? configure CPPFLAGS= |
| 17:51.44 | *** join/#asterisk nickfennell (~nick@i-195-137-23-30.freedom2surf.net) |
| 17:52.01 | nickfennell | hey peeps, can someone point me at a sip alg project for linux |
| 17:52.04 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 17:52.06 | nickfennell | i had a couple of names but I can't find them now |
| 17:55.02 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 17:57.39 | justdave | submits another patch |
| 17:57.45 | justdave | that makes 5. :) |
| 17:58.01 | justdave | and the SRPM now builds out-of-the-box on rhel5 as long as you have the prereqs |
| 17:58.08 | justdave | (after my patches are applied) |
| 17:58.10 | justdave | https://issues.asterisk.org/search.php?project_id=1&reporter_id=6185&sticky_issues=on&sortby=last_updated&dir=DESC&hide_status_id=90 |
| 17:59.11 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 17:59.14 | nickfennell | Hmmm ok, netfilter has sip alg functionality |
| 18:00.43 | *** join/#asterisk LtBrenton (~LtBrenton@0.91.112.87.dyn.plus.net) |
| 18:00.49 | LtBrenton | hey |
| 18:01.24 | LtBrenton | setting up an asterisk server on a VPS, at the moment it's not firewalled, but having no end of trouble getting my grandstream phone to hook up to it |
| 18:02.08 | LtBrenton | I've got it to the point where I can force a call with channel originate and it'll work properly, but if I try and originate a call from my phone, the SIP exchange is successful but I get no audio |
| 18:02.34 | justdave | what version of asterisk? |
| 18:02.46 | LtBrenton | 1.8.0 |
| 18:02.54 | justdave | and is one or the other of the server or the phone behind a NAT? |
| 18:03.03 | LtBrenton | phone's behind a NAT with dynamic IP |
| 18:03.18 | justdave | you have nat=yes on the sip.conf entry for that phone? |
| 18:03.33 | LtBrenton | yeah, but changing that makes no difference |
| 18:03.57 | justdave | ok, so you're hitting the same bug I am then |
| 18:04.14 | LtBrenton | ah...downgrade then? :P |
| 18:04.14 | justdave | I can't get it to send rtp traffic to the originating IP in 1.8.x so far |
| 18:04.39 | justdave | it keeps using the IP provided in the SIP headers, which is wrong if it's behind nat |
| 18:04.40 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 18:04.53 | *** join/#asterisk hfb (~hfb@pool-96-229-100-244.lsanca.dsl-w.verizon.net) |
| 18:05.21 | LtBrenton | yeah, i've managed to STUN the right IP from the phone, otherwise it wouldn't even register |
| 18:05.25 | LtBrenton | it'd throw Wrong Password |
| 18:05.26 | justdave | I'm just now deploying 1.8.1 as we speek to see if it fixes it |
| 18:05.39 | justdave | there's a couple related line-items in the changelog, so I'm hopeful |
| 18:07.13 | *** join/#asterisk atan3 (~atan@unaffiliated/atan) |
| 18:07.28 | justdave | and no, it doesn't fix it. :( |
| 18:08.05 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 18:09.23 | LtBrenton | justdave, did it work in 1.6.x? |
| 18:09.29 | *** join/#asterisk imox1234 (~imox1234@p4FC5C79F.dip0.t-ipconnect.de) |
| 18:09.33 | justdave | LtBrenton: unknown, never used 1.6 |
| 18:09.36 | justdave | it worked in 1.4 though |
| 18:10.30 | justdave | in theory, putting nat=yes on a sip.conf entry is supposed to make asterisk ignore the IP address given in the SIP headers and make it send traffic to the originating IP address instead |
| 18:10.52 | justdave | in 1.8 it's not doing that, and it's still sending the RTP traffic to the address requested in the headers |
| 18:10.53 | LtBrenton | yeah, but in 1.8 it seems to even go so far as throw an authentication shitfit if the IPs don't match |
| 18:11.11 | justdave | dunno, I haven't been having that problem |
| 18:11.18 | justdave | as long as you have host=dynamic on it |
| 18:11.38 | justdave | SIP registers and negotiates just fine |
| 18:11.46 | justdave | the RTP traffic just isn't getting sent to the right place |
| 18:11.50 | LtBrenton | yeah, but it ditches the auth if the header IP is 192.x and the originator is a WAN address |
| 18:11.55 | LtBrenton | throws it out with wrong password on register |
| 18:12.06 | justdave | that doesn't sound right. |
| 18:12.28 | LtBrenton | if this doesn't work i'm rolling back to 1.6 |
| 18:13.09 | LtBrenton | it's only an experimental thing right now but it's going production pretty much as soon as I've jogged my memory enough :P |
| 18:14.26 | nickfennell | anyone used OpenSBC? |
| 18:14.34 | LtBrenton | also, dear god I forgot how much of a long-ass compile this is |
| 18:15.22 | *** join/#asterisk ickmund (~ickmund@cli-5b7ee407.bcn.adamo.es) |
| 18:19.54 | *** part/#asterisk oneseventeen (~onesevent@web64.webfaction.com) |
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| 18:20.39 | tvc123 | ok I have a newb question ... in iax.conf I want to speicify that register be sent with a md5 password is that possible or do I just need to md5 the string? |
| 18:20.55 | oneseventeen | [TK]D-Fender: thanks for the help! I now have working AGI scripts!! |
| 18:28.27 | justdave | tvc123: auth=md5 I think |
| 18:28.46 | *** join/#asterisk blee (~blee@99-117-188-231.lightspeed.dybhfl.sbcglobal.net) |
| 18:29.29 | *** join/#asterisk atan3 (~atan@unaffiliated/atan) |
| 18:29.47 | blee | does anyone know how to change what priority voicemail 0 exits go to? currently mine transfers to a ext o, priorty 0, and its failing |
| 18:30.15 | justdave | tvc123: normally I'd point you at http://www.voip-info.org/wiki/view/Asterisk+config+iax.conf but it seems to be lacking a lot of the good examples and param descriptions that the sip.conf version has |
| 18:30.15 | blee | i tried to set my extension 0 to start at priority 0, and it doesnt like that, can the voicemail app change? |
| 18:30.36 | blee | extension o rather |
| 18:30.57 | p3nguin | blee: If you are in voicemail and press 0? It goes to extension o. And all extensions start with priority 1. |
| 18:31.23 | blee | p3nguin: correct, asterisk is doing a Goto(o,0) |
| 18:31.32 | p3nguin | blee: Show me. |
| 18:31.32 | blee | i dont know why |
| 18:32.31 | blee | p3nguin: http://pastebin.ca/2014960 |
| 18:32.56 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 18:33.09 | blee | works in an older version of asterisk I use |
| 18:33.32 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 18:34.42 | p3nguin | blee: I was hoping for something a little more detailed. This just shows me that 039, o, 0 exited. I don't know why it tried to run in the first place. Any other info you can provide about this issue? |
| 18:35.03 | blee | p3nguin: sorry, perhaps a longer PB will help |
| 18:35.18 | p3nguin | blee: What is your verbose level? |
| 18:35.25 | blee | 3 |
| 18:35.30 | p3nguin | okay |
| 18:35.38 | blee | here is the full PB |
| 18:35.40 | blee | http://pastebin.ca/2014964 |
| 18:36.04 | blee | I call into voicemail, leave a message, dial #, press 0 for the operator, and I expect it to go to ext 0, priority 1 |
| 18:36.12 | blee | ext o rather |
| 18:37.20 | blee | I can confirm, that in context 039, I indeed have exten => o,1,Goto(voicemail_zero_exit|1) |
| 18:37.27 | p3nguin | That's what I would expect, too. What asterisk version are you using that this is happening? |
| 18:37.39 | blee | 1.4.32 |
| 18:37.59 | *** join/#asterisk atan3 (~atan@unaffiliated/atan) |
| 18:38.16 | blee | I was hoping there was an *easy* way to fix this other than upgrading asterisk |
| 18:38.18 | p3nguin | Is there a specific reason you don't use the most recent version within that branch? |
| 18:38.29 | blee | our dialplan is old, and it works |
| 18:38.51 | p3nguin | And it should continue to work when you upgrade to the most recent version in the same branch, too. |
| 18:39.12 | p3nguin | Without the problem you're experiencing, even. |
| 18:39.32 | blee | this is probably true, if I need to do that I will |
| 18:39.43 | p3nguin | You need to do that. |
| 18:39.44 | blee | just a strange behavior that perhaps I would be able to work around without taking us down |
| 18:40.28 | p3nguin | Old versions are typically not supported because bug have been fixed, things have been changed. Assume the current version is the most-working version. |
| 18:41.10 | blee | most definitely, i think another reason my colleague didnt want to upgrade is the syntax change from | to , |
| 18:41.17 | blee | but anyways, no worries |
| 18:41.18 | tvc123 | justdave: yeah I looked at the voip-info site it seemed to reference that auth=md5 but that is for the incoming registration I'm looking to register going out |
| 18:41.32 | p3nguin | What's odd is that I know I would have used 1.4.32, and I didn't encounter this pri 0 behavior. |
| 18:41.47 | tvc123 | justdave: and you can use brackets to specify that it is rsa |
| 18:42.01 | p3nguin | blee: It changed from a pipeline to a comma a long time before the version you are using now, just for the record. |
| 18:42.06 | tvc123 | but there isn't any real info on how to send an md5 |
| 18:42.59 | blee | p3nguin: lol :-( |
| 18:43.05 | weta | Is there a way I can debug asterisk email? I have serveremail set to one thing in voicemail.conf. In my smtp logs I can see the messages are being rejected upstream with a different from email. using the mail command from the command line works fine. Thanks. |
| 18:44.08 | p3nguin | blee: It's pretty easy to run sed 's/|/,/g' against the dialplan, though. |
| 18:46.10 | blee | p3nguin: ill give it a shot, i im relatively new to asterisk and this dilplan is rather daunting |
| 18:46.11 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 18:46.27 | blee | scared that just replacing all the bars wont go as smoothly as I would like |
| 18:46.50 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 18:47.18 | ChannelZ | I can't think of any other uses of the pipe unless you had a bunch of logical expressions |
| 18:47.39 | leifmadsen | blee: so try app_set=1.4 under [options] in asterisk.conf |
| 18:48.29 | p3nguin | You could always output your changes to a new file and check it before putting it in place. sed 's/|/,/g' extensions.conf > extensions.conf.new |
| 18:49.26 | p3nguin | Check the new file. If it looks good, "mv extensions.conf extensions.conf.1.2 ; mv extensions.conf.new extensions.conf" |
| 18:49.33 | leifmadsen | +1 |
| 18:50.04 | ChannelZ | and if it doesn't work, well phone calls are overrated anyway. |
| 18:50.10 | blee | haha |
| 18:50.18 | blee | indeed, just forward them all to the boss |
| 18:50.18 | p3nguin | If it doesn't look good, don't put it in place. |
| 18:51.07 | ChannelZ | the only time the phone rings is when people want something from you anyway. It's a bother. |
| 18:51.14 | blee | Can I have 2 versions of asterisk built at the same time, and just move between the 2? |
| 18:52.18 | p3nguin | Not with both running? Sure. |
| 18:53.42 | WIMPy | Well, if you're carefully chosing directories you should be able to have two different version running. |
| 18:54.08 | p3nguin | When I upgrade, I usually build my new package, install the new package, then, when ready, I issue "restart now" on the CLI. |
| 18:54.13 | WIMPy | I was actually thinking about doing so on a box with shitty hardware that only works with an old zaptel version. |
| 18:55.27 | blee | what version of asterisk began enforcing commas only? |
| 18:55.48 | blee | ie, i could probably try 1.4.38 and get away with my current dialplan right? |
| 18:56.45 | p3nguin | I would change my pipes to commas right now. Then upgrade. |
| 18:57.31 | p3nguin | Least significant change first, in my opinion. |
| 19:09.45 | yonahw | anybody know of a way to catch dtmf tones and not pass them on to the other leg? What I am really trying to do is enable recording without other side hearing dtmf from dialing *1. |
| 19:10.38 | p3nguin | If you have automon enabled, does *1 pass through anyway? |
| 19:10.46 | p3nguin | I thought it didn't. |
| 19:11.31 | yonahw | p3nguin: maybe that is my problem. I thought it wasn't passing through and now it is. Maybe something changed that I'm not aware of but I don't think so. |
| 19:11.39 | justdave | I think that's what features.conf is for |
| 19:11.58 | justdave | you can define stuff that's interpreted during calls in there |
| 19:12.55 | p3nguin | Set your automon for *1 and make sure you include the w option in the Dial() command. I don't think the other side will hear the *1 being dialed. |
| 19:13.47 | WIMPy | Well if it is real DTMF I think it can only be filtered partially. |
| 19:14.01 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:14.55 | yonahw | I have in features.conf a [featuremap] automon => *1 and I have in extensions.conf under [globals] DYNAMIC_FEATURES => automon but caller is still hearing dtmf tones |
| 19:15.26 | p3nguin | Does automon turn on when they dial *1? |
| 19:16.14 | yonahw | you know I though it was but now it seems like there are less recordings than there should be. I am going to have to investigate a little more, I wonder during which cases automon isn't being activated |
| 19:16.46 | p3nguin | If the w option is not in the Dial() command in the extension being called, I don't see any way it could ever be turned on. |
| 19:16.54 | yonahw | oh it turns out that I don't have w in the outbound calls |
| 19:16.58 | yonahw | that must be the problem |
| 19:17.24 | p3nguin | That could also be why *1 is being heard on the other side of the call. |
| 19:18.21 | jpmcallister | I'm trying to use dynamic realtime to configure sip peers. I'd like to use templates. I noticed an option template in sip.conf.sample and create a column template at the table. But asterisk seems to ignore it. Is there any way to use templates with realtime? |
| 19:18.21 | yonahw | indeed, I'm going to accept it as most likely explanation |
| 19:18.47 | jpmcallister | The other options are working just fine |
| 19:24.46 | *** join/#asterisk Gary_B (~Gary_B@85.211.222.128) |
| 19:26.35 | leifmadsen | jpmcallister: I don't believe realtime can support templates. This question was actually asked and answered yesterday. |
| 19:27.28 | leifmadsen | blee: 1.6.0 I think enforced... can't remember. 1.6.2 definitely does. |
| 19:27.43 | weta | in case it might be of value to anyone here. I fixed my issue with email. I had to set postfix (my MTA) to re-write the local email address to one that was valid externally. I'm not sure why the emailserver setting in asterisk didn't do this... possibly in the way it interacts with postfix. |
| 19:28.49 | leifmadsen | weta: because asterisk is not an MTA -- that is a function of the mail server. Asterisk can only do so much. You could have also tried setting "serveremail" to something other than "asterisk" if you didn't already. |
| 19:34.01 | weta | Thanks leifmadsen. I had serveremail set to a valid email address and it was falling through to the default local email user in postfix. I'll try asterisk and map asterisk to the right external address I want to see if that works right. Appreciate the reply. |
| 19:38.23 | *** part/#asterisk Gary_B (~Gary_B@85.211.222.128) |
| 19:47.20 | jpmcallister | leifmadsen: tank you! |
| 19:50.49 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 19:50.55 | chandoo | hi |
| 19:50.57 | *** join/#asterisk CraigW76 (~techcaw@cl-427.dal-01.us.sixxs.net) |
| 19:51.25 | chandoo | i am thinking of buying acn 3000 video phone, can i get it working with out service from ACN |
| 19:52.27 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 19:52.45 | justdave | ok, so doing some packet-tracing, the difference between the working server that has audio and the one that doesn't have audio is that the working one is transmitting this as part of the sip setup instructions: |
| 19:52.50 | justdave | a=silenceSupp:off - - - - |
| 19:52.53 | justdave | and the non-working one is not sending that line |
| 19:53.15 | justdave | I can't find any related differences in config... changed default perhaps? what option would control that? |
| 19:53.39 | p3nguin | I guess it's from the phone. |
| 19:53.59 | *** join/#asterisk CraigW76 (~techcaw@cl-427.dal-01.us.sixxs.net) |
| 19:54.15 | justdave | this is asterisk sending instructions to the phone to disable the silence suppression |
| 19:54.26 | justdave | it's in the packet getting sent by asterisk to the phone during setup |
| 19:55.11 | p3nguin | Since Asterisk doesn't support silence suppression, it has to be in response to something the phone sent. |
| 19:57.14 | justdave | it's the same phone |
| 19:57.26 | justdave | only change is changing the IP of the server it's connecting to |
| 19:58.41 | p3nguin | So both Asterisk systems are identical? Same version, same build options, same configs? |
| 19:59.09 | justdave | same configs yes, same version, no. right now one of them is 1.4.26 and the other is 1.4.37 |
| 19:59.31 | justdave | (1.4.26 is the working one) |
| 19:59.39 | justdave | that's why I asked earlier if it was changed defaults |
| 20:12.39 | *** part/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 20:23.41 | *** part/#asterisk oneseventeen (~onesevent@web64.webfaction.com) |
| 20:24.31 | p3nguin | Is a SATA 3 cable backward compatible to SATA 2 drives and system boards? |
| 20:26.41 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 20:31.21 | blee | p3nguin: SATA cables are no different from SATA 3/2 |
| 20:31.23 | blee | or boards |
| 20:31.59 | p3nguin | Well, SATA 1 cables will not transfer at SATA 2 or SATA 3 speeds. So there's some difference. |
| 20:34.54 | blee | Hmm that is not what I've read or am reading |
| 20:35.01 | blee | granted, off wikipedia so :-o |
| 20:39.01 | *** join/#asterisk cmnky (debian-tor@gateway/tor-sasl/cmnky) |
| 20:53.44 | citywok | p3nguin: really? i had no idea they were different. lol. |
| 20:54.47 | p3nguin | I don't have the physical evidence through my own testing, but manufacturers and other people's tests say the early cables don't perform well with new faster drives. |
| 20:55.23 | blee | Im sure theres merit to that, but i bet its because cable manufacturers took advantage o the fact that they could cut corners |
| 20:55.33 | blee | if built to spec i bet they would work |
| 20:55.38 | blee | as expected |
| 20:57.46 | Chainsaw | p3nguin: This is true, shielding expectations have risen with newer standards. |
| 20:58.26 | p3nguin | blee: I can believe that. |
| 20:58.48 | Chainsaw | Cheap & nasty eBay SATA cables don't work very well above 1.5Gb/sec. |
| 20:58.52 | Chainsaw | This is entirely true. |
| 20:59.28 | Chainsaw | But the SATA cables that came bundled with a very old mainboard work beautifully on 3Gb/sec. |
| 20:59.43 | Chainsaw | The most annoying thing is how similar they look. Both red with black connectors. |
| 21:00.28 | blee | This makes me wonder about my Sata cables |
| 21:00.47 | blee | yep, fully expect my computer to be in pieces tonight |
| 21:00.57 | blee | at least the SATA cable on my SSD |
| 21:01.46 | p3nguin | SMART and dd might be useful to determine if you have problems with the current cables. |
| 21:02.04 | Chainsaw | dmesg will tell you everything. |
| 21:02.18 | Chainsaw | You will see the error handler kicking in and nudging the speeds down until it works. |
| 21:12.00 | alphaz | hey p3nguin, i dunno if you would know.. but i'll ask. would you know how to route calls internally? say i have a ton of did's like 000 0001 to 000 0020 and .. like 111 1112 to 111 1150.. is there a way to route these internally? like.. if someone calls 000 0020, to route it to.. whoever has that did assigned? instead of using a trunk to go out then come back in on 000 0020? |
| 21:12.32 | alphaz | by someone calls i mean someone internal calls 000 0020 |
| 21:13.56 | p3nguin | Yes. Create an extension that matches the call, which uses the Dial() command to make a call to the device you wish to receive the call. This is Asterisk 101 stuff. Maybe you should read The Book again. |
| 21:14.10 | p3nguin | ~book |
| 21:14.10 | 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, or http://ofps.oreilly.com/ |
| 21:15.38 | russellb | ~newbook |
| 21:15.38 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/ |
| 21:15.40 | jdoe | unless you explicitly bought a high end SSD, it almost certainly can't max out 1.5Gb/s anyway, so the point is probably moot. |
| 21:16.09 | alphaz | and for sata, its different cable specs |
| 21:16.14 | alphaz | but the connector is still the same. |
| 21:16.24 | jdoe | really? neat. |
| 21:20.09 | alphaz | and p3nguin. no thats not what i mean i can't make the extension 000 0020. |
| 21:20.22 | p3nguin | Why not? |
| 21:20.33 | p3nguin | If I can, surely you can. |
| 21:20.40 | alphaz | because that extension is called 12345 |
| 21:20.47 | p3nguin | That doesn't make sense. |
| 21:20.51 | alphaz | sure it does... |
| 21:20.57 | alphaz | i have extension called 12345 |
| 21:21.14 | alphaz | i have another extension called 67890 |
| 21:21.20 | p3nguin | It makes sense that extension 000 0020 can't be made because extension 12345 exists? No, that doesn't make sense at all. |
| 21:21.52 | alphaz | and when 67890 calls 000 0020, i want it to route internally to 12345 |
| 21:22.04 | p3nguin | Yeah, it's called a dial plan. |
| 21:22.18 | p3nguin | Learn it, build one, use it, make calls. |
| 21:22.30 | alphaz | thats unreasonable... i'm not gonna make 1000 dialplans one for each did. |
| 21:22.38 | *** join/#asterisk nny (~Scott@cpe-174-107-201-103.sc.res.rr.com) |
| 21:22.51 | WIMPy | So how did you make your extensions? |
| 21:22.55 | p3nguin | 67890 calling to something makes 67890 a device name. Create the necessary extension to call the other device as needed. |
| 21:22.58 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 21:23.43 | p3nguin | You'll create an extension for every DID you want routed to a specific phone, unless you devise some type of scheme where variables will end up being used. |
| 21:24.02 | alphaz | ok i'm beginning to get it i think.. |
| 21:24.07 | nny | http://www.sangoma.com/products/hardware_products/fax.html considering this for faxing with a PRI system i am building, any thoughts or experiences? |
| 21:24.17 | alphaz | so then just use dial to 12345? |
| 21:24.22 | alphaz | from that 000 0020 extension |
| 21:24.25 | alphaz | right? |
| 21:25.02 | alphaz | hmm i guess that makes sense.. |
| 21:25.04 | p3nguin | If the device is 12345, and the extension is 000 0020, you'd end up with something like this: exten => 0000020,1,Dial(SIP/12345,30) |
| 21:25.50 | alphaz | i guess i can match multiple too? like 000002.,1,Dial(...) |
| 21:26.23 | p3nguin | You can use patterns, but you have to figure out some way to use patterns in the Dial() command, too, which is usually by a variable. |
| 21:27.01 | alphaz | so first i'd have to set($some={$exten} or something? |
| 21:27.33 | p3nguin | exten => _000002.,1,Dial(SIP/1234${EXTEN:6,30) |
| 21:27.34 | WIMPy | There are a lot of ways to do it. |
| 21:27.49 | p3nguin | 0000020 would dial 12340 |
| 21:27.54 | p3nguin | 0000021 would dial 12341 |
| 21:27.59 | p3nguin | 0000022 would dial 12342 |
| 21:28.01 | p3nguin | and so on. |
| 21:28.05 | alphaz | thats pretty interesting |
| 21:28.42 | p3nguin | as long as you fix my typo, that is. |
| 21:28.43 | alphaz | so ${EXTEN:6,30} takes like from the 7th digit? |
| 21:28.46 | alphaz | for the 30th? |
| 21:28.56 | p3nguin | exten => _000002.,1,Dial(SIP/1234${EXTEN:6},30) |
| 21:29.01 | alphaz | oh |
| 21:29.07 | alphaz | so takes the 7th digit |
| 21:29.07 | alphaz | only |
| 21:29.19 | alphaz | due to.. 0 arrays |
| 21:29.22 | alphaz | i geuss |
| 21:29.25 | p3nguin | ${EXTEN:6} offsets 6 characters |
| 21:29.37 | p3nguin | so it starts at the 7th. |
| 21:30.14 | alphaz | hehe i guess that makes sense too.. too used to "programming" languages with array position 0 . so i figure the array position will be 6 which is the 7th array element |
| 21:30.20 | p3nguin | 00000255555 would dial 123455555 |
| 21:30.28 | alphaz | would it? |
| 21:30.41 | alphaz | takes everything after the offset? |
| 21:30.49 | p3nguin | offset is 6, so 7 to the end. |
| 21:31.26 | alphaz | thats pretty neat. |
| 21:31.27 | alphaz | hmm |
| 21:31.29 | alphaz | lemme go try that |
| 21:31.46 | p3nguin | You can also do something like ${EXTEN:6:2} to offset 6 and take only 2 more. |
| 21:32.08 | alphaz | ooh ranges.. |
| 21:32.09 | alphaz | thats useful |
| 21:32.10 | WIMPy | You'd better put that in to your pattern. |
| 21:32.31 | alphaz | put what into the pattern? |
| 21:32.39 | p3nguin | Yeah: exten => _000002[0-9],1,Dial(...) |
| 21:32.47 | alphaz | ah. |
| 21:32.57 | alphaz | this sounds like some take on regex |
| 21:33.01 | WIMPy | The fact that you only want two more digits and not how ever may there might be. |
| 21:33.12 | alphaz | is it using regex? |
| 21:33.22 | p3nguin | Pretty much, yes. |
| 21:33.26 | alphaz | ah. |
| 21:33.29 | WIMPy | No. IT's a lot simpler. |
| 21:33.37 | alphaz | oh. |
| 21:33.43 | seanbright | more like globbing |
| 21:33.52 | alphaz | oooh. |
| 21:33.54 | p3nguin | I don't think it will accept all possible regexp in there, but it works on the same principle. |
| 21:34.10 | p3nguin | _000002[0123489] |
| 21:34.12 | WIMPy | But you really shoult read one of the books. They are really useful. |
| 21:34.19 | p3nguin | _000002[0-489] |
| 21:34.32 | alphaz | books and me don't generally agree. |
| 21:34.38 | alphaz | i turn on auto mode after a few pages |
| 21:34.49 | p3nguin | At least use it for reference. |
| 21:34.52 | alphaz | ya |
| 21:34.56 | alphaz | that may be a good idea |
| 21:35.51 | alphaz | i think i'm getting pretty confused becuase stupid freepbx is just doin all sorts of stuff that i wouldnt do if i were just doin straight asterisk |
| 21:36.15 | alphaz | just figured out that i can append stuff to an extension definition yesterday i was like wth is this. |
| 21:36.19 | p3nguin | That's why we don't support FreePBX here. At all. |
| 21:36.19 | WIMPy | That's what it's known for. |
| 21:37.38 | alphaz | i wonder if theres a way to do all the sip.conf in sql... |
| 21:37.40 | alphaz | that would be nice.. |
| 21:37.58 | WIMPy | That's called realtime. |
| 21:38.28 | alphaz | cuz then i could make a stupid interface myself to just change stupid rows instead of typing all sorts of shit all over the place.. |
| 21:38.55 | alphaz | lemme look up realtime |
| 21:39.34 | *** join/#asterisk ChUbB (~IceChat7@cpc2-aztw12-0-0-cust229.aztw.cable.virginmedia.com) |
| 21:39.36 | WIMPy | You can also just generate some text file to incluse in your sip.conf. |
| 21:39.44 | alphaz | hmm thats true too |
| 21:41.45 | *** join/#asterisk mateu (~mateu@suryahunter.com) |
| 21:42.04 | alphaz | but i think if you do it that way you'll have to reload each time? |
| 21:42.58 | WIMPy | Just what you changed. |
| 21:43.26 | alphaz | ya. well i mean if you added an extension. you'd have to reload sip... |
| 21:43.29 | WIMPy | And that shouldn't have any side effects. |
| 21:44.34 | WIMPy | Yes. When I'm done generating the confs i do an asterik -rx "sip reload"; asterisk -rx "extensions reload". No big deal. |
| 21:44.55 | alphaz | oh |
| 21:45.03 | alphaz | hmm |
| 21:46.10 | alphaz | ok wait so when i create an extension say 000002X, how does it match that.. because i start with [000002???] at the top.. cuz i'm not in the exten => patterh part yet.. |
| 21:48.00 | WIMPy | I don't understand that question. |
| 21:49.59 | alphaz | ok like.. what context would that be under? |
| 21:50.22 | alphaz | the exten => _000002., |
| 21:52.18 | p3nguin | <alphaz> ya. well i mean if you added an extension. you'd have to reload sip... <--- No, if you add an extension, you have to reload extensions.conf with dialplan reload. |
| 21:53.54 | p3nguin | The things in square brackets in extensions.conf are the contexts. |
| 21:54.00 | alphaz | ya |
| 21:54.03 | alphaz | that i get |
| 21:54.09 | p3nguin | The lines that start with exten are extensions. |
| 21:54.12 | alphaz | oh |
| 21:54.14 | alphaz | ok |
| 21:54.39 | p3nguin | By the way, this is all in the book. |
| 21:54.44 | alphaz | but then usually my sip devices use the from-internal context |
| 21:55.05 | p3nguin | That's a sane context, in my opinion. |
| 21:55.55 | alphaz | but does that i mean i'll have to modify the from-internal context to include another context i make that matches the 000stuff? |
| 21:57.19 | p3nguin | Those DIDs (extensions) are dialed from outside, so you would put them in a relevant context. You should have a sip peer already defined in sip.conf that matches the system sending calls to those DIDs; the context is defined there. |
| 21:57.58 | p3nguin | Something like context=from-itsp |
| 21:58.07 | alphaz | yes when they dialed from outside that yes. |
| 21:58.10 | p3nguin | or context=voipms-inbound |
| 21:58.14 | alphaz | but when they dialed from inside. |
| 21:58.26 | p3nguin | You want inside people to dial the outside phone number? |
| 21:58.30 | alphaz | yes. |
| 21:58.35 | alphaz | if they dial the outside number |
| 21:58.40 | alphaz | i want it to go route inside |
| 21:58.43 | alphaz | isntead of going to itsp |
| 21:58.44 | alphaz | then back |
| 21:59.18 | p3nguin | Then you have two choices: include the inbound context in your internal context, or duplicate all the DID extensions inside the internal context. |
| 21:59.22 | Tech_Travis | does anyone have recommendations on sip phones outside the firewall? |
| 22:00.01 | alphaz | tech: grandstream works pretty well i've found, cisco takes a bit more fiddling and maybe some port opening. |
| 22:00.24 | alphaz | hmm i'll try to include it in my internal context |
| 22:00.32 | p3nguin | Never include an outbound context where a person dialing inbound from outside can access it. You don't want people outside calling back out on your dime. |
| 22:00.51 | alphaz | hmm |
| 22:01.02 | alphaz | thats one complicated scenario.. it just flew right over my head |
| 22:01.20 | alphaz | outbound.. context.. |
| 22:01.24 | alphaz | how does that even happen |
| 22:01.37 | p3nguin | You don't want Joe Blow to call inbound and be able to dial back outbound and call the Dominican Republic or something. |
| 22:01.42 | alphaz | no i understand |
| 22:01.44 | alphaz | but |
| 22:01.48 | alphaz | i mean context wise |
| 22:01.51 | alphaz | how would that even happen |
| 22:02.06 | alphaz | if hes already calling in, |
| 22:02.11 | alphaz | how could he call out |
| 22:02.14 | phix | :D |
| 22:02.19 | p3nguin | Some people include other contexts without drawing a picture as to how the calls can flow. |
| 22:02.55 | alphaz | but how would someone initiate a call like that |
| 22:03.05 | p3nguin | Such as inbound includes internal, internal includes outbound. Now you can call in, get into the internal context, and call back outbound. |
| 22:03.09 | alphaz | if joe blow calls 000 0020 how would he dial 01 001 92817398 |
| 22:03.21 | p3nguin | Usually it happens via IVR menus. |
| 22:03.24 | alphaz | oh |
| 22:03.37 | p3nguin | I was just saying to think before you include. |
| 22:04.21 | alphaz | well i'd include a context that i create with one line :\ exten => _000002.,... dial(bla).. i'm not sophisticated enough to tie in contexts like that lol.. |
| 22:04.32 | alphaz | hehe i haven't gotten that far yet.. (IVR) |
| 22:04.38 | p3nguin | I should get a new bumper sticker: Don't drink and include => |
| 22:04.52 | thehar | rofl |
| 22:04.56 | alphaz | haha |
| 22:05.00 | thehar | p3nguin +1 |
| 22:05.25 | Tech_Travis | alphaz: Thanks, Currently using Cisco but don't want to use TFTP outside the firewall. I'll check out Grandstreams. |
| 22:05.31 | alphaz | exactly |
| 22:05.33 | alphaz | tftp. |
| 22:05.36 | alphaz | but what you can do |
| 22:05.39 | alphaz | is configure them first |
| 22:05.42 | thehar | Grandstream == pain |
| 22:05.45 | alphaz | then after that they can work without the tftp |
| 22:05.48 | thehar | horrible horrible pain |
| 22:05.51 | alphaz | thehar |
| 22:05.53 | alphaz | not really |
| 22:05.58 | alphaz | took me like 2 min to configure one |
| 22:06.03 | alphaz | just on the phone itself type in the info |
| 22:06.06 | alphaz | and you done :\ |
| 22:06.11 | thehar | configuring is cake |
| 22:06.14 | Tech_Travis | true, but if one of the users ever does a complete reset they're hosed. |
| 22:06.20 | alphaz | ya |
| 22:06.23 | alphaz | thats very true. |
| 22:06.35 | alphaz | cisco painnnnnn if you giving to users. outside |
| 22:06.40 | p3nguin | If you load the SIP firmware onto the Cisco phones once, they really don't need a tftpd after that. They do take longer to boot up if it doesn't exist, but they still do load and work. |
| 22:06.42 | alphaz | tftp = dont let it out -_- |
| 22:07.15 | Tech_Travis | never ever ever if I can another route. |
| 22:07.23 | Tech_Travis | can find |
| 22:07.24 | alphaz | yup p3nguin, if only theres a way to make that stupid configuring vlan thing go away faster -_- longest thing ever. |
| 22:07.26 | p3nguin | VPN? |
| 22:07.54 | alphaz | i dont think the ciscos do vpn . you'd have to vpn the whole network together. |
| 22:08.03 | *** part/#asterisk superm1 (~superm1@ubuntu/member/superm1) |
| 22:08.24 | alphaz | and well i mean if you giving the phone to client.. i dont think you want them to be "in" your network hehe. |
| 22:09.08 | Tech_Travis | yeah, I don't think they do VPN either. These are for our guys to work from home so VPN would be more preferrable than TFTP outside. |
| 22:09.23 | p3nguin | I need to ask my friend how his is configured. He runs a Cisco at home to work from home, and the company requires him to use a VPN. |
| 22:09.37 | Tech_Travis | oooh. |
| 22:10.06 | alphaz | well for sure you'd connect to vpn with either your router, or a separate router. |
| 22:10.11 | alphaz | then from that router plug to cisco phone |
| 22:10.15 | alphaz | then you'll be happy. |
| 22:10.16 | alphaz | i guess |
| 22:10.31 | p3nguin | If the phone doesn't have a VPN client, then yeah you'd have to have some other end point. |
| 22:10.36 | alphaz | yup. |
| 22:10.42 | alphaz | ah well. |
| 22:11.01 | alphaz | i just.. configured and brought it somewhere else.. no need for tftp.. |
| 22:11.05 | alphaz | the settings generally stay |
| 22:11.17 | alphaz | unless you punch in super secret patterns that reset the phone to oblivion |
| 22:12.21 | alphaz | thx for your help btw p3ng |
| 22:12.25 | Tech_Travis | well, our user does have 7 year old so it may be possible to crack that code. |
| 22:12.31 | alphaz | well.. |
| 22:12.33 | alphaz | its like |
| 22:12.41 | alphaz | hold pound on power up |
| 22:12.47 | alphaz | then within those like 30 sec or whatever |
| 22:12.50 | Tech_Travis | 1-9 star 0 pound. |
| 22:12.52 | alphaz | 123456789 |
| 22:12.53 | alphaz | ya |
| 22:12.57 | alphaz | star0pound |
| 22:13.19 | alphaz | not very likely someone will do that while booting then holding pound |
| 22:13.26 | alphaz | by accident. |
| 22:14.00 | p3nguin | I do it every time I go into someone's office and they leave me alone for a few minutes. |
| 22:14.06 | alphaz | haha |
| 22:14.08 | alphaz | rofl. |
| 22:14.19 | alphaz | thats not exactly an "accident" :D |
| 22:14.23 | Tech_Travis | that'll teach them. |
| 22:14.39 | alphaz | but i mean all they'd have to do is bring it back to the office and plug it in for a min :\ |
| 22:14.57 | Tech_Travis | we're in California, he's in Germany. |
| 22:15.01 | alphaz | rofl |
| 22:15.14 | alphaz | hmm 300$ fedex same day? :D |
| 22:15.18 | thehar | is in California as well! |
| 22:15.25 | alphaz | i'm in Ca... nada |
| 22:16.12 | alphaz | yes we have phones here in canada.. :\ we upgraded from telegrams last month |
| 22:16.22 | thehar | lol |
| 22:17.10 | thehar | depends where in Canada |
| 22:17.59 | alphaz | well i mean unless you're in the northwest territories or yukon or some rural town.. i think its pretty safe bet you have some sort of modern society around. |
| 22:18.18 | *** join/#asterisk TdM2 (~TdM2@c-24-2-244-4.hsd1.ma.comcast.net) |
| 22:20.29 | alphaz | i'ma go try that magical extensions stuff bbl |
| 22:20.46 | xSmurf | alphaz: we're still waiting on that thing they call internetz |
| 22:20.58 | xSmurf | I heard Bell was trying to get it working, but they still haven't figured it out yet |
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| 22:33.06 | alphaz | lol |
| 22:33.10 | alphaz | inter what |
| 22:33.11 | alphaz | ? |
| 22:33.19 | alphaz | sounds like some kind of high tech stuff |
| 22:33.32 | alphaz | last i heard we were doing dial up bbs stuff |
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| 22:42.31 | plut0 | anyone know how to get fax receive working on 1.8? it doesn't seem to recognize an incoming fax |
| 22:42.47 | alphaz | fax using what |
| 22:42.56 | alphaz | the digium scam channels? |
| 22:43.11 | plut0 | receivefax() |
| 22:43.40 | alphaz | sorry, cant help you there.. i dont do digium fax. they scam you. so unreasonable for price per channel. |
| 22:43.54 | alphaz | i just went t38modem or iaxmodem or something of the sort |
| 22:44.03 | plut0 | i'm doing it over sip |
| 22:44.20 | alphaz | ya so am i. |
| 22:44.20 | plut0 | not sure what you're talking about |
| 22:44.37 | plut0 | ok great |
| 22:44.40 | alphaz | you connect t38modem or iaxmodem to asterisk |
| 22:44.51 | plut0 | whats that? |
| 22:45.40 | alphaz | its.. a fax modem thing that connects to asterisk |
| 22:46.06 | alphaz | software fax modem thing. |
| 22:46.09 | alphaz | so when someone calls.. you redirect to the t38modem or iaxmodem. and it'll receive fax |
| 22:46.10 | plut0 | alphaz: have you tried with receivefax() ? |
| 22:46.12 | alphaz | using hylafax |
| 22:46.15 | alphaz | no i haven't |
| 22:46.23 | plut0 | ok, thats what i'm looking for help with |
| 22:46.26 | alphaz | k |
| 22:46.45 | alphaz | but as far as i've read.. |
| 22:46.48 | alphaz | theres nothign to it |
| 22:46.51 | alphaz | for the asterisk |
| 22:46.52 | alphaz | one |
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| 23:15.05 | tvc123 | I'm hoping to get a little bit of best practice advice ... I'm running 1.8.0 compiled from source ... I just found a bug fixed in 1.8.1 ... should I be running from the trunk? is there an easy way to set the same compile options without reconfigureing it again? |
| 23:16.38 | tvc123 | is there some sort of reference for best practice while upgrading? |
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| 23:18.33 | marl_scot | hi folks, am trying to test sip indicators, (flashing when on call etc) can anyone recomend a sip softphone for wind*ws that shows the line states? |
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| 23:30.47 | plut0 | anyone have any luck getting fax detect working on 1.8? |
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| 23:55.27 | marl_scot | hi, ive come accross this before but cant find the answer now :( I have been testing 1.8 with 2 softphones, they appeared to be working a short while ago, but now i keep getting 503 service unavailbe when i try and answer a call from one sip on the other, only thing i can see that sparked alarms was that i am seeing my external ip address under sip debug for hte sip phones, when everything im testing is internal, anyone got any pointers? |
| 23:56.36 | WIMPy | What did you put in to the phones as registrar/proxy? |
| 23:56.55 | marl_scot | local ip of the * box |
| 23:58.05 | WIMPy | And where exactely do you see the external one? |
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