| 00:01.19 | jc319 | Is there a particularly easy to use Windows SIP softphone that might work better than zoiper/x-lite? just a basic dialpad/quickdial pad and a voicemail button? |
| 00:28.28 | *** join/#asterisk pdtpatrick__ (~pdtpatric@ip68-4-0-113.pv.oc.cox.net) |
| 00:32.32 | *** join/#asterisk voxter (~hardcore@macpro.daytonhome.voxter.net) |
| 00:32.38 | voxter | any of you ever work with chan_unistim.so ? |
| 00:32.55 | voxter | my asterisk, (1.8) appears to attempt to parse the config file, then "hang" and never fully load it. |
| 00:36.39 | Lantizia | tzafrir_laptop, hey! guess what I've remembered by question like 2 days later! :P |
| 00:36.58 | Lantizia | *my |
| 00:37.02 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 00:42.11 | Mango | wtf! |
| 00:42.25 | Mango | I just realized that the microphone for the handsfree portion of my Linksys SPA-921 is on the back of the phone. |
| 00:42.29 | Mango | What's it doing there!? |
| 00:43.17 | voxter | aastra did that too on the 31i, stupid as hell |
| 00:43.38 | Mango | I wonder if I can mod it. |
| 00:44.19 | *** join/#asterisk Phlunk3 (~phlunk3@203.89.166.247) |
| 00:52.59 | leifmadsen | cj: make sure you file an errata on the o'reilly site so I can look into it |
| 00:53.38 | WiretapWork | you know why they do that I assume? |
| 00:53.42 | WiretapWork | Mango, |
| 00:57.05 | Mango | Why? To make it farther away from the speaker? |
| 00:57.32 | WiretapWork | to prevent feedback, wind noise and echo |
| 00:57.38 | Mango | ah |
| 01:04.34 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 01:10.03 | *** join/#asterisk Kumbang (~unknown@180.245.137.5) |
| 01:19.10 | jc319 | Is anyone using some kind of basic heartbeat script/method to check if primary server is down and launch 2nd server? |
| 01:19.27 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-231-136-192.tc.ph.cox.net) |
| 01:32.34 | *** join/#asterisk nightrid3r (~kvirc@91.176.11.145) |
| 01:35.47 | leifmadsen | jc319: yes I've done that with LinuxHA -- any of the basic scripts out there showing how to launch an application when a server falls over should be enough for you |
| 01:44.08 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-231-136-192.tc.ph.cox.net) |
| 01:45.52 | *** join/#asterisk nightrid3r (~kvirc@91.176.19.225) |
| 01:46.36 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 01:47.36 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 01:52.46 | *** join/#asterisk X-Rob (~Rob@eth2083.qld.adsl.internode.on.net) |
| 02:00.51 | *** join/#asterisk shmaltz (~chatzilla@mail.dmaven.com) |
| 02:00.57 | shmaltz | hi everyone |
| 02:03.19 | shmaltz | ~anyone here? |
| 02:03.19 | infobot | No, we're all bots :p |
| 02:03.41 | WiretapWork | shmaltz, |
| 02:03.43 | WiretapWork | ~ask |
| 02:03.44 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 02:04.09 | shmaltz | WiretapWork, no questions here, just bored. Want to talk? |
| 02:04.21 | WiretapWork | nope, I'm working |
| 02:05.01 | shmaltz | what are you working on? |
| 02:05.32 | WiretapWork | $dayJob |
| 02:06.08 | shmaltz | anyway I can help you working now? |
| 02:06.12 | WiretapWork | nope |
| 02:06.59 | shmaltz | ouch, would you at least join me with my beer? |
| 02:18.04 | *** join/#asterisk PhoenixMage (~Phoenix@ncao.vtcif.telstra.com.au) |
| 02:19.04 | *** join/#asterisk gruvfunk (~chatzilla@cpe-68-172-221-157.hvc.res.rr.com) |
| 02:19.39 | p3nguin | Is there any reason to use an unusual extension for accepting ISN calls from the internet? If my local extension number is 4321, would 4321*123 be a sensible numbering convention to reach 4321 on ISN domain 123? Any reason to use a fancy "code" for inbound calling? |
| 02:20.00 | gruvfunk | paulc: when you get a chance, can you ping me about that favor? |
| 02:21.53 | WiretapWork | p3nguin, generally some obfuscation is desired |
| 02:22.11 | WiretapWork | p3nguin, as publicising an internal extension could be considered a security risk |
| 02:22.37 | WiretapWork | I just prefix 020955 for mine, but thereare probably better ways |
| 02:23.10 | WiretapWork | it definitely makes sense, with ISN to dump calls if they don't match ISN inbound numbers, if they come in anonymous |
| 02:23.45 | *** join/#asterisk Tech_Travis (~Travis@cpe-76-168-191-127.socal.res.rr.com) |
| 02:25.12 | p3nguin | How did you arrive at that prefix? |
| 02:26.16 | *** join/#asterisk sourcode (~code@ppp-61-90-7-128.revip.asianet.co.th) |
| 02:27.11 | WiretapWork | started with an invalid area code for this locale (020) and added a 'bogon' sub-area prefix (955) |
| 02:28.19 | p3nguin | What is 955 actually used for? |
| 02:28.21 | WiretapWork | basically it makes it possible for people with stupid dialplans (no outbound dialling prefix) to call the number |
| 02:28.24 | WiretapWork | nothing at all |
| 02:28.39 | PhoenixMage | hmmm, why is my 7975G with SIP seeing "1" when I dial the 100 extension? |
| 02:28.43 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-231-136-192.tc.ph.cox.net) |
| 02:28.45 | p3nguin | I was thinking it might be like our 555 prefix. |
| 02:29.04 | shmaltz | ~isn |
| 02:29.04 | infobot | from memory, isn is ITAD Subscriber Number (see 'itad'). An ISN is a method of dialing SIP URI's via a standard keypad on a telephone. Because of the alphanumeric nature of SIP URIs, it is difficult to dial them via the keypad on your phone. The use of ISN numbers simplifies this by utilizing DNS lookups to map the ISN number to a domain. See http://www.freenum.org for more information. |
| 02:29.10 | WiretapWork | 0N955 is owned by an alternate PTSP who isn't using it yet |
| 02:29.12 | p3nguin | Using extensions starting with 1 is usually a terrible idea. |
| 02:29.23 | Grady2000 | sys.stdout.write('EXEC Dial(SIP/5665,20)\n') anyone know why this is incorrect syntax? asterisk does not accept it and yes 5665 is a registered sip extension |
| 02:29.25 | PhoenixMage | p3nguin: Is more just me playing around |
| 02:29.26 | WiretapWork | there are no 'invalid' sub-area prefixes in NZ |
| 02:29.32 | p3nguin | oh |
| 02:29.47 | WiretapWork | but '020' is an invalid area code, so it helps |
| 02:30.03 | Grady2000 | that command works fine in extenstions.conf |
| 02:30.12 | Grady2000 | but not in my agi script |
| 02:30.26 | WiretapWork | (all area codes in NZ are 2 digit except for 02x, which is 3 and normally used for cellphones or premium rate calls, 020 is unroutable) |
| 02:30.29 | p3nguin | phoenixmage: Look at your dialplan in the phone. You'll probably understand what happened and why extensions starting with 1 is a bad idea. |
| 02:31.33 | p3nguin | How would I go about finding out those area codes in North America? |
| 02:32.10 | WiretapWork | no idea |
| 02:32.25 | WiretapWork | I only know because I used to work in telco and have kept the spreadsheet with all the code allocations on it :P |
| 02:32.27 | ectospasm | http://www.bennetyee.org/ucsd-pages/area.html |
| 02:32.28 | p3nguin | I'll check NANP. |
| 02:32.30 | shmaltz | p3nguin, what area code do you want to findout? |
| 02:32.39 | p3nguin | any that are not able to be routed |
| 02:32.56 | ectospasm | ^ that page has a more or less complete listing of NANP area codes. |
| 02:32.59 | shmaltz | p3nguin, for starters anything starting with 1 or 0 or x11 |
| 02:33.03 | p3nguin | I'll check it. |
| 02:33.38 | shmaltz | but you wouldn't want to use x11 or 1 anyhow |
| 02:33.51 | shmaltz | but 0 looks good to me |
| 02:33.57 | p3nguin | I've got an idea, based on CallCentric's 777 area code. I'll use that list to see if I can make it work. |
| 02:33.58 | shmaltz | so does 11 |
| 02:34.21 | WiretapWork | shmaltz, 011 could end up with accidental international routing |
| 02:34.48 | shmaltz | wirteapwork, thats why I said no x11 but 0 or 11 |
| 02:34.49 | WiretapWork | p3nguin, the main reason is that there WILL be pbx admins out there that won't do their dialplan matching for ISN based on the presence of a '*' in the number :P |
| 02:35.13 | p3nguin | I don't plan to accept * in the numbers. |
| 02:35.20 | PhoenixMage | p3nguin: I didnt think the phone had an inbuilt dialplan as I have no dialplan.xml on the tftp server |
| 02:35.36 | *** join/#asterisk chrisjunkie (~Chris@2001:4428:22d:2:208:2ff:fe7e:6312) |
| 02:36.01 | WiretapWork | p3nguin, no, I mean outbound matching |
| 02:36.06 | p3nguin | oh |
| 02:36.15 | WiretapWork | since the incoming number does not generally contain the '*' |
| 02:36.35 | WiretapWork | (you will want to set your callerID in some manner so that the calls appear to come from your full ISN number) |
| 02:36.38 | p3nguin | For outbound, I use a special prefix and parse out just the extension portion of the number. |
| 02:36.52 | WiretapWork | yep, that's correct |
| 02:36.54 | p3nguin | I still have to work on ISN caller ID. |
| 02:37.38 | WiretapWork | I can dig up my dialplan if you want, its a modified version of the freepbx outbound, but it sets up the ISN correctly |
| 02:38.27 | p3nguin | It might help to see what you're doing with caller ID. |
| 02:38.52 | p3nguin | Otherwise, my calls will send the configured callerid value from the phone's peer entry. |
| 02:41.19 | WiretapWork | this is the outbound route: http://dev.inetpro.org/pastebin/642 |
| 02:42.03 | WiretapWork | it takes the phone's peer entry and wraps it in the ISN DID |
| 02:42.55 | WiretapWork | obviously I'm not ISN 1234, just an example :P |
| 02:43.23 | p3nguin | I know. |
| 02:44.19 | WiretapWork | I did have to completely modify the way FreePBX handles outbound callerID to stop it sending 020955092823180*1410 :P |
| 02:44.39 | p3nguin | hmm |
| 02:44.42 | WiretapWork | (normal outbound CID, wrapped in ISN data) |
| 02:46.55 | p3nguin | I see where I went wrong in my outgoing CID. I was using @ in it. |
| 02:47.09 | p3nguin | @domain rather than *ISN |
| 02:47.21 | WiretapWork | I've also changed it so that inbound calls get the outbound dialling prefix prepended to them now |
| 02:47.30 | p3nguin | I must have done that pretty late at night. |
| 02:47.35 | WiretapWork | and you don't want to do that p3nguin, as the idea behind CID is that people can call you back :P |
| 02:47.44 | WiretapWork | most people who use ISN don't have SIP URI dialling set up |
| 02:48.16 | p3nguin | Yeah, now that I see what I did, it makes sense to NOT do that. I know ISN is to aid in the calling of SIP URI from a standard phone keypad. |
| 02:48.40 | WiretapWork | I personally feel that ISN holds great promise for the future of telecommunications |
| 02:49.00 | WiretapWork | as it removes the ITSP from the mix |
| 02:49.48 | p3nguin | It'll keep me from having to create an extension in my dial plan to call you@sip.your.domain. |
| 02:50.26 | WiretapWork | yeah |
| 02:51.18 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 02:51.59 | p3nguin | I have a bunch of those for people that use SIP URIs but not ISN. |
| 02:52.17 | WiretapWork | yep |
| 02:52.19 | p3nguin | It's really not practical, but it's my PBX so I basically do what I want. |
| 02:52.25 | WiretapWork | the two tend to be mutually exclusive |
| 02:53.05 | p3nguin | As long as I don't break anything, no one has any right to scream at me over it. |
| 02:53.11 | WiretapWork | hahaha |
| 03:02.44 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 03:05.40 | p3nguin | There. That should do it. |
| 03:05.57 | *** join/#asterisk Micc (~Micc@c-98-232-41-66.hsd1.wa.comcast.net) |
| 03:06.24 | Micc | anyone know how to setup a mitel phone? |
| 03:07.22 | p3nguin | I fixed the outbound ISN CID and also added the outgoing dial prefix to the inbound ISN CID. |
| 03:07.49 | Micc | I've got a mitel 5224 that seems to need a tftp server. Isn't there a web interface? |
| 03:10.11 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 03:16.28 | *** join/#asterisk freakazoid0223 (~matt@pool-173-49-209-91.phlapa.fios.verizon.net) |
| 03:18.34 | *** join/#asterisk radic (~radic@dslb-178-002-232-170.pools.arcor-ip.net) |
| 03:20.13 | *** join/#asterisk MaliutaLap (~biteme@kiev.lusan.id.au) |
| 03:39.20 | pabelanger | Micc: no |
| 03:41.55 | *** join/#asterisk vinhdizzo (~vinh@pcp038194pcs.islay.reshall.calpoly.edu) |
| 03:44.06 | WiretapWork | Micc, most commercial phones don't have webinterfaces |
| 04:00.41 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 04:01.42 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 04:10.10 | p3nguin | How many people actually use ISN for everyday phone calls? |
| 04:10.42 | WiretapWork | not many yet, but I think we'll see it growing |
| 04:13.06 | p3nguin | I bet in a few years from now, I won't have had a single legitimate (not a test) call using it. |
| 04:13.31 | p3nguin | It's going to be like IPv6... just a novelty for the geeks. |
| 04:13.54 | WiretapWork | lol |
| 04:14.05 | WiretapWork | there is a lot of pressure to move to IPv6 here |
| 04:14.11 | p3nguin | I can imagine. |
| 04:14.21 | p3nguin | The pressure should have been on YEARS ago. |
| 04:14.31 | WiretapWork | agreed |
| 04:20.32 | *** join/#asterisk Ycarene (~Ycarene@24-116-61-193.cpe.cableone.net) |
| 04:21.19 | *** join/#asterisk |ance|ott (~albertoll@cable201-232-157-117.epm.net.co) |
| 04:24.20 | PhoenixMage | WiretapWork: Thanks for the tips on tcp sip on the 7975, all working now |
| 04:24.28 | WiretapWork | no problem |
| 04:24.46 | WiretapWork | PhoenixMage, do you need BLF pickup dialplan? |
| 04:24.57 | PhoenixMage | WiretapWork: Nah, more playing atm |
| 04:25.07 | WiretapWork | sweet, I have it at the ready if needed |
| 04:25.10 | PhoenixMage | I plan to compile 1.8 soon atm I am just using astlinux |
| 04:25.24 | WiretapWork | ah |
| 04:25.27 | PhoenixMage | When I compile I will use the patches you mentioned to enable |
| 04:25.44 | WiretapWork | ah, right, you don't even have BLF itself operational :P |
| 04:26.06 | PhoenixMage | indeed |
| 04:29.17 | PhoenixMage | Was not working for a while until I found something about using USECALLMANAGER in the lines <proxy> config |
| 04:29.52 | WiretapWork | eh? |
| 04:30.00 | WiretapWork | the config on my page should have worked |
| 04:30.09 | *** join/#asterisk joseph (~joe@unaffiliated/joseph) |
| 04:30.19 | *** part/#asterisk joseph (~joe@unaffiliated/joseph) |
| 04:30.33 | WiretapWork | or are you using SIP/UDP (as I never got that working) |
| 04:31.56 | PhoenixMage | Nah its SIP TCP |
| 04:32.15 | PhoenixMage | <proxy>USECALLMANAGER</proxy> |
| 04:32.15 | PhoenixMage | I need to put that in my cnf.xml |
| 04:33.07 | WiretapWork | oh wow, I completely forgot to even put the config file on my page *embarrassed* |
| 04:33.29 | PhoenixMage | Thinking of dropping back to 1.6 just for chan-sccp-b |
| 04:33.37 | PhoenixMage | For my 7925 |
| 04:33.43 | WiretapWork | tempting, it is |
| 04:37.52 | Ycarene | Will asterisk work with winmodems? |
| 04:39.12 | WiretapWork | no |
| 04:39.23 | WiretapWork | I am assuming you want to use a winmodem as an FXO |
| 04:39.30 | WiretapWork | winmodems are not designed to carry voice |
| 04:39.47 | WiretapWork | you CAN use an Intel 536EP as an X100P clone, but don't expect good performance |
| 04:45.29 | *** join/#asterisk moy_ (~moy@69.157.45.156) |
| 04:50.37 | *** join/#asterisk Scorp1us (~as@pool-72-81-130-240.bltmmd.fios.verizon.net) |
| 04:50.53 | Scorp1us | Hi I just installed AsteriskNow and it says to go to a web page |
| 04:51.03 | Scorp1us | but there is no http server running |
| 04:51.32 | |ance|ott | jejeje |
| 04:51.46 | DND | hi guys. i have this: Executing [s@macro-dialout-trunk:21] i wanted to know what "trunk:21" means |
| 04:52.05 | Scorp1us | what do I need to do to get it running? |
| 04:52.26 | |ance|ott | <PROTECTED> |
| 04:52.36 | *** join/#asterisk irroot (~irroot@dsl-185-122-97.dynamic.wa.co.za) |
| 04:52.42 | |ance|ott | excuseme |
| 04:52.44 | Scorp1us | there is no httpd |
| 04:52.48 | |ance|ott | <PROTECTED> |
| 04:52.52 | DND | thats impossible |
| 04:52.58 | |ance|ott | mmm |
| 04:53.03 | DND | unless you setup asterisk without freepbx or asteriskgui |
| 04:53.18 | Scorp1us | ah, i did the last one |
| 04:53.26 | DND | nope no httpd there |
| 04:53.28 | Scorp1us | asteriskgui |
| 04:53.33 | DND | asteriskgui? |
| 04:53.36 | irroot | mawnin |
| 04:53.41 | Scorp1us | but there is no gui... just a prompt |
| 04:53.42 | |ance|ott | then i dont remenber that |
| 04:54.02 | DND | asterisknow doesnt have gui |
| 04:54.05 | DND | command line |
| 04:54.11 | DND | you access the web panel from another computer |
| 04:54.17 | |ance|ott | asterisknow has a freepbx |
| 04:54.27 | Scorp1us | but there is no httd installed... |
| 04:54.38 | |ance|ott | mmmm |
| 04:54.45 | Scorp1us | so how do I access the web panel |
| 04:54.54 | DND | access it from another computer |
| 04:55.01 | DND | using the ip address of the server |
| 04:55.03 | |ance|ott | http://ip |
| 04:55.08 | Scorp1us | there is no httpd installed |
| 04:55.16 | |ance|ott | do you hace ssh acces? |
| 04:55.24 | |ance|ott | *have |
| 04:55.31 | Scorp1us | i'm on it in a VM, I'm on the root console |
| 04:55.52 | DND | try: service httpd start |
| 04:55.52 | |ance|ott | mmm can you make ping from another computr? |
| 04:56.13 | DND | |ance|ott, the prob is he says it doesnt have httpd |
| 04:56.28 | irroot | httpd will most likely be apache |
| 04:56.34 | |ance|ott | but you know, users |
| 04:56.44 | DND | in my *now its called httpd |
| 04:56.45 | Scorp1us | hrm. I can't ping it but i have it set for NAT |
| 04:56.57 | |ance|ott | jajajaajaja |
| 04:57.02 | WiretapWork | Scorp1us, you cannot access a web panel without a web server service |
| 04:57.04 | |ance|ott | xD |
| 04:57.12 | Scorp1us | right. |
| 04:57.13 | WiretapWork | |ance|ott, in english, it is 'hahaha' |
| 04:57.23 | DND | but if he installed *now with *gui, there should be one running |
| 04:57.29 | Scorp1us | i can't beleive he's typing his acent |
| 04:57.30 | WiretapWork | Scorp1us, if nothing is listening on port 80, how do you expect to talk to it? |
| 04:57.30 | |ance|ott | im from other country men but thank you |
| 04:57.52 | Scorp1us | wait. i'm saying there is nothing there listening |
| 04:57.55 | WiretapWork | Scorp1us, spanish/portuguese 'j' makes a similar sound to the english 'h', so is used for laughing instead |
| 04:58.13 | |ance|ott | thts right wire |
| 04:59.09 | |ance|ott | netstat -putan | grep 80 |
| 04:59.25 | DND | hmmprobably a bad install? |
| 04:59.36 | |ance|ott | yeah i think that DND |
| 04:59.40 | |ance|ott | too |
| 04:59.59 | DND | because from the boot splash of *now, you will have to choose from 6 options |
| 05:00.06 | Scorp1us | yeah, there's soem weird thigns afoot |
| 05:01.00 | DND | * 1.6 with freepbx, * 1.4 with freepbx, * 1.6 with *gui, * 1.4 with *gui, * 1.6 without gui, * 1.4 without gui |
| 05:01.53 | DND | Scorp1us, try re-burning it using a slow burning speed then re-do the installation if its a fresh install |
| 05:02.54 | Scorp1us | so, which one do I want? I'm not reall trying to run a PBX |
| 05:02.55 | |ance|ott | jejeje xD |
| 05:03.04 | Scorp1us | I just need asterisk and some scripts |
| 05:03.09 | Scorp1us | for IVR |
| 05:03.15 | *** join/#asterisk irroot (~irroot@196.44.226.250) |
| 05:03.31 | |ance|ott | so you dont need a freepbx interface |
| 05:03.37 | |ance|ott | whats the problem? |
| 05:04.01 | Scorp1us | i was tryin to use the webgui to set it up |
| 05:04.08 | DND | then you need freepbx |
| 05:04.12 | DND | or *gui |
| 05:04.23 | |ance|ott | yeah |
| 05:04.25 | Scorp1us | i need to set it up for some viop stuff |
| 05:04.30 | Scorp1us | voip* |
| 05:04.39 | DND | ifyou opt to use asterisk with webgui, you will have to use the webgui forever |
| 05:05.01 | Scorp1us | hrm. and i have commitment problems |
| 05:05.20 | Scorp1us | what is the the binding factor? |
| 05:07.31 | WiretapWork | the fact config is autogenerated |
| 05:08.24 | Scorp1us | ah thanks |
| 05:08.34 | |ance|ott | my god if you dont know fix it, just reinstall |
| 05:08.40 | WiretapWork | ^ |
| 05:08.52 | |ance|ott | xD |
| 05:08.54 | WiretapWork | don't waste your time fucking around with something you don't understand |
| 05:09.18 | Scorp1us | if I did that, I'd never learn anything |
| 05:09.36 | |ance|ott | jajajajajajja |
| 05:09.59 | |ance|ott | you have to read asterisknow for dummies |
| 05:10.02 | |ance|ott | book |
| 05:10.09 | WiretapWork | ~thebook |
| 05:10.09 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 05:10.29 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 05:10.35 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
| 05:10.39 | |ance|ott | is available for reading online |
| 05:10.42 | |ance|ott | ;p |
| 05:10.53 | Scorp1us | oh, i made progress my fucking around with it |
| 05:11.02 | DND | i now have other things to worry now. |
| 05:11.05 | Scorp1us | not it says file not found inmy web browser |
| 05:11.15 | Scorp1us | now* |
| 05:11.17 | PhoenixMage | Any recommendations for a softphone for windows? |
| 05:11.25 | DND | x-lite |
| 05:11.32 | DND | portgo |
| 05:11.36 | |ance|ott | x-lite sure |
| 05:11.47 | |ance|ott | wich winndows version? |
| 05:11.52 | PhoenixMage | 7 |
| 05:11.56 | |ance|ott | mmmm |
| 05:11.56 | DND | sometimes portgo has better audio quality |
| 05:12.06 | |ance|ott | xlite version 3 |
| 05:12.26 | DND | yeah i prefer that old version than the new one |
| 05:12.27 | PhoenixMage | I dont mind paying for something that has excellent features |
| 05:12.50 | |ance|ott | if you will have problems with x-lite softphonw, you can try with zoiper |
| 05:12.57 | DND | the new "free" one doesnt even have auto conference |
| 05:13.40 | PhoenixMage | I have 3cx atm doesnt seem too bad |
| 05:14.22 | DND | PhoenixMage, seems much lighter than x-lite 4 |
| 05:14.39 | PhoenixMage | dled it cos thats what I have my iphone and wanted to play rather then search for a client |
| 05:14.55 | DND | wow it has transfer button for free? |
| 05:15.25 | WiretapWork | PhoenixMage, the 3CX sipphone works on asterisk |
| 05:15.37 | WiretapWork | and the 3CX iphone client sucks |
| 05:15.58 | PhoenixMage | WiretapWork: Yeah I have it wokring, just looking for something better |
| 05:16.08 | Scorp1us | so now it says: http://192.168.1.6:8088/static/config/index.html |
| 05:16.21 | ChannelZ | uses Zoiper |
| 05:16.28 | Scorp1us | butit says "Not Found" |
| 05:17.05 | DND | wow thanks PhoenixMage, seems i can change our x-lite now |
| 05:18.24 | PhoenixMage | DND: Sorry? |
| 05:18.42 | DND | you mentioned 3cx |
| 05:18.49 | PhoenixMage | oh |
| 05:18.53 | PhoenixMage | np |
| 05:18.59 | DND | and seems it will solve our problem about call transfer |
| 05:26.32 | *** join/#asterisk thumbs (1000@unaffiliated/thumbs) |
| 05:26.53 | *** part/#asterisk thumbs (1000@unaffiliated/thumbs) |
| 05:31.59 | DND | guys any idea why my softphone cannot make outgoing but if i do originate command it works? |
| 05:32.29 | DND | here's the call log: http://pastebin.com/u8xPBPuD |
| 05:37.50 | ChannelZ | what is the Originate line you are using? |
| 05:37.57 | Scorp1us | ok, so i cna't seem to figure out how to fix this 404 error |
| 05:41.45 | *** join/#asterisk mKn0wt (~Taisigue@190.181.165.171) |
| 05:58.14 | Grady2000 | any know why my script won't accept: sys.stdout.write("EXEC Dial(SIP/3401,20)") ? and yes 3401 is a valid sip registered extension? |
| 06:06.16 | Grady2000 | this works on extensions.conf but not in my script |
| 06:11.43 | |ance|ott | AGI? |
| 06:13.56 | Grady2000 | yes |
| 06:14.10 | Grady2000 | how do i dial sip from an agi script? |
| 06:14.28 | |ance|ott | in php? |
| 06:14.37 | Grady2000 | python and pyst |
| 06:14.44 | Grady2000 | but sure whats the php command? |
| 06:15.16 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 06:15.31 | |ance|ott | i recommend to you a book asterisk gateway interface |
| 06:15.36 | |ance|ott | had you read it? |
| 06:16.23 | Grady2000 | no where can i get the link? |
| 06:16.46 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:16.46 | |ance|ott | i have it in my computer xD |
| 06:17.01 | |ance|ott | its so difficult to find |
| 06:17.43 | |ance|ott | can i send file here? |
| 06:18.39 | Grady2000 | irc file send yes |
| 06:18.52 | ectospasm | it's called DCC |
| 06:19.12 | ectospasm | although it might be easier to post it to some filesharing service, and provide a link |
| 06:19.25 | ectospasm | Not many folks accept DCC randomly |
| 06:22.13 | |ance|ott | http://phpagi.sourceforge.net/phpagi22/api-docs/ |
| 06:22.25 | |ance|ott | array, exec_dial (string $type, string $identifier, [integer $timeout = NULL], [string $options = NULL], [string $url = NULL]) |
| 06:22.33 | DND | guys im getting response 405 "method not allowed" |
| 06:22.37 | DND | what does it mean? |
| 06:22.59 | ectospasm | DND: means you're sending a SIP message which the receiver doesn't allow |
| 06:23.04 | |ance|ott | NAT |
| 06:23.12 | DND | hmm i allowed nat |
| 06:23.16 | DND | im disabling it now |
| 06:23.23 | ectospasm | No, it's not a NAT issue |
| 06:23.44 | |ance|ott | yes maybe |
| 06:23.46 | *** join/#asterisk Kumbang (~unknown@180.245.137.5) |
| 06:24.13 | ectospasm | In an OPTIONS message, an endpoint will respond with a list of messages that they allow, and I suppose you're not sending a message type in that list. |
| 06:24.26 | ectospasm | ...see SIP debug for details. |
| 06:24.26 | DND | does this affect something like not making outgoing call? |
| 06:24.35 | ectospasm | DND: not necessarily |
| 06:24.48 | DND | its gone now i reverted it back |
| 06:25.03 | DND | now my biggest problem is x-lite cannot make outgoing calls |
| 06:25.18 | DND | but i can make calls using the originate command |
| 06:25.34 | DND | ectospasm, here's the call log: http://pastebin.com/u8xPBPuD |
| 06:27.11 | ectospasm | DND: that doesn't have SIP debug in it |
| 06:28.04 | DND | ectospasm, ok will enable |
| 06:28.27 | ectospasm | actually, SIP is easier to troubleshoot in tcpdump or wireshark |
| 06:28.53 | ectospasm | use tcpdump or wireshark to capture, wireshark to analyze |
| 06:29.36 | DND | ectospasm, should i disable core verbose or enable it? |
| 06:29.52 | ectospasm | enable it |
| 06:29.56 | *** join/#asterisk UnixDev (~UnixDev@unaffiliated/unixdev) |
| 06:30.05 | ectospasm | ...although a packet capture would be easier to look through |
| 06:31.01 | ectospasm | ...although, now that I look at it, this may be a problem with DAHDI, since it never quite connects |
| 06:31.11 | UnixDev | hi, im using asterisk 1.8 svn, but it seems somewhere along the line in 1.8.x sendrpid started behaving differently...not just sending rpid on the initial invite, but continuing to try to reinvite when a call is transferred or sent to another extension.... how can I revert to the old behavior of only sending it once? |
| 06:31.33 | ectospasm | UnixDev: when did you download your SVN version? |
| 06:31.58 | UnixDev | ectospasm: recently...i dont remember what rev we were using before...we did not notice this |
| 06:32.05 | UnixDev | and upgraded hundreds of systems |
| 06:32.16 | UnixDev | now they are all broken |
| 06:32.28 | *** join/#asterisk boazb (~b@bzq-218-195-107.red.bezeqint.net) |
| 06:32.49 | ectospasm | there was a recent revision released for rpid stuff... like last week or so |
| 06:33.17 | ectospasm | ...can't remember exactly what needed to be changed, or the revision that covers it... |
| 06:33.28 | UnixDev | does it have to do with rpid reinvites? |
| 06:33.54 | *** join/#asterisk vinhdizzo (~vinh@pcp038194pcs.islay.reshall.calpoly.edu) |
| 06:34.00 | UnixDev | my issue is that...i just want it once...reinvites are taxing and not necessary |
| 06:34.12 | UnixDev | for rpid update only |
| 06:35.01 | ectospasm | I can't seem to find it |
| 06:35.17 | UnixDev | :( |
| 06:35.52 | DND | ectospasm, you think its the span configuration? |
| 06:38.35 | ectospasm | UnixDev: yeah, I don't see it. Maybe your searching of http://issues.asterisk.org/jira might yield better results. |
| 06:38.48 | ectospasm | DND: I dunno, you said you can run the originate command that goes through that DAHDI trunk? |
| 06:39.04 | DND | yes i can do originate |
| 06:40.09 | ectospasm | ...through the DAHDI/g0 trunk? |
| 06:40.13 | DND | ectospasm, one question, what does 300 means in this: Dial("SIP/1600-00000053", "DAHDI/g0/0502374530|300|") in new stack |
| 06:40.41 | ectospasm | DND: see "core show application Dial" in the Asterisk CLI |
| 06:40.49 | ectospasm | I can't remember what that field means offhand |
| 06:40.53 | ectospasm | Probably a timeout |
| 06:40.59 | kaldemar | it's the timeout. |
| 06:41.13 | DND | the command i use is: originate DAHDI/g0/0502222222 extension 1600@from-internal |
| 06:41.49 | ectospasm | different number... |
| 06:41.58 | ectospasm | try the same number |
| 06:42.18 | Haraken | jc319, finally got around to installing openxml. |
| 06:44.16 | wdoekes2 | UnixDev: colp? update_connected... stuff? |
| 06:44.24 | kaldemar | DND: enable debug on the span |
| 06:45.00 | ectospasm | DND: what kind of DAHDI span is this? Digital? Analog? |
| 06:45.05 | DND | now it says Primary D-Channel on span 1 down |
| 06:45.07 | DND | then up |
| 06:45.10 | DND | then down again |
| 06:45.12 | DND | its ISDN |
| 06:45.12 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 06:45.23 | ectospasm | DND: sounds like your D-channel is bouncing |
| 06:45.41 | DND | im doing reboot it only happens today |
| 06:45.51 | ectospasm | what does that mean? |
| 06:46.08 | ectospasm | ...it started today after a reboot? |
| 06:46.19 | DND | no i havent rebooted yet |
| 06:46.26 | Grady2000 | anyone here know the correct syntax for "exec dial SIP"? i have a registered sip that works with extensions.conf but when i use this command in my python agi: "exec dial sip/5400" i get no luck and just a crash... 5400 is a sip extension |
| 06:46.41 | DND | let me try a restart |
| 06:46.59 | Grady2000 | sys.stdout.write("EXEC DIAL SIP/5400") |
| 06:47.02 | ectospasm | not sure if that will fix it DND... |
| 06:47.26 | wdoekes2 | UnixDev: possibly r317670 in 1.8? |
| 06:48.44 | kaldemar | Grady2000: "manager show commands topic exec". enable agi debug and look at CLI when you make a call. |
| 06:53.02 | *** join/#asterisk kwk (~kleine@carbon.gonicus.de) |
| 06:53.30 | kwk | Hi! |
| 06:53.51 | Grady2000 | how do i enable agi debug? |
| 06:55.18 | kaldemar | Grady2000: in the CLI, "agi set debug on". hitting tab will give you options in the CLI. |
| 06:56.03 | kwk | In Asterisk 1.8.4 CLI I would like to execute this command: "odbc show status". But the command cannot be found. What do I need to compile with asterisk to enable ODBC? |
| 06:56.35 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:56.36 | wdoekes2 | kwk: if you don't get res_odbc in make menuselect, you're missing deps, like unixodbc-dev |
| 06:56.40 | schmidts | good morning |
| 06:56.44 | wdoekes2 | morning |
| 06:56.47 | UnixDev | wdoekes2: do you think that is the patch that broke it? its a pretty serious problem |
| 06:57.11 | wdoekes2 | I have no idea.. I just grepped a bit through the source to find relevant changes |
| 06:58.04 | Grady2000 | just turned it on and made a call and no output to cli |
| 06:59.42 | Grady2000 | is anyone here using SIP dialing in an agi? |
| 06:59.42 | kwk | wdoekes2: I have "Resource Modules->res_odbc and res_config_odbc" enabled. |
| 07:00.15 | DND | ectospasm, seems its bouncing |
| 07:00.28 | DND | is this a telco issue or hardware issue? |
| 07:00.36 | ectospasm | could be either |
| 07:00.53 | ectospasm | typically it's telco, unless you have bad hardware |
| 07:01.03 | wdoekes2 | if res_*odbc.so modules are built, then it's a configuration problem. start asterisk with -c and look for warnings about odbc |
| 07:01.11 | kwk | wdoekes2: But the installed executable doesn't link to any ODBC library as "ldd" reveals: ldd `which asterisk` | grep -i odbc |
| 07:01.11 | *** join/#asterisk Tim_Toady (~moi@79.103.30.231) |
| 07:01.28 | wdoekes2 | kwk: do ldd on the module |
| 07:01.57 | kwk | wdoekes2: Where the modules installed? |
| 07:02.47 | wdoekes2 | grep mod /etc/asterisk/asterisk.conf |
| 07:06.13 | kwk | wdoekes2: ok, I think I've found the issue. in /etc/asterisk/modules.conf there's this line "noload => res_odbc.so". |
| 07:06.28 | wdoekes2 | that would indeed cause it not to load ;P |
| 07:06.33 | kwk | :) |
| 07:09.50 | Wiretap7 | Grady2000, tried 'Dial(SIP/5400,20)' ? |
| 07:10.12 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 07:11.56 | *** part/#asterisk Tech_Travis (~Travis@cpe-76-168-191-127.socal.res.rr.com) |
| 07:12.24 | kwk | wdoekes2: wow, when I write "load => res_odbc.so" in /etc/asterisk/modules.conf I get a crash (core dump). |
| 07:12.56 | Grady2000 | will do |
| 07:12.58 | wdoekes2 | that's not good ;) bp the bt |
| 07:13.44 | wdoekes2 | but before that, you may want to ascertain that the modules are for the right version |
| 07:13.57 | wdoekes2 | (e.g. by rm'ing all modules and re-doing make install) |
| 07:16.08 | kwk | wdoekes2: will do that but with "preload => res_odbc.so" and "preload => res_config_odbc.so" I don't get a core dump anymore. But the command isn't there neither |
| 07:16.25 | kwk | I mean the odbc command in CLI |
| 07:16.34 | wdoekes2 | did you see any warnings with -c ? |
| 07:16.45 | wdoekes2 | (or perhaps -c -v ) |
| 07:17.03 | wdoekes2 | (e.g. about missing or broken config files) |
| 07:19.08 | *** join/#asterisk irroot (~irroot@dsl-185-122-97.dynamic.wa.co.za) |
| 07:20.37 | Grady2000 | nope on that syntax wiretap |
| 07:20.42 | kwk | wdoekes2: I didn't see any warning when opening the CLI with the parameters -c -v but "cat /var/log/asterisk/full | grep -i odbc" reveals this: http://www.pasteall.org/22510 Not looking good |
| 07:22.16 | *** join/#asterisk wwgd (~wwgd@208.79.14.130) |
| 07:24.44 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 07:25.10 | kwk | wdoekes2: the first line says that the symbol "ast_odbc_clear_cache" could not be found in "res_config_odbc.so". I looked into this and found this: http://www.pasteall.org/22511 |
| 07:25.22 | kwk | wdoekes2: I hope this helps to find the problem |
| 07:25.53 | *** join/#asterisk mha_ (~mha@109.161.140.230) |
| 07:26.21 | kwk | wdoekes2: It looks like "ast_odbc_clear_cache" is defined in res_odbc.so and not in res_config_odbc.so, right? |
| 07:26.28 | Wiretap7 | Grady2000, are you terminating with a \n? |
| 07:26.53 | wdoekes2 | yes.. which is like it should be.. res_config_odbc depends on res_odbc |
| 07:27.20 | mha_ | any ftp site to download free asterisk with GUI ? in particular latest version available |
| 07:27.40 | ectospasm | mha_: AsteriskNOW: http://asterisknow.org |
| 07:27.44 | Wiretap7 | mha_, your google fu sucks |
| 07:27.57 | Wiretap7 | ~freepbx |
| 07:27.57 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 07:28.02 | wdoekes2 | kwk: do the rm and the make install before you continue |
| 07:28.12 | kwk | @wdoekes2: yes, but "res_config_odbc.so" is not linking against "res_odbc.so" http://www.pasteall.org/22512 |
| 07:28.18 | irroot | http://tinyurl.com/2rfwr <- best site to get all you want |
| 07:28.23 | kwk | @wdoekes2: ok will do |
| 07:30.01 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:33.07 | kwk | @wdoekes2: I did make install after renaming "/usr/lib/asterisk/modules/" to "modules.bak" Now all the modules are reinstalled but the error written to /var/log/asterisk/full are the same. |
| 07:35.00 | *** join/#asterisk rshah (~sabayonus@pc1.jmtech2-unet.ocn.ne.jp) |
| 07:37.04 | wdoekes2 | kwk: ok.. well it looks like the res_odbc.so doesn't get loaded (properly).. so you would have to look into why that is.. I don't know why you don't get an earlier error |
| 07:37.28 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-uvyowkcfpnuuvlyg) |
| 07:38.08 | wdoekes2 | (as for ldd not showing res_odbc, that's correct too.. see compile-time vs run-time dynamic loading) |
| 07:38.10 | kwk | @wdoekes2: I will wait until later this day when I can install a precompiled asterisk 1.8.4 from which I know that ODBC works fine |
| 07:39.59 | wdoekes2 | 'strace -- asterisk -c 2>&1 | grep res_odbc -C10' might give some clues too, but other than that, I'm out of ideas |
| 07:40.41 | kwk | @wdoekes2: I will try this. Thank you for your help. |
| 07:41.38 | kwk | @wdoekes2: I usually start asterisk with "service asterisk start" that doesn't work with your command, right? |
| 07:42.02 | wdoekes2 | you should service stop asterisk first |
| 07:42.18 | wdoekes2 | -c runs asterisk in the foreground |
| 07:43.35 | kwk | @wdoekes2: this command runs like forever |
| 07:45.06 | rshah | hello, i have installed asterisk in a machine with 2 ips. |
| 07:45.08 | rshah | first is local ip e.g 123.123.123.123 and other is public ip e.g 213.213.213.213. |
| 07:45.13 | rshah | So local client will register to local ip, and external client will register to public ip. |
| 07:45.19 | rshah | is it possible to call external account from local account? |
| 07:45.31 | rshah | or is that bad design? |
| 07:45.56 | rshah | i made asterisk to listen to all ips |
| 07:46.44 | ectospasm | should work with the Dial application |
| 07:48.02 | kwk | @wdoekes2: "strace -- asterisk -c 2>&1 | grep res_odbc -C10" won't write anything. From another console I can login to the CLI already and asterisk is up and running |
| 07:48.16 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 07:48.24 | rshah | i always get restransmitting INVITE in asterisk console. |
| 07:48.49 | ectospasm | need more detail |
| 07:49.16 | wdoekes2 | ok.. and grep odbc without res_ ? |
| 07:50.13 | wdoekes2 | (it will indeed run forever as you're holding the main console) |
| 07:51.05 | kwk | @wdoekes: that did the trick: http://www.pasteall.org/22513 |
| 07:51.58 | kwk | @wdoekes2: that did the trick: http://www.pasteall.org/22513 |
| 07:52.34 | wdoekes2 | well.. the trick says that it's not loading res_odbc.so |
| 07:52.40 | wdoekes2 | and without that, you're missing symbols |
| 07:52.53 | wdoekes2 | the SQLFetch and ast_odbc_cache_thing |
| 07:53.41 | wdoekes2 | go back to modules.conf and fix until it open()'s res_odbc |
| 07:54.04 | kwk | @wdoekes2: ok |
| 07:55.13 | *** part/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 07:56.34 | rshah | ectospasm: i'm running asterisk 1.6 on centos 5.5. I tried to install asterisk which need to listen to 2 ips. both external and local client are registered. i saw the status using sip show peers. I tried make a call from local account to external account. |
| 07:56.54 | rshah | In asterisk console Retransmitting #6 (NAT) to 213.213.213.213:5060: INVITE sip:externalaccount@213.213.213.213 SIP/2.0 |
| 07:58.06 | rshah | in my sip.conf, i put nat=yes, externip=213.213.213.213 set localnet to both ips |
| 07:58.51 | kwk | @wdoekes2: yeaha! I've fixedit |
| 07:59.58 | kwk | I've uncommented the "preload => res_odbc.so" and "preload => res_config_odbc.so" but left "noload => res_odbc.so" in it's place. stupid me. Thank you for your help @wdoekes2. |
| 08:00.08 | *** join/#asterisk cneb3000 (~Ben@87.127.15.113) |
| 08:00.25 | wdoekes2 | np :) |
| 08:00.37 | cneb3000 | how do fellow astrisk-teers |
| 08:01.03 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 08:01.05 | cneb3000 | today we play with a SIP call generator :) - http://sourceforge.net/projects/sipp/ |
| 08:02.19 | ectospasm | rshah: shouldn't need NAT unless the phone connecting to the 213... address is behind a NAT |
| 08:03.20 | ectospasm | rshah: "Retransmitting #N..." means that Asterisk can't send a call to the endpoint connected to that interface |
| 08:03.40 | rshah | ectospasm: if both register to external ip, and i removed the nat, everything run perfect |
| 08:03.58 | rshah | or if i make asterisk only listen to external ip |
| 08:05.06 | ectospasm | rshah: I'd need to see a tcpdump packet capture of the nonworking setup. |
| 08:08.04 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:09.21 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:11.51 | *** join/#asterisk gurra (~gurra__@unaffiliated/gurra) |
| 08:14.52 | AdvoWork | you know the localnet setting in asterisk, how can i make sure mines correct? i think its 192.168.0.0/255.255.255.0 but want to make sure thats ok? |
| 08:16.02 | Haraken | who do you guys use for your voip provider? |
| 08:19.25 | rshah | ectospasm: i tried dump on eth1(public ip), but cannot get anything :( sorry. |
| 08:20.04 | rshah | only eth0 give some packages |
| 08:20.10 | kaldemar | AdvoWork: that is a valid setting. you just need to know that is matches your network. |
| 08:21.35 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 08:22.50 | AdvoWork | kaldemar, i think it does, but ive got 2 ways of accessing it, phones are all on 1.* but i can acccess the freepbx gui on either 0... or 1.... ? |
| 08:23.22 | AdvoWork | is it localip of the network or can i specify the ip of the asterisk server? |
| 08:24.12 | AdvoWork | and i also keep getting loads of "Target address 78.46.43.9 is not local, substituting externip" |
| 08:24.24 | *** join/#asterisk coppice (~coppice@m121-202-20-144.smartone-vodafone.com) |
| 08:24.46 | kaldemar | AdvoWork: locanet is meant for local telephony devices. |
| 08:25.11 | kaldemar | AdvoWork: you should get that. |
| 08:27.30 | AdvoWork | yeah so that is correct then, i just cant work out whats going on with this, spent hours and hours trying to debug it, but still nothing |
| 08:27.35 | kwk | bye |
| 08:27.36 | *** join/#asterisk wonderworld (~ww@port-92-201-203-104.dynamic.qsc.de) |
| 08:27.36 | *** part/#asterisk kwk (~kleine@carbon.gonicus.de) |
| 08:28.19 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 08:33.41 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 08:37.45 | cneb3000 | Haraken: UK here. I use Gamma Telecom. |
| 08:38.15 | Haraken | ahh |
| 08:38.31 | ectospasm | rshah: what's the tcpdump command you're using? |
| 08:38.55 | cneb3000 | Quite an established carrier.. only ever had minor problems. They actually give you good support too, which is lacking in a lot of VoIP carriers I'd say |
| 08:41.08 | rshah | ectospasm: i tried to tcpdump -i eth1 |
| 08:42.45 | ectospasm | rshah: try tcpdump -s0 -i eth0 and eth1 -w output.pcap |
| 08:44.58 | rshah | ectospasm: syntax error |
| 08:45.15 | ectospasm | rshah: man tcpdump then |
| 08:48.21 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 08:48.38 | *** join/#asterisk Sheeplet (~multivac@41-135-86-171.dsl.mweb.co.za) |
| 08:49.22 | Sheeplet | hi all |
| 08:49.31 | mandla | Morning |
| 08:50.00 | irroot | morning mandla |
| 08:50.32 | irroot | mandla was in your cousins part of the world this weekend the pilanesburg |
| 08:52.10 | mandla | irroot, is it, how was your weekend though/ |
| 08:52.36 | irroot | was good thx maybe make some progress today |
| 08:54.01 | mandla | Yah, iv made some progress myself, but its not working as expected. il send you something in a minute. |
| 09:02.13 | mandla | irroot, http://pastebin.com/MpdcsLFA |
| 09:02.28 | mandla | irroot, check your private channel |
| 09:05.22 | AdvoWork | im debugging a peer, and it shows: retransmitting #2 (no NAT) to 78.46.43.9:5060: but the next time it will show NAT, then No Nat.. any idea please? ive got nat=yes set |
| 09:08.49 | kaldemar | AdvoWork: is the peer behind a NAT? |
| 09:10.07 | kaldemar | AdvoWork: is your issue still that your asterisk doesn't register to the provider? |
| 09:12.18 | wdoekes2 | AdvoWork: the retransmits aren't switching between nat and no-nat, are they? it's a different transaction, right? is it on request or response? does the Via port correspond to the received port? |
| 09:16.10 | AdvoWork | kaldemar, yeah behind a nat router, and yeah thats the same problem |
| 09:18.19 | AdvoWork | wdoekes2, well, my settings specified are: http://pastebin.com/pkbFXVBC and it looks like they are switching, i just dont know how.. also it all seems to be on port 5060. |
| 09:19.09 | AdvoWork | ive also spotted: From: "Unknown" <sip:Unknown@myexternalip>;tag=as225d70f9 in the OPTIONS request.. when i do the REGISTER request, it does from sip:accountnumber@.... which is weird, i wonder if its the Unknown causing the problem? |
| 09:20.46 | kaldemar | AdvoWork: behind a NAT router, you or the peer? |
| 09:21.29 | kaldemar | AdvoWork: have you started to verify that the problem is not in your network? |
| 09:26.36 | AdvoWork | kaldemar, basically, ive got a line/router for the phones, which is a nat router. everything on asterisk is configured to go through that. We do similar for another company, which works fine.. so i honestly dont know. ive tried to packet captures, and ive asked the company to do some, they say they are receiving nothing, so im struggling what to try next really |
| 09:28.00 | AdvoWork | also, sip show registry shows: sip.mydivert.com:5060 myaccountnnumber 120 Request Sent but ive added defaultexpirey=90 maxexpirey=90 recently as a test, so is it not picking that up either? |
| 09:29.29 | kaldemar | AdvoWork: you can't configure anything in asterisk to go through some point in the network. that is the OS's job. do a traceroute from the asterisk box to see the hops in your network and see if one of them interferes with SIP traffic. |
| 09:30.16 | *** join/#asterisk coppice (~coppice@m121-202-20-144.smartone-vodafone.com) |
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| 09:30.48 | kaldemar | AdvoWork: they are spelled "maxexpiry" and "defaultexpiry", not "-expirey". |
| 09:31.03 | AdvoWork | so on my asterisk, do a traceroute to the sip provider? |
| 09:31.13 | AdvoWork | ahh, ok, thats a copy paste from their support page(worrying).. |
| 09:33.40 | AdvoWork | this is the result of the tracepath http://pastebin.com/kj5Nwe6Z |
| 09:33.59 | kaldemar | AdvoWork: "qualify=3600&yes" <-- & can't be used there, either "yes" or a numeric value; "allow=alaw&ulaw&gsm&slinear&ilbc" <-- & can't be used there, have a single value per an allow line; a peer section should not have externip or localnet settings, |
| 09:34.28 | kaldemar | AdvoWork: from now on, if you pastebin settings, copy them from your sip.conf, not a GUI. |
| 09:36.07 | kaldemar | AdvoWork: now go through the path that you control and make sure that they don't interfere. start with 192.168.0.4. also, have you made sure that your router does not have a SIP ALG (application level gateway) enabled? |
| 09:38.35 | AdvoWork | kaldemar, ahh ok sorry, taken from sip.conf instead: http://pastebin.com/PmBwpV55 |
| 09:40.05 | AdvoWork | and there is no sip alg enabled, just checked |
| 09:40.39 | AdvoWork | how would i know if something is interfearing, what kind of things should i be checking for, doing a tracepath type thing from the asterisk server to 0.4(thats the router by the way) and so on? |
| 09:40.46 | *** join/#asterisk orn (~orn@rtr1.sh23.sip.is) |
| 09:42.18 | jc319 | leifmadsen: thanks, I will have a look |
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| 09:44.41 | jc319 | Haraken: Good news, how do you like it so far? |
| 09:51.13 | wdoekes2 | AdvoWork: externip/localnet are not per-device settings, they should be in [general] |
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| 09:56.44 | AdvoWork | wdoekes2, i have these: sip_general_custom.conf sip_general_additional.conf which one would it be? |
| 09:56.58 | wdoekes2 | ~freepbx |
| 09:56.58 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 10:02.46 | kaldemar | AdvoWork: you already did a tracepath and there seems to be a single hop to the router, so tracepathing to is would be pointless. how would you know what is interfering? examine your network. that's about all one can answer without additional information. see if the packets go out of your network. |
| 10:07.17 | AdvoWork | kaldemar, ive done a tcpdump on the ip of the sip provider, and from what i can see, its going out of that system(asterisk) would i then pretty much do the same on the server that controls dhcp/dns as well to see the same, and so on? |
| 10:07.47 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:09.10 | AdvoWork | wdoekes2, actually, i know you say about freepbx, but ive just checked, sip_nat.conf contains: Externip = externalip localnet = 192.168.0.0/255.255.0.0 anyway, so its already got that |
| 10:16.35 | kaldemar | AdvoWork: it's irrelevant what control dhcp/dns, router is what you should be looking at. |
| 10:17.51 | AdvoWork | kaldemar, just one thing, ive done a tcpdump on the server that controls dhcp/dns etc, and its showing: http://pastebin.com/8Yy1h0Qn do you know if that means its using a different port? ie 192.168.0.204.34105 > 192.168.0.202.53: 38013+ SRV? _sip._udp.sip.mydivert.com. (44) 38013, should that not be 5060 as normal? |
| 10:19.03 | kaldemar | AdvoWork: those are DNS queries, not SIP traffic. |
| 10:20.18 | kaldemar | AdvoWork: do you know what a router is and what DNS and DHCP are? |
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| 10:26.00 | AdvoWork | yeah i do kaldemar |
| 10:27.11 | AdvoWork | im logging things on the router, or trying to, but seeing nothing, so i dont know if its actually getting to that, but the trace showed no problem |
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| 11:27.33 | skrusty | afternoon |
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| 11:56.44 | uk101man | hi, anyone know how to get calendar support working in 1.8 when installed via yum? |
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| 11:57.55 | Chainsaw | uk101man: The very first thing they're going to tell you to do here is install from source. |
| 11:58.03 | Chainsaw | uk101man: So you may want to save yourself some time and do that first. |
| 11:58.28 | leifmadsen | you'd have to make sure there are modules to even support calendaring from yum -- my guess is they don't exist |
| 11:58.36 | leifmadsen | so.... see above |
| 11:58.48 | uk101man | thats what i thought |
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| 12:14.38 | puzzled | hi |
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| 13:12.32 | *** join/#asterisk emsLinux (~dave@190.71.3.255) |
| 13:13.37 | emsLinux | Good morning people, i got a question, for you, whats the best way to record sounds for IVRs in Linux? I got a lot of problems with these files, also, i dont know how to create ulaw/alaw files. |
| 13:14.49 | irroot | emsLinux with a voice artist in a recording studio as a hi quality wav file |
| 13:15.13 | irroot | then use sox to generate .alaw .ulaw .gsm .g722 as you wish |
| 13:15.21 | WIMPy | Speak into a phone. Use any recording software on your PC. And if you're using more that just the one codec, save them as wav with one channel and 8ksps rate. |
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| 13:16.44 | Katty | morning |
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| 13:17.30 | leifmadsen | emsLinux: I just record it with my Polycom and they sound quite good ( just need to be in a quiet room) |
| 13:17.40 | emsLinux | irroot no money for the voice artist yet, so i need to do it in my PC, i'm not very good using sox yet, but im gonna try |
| 13:17.49 | leifmadsen | just use Asterisk :) |
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| 13:18.18 | WIMPy | emsLinux: Use audacity and directly save the right format from there. |
| 13:18.47 | emsLinux | from the phone could be an option, btw, any of you guys had any problem before with Google Voice incoming calls and IVRs? |
| 13:19.59 | leifmadsen | emsLinux: http://ofps.oreilly.com/titles/9780596517342/asterisk-AA.html#Autoattendant_id288013 |
| 13:20.40 | emsLinux | WIMPy I have beed using Audacity, saving the files as wav with 8KHz and 16 Bits, but FreePBX convert them to slin and sounds really crappy, also, theres a lot of times Google Voice can't complete the call and the IVR never answer, the CLI shows error with the format |
| 13:21.28 | WIMPy | That must bee a FreePBX issue. |
| 13:21.33 | WIMPy | ~freepbx |
| 13:21.33 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 13:23.21 | emsLinux | WINPy hahaha, i know i won't find any FreePBX help here, but i hate the FreePBX channel, dont worry, im planning to implemente the IVR manually with .wav files or maybe change the files to uLaw/aLaw |
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| 13:27.38 | gopal_ | any idea about to configure hearbeat server with E1 card |
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| 13:29.05 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 13:32.50 | irroot | morning fair maiden |
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| 13:35.09 | kchehab | hi ppl |
| 13:35.56 | kchehab | i am using sipp to test asterisk performence ,but when i limit the calls to 150 ,only 100 active calls on asterisk found ?why |
| 13:37.29 | tuxx- | fair maiden? http://imgur.com/Unt4j |
| 13:40.15 | kchehab | russellb any ideas ,is there a limitation on asterisk 1.6 ?i should by asterisk business edition ?> |
| 13:40.29 | irroot | ROFLMAO friend says she is preagnant i look at her congragulate her ask her if she knows who the mother is yet ... she had to think it through |
| 13:40.36 | leifmadsen | limitation? |
| 13:40.39 | leifmadsen | there are no limitations |
| 13:40.48 | sxpert | irroot: hahaha |
| 13:44.07 | kchehab | @leifmadsen why i cant reach more than 100 active calls |
| 13:45.12 | kchehab | i already add my sip.conf [general] call-limit=250 |
| 13:48.31 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:49.09 | kchehab | russellb any idea ? |
| 13:49.33 | russellb | why me? |
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| 13:50.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 14:03.41 | Scorpio2007 | im using asterisk 1.8 and have configured sip realtime using mysql backend .. for some reason i get error msg no comptabile codec even though in my sipfriends table i have stated allow = ulaw |
| 14:03.44 | Scorpio2007 | any ideas please? |
| 14:04.01 | Scorpio2007 | the registeration and everything succeeds |
| 14:08.15 | Scorpio2007 | Codecs : 0x0 (nothing) but in my sipfriends table its allow = ulaw |
| 14:08.34 | irroot | maybe ulaw is not what is compatible ?? |
| 14:08.54 | irroot | you connecting to a ulaw line ?? |
| 14:09.04 | Scorpio2007 | the other side states ulaw |
| 14:09.12 | Scorpio2007 | here is a sip debug from the machine that is notw orking |
| 14:09.13 | Scorpio2007 | Capabilities: us - 0x0 (nothing), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) |
| 14:09.28 | *** part/#asterisk emsLinux (~dave@190.71.3.255) |
| 14:10.53 | irroot | "sip show peer" look for codecs |
| 14:11.20 | Scorpio2007 | yah the codec states 0x0 nothing for all the peers |
| 14:11.27 | Scorpio2007 | its almost like its not reading the allow from the database |
| 14:12.49 | *** join/#asterisk epsolon77 (~jstann@70-91-80-37-washingtondc.hfc.comcastbusiness.net) |
| 14:12.56 | epsolon77 | hello all |
| 14:14.34 | epsolon77 | anyone used the Fonality version of asterisk? |
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| 14:21.58 | Scorpio2007 | ok so heres my answer from the previsou question . |
| 14:22.10 | Scorpio2007 | asterisk 1.8 you can no longer state disallow = all and allow = ulaw in the database .. |
| 14:22.42 | Scorpio2007 | when you state disallow = all in sipfriends it does not load any codec no matter what you set the allow to be |
| 14:23.06 | Scorpio2007 | so explicitly set what to be disallowed and allowed in sipfriends for the codecs |
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| 14:32.46 | *** join/#asterisk DanFromUK (DanFromUK@2.27.10.118) |
| 14:34.33 | DanFromUK | Hi, I have a potential client that tells me, he is able to see which DID has been dialled when he receives a call. He uses Eyebeam. Is this possible if he has only one sip registration? |
| 14:36.03 | beek | DanFromUK: See it how, via CallerID? |
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| 14:36.52 | Katty | yawns |
| 14:36.55 | Katty | shugs some tea |
| 14:37.02 | Katty | s/shugs/chugs/ |
| 14:37.16 | beek | offers the sugar bowl to Katty |
| 14:37.34 | Katty | would rather have the caffeine bowl |
| 14:37.50 | Chainsaw | Mmm, yes. Sweet sweet caffeine. Replaces sleep. |
| 14:37.54 | bobb_WU | hello everybody |
| 14:38.23 | Katty | Chainsaw: pff, doesn't replace sleep |
| 14:38.28 | Katty | Chainsaw: just makes awake better |
| 14:38.31 | irroot | caffine is nectar of the gods |
| 14:38.37 | bobb_WU | can anybody help me with a question? i'm trying to figure out how to manually busy-out a dahdi line and bring it back into service without having to restart dahdi (which destroys all active analog connections) |
| 14:39.20 | DanFromUK | beek: maybe, but then they can't dial from the missed call list easily. |
| 14:40.07 | beek | DanFromUK: I think I misread your question... you said "he is able to see..." . Did you mean to say "he wants to be able to see?" |
| 14:40.43 | DanFromUK | he's currently with another company. they have set things up so he can see the DID that has been dialled. |
| 14:40.55 | DanFromUK | he wants to switch to our service, and have the same feature |
| 14:41.03 | beek | Perhaps they did it via the callerid text? That's how I do it at some of my client's. |
| 14:41.04 | WIMPy | bobb_WU: exten => _X.,1,Busy()? |
| 14:41.39 | DanFromUK | beek: so you change the CALLERID(name) = DIALLED_DID ? |
| 14:41.55 | DanFromUK | and the CALLERID(number) still shows the callers id? |
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| 14:42.19 | WIMPy | DanFromUK: You can change it to whatever you want. |
| 14:42.20 | beek | Yes... I have one client in particular who runs multiple companies through the same people. They just wanted to know what inbound caller dialed. |
| 14:42.24 | bobb_WU | WIMPY: will that busy out the circuit? or just send a busy tone? |
| 14:42.49 | bobb_WU | we're looking to reset a phone line from a webpage |
| 14:42.52 | WIMPy | bobb_WU: It will signal a busy. It won;t send tones. |
| 14:43.07 | WIMPy | bobb_WU: What does that mean? |
| 14:43.07 | irroot | FXO lines ?? unplug it drops voltage sees red alarm ?? |
| 14:43.26 | DanFromUK | ok, thanks. i'll give that a go. |
| 14:44.11 | WIMPy | DanFromUK: But you might have to ensure the presentation is set to allowed. |
| 14:44.46 | bobb_WU | we need it so we can remotely disable (and then re-enable) phones |
| 14:44.55 | engrxyz | anyone have some suggestion for a cheapest GSM gateway? |
| 14:45.02 | bobb_WU | so we figured out how to disable the line: "dahdi destroy channel x" |
| 14:45.23 | DanFromUK | ok, thanks |
| 14:45.25 | bobb_WU | but to restore the channel, we have to do: "dahdi restart" which destroys all the lines. is there another way |
| 14:45.27 | bobb_WU | ? |
| 14:45.30 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-4.saucontech.com) |
| 14:45.50 | WIMPy | engrxyz: ebay |
| 14:45.53 | bobb_WU | -not destroys them all- rather it disconnects all analog calls |
| 14:46.13 | engrxyz | WIMPy, : what brand |
| 14:46.38 | WIMPy | engrxyz: I don;t know if they can afford brand names. |
| 14:47.52 | WIMPy | engrxyz: You could also use a BT capable phone or an USB stick. |
| 14:48.26 | engrxyz | WIMPy,: BT what do GSM gateway? |
| 14:48.44 | engrxyz | I am not sure if you understand what is a GSM gateway |
| 14:56.50 | *** join/#asterisk fireman_biff (~biff@65.48.133.103) |
| 14:58.18 | fireman_biff | My PBX doesn't seem to be getting DID information and the CLI has a line which includes " Set("DAHDI/5-1", "__FROM_DID=s") ". Is there anyway to determine definitively whether the problem is with the PBX or the provider? (its analog lines) |
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| 14:59.14 | irroot | that line will do squat fireman_biff |
| 14:59.22 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 14:59.38 | irroot | SET(_FROM_DID=${EXTEN}) |
| 14:59.45 | fireman_biff | irroot: oh, i thought that was showing that it didnt know the DID |
| 15:00.06 | fireman_biff | but in any case, inbound calls can't route based on the DID at the moment |
| 15:00.11 | irroot | or in somcases need to use ${CALLERID(dnid)} |
| 15:00.22 | fireman_biff | i had to set up a catch all to send all calls to the same destination |
| 15:00.32 | irroot | have you got a exten match on inbound |
| 15:00.39 | *** join/#asterisk davlefou (~david@41.225.54.206) |
| 15:00.56 | irroot | the context in dialplan where calls begin |
| 15:01.08 | irroot | exten => _XXXX,1,..... |
| 15:01.10 | fireman_biff | its not a hand-coded dial plan, the system is running elastix |
| 15:01.19 | irroot | if your inbound is 4 digit |
| 15:01.40 | *** join/#asterisk jameswf (~james@unaffiliated/jameswf-home) |
| 15:01.43 | irroot | does not like gui's .... eish |
| 15:02.15 | fireman_biff | I dont know the dialplan well enough to answer your question |
| 15:02.31 | fireman_biff | i know the line i pasted is the first to appear when i call |
| 15:02.37 | fireman_biff | at least with verbose=3 |
| 15:02.45 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 15:03.06 | fireman_biff | well, second after " Starting simple switch on 'DAHDI/5-1' " |
| 15:03.40 | fireman_biff | is there a way to tell if the DID info is being received by the PBX at all? |
| 15:04.31 | irroot | see where the call routes and check if immeadiate is set or not |
| 15:05.19 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 15:05.30 | WIMPy | There's no "simple switch" with immediate. |
| 15:06.00 | fireman_biff | the call only routes somewhere if I have an inbound route set that doesn't check the DID. If all the inbound routes check the DID then the cli says something like, no DID or CLID matches |
| 15:06.05 | irroot | ah true that |
| 15:06.11 | fireman_biff | what do you mean by immediate? |
| 15:06.49 | irroot | nm fireman_biff its a setting in the system that does not apply as you have simple switch on |
| 15:06.50 | WIMPy | I didn't get the beginning. Waht kind of line and what's happening after that "simple switch" thing? |
| 15:07.13 | *** join/#asterisk cneb3000 (~Ben@87.127.15.113) |
| 15:09.08 | fireman_biff | WIMPy: here's the first few lines of logs: http://pastebin.com/h65h5n1q |
| 15:09.14 | fireman_biff | the problem is that I dont seem to be getting DID info |
| 15:09.21 | fireman_biff | so the inbound routes aren't working |
| 15:09.39 | fireman_biff | I had to set up a catch all to send all calls to a single destination as a workaround |
| 15:10.02 | fireman_biff | btw in paste bin I replaced my CLID with XXX... but the CLID showed properly |
| 15:10.16 | fireman_biff | all the lines are analog |
| 15:10.33 | *** join/#asterisk timholum1 (~chatzilla@68-117-120-138.static.eucl.wi.charter.com) |
| 15:10.33 | WIMPy | Analog is evil. |
| 15:10.53 | fireman_biff | WIMPy: I agree with that one |
| 15:11.04 | WIMPy | But I wonder about "simple switch" and s. Whouldn't you get to t if nothing happens? |
| 15:11.22 | fireman_biff | well right now I have the catch all set up |
| 15:11.28 | timholum1 | Is there a way to exicute an agi script without waiting for a result back to go on to the next process? |
| 15:11.30 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:11.30 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:11.37 | fireman_biff | before i was getting a message like, no did or clid match |
| 15:11.47 | fireman_biff | but i cant remember what else appeared |
| 15:11.51 | fireman_biff | there were only like 4 lines |
| 15:11.58 | jameswf | ~freepbx |
| 15:11.58 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 15:11.59 | fireman_biff | and i heard an error message |
| 15:12.31 | *** join/#asterisk moy_ (~moy@173.239.155.74) |
| 15:12.59 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 15:14.21 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 15:14.35 | fireman_biff | so is there a way for asterisk to tell me definitively if the DID info is being seen anywhere or not? |
| 15:17.24 | kchehab | @russellb becuase you helped me before :) |
| 15:17.59 | kchehab | guys any idea: i am using sipp to test asterisk performence ,but when i limit the calls to 150 ,only 100 active calls on asterisk found ?why |
| 15:20.43 | *** join/#asterisk ioioxoo (~aceioio@93-96-168-138.zone4.bethere.co.uk) |
| 15:21.03 | kaldemar | kchehab: is sipp trying to make more calls? |
| 15:22.33 | kaldemar | kchehab: what does asterisk do when it gets a call from sipp? what's your SIP configuration like? what does your dialplan do? what do you see in CLI? |
| 15:22.34 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:24.14 | Haraken | jc319, honestly I can't find much use for openxml. it's better than anything I had on the services button before (nothing) but it's mostly for the ooo ahh affect. I might find some real uses for it if/when i setup a voip for a small business |
| 15:25.12 | Haraken | jc319, if you add a contact to openxml, are you able to choose which line it uses to dial? so far I've only been able to get contacts to dial the primary line |
| 15:25.24 | Haraken | dial from |
| 15:29.25 | jameswf | OT: Do you think the white house uses Asterisk. I know the military does. http://caivn.org/article/2011/06/18/internet-activists-crash-white-house-phone-lines-calling-end-war-drugs |
| 15:32.46 | kchehab | kaldemar yes sipp is generating calls |
| 15:33.39 | kchehab | kaldemar but i configured sipp to send 250 calls and sipp is sending 250 ,asterisk only answer 100 concurent call |
| 15:40.30 | *** join/#asterisk davlefou (~david@41.225.99.237) |
| 15:42.28 | kchehab | f |
| 15:46.57 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:48.38 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 15:49.29 | *** part/#asterisk irroot (~irroot@dsl-185-122-97.dynamic.wa.co.za) |
| 15:50.54 | *** join/#asterisk scalex000 (~chatzilla@186.6.171.203) |
| 15:52.51 | scalex000 | hello |
| 15:52.55 | scalex000 | good morning |
| 15:53.15 | scalex000 | which is the command in astersik 1.6.2 to reload all |
| 15:53.37 | russellb | *CLI> core reload |
| 15:53.41 | russellb | if that's not there, then *CLI> module reload |
| 15:54.00 | scalex000 | ok |
| 15:54.33 | scalex000 | its there a option to pause |
| 15:54.44 | scalex000 | i need to check if I get error |
| 15:56.36 | scalex000 | russellb, this error its important http://pastebin.com/rD7gdcz4 |
| 15:56.56 | scalex000 | russellb, I never setup clialias |
| 15:57.09 | russellb | you can ignore it then |
| 16:01.09 | *** part/#asterisk fireman_biff (~biff@65.48.133.103) |
| 16:10.39 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 16:17.14 | *** join/#asterisk Gugge (~gugge@91.208.16.1) |
| 16:17.48 | *** join/#asterisk drudge` (~drudge@unaffiliated/drudge/x-837452) |
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| 16:39.29 | *** join/#asterisk vfabi (~fabi@host-static-109-185-197-18.moldtelecom.md) |
| 16:48.51 | *** join/#asterisk leroybuckingham (43350083@gateway/web/freenode/ip.67.53.0.131) |
| 16:52.07 | leroybuckingham | hey I'm trying to send a fax with "channel originate" and based on the fax-tx dialplan in the FFA manual, but it seems to be failing as soon as the SendFax app is hit. turning on fax debug doesn't seem to be telling me anything about SendFax. Console output is here http://pastebin.com/p8bh7YZ3 |
| 16:52.36 | leroybuckingham | i know the fax module is installed right, all the commands are there and i can get it to beep in my ear. |
| 16:53.35 | leroybuckingham | Is there anything I can do to get a more verbose error message? Could it have something to do with the way the file is formatted? |
| 16:56.38 | *** join/#asterisk irroot (~irroot@41.52.74.96) |
| 17:02.31 | *** join/#asterisk thehar (thehar@diddlebox.thehar.com) |
| 17:03.27 | leroybuckingham | ah, it was the formatting. nevermind. cheers. |
| 17:04.32 | Katty | wibbles |
| 17:04.43 | *** join/#asterisk hdiogenes (~humberto@201.76.154.125.intranet.digi.com.br) |
| 17:06.15 | *** join/#asterisk JuStIcIa_ (~artur0@190.167.57.187) |
| 17:08.56 | tzafrir_laptop | Anybody here uses chan_capi? |
| 17:09.01 | tzafrir_laptop | With asterisk 1.8? |
| 17:13.50 | *** join/#asterisk Gugge (~gugge@91.208.16.1) |
| 17:15.09 | WIMPy | I didn't dare to try it, yet. |
| 17:17.10 | leroybuckingham | okay now onto my bad noob question... I'll likely be doing something in AJAM instead of the CLI to originate this fax transmission, but how can my external system set the filename? In the context I have the line exten =>s,n,Set(FAXFILE=/tmp/test.fax), but I obviously won't be able to hardcode a filename in there |
| 17:19.40 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 17:20.46 | leifmadsen | Asterisk 1.4.42-rc2 is now available: http://www.asterisk.org/node/51645 |
| 17:22.10 | irroot | liefmadsen it served us well RIP 1.4 :P |
| 17:23.11 | WIMPy | has just upgraded to Linux 3. Now waiting for Asterisk 2. :-) |
| 17:24.58 | irroot | WIMPy will be doing v3 soon myself |
| 17:25.15 | irroot | you cant edit .version make it 2.0 :P |
| 17:26.10 | _Corey_ | 3.0rc3... man I feel old now |
| 17:27.50 | *** join/#asterisk cVsup (~cVsup@189.107.145.55) |
| 17:28.41 | *** join/#asterisk wonderworld (~ww@port-92-201-203-104.dynamic.qsc.de) |
| 17:31.07 | *** join/#asterisk dlublink (~david@76-10-163-98.dsl.teksavvy.com) |
| 17:34.06 | cVsup | somebody can say about sruffel? |
| 17:38.27 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 17:42.48 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:49.27 | fullstop | Meh.. 3.0 == 2.6.4X |
| 17:49.54 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-nvihhhywclgmajgw) |
| 17:50.07 | fullstop | My first kernel, after freebsd, was something early in the 2.0 line. Slackware with the root and boot floppies. |
| 17:52.57 | irroot | fullstop slackware was first distro i used with a 1.x kernel upgrading to 2.0 was pain |
| 17:53.08 | irroot | supporting elf and a.out |
| 17:56.00 | fullstop | irroot: I jumped on shortly after that change. My system had support for both formats. |
| 17:56.12 | leifmadsen | redoing the kernel numbering just because "2.6.x where x has gotten too large" is s stupid reason |
| 17:56.34 | fullstop | To be fair, 2.6.40 is vastly different from, say, 2.6.9. |
| 17:56.45 | irroot | liefmadsen i believe the true reason is the 3decade of linux |
| 17:56.56 | leifmadsen | still a rediculous reason :) |
| 17:57.26 | fish-bulb | 3.0 sounds cooler too |
| 17:57.36 | fullstop | I mean, come on.. Windows is up to version 7 now. And linux, still back at 2? |
| 17:57.48 | irroot | fish-bulb did not do much for windows :P |
| 17:57.55 | fullstop | By this logic, clearly Windows is superior. |
| 17:57.57 | fish-bulb | yeah, there is some major catching up to do |
| 17:58.18 | leifmadsen | releases Asterisk XX |
| 17:58.21 | fish-bulb | fullstop: yep, and OS X still more so |
| 17:58.28 | *** join/#asterisk sourcode (~code@ppp-61-90-7-128.revip.asianet.co.th) |
| 17:58.59 | fullstop | I learned something about OS X the other day.. |
| 17:59.20 | fullstop | 10.6 for a 2010 iMac != 10.6 for a 2011 iMac. |
| 17:59.39 | fullstop | Despite the version number being the same, there is an additional build number... |
| 17:59.45 | irroot | forks asterisk realeases iCall XX |
| 17:59.45 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 18:00.05 | fullstop | and, a disk with 10.6 from an older computer will not boot on a newer mac. |
| 18:00.36 | fullstop | forks iCall and releases iCall XXX, which is vastly more successful in certain markets. |
| 18:02.35 | irroot | and fullstop draw the jobs card and get sent to siberia to hard labor camp |
| 18:05.08 | *** join/#asterisk contrabanda (contraband@188.123.143.40) |
| 18:05.10 | *** join/#asterisk dlublink (~david@76-10-163-98.dsl.teksavvy.com) |
| 18:05.23 | dlublink | Can I read variables from another channel than the current channel in my dialplan ? |
| 18:05.56 | leifmadsen | dlublink: see SHARE() |
| 18:06.02 | leifmadsen | SHARED() rather |
| 18:06.17 | *** part/#asterisk contrabanda (contraband@188.123.143.40) |
| 18:07.20 | dlublink | ok, so the way to read the variables is I have to set all the variables I want shared twice ? Once using SHARED and once using the normal Set function. ? |
| 18:08.03 | dlublink | I'll look through the doc for SHARED |
| 18:08.03 | dlublink | thx |
| 18:13.56 | *** join/#asterisk moodyy (~chatzilla@estrela-adm.nortenet.pt) |
| 18:14.54 | cj | leifmadsen: I couldn't get DUNDi working with the SIP transport, but IAX2 seems to work alright |
| 18:16.23 | cj | I so tired. |
| 18:16.35 | leifmadsen | cj: ok |
| 18:17.08 | cj | it could be a bug in 1.8 rather than the docs. I followed the docs but s/SIP/IAX2/ and things seem to be okay |
| 18:17.45 | leifmadsen | I'm not sure what part is wrong or not working :) |
| 18:18.43 | cj | leifmadsen: it seems to match this: |
| 18:18.44 | cj | 11:15 < cj> http://www.voip-info.org/boards/index.php?t=22805 |
| 18:20.05 | WIMPy | Looks like it recognises the username as extension. |
| 18:20.16 | cj | right |
| 18:23.04 | cj | my next step is to have asterisk not return 'unavailable' for extensions which are registered on a DUNDi peer... I'm guessing this has something to do with the regcontext/regexten settings |
| 18:23.14 | *** join/#asterisk whit_ (~whit@bas3-windsor12-1128741676.dsl.bell.ca) |
| 18:23.53 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 18:24.05 | *** join/#asterisk dlublink (~david@76.10.163.98) |
| 18:24.49 | dlublink | I read the page about SHARED() which refered me to IMPORT(). Import seems to do what I need, it seems to be read only, but I don't need to write variables, just read them. So IMPORT is what I needed |
| 18:25.26 | bobb_WU | how do you show all the active calls on an asterisk instance |
| 18:25.27 | bobb_WU | ? |
| 18:25.42 | whit_ | bobb_WU: core show channels |
| 18:25.46 | WIMPy | core show channels |
| 18:25.58 | bobb_WU | thanks! |
| 18:26.50 | moodyy | hi, all, how can i send user-to-user info (used in isdn) in sip |
| 18:27.01 | moodyy | »? |
| 18:27.11 | moodyy | thanks in advance :) |
| 18:27.26 | WIMPy | moodyy: As a text. Or not. |
| 18:27.34 | moodyy | yes |
| 18:27.37 | moodyy | text |
| 18:27.46 | WIMPy | yes |
| 18:27.50 | WIMPy | :-) |
| 18:28.17 | WIMPy | SendText() |
| 18:28.48 | WIMPy | But do you get the data somewhere at all? |
| 18:28.59 | *** join/#asterisk binbash_ (~peter@a83-161-240-87.adsl.xs4all.nl) |
| 18:30.51 | moodyy | i want to send this info in the initial invite, that info is read by the other party, but the other party has an isdn connection |
| 18:31.46 | WIMPy | SIPAddHeader() |
| 18:32.47 | *** join/#asterisk deadpigeon (~deadpigeo@office.xpressamerica.net) |
| 18:33.20 | moodyy | ok, i think that's it, thanks |
| 18:43.50 | *** join/#asterisk pabelanger (~pabelange@2607:f2c0:a000:166:beae:c5ff:fe3e:b315) |
| 18:43.50 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:49.42 | Katty | AHHHHHHHHHHHHhhhiiiiiieeeeeeeeeeeeeeee /sob |
| 18:52.57 | jaytee | what's wrong? |
| 18:56.10 | scalex000 | russellb, hello |
| 18:56.23 | scalex000 | russellb, are you there? |
| 18:57.29 | Katty | it's monday |
| 18:57.29 | scalex000 | festival can use different voice |
| 18:57.32 | Katty | that's what's wrong :< |
| 18:58.08 | tzanger | mondays aren't bad |
| 18:58.08 | tzanger | they lead to fridays. |
| 18:58.08 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 18:58.09 | jaytee | yeah..... mondays suck |
| 18:58.49 | jaytee | unless it's a 3 day weekend with monday off |
| 19:00.18 | Katty | too much anxiety today |
| 19:00.28 | carrar | Katty's got a case of the moondays |
| 19:00.33 | Katty | but i did get on the scale this morning |
| 19:00.36 | Katty | and i've dropped 10lbs |
| 19:00.37 | Katty | horay! |
| 19:00.41 | carrar | woo hoo! PICS!! |
| 19:00.58 | Katty | i've not had a new one taken since 10lbs ago |
| 19:10.34 | *** join/#asterisk rutski (~rutski@96.56.54.186) |
| 19:10.47 | rutski | after running "make install" on the asterisk source I get this warning: "YOU MUST READ THE SECURITY DOCUMENT" |
| 19:10.59 | rutski | but they don't mention where or what that document is :-p |
| 19:14.28 | serafie | rutski: README-SERIOUSLY.best-practices.txt |
| 19:14.37 | serafie | should be in your source root. |
| 19:15.08 | rutski | nifty |
| 19:15.11 | rutski | nice file name :) |
| 19:15.38 | dr0ck | srsly |
| 19:16.57 | leifmadsen | :) |
| 19:17.02 | leifmadsen | bows |
| 19:17.34 | Katty | YOU |
| 19:17.34 | tzanger | OH |
| 19:17.35 | tzanger | leifmadsen: |
| 19:17.36 | Katty | WHY I OUTTA |
| 19:17.36 | Katty | JUST |
| 19:17.38 | WIMPy | rm -srsly |
| 19:17.38 | Katty | HUG YOU TO BITS |
| 19:17.47 | Katty | hugs leifmadsen to bits. |
| 19:18.33 | tzanger | 1.6.2.9 seems to be an asterisk version I'm stuck using... besides being 1.6, is there anything that I should be ZOMGOHNOES about with this version? |
| 19:19.10 | Katty | make sure you have cookie offersings. |
| 19:19.17 | leifmadsen | tzanger: probably just security releases if it works for oyu :) |
| 19:20.03 | tzanger | not sure if it's gonna work for me yet. people are asking me if they really need to move off of it or if it is okay, given that later versions apparently want later versions of libs that the vendor refuses to update |
| 19:20.46 | *** join/#asterisk irroot (~irroot@197.173.77.10) |
| 19:21.17 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:24.46 | tzanger | leifmadsen: is there a cooler way to browse the changelog than just looking at the textfile? it's kind of difficult to parse out |
| 19:24.49 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 19:28.17 | leifmadsen | not really |
| 19:28.19 | leifmadsen | svn log? |
| 19:28.28 | leifmadsen | basically the same thing |
| 19:28.46 | Chainsaw | tzanger: You could pay leifmadsen to read it to you as a bedtime story. |
| 19:28.53 | Chainsaw | tzanger: I think that would be the coolest way. |
| 19:32.04 | tzanger | I don't like his bedtime stories |
| 19:37.23 | serafie | tzanger: use JIRA's issue navigator to find issues in Asterisk for which status is closed, resolution is fixed, and choose a "fix version" |
| 19:38.01 | serafie | there is a good number of results for 1.8.5 and going forward |
| 19:38.08 | serafie | though past versions will be sparse. |
| 19:39.33 | scalex000 | hello, I need to know how can I use another voice using swift that is not default voice |
| 19:40.33 | _Corey_ | scalex000: "swift -n Allison" where "Allison" is the name of the voice you want |
| 19:41.04 | scalex000 | but I need to use in dialplan its that possible |
| 19:42.13 | Katty | hmm |
| 19:42.27 | Katty | maybe i should record some voice overs for asterisk |
| 19:45.33 | _Corey_ | scalex000: I use it primarily w/AGIs and external to the dialplan... If you post an example of what you have now I will look |
| 19:46.20 | scalex000 | _Corey_, I want to make sure, I can use 2 voices in different option, |
| 19:46.40 | scalex000 | _Corey_, but now I only can use 1 |
| 19:46.43 | _Corey_ | You should be able to use as many as you have licensed |
| 19:47.18 | scalex000 | _Corey_, ok |
| 19:47.47 | scalex000 | _Corey_, but do you know how to change in dial plan, I try to test like they said using xmpp |
| 19:47.50 | scalex000 | but not work |
| 19:51.35 | scalex000 | _Corey_, what do you think? |
| 19:51.58 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:52.52 | _Corey_ | I don't use it that way, so I don't know unless you post more detail |
| 19:53.30 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 19:53.36 | *** join/#asterisk JParr (~JParr@24.244.133.104) |
| 19:53.40 | DrDigital | Wiretap7, you around? |
| 19:53.41 | _Corey_ | ... |
| 19:54.36 | scalex000 | _Corey_, let me tell you what I want, I want to use 2 voices, in sometime I want to use female and sometime male, but Im not sure swift application can do it |
| 19:54.57 | scalex000 | _Corey_, on 1 extension |
| 19:55.24 | _Corey_ | Yeah, the application can support multiple voices. Maybe contact Cepstral for help |
| 19:55.32 | JParr | im trying to prevent chan_ooh323 from marking its calls as data calls in the information element, is there a way to do this? |
| 19:59.26 | *** join/#asterisk b0ot (~Jinxed---@147.177.56.129) |
| 19:59.51 | b0ot | Has anyone ever gotten a Grandstream GXV 6000 to work with CME |
| 20:01.32 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 20:11.02 | cj | would like to try out a Grandstream |
| 20:11.18 | cj | in case Grandstream is listening and in the helping out a new clec mood |
| 20:12.33 | scalex000 | _Corey_ Thank you work with license |
| 20:16.37 | *** join/#asterisk devmikey (~irc@ip-209-215-165-114.browardlibrary.org) |
| 20:17.48 | devmikey | Question: I have a SIP account with a 3rd party provider and unfortunately frequently when I place calls to PSTNs it disconnects me after like 10 seconds. Is there a typical reason why? |
| 20:21.40 | p3nguin | devmikey: Enable sip debug, make a call, wait for the disconnect, copy the full debug output, paste it in the pastebin. |
| 20:22.14 | p3nguin | ~grandstream |
| 20:22.14 | infobot | i guess grandstream is the Yugo of VoIP hardware. Run... Run away now. Though, therealcircut says that they're not that bad. |
| 20:22.20 | p3nguin | ~gs |
| 20:22.20 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 20:22.32 | p3nguin | looks at cj |
| 20:24.54 | *** join/#asterisk jnix_ (~jnix@64-233-209-72.static.nap.wideopenwest.com) |
| 20:25.08 | *** join/#asterisk rightie (~rightie@wsip-24-249-29-9.ri.ri.cox.net) |
| 20:25.28 | devmikey | I don't have the logs I don't think |
| 20:25.38 | devmikey | *3rd party provider* |
| 20:27.51 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 20:28.21 | jnix_ | I'm trying to use an SPA8000 to bring pots lines into an asterisk server. Even though the SPA seems to be registering, and I've got a very basic monkey noise dial plan, the system doesn't seem to pick up. can anyone point me back in the right direction? I'm very new to Asterisk--this is my first install. |
| 20:29.16 | *** join/#asterisk WiretapWork_ (~Wiretap@unaffiliated/wiretap) |
| 20:30.05 | p3nguin | devmikey: You don't have access to your own Asterisk system? |
| 20:30.30 | devmikey | I'm paying a 3rd party for the PSTN access |
| 20:31.10 | p3nguin | devmikey: So you don't have access to your own Asterisk system? |
| 20:31.25 | devmikey | I guess not |
| 20:32.02 | p3nguin | devmikey: The only thing that I can think of is a problem with NAT closing down the connection when it shouldn't. |
| 20:32.17 | devmikey | let me see if i'm using NAT |
| 20:32.54 | devmikey | Yes I am |
| 20:33.40 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 20:33.48 | *** join/#asterisk E-bola (~bola@x1-6-00-13-46-83-e5-04.k888.webspeed.dk) |
| 20:34.45 | p3nguin | devmikey: Are you using more than one phone? |
| 20:34.49 | devmikey | No |
| 20:35.08 | devmikey | Maybe I'll try STUN |
| 20:35.24 | p3nguin | Sounds like it's worth a try. |
| 20:35.44 | jaytee | on the SPA-8000 I had to have nat=yes for the sip accounts for the SPA-8000 |
| 20:35.56 | WIMPy | Can you hear the other end in those 10s? |
| 20:41.15 | *** join/#asterisk nny (~SM@174.107.223.14) |
| 20:41.41 | jnix_ | jaytee: perhaps I'm trying to do this backwards. What I want is for the SPA to pass an incoming call on the pots line to the asterisk system (eventually so that it rings on a registered sip phone) |
| 20:42.31 | jnix_ | currently it reads "registered", but nothing ever answers, and eventually my telco voicemail picks up |
| 20:42.45 | WiretapWork_ | jnix_, sounds like a dialplan issue |
| 20:43.02 | scalex000 | _Corey_, hi its too difficult to install AGI |
| 20:43.05 | WiretapWork_ | well, that depends if the SPA is actually noticing the line ringing |
| 20:44.06 | jnix_ | the status page says "idle" while I know it to be ringing |
| 20:44.24 | jnix_ | so perhaps it isn't noticing the line ringing |
| 20:45.08 | nny | trying to work with sangoma to install a transcoder card, getting Error loading module 'codec_sangoma.so': /usr/lib/asterisk/modules/codec_sangoma.so: undefined symbol: ast_rtp_set_peer -------> WARNING[9901] loader.c: Module 'codec_sangoma.so' could not be loaded. |
| 20:45.31 | p3nguin | install AGI? |
| 20:45.36 | nny | this is 1.8.4.2-1 from digium repo fwiw |
| 20:46.19 | nny | p3nguin: AGI? not being used if you mean the gui thing for asterisk |
| 20:46.45 | nny | p3nguin: er AGI is gateway interface, is it needed? |
| 20:46.48 | p3nguin | <scalex000> _Corey_, hi its too difficult to install AGI <--- curious about this |
| 20:46.59 | nny | p3nguin: oh sorry wrong person, thanks |
| 20:47.18 | scalex000 | p3nguin, yes |
| 20:47.42 | scalex000 | p3nguin, how to install, very difficult or is easier |
| 20:47.44 | jaytee | jnix, the SPA8000 as I recall is an 8 port FXS ATA adapter. it has 8 FXS ports, not FXO ports for POTS lines. This would not work with POTS lines plugged into it. It is only for plugging in analog phones |
| 20:47.54 | p3nguin | That doesn't make any sense to me. AGI isn't something you "install," as far as I know. |
| 20:48.11 | _Corey_ | p3nguin: He was asking about playing different voices with Cepstral |
| 20:48.49 | jnix_ | ug. I thought it could do either. Can someone recommend a straightforward standalone FXO device? |
| 20:48.51 | p3nguin | That doesn't make the statement make any more sense to me than before you said it. |
| 20:49.09 | _Corey_ | Yeah, I think there's a translation issue |
| 20:49.36 | jnix_ | I'm running asterisk on a Mac mini, so I can't use an internal card for the FXO |
| 20:49.52 | _Corey_ | Seems he's trying to do it within the dialplan, confused about the command argument I supplied and I replied that I usually use AGI and that Cepstral may be more help |
| 20:49.55 | p3nguin | The SPA-3102 has one FXO and one FXS. Would that be of any use? |
| 20:49.57 | _Corey_ | now you're up to speed :) |
| 20:50.29 | scalex000 | _corey_, forget this |
| 20:50.47 | scalex000 | _corey_, I registered the voice and work |
| 20:50.55 | scalex000 | XMPP |
| 20:51.01 | jnix_ | perhaps. I have 2 lines, but they are both carried on one of the jacks. can the spa-3102 handle 2 lines on a single fxo port? |
| 20:51.31 | _Corey_ | scalex000: Glad to hear :) |
| 20:52.00 | p3nguin | On a different topic, chan_sccp-b is reported to be "close" to having Asterisk 1.8 support. |
| 20:52.23 | _Corey_ | hmmm |
| 20:52.28 | _Corey_ | Glad to hear it :) |
| 20:52.54 | scalex000 | _corey_, they need to put on website to use more than 1 voice |
| 20:53.11 | p3nguin | If you're interested in testing it, they will provide the development build's source if you request it. |
| 20:53.28 | p3nguin | I'd rather just use it when it's ready. |
| 20:53.47 | _Corey_ | :) me too |
| 20:54.29 | p3nguin | I can test it and say that it doesn't work, but that's not going to be very helpful. |
| 20:54.42 | WiretapWork_ | p3nguin, how close is close? |
| 20:54.46 | scalex000 | hey, how to make a beep |
| 20:54.52 | p3nguin | I wish I could say how close. |
| 20:55.02 | scalex000 | I try to find all command I can use on dialplan |
| 20:55.04 | scalex000 | but :( |
| 20:55.19 | p3nguin | They just mentioned that they've had it working but then it wasn't working. |
| 20:55.40 | WiretapWork_ | p3nguin, the 'research' branch is available in svn, it doesn't compile against 1.8.4 |
| 20:55.54 | WiretapWork_ | I think it compiles against 1.8.0 |
| 20:56.02 | p3nguin | yeah, probably |
| 20:56.04 | WiretapWork_ | they still regard 1.8.x as 'unstable' |
| 20:56.11 | WiretapWork_ | so that's why theyre so slack about it |
| 20:56.11 | p3nguin | I have no reason to use 1.8.0, though. |
| 20:59.57 | WiretapWork_ | I think I need to get in more of a flap, so people don't load me with more and more and more work |
| 21:00.05 | WiretapWork_ | because I look like I have it under control |
| 21:01.34 | p3nguin | Start talking to yourself, mumbling how there's so much work to do and that you'll "never get all of this stuff done." |
| 21:01.45 | nny | does anyone know if the digium repo for asterisk includes the headers? |
| 21:01.47 | nny | er |
| 21:02.16 | nny | rephrase, if asterisk18-core.x86_64 includes the headers needed to compile third party modules* |
| 21:02.31 | WiretapWork_ | nny, the undefined symbol error you're getting pertains to asterisk versions >1.8.2 |
| 21:02.39 | WiretapWork_ | as RTP was re-engineered afaik |
| 21:02.47 | WiretapWork_ | same reason chan_sccp-b won't compile |
| 21:03.13 | nny | WiretapWork_: oh, thanks, so the vendor needs to update to match > 1.8.2? |
| 21:03.32 | nny | (er rather the changes re-engineered since 1.8.2) |
| 21:03.46 | WiretapWork_ | it would seem that way |
| 21:03.56 | nny | WiretapWork_: ok will inform them ,thanks |
| 21:04.55 | *** join/#asterisk ketas- (ketas@ketas6-sixxs.si.pri.ee) |
| 21:06.16 | devmikey | Is it possible for SIP to use SSL? |
| 21:06.48 | WiretapWork_ | fantastic, my head is now spinning |
| 21:07.57 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 21:09.15 | russellb | devmikey: yes. |
| 21:09.16 | *** part/#asterisk jnix_ (~jnix@64-233-209-72.static.nap.wideopenwest.com) |
| 21:09.19 | russellb | ~securecalls |
| 21:10.31 | devmikey | I presume your client would have to support it |
| 21:10.37 | russellb | infobot: securecalls is <reply> For a tutorial on setting up secure calls with Asterisk, see this page on the Asterisk project wiki: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial |
| 21:10.37 | infobot | okay, russellb |
| 21:10.47 | russellb | devmikey: yep. |
| 21:10.56 | devmikey | I doubt mine does |
| 21:12.12 | devmikey | Does xlite support ssl? |
| 21:12.48 | WiretapWork_ | ~google |
| 21:12.49 | infobot | methinks google is http://google.com |
| 21:13.19 | devmikey | If you have to google it, the info is badly organized |
| 21:13.32 | WiretapWork_ | not really |
| 21:13.36 | devmikey | Yes really |
| 21:13.48 | WiretapWork_ | if you have to google it, it means you're not too lazy to find it for yourself |
| 21:13.57 | devmikey | Nope |
| 21:14.06 | WiretapWork_ | instead of expecting everyone to run around and find the answers for you |
| 21:14.19 | russellb | WiretapWork_: you have been counter-trolled. |
| 21:14.26 | russellb | everyone calm down now. |
| 21:14.43 | WiretapWork_ | russellb, telling someone to calm down, is about the least effective way to get them to do so |
| 21:15.00 | russellb | alrighty. |
| 21:15.01 | WiretapWork_ | regardless, despite my stress levels, I'm not actually raging at devmikey |
| 21:17.17 | WiretapWork_ | I wouldn't mind a troll to bitch out right now actually, it might help my stress levels, or maybe one of those scam phonecalls |
| 21:18.16 | E-bola | lol |
| 21:18.19 | E-bola | I love irc :) |
| 21:18.49 | *** join/#asterisk dailylinux (~test@88.87.48.55) |
| 21:19.01 | devmikey | Well I'm raging at my DID provider |
| 21:19.12 | _Corey_ | rages against the machine |
| 21:19.17 | *** join/#asterisk lwizardl (~james@c-68-60-84-225.hsd1.mi.comcast.net) |
| 21:19.20 | lwizardl | hello |
| 21:20.25 | *** join/#asterisk Wiretap_Work (~Wiretap@unaffiliated/wiretap) |
| 21:22.17 | lwizardl | I am planning to use Broadvoice for my VOIP provider and was trying to figure out what type of hardware card would be best used to allow me to connect a standard business/house phone to the system. |
| 21:22.52 | lwizardl | is that a FXO or a FXS? |
| 21:23.07 | russellb | a phone or phone line? |
| 21:23.27 | russellb | phone, FXS |
| 21:23.43 | *** join/#asterisk WiretapWork_ (~Wiretap@unaffiliated/wiretap) |
| 21:23.51 | lwizardl | phone I would think. The only phone service will be the VOIP line |
| 21:24.07 | russellb | k, FXS then. |
| 21:24.12 | lwizardl | k |
| 21:27.21 | lwizardl | and fxo are used for ? |
| 21:27.45 | beek | lwizardl: I remember as FX[central]OFFICE, and FX-STATION (deskset) |
| 21:27.49 | lwizardl | reason why is I am looking at a card that has 2x fxo & 2x fxs |
| 21:28.20 | lwizardl | ok this was further down the auction |
| 21:28.20 | beek | fxo points to the central Office, fxs points toward the station |
| 21:28.21 | lwizardl | FXO modules are used to plug existing analog telephone lines into your phone system and FXS modules are used to plug existing analog telephone. |
| 21:28.24 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:28.29 | Qwell | auction? as in ebay? |
| 21:28.35 | Qwell | It's a crap clone card. Buy elsewhere. |
| 21:28.35 | lwizardl | Qwell, yes |
| 21:28.59 | russellb | shakes his fist at China |
| 21:29.08 | Qwell | (otherwise, we're going to laugh at you when you come back later and try to get it working) |
| 21:29.10 | lwizardl | this was what I was looking at, ebay is always my starting point then i look for other sellers and prefer authorized sellers |
| 21:29.11 | beek | flips them the bird. |
| 21:29.11 | lwizardl | http://cgi.ebay.com/TDM410P-2FXO-2FXS-asterisk-card-tdm400p-a400p-elastix-/120715106707?pt=LH_DefaultDomain_0&hash=item1c1b2e5993 |
| 21:29.49 | beek | lwizardl: Here's a novel idea: buy a Digium card. |
| 21:29.57 | Qwell | NOT from ebay. |
| 21:30.17 | Qwell | also 100% positive feedback != reputable |
| 21:30.33 | beek | lwizardl: http://store.digium.com/products.php?category_id=20 |
| 21:30.59 | lwizardl | yeah already had that page up |
| 21:32.07 | lwizardl | man does broadvoice ever answer their phones lol been waiting for 20 mins already to their music breaking up all the time |
| 21:32.25 | beek | lwizardl: How many phones are you talking about? |
| 21:32.55 | lwizardl | beek, was looking at 2 separate numbers, and maybe 2 phones per line |
| 21:33.23 | p3nguin | You could just get IP phones and forget the card. |
| 21:33.41 | p3nguin | all VoIP all the way |
| 21:34.10 | p3nguin | I think you'll spend more on a card than you'll spend on two IP phones. |
| 21:34.21 | lwizardl | yeah I was using a SIP based service and kinda trying to remove windows from my business life like my home life. and so now looking to get a real service and not just the MJ |
| 21:34.24 | citywok | yea. unless you already have the phones and you really want to use them... |
| 21:34.27 | p3nguin | And the IP phones will have far better VoIP features. |
| 21:34.54 | citywok | pure sip is the way to go, you can do so much more |
| 21:34.56 | lwizardl | citywok, yeah I have some cordless phones I was wanting to use |
| 21:35.17 | citywok | lwizardl: if you want to use your old dect cordless phones or something it may be easier to use an ATA |
| 21:35.18 | Chainsaw | lwizardl: You can get SIP-enabled DECT phones these days. Siemens make them. |
| 21:35.33 | lwizardl | citywok, well being a magicjack user kinda want to move away from anything like that |
| 21:35.34 | citywok | or as Chainsaw suggested, you can get sip-dect phones |
| 21:35.49 | p3nguin | Two phones for around $120 total or a card for like $250... for me it is a no-brainer. |
| 21:35.52 | citywok | lwizardl: we use ATA's for our polycom conference systems that are pre-voip |
| 21:35.52 | WIMPy | A sip dect base that is. |
| 21:36.10 | citywok | if they are good enough for that, i'm pretty sure they're good enough for your dect phone :) |
| 21:36.38 | Chainsaw | citywok: Ah yeah, they make nice boardroom phones. |
| 21:36.40 | p3nguin | Do you really expect to get better service and/or quality from a card as opposed to a couple ATAs? |
| 21:36.44 | Chainsaw | citywok: Mine's behind a Patton 4118. |
| 21:37.12 | citywok | Chainsaw: we've got a polycom 4 channel sound processor with all 4 mic's, the thing was like 6grand or something. |
| 21:37.29 | p3nguin | If you've got the money to blow on a card, then you've got the money to spend on nice IP phones and you can afford to throw out the old cordless phones. |
| 21:37.37 | citywok | a year later we put in our asterisk solution. whoops. it would have been nicer to get the SIP version :P |
| 21:37.56 | citywok | lwizardl: p3nguin speaks good advice. a new set of sip dect phones is cheaper than the card you are trying to buy to salvage them. |
| 21:38.42 | WIMPy | Guys, a dect phone is a dect phone. None of them know anything about sip. You want a sip dect _base_. |
| 21:39.28 | p3nguin | I have two modes of thought: how much does it cost, and how fantastic is the product? |
| 21:39.46 | p3nguin | Something that costs a bunch of money and provides no features is not something I would buy. |
| 21:40.52 | citywok | WIMPy: sorry, i keep mis-speaking. s/dectphone/dectbasestation/ |
| 21:41.29 | p3nguin | You could get a couple Polycom phones and get virtually all the necessary (and usually desired) features that a good IP phone should offer and that no analog phone could pretend to offer. |
| 21:41.50 | WIMPy | citywok: you're not the only one. But it's worth noting that you can use any dect phone (except the really early models) with a sip base. |
| 21:42.17 | Chainsaw | p3nguin: The SoundPoint IP670 is downright sexy, yes. |
| 21:42.21 | WIMPy | Especially if you already got the phones. |
| 21:42.23 | p3nguin | Or the new like of Cisco SIP phones, if that's the sort of thing you prefer. I don't have any experience with the new 500 series phones, so I don't know what they do. |
| 21:42.42 | p3nguin | s/like/line/ |
| 21:42.50 | citywok | for polycom I only have an IP650 and it's pretty nice. annoying to configure automatically, the aastra's are much easier for that. |
| 21:43.15 | citywok | but the polycom web browser is so much nicer using xhtml rather than a custom xml format. |
| 21:43.48 | Chainsaw | p3nguin: I have had the 7960. As far Cisco phones go... I'm cured. |
| 21:43.55 | p3nguin | Don't most people have a computer on their desk with their phone? I don't see the need for a browser on the phone when there's a computer right there. |
| 21:44.05 | p3nguin | chainsaw: You don't like the legacy phones? |
| 21:44.16 | Chainsaw | p3nguin: Provisioning over TFTP, UDP-only SIP... |
| 21:44.18 | citywok | p3nguin: i use it for provisioning, voicemail app, company direcotry |
| 21:44.39 | WIMPy | p3nguin: Don't those computers all have sound I/O or USB? So why do you need a phone in the first place? |
| 21:44.54 | Chainsaw | p3nguin: Not to mention having to telnet in to virtually push buttons for some features. The SIP firmware is just hateful. |
| 21:44.54 | Nugget | telnet is eeeeeeevil! |
| 21:44.55 | citywok | plug in a brand new phone at my company and it presents you a list of available extensions. select one and the phone reboots and instantly becomes that extension. |
| 21:45.18 | p3nguin | I currently use 7960, 7940, and 7912 with Asterisk. I'm not dissatisfied. |
| 21:45.43 | Chainsaw | p3nguin: Unless you want to do something like automatic provisioning. |
| 21:45.44 | p3nguin | I used to use SIP, but now I use SCCP on them. |
| 21:46.02 | Chainsaw | p3nguin: My ISDN & analog gateways speak SIP. |
| 21:46.05 | Chainsaw | p3nguin: Can't do that. |
| 21:46.26 | Chainsaw | p3nguin: (Well unless Asterisk ever becomes stable enough to always have it in the media path as a translator) |
| 21:46.33 | p3nguin | I used SIP until I found out that I get a much better feature set with SCCP. |
| 21:46.46 | Chainsaw | p3nguin: That hardware was constructed for SCCP, yes. And it shows. |
| 21:46.55 | p3nguin | yep |
| 21:47.09 | p3nguin | But SIP wasn't _that_ bad on the phones. |
| 21:47.17 | p3nguin | I used SIP for quite some time, actually. |
| 21:47.25 | Chainsaw | p3nguin: 3.12 broke caller ID. |
| 21:47.31 | p3nguin | 8.12 |
| 21:47.36 | p3nguin | I used 8.11 |
| 21:47.36 | Chainsaw | p3nguin: As you wish. The .12 |
| 21:47.44 | Chainsaw | p3nguin: Every release had a new surprise. |
| 21:48.02 | beek | russellb: Tell your sales guys that there's a broken link on http://www.switchvox.com/catalog/bundles.php. The broken link is: http://www.digium.com/en/products/switchvox/features.php |
| 21:48.19 | p3nguin | I don't recall what version I started out using, but once I got 8.11 I didn't change it again. |
| 21:48.30 | WIMPy | Well, if you buy a CUCM, your problems will magically vanish. |
| 21:48.39 | Chainsaw | p3nguin: That's the one that is least broken, correct. |
| 21:48.41 | citywok | but then you wouldn't be in #asterisk |
| 21:48.45 | WIMPy | Or at least that's waht they'll tell you :-) |
| 21:49.39 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 21:50.42 | *** join/#asterisk moy_ (~moy@173.239.155.74) |
| 21:52.26 | *** join/#asterisk jeffspeff (~jeffspeff@173-11-144-149-houston.txt.hfc.comcastbusiness.net) |
| 21:54.31 | jeffspeff | i'm having an issue trying to compile my asterisk build. i've downloaded the latest source, i ran configure and make menuselect, but it errors after a bit of ./make saying "../res/res_adsi.o:/asterisk-1.8.4.2/res/res_adsi.c:362: first defined here collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make: *** [main] Error 2" |
| 21:55.42 | WiretapWork_ | jeffspeff, that is not sufficient data, pastebin the entire falllover |
| 21:55.44 | WiretapWork_ | ~pb |
| 21:55.44 | 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.us , 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. |
| 21:56.32 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 21:56.52 | *** join/#asterisk jks (jks@193.189.93.254) |
| 21:57.01 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 21:57.14 | jeffspeff | WiretapWork_, http://pastebin.com/WsDU4Dfq |
| 21:57.19 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 21:57.23 | jeffspeff | that's as much as my putty screen will show |
| 21:57.43 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 21:57.56 | citywok | jeffspeff: if you click the top left icon, click change settings, and you can change the putty scrollback from 200 to say 2000 lines. or enable putty logging to file. |
| 21:58.15 | WiretapWork_ | jeffspeff, you have fucked something up spectacularly, just sayin |
| 21:58.26 | Qwell | What he said. ^ |
| 21:58.29 | jeffspeff | lol, delete the directory and try again? |
| 21:58.35 | WiretapWork_ | yep |
| 21:58.40 | WiretapWork_ | untar a fresh copy of the tar.gz |
| 21:58.55 | WiretapWork_ | ./configure && make menuselect, then make |
| 22:00.20 | jeffspeff | ok, thanks for the confirmation, i'll try again later. |
| 22:03.50 | scalex000 | hello |
| 22:03.56 | *** join/#asterisk lwizardl (~james@c-68-60-84-225.hsd1.mi.comcast.net) |
| 22:04.56 | scalex000 | I try to create a IVR, how can I get the number user dial |
| 22:05.15 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 22:06.02 | WIMPy | scalex000: Read() or use extensiond with Background() or WaitExten(). |
| 22:06.15 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
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| 22:30.11 | lwizardl | for a single fxo card like the 1TDM410PELF using it for 2 phone lines would I need a very strong system to work as a pbx. or would just about any fairly recent machine handle it great |
| 22:31.40 | Qwell | a pentium 2 would handle that many calls just fine |
| 22:32.35 | lwizardl | thats what I thought. so just about any machine you could buy today would work great. so I could in theory just go to bestbuy and get a generic emachines and be fine |
| 22:33.11 | WIMPy | Just make sure it has a slot for the card(s) you want to put in. |
| 22:33.35 | lwizardl | yeah as long as it has either pci or pcie 1/8x |
| 22:33.44 | lwizardl | it should be fine |
| 22:39.11 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 22:43.28 | jc319 | Finally found a way to override N. American ITSP's enforced +1 prefix |
| 22:43.46 | *** join/#asterisk tamiel (~tamiel@ip-28.net-81-220-88.toulouse.rev.numericable.fr) |
| 22:51.06 | JParr | ok, this is odd, default install of 1.6 on a plain vanilla ubuntu install, install the chan_h323 module, start asterisk, and it hangs at 99% cpu |
| 22:51.12 | JParr | noload the h323 module, and its happy |
| 22:55.56 | *** join/#asterisk skirmisha (~vk@95-42-47-30.btc-net.bg) |
| 22:56.06 | skirmisha | hi |
| 22:56.30 | skirmisha | guys how can i make multiply match in gotoif function |
| 22:56.35 | WiretapWork_ | JParr, ensure you are compiling it correctly, prepackaged installs do not work properly |
| 22:56.46 | WiretapWork_ | skirmisha, your question is too vague to answer |
| 22:56.52 | skirmisha | can i use ${a} = "a|b|c" |
| 22:57.42 | skirmisha | can i have that in one condition |
| 22:58.07 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 22:58.34 | WiretapWork_ | please wait for someone to respond who knows the answer to your question |
| 22:58.56 | skirmisha | i hope so |
| 22:59.08 | skirmisha | i tested it, but looks like not working as i want it |
| 23:01.11 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 23:02.35 | skirmisha | come on guys |
| 23:02.46 | skirmisha | can someone confirm it |
| 23:03.39 | WIMPy | skirmisha: That won't work. But there is a regex function that might be useful. |
| 23:06.18 | skirmisha | WIMPy, i understand. Any idea how to use it with regexp + gotoif |
| 23:08.07 | *** join/#asterisk Scorp1us (~as@pool-72-81-130-240.bltmmd.fios.verizon.net) |
| 23:08.25 | WIMPy | No, I'm not even sure it works. Haven't looked into that. |
| 23:08.37 | Scorp1us | is it possible to run a asterisk machine and a nother VIOP device behind a firewall if both have their own DID? |
| 23:08.55 | WiretapWork_ | Scorp1us, no |
| 23:09.03 | WiretapWork_ | if they both have their own IPs, sure |
| 23:09.15 | WiretapWork_ | if they share one, no, not really |
| 23:09.38 | Scorp1us | hrm ok. looks like I'm getting another IP |
| 23:09.48 | WiretapWork_ | why do you even need to though? |
| 23:09.54 | WIMPy | As long as they don't use the same port, that should work. |
| 23:10.06 | WiretapWork_ | register the 'other device' to asterisk, and let asterisk handle the routing |
| 23:10.18 | JParr | yeah, you should be fine with two devices behind one nat firewall |
| 23:10.25 | Scorp1us | well I have a Linksys viop adapter for my home phone, and I'm starting a project with askterisk |
| 23:10.27 | JParr | assuming both aren't trying to be servers on the same port |
| 23:10.47 | Scorp1us | I've got a number and the registration working ok, but everything goes to my linksys |
| 23:10.48 | WiretapWork_ | WIMPy, I tend to deal with non-registering trunks mostly, i.e. static peers |
| 23:11.01 | WiretapWork_ | WIMPy, so my views are somewhat tainted toward the enterprise :P |
| 23:11.21 | WiretapWork_ | Scorp1us, your explaining skills seem to be impaired right now |
| 23:11.28 | WIMPy | WiretapWork_: How does that matter? |
| 23:11.56 | WiretapWork_ | WIMPy, they tend not to support anything but port 5060 |
| 23:12.32 | WIMPy | That's a (senseless) limitation of your suppliers then. |
| 23:12.38 | *** join/#asterisk mindCrime (~chatzilla@nat/redhat/x-ofezhdkjygtmdkrd) |
| 23:12.44 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 23:13.09 | WiretapWork_ | of course, it doesn't neccessarily make it a good idea :P |
| 23:13.49 | WIMPy | SIP is not a good idea, to start with, so what? |
| 23:14.02 | WiretapWork_ | really, why do you feel that? |
| 23:14.17 | Scorp1us | my explaining skills are impared? |
| 23:14.24 | WIMPy | It's a mess. |
| 23:14.45 | WiretapWork_ | you said all calls go to your linksys, are you implying that your asterisk is up but call routing is messed, or that you want this to happen ,etc, etc ,etc ,etc |
| 23:15.59 | Scorp1us | well the asterisk system registered itself it seems because when I call the number I put into the registration for it, it rings back to me, but rather than asterisk picking it up, the linksys rinks |
| 23:16.02 | Scorp1us | rings* |
| 23:16.20 | WiretapWork_ | because you're not doing nat properly |
| 23:16.29 | WiretapWork_ | and you have them both registering with the same port |
| 23:16.33 | WiretapWork_ | which the linksys has control over |
| 23:16.43 | Scorp1us | yeah, I gotta figure out how to configure that. |
| 23:19.37 | *** join/#asterisk jc319 (~jc318@78-86-169-203.dsl.cnl.uk.net) |
| 23:23.49 | timholum1 | for some reason Asterisk is telling me that "sip show peers" no such command? |
| 23:24.11 | jc319 | what version do you have? |
| 23:24.15 | WiretapWork_ | make sure yo uhave the module logo |
| 23:24.17 | WiretapWork_ | loaded* |
| 23:24.19 | WiretapWork_ | wtf logo? |
| 23:24.22 | timholum1 | I know I did it earlyer today? could it be due to a misconfigured sip.conf? ( asterisk 1.8.3 |
| 23:24.23 | WiretapWork_ | <-- braindead |
| 23:24.39 | WiretapWork_ | core show modules |
| 23:24.40 | WIMPy | timholum1: Possible, yes |
| 23:24.42 | WiretapWork_ | make sure sip is in there |
| 23:25.02 | p3nguin | module show like sip |
| 23:26.09 | timholum1 | it tels me chan_sip.so loaded |
| 23:26.12 | timholum1 | use count 0 |
| 23:26.49 | timholum1 | reload chan_sip.so gives me an error |
| 23:27.15 | p3nguin | module reload chan_sip |
| 23:27.24 | jc319 | how about core stop now and re starting asterisk |
| 23:27.55 | jc319 | p3nguin: Thanks for all the help, everything works now and about 2 hours ago I've fixed the +1 issue |
| 23:28.02 | jc319 | (prefix problem due to carrier) |
| 23:28.05 | p3nguin | I need to provide wireless service to a network device. Would anyone recommend a simple wireless bridge instead of a regular access point? |
| 23:28.44 | p3nguin | jc319: I'm glad you're up and going. What was the +1 issue? |
| 23:29.51 | jc319 | Whatever I put into caller id field did not matter, ITSP enforced a +1 prefix, so it would end up +1<my_callerID_variable> on the other party's screen which practically limited scope of this voip to family and friends |
| 23:29.52 | WiretapWork_ | p3nguin, I like to use the old reliable WRT54GL |
| 23:30.07 | p3nguin | I'm trying to go as cheap as possible. |
| 23:30.12 | WiretapWork_ | jc319, that's positively retarded |
| 23:30.25 | WiretapWork_ | p3nguin, they're not pricy |
| 23:30.36 | *** join/#asterisk saxa (~sasa@189.26.255.43) |
| 23:30.41 | scalex000 | how to run a ODBC function |
| 23:30.42 | WiretapWork_ | and damnfuckit, I'm going to get lunch before I beat someone up :P |
| 23:30.43 | scalex000 | ? |
| 23:30.52 | p3nguin | I just need a basic 802.11 b/g with whatever security implementation that's available. |
| 23:30.55 | scalex000 | I need to check if I write sql syntax |
| 23:31.05 | jc319 | WiretapWork_: I guess it does not matter if your number starts with +1 anyway, however it is unusable in the UK... |
| 23:31.09 | scalex000 | correct |
| 23:32.34 | p3nguin | We in North America have 10-digit phone numbers, and +1 isn't part of it. |
| 23:32.56 | p3nguin | But we often need to dial the 1 before the 10-digit number to make a call. |
| 23:32.59 | jc319 | Yeah but you can call the same number with +1 <10dig> right? |
| 23:33.05 | WIMPy | It is. You just don't have to dial it, if you're already there. |
| 23:33.36 | jc319 | Yeah because it is in your scope (I'm sure 'scope' is not the right word but gives an idea). or context perhaps |
| 23:33.39 | timholum1 | Ok I think I got it fixed, at least its loaded anyway :) |
| 23:33.45 | p3nguin | 1 isn't part of the phone number any more than 011 is part of jc319's phone number. |
| 23:33.52 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 23:34.20 | WIMPy | Off course 1 is part of north american phone numbers. |
| 23:34.20 | jc319 | it's nation/area/town whatever code, so it is part of the number if you are outside that context |
| 23:35.08 | WIMPy | It's always a coutry code, optionally an area code, the subscriber number and optionally an extension. |
| 23:35.24 | jc319 | For example say my mobile is 9940 8977 in my mobile operator's network. If you're in the same mobile operator, it is 9940 8977 for you, if you've another mobile operator it is 075 9940 8977 (same nation/area) and 44 75 9940 8977 in another country etc. |
| 23:35.58 | p3nguin | But I would have to dial 011 44 75 9940 8977. That doesn't make 011 part of your phone number. |
| 23:36.00 | jc319 | But as WIMPy said full number is always there, some people never dial so they don't know but it's still there.. |
| 23:36.03 | WIMPy | Exactely. |
| 23:36.10 | jc319 | 011 is specific to you |
| 23:36.25 | WIMPy | no. 011 is your international access. |
| 23:36.28 | p3nguin | Me as in the entire North America, I guess. |
| 23:36.29 | jc319 | 44 is global (ie it is standard for any network except mine) |
| 23:36.44 | jc319 | yes it is a big context indeed, still not as big as the world. |
| 23:36.51 | lwizardl | is there a good source for addons to asterisk addons, and sounds |
| 23:36.58 | WIMPy | :-) |
| 23:36.59 | paulc | jc319: YOu can't dial 1+10 for a local call though.. In the UK, I can dial 888222 as my local number, or 01534 888222 and the call will complete. In North America 604 257 5757 will work, but 1-604 257 5757 won't |
| 23:37.37 | WIMPy | 011-1-.......... might work. |
| 23:37.40 | paulc | (which has always irked me - why not allow full national dialing all the time? "Because the customer won't know if it's a toll call or not" apparently) |
| 23:37.56 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 23:38.00 | paulc | WIMPy: you reckon? Dial out then in again, internationally? *grins* lemme see here.. |
| 23:38.20 | paulc | nope.. "The call you made cannot be completed as dialed..." |
| 23:38.33 | WIMPy | It might not even be a national thing, but operator specific |
| 23:38.45 | jc319 | In a country it may be different than standard but with my vast experience (of three countries that I call frequently) my theory works. Need more input to see if it is global though. I always assumed this was a global rule or something. |
| 23:38.48 | paulc | ooh - interestnigly, I jsut dialed 1604 2575757 and it worked.. never used to.. |
| 23:39.05 | WIMPy | But you sure can do it with a mobile, even an US one and dial +1.......... |
| 23:39.21 | paulc | jc319: I think most civilised countries allow dialing of full national numbers when it's a local call. |
| 23:39.39 | paulc | WIMPy: Ah yes - mobile networks tend to be a bit more "consistent" |
| 23:39.42 | WIMPy | Yes |
| 23:39.43 | jc319 | so it is always [00-or your international dialing code-set by your operator-whateveritis] + [country code] + [regional code] + [local number] |
| 23:40.13 | WIMPy | Not all countries have area codes. |
| 23:40.25 | WIMPy | It's always a coutry code, optionally an area code, the subscriber number and optionally an extension. |
| 23:40.44 | jc319 | So for a London number >> (00) + 44 + 20 + 84441111 |
| 23:41.58 | WIMPy | A closed number plan like NANP doesn't allow you to manage your extensions yoursef, for example. While e.g. Denmark has all national numbers. |
| 23:42.15 | WIMPy | And it's closed as well. |
| 23:42.58 | jc319 | Can you explain 'manage your extensions' bit? |
| 23:43.20 | WIMPy | On the other side, Germany has an open number plan with area codes of different langt. |
| 23:43.54 | WIMPy | Big cities have short area codes and long subscriber numbers while small cities have long area codes and short subscriber numbers. |
| 23:44.49 | p3nguin | The only way we can manage our extensions is by getting a block of numbers assigned to the organization. But we can't manage them outside of the block. |
| 23:44.50 | WIMPy | In some places you only get a base number that's routed to you and you can just configure your extensions locally. Being totally free in choic if the length of those extensions. |
| 23:45.51 | WIMPy | You set of numbers is clearly defined. |
| 23:45.56 | paulc | WIMPy: like in Germany.. where the main number is xxxxxx-0 but your direct dial extension might be xxxxxx-201 for example, right? |
| 23:45.58 | WIMPy | Your... |
| 23:46.05 | jc319 | For example if the 'normal' # is 604 345 5757, you take the whole '604 257 57xx' and manage the last 2 digits? |
| 23:46.15 | WIMPy | exactely. |
| 23:46.22 | jc319 | It sounds expensive |
| 23:46.38 | jc319 | Do this numbers not run out, like IPv4 |
| 23:46.54 | WIMPy | In germany there's a tradition of wasting 10% of your numbers for the attendant. |
| 23:47.20 | WIMPy | But that's personal choice. |
| 23:47.33 | paulc | jc319: They don't do it that way in North America. If you want 100 numbers, you buy 100 numbers.. and they're all routed to your PRI or whatever.. whereas in Germany you get a "prefix" and what you have after that is more down to you and your routing |
| 23:48.08 | WIMPy | If you need mor numbers, you just make them longer. |
| 23:48.09 | paulc | WIMPy: In Germany, do you have to tell the telco how many digits follow the prefix? Or is it just on timeout? (how do they deal with "0" vs "201" for example - a delay/timeout to complete the call to xxxxxx-0?) |
| 23:48.25 | jc319 | In the same example if you have '604 257 57xx' assigned to you, can you make '604 257 57xxZZZZZ' a valid number? |
| 23:48.51 | WIMPy | However it's no longer guaranteed they can be reached from abroad then. But adding two extra digits works from most countries. |
| 23:49.00 | paulc | no, because all north american numbers are 10 digits.. but if you were talking in Germany, your "assignment" would be "604 25757" and you take whatever comes next |
| 23:49.44 | WIMPy | paulc: No, if you have a match, the call is connected. |
| 23:50.06 | WIMPy | The timeout thing is used in Switzerland, IIRC. |
| 23:50.22 | WIMPy | So for the attendant you just dial the base number without any extension. |
| 23:50.36 | paulc | is that overlap dialing vs en-bloc dialing then? |
| 23:51.01 | WIMPy | There's not neccessarily a relation. |
| 23:51.39 | WIMPy | You will receive it the same way it's sent. |
| 23:52.17 | WIMPy | So if the caller uses en-block dialling, you don't need a timeout, even if you have overlapping extensions. |
| 23:52.33 | WIMPy | If the caller uses overlap dialling, you need the timenout in that case. |
| 23:53.46 | WIMPy | And if you're on ISDN you can extend it much further by using subadresses :-) |
| 23:54.21 | paulc | ..provided the user can set the subaddress :-) |
| 23:54.25 | paulc | The joys of ISDN :) |
| 23:54.47 | WIMPy | There was an "if"... |
| 23:58.44 | WIMPy | What is more common is service based routing. |
| 23:59.29 | jc319 | is anyone using iNum? |
| 23:59.31 | WIMPy | That's what ppl often experience when trying to send faxes via iaxmodem, as they will be sent as speech but default. |
| 23:59.37 | cj | moo |