| 00:02.22 | *** join/#asterisk Techie-Micheal (n=Techie-M@phpbb/leader/Techie-Micheal) |
| 00:02.29 | Techie-Micheal | ~ztdummy |
| 00:02.30 | jbot | [ztdummy] a driver that interacts with zaptel to provide a timing source to Asterisk. On 2.4.x kernals, timing is obtained from a UHCI USB controller. It will not work with OHCI controllers. On 2.6.0 and later kernels, the timing is provided by the kernel, thus no hardware is required at all. |
| 00:02.50 | Techie-Micheal | Blah, that's not any help to me. |
| 00:04.56 | *** join/#asterisk andresmujica (n=andresmu@190.24.227.202) |
| 00:08.34 | drako | manfred*CLI> core show channels |
| 00:08.34 | drako | Channel Location State Application(Data) |
| 00:08.34 | drako | SIP/pstn-spa3k-0880f (None) Up Bridged Call(SIP/pstn-spa3k-08 |
| 00:08.34 | drako | SIP/pstn-spa3k-0880a s@macro-pstn:4 Up Dial(SIP/02127511493@pstn-spa3 |
| 00:08.34 | drako | 2 active channels |
| 00:08.35 | drako | 1 active call |
| 00:08.39 | drako | manfred*CLI> |
| 00:08.44 | drako | how i can take this call, is in a loop and keep the line busy |
| 00:08.54 | drako | how i can take the call or how i can prevent it to happen |
| 00:11.00 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 00:11.15 | [TK]D-Fender | Techie-Micheal, perhaps if you asked a specific question... |
| 00:13.02 | [TK]D-Fender | drako, "soft hangup SIP/pstn-spa3k-0880f" |
| 00:13.38 | drako | [TK]D-Fender, thank you. Now how can I prevent it to happen? |
| 00:17.55 | *** join/#asterisk TheCops (n=henri@got.securebinary.com) |
| 00:22.29 | [TK]D-Fender | drako, Dunno... |
| 00:22.40 | Techie-Micheal | [TK]D-Fender: Sorry, digging around on voip-info.org. Trying to figure out why Asterisk won't recompile with zapata/ztdummy stuff so I can use MeetMe. |
| 00:22.48 | [TK]D-Fender | drako, maybe do all of your dials with an absolute timeout |
| 00:23.00 | [TK]D-Fender | Techie-Micheal, Using 2.6 kernel? |
| 00:23.06 | [TK]D-Fender | Techie-Micheal, and what ver of *? |
| 00:23.11 | Techie-Micheal | Yes. 2.6.18 |
| 00:24.12 | Techie-Micheal | zaptel 1.4.0 and asterisk 1.4.0 (rebuilding with zaptel 1.4.4 and 14.49 now to see if that fixes it) |
| 00:24.38 | [TK]D-Fender | Techie-Micheal, You need to pick ztdummy in menuselect, and compile * afterwards. |
| 00:25.08 | Techie-Micheal | Ah. I've never compiled with menuselect. heh. |
| 00:25.11 | [TK]D-Fender | Techie-Micheal, Once you've done all that and redid your install, do "modprobe zaptel" , "modprobe ztdummy" , "ztcfg -vvvv" |
| 00:25.31 | [TK]D-Fender | Techie-Micheal, then start up * |
| 00:25.38 | TheCops | someone tried oslec echo canceller ? |
| 00:25.46 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 00:25.54 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 00:25.59 | *** join/#asterisk hohum (n=dcorbe@dhcp64-134-231-170.shs.nyc.wayport.net) |
| 00:29.20 | *** join/#asterisk VOiCi (n=o@132-199.sh.cgocable.ca) |
| 00:29.31 | VOiCi | <PROTECTED> |
| 00:29.34 | coppice | oslec works pretty well |
| 00:29.38 | VOiCi | i heard that there was a default time set somewhere, but i dont know where, could somebody help me |
| 00:32.27 | ZeroPing | [TK]D-Fender: I've got Asterisk installing on a box now. Can you help me configure my 7940 to work with it? |
| 00:34.53 | Techie-Micheal | [TK]D-Fender: I don't see ztdummy in menuselect. Am I missing something? |
| 00:35.34 | [TK]D-Fender | ZeroPing, Sorry, no Cisco experience. |
| 00:35.44 | ZeroPing | Ah, know of anyone here that does? |
| 00:35.59 | [TK]D-Fender | VOiCi, "Wait(5)" |
| 00:36.13 | Techie-Micheal | ZeroPing: I have Cisco experience, just not on their VoIP products, sorry. |
| 00:36.43 | ZeroPing | Ah, that's cool. I guess I'll just have to wait for VOiCi or Wait(5) |
| 00:37.59 | [TK]D-Fender | ZeroPing, ... that was my answer to another person... not you |
| 00:38.10 | ZeroPing | Oh, sorry. |
| 00:39.46 | coldsteal | any of u have a Sipura SPA-2100? |
| 00:42.38 | VOiCi | thanks [TK]D-Fender |
| 00:42.45 | VOiCi | <3 |
| 00:43.16 | [TK]D-Fender | coldsteal, |
| 00:43.17 | [TK]D-Fender | ~ask |
| 00:43.17 | jbot | 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. |
| 00:43.33 | *** join/#asterisk rue_mohr (n=rue@h24-207-90-24.cst.dccnet.com) |
| 00:43.51 | rue_mohr | off the top of your head, anyone know the max loop current of a pots circuit? |
| 00:44.16 | *** join/#asterisk saftsack (n=saftsack@pD9E067DC.dip.t-dialin.net) |
| 00:44.31 | *** join/#asterisk MindTheGap (n=iote@201.80.197.219) |
| 00:44.31 | coppice | depends where you are. typically they run at 35mA |
| 00:44.37 | rue_mohr | cool |
| 00:44.59 | coldsteal | ok... does anyone have a Sipura SPA-2100 im trying to figure out how to use it as an extension im kina confused cuz it has a wan and lan port |
| 00:45.00 | rue_mohr | I think I have a 100mA panel meter around |
| 00:45.07 | coldsteal | [TK]D-Fender: is that better? |
| 00:45.22 | [TK]D-Fender | coldsteal, considerably. |
| 00:45.29 | coldsteal | ;) |
| 00:46.32 | [TK]D-Fender | coldsteal, get the pamphlet that lists the DMT codes for enabling the web interface on it and do so, plug the WAN port onto your network and use the DTMF code to have it tell you the IP its using. Then you can go configure it with a web browser |
| 00:47.08 | Techie-Micheal | [TK]D-Fender: Thanks, I got MeetMe working. Not quite sure how, since I didn't see ztdummy in menuselect, but hey, it works. :P |
| 00:48.01 | [TK]D-Fender | Techie-Micheal, You're welcome. |
| 00:48.01 | coldsteal | [TK]D-Fender: do u know where i could get that info |
| 00:48.24 | [TK]D-Fender | coldsteal, should be on the pamphlet your unit came with. If not, co GL it from Linksys's website |
| 00:48.36 | [TK]D-Fender | coldsteal, its the "Quick Install" guide |
| 00:48.47 | coldsteal | [TK]D-Fender: is not a linksys |
| 00:49.34 | [TK]D-Fender | coldsteal, Same this as the Linksys. Linksys bought sipura. when in doubt do to www.sipura.com as well |
| 00:50.12 | [TK]D-Fender | ok, thats it for now... out for a few hours. Later all |
| 00:50.24 | coldsteal | by and thanks |
| 00:51.17 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 00:56.02 | *** join/#asterisk ming_zy1 (n=ming_zym@124.254.54.183) |
| 01:09.39 | ZeroPing | Anyone around that has some Cisco 79xx experience? |
| 01:27.13 | LoneShadow | was there a bug in 1.2.16 for iax2 in resolving IP address for register command ? |
| 01:33.39 | x86 | 1.2.16 is oldskewl |
| 01:49.42 | LoneShadow | x86: dont have space to install 1.4.x on my router :( |
| 01:50.01 | x86 | 1.2.23? |
| 01:50.26 | *** join/#asterisk D-side (n=brian@pool-72-76-175-196.nwrknj.east.verizon.net) |
| 01:50.44 | *** part/#asterisk D-side (n=brian@pool-72-76-175-196.nwrknj.east.verizon.net) |
| 01:52.29 | LoneShadow | hmm, will have to figure out a way to compile that package |
| 01:52.46 | LoneShadow | was hoping I could just compile the iax2 module with the bug fix :D |
| 01:53.20 | tzafrir_home | what router is it? |
| 01:53.50 | *** join/#asterisk IraqiGeek (n=IraqiGee@bl4-154-191.dsl.telepac.pt) |
| 01:53.55 | IraqiGeek | hi all |
| 01:54.26 | IraqiGeek | i'm having some problems with extensions.conf |
| 01:54.32 | LoneShadow | Linksys WRT54G |
| 01:54.53 | IraqiGeek | i have configured two 3 extensions, 101 and 102, and a dummy extension at 110 |
| 01:55.02 | LoneShadow | I should have put the GS version instead, it has 8MB of compact flash |
| 01:55.16 | IraqiGeek | both 101 and 102 can dial 110, which plays back the extension number dialed |
| 01:55.32 | IraqiGeek | however, sometimes 101 and 102 can dial each other, other times they cant |
| 01:56.00 | IraqiGeek | any ideas why that is happening? |
| 01:56.38 | LoneShadow | IraqiGeek: did you try comparing the debugs ? |
| 01:56.52 | LoneShadow | you could do type "asterisk -vvvgcr" |
| 01:56.52 | IraqiGeek | can i get those on a file? |
| 01:56.57 | IraqiGeek | im new to asterisk |
| 01:57.09 | IraqiGeek | and its running on a remote machine, so im on ssh |
| 01:57.17 | LoneShadow | type that on a console window, should put you into asterisk console prompt |
| 01:57.29 | tzafrir_home | or, if asterisk is already running: asterisk -rvvv |
| 01:58.00 | tzafrir_home | this will start a remote CLI session . The three v-s set verbosity to at least 3 |
| 01:58.14 | IraqiGeek | yeah, read about -v |
| 01:58.23 | IraqiGeek | but didnt know i could get there when asterisk is running |
| 01:58.33 | tzafrir_home | With this verbosity level you should be able to get pretty good details of what's going on |
| 01:58.52 | LoneShadow | its a good way to learn asterisk as well :D |
| 01:59.07 | tzafrir_home | If this still doesn't help you understand what's going on, pastebin what you see in there on a failed call |
| 01:59.09 | tzafrir_home | ~pb |
| 01:59.10 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 01:59.16 | IraqiGeek | well, now both extensions r ringing fine |
| 01:59.28 | LoneShadow | save those debugs when its fine :) |
| 01:59.36 | LoneShadow | and try to make it fail and compare the debugs :P |
| 01:59.52 | IraqiGeek | put to 3 v's, as i didnt get anything on 2 v's |
| 02:00.08 | LoneShadow | once you are connected, you could set verbosity I think |
| 02:00.32 | LoneShadow | "set verbose <level>" |
| 02:00.41 | LoneShadow | incase you are not getting sufficient debug info |
| 02:01.00 | IraqiGeek | The 'set verbose' command is deprecated and will be removed in a future release. Please use 'core set verbose' instead. |
| 02:01.26 | LoneShadow | use core set verbose then :D |
| 02:02.11 | LoneShadow | tzafrir_home, x86: is there a way to check what commmits went into iax module since 1.2.16 ? |
| 02:02.21 | LoneShadow | maybe I can patch a fix |
| 02:02.37 | LoneShadow | maybe get a history of changes with checkin comments |
| 02:03.34 | IraqiGeek | @LoneShadow, http://pastebin.com/d5097d607 |
| 02:03.41 | tzafrir_home | something of the sort of: svn diff http://svn.digium.com/asterisk/tags/1.2.16/channels http://svn.digium.com/svn/asterisk/branches/1.2/channels |
| 02:03.46 | *** join/#asterisk themayor (n=jack@dsl081-200-011.nyc2.dsl.speakeasy.net) |
| 02:03.53 | tzafrir_home | chan_iax has more than one file |
| 02:05.36 | tzafrir_home | codecs mis-match between the two? |
| 02:06.12 | IraqiGeek | LoneShadow, seems to be working fine after I made a system restart |
| 02:06.12 | tzafrir_home | shouldn't be. Asterisk would have transcoded in that case |
| 02:08.48 | LoneShadow | Jul 28 14:21:37 WARNING[2756]: chan_iax2.c:5615 iax2_register: Host 'my-server.com' not found at line 201 |
| 02:08.55 | LoneShadow | tzafrir_home: I get this error |
| 02:08.57 | IraqiGeek | ok, the other problem i'm having is that im defining an outbound SIP trunk as follows: |
| 02:09.00 | IraqiGeek | exten => _9X.,1,Playback(${EXTEN:1}) |
| 02:09.00 | IraqiGeek | exten => _9X.,n,Dial(${SIPTrunk}/${EXTEN:1}) |
| 02:09.33 | LoneShadow | so iax2 register works most of the time, fails every now and then |
| 02:09.47 | IraqiGeek | but when I dial anything beginning with 9, i get the "the person u're calling is unavailable.." |
| 02:10.04 | tzafrir_home | instead of: Playback(${EXTEN:1}), use SayDigits(${EXTEN:1}) |
| 02:10.14 | IraqiGeek | and no debug messages in asterisk set at verbose 9 |
| 02:11.04 | tzafrir_home | What is SIPTrunk? "SIP/something"? |
| 02:11.26 | IraqiGeek | yep |
| 02:11.42 | IraqiGeek | and i checked the spelling several times |
| 02:11.53 | IraqiGeek | im not even getting asterisk to say the digits |
| 02:11.57 | IraqiGeek | i restarted asterisk |
| 02:12.04 | tzafrir_home | Are you sure you got to the dial command? |
| 02:12.29 | IraqiGeek | im not getting you? |
| 02:12.40 | tzafrir_home | you should learn to use 'reload' and 'show dialplan (or whatever it's called in 1.4)... |
| 02:13.05 | LoneShadow | type "extensions reload" |
| 02:13.24 | tzafrir_home | Playback(${EXTEN:1} will try to play a file called, say 12345678.wav |
| 02:13.35 | x86 | indeed |
| 02:13.38 | IraqiGeek | changed that to PlayDigits |
| 02:13.51 | IraqiGeek | *SayDigits |
| 02:14.16 | tzafrir_home | So look at the CLI trace. Maybe you don't actually get there |
| 02:14.21 | *** join/#asterisk kavit (n=kavit@ppp167-236-231.static.internode.on.net) |
| 02:14.23 | tzafrir_home | incorrect context or something |
| 02:14.31 | LoneShadow | svn: PROPFIND request failed on '/view/asterisk/branches/1.2/channels' |
| 02:14.33 | IraqiGeek | nothing in the cli |
| 02:14.54 | LoneShadow | some error in http://svn.digium.com/view/asterisk/asterisk/branches/ |
| 02:14.56 | tzafrir_home | LoneShadow, replace "view" with "svn" |
| 02:15.24 | LoneShadow | ok I tried svn diff http://svn.digium.com/asterisk/tags/view/1.2.16/channels http://svn.digium.com/svn/asterisk/branches/1.2/channels |
| 02:15.40 | LoneShadow | even this fails |
| 02:15.46 | tzafrir_home | LoneShadow, replace "view" with "svn" |
| 02:16.02 | LoneShadow | still same error |
| 02:16.10 | LoneShadow | svn: 'http://svn.digium.com/asterisk/tags/svn/1.2.16/channels' isn't in the same repository as 'http://svn.digium.com/svn/asterisk' |
| 02:16.19 | tzafrir_home | http://svn.digium.com/asterisk/tags/svn/1.2.16/channels |
| 02:16.40 | IraqiGeek | at least i should get asterisk to say the digits |
| 02:16.47 | tzafrir_home | oops |
| 02:16.55 | tzafrir_home | http://svn.digium.com/svn/asterisk/tags/1.2.16/channels |
| 02:17.26 | tzafrir_home | IraqiGeek, again: this means you got to the wrong place in the dialplan. |
| 02:17.29 | LoneShadow | cool :D |
| 02:17.34 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:17.38 | tzafrir_home | What do you see in the CLI trace? |
| 02:18.01 | IraqiGeek | i added the SayDigits exten in the internal context, and it says the digits |
| 02:18.15 | IraqiGeek | before I had that in an [outbound] context |
| 02:18.56 | tzafrir_home | What makes you think you really get into the [outbound] context? |
| 02:19.09 | IraqiGeek | apparently i wasnt |
| 02:20.04 | IraqiGeek | well, im reading Asterisk: The future of telephony |
| 02:20.16 | IraqiGeek | and they put the outboud trunks on a separate context |
| 02:20.28 | IraqiGeek | at least thats what i understood |
| 02:25.28 | LoneShadow | tzafrir_home: so these diffs are between 1.2.16 and 1.2.23 ? |
| 02:26.09 | tzafrir_home | right |
| 02:26.24 | tzafrir_home | or actually: a bit beyond 1.2.23 |
| 02:26.27 | LoneShadow | anyway I can get commit comments ? |
| 02:26.33 | LoneShadow | like history of changes ? |
| 02:26.36 | Qwell | svn log |
| 02:26.46 | Qwell | or just read the 1.2.23 ChangeLog |
| 02:28.11 | *** join/#asterisk kimo_sabe (i=nick@zappa.azrackspace.net) |
| 02:28.14 | IraqiGeek | is it "canrenvite" or "canreinvite" in sip.conf? |
| 02:29.13 | IraqiGeek | and what should be the context of the outbound SIP trunk? |
| 02:31.28 | tzafrir_home | the context is what you choose it to be |
| 02:31.53 | IraqiGeek | yeah, but i still dont get its effect |
| 02:33.03 | tzafrir_home | you enter the dialplan in the context that you set in the channel's config |
| 02:33.07 | IraqiGeek | I got only one outbound/inbound SIP trunk |
| 02:33.23 | IraqiGeek | and three internal SIP extensions |
| 02:33.31 | IraqiGeek | the extensions are under [internal] |
| 02:33.39 | tzafrir_home | e.g.: if you set context=internal in sip.conf for some phone, this is where it enters the dialplan |
| 02:33.45 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 02:34.06 | IraqiGeek | oh, ok |
| 02:34.15 | IraqiGeek | but it doesnt affect outbound calls |
| 02:34.52 | IraqiGeek | right? |
| 02:34.53 | tzafrir_home | you can use 'include=>' |
| 02:35.03 | LoneShadow | hmm ast_gethostbyname() failed to resolve, wonder if its not a bug of asterisk |
| 02:36.31 | LoneShadow | if I were to store the IP address of the hostname in say /tmp/server-ip.txt |
| 02:36.47 | LoneShadow | is there anyway I can make asterisk to use that file while registering ? |
| 02:41.03 | LoneShadow | or can I invoke a iax2 register call from cli ? |
| 02:42.25 | *** join/#asterisk bintut (n=bintut@cm229.gamma181.maxonline.com.sg) |
| 02:42.45 | bintut | anyone knows where can i read about "ignorepat"? |
| 02:42.53 | IraqiGeek | now i'm trying to make outbound calls, and this is what I get: http://pastebin.com/d12cad0d4 |
| 02:45.08 | LoneShadow | aha I can have #include iax_custom.conf, wonder if I can have #include /tmp/iax_custom.conf instead |
| 02:45.51 | bintut | anybody knows if it's possible to have additional options or arguments with ignorepat? |
| 02:46.07 | kimo_sabe | bintut: I don't think so, it's a pretty simple option |
| 02:46.14 | tzafrir_home | LoneShadow, you can |
| 02:46.42 | LoneShadow | is it #include "/tmp/iax_custom.conf" or |
| 02:46.52 | LoneShadow | just #include /tmp/iax_custom.conf |
| 02:46.59 | LoneShadow | oh well, I guess I can try one :) |
| 02:47.05 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 02:47.16 | IraqiGeek | anybody? |
| 02:47.18 | LoneShadow | now to write a script to generate my register commands with IP :D |
| 02:48.35 | *** part/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 02:49.21 | tzafrir_home | IraqiGeek, authentication problem? |
| 02:49.26 | tzafrir_home | not really sure |
| 02:49.51 | IraqiGeek | this is what my SIP trunk looks like in sip.conf: http://pastebin.com/d76ebdb15 |
| 02:51.08 | IraqiGeek | rechecked username and pass several times |
| 02:53.52 | IraqiGeek | apparently i need to make more than one reload to make asterisk accept changes... |
| 02:57.49 | IraqiGeek | do i need to do anything other than defining an incoming context in extensions.conf and having my outbound/inbound SIP trunk's context set to incoming? |
| 03:01.53 | IraqiGeek | when I dial my DID from voipstunt, it says ringing then call ended |
| 03:03.21 | IraqiGeek | my incoming context has: |
| 03:03.21 | IraqiGeek | exten => s,1,Answer() |
| 03:03.21 | IraqiGeek | exten => s,n,Playback(enter-ext-of-person) |
| 03:04.29 | TheCops | Someone tried the oslec echo canceller ? |
| 03:07.05 | IraqiGeek | anyone can help? |
| 03:07.53 | ManxPower | IraqiGeek: Do you understand what extension "s" matches? |
| 03:07.59 | IraqiGeek | start? |
| 03:08.12 | ManxPower | Officially. And what digits would it match? |
| 03:08.32 | IraqiGeek | well, that i donno |
| 03:08.50 | ManxPower | I call it (s)tupid. As in, it is only matched when the tech that the call arrives on is too stupid to pass the dialed digits. Most sip providers are not that stupid. |
| 03:09.07 | ManxPower | Doesn't voipstunt have and asterisk exensions.conf examples? |
| 03:09.29 | kimo_sabe | IraqiGeek: what he's getting at is that SIP calls don't come in as "someone is calling (s)", the come in as "someone is calling for XXX number" |
| 03:09.47 | IraqiGeek | my SIP trunk is from voicestick, but im testing it from voipstunt |
| 03:09.54 | kimo_sabe | IraqiGeek: so you probably want something like exten => _.,1,Answer() |
| 03:09.59 | ManxPower | kimo_sabe: SIP calls from most providers, at least |
| 03:10.11 | kimo_sabe | ManxPower: even from phones |
| 03:10.12 | ManxPower | kimo_sabe: you should be banned from the channel for telling someone to use _. |
| 03:10.38 | kimo_sabe | ManxPower: what's the prefered catch-all? |
| 03:10.59 | ManxPower | kimo_sabe: The prefered catch all is to not use one. |
| 03:11.07 | ManxPower | If you MUST use one at least use _X. |
| 03:11.11 | IraqiGeek | so, should I still have an s extension in incoming or remove it? |
| 03:11.32 | bintut | kimo_sabe: sorry for the late reply. anyway, i was thinking after dialing 9 for outbound call, there will be a 1 second pause then dialtone again like how the old traditional pbx does. is it possible? |
| 03:11.39 | IraqiGeek | ManxPower, I read enough to know to never use _. |
| 03:11.40 | ManxPower | Then a call hangs up asterisk runs exten => h if it exists. _. will match "h" and so your dialplan will process the call TWICE |
| 03:12.01 | kimo_sabe | ManxPower: good reason |
| 03:12.01 | ManxPower | IraqiGeek: you need to know what destination your provider is sending you. |
| 03:12.11 | ManxPower | you can usually assume exten => yourdidnumber,1,whatever |
| 03:12.44 | ManxPower | kimo_sabe: imagine exten => _.,1,Dial(Zap/g1/${EXTEN}) |
| 03:12.46 | LoneShadow | tzafrir_home, x86: can I use register=log:passwd@${CUSTOM_VARIABLE} |
| 03:12.53 | ManxPower | The poor person you are calling will be called twice. |
| 03:13.11 | ManxPower | every single time you make a call |
| 03:13.26 | tzafrir_home | LoneShadow, no. those variables are only used in extensions.conf |
| 03:13.38 | ManxPower | although technically, since EXTEN is h at the time asterisk will try to call the telephone number h |
| 03:13.47 | LoneShadow | ok thanks |
| 03:13.59 | ManxPower | unless you hardcoded the phone number instead of using ${EXTEN} |
| 03:14.06 | IraqiGeek | so, this should work: |
| 03:14.07 | IraqiGeek | exten => _X.,1,Answer() |
| 03:14.07 | IraqiGeek | exten => _X.,n,Playback(enter-ext-of-person) |
| 03:14.17 | IraqiGeek | under incoming |
| 03:14.30 | ManxPower | IraqiGeek: I can only answer that if I knew what your provider is sending you. But it is a good place to start. |
| 03:14.46 | IraqiGeek | i donno either |
| 03:14.48 | ManxPower | but that will cause OTHER issues down the road. |
| 03:15.10 | ManxPower | IraqiGeek: Do this. exten => _X.,1,Noop(DIALED NUMBER IS ${EXTEN}) |
| 03:15.15 | ManxPower | then call your DID and watch the console. |
| 03:15.17 | IraqiGeek | yet, still not connecting |
| 03:15.26 | ManxPower | whatever it tells you the dialed number is, is what you put in the exten line. |
| 03:15.36 | ManxPower | IraqiGeek: what does "sip show registry" show? |
| 03:15.54 | ManxPower | and do you have a /something at the end of your register line? |
| 03:16.02 | ManxPower | (in sip.conf) |
| 03:16.17 | IraqiGeek | host, username, refresh, state, reg, and time |
| 03:16.37 | ManxPower | that is all? |
| 03:16.41 | ManxPower | no actual information? |
| 03:16.44 | IraqiGeek | in show registry, yep |
| 03:16.47 | IraqiGeek | no info |
| 03:16.59 | IraqiGeek | what do you mean /something in sip.conf? |
| 03:17.04 | ManxPower | Well, then you are not registered to your provider and your provider doesn't know what IP address to send the call to. |
| 03:17.09 | IraqiGeek | not that i have any line starting with / |
| 03:17.27 | IraqiGeek | but im able to make calls through that sip trunk |
| 03:17.47 | IraqiGeek | or do i have to configure some other file to force asterisk to register that sip trunk? |
| 03:17.53 | ManxPower | Outgoing calls have NOTHING WHATSOEVER to do with incoming calls |
| 03:18.13 | kimo_sabe | IraqiGeek: look for a register => line example in sip.conf |
| 03:18.19 | ManxPower | IraqiGeek: Please step away from the keyboard, and go read.... |
| 03:18.21 | ManxPower | ~thebook |
| 03:18.22 | jbot | thebook is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:18.35 | IraqiGeek | im reading that |
| 03:19.08 | ManxPower | real the part about sip registration |
| 03:19.12 | ManxPower | read that is |
| 03:20.33 | IraqiGeek | and that statement should be in sip.conf? |
| 03:20.38 | IraqiGeek | or in extensions.conf? |
| 03:20.39 | ManxPower | yes. |
| 03:20.47 | ManxPower | register => whatever |
| 03:20.52 | kimo_sabe | IraqiGeek: sip.conf, |
| 03:21.05 | ManxPower | but just putting the line in there does not give you the basic info to be able to do stuff |
| 03:21.13 | IraqiGeek | and i take it, it doesnt matter where on the file that line appears |
| 03:21.14 | ManxPower | You need to UNDERSTAND it. |
| 03:21.22 | ManxPower | IraqiGeek: of course it matters. |
| 03:21.31 | IraqiGeek | ok, where then? |
| 03:21.42 | ManxPower | Why do I get the feeling I keep giving you information that you should be finding in the book? |
| 03:22.01 | IraqiGeek | sorry, but i've been reading through the book for 3 days now |
| 03:22.02 | ManxPower | And unless you are buying me dinner and drinks, I'm not holding your hand for this. |
| 03:22.08 | *** part/#asterisk andresmujica (n=andresmu@190.24.227.202) |
| 03:22.29 | IraqiGeek | max, if u're in portugal, i'd be glad to |
| 03:22.42 | IraqiGeek | *manx |
| 03:22.51 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:22.56 | bkruse_home | ManxPower: ask him for a plain ticket, anyone can always use a vacation |
| 03:23.06 | IraqiGeek | :) |
| 03:23.48 | ManxPower | bkruse_home: Somehow I don't think he is my type. |
| 03:24.34 | bkruse_home | ManxPower: ha :P |
| 03:24.51 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:25.02 | ManxPower | IraqiGeek: look on page 73 of The Book |
| 03:25.09 | ManxPower | The format is similar for IAX as for SIP |
| 03:25.15 | IraqiGeek | 173 |
| 03:25.35 | IraqiGeek | 172 is register statements |
| 03:25.40 | IraqiGeek | :) |
| 03:25.44 | ManxPower | Also look on page 214 |
| 03:26.18 | *** join/#asterisk asnet (n=asn@geekon.net) |
| 03:26.25 | bkruse_home | ManxPower: have you read the new book? |
| 03:26.29 | ManxPower | On MY book page 73 gives you and example |
| 03:26.33 | bkruse_home | a:tfot2? |
| 03:26.38 | ManxPower | bkruse_home: nope. |
| 03:26.48 | ManxPower | is it finished and sent to printing? |
| 03:26.52 | bkruse_home | ManxPower: good read, we will have to teach jbot all the new places! |
| 03:26.59 | bkruse_home | no, im pretty sure it isnt :/ |
| 03:27.03 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 03:27.13 | ManxPower | I'd be happy to give it a read and give feedback |
| 03:27.26 | IraqiGeek | added register in general, now cli is saying its timing out |
| 03:27.50 | ManxPower | That means asterisk is not getting a response. |
| 03:27.57 | Strom_C | bkruse_home: where may I obtain the new book? |
| 03:28.06 | Strom_C | is it floating around in VPN land somewhere? |
| 03:28.07 | ManxPower | firewall problem or wrong hostname to register to is the most common causes |
| 03:28.09 | *** join/#asterisk tuxd00d (n=tuxinato@166.129.21.238) |
| 03:28.26 | bkruse_home | Strom_C: it is in vpn land somewhere, let me check if its still on my lappy |
| 03:28.29 | IraqiGeek | its a vps with a public IP |
| 03:28.35 | ManxPower | I might be able to give some feed back, at least on the technical stuff. |
| 03:29.00 | ManxPower | Since it is 1.4 specific, I don't know how much my 1.2-centric world will help. |
| 03:29.02 | bintut | hello all.. where can i find the descriptions of all the sound files inside the /usr/share/asterisk/sounds ? |
| 03:29.14 | ManxPower | bintut: sounds.txt in the asterisk source code directory |
| 03:29.20 | bkruse_home | ManxPower: partly because TONS of people still use 1.2! |
| 03:29.26 | IraqiGeek | arent the file names enough to understand what each file does? |
| 03:29.28 | ManxPower | bintut: but they are in /var/lib/asterisk/sounds |
| 03:29.50 | tzafrir_home | IraqiGeek, now what does "tt-weasels" mean? |
| 03:30.02 | ManxPower | bintut: of course if you are using an RPM or other package, then ask the person that did the packaging, not us. |
| 03:30.24 | bkruse_home | any vim foo'ers in here, that know how to get rid of my stupid ^M dos returns at the end of my file? |
| 03:30.53 | Strom_C | http://www.stromcarlson.com/misc/strom_still_dangerous.jpg |
| 03:30.56 | ManxPower | bkruse_home: I could tell you for joe, maybe even a sed filter. |
| 03:31.02 | bintut | ManxPower: i'm running debian etch here and the sound files are in /usr/share/asterisk/sounds/ |
| 03:31.11 | Strom_C | bkruse: |
| 03:31.14 | Strom_C | maybe try this |
| 03:31.19 | tzafrir_home | :%s/^M$/ "the ^M here is ctrl-V ctrl-M |
| 03:31.22 | bkruse_home | i bet you can just do it with %s/^M//g |
| 03:31.25 | IraqiGeek | manx, got me on that one |
| 03:31.27 | tzafrir_home | oops: |
| 03:31.29 | bkruse_home | ahh! i couldnt find that char |
| 03:31.29 | bkruse_home | let me try |
| 03:31.32 | tzafrir_home | :%s/^M$// |
| 03:31.40 | Strom_C | :%s/.$//g |
| 03:31.47 | bkruse_home | Strom_C: ++ |
| 03:31.50 | bkruse_home | thank you so much strom! |
| 03:32.05 | bkruse_home | i could not remember how to get type those ^M characters |
| 03:32.09 | Strom_C | :D |
| 03:32.14 | tzafrir_home | Strom_C, but what if there wasn't a CR at the end of line? |
| 03:32.21 | Strom_C | tzafrir: well then you'd be fucked. |
| 03:32.37 | bkruse_home | tzafrir_home: there is, it just got thrown in there somewhere down thel ine of transporting to my laptop |
| 03:33.06 | ManxPower | \r is ^m |
| 03:38.05 | tzafrir_home | well, sysfs is cool, but it'll have to wait for a while... |
| 03:39.11 | bintut | is there a way to ask the first person in the meetme conference to record or not the entire discussions of the said session? |
| 03:40.10 | enioreh | bintut: Monitor may help you |
| 03:40.21 | enioreh | Monitor() followed by Dial() |
| 03:40.23 | bintut | enioreh: lemme check that app |
| 03:40.32 | enioreh | going to sleep |
| 03:42.52 | ManxPower | bintut: you would have to design an IVR of some sort to check the number of users in a meetme room, determine if this caller is the first caller, then give the caller a choice as to record or not. if they do want to record, then run Monitor on their cannel |
| 03:42.55 | ManxPower | channel too |
| 03:43.17 | ManxPower | The answer to most "Can I..." questions is "Yes, your write it in a dialplan" |
| 03:43.24 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
| 03:44.48 | IraqiGeek | can i specify a SIP proxy in a register statement? |
| 03:45.02 | Techie-Micheal | Yes |
| 03:45.10 | ManxPower | IraqiGeek: Usually, yes |
| 03:45.39 | IraqiGeek | looking at the sip.conf page at voip-info.org, and cant find anything bout that there |
| 03:45.43 | IraqiGeek | so, can i ask how? |
| 03:45.52 | ManxPower | IraqiGeek: just put it in. |
| 03:46.04 | IraqiGeek | in where? |
| 03:46.06 | ManxPower | as the hostname of the machine you are registering to |
| 03:46.27 | ManxPower | register => sipuserid:sipassword@remotehosttoregisterto |
| 03:47.15 | IraqiGeek | and where does the sip proxy fit into that? remotehosttoregisterto? |
| 03:47.44 | ManxPower | YES! |
| 03:48.05 | ManxPower | ALL registration does is tell the remote host what IP address is associated with your userid/password. That is all. It does nothing else. |
| 03:48.40 | ManxPower | And the only reason you need to do that is to receive calls from the remote machine if your IP address changes frequently. |
| 03:49.01 | ManxPower | you CAN use register with a static IP, but there is no TECHNICAL reason to do so. |
| 03:49.14 | IraqiGeek | its a static IP, but anyway |
| 03:49.56 | ManxPower | you provider would have to put in your IP address into your account on their system if you didn't want to register to them. Most ITSPs don't want that extra work and do have EVERYONE register. |
| 03:50.07 | IraqiGeek | in x-lite, i have to specify i2telecom.com in domain, and sbc1.voicestick.com as domain proxy |
| 03:50.36 | IraqiGeek | and unless i specify sbc1.voicestick.com, x-lite or any other sip phone i tried wont connect |
| 03:50.41 | ManxPower | that has nothing to do with registration |
| 03:50.51 | ManxPower | that has to do with sending calls to the remote system |
| 03:51.08 | IraqiGeek | i thought that too, but it wont register if i dont do that |
| 03:51.18 | ManxPower | If they use different domains and proxies, then your setup has just gotten a hell of a lot more complicated. |
| 03:51.25 | IraqiGeek | i even opened a support ticket about it, and they said i have to do that?! |
| 03:51.28 | ManxPower | IraqiGeek: look at /path/to/src/asterisk/configs/sip.conf.sample |
| 03:53.35 | bintut | which do you think is the right app to use to record the conversations inside the meetme conference, monitor() or mixmonitor()? |
| 03:54.08 | ManxPower | bintut: I have no opinion on that. do a "show application" for each of them and then you decide. |
| 03:56.10 | IraqiGeek | what is authuser in user[:secret[:authuser]]@host |
| 03:56.22 | bintut | ManxPower: ok. i think i'll try the mixmonitor() first |
| 04:01.59 | IraqiGeek | what the heck, i'll open a support ticket with voicestick about this, and let them give me the answer |
| 04:02.07 | IraqiGeek | tnx all, and have a good night |
| 04:03.33 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 04:04.56 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 04:05.07 | bintut | anyone here have a copy of the sounds.txt file? could you please post it on any available pastebin on the internet? |
| 04:06.57 | kiscokid | where is it usually found? |
| 04:07.42 | bintut | kiscokid: maybe from your source tree of your asterisk? |
| 04:11.58 | bintut | kiscokid: don't bother.. i have to leave already.. |
| 04:12.06 | bintut | thanks all.. gtg for now.. |
| 04:12.14 | kiscokid | ok |
| 04:13.17 | *** join/#asterisk geniesis (n=geniesis@c220-239-144-224.eburwd1.vic.optusnet.com.au) |
| 04:13.27 | MrTelephone | is there support for sip messaging yet? |
| 04:14.31 | kiscokid | what is sip messaging? |
| 04:14.54 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 04:15.24 | MrTelephone | like msn on but its through the sip stack |
| 04:15.33 | MrTelephone | never seen that on the polycom phones? |
| 04:16.05 | kiscokid | yeah I have a Polycom. I was wondering what it was for |
| 04:16.51 | kiscokid | not sure I'd want to type text on a phone keyboard when my computer is sitting next to the phone |
| 04:18.52 | *** join/#asterisk mogorman (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 04:18.52 | *** mode/#asterisk [+o mogorman] by ChanServ |
| 04:18.59 | geniesis | I've got a problem with the TDM844B digium card. For some reason when i call out using the FXS port and it gets routed through the FXO port, after the call has ended the FXO port then has excessive noise on the RX line. However if i plug a phone in parallel with FXO port and the land line, i don't hear any noise whatsoever, so it seems to be an added artifact of the digium drivers. I have set everything up according to |
| 04:18.59 | geniesis | whats needed in austrlia. But I can't think of anything else that could be causing the problem. |
| 04:19.07 | kimo_sabe | kiscokid: my thoughts exactly |
| 04:19.59 | geniesis | the only reason for messaging via sip would be for systems that support SMS |
| 04:20.21 | kiscokid | messaging to cell phones? |
| 04:21.47 | MrTelephone | that sms i thnk |
| 04:22.58 | geniesis | yes |
| 04:23.09 | geniesis | there are a lot of telcos now that support landline sms |
| 04:27.09 | MrTelephone | how do you send an sms message |
| 04:27.17 | MrTelephone | you have to subscribe |
| 04:27.28 | coppice | with an SMS phone. I have one right beside me |
| 04:27.38 | MrTelephone | its such a money grab that crap |
| 04:27.58 | MrTelephone | .50 cents to call a tv add to vote |
| 04:27.59 | MrTelephone | wtf |
| 04:28.05 | coppice | we don't pay. even for the phone |
| 04:28.26 | geniesis | in any case, any one had the troubles i have with the digium card? |
| 04:29.20 | MrTelephone | geniesis, disable echo cancel |
| 04:29.32 | MrTelephone | yeah i hear of lots of problems with those cards, not just digium |
| 04:29.44 | MrTelephone | try to disable echo cancel and see if you get the noise |
| 04:29.55 | MrTelephone | i get odd distortion on my t1 all the time |
| 04:30.10 | MrTelephone | run your zttest and make sure your hitting irqs at 99.98% or higher |
| 04:30.26 | MrTelephone | fxs to sip is ok? |
| 04:30.38 | geniesis | yup. fxs to sip is fine |
| 04:30.47 | MrTelephone | sip to fxo is fine? |
| 04:30.53 | geniesis | and pstn->fxo->fxs->phone is also fine |
| 04:31.00 | MrTelephone | wierd eh |
| 04:31.06 | MrTelephone | but calls out are shit? |
| 04:31.20 | geniesis | it takes one call out to start the noise |
| 04:31.26 | geniesis | once the noise starts you can't stop it |
| 04:31.34 | shido6 | turn off echo training |
| 04:31.41 | geniesis | and calls in any direction on the fxo port is stuffed |
| 04:31.45 | MrTelephone | yeah turn off all the echo crap |
| 04:32.00 | geniesis | just on the fxo? |
| 04:32.06 | MrTelephone | on the fxs |
| 04:32.16 | MrTelephone | echo cancel only one one side |
| 04:32.20 | MrTelephone | on one side |
| 04:32.32 | MrTelephone | i think i solved my issue by echo cancelling the telco side |
| 04:32.41 | MrTelephone | and turning off the other crap |
| 04:32.58 | MrTelephone | i could be on crack though |
| 04:33.32 | geniesis | well...reloading now |
| 04:33.38 | MrTelephone | there's always gunshots going off here |
| 04:33.43 | MrTelephone | people are shooting bears |
| 04:35.06 | MrTelephone | <PROTECTED> |
| 04:35.36 | geniesis | nope...that didn't work....i have fxo echocancel on, and fxs echocancel off |
| 04:36.21 | shido6 | what did you do to initialize your settings? |
| 04:36.48 | geniesis | asterisk stop && zaptel restart && asterisk start |
| 04:36.52 | *** join/#asterisk ZeroPing (n=none@adsl-217-188-53.owb.bellsouth.net) |
| 04:36.56 | geniesis | in the init.d dir |
| 04:37.16 | shido6 | turn off all the echo can settings and try again |
| 04:37.26 | shido6 | fxs, fxo, everything |
| 04:37.44 | MrTelephone | what kind of sound is it making i wonder |
| 04:39.31 | geniesis | MrTelephone: its pretty much very loud white noise |
| 04:39.34 | TheCops | Someone is using SLA support for Asterisk ? I have something weird, when station2 is using line1, and station1 trying to take that line, it merge to station2's line1 and you can listen the call. It is supposed to denied to call. someone heard something about that ? |
| 04:39.39 | geniesis | like when you tune a radio to a blank station |
| 04:40.25 | MrTelephone | like a fax transmission too? |
| 04:40.32 | geniesis | no |
| 04:40.42 | geniesis | doesn't sound like modem breath |
| 04:40.52 | MrTelephone | what version of zaptel |
| 04:40.55 | geniesis | 1.4.3 |
| 04:41.59 | MrTelephone | try and update |
| 04:42.20 | MrTelephone | sounds like a software bug too me.. best try a different release |
| 04:42.43 | geniesis | i just did a yum -y update about an hour ago |
| 04:43.03 | geniesis | also tried the 1.4.4 drivers as well |
| 04:44.52 | MrTelephone | only thing to do now would be to contact digium |
| 04:45.06 | geniesis | hrmm |
| 04:45.07 | geniesis | damn |
| 04:46.15 | MrTelephone | well i'd ask these guys during the day |
| 04:46.21 | MrTelephone | everyone is gone |
| 04:47.36 | *** join/#asterisk ming_zy1 (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 04:49.01 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 04:52.56 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-182-96.hsd1.or.comcast.net) |
| 05:08.07 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 05:08.45 | bkruse_home | anyone have a box with a t1 card i can test some code on? geez |
| 05:16.59 | Nugget | what kind of code? |
| 05:17.28 | coldsteal | anyone know of a itsp that allows CID spoofing? |
| 05:19.50 | *** part/#asterisk rue_mohr (n=rue@h24-207-90-24.cst.dccnet.com) |
| 05:20.16 | bkruse_home | Nugget: nevermind, i found a random box on the digium vpn, itll do |
| 05:20.22 | *** part/#asterisk foo (n=foo@unaffiliated/foo) |
| 05:20.23 | bkruse_home | so digital card detection code |
| 05:20.38 | *** join/#asterisk keith4_ (n=keith@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 05:23.03 | bkruse_home | geniesis: have you tried fxotune? |
| 05:24.26 | keith4_ | what are potential reasons why a command that runs fine in the shell doesn't seem to work via System() in the dial plan? |
| 05:25.21 | Nugget | okey |
| 05:25.36 | kimo_sabe | keith4_: different environment variables. Different user if you arn't running asterisk as root |
| 05:25.47 | Nugget | keith4_: Your shell path is more comprehensive than the system shell path? |
| 05:25.57 | keith4_ | full paths in system() |
| 05:26.07 | Nugget | define "full" :) |
| 05:26.19 | keith4_ | uh |
| 05:26.24 | keith4_ | /usr/local/bin/foo |
| 05:26.29 | coldsteal | like /sbin/hwclock |
| 05:27.07 | Nugget | It wouldn't shock me that /usr/local/bin wasn't in /bin/sh's default path. |
| 05:27.26 | keith4_ | ... what? |
| 05:27.35 | Nugget | which word was unclear? |
| 05:27.45 | keith4_ | it was clear |
| 05:27.52 | keith4_ | but i don't see how it's relevent |
| 05:28.32 | Nugget | I was just suggesting that perhaps the command you're trying to run isn't in the path when invoked from system, which I'm SURE you are aware is likely to be very different than the shell experience you enjoy when you're logged in with bash or tcsh or whatever. |
| 05:28.59 | Nugget | in particular, there's no reason to expect that anything in /usr/local/ will be in the path unless you've gone out of your way to ensure that it is |
| 05:29.00 | keith4_ | which is why I said that i'm using full paths in the system() call |
| 05:29.12 | Nugget | you didn't say that. What you said was sort of unclear. |
| 05:29.12 | *** join/#asterisk [LiFE] (i=life@unaffiliated/life/x-0000003) |
| 05:29.23 | [LiFE] | anyone have experience with DIDWW? |
| 05:29.26 | Nugget | I had no idea what you were trying to say |
| 05:29.30 | coldsteal | http://www.voip-info.org/wiki-Asterisk+cmd+System |
| 05:29.51 | keith4_ | right |
| 05:29.53 | geniesis | bkruse_home: yep i've tried fxotune...didn't do anything |
| 05:30.15 | bkruse_home | geniesis: did not apply any settings you mean? |
| 05:30.27 | bkruse_home | as in, fxotune.conf did not write values? |
| 05:30.29 | geniesis | as in it didn't make any different |
| 05:30.38 | keith4_ | so.... System(/usr/local/bin/foo bar) doesn't work, but if I paste '/usr/local/bin/foo bar' in the shell, it works fine.. |
| 05:30.40 | geniesis | i still get the whitenoise problem |
| 05:30.48 | keith4_ | and the CLI doesn't seem to show anything, even with debug 10 |
| 05:30.55 | keith4_ | and the log doesn't seem to show anything either |
| 05:31.41 | geniesis | hrmm...the new update seems to have caused this error now "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)" |
| 05:31.51 | bkruse_home | lol |
| 05:31.53 | bkruse_home | asterisk -g |
| 05:31.55 | bkruse_home | asterisk -r |
| 05:32.05 | coldsteal | keith4_: try like system( /bin/touch /file.txt ) |
| 05:32.41 | bkruse_home | keith4_: is it in quotes? |
| 05:32.54 | keith4_ | it's not in quotes. should it be? |
| 05:33.02 | keith4_ | could it be a permissions problem? |
| 05:33.16 | keith4_ | seems to be that * is overly fussy about permissions, like with callfiles and such |
| 05:33.31 | bkruse_home | it could be permission (ps aux | grep asterisk) I run asterisk as root |
| 05:33.33 | bkruse_home | put it in quotes |
| 05:33.43 | bkruse_home | system("whatev goes here") |
| 05:33.43 | coldsteal | keith4_: and it should be for security reasions |
| 05:33.59 | keith4_ | i agree. i don't run * as root, either |
| 05:34.02 | keith4_ | should I be? |
| 05:34.05 | bkruse_home | I do |
| 05:34.20 | bkruse_home | if you have a secure asterisk setup, its fine, i have no problem with it |
| 05:34.26 | *** part/#asterisk [LiFE] (i=life@unaffiliated/life/x-0000003) |
| 05:34.32 | coldsteal | keith4_: lol i do but i trust some other security like selinux |
| 05:34.48 | bkruse_home | coldsteal: asterisk + seline = good luck |
| 05:35.01 | keith4_ | could failed variable substitution cause a system command to fail? |
| 05:35.09 | bkruse_home | Of course you should practice good security, but I am usually deving |
| 05:35.25 | bkruse_home | its best to start with what works and work to security, than start super secure and try to get something working. |
| 05:35.50 | geniesis | seems as tho asterisk doesn't create the asterisk folder if it doesn't exist |
| 05:38.57 | *** join/#asterisk mtaht4 (n=m@pool-70-109-104-250.alb.east.verizon.net) |
| 05:40.41 | coldsteal | any of u have telasip? |
| 05:42.29 | keith4_ | if I use ${CALLERIDNUM} in a command, how exactly does that get expanded? any quotes or anything? |
| 05:43.40 | sweeper | keith4_: no quotes, just a string |
| 05:43.55 | keith4_ | ok thanks |
| 05:44.10 | sweeper | how do I check if a variable is empty? ${VAR} = "" doesn't seem to work :/ |
| 05:44.26 | coldsteal | anyone know of a voip provider that allows Caller ID spoofing? |
| 05:44.45 | coldsteal | sweeper: use == |
| 05:44.54 | sweeper | coldsteal: uh, most allow you to set your own caller id, afaik |
| 05:45.37 | coldsteal | sweeper: lol yeah but im talking about spoofing |
| 05:45.48 | sweeper | uh |
| 05:45.59 | sweeper | what's the difference? :v |
| 05:46.05 | sweeper | it still shows up as whatever you want it to |
| 05:46.23 | sweeper | just set the channel variables to whatever before you do the dial |
| 05:47.10 | coldsteal | im trying to spoof a cid and some providers dont allow it |
| 05:47.51 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 05:48.45 | keith4_ | does ${CALLERIDNUM} include < and > ? |
| 05:50.40 | kimo_sabe | keith4_: throw it in a NoOp to see |
| 05:51.13 | keith4_ | see in debug? |
| 05:53.03 | sweeper | coldsteal: eh, I've never used one that didn't allow it. I mean, I'm not even paying them for a DID, so there's no VALID cid to be had anyways |
| 05:53.44 | bkruse_home | keith4_: use ${CALLERID(num)} |
| 05:55.47 | bkruse_home | gah i cannot find a machine with a t1 card! bah! |
| 05:56.07 | kimo_sabe | bkruse_home: what are you trying to test? |
| 05:56.26 | keith4_ | bkruse_home: didn't seem to make a difference |
| 05:56.29 | bkruse_home | kimo_sabe: some digital card auto detecting stuff |
| 05:56.52 | kimo_sabe | bkruse_home: now that I think about it I might have one with a tdm2xxp (I htink) in it |
| 05:57.36 | kimo_sabe | bkruse_home: I think it might even be linking to a channel bank |
| 05:57.53 | bkruse_home | that would work great |
| 05:58.01 | bkruse_home | one thats back up to another t1 would be great also |
| 05:58.10 | bkruse_home | (eg it will come out of red alarm with esf/b8zs of similar) |
| 05:59.29 | kimo_sabe | bkruse_home: Lemmi see if it's still up, there've been some power outages lately |
| 05:59.38 | bkruse_home | yay, thanks |
| 06:00.00 | keith4_ | alright, need some sleep. thanks guys |
| 06:04.32 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 06:10.51 | kimo_sabe | bkruse_home: bah, I think the storms got it. It was only a testing system so it's not on a UPS or set to auto powerup |
| 06:12.34 | *** join/#asterisk hijacked (n=argh@cerebus.clandestineresearch.com) |
| 06:12.40 | Qwell | bkruse_home: see redbull :D |
| 06:13.04 | bkruse_home | Qwell: done |
| 06:13.08 | bkruse_home | kimo_sabe: darnit :[ |
| 06:17.09 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 06:22.54 | Qwell | doubt it'll work :D |
| 06:23.27 | Qwell | oh, it did, nice |
| 06:23.35 | Qwell | bkruse_home: that's totally my new ringtone |
| 06:25.38 | *** join/#asterisk hfb (n=hfb@75.80.37.175) |
| 06:29.17 | Juggie | Corydon76-home |
| 06:30.55 | *** join/#asterisk shinao1 (n=shinao1@41.205.186.82) |
| 06:31.45 | Corydon76-home | Juggie: what? |
| 06:34.40 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:43.16 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 06:48.09 | Juggie | i wanted to talk to you about a couple of old agi commits |
| 06:48.14 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:48.14 | Juggie | which seem to have caused a behaviour change |
| 06:48.34 | sweeper | the smackdown cometh! |
| 06:51.52 | Juggie | heh, i cant put the smackdown :) |
| 06:53.59 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 07:01.19 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 07:01.48 | ZX81 | :) hi all |
| 07:03.08 | ZX81 | ~adn |
| 07:03.09 | jbot | hmm... adn is is the Asterisk Daily News - http://www.sineapps.com/news.php for HTML and http://feeds.feedburner.com/asterisknews for RSS |
| 07:03.39 | ZX81 | jbot: no adn is hmm... adn is is the Asterisk Daily News - http://www.venturevoip.com/news.php for HTML and http://feeds.feedburner.com/asterisknews for RSS |
| 07:03.39 | jbot | okay, ZX81 |
| 07:03.43 | ZX81 | ~adm |
| 07:04.05 | ZX81 | mmm I can't spell |
| 07:04.08 | ZX81 | ~adn |
| 07:04.09 | jbot | from memory, adn is hmm... adn is is the Asterisk Daily News - http://www.venturevoip.com/news.php for HTML and http://feeds.feedburner.com/asterisknews for RSS |
| 07:06.26 | *** join/#asterisk MrMister2 (n=mrmister@89-180-58-247.net.novis.pt) |
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| 07:28.04 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 07:38.13 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 07:55.19 | CrazyTux[m] | anyone know what the latest version for asterisk addons is? |
| 07:59.29 | bkruse_home | anyone done snmp stuff here, with asterisk/ |
| 07:59.30 | bkruse_home | ? |
| 08:12.57 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 08:17.53 | geniesis | hrmm...i crashed the zaptel driver by using the "-p: Get a pre-echocanceled stream." switch in the ztmonitor util. |
| 08:18.14 | *** join/#asterisk hang3r (n=hang3r@124-168-236-35.dyn.iinet.net.au) |
| 08:57.55 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 09:53.44 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
| 10:03.17 | *** join/#asterisk toot (n=toot@84.19.255.123) |
| 10:13.23 | *** join/#asterisk Paul_UK (i=Paul_UK@78.32.14.83) |
| 10:15.33 | Paul_UK | hey all, i have an asterisk box fully working, but now i need to get my head round the terminalogy. i have a inbound trunk, extensions, queue, ring/hunt groups. if i want to create a set of agents to recieve calls from an inbound trunk, is it inbound trunk --> queue --> ring group --> extensions ? |
| 10:22.00 | sheldonh | Paul_UK: i'm not much of a cluebat on user pbxs, but i suspect queues and ring groups operate at the same level, so you'd use one or the other? |
| 10:24.20 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 10:24.41 | Paul_UK | sheldonh, no i was lead to believe you could have multiple queues, but then different agents in those queues and their groups would either be ring all or hunt |
| 10:26.36 | yonahw-work | Paul_UK: I believe you are referring to the ring strategy within a queue as well as agents within queues and their priorities and such, try checking out http://www.voip-info.org/wiki-Asterisk+call+queues |
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| 11:19.10 | *** join/#asterisk saftsack (n=oliver@p54A7D03A.dip.t-dialin.net) |
| 11:20.39 | tzafrir | geniesis, hmmm... nice |
| 11:20.50 | tzafrir | could you please provide more details? |
| 11:21.00 | tzafrir | (or file a bug report) |
| 11:23.48 | *** join/#asterisk af_ (n=getsmart@81-174-46-138.dynamic.ngi.it) |
| 11:39.17 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
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| 12:04.45 | *** join/#asterisk saftsack (n=oliver@p54A7D109.dip.t-dialin.net) |
| 12:21.47 | tzafrir | ~seen nextime |
| 12:21.54 | jbot | nextime <n=nextime@unaffiliated/nextime> was last seen on IRC in channel #asterisk, 42d 14h 57m 52s ago, saying: 'anyone using mISDN 1.2.0?'. |
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| 12:24.50 | *** join/#asterisk shinao1 (n=shinao1@41.205.184.94) |
| 12:25.22 | sweeper | how can I check the build options of an asterisk install? |
| 12:40.19 | tzafrir | a /proc/config.gz-like in the help text? |
| 12:40.28 | tzafrir | show config? |
| 12:42.20 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 12:45.05 | *** join/#asterisk saftsack (n=oliver@p54A7E3E5.dip.t-dialin.net) |
| 12:46.27 | sweeper | nah |
| 12:46.42 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 12:49.33 | *** join/#asterisk masus (n=tet@88.248.14.186) |
| 12:49.50 | masus | hi all can anybody tell me on which os is asterisk developed |
| 12:52.07 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 12:52.11 | masus | debian ? centos ? |
| 12:53.49 | JT | linux |
| 12:53.55 | JT | distro doesn't matter that much |
| 12:54.13 | masus | hmmm i would like to learn the best one |
| 12:54.18 | masus | so i have no problems |
| 12:54.41 | masus | after install |
| 12:54.43 | JT | it's a matter of preference |
| 12:54.46 | JT | i prefer debian |
| 12:55.02 | masus | ok that was the answer ;) |
| 13:11.27 | mvanbaak | I prefer debian as well |
| 13:12.22 | geniesis | tzafrir: zaptel ver 1.4.3 on trixbox install (centos 4.5 final) command to create the crash is ztmonitor 6 -m -p -r streamrx.raw -t streamtx.raw -R streampreechorx.raw -T streampreechotx.raw , relevant dmesg output=Kernel panic - not syncing: /usr/src/redhat/BUILD/zaptel-1.4.3/zaptel-base.c:6944: spin_lock(/usr/src/redhat/BUILD/zaptel-1.4.3/zaptel-base.c:e09b56d4) already locked by /usr/src/redhat/BUILD/zaptel-1.4.3/za |
| 13:12.22 | geniesis | ptel-base.c/3953 |
| 13:13.09 | *** join/#asterisk saftsack (n=oliver@p54A7E190.dip.t-dialin.net) |
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| 13:17.26 | *** join/#asterisk SwK_ (n=SwK@24.248.196.141) |
| 13:17.36 | emacsen | Hi. A dumb newbie question. When I installed Asterisk, it came with a dozen or so config files. I want to start very simple. Which of those files is essential and which (if any) can I delete for now? |
| 13:19.22 | masus | debian lokks like freebsd |
| 13:19.56 | tzafrir | trixbox is centos 4.5 but with centos 4.3 kernel |
| 13:20.52 | tzafrir | geniesis, any chance you have a more complete trace? |
| 13:21.14 | tzafrir | idealy: a serial console connected to another computer... |
| 13:22.39 | emacsen | sorry, 64 files :) |
| 13:22.42 | shido6 | good |
| 13:22.43 | shido6 | question |
| 13:22.50 | shido6 | a lot of asterisk users dont go that route |
| 13:22.53 | shido6 | you should |
| 13:23.03 | shido6 | delete the modules or dont load the modules you dont need |
| 13:23.31 | shido6 | check /usr/lib/asterisk/modules and look at all those modules. There are a lot |
| 13:24.03 | emacsen | shido6: ok, and they'll match to the config filenames? |
| 13:24.03 | shido6 | taking up resources you could use for other tasks and not have to worry about voice quality or throwing in a faster processor, etc. |
| 13:24.10 | shido6 | you almost got it... |
| 13:24.25 | shido6 | modules.conf will tell asterisk what modules to load and which to ignore |
| 13:24.42 | emacsen | shido6: I don't even have an analog card. All I need is AIX2 to my VOIP provider, SIP to my ATA, voicemail and maybe FWD AIX connectivity |
| 13:24.44 | shido6 | we should start with , "What do you want to DO with Asterisk?" |
| 13:25.06 | shido6 | AIX is an IBM distro, you mean IAX :) |
| 13:25.11 | emacsen | IAX, sorry |
| 13:25.11 | sheldonh | hehe |
| 13:25.24 | geniesis | tzafrir: yup |
| 13:25.33 | sheldonh | i believe AIX2 is now called RHEL ;) |
| 13:25.44 | geniesis | tzafrir: i can make a bug report |
| 13:26.04 | emacsen | shido6: *shudder* irix... |
| 13:26.19 | shido6 | hehe |
| 13:26.20 | shido6 | ok |
| 13:26.26 | emacsen | shido6: unless you like binaries in /etc and all your apps suid root... anyway back to this |
| 13:26.27 | shido6 | I willbe back in about 30 minutes |
| 13:27.22 | sheldonh | <PROTECTED> |
| 13:27.31 | sheldonh | you get moron admins on every platform :) |
| 13:28.00 | emacsen | sheldonh: no... on irix they put ifconfig in etc (and maybe others, I forget) |
| 13:28.10 | sheldonh | wow |
| 13:28.16 | sheldonh | that's very special |
| 13:29.12 | emacsen | ...yeah... |
| 13:32.36 | *** join/#asterisk Falle (n=falle@87.ftth2.cust.cen2.sksk.se.borderlight.net) |
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| 13:37.49 | Sci_05 | can someone give me a hand, I upgrade from 1.4.5 to 1.4.9 and I can't get anything to work. I have pasted my testing info at http://www.pastebin.ca/638012 |
| 13:38.25 | Sci_05 | Everything works fine when I use 1.4.5 but in 1.4.9 I am not even seeing the calls come in and yet they get put into some other context that they shouldn't be |
| 13:39.59 | *** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-185.range81-152.btcentralplus.com) |
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| 13:44.06 | geniesis | tzafrir: got an email i can send the console output to |
| 13:45.45 | Sci_05 | geniesis: use pastbin |
| 13:45.49 | Sci_05 | ~pb |
| 13:45.51 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:57.04 | *** part/#asterisk masus (n=tet@88.248.14.186) |
| 14:07.30 | Sci_05 | why doesn't 1.4.9 like me?!?!?! |
| 14:10.30 | tzafrir | Sci_05, you got extension "s" |
| 14:10.50 | tzafrir | Any specific extension before that? |
| 14:11.13 | Sci_05 | no I don't |
| 14:11.19 | tzafrir | Is it an incoming call? |
| 14:11.25 | Sci_05 | yes |
| 14:11.48 | Sci_05 | when I get calls in I should at least be seeing this "- Executing [9702406605@did_context:1] Answer("Zap/1-1", "") in new stack" but I don't even get that in 1.4.9 |
| 14:12.21 | Sci_05 | if I install verions 1.4.5 everything works a-ok but not in 1.4.9 |
| 14:14.57 | shido6 | Zzz |
| 14:17.09 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-176-115.buff.east.verizon.net) |
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| 14:19.44 | SuPrSluG | anyone use fxo gateways such as audio codes? any thoughts on quality vs pci card? |
| 14:20.19 | shido6 | takes less time to reboot :) |
| 14:23.32 | SuPrSluG | are they any good? i'd like to use one because i'd rather not place the server where the telco drop is. |
| 14:25.25 | *** join/#asterisk Penggu (n=me@203-213-102-59-nme-ts7-2600.tpgi.com.au) |
| 14:28.35 | *** join/#asterisk ivanfm (n=ivanfm@c906b486.virtua.com.br) |
| 14:28.48 | SuPrSluG | ~fxo gateway |
| 14:30.23 | shido6 | go for it |
| 14:30.33 | shido6 | it has a bit of a learning curve |
| 14:30.40 | shido6 | but it works |
| 14:39.04 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
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| 15:02.06 | *** join/#asterisk [1]kimc (n=kimc@c-68-41-255-85.hsd1.mi.comcast.net) |
| 15:03.33 | *** join/#asterisk DragoraN (n=dragoran@217.67.19.74) |
| 15:03.34 | DragoraN | hi |
| 15:03.42 | [1]kimc | heh |
| 15:04.41 | DragoraN | is it possible to make changes to SIP ip phone display with sip? does sip protocol support this? |
| 15:04.46 | DragoraN | buttons.. etc. |
| 15:04.57 | shido6 | ? |
| 15:05.06 | shido6 | what phone and what do you want to do specifically? |
| 15:06.03 | *** join/#asterisk kimc_ (n=kimc@c-68-41-255-85.hsd1.mi.comcast.net) |
| 15:06.37 | DragoraN | for example cisco ip phone with sip firmware, and i want to control layout of the phone |
| 15:07.22 | shido6 | you can control the services, messages, directory, and line labels... |
| 15:07.32 | [1]kimc | on a cisco 7960 you need to progarm the phone using tftp to upload a configuration file into it |
| 15:07.35 | shido6 | user/pass for each line. |
| 15:08.06 | shido6 | are you trying to tweak the softbuttons outside of the line buttons? |
| 15:08.31 | DragoraN | i dont know |
| 15:09.00 | shido6 | what do you want your phone to do? |
| 15:12.07 | DragoraN | write own texts to display |
| 15:12.08 | *** join/#asterisk ZeroPing (n=none@adsl-217-188-53.owb.bellsouth.net) |
| 15:15.32 | shido6 | u can label the lines |
| 15:16.03 | shido6 | and services |
| 15:17.03 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 15:21.06 | *** join/#asterisk joaop (n=taken@201-67-131-64.bnut3700.dsl.brasiltelecom.net.br) |
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| 15:23.59 | *** join/#asterisk joaop (n=taken@201-67-131-64.bnut3700.dsl.brasiltelecom.net.br) |
| 15:33.22 | joaop | there a brazilian asterisk channel in f. node? |
| 15:33.28 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 15:47.49 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 15:51.01 | *** join/#asterisk NoIdea465 (n=NoIdea46@d53-64-116-212.nap.wideopenwest.com) |
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| 15:59.27 | SuPrSluG | anyone try the Grandstream GXP-2020 |
| 15:59.53 | coppice | It looks nice in the pictures :-) |
| 16:01.24 | tzafrir | joaop, someone bothered registering #asterisk-br . It is empty right now |
| 16:02.25 | joaop | tzafrir, after a search i found the wierd channel name #asteriskbrasil.org :P |
| 16:02.56 | tzafrir | someone ought to put a reference between those two channels... |
| 16:03.04 | tzafrir | or a forwarder |
| 16:03.17 | joaop | it would be nice |
| 16:05.59 | Strom_C | I think Grandstream's mission statement has to include the phrase "ensure that no one can undercut us on price or quality..." |
| 16:06.16 | *** part/#asterisk awannabe (n=gti@ip24-251-135-202.ph.ph.cox.net) |
| 16:06.54 | NoIdea465 | On the flip side, I have a Cisco 7970 that is giving me hell |
| 16:07.04 | Qwell | NoIdea465: send it here, I'll make it happy |
| 16:07.08 | Nugget | That's what Ciscos do. |
| 16:07.11 | Qwell | (and give it a good home) |
| 16:07.43 | Strom_C | Qwell: will you love it and feed it and brush its hair and care for it for ever and ever ?!!!?!? |
| 16:07.47 | Qwell | yes |
| 16:07.50 | Strom_C | awwwwwwwwwwwwwwwwwwwwwwwwwww |
| 16:07.58 | Qwell | That's totally gonna work one of these days - just watch |
| 16:08.04 | NoIdea465 | And wipe finger prints off the screen every night? |
| 16:08.06 | Strom_C | hahahahaha |
| 16:08.18 | Qwell | NoIdea465: every hour |
| 16:08.39 | Qwell | and it'll run skinny...it'll be happy |
| 16:08.44 | NoIdea465 | That's dedication |
| 16:08.54 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 16:09.00 | Qwell | right [TK]D-Fender ? |
| 16:09.31 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 16:09.55 | NoIdea465 | Would anyone out there happen to know where I can get 7970 Firmware? |
| 16:10.01 | Qwell | cisco.com |
| 16:10.07 | Qwell | it'll cost you like $150 though |
| 16:10.44 | NoIdea465 | I tried getting a smartnet contract but not many will sell it |
| 16:10.51 | Qwell | yeha |
| 16:10.55 | Qwell | welcome to Cisco |
| 16:11.31 | ManxPower | technically getting a smartnet contract does not give you a LICENSE for the SIP firmware. |
| 16:11.34 | Qwell | ooo |
| 16:11.39 | hi365 | what would i have to put in to a contex for disa? |
| 16:11.45 | hi365 | is this line enough? |
| 16:11.46 | Qwell | ManxPower: this is true |
| 16:11.47 | ManxPower | it just gives you ACCESS to it, like getting MS Office off bittorrent. |
| 16:11.51 | Qwell | so it'll cost you like $250 |
| 16:11.56 | Qwell | to be "legit" |
| 16:12.01 | Qwell | (PER PHONE) |
| 16:12.03 | hi365 | exten => s,n,Dial(Local/${EXTEN}@from-internal) |
| 16:12.15 | Qwell | somebody remind me to make my Cisco spoof image later |
| 16:12.20 | Qwell | brb |
| 16:12.25 | NoIdea465 | I had it conecting via sccp to * |
| 16:12.43 | ManxPower | hi365: Think of Local/ as a fancy Goto. |
| 16:12.53 | *** join/#asterisk saftsack (n=oliver@p54A7E383.dip.t-dialin.net) |
| 16:13.11 | hi365 | ManxPower: so it *should* work then - right? |
| 16:13.14 | ManxPower | hi365: there is NO reason to use ${EXTEN} with Dial unless you have a pattern match. |
| 16:13.24 | ManxPower | hi365: well EXTEN = s at that point |
| 16:13.44 | hi365 | ManxPower: ok. from the begining: im trying to set my callerid based on the number im calling |
| 16:13.51 | ManxPower | I don't know why that is so hard for everyone to understand, but I was confused about that as well. |
| 16:14.08 | ManxPower | hi365: well you would not be using exten s |
| 16:14.32 | hi365 | you can do anything to a disa number per-say, unless you pass it to a custom context ... |
| 16:14.38 | ManxPower | "s" is only matched when asterisk doesn't KNOW the dialed number, like if the call comes in on an FXS |
| 16:14.42 | ManxPower | ..er..FXO port. |
| 16:14.55 | CoaxD | Data's name[1] is properly pronounced (in IPA) /'dei.t./ (or day-tuh) as opposed to the alternative pronunciations /'d.:.t./ (dah-ta)or /'dæt../ (datt-a). When Data corrects Dr. Katherine Pulaski for using the latter pronunciation, Pulaski asks, "What's the difference?" Data replies, "One is my name, the other is not".[6] |
| 16:15.53 | Strom_C | winning the irrelevant comment award for today... |
| 16:15.54 | hi365 | ManxPower: so i tried passing it to a contex the read set(calleris=bla), dial(local/{extne}@from-internal) |
| 16:16.04 | CoaxD | Strom_C: Hell yes! Thats me! |
| 16:16.28 | hi365 | ManxPower: is there a different variable that i should be usig in the dail string? |
| 16:16.43 | ManxPower | hi365: exten => s will NEVER be used because asterisk knows the dialed number. |
| 16:16.46 | ManxPower | you need a mattern match |
| 16:17.13 | ManxPower | Sorry, but I'm not really in the mood to teach you Asterisk. Someone else will have to do it. |
| 16:17.18 | hi365 | ManxPower: so i tried doing something like _x.,n,Dial (local........ |
| 16:17.26 | hi365 | but i got the same results |
| 16:18.02 | CoaxD | hi365: I am available to set you up for the rate of $79.99/hr. Please pass me a credit card number and we can proceed. |
| 16:18.40 | hi365 | CoaxD: please post ads to the biz mailing list. this is a supportt channel |
| 16:18.51 | ManxPower | hi365: exten _X. is a good start. |
| 16:19.08 | hi365 | ManxPower: i will try it again |
| 16:19.30 | ManxPower | I can tell you exten "s" is not what you want, based on my understanding of what you are doing. |
| 16:19.48 | ManxPower | hi365: Try putting in the ACTUAL number instead of ${EXTEN}, at least for testing. |
| 16:20.06 | hi365 | ManxPower: will do |
| 16:21.21 | hi365 | ManxPower: http://pastebin.ca/638180 <--- my dial plan |
| 16:21.33 | hi365 | disa gives me a busy signal after the first digit |
| 16:26.21 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 16:26.31 | [TK]D-Fender | hi365, "s" with never get touched in |
| 16:26.32 | [TK]D-Fender | [moshe-dialplan] |
| 16:26.41 | [TK]D-Fender | hi365, and exten => _x.,n,Dial(Local/*099@from-internal) |
| 16:26.57 | [TK]D-Fender | hi365, has no "1" priority. There is NOTHING you can dial in that context |
| 16:27.44 | ai-a | Fender: hmm, cant you include moshe-dialplan,, inside something that already has _x.,1, ... before it ? |
| 16:28.33 | ManxPower | ai-a: You can, but it does not usually work the way you expect it to and it is prone to errors and I do not recommend it. |
| 16:28.57 | ai-a | *now has an anoying habbit of overuse of include default |
| 16:30.18 | hi365 | [TK]D-Fender: if [moshe-dialplan] has just 2 lines, exten => s,1,Answer and exten => _X.,n,Dial(Local/${EXTEN}@from-internal), disa still returns a busytone after 1 digit |
| 16:30.57 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 16:30.59 | [TK]D-Fender | hi365, exten => _X.,n,Dial has no *1* priority an "s" is USELESS and can never be called |
| 16:31.04 | hi365 | [TK]D-Fender: is ${EXTEN} considered s still at that point? |
| 16:31.41 | [TK]D-Fender | hi365, you have completely misunderstood the dialplan & DISA |
| 16:31.50 | *** join/#asterisk joaop (n=taken@201-67-131-64.bnut3700.dsl.brasiltelecom.net.br) |
| 16:31.55 | [TK]D-Fender | hi365, DISA gives your dialtone and matches against the tarrget context |
| 16:32.33 | [TK]D-Fender | hi365, You can't dial "s", and there is no exten to match |
| 16:33.28 | [TK]D-Fender | hi365, just picture what you could tdo if you pointed a SIP softphone's account directly at that context. It could dial nothing. |
| 16:33.33 | hi365 | [TK]D-Fender: i got that much. i was hoping however that disa would PASS the dialed number to the context and the context would handelthe dialing and stuff |
| 16:33.40 | hi365 | right. i see your point |
| 16:34.01 | [TK]D-Fender | hi365, it does, but you can't have " _X.,n," without a " _X.,1," <------------ |
| 16:34.21 | hi365 | ah! gotchya |
| 16:35.08 | Qwell | hmm |
| 16:35.14 | Qwell | my photoshop skills suck |
| 16:35.34 | Qwell | it would've been far funnier with an image to go along with it, but I'll describe what I had in mind :p |
| 16:36.02 | Qwell | Cisco: The Human Network, placed on an image from The Matrix, with all the people in the little pods... |
| 16:36.18 | Qwell | hilarity would have ensued, but alas |
| 16:37.23 | Techie-Micheal | lol |
| 16:37.50 | NoIdea465 | Lol |
| 16:37.52 | Strom_C | lawl |
| 16:37.54 | *** join/#asterisk saftsack (n=oliver@p54A7D292.dip.t-dialin.net) |
| 16:38.10 | NoIdea465 | My 7970 Makes me feel important |
| 16:38.48 | wothinn | I'm trying to build a conference bridge with app_conference and need to be able to play enter-conf-call-number then accept some digits... for the life of me, I can't figure out how to accept the digits I'm trying to get the user to dial next. |
| 16:38.56 | wothinn | Can anyone point me at the right dialplan application? |
| 16:39.00 | ManxPower | NoIdea465: Give it time, eventually your 7970 will make you feel like you wish you had bought a Polycom 8-) |
| 16:39.26 | Qwell | unfortunately, polycom doesn't have anything comparable in features to the 7970... |
| 16:39.29 | NoIdea465 | Right now its asking for "term70.default.loads" |
| 16:39.35 | Qwell | maybe there's no market for it |
| 16:39.37 | [TK]D-Fender | wothinn, "show application read" |
| 16:39.45 | wothinn | Thanks. |
| 16:39.55 | ManxPower | Qwell: what features? |
| 16:40.01 | Qwell | color touchscreen for one |
| 16:40.21 | ManxPower | Yes, of course. My life is meaningless without that! |
| 16:40.34 | ManxPower | 8-) |
| 16:41.01 | NoIdea465 | The cisco also has the ability to impress co-workers |
| 16:41.02 | ManxPower | I just don't like Cisco's firmware licensing as well as the fact their SIP load has far fewer features their their SCCP loads. |
| 16:41.54 | NoIdea465 | I would be fine with sccp if I could just get the thing to boot after I reset it |
| 16:42.10 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 16:42.11 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 16:42.34 | hi365 | [TK]D-Fender: ManxPower: thanks guys! |
| 16:45.59 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 16:49.27 | [TK]D-Fender | hi365, glad you've begun to understand it all now... |
| 16:50.31 | joaop | have some way to use ekiga with blaster phone? |
| 16:51.55 | [TK]D-Fender | joaop, explain this "blaster phone".... |
| 16:52.16 | joaop | ups, sry, it's a voip operator |
| 16:52.29 | Qwell | what does that have to do with asterisk? |
| 16:52.50 | *** join/#asterisk bmg505 (n=leon@196.209.176.104) |
| 16:53.36 | joaop | nothing.. im still reading the teory and taking a look on ekiga |
| 16:54.16 | joaop | *theory with asterisk |
| 16:55.57 | *** join/#asterisk saftsack (n=oliver@p54A7C71F.dip.t-dialin.net) |
| 16:59.26 | *** join/#asterisk hohum (n=dcorbe@dhcp64-134-231-170.shs.nyc.wayport.net) |
| 17:00.09 | *** join/#asterisk hfonteboa (i=_@BHE200150063006.res-com.wayinternet.com.br) |
| 17:11.04 | *** join/#asterisk ManxPower (n=manxpowe@71-8-56-17.dhcp.leds.al.charter.com) |
| 17:11.43 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 17:12.13 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 17:15.39 | Sci_05 | anyone in here using 1.4.9? |
| 17:17.09 | *** join/#asterisk saftsack (n=oliver@p54A7CF04.dip.t-dialin.net) |
| 17:18.26 | sheldonh | Sci_05: not until i can figure out a sensible emergency rollback plan for debian :) |
| 17:19.09 | Sci_05 | I am using debian as well, and I went from 1.4.5 to 1.4.9 and now I can't even see test calls comming in over the pri when using the 1.4.9 |
| 17:19.10 | *** part/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 17:19.45 | Sci_05 | sheldonh: I just have both 1.4.5 and 1.4.9 compiled and everything and just do a make install and restart asterisk to jump between the 2 |
| 17:20.18 | Sci_05 | sheldonh: can you check this out and see if I am making a mistake somewhere I don't see http://www.pastebin.ca/638012 |
| 17:21.58 | ManxPower | You should ALWAYS upgrade your zaptel at the same time. |
| 17:22.02 | ManxPower | expecially in 1.4 |
| 17:22.28 | Sci_05 | yes zaptel is all up to date |
| 17:24.00 | *** join/#asterisk file2 (n=IrcNet@asterisk/developer-and-muffin-lover/file) |
| 17:24.00 | *** mode/#asterisk [+o file2] by ChanServ |
| 17:24.09 | file2 | moo. |
| 17:24.16 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 17:24.19 | Sci_05 | boo |
| 17:24.25 | Qwell | FILE |
| 17:24.30 | Qwell | Let's start an MVNO :p |
| 17:25.02 | file2 | hahahaha. |
| 17:25.17 | file2 | yessss... let's doug it. |
| 17:25.27 | file2 | erm do it. |
| 17:26.27 | [TK]D-Fender | Sci_05, == Unknown extension 's' in context 'did_context' requested <- sure doesn't LOOK like PRI |
| 17:26.43 | Qwell | I wonder what is involved in doing that |
| 17:27.08 | [TK]D-Fender | Sci_05, in fact -- Starting simple switch on 'Zap/1-1' screams "yay I'm dumb analog" |
| 17:27.18 | ManxPower | or a CT1 |
| 17:27.24 | Sci_05 | [TK]D-Fender: sorry its not a pri, its a e&m |
| 17:27.33 | file2 | buying airtime from a major carrier. |
| 17:27.54 | coppice | MVNOs come in many shapes and sizes, from pure marketing operations, to people who own everything but base stations |
| 17:27.56 | [TK]D-Fender | Sci_05, And what you've shown us is clearly a dialplan error |
| 17:28.16 | ManxPower | looks like your E&M is not getting the DID. |
| 17:28.26 | Qwell | coppice: well, seeing as how I would only invest about $10... chances are, we won't own much :p |
| 17:28.42 | Sci_05 | then why would it work ok in 1.4.5 and not work at all 1.4.9? |
| 17:28.44 | [TK]D-Fender | Sci_05, you didn't even show us [did_context] that its looking for, and it tells your ight to your face that its looking for "s" in there and you don't have one. |
| 17:28.54 | [TK]D-Fender | Sci_05, You screwed something else up. |
| 17:29.07 | ManxPower | In the USA Sprint seems to me most into doing MVNO |
| 17:29.08 | NoIdea465 | coppice, maybe you can just buy out 'amp'd'mobile |
| 17:29.54 | Qwell | yeah, but they're CDMA |
| 17:29.57 | Qwell | so meh |
| 17:30.00 | ZeroPing | Can someone help me configure my Cisco 7940? |
| 17:30.04 | Sci_05 | [TK]D-Fender: then why don't i have this problem when I run the 1.4.5? |
| 17:30.38 | [TK]D-Fender | Sci_05, that error is self explanitory. Forge whatever you though you had during 1.4.5. You know what needs to be done, so go do it. |
| 17:30.47 | [TK]D-Fender | forget* |
| 17:31.15 | coppice | I love the recent publicity from qulcomm about how they are winning the 3G war, in the same year most of the world's CDMA networks shut down |
| 17:31.15 | ManxPower | Sci_05: It is pretty common for stuff to break when you upgrade Asterisk |
| 17:31.38 | ManxPower | they are doing well in the USA> |
| 17:31.49 | ManxPower | both sprint and verizon ises Qualcomm stuff |
| 17:31.50 | [TK]D-Fender | And canada |
| 17:32.11 | [TK]D-Fender | Sci_05, "s" **EXTEN** in [did_context] |
| 17:32.31 | [TK]D-Fender | Sci_05, no mystery, its telling you to your face. wake up time... |
| 17:32.51 | ManxPower | Sci_05: Um, why are you using an Asterisk release that is 5 days old -- that has not been out long enough for people to discover problems with it. |
| 17:33.22 | Sci_05 | security |
| 17:34.16 | ManxPower | that is the best reason I've heard |
| 17:34.48 | ManxPower | I sometimes forget that people connect Asterisk to "things on the internet". |
| 17:35.21 | ManxPower | 1.2's lack of an RTP jitter buffer nixed that idea for us. |
| 17:35.21 | NoIdea465 | wow |
| 17:36.14 | ManxPower | In my world users and management demand things be reliable -- so we use PRIs, not VoIP for connection to PSTN |
| 17:38.06 | ManxPower | We DO use VoIP all the time -- but only on the LAN and corporate WAN (that has QoS) |
| 17:39.22 | Sci_05 | ManxPower: voip+noQoS=very bad day |
| 17:40.33 | ManxPower | Sci_05: usually the day ends with the villagers storming the NOC waving torches and screaming "Burn the geek!" |
| 17:40.51 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 17:40.57 | ManxPower | Hell, they do that anyway, but I try to give them as few excuses to do so as I can. |
| 17:41.16 | coppice | shouldn't there be pitch forks too? |
| 17:41.41 | ManxPower | Sci_05: no QoS can work fairly well IF you have RTP jitterbuffers on both endpoints AND you have a decent internet connection. |
| 17:41.51 | NoIdea465 | Any sharp object will do |
| 17:42.15 | Sci_05 | ManxPower: I am over by Gary, indiana, no torches and sharp object around here...just plenty of guns.... |
| 17:42.44 | ManxPower | Sci_05: I grew up in Holland MI, they use wooden shoes there. |
| 17:42.45 | Sci_05 | the phrase "shoot first" comes ot mind |
| 17:42.51 | Sci_05 | lol |
| 17:43.07 | NoIdea465 | Sci_05, we are surprisingly close |
| 17:43.08 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 17:43.23 | Sci_05 | NoIdea465: where you located? |
| 17:43.41 | NoIdea465 | If you hear screaming, I bricked my phone |
| 17:46.50 | *** join/#asterisk saftsack (n=oliver@p54A7D930.dip.t-dialin.net) |
| 17:55.47 | *** join/#asterisk zydrunas_ (n=zydrunas@24-119-29-130.cpe.cableone.net) |
| 17:58.08 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 18:01.49 | mvanbaak | ~pb |
| 18:01.50 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:03.58 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-82-81-220-132.cablep.bezeqint.net) |
| 18:08.57 | *** join/#asterisk keith4_ (n=keith@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 18:11.28 | keith4_ | quiet today? |
| 18:11.37 | sweeper | a bit |
| 18:12.10 | mvanbaak | yeah |
| 18:12.13 | mvanbaak | must be holidays |
| 18:12.47 | mvanbaak | hhmm, the asterisk-users list is still slow |
| 18:12.47 | keith4_ | well, i'll liven it up a bit |
| 18:13.13 | keith4_ | i rebooted my asterisk box for a new kernel, and now am seeing some weird behavior |
| 18:13.26 | keith4_ | ztcfg -vv shows: Changing signalling on channel 1 from Unused to FXO Kewlstart |
| 18:13.26 | keith4_ | ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25) |
| 18:13.31 | mvanbaak | !define weird behavior |
| 18:13.33 | keith4_ | with digium hardware |
| 18:13.39 | Strom_C | keith4: did you compile zaptel for the new kernel? |
| 18:14.00 | keith4_ | well, it's the same version |
| 18:14.07 | keith4_ | just a security update, afaik |
| 18:14.12 | keith4_ | still need to compile? |
| 18:14.16 | Strom_C | be on the safe side and recompile it |
| 18:14.22 | Strom_C | see if that fixes the problem |
| 18:14.23 | keith4_ | ok |
| 18:15.10 | keith4_ | actually, it's packaged (debian) |
| 18:15.14 | mvanbaak | when you install a new kernel you have to recompile the zaptel kernel modules |
| 18:16.36 | *** join/#asterisk hFonteBoa (i=_@BHE200150063006.res-com.wayinternet.com.br) |
| 18:16.59 | *** join/#asterisk ikey1 (i=ikey@219.64.64.40) |
| 18:17.15 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 18:17.29 | keith4_ | how about wctdm? |
| 18:17.36 | mvanbaak | too |
| 18:17.45 | mvanbaak | you should recompile the whole zaptel sources |
| 18:18.11 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
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| 18:23.17 | *** part/#asterisk ikey1 (i=ikey@219.64.64.40) |
| 18:23.46 | keith4_ | oh... it's not packaged. i built that package. d'oh |
| 18:24.02 | keith4_ | that would explain it |
| 18:24.11 | Strom_C | heh |
| 18:24.37 | keith4_ | too many beers ago |
| 18:28.54 | *** join/#asterisk C6vette (n=Darth_Ky@ip70-176-167-93.ph.ph.cox.net) |
| 18:32.04 | tzafrir_laptop | keith4_, if it's the same kernel version string (uname -r, /proc/version), then no need to rebuild any module |
| 18:32.06 | karleeto | lol |
| 18:32.11 | tzafrir_laptop | Certainly not Zaptel |
| 18:32.13 | karleeto | i need beer |
| 18:32.50 | tzafrir_laptop | (also tested with packaged zaptel-modules- packages...) |
| 18:33.36 | *** part/#asterisk C6vette (n=Darth_Ky@ip70-176-167-93.ph.ph.cox.net) |
| 18:35.12 | tzafrir_laptop | connection reset by beer |
| 18:36.02 | *** join/#asterisk karleeto (n=karl@techwifi.franklincomputer.net) |
| 18:36.25 | keith4_ | mvanbaak: thanks, that did it |
| 18:36.30 | keith4_ | i don't know why I didn't think of that |
| 18:36.36 | mvanbaak | :) |
| 18:36.40 | mvanbaak | you're welcome |
| 18:36.41 | keith4_ | tzafrir_laptop: turns out it was a different revision |
| 18:38.27 | keith4_ | exactly what I just did |
| 18:38.43 | *** join/#asterisk NoIdea465 (n=NoIdea46@d53-64-116-212.nap.wideopenwest.com) |
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| 19:03.20 | MrTelephone | i signed up for voicemeup and its not working now |
| 19:03.32 | MrTelephone | how can someone run a business if it goes down all the time |
| 19:04.09 | NoIdea465 | Cisco gets away with it |
| 19:05.15 | *** join/#asterisk hfb (n=hfb@75.80.37.175) |
| 19:06.08 | Techie-Micheal | NoIdea465: Just a guess, but you aren't a fan of Cisco, are you? :P |
| 19:07.44 | JunK-Y | MrTelephone: sign with unlimitel, you wont get problems. |
| 19:08.17 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:10.33 | NoIdea465 | Im trying to upgrade a 7970 to 70.8-0-4SR3S It keeps requesting jar70sip.8-0-3-34.sbn when jar70sip.8-0-3-33.sbn is what was included in the COP file |
| 19:10.59 | NoIdea465 | changing the file name does not seem to help |
| 19:12.13 | sweeper | exten => s,2,GotoIf(["${TARGET}" = ""]?5:forward,1) <-- this should go to 5 if ${TARGET} is nil, right? |
| 19:12.39 | mvanbaak | sweeper: you forgot a $ |
| 19:12.46 | mvanbaak | GotoIf($[ |
| 19:12.50 | sweeper | aha! |
| 19:13.58 | mvanbaak | and I would do: GotoIf($["x${TARGET}" = "x" |
| 19:14.06 | mvanbaak | but maybe that's old habits |
| 19:15.14 | Techie-Micheal | NoIdea465: Hm. Unfortunately I know very little about Cisco VoIP (I dealt more with their routers, switch, and firewalls), but what I did do with their VoIP, the upgrades always seemed to work. |
| 19:18.40 | NoIdea465 | Looks like I need an older version of the firmware since I am trying to load from a factory reset... |
| 19:19.36 | *** join/#asterisk ManxPower (n=manxpowe@71-8-56-17.dhcp.leds.al.charter.com) |
| 19:21.02 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:21.39 | NoIdea465 | Are older versions of sccp firmware for the 7970 public? |
| 19:22.05 | *** join/#asterisk ZeroPing (n=none@adsl-217-188-53.owb.bellsouth.net) |
| 19:23.11 | mvanbaak | NoIdea465: nopes |
| 19:23.50 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 19:27.02 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 19:27.13 | NoIdea465 | any ideas on where I could get my hands on it? |
| 19:27.40 | mvanbaak | cisco ftp |
| 19:27.47 | mvanbaak | but you need an account for that |
| 19:28.04 | NoIdea465 | I have no smartnet :( |
| 19:28.39 | mvanbaak | then you cannot get the firmware |
| 19:30.23 | NoIdea465 | Shutdown ;( |
| 19:32.29 | NoIdea465 | Any recomendations for a contract provider? One without a minimum order? |
| 19:33.01 | mvanbaak | no idea |
| 19:33.06 | mvanbaak | I only know one in .nl |
| 19:33.39 | NoIdea465 | hmmmm |
| 19:50.34 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 19:52.22 | *** join/#asterisk MdeP (n=mdep@119-81-22-190.adsl.tie.cl) |
| 19:55.19 | Bilbolodz | I've question about T.38: how to check is it working or not? |
| 19:55.38 | mvanbaak | asterisk only has T.38 passthru |
| 19:56.42 | Bilbolodz | I've configuration: PSTN <-> SPA-3102 <-> Asterisk <-> SPA-2102 <-> Analog FAX |
| 19:57.12 | Bilbolodz | Should this configuration working with T.38? |
| 19:57.21 | Bilbolodz | works |
| 19:57.23 | Bilbolodz | sorry |
| 19:59.01 | Bilbolodz | I was trying to turn on debug but I can't find any "t.38 debug commands" |
| 19:59.15 | Bilbolodz | so how to check it? |
| 20:00.06 | Bilbolodz | Faxes are sent and received correctly but is it T.38 or just plain 711u fax pass thru |
| 20:00.17 | Bilbolodz | ? |
| 20:02.29 | mvanbaak | check the fax machine |
| 20:03.33 | Bilbolodz | and |
| 20:03.34 | Bilbolodz | ? |
| 20:03.42 | Bilbolodz | What should I look for |
| 20:03.44 | Bilbolodz | ? |
| 20:03.49 | mvanbaak | dont know |
| 20:04.09 | mvanbaak | I have no experience with T.38 |
| 20:04.15 | Bilbolodz | ok |
| 20:04.16 | Bilbolodz | thanks |
| 20:13.31 | *** join/#asterisk hfonteboa (i=_@BHE200150060080.res-com.wayinternet.com.br) |
| 20:19.48 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:21.08 | af_ | there is a specific channel on asteriskgui? |
| 20:23.47 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 20:32.03 | joaop | i done a minimal configuration for asterisk, and im logged with Xlite, how can i test it ? |
| 20:32.24 | joaop | there is no dialplan and no extension configured yet |
| 20:33.37 | af_ | configure echo |
| 20:33.49 | af_ | look at extension 600 in the sample extensions.conf |
| 20:36.20 | joaop | ÓMG, it works |
| 20:36.24 | joaop | 600 |
| 20:36.26 | joaop | ups |
| 20:40.52 | *** join/#asterisk pc500 (i=pc500@66-225-36-13.dynamic.tbc.net) |
| 20:41.38 | pc500 | When I try to use the conference feature on my cisco 7960 I get the following on my asterisk box: |
| 20:41.38 | pc500 | Jul 29 17:21:12 NOTICE[7554]: chan_sip.c:3427 process_sdp: No compatible codecs! |
| 20:41.44 | pc500 | Any ideas? |
| 20:47.34 | *** join/#asterisk bbryant (n=12243@c-68-59-20-153.hsd1.sc.comcast.net) |
| 20:48.03 | *** join/#asterisk DrukenHME (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 20:48.25 | coldsteal | Oo |
| 20:48.26 | coldsteal | Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 20:49.01 | coldsteal | lol i fixed it |
| 20:49.35 | denon | it wasn't broken, you just didn't have asterisk running :) |
| 20:50.02 | CoaxD | hi denon. |
| 20:51.23 | *** join/#asterisk Rickalicious (n=Rickalic@c-76-29-83-134.hsd1.il.comcast.net) |
| 20:51.25 | denon | howdy |
| 20:51.42 | Rickalicious | howdy |
| 20:51.54 | denon | actually I was saying howdy to CoaxD |
| 20:52.03 | denon | but Nugget is saying howdy to you |
| 20:52.19 | Rickalicious | dose anybody know where I can find the patch for chan_oss.c that lets me set the console/dsp to answer, or no answer in the dial context?> |
| 20:52.30 | Nugget | That's just how I roll. |
| 20:52.40 | pc500 | How do I tell what codec my SIP phone is trying to use with conference? I can make phone calls all I want, but the conference functino is broken on my cisco 7960. Worked fine a a sipura adapter. the following error occurs: |
| 20:52.41 | pc500 | Jul 29 17:21:12 NOTICE[7554]: chan_sip.c:3427 process_sdp: No compatible codecs! |
| 20:53.54 | Rickalicious | I am trying to pathc the chan_oss.c file, and keep getting an error. I thought someone here might know about it... |
| 20:54.17 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 20:54.46 | CoaxD | denon: how ya been? |
| 20:55.03 | denon | swamped, you? |
| 20:55.19 | CoaxD | denon: Bout same |
| 20:55.40 | CoaxD | denon: workin' for an outfit these days that does CTI.. consoles, speech recog, paging, etc... |
| 20:55.40 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 20:55.52 | denon | ah, fun |
| 20:55.57 | denon | miss the old isp days? |
| 20:56.05 | CoaxD | denon: Nahh, still in it. hehe |
| 20:56.13 | SwK_ | anyone around with good .ca term rates? need t38 enabled 6/6 billing |
| 20:56.21 | SwK_ | msg me if you do |
| 20:56.29 | denon | lotsa irons in the fire |
| 20:56.41 | CoaxD | denon: oh yeah... never ends.. |
| 20:56.43 | denon | so you going to score me some fun freebie licenses of your company's tech? |
| 20:57.04 | denon | the home ivr could use some speech recog :) |
| 20:57.22 | delmar | ooo speech recog? |
| 20:57.48 | delmar | wouldn't trust that on pstnline tho eh, |
| 20:58.06 | pc500 | I'd rather punch numbers that talk to some computer, thank you. IT's far quicker :). What's worse is when they ask you to speak your account number to one of those things. |
| 20:58.06 | *** join/#asterisk NoIdea465 (n=NoIdea46@d53-64-116-212.nap.wideopenwest.com) |
| 20:58.17 | pc500 | Is it really that hard for someone to hit a number on a phone? |
| 20:58.22 | CoaxD | delmar: Yeah, its good stuff. company directory stuff |
| 20:58.34 | CoaxD | definitely not open source though... cant have everything :) |
| 20:58.46 | denon | pc500: sure, for some things .. but picture "Dial 1 for bob .... dial 436 for frank .. " |
| 20:59.27 | delmar | CoaxD, yeah.. gotta put food on the table somehow. |
| 20:59.39 | SwK_ | all i gotta say about speach recog is "name connector" thats the best pile of speech rec i have ever used |
| 20:59.39 | denon | "..dial 894 for jim" |
| 20:59.40 | pc500 | denon - Yeah, that works, but don't ask me to say Yes or No, don't ask me to speak what I want out of a list of 3 items, and by god (DELTA!) don't make me speak my medaillion number in digit by digit and then correct each one that I got wrong when I could have typed the thing far quicker. |
| 21:00.05 | denon | hehe |
| 21:00.10 | denon | most speech recog will let you dtmf it too |
| 21:00.20 | denon | sometimes they tell you, sometimes the dont |
| 21:00.23 | denon | s/the/they/ |
| 21:00.24 | pc500 | denon - Often times you can (1 for yes, 2 for no), or hit it, but they don't tell you you can. |
| 21:01.08 | pc500 | denon - YOu ever done a major upgrade? Talking 1.0.9 > current 1.4? Should I just re-write the configs from scratch, since I know most of the stuff is busted in 1.2 anyways, rather than hoping to upgade them piece by piece? |
| 21:01.11 | CoaxD | yeah, our stuff does dtmf as well... but the whole solution backends on Nuance, which...well...yeah, pretty much locks you to win32 |
| 21:01.16 | delmar | pc500, i agree.. some peoples idea of a good speach recog system is pathetic. Our NZ Telco has setup one recently thats not too bad... she asks what u want... then confirms what she thinks u were asking about (yes/no) and goes from there. |
| 21:01.32 | delmar | pc500, she wouldn't give me her phone number tho. |
| 21:01.44 | CoaxD | Its pretty stable when all you use the servers for is that one single task... good uptimes, etc.. with good hardware |
| 21:01.49 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 21:01.58 | pc500 | rofl. yeah, the claria %@#$ on sprint pcs is bad too. |
| 21:02.12 | CoaxD | claire from sprint sux! |
| 21:02.12 | SwK_ | delmar, sounds like tmobiles ASR menus |
| 21:02.13 | pc500 | delmar - I think some people in this society can't fathem pressing dtmf tones, so they find it easier to talk to them I guess. |
| 21:02.33 | denon | I find it's easier to whistle them |
| 21:02.45 | denon | no need to reach for a button |
| 21:02.46 | SwK_ | what I REALLY hate is menu hells w/ no operator option |
| 21:02.59 | delmar | pc500, the least they could do is allow the use of 1 for yes 2 for no or something .. as well as speach. |
| 21:03.00 | denon | 'course whistling multiple frequencies at once is tricky |
| 21:03.03 | pc500 | denon - Ever done a major upgade? 1.0.9 > 1.4.x? Can I still use USB port for timing device? |
| 21:03.12 | pc500 | delmar - They don't always do it. ANd in some cases, speech doesn't make sense. |
| 21:03.13 | CoaxD | swk: First rule of thumb in any ivr or speech recog app is to allow someone to press 0 or say 'operator' and immediately bust them out |
| 21:03.21 | pc500 | CoaxD - Never! |
| 21:03.24 | denon | pc500: use ztdummy on 2.6 if you dont have any hardware |
| 21:03.34 | delmar | pc500, heading in the right direction tho eh?. |
| 21:03.40 | SwK_ | CoaxD, i know that and you know that... but I can show you ivr's where that does not work |
| 21:03.42 | pc500 | denon - Ok, I Can't remember, but I used some usb-hack to use the usb port or something. |
| 21:03.47 | CoaxD | swk: Oh, i know |
| 21:03.49 | SwK_ | gov't agencies are the worse for that shit |
| 21:03.56 | denon | pc500: ztdummy is the way to go |
| 21:03.59 | CoaxD | swk: Its the little things people forget.. |
| 21:03.59 | denon | well, hardware is .. |
| 21:04.03 | denon | but ztdummy is the cheap way to go |
| 21:04.07 | pc500 | denon - Does ztdummy work good for music on hold and meetme (makes it not soundl ike shit)? |
| 21:04.10 | SwK_ | forget my ass they do it on purpose |
| 21:04.22 | delmar | pc500, ive had plenty of success with ztdummy. |
| 21:04.23 | denon | pc500: it works fine |
| 21:04.24 | pc500 | denon - Can you still use any cheap $20 modem as a timing device? I'm in a purely SIP enviornment. |
| 21:04.31 | denon | pc500: yes |
| 21:04.56 | SwK_ | pc500 it works fine.... just really needs a 2.6 kernel which most everyone has now |
| 21:04.57 | denon | but just use zt, it's probably fine for you |
| 21:04.57 | pc500 | denon - digium don't have them for sale anymore, but seen them on fleabay |
| 21:05.02 | denon | SwK_: not really true, I still run a *ton* of 2.4 |
| 21:05.23 | pc500 | Let's see what happens |
| 21:05.25 | SwK_ | denon: yeah but the timers ztdummy uses are more accurate in 2.6 |
| 21:05.25 | CoaxD | not: You couldnt really ever just use 'any cheap $20 modem' as a timing device |
| 21:05.26 | delmar | pc500, x100's are still around. i have about 8 in my workshop doing nuthin. lol. |
| 21:05.29 | pc500 | I dout this upgrade will work. |
| 21:05.34 | CoaxD | only the x100p model |
| 21:05.39 | pc500 | delmar - They're cheap timing devices. |
| 21:05.43 | delmar | pc500, upgrading to 1.4? |
| 21:05.49 | pc500 | CoaxD - "asterisk" on ebay finds the $9.99 clones. |
| 21:05.53 | pc500 | delmar - Yes, from 1.0.9 |
| 21:05.54 | CoaxD | pc500: Yeah |
| 21:06.07 | denon | pc500: rewrite your dialplan entirely |
| 21:06.09 | delmar | pc500, plenty of config diffs to look out for. it wont be smooth. |
| 21:06.12 | denon | unless it's huge |
| 21:06.14 | CoaxD | pc500: They're just the same brand/make/model modem.. my point was, not 'any cheap modem' wil do it. it had to be that one |
| 21:06.21 | denon | you'll find a lot of advantages to using the new stuff |
| 21:06.23 | CoaxD | i still have x100p's (and cheap clones) in service, btw |
| 21:06.34 | pc500 | Yeah, I don't like how asterisk CHANGES commands instead of building on existing config architecture. It needs to be stabilized. Look to Cisco IOS for a good example :) |
| 21:06.36 | CoaxD | they never really did ever get the damn callerid right on those |
| 21:06.41 | delmar | CoaxD, had too many echo issues. they are crap imo |
| 21:06.51 | CoaxD | delmar: yeep. |
| 21:06.59 | delmar | CoaxD, dont think i had callerID issues... |
| 21:07.03 | CoaxD | delmar: used to have those problems... |
| 21:07.04 | pc500 | delmar - I did 1.2 about 2 years ago and lots of stuff changed, so I just rolled back and left it alone. |
| 21:07.10 | denon | pc500: there are some pretty good leaps toward standardizing - look at Set() for example |
| 21:07.19 | denon | compared to the way it used to be, SetWhateverName() |
| 21:07.20 | delmar | pc500, yep. reminds me of amavisd-new. |
| 21:07.23 | CoaxD | delmar: Yeah, callerID issues were continual problem w/ x100p |
| 21:07.25 | pc500 | delmar - I'm having a Music on hold and silence supression issue now, and I can't get any support for this old junk. |
| 21:07.38 | delmar | pc500, hate it when things dont upgrade nicely and waste my weekends. |
| 21:07.42 | pc500 | ~2004/2005 era. |
| 21:08.08 | pc500 | How do I save my g729 codec? |
| 21:08.18 | pc500 | I don't remember how I installed it, but I remember it was a pain. |
| 21:08.27 | delmar | CoaxD, maybe its the NZ system but I think callerID always worked... i just couldn't stand the echo, and now for some reason my tdm400 is doing odd things of late... suspect the new zaptel. |
| 21:08.40 | delmar | pc500, its in the asterisk modules dir. |
| 21:08.43 | CoaxD | delmar: heh |
| 21:08.46 | pc500 | delmar - IS it free now? |
| 21:08.55 | pc500 | delmar - I had to pay like $20 or something and manually compile it in. |
| 21:08.57 | CoaxD | delmar: Thing is, TDM400 pretty much eliminated the callerid probs |
| 21:09.02 | delmar | pc500, but if u are upgrading kernel as well ( as you should ) as well as Asterisk, id grab a new codec. |
| 21:09.03 | CoaxD | delmar: For whatever reason.. |
| 21:09.08 | denon | pc500: http://ftp.digium.com/pub/telephony/codec_g729/README |
| 21:09.13 | denon | look at "Backup Procedure" |
| 21:09.15 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 21:09.18 | CoaxD | delmar: it was just the x100p that hated people |
| 21:09.24 | pc500 | denon - I don't kow if I bought it from digium though |
| 21:09.47 | CoaxD | delmar; I finally got out ztmonitor and a set of speakers and listened to the damn fsk... tuned things down to where it wasnt overdriven and wierd sounding... Got it 90% after that |
| 21:10.10 | denon | then it's hard to say if it'll even work under 1.4 |
| 21:10.28 | delmar | CoaxD, what i didnt get was how random it was... some calls fine.. others not.. outgoing calls were usually ok, echo vanished after 15-30secs but incoming calls... omg... and even the tdm400 gets it's knickers in a twist on iincoming calls. |
| 21:10.49 | pc500 | I have no /var/lib/asterisk/licenses |
| 21:10.50 | delmar | CoaxD, yeah i need to re-tune too I think |
| 21:10.51 | CoaxD | delmar: yeah.. we had the same issue |
| 21:11.19 | CoaxD | delmar: Secret was to turn the inputs way down at the POTS inputs, but turn up the gain on the outputs to the fxs to compensate |
| 21:11.23 | delmar | pc500, g729 isn't officially free, but you can certainly get it if you want. |
| 21:11.39 | pc500 | delmar - This is the file: "/usr/lib/asterisk/modules/codec_g729-gcc-pentium3.so" |
| 21:11.43 | CoaxD | delmar: (God, i hate POTS.) |
| 21:11.58 | pc500 | Yeah, I don't even have the licenses directory, that must be new. It was just a download. |
| 21:12.00 | CoaxD | delmar: I wish we could all afford a frickin dialogic port for fxs/fxo. |
| 21:12.04 | CoaxD | delmar: Thats where its at |
| 21:12.28 | delmar | CoaxD, yup. and my wifes mother keeps bloody ringing on the POTS line even after I have told her a dozen times to ring the Auckland DID .. which is crystal clear. |
| 21:12.59 | CoaxD | delmar: hahahha |
| 21:13.02 | pc500 | delmar - Can I reused that file? I don't care, it's 3 of us on a home pbx. |
| 21:13.07 | delmar | CoaxD, i wish i could afford a T1 card :P. I have a spare Carrier Access channel bank with FXO's in it.. in the shed. |
| 21:13.31 | CoaxD | delmar: The TE100P's are lookin pretty cheap on ebay these days |
| 21:13.34 | delmar | pc500, yeah but your license is bound to the MAC address of your lan card.. I believe... just don't change that. |
| 21:13.41 | CoaxD | delmar: Course, didnt you say you're NZ? Yeah, you need E1 stuff i think |
| 21:13.42 | delmar | CoaxD, what kind price? |
| 21:13.45 | Rickalicious | does anybody know where I can find the patch for chan_oss.c that lets me set the console/dsp/answer, or console/dsp/noanswer in the dial context? i am trying to find the damn patch for it>? |
| 21:13.52 | pc500 | delmar - It was just a file download, no keys, no mac, must be old. |
| 21:13.55 | CoaxD | delmar: I forget |
| 21:14.02 | pc500 | I have a quad T-1 card pulled out of some proprietary "clarent" voip phone system. |
| 21:14.09 | pc500 | Would be nice if it worked in *. |
| 21:14.13 | delmar | CoaxD, no we only need E1 if we wanna interface with the Telco... and as a small business home/office.. that will happen when hell freezes over |
| 21:14.20 | pc500 | 4 t1 ports on a full length pci card =) |
| 21:14.20 | CoaxD | delmar: Ahhh |
| 21:14.21 | delmar | CoaxD, the channel banks are T1 connection |
| 21:14.48 | delmar | pc500, droooohl |
| 21:15.00 | delmar | pc500, what brand? send here and ill test it for you hehe |
| 21:15.11 | CoaxD | shit, the quad T1 cards are cheap now |
| 21:15.11 | pc500 | delmar - I got a few, I might just do that :) |
| 21:15.16 | CoaxD | the clones are even cheaper |
| 21:15.23 | CoaxD | (but. i wouldnt buy those.) |
| 21:15.33 | delmar | pc500, damn freight here to NZ will kill you :( |
| 21:15.43 | delmar | pc500, so how many g29 licenses did u buy etc? |
| 21:15.57 | delmar | g729* even |
| 21:16.01 | pc500 | delmar - I think it came out of one of these: http://cgi.ebay.com/Clarent-Gateway-1200-VoIP-Server-Dual-P3-NMS-Audiocodes_W0QQitemZ300131408782QQihZ020QQcategoryZ1484QQrdZ1QQssPageNameZWD1VQQcmdZViewItem |
| 21:16.07 | pc500 | delmar - 1 |
| 21:16.31 | CoaxD | shit, the quad T1 clone cards built on the zaptel interface are $299 |
| 21:16.42 | delmar | CoaxD, USD? |
| 21:16.47 | CoaxD | delmar: yeah |
| 21:16.49 | pc500 | delmar - Infact it is one of those cards. |
| 21:16.51 | delmar | still more than my wife would let me spend :( |
| 21:17.00 | pc500 | Got 4 T-1s laying around? |
| 21:17.14 | CoaxD | delmar: Those are the clones, tho. I never advise those. digium will tell ya where to go when it dont work |
| 21:17.19 | pc500 | Why deal with a $299 card when the monthly recurring... is much higher. |
| 21:17.21 | delmar | pc500, lookin... trying to see what make/model it is |
| 21:17.27 | fujin | get a cisco |
| 21:17.40 | delmar | I never call Digium anymore |
| 21:17.52 | CoaxD | delmar: T1 is different. sometimes you need to |
| 21:17.53 | delmar | I think I know more than they do sometimes. |
| 21:18.11 | CoaxD | delmar: That isnt uncommon especially for niche knowledge |
| 21:18.17 | pc500 | delmar - Can I install ztdummy after the fact? Doe sit have to be installed manually? |
| 21:18.25 | CoaxD | delmar: Hard to get good people... (I work for a software company, man.) |
| 21:18.44 | delmar | I wanted their advise on a GOOD QoS solution... the RN300 from Ranchnetworks was an awesome concept.. dunno why they folded.... so I phoned Digium.... and emailed.. and they just said... " get a seperate DSL connection for VoIp.." |
| 21:18.46 | pc500 | There are tons of software engineers. Finding a good one, now that's another matter. |
| 21:18.47 | CoaxD | delmar: This is why you have 'tier1', 'tier2', 'tier3', etc. You first talk to the folks who don't know shit |
| 21:19.00 | CoaxD | i work in the 'folks who dont know shit' dept |
| 21:19.05 | delmar | still.. i appreciate Asterisk.. bloody awesome. |
| 21:19.16 | CoaxD | (I support the folks who dont know shit so they dont have to escalate to the 'we know some shit' dept.) |
| 21:19.22 | pc500 | delmar - QoS is easy under one condition: You consistently have a given bandwidth figure. If that part fails, your screwed. |
| 21:19.35 | CoaxD | pretty soon, i'm moving to the 'we know some shit' dept. |
| 21:19.41 | delmar | pc500, exactly.. and DSL is all over the place. |
| 21:19.50 | pc500 | delmar - You artificially cap yourself lower. |
| 21:20.08 | delmar | pc500, once they offer the ability to subscribe to multiple services hence more than one PVC here in NZ... it will rock |
| 21:20.28 | delmar | I have the new SpeedTouch / Thompson router.. ST608WL... |
| 21:20.33 | delmar | man the QoS in that just rocks... |
| 21:20.44 | delmar | but my wife's torrenting can still mess up calls.. |
| 21:20.46 | pc500 | delmar - Some cisco stuff now can prioritize what's the the queues/buffers based on flagging... so P2P could use all B/w but give 64k for a clean voip call wehn needed. BUt once it leaves your router you're screwed. If it's buffered up on your cable modem or ISP, nothing you can do. |
| 21:21.02 | delmar | but its the closest ive found.. even my Linux box acting as router doing QoS is not as good. |
| 21:21.37 | delmar | my next task tho.. is to play with all that P2P QoS stuff u can do on your linux router now. |
| 21:21.46 | pc500 | delmar - If your ISP works with you for QoS it's easy. But not happening due to cpu load on there router. Just tell them to keep the buffers small and deal with it. YOu can onyl really control the upstream. |
| 21:21.48 | delmar | that should keep our torrents from messing things :P |
| 21:22.06 | CoaxD | voip is a pain in the ass |
| 21:22.18 | CoaxD | if you need to RELY on it, use regulated carriers. |
| 21:22.27 | CoaxD | and dedicated links. |
| 21:22.34 | delmar | pc500, our Telco is anal. the entire DSL system is run by ONE Telco.... conflicted interests.. why should they want VoIp to work right? that hurts their Tolls business. |
| 21:22.39 | pc500 | SIP is good to the server, for business use you must use PRI to go out. |
| 21:22.43 | *** join/#asterisk zydrunas_ (n=zydrunas@24-119-29-130.cpe.cableone.net) |
| 21:22.58 | mvanbaak | why ? |
| 21:23.02 | pc500 | delmar - QoS on ISP router won't happen without expensive link (t1 or higher). |
| 21:23.17 | pc500 | mvanbaak - Reliability... in my opinion. |
| 21:23.19 | mvanbaak | I have QoS on default DSL |
| 21:23.27 | pc500 | mvanbaak - On your Side. Not on your ISPs side. |
| 21:23.27 | delmar | pc500, oh you wanna install ztdummy ? if u compiled and installed Zaptel, drop Asterisk, modprobe 'ztdummy' then start Asterisk again |
| 21:23.36 | mvanbaak | pc500: on ISP side as well |
| 21:23.38 | pc500 | delmar - Ok, so I can worry about it later, thanks. |
| 21:23.51 | delmar | pc500, yep. is this a home box? |
| 21:23.55 | pc500 | mvanbaak - ISPs don't normally QoS for VoIP (except maybe they're own). IT takes way too much cpu processing on their router. |
| 21:23.58 | pc500 | delmar - This one is, yes. |
| 21:24.02 | delmar | pc500, you dont tie into a pstn? |
| 21:24.06 | pc500 | no |
| 21:24.15 | mvanbaak | pc500: depends on the ISP |
| 21:24.26 | mvanbaak | if you have cheapass shitty ISP you are right |
| 21:24.32 | pc500 | mvanbaak - Very few. Infact I gather any major one doesn't. |
| 21:24.33 | mvanbaak | but a good ISP will offer it |
| 21:24.40 | mvanbaak | pc500: what country ? |
| 21:24.43 | pc500 | mvanbaak - USA |
| 21:24.54 | mvanbaak | here in .nl there are various ISP's that offer it |
| 21:24.59 | ManxPower | IF you have a dedicated T-1 to your ITSP, then you might be able to get QoS |
| 21:25.09 | pc500 | mvanbaak - It's pulling tooth to get them to modify their router config for QoS as I want it. But they won't touch it unless I pay for a T1 line. |
| 21:25.12 | delmar | pc500, there are two providers here in NZ now.. iTalk (slingshot) and VFX (Xnet/worls exchange) .. doing VoIp / DID's etc... im on Xnet for my DSL and use iTalk for DID.. and it works really well... because Xnet have some QoS happening for their VoIP product... |
| 21:25.15 | pc500 | Not for a $40/month line. |
| 21:25.25 | delmar | so essentially.. it pays to be on a DSL provider that has VoIP interests... |
| 21:25.48 | mvanbaak | pc500: all business DSL providers here config the DSL line as you want |
| 21:25.49 | pc500 | delmar - They don't always prioritize VOIP traffic, something they prioritize traffic going to/from a certain server :) Their own. |
| 21:25.55 | mvanbaak | QoS as you specify it |
| 21:25.58 | mvanbaak | no problem at all |
| 21:25.58 | delmar | but the physical network.. is all TelecomNZ so the QoS for VoIP at the RAN's and exchanges/dslam's is pita. |
| 21:26.20 | delmar | mvanbaak, wheres that? sounds awesome. |
| 21:26.20 | mvanbaak | even for the 128/128 lines |
| 21:26.23 | mvanbaak | .nl |
| 21:26.38 | delmar | ah yep. u guys are able to have more than 1 PVC too right? |
| 21:26.48 | delmar | so u can order more than 1 pipe on the physical DSL line. |
| 21:26.48 | mvanbaak | yup |
| 21:26.51 | mvanbaak | no problem at all |
| 21:26.57 | delmar | exactly what the ST608WL is setup for... |
| 21:26.59 | mvanbaak | 2 is default |
| 21:27.12 | delmar | soon, we will have 1 for internet and 1 small one for voice. |
| 21:27.26 | delmar | I think it was written into the legislation push the government is doing too. |
| 21:27.26 | mvanbaak | I have a default line here at home |
| 21:27.38 | mvanbaak | 2 8000/1024 dsl lines running on it |
| 21:27.48 | delmar | see.. our government has to force the Telco to do stuff to make things work right... pathetic really. |
| 21:27.50 | mvanbaak | both with QoS on my side _and_ ISP side |
| 21:28.04 | pc500 | mvanbaak - Now, get them to do multilink PPP for 16/2. that's never happening :) |
| 21:28.10 | mvanbaak | line is 35us$/month |
| 21:28.12 | delmar | mvanbaak, nice. AND you have hot chicks over there.. man u got it all :P |
| 21:28.21 | mvanbaak | pc500: they do |
| 21:28.29 | mvanbaak | I can get redundancy setup |
| 21:28.35 | mvanbaak | for another 10us$/month |
| 21:28.51 | mvanbaak | delmar: and legal canabis :))) |
| 21:28.55 | pc500 | mvanbaak - That's not multilink PPP. |
| 21:29.00 | delmar | mvanbaak, it all makes too much sense.. thats why it dont happen. |
| 21:29.09 | pc500 | mvanbaak - Will require expensive cisco equipment on your side. It's one bonded (not fault tolerant) link. |
| 21:29.13 | delmar | mvanbaak, yeah but our canabis is stronger than anywhere in the world :P |
| 21:29.21 | mvanbaak | pc500: cisco or OpenBSD |
| 21:29.30 | delmar | mvanbaak, dont go near it myself. linked to many health issues. |
| 21:29.41 | mvanbaak | or linux bonding |
| 21:29.42 | pc500 | mvanbaak - Could be done if ADSL cards existed, yes. |
| 21:29.56 | mvanbaak | pc500: adsl cards DO exist |
| 21:29.56 | pc500 | mvanbaak - Still can't send a single TCP flow over both lines. |
| 21:30.21 | mvanbaak | pc500: with bonding you can really bond them |
| 21:30.33 | mvanbaak | so packet1 will use line1 and packet2 will use line2 |
| 21:30.37 | mvanbaak | no problem with that |
| 21:30.42 | mvanbaak | I'm using that all the time |
| 21:30.52 | pc500 | mvanbaak - Nope, but each transfer is limited to 7 megabit, but two transfers can get 14. |
| 21:31.07 | pc500 | With MPPP on both sides, it's one big'old'pipe... hehe |
| 21:31.12 | pc500 | But with 7 megs, nobody really cares anymore. |
| 21:31.12 | delmar | pc500, no idea if that quad T1 is supported by Asterisk... from that link u provided. |
| 21:31.20 | mvanbaak | pc500: indeed |
| 21:31.25 | pc500 | delmar - Unlikely, it seems much other than digium isn't. |
| 21:31.47 | pc500 | delmar - Haveyou used the imap option? Is it for storing/retreiving voicemail? |
| 21:31.49 | mvanbaak | pc500: it all depends on the remote end |
| 21:31.56 | pc500 | delmar - Anyone come up with a usable exchange integration method? |
| 21:31.56 | delmar | the entire box is buy now .. for the price of a digium T1 card new. |
| 21:32.03 | pc500 | mvanbaak - Yes, multilink PPP, processor intensive. |
| 21:32.15 | delmar | pc500, imap option where? |
| 21:32.18 | pc500 | delmar - Ya. |
| 21:32.26 | pc500 | delmar - IT's under voice build options, make menuselect. |
| 21:32.42 | mvanbaak | pc500: not that intensive. I mean, the current junipers come with CPU's more powerfull then my laptop |
| 21:33.28 | mvanbaak | if your ISP is cheapass they will use stuff that is outdated and they wont be able to provide good services |
| 21:33.45 | mvanbaak | but here in .nl most ISP's use really great stuff in their network |
| 21:33.57 | pc500 | mvanbaak - It will bring 2800 router to it's knees. Combined with the overhead of the wic-1-adsl interface and small packet size/deframentation. A lot of the Juniper stuff is more powerful. |
| 21:34.38 | pc500 | mvanbaak - Here in the US, it's typical for ISPs to refuse to do such services. Especially for $80/month. |
| 21:34.46 | pc500 | There's just no money in it. |
| 21:34.53 | mvanbaak | 80/month is a highend line here |
| 21:34.59 | pc500 | That's 2 dsl lines here. |
| 21:35.11 | delmar | pc500, no idea... looks interesting. |
| 21:35.21 | mvanbaak | my 8000/1024 kbps dsl line is 30 us$/month |
| 21:35.23 | pc500 | delmar - Why would you want to embed modules? |
| 21:35.55 | pc500 | Nice -- MoH in native formats for better sound quaility :) |
| 21:35.58 | delmar | pc500, i wouldn't make something modular only to embed it... |
| 21:36.06 | ManxPower | speaking of NL, I am interested in living in NL for 2-3 months this winter and am interested in housing. If anyone has leads or suggestions, please /msg me |
| 21:36.55 | mvanbaak | ManxPower: where in .nl ? |
| 21:36.58 | delmar | ManxPower, head to the pub when u land and score some young wench and have her take you home to be her bedroom slave for 2-3months. |
| 21:37.08 | ManxPower | I figure that if I'm going to be cold and miserable, I might as well be cold and miserable in .NL or .BE instead of on top of a mountian in Alabama |
| 21:37.35 | ManxPower | mvanbaak: doesn't really matter all that much. as I'll be working remotely for customers in USA. |
| 21:37.42 | mvanbaak | ah |
| 21:37.43 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 21:37.51 | ManxPower | And (from an american point of view) EVERYTHING is very close. |
| 21:37.52 | mvanbaak | well, you can stay at my place for a couple of weeks |
| 21:38.23 | ManxPower | mvanbaak: I was more interested in a place of my own, cheap apartment or something like that. |
| 21:38.32 | mvanbaak | 2-3 months is too much to stay at my place |
| 21:38.48 | ManxPower | for a few weeks I can stay with a friend of my grandmothers -- but any longer than that it not workable |
| 21:39.24 | ManxPower | same goes for most anyone I might stay with. Staying with someone for a few weeks would be good for looking for a short term apartment |
| 21:40.14 | ManxPower | Belgium is also fine, BTW. |
| 21:40.21 | mvanbaak | ManxPower: I'll ask around |
| 21:40.29 | ManxPower | mvanbaak: Thanks! |
| 21:40.39 | mvanbaak | we live in a place where there's a lot of cheap houses |
| 21:40.53 | mvanbaak | so maybe I can find something you can get for a couple of monts |
| 21:40.53 | delmar | ManxPower, staying with someone for a couple of weeks, where you spend most time at the house working remotely, can get tedious. different if its a couple of weeks for someone on holiday, heading out doing the tourist thing or going out drinking with the gurls :P |
| 21:41.16 | pc500 | delmar - Roomates only work if they're never home :) |
| 21:41.23 | mvanbaak | ManxPower: private chat is logged :) |
| 21:41.38 | pc500 | ManxPower - Bring your own internet :) |
| 21:41.39 | pc500 | hehe |
| 21:42.04 | mvanbaak | I'll be gone from 8AM till 6PM |
| 21:42.37 | mvanbaak | I live in a wooden holiday home |
| 21:42.43 | pc500 | ? |
| 21:42.46 | pc500 | What's that? :) |
| 21:42.49 | mvanbaak | maybe this park has more for rent |
| 21:42.58 | mvanbaak | pc500: lemme lookup a nice pic of my house |
| 21:43.00 | pc500 | Short term rentals are hard though... no landlords want to deal with it. |
| 21:43.17 | *** join/#asterisk Stridernzl (n=neville@125-239-179-163.jetstream.xtra.co.nz) |
| 21:43.18 | delmar | yeah who would. |
| 21:43.30 | pc500 | Lots of vacancy, move in, move out, etc. |
| 21:43.34 | pc500 | If they do, they charge extra for it. |
| 21:43.46 | pc500 | delmar - Can you run a 5 line office with a pentium 2 or 3 asterisk box? |
| 21:44.03 | delmar | pc500, p3 sure. |
| 21:44.09 | pc500 | delmar - I didn't know if a box running at heavier loads affects quality at all. |
| 21:44.18 | pc500 | Even if free cpu is available :) |
| 21:44.19 | mvanbaak | pc500: http://images.vanbaak.info/displayimage.php?album=36&pos=3 |
| 21:44.22 | mvanbaak | that's the outside |
| 21:44.27 | ManxPower | delmar: Exactly! |
| 21:44.34 | ManxPower | which is why I want a place of my own |
| 21:44.52 | mvanbaak | http://images.vanbaak.info/displayimage.php?album=36&pos=15 |
| 21:44.56 | pc500 | like a cabin in the woods :) |
| 21:44.56 | mvanbaak | that's my livingroom |
| 21:44.57 | pc500 | Is that you? |
| 21:45.12 | ManxPower | I actually DO live in a cabin in the woods on the top of a mountian |
| 21:45.29 | mvanbaak | pc500: the one with the yellow shirt |
| 21:45.31 | mvanbaak | that's me |
| 21:45.34 | pc500 | ManxPower - except the high speed internet problem... and the woods. |
| 21:45.53 | ManxPower | delmar: Also since I'mm be mostly working on USA Central time (GMT/UTC - 6)...... |
| 21:46.01 | mvanbaak | pc500: this is me: http://images.vanbaak.info/displayimage.php?album=36&pos=10 |
| 21:46.09 | delmar | yeah of course.. TZ difference etc. |
| 21:46.34 | pc500 | throwing knifes? |
| 21:46.48 | mvanbaak | lol |
| 21:46.53 | mvanbaak | bbq-ing |
| 21:47.00 | delmar | the shrine ! |
| 21:47.12 | pc500 | Looks like you're holding onto the one by the blade... probably an optical illusion. |
| 21:47.22 | ManxPower | I'm a bit....eccentric anyway |
| 21:47.37 | pc500 | And one of them fruity machines :) |
| 21:47.40 | pc500 | good choice... |
| 21:47.40 | mvanbaak | pc500: holding a knife and a kitchen lighter |
| 21:47.41 | ManxPower | what is .NL's UTC offset? |
| 21:47.49 | mvanbaak | Sun Jul 29 23:47:49 CEST 2007 |
| 21:47.53 | mvanbaak | <PROTECTED> |
| 21:47.55 | pc500 | go to bed. |
| 21:47.58 | delmar | mvanbaak, wheres the pic of your.. plantation :P |
| 21:48.02 | pc500 | hehe |
| 21:48.10 | mvanbaak | lol delmar |
| 21:48.16 | mvanbaak | I dont grow it myself |
| 21:48.16 | pc500 | Are a lot of houses like that in your country? |
| 21:48.38 | mvanbaak | pc500: depends on where in .nl you are. there are a lot in my area |
| 21:48.53 | mvanbaak | a lot of germanz and french ppl are spending their holidays here |
| 21:49.01 | mvanbaak | legally I'm not allowed to live here |
| 21:49.04 | pc500 | why? |
| 21:49.04 | delmar | mvanbaak, what city/area are u in ? |
| 21:49.15 | mvanbaak | it's a holiday cabin |
| 21:49.20 | pc500 | What's that mean? |
| 21:49.29 | mvanbaak | it's not meant for day-to-day family to live in |
| 21:49.33 | mvanbaak | but it's cheap |
| 21:49.42 | mvanbaak | I payed 28.000 us $ for it |
| 21:49.51 | pc500 | as in, own? |
| 21:49.54 | mvanbaak | yeah |
| 21:49.57 | pc500 | not bad, if the rent isn't to bad. |
| 21:50.02 | mvanbaak | no rent |
| 21:50.04 | mvanbaak | it's mine |
| 21:50.06 | delmar | no rent lol |
| 21:50.16 | mvanbaak | I own the place |
| 21:50.21 | delmar | thats the way to go |
| 21:50.27 | delmar | no lease on the land ? |
| 21:50.31 | mvanbaak | nope |
| 21:50.40 | mvanbaak | 300m2 of land |
| 21:50.48 | mvanbaak | 305 actually |
| 21:50.59 | delmar | just the rates, insurance, and mortgage if u have one.. but for $28k im sure that wouldnt be the case. |
| 21:51.15 | mvanbaak | mongage is like 300 us$/month |
| 21:51.19 | pc500 | Thoguht there would be upkeep, etc :) |
| 21:51.23 | mvanbaak | insurance is 61$/month |
| 21:51.23 | pc500 | How are wages there compared to the US? |
| 21:51.33 | delmar | mvanbaak, thats a small plot tho |
| 21:53.06 | mvanbaak | it's cheap |
| 21:53.06 | delmar | mvanbaak, got a 1000m2 section here. |
| 21:53.06 | mvanbaak | that's why a lot of young ppl are living in places like this |
| 21:53.06 | pc500 | Do I install the sampels (make samples) or not? |
| 21:53.06 | mvanbaak | a normal hous will be like 250.000$ |
| 21:53.06 | pc500 | The ones a long time ago were terrible. |
| 21:53.06 | delmar | pc500, i wouldnt if u want to keep your existing stuff. |
| 21:53.06 | ManxPower | I would have to find a place for $500 or less per month. It doesn't have to be nice, but it does need internet service available |
| 21:53.06 | pc500 | delmar - Already removed it. |
| 21:53.12 | delmar | ManxPower, get a place next door or close to mvanbaak and run a cable or wireless :P |
| 21:53.24 | delmar | pc500, then do wahtever :P |
| 21:53.37 | mvanbaak | ManxPower: I'll ask around this week |
| 21:53.40 | pc500 | delmar - Ok, I remeber the old ones were terrible (And set up lots of extensions, non-commented out stuff). |
| 21:54.07 | ManxPower | mvanbaak: Thanks. I'm thinking about dec, jan, feb or jan, feb, mar |
| 21:54.10 | ManxPower | gotta run! |
| 21:54.14 | mvanbaak | delmar: my neighbours house has UTP to my switch :) |
| 21:54.26 | pc500 | mvanbaak - burried? :) |
| 21:54.29 | delmar | mvanbaak, hah. same here :P |
| 21:54.44 | mvanbaak | pc500: like 15cm deep yeah |
| 21:54.50 | pc500 | split the costs, what the hell =) |
| 21:54.54 | mvanbaak | indeed |
| 21:54.58 | pc500 | mvanbaak - Why not wireless? |
| 21:55.01 | *** join/#asterisk matt_ (n=matt@2001:770:168:1:220:edff:feb4:7c9d) |
| 21:55.04 | delmar | yeah i just sliced into the grass and ticked it under a little bit so the mower wont run it over. |
| 21:55.17 | mvanbaak | pc500: because the house has a steel skeleton. it acts like cage of ferraday |
| 21:55.24 | mvanbaak | bad wireless signal outside |
| 21:55.31 | pc500 | mvanbaak - STill something that close out to be workable. |
| 21:55.32 | mvanbaak | delmar: same here |
| 21:55.46 | pc500 | delmar - How long is your cable run? |
| 21:55.55 | mvanbaak | pc500: my 54mbit apply airport gave my neighbour a 1kbps link |
| 21:55.58 | delmar | could probably get reception off a web piece of string if its close enuf. |
| 21:56.07 | delmar | pc500, not more than 50m. |
| 21:56.22 | mvanbaak | s/apply/apple/ |
| 21:56.30 | delmar | most of that is distance under the houses . |
| 21:56.41 | pc500 | What security documents am I supposed to be reading for asterisk? |
| 21:56.51 | pc500 | delmar - My closest neighbor is beyond cat5 reach =) |
| 21:56.58 | pc500 | Just hope it doesn't freeze or get eaten... hehe |
| 21:57.15 | delmar | pc500, what distance? |
| 21:57.46 | mvanbaak | pc500: winter is fine. I had to redo the CAT5 once because father of neighbour did not know about the cable and dug out a tree |
| 21:57.52 | pc500 | delmar - half a football field? |
| 21:57.53 | mvanbaak | of course he cut the cable |
| 21:58.02 | delmar | pc500, no problem. |
| 21:58.16 | mvanbaak | so we had to redo the cat5 |
| 21:58.18 | mvanbaak | ah well |
| 21:58.23 | pc500 | delmar - IT's about an acre away or so. |
| 21:58.27 | mvanbaak | I get 100m cat5 for 15$ |
| 21:58.35 | delmar | pc500, should be able to get 100mbit at 350m. |
| 21:58.35 | pc500 | delmar - It's out of spec though, isn't 100m max? |
| 21:58.46 | delmar | pc500, but for all the effort... wireless would be the answer. |
| 21:58.46 | mvanbaak | 1000m max |
| 21:58.52 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 21:58.55 | mvanbaak | shielded |
| 21:58.56 | pc500 | for 100baseT? |
| 21:59.03 | mvanbaak | I think |
| 21:59.08 | mvanbaak | it's a bit rusty |
| 21:59.14 | mvanbaak | I use fiber nowedays |
| 21:59.44 | pc500 | then you need conduit :) |
| 22:00.18 | pc500 | 100 meters max (http://en.wikipedia.org/wiki/Category_5_cable) |
| 22:00.20 | delmar | fibre is the way to go for those sorta distances for sure.. now that its cheap enough but.. for something that far but within 1km .. i would just whip up a couple of cheap AP's on the roof.. maybe some little 8dB or 15dB antenna's. |
| 22:00.45 | pc500 | delmar - Yeah really, why bother with the cable :) |
| 22:01.00 | mvanbaak | indeed |
| 22:01.23 | delmar | pc500, dunno about the max being 100m... I have had more than 350m 100mbit no problems with cat5eS. |
| 22:01.56 | delmar | 10mbit will go even further. |
| 22:02.01 | CoaxD | heh. this lumenvox speech recog stuff looks promising |
| 22:02.58 | delmar | totally off channel topic (but i cant find answers) .. anyone good with vTiger CRM ? |
| 22:03.23 | pc500 | Where's a good source of cheap origination? |
| 22:03.46 | pc500 | Not a fly by night gone next week company :) |
| 22:04.19 | CoaxD | pc500: There exists both in one package? |
| 22:04.34 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 22:04.40 | pc500 | CoaxD - huh? |
| 22:04.48 | CoaxD | <pc500> Where's a good source of cheap origination? |
| 22:04.48 | CoaxD | <pc500> Not a fly by night gone next week company :) |
| 22:04.57 | delmar | pc500, u cant get cheap AND not fly by night. |
| 22:04.59 | pc500 | CoaxD - Ahh, I get it :). No seriously, something that's fair, but the number won't disappear in 6 months. |
| 22:05.11 | CoaxD | Every single time i give nufone another try.. |
| 22:05.13 | CoaxD | Every SINGLE time... |
| 22:05.25 | pc500 | nufone had some problems, can't recall what, but I ditched them. |
| 22:05.36 | pc500 | voipjet.com has cheap termination and has never had any problems with me. |
| 22:05.43 | pc500 | I need something like that for origination. |
| 22:05.47 | CoaxD | pc500: Yeah, well, this damn 800 number i have from them is screwed. |
| 22:06.00 | CoaxD | i try to do their user login thing, and it blows up |
| 22:06.02 | delmar | pc500, yeah john is quite customer focused... if a bit of a strange bloke at times. |
| 22:06.08 | CoaxD | cant even submit a fucking ticket |
| 22:06.09 | pc500 | Termination is easier, because you can switch providers anyways in 10 seconds. orgiinaiton has your phone number. |
| 22:06.20 | pc500 | delmar - who's john, voipjet? |
| 22:06.29 | delmar | yep |
| 22:06.30 | CoaxD | i remember when voipjet got started |
| 22:06.35 | CoaxD | i helped him build out his SMTP farm. *chuckle* |
| 22:06.38 | pc500 | delmar - YEah, no support unless you do at least business voulmes in calls (he hates us home users), but it still works. |
| 22:06.46 | pc500 | smtp? |
| 22:06.49 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 22:07.11 | CoaxD | yeah. he understood voip real good. but didn't know how to admin an email solution for shit. *chuckle* |
| 22:07.16 | delmar | i wouldnt say that.. he never seemed to have any issues supporting us, but I don't use voiojet anymore. |
| 22:07.23 | delmar | voipjet* even. |
| 22:07.26 | SwK_ | hmm since you are talking about term... know anyone with t38 term to canada? |
| 22:07.53 | pc500 | delmar - never replies, sets all the servers to $25 account minimum except one, etc. |
| 22:08.07 | delmar | pc500, yeah I noticed that of late. |
| 22:08.44 | delmar | Pc500, and I had issues getting him to reply on a few things until i got my wife to email him directly (cuz it was her account anyway).. then he replied.. it was like he had some kinda whitelist going on... |
| 22:09.03 | mvanbaak | lol |
| 22:09.15 | mvanbaak | SMTP is one of the easiest things to setup |
| 22:09.24 | mvanbaak | maybe DNS is more simple |
| 22:09.27 | mvanbaak | but it's close |
| 22:09.30 | pc500 | dns is a PITA. |
| 22:09.37 | pc500 | not hard, but a pain |
| 22:09.51 | mvanbaak | why ? |
| 22:09.52 | pc500 | any reason to not enable reinvites (other than you can't record calls)? |
| 22:09.58 | pc500 | mvanbaak - Just bind in general. |
| 22:09.59 | delmar | mvanbaak, yeah.. man apt-get :P |
| 22:10.05 | mvanbaak | lol delmar |
| 22:10.18 | mvanbaak | pc500: reinvites & nat != go |
| 22:10.25 | pc500 | mvanbaak - It used to always have security patches too, but not as of late. |
| 22:10.36 | pc500 | mvanbaak - yeah, but more latency. Grin. |
| 22:10.50 | mvanbaak | yup |
| 22:10.57 | delmar | nah u dont want reinvites when nat is going on. big issues otherwise. |
| 22:11.06 | mvanbaak | indeed |
| 22:11.22 | SwK_ | reinvites + nat not nessecarily == bad... |
| 22:11.23 | delmar | i wish more gear would support IAX. |
| 22:11.31 | SwK_ | how else are you gonna run t38 |
| 22:11.34 | delmar | IAX doesn't care about nat. it just works. |
| 22:11.40 | mvanbaak | reinvite means the rtp stream is going from endstation to endstation directly |
| 22:11.41 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:11.48 | SwK_ | mvanbaak, no it doesnt |
| 22:11.50 | delmar | t38 is bad. |
| 22:11.51 | mvanbaak | that wont work when nat is involved |
| 22:12.06 | SwK_ | mvanbaak, reinvite simply means change in media stream |
| 22:12.14 | mvanbaak | ok ok |
| 22:12.32 | mvanbaak | _MOST_ of the time, reinvite means station to station rtp |
| 22:12.38 | mvanbaak | you dont want that with nat |
| 22:12.45 | SwK_ | no |
| 22:13.03 | SwK_ | it means you are changing the media some how... |
| 22:13.43 | SwK_ | that includes changing from codec X to codec y, as well as redirecting the media to different endpoint... and most of the time is totally dependant on your system |
| 22:14.06 | delmar | and usually works nicely... i see sip handoffs on the console all the time... |
| 22:14.08 | mvanbaak | codecs are negotiated at call setup |
| 22:14.11 | delmar | you don't even notice its happened. |
| 22:14.21 | mvanbaak | most of the time |
| 22:14.24 | pc500 | delmar - WIthout the make sapmles, is there any base configs or no? |
| 22:14.24 | SwK_ | 75% + of the calls I do are re-invited because they are fax... and fax calls always start in ulaw, and switch to t38 mode after the termination end answers |
| 22:14.37 | mvanbaak | ah |
| 22:14.40 | delmar | yeah i thought a sip handoff would only occur if the SAME codecs could be negotiated. |
| 22:14.43 | mvanbaak | we dont do fax |
| 22:14.43 | SwK_ | pc500, no... |
| 22:14.52 | delmar | pc500, just do make samples. |
| 22:15.05 | delmar | fax can sodd off |
| 22:15.14 | pc500 | delmar - Gotcha, my config is fairly polluted. |
| 22:15.15 | delmar | anyone still using a fax machine can also.. sodd off. |
| 22:15.17 | pc500 | Fax must die! |
| 22:15.23 | mvanbaak | our ITSP offers fax handling directly on their E1 interfaces |
| 22:15.28 | mvanbaak | so we let them handle it |
| 22:16.04 | delmar | no more talk of t38 or faxes... or... |
| 22:16.10 | delmar | I will upload a virus to your mothership ! |
| 22:16.13 | pc500 | Seriously, email & PDF/TIF is much better. Unless your industry must have it for legal reasons. |
| 22:16.17 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 22:16.19 | pc500 | It's 25 years old and antiquated. |
| 22:16.26 | NoIdea465 | I want to ofice-space my fax |
| 22:16.42 | mvanbaak | office space++ |
| 22:16.42 | delmar | pc500, more than that. Captain Cook and Able Tasman both used my fax machine.. I swear. |
| 22:16.49 | SwK_ | delmar: until someone makes a fucking machine where a) it has an ethernet port and b) has an easy to use UI like a faxmachine, faxes will not die... its too deeply entrenched in business |
| 22:17.32 | pc500 | SwK_ - Seriously, I know. Maybe a e-mail button is a start, or a way to map "fax" numbers for electronic delivery. |
| 22:17.33 | delmar | SwK_, the all-in-one machines for $199 all have ethernet now. |
| 22:17.38 | pc500 | Like a database of some sort. |
| 22:17.50 | delmar | NoCarrier, pice when u do plz. |
| 22:17.54 | delmar | pic* |
| 22:18.07 | SwK_ | delmar, maybe so, but non of them work like a fax machine |
| 22:18.17 | SwK_ | you still have to go to a computer and do something |
| 22:18.19 | mvanbaak | fax == evil |
| 22:18.34 | SwK_ | fax == profitable |
| 22:18.52 | delmar | SwK_, they are all-in-one... you can send a fax from your desktop, or lift the lid, put page in.. dial the number.. and hit the spam button |
| 22:19.03 | delmar | cuz thats about all they are good for ... sending spam. :P |
| 22:19.12 | SwK_ | delmar but it still wont send the fax over IP when you push in the number and hit send |
| 22:19.26 | SwK_ | until that time, 'faxing' will still be around |
| 22:19.43 | delmar | SwK_, oh i see what u are saying... yeah.. need some kinda Enum type number to IP translation |
| 22:20.16 | delmar | SwK_, scan-to-email is a feature being rolled out in many products now. |
| 22:20.23 | SwK_ | hell just a 'faxmachine' with t38 and ethernet would change things |
| 22:20.54 | SwK_ | delmar, scan2email has been around forever, but its still more complex then just dialing a number and hitting send |
| 22:21.43 | delmar | SwK_, not so much. I was in a copy store that had one of the new Cannon machines... it scans.. u enter the address and press send.. same as you would enter the number and press Start. |
| 22:22.16 | delmar | sure is getting better |
| 22:22.36 | delmar | but i agree.. fax machines are gonna be around way longer than they should be. |
| 22:23.44 | Nugget | easy to say with confidence because it's already true. :) |
| 22:25.02 | pc500 | delmar - Where do you get the free g729 for home use? :) |
| 22:25.22 | delmar | pc500, google is your friend :P |
| 22:25.28 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 22:25.34 | pc500 | delmar - Yeah, the auto-pdf things that take e-mails work good. People don't know how to send, most people know how to open a pdf |
| 22:26.09 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:26.10 | delmar | pc500, ill dig out the link .. sec |
| 22:26.21 | pc500 | delmar - how bad is music on hold without a timing device (compared to ztdummy)? |
| 22:26.26 | pc500 | thx |
| 22:26.39 | delmar | pc500, actually its fine. |
| 22:27.00 | joaop | if a have a local asterisk server runing and working with login/pass in xlite for example, can someone connect here outside of my home network(internet) ? i have only 5060 opened |
| 22:27.06 | delmar | pc500, i dont even have a timing source on my colo box and play ivr's and music on it without issues. |
| 22:27.36 | delmar | joaop, would need more than that going on for the media path to work. |
| 22:28.13 | pc500 | yikes, they split set callerid to set callerid ANI and ALL :( |
| 22:28.23 | delmar | yup |
| 22:28.31 | mvanbaak | zzzzzzzzzzzzzzzzzzzz |
| 22:28.36 | delmar | messed up my callerID good when i upgraded |
| 22:28.55 | delmar | mvanbaak, dude your cat is standing on your z key |
| 22:28.55 | joaop | delmar, i meant use xlite over my network with my ip user/pass, it will not work ? |
| 22:29.58 | delmar | joaop, if they know the user/pass and u have that port open they can authenticate and attempt calls but i doubt much else would happen if thats the only port open. |
| 22:30.02 | delmar | wouldnt be any audio |
| 22:30.21 | pc500 | I'll be here all day! |
| 22:30.28 | pc500 | Next error: [Jul 29 19:10:47] WARNING[11722]: pbx.c:1797 pbx_extension_helper: No application 'SetVar' for extension (d |
| 22:30.29 | pc500 | :) |
| 22:30.47 | delmar | its not hard. u will get there :P |
| 22:31.19 | pc500 | delmar Now I have to rewrite this :( |
| 22:31.19 | pc500 | exten => _1NXXNXXXXXX,2,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}) |
| 22:31.19 | pc500 | exten => _1NXXNXXXXXX,3,Monitor(wav,${CALLFILENAME},m) |
| 22:31.22 | mvanbaak | delmar: 6am is wakeup time |
| 22:32.00 | pc500 | delmar - Any ideas where to start? |
| 22:32.01 | delmar | pc500, Set |
| 22:32.13 | pc500 | thx |
| 22:32.53 | delmar | the site i used to get the codec from appears to be gone |
| 22:33.18 | pc500 | delmar - I'm geting a million of there: [Jul 29 19:13:45] WARNING[12336]: chan_sip.c:3548 sip_write: Asked to transmit frame type 4, while native formats is 0x100 (g729)(256) read/write = 0x100 (g729)(256)/0x100 (g729)(256) |
| 22:33.25 | pc500 | Like, thousands, 1 per packet. |
| 22:33.30 | pc500 | IS that called no g729 installed? :) |
| 22:33.38 | pc500 | Seems there'd be a better way to log it... |
| 22:34.16 | mvanbaak | looks like you dont have g729 licenses :) |
| 22:34.19 | delmar | pc500, not sure what thats all about dude. |
| 22:34.32 | delmar | that would be my guess.. no license. |
| 22:34.53 | delmar | u would think it would say.. " get a license u knob" or something useful |
| 22:34.59 | mvanbaak | lol |
| 22:35.08 | pc500 | yeah, really, because I guess it loads the codec but won't use it without license. |
| 22:35.16 | *** join/#asterisk mltlnx (n=mltlnx@cpe-69-203-218-147.nyc.res.rr.com) |
| 22:35.39 | delmar | pc500, so yeah.. no more SetVar .. its just Set |
| 22:35.53 | delmar | ie Set(CALLERID(all)=KnobHead <123456> |
| 22:36.02 | mltlnx | Hey there, Im using a .call file to place calls.....Is there a way to display the called number on the console? |
| 22:36.53 | delmar | mltlnx, something like exten => s,n,NoOp(${CALLERID(all)}) |
| 22:37.11 | delmar | drop that line in there somewhere. |
| 22:37.16 | mltlnx | thanks ill try now. |
| 22:37.59 | pc500 | Anyone use a cisco 7960 ip phone? |
| 22:38.14 | pc500 | EVen with no tftp server, it tries to download a config for 3 minutes on startup. |
| 22:38.23 | pc500 | Before using the manual config... it's annoying |
| 22:38.47 | delmar | pc500, hit it |
| 22:39.04 | pc500 | rofl |
| 22:39.04 | delmar | its always fun to bash expensive cisco stuff |
| 22:39.12 | pc500 | Picked them off craigslist for $50/pop :) |
| 22:39.21 | JT | that probably only costs $50 to make |
| 22:39.36 | pc500 | delmar - NEvermind the fact they're worth $100 each on ebay... hehe |
| 22:40.02 | delmar | pc500, only $100? finally.. they have come down in price. |
| 22:40.10 | pc500 | Yeah, for a 7960 |
| 22:40.27 | pc500 | delmar - Ok, next topic: music on hold doesn't work right. the caller hears music cut in and out. possible silence supression problem. |
| 22:40.29 | delmar | had to happen.. the way government departments have been soaking them up, then splashing out on the next model as soon as it comes out... |
| 22:40.30 | pc500 | Any ideas? |
| 22:40.46 | JT | what's the point of buying a cisco from ebay other than testing? |
| 22:40.51 | pc500 | delmar - That's the other thing I wonder, phone systems need 15 years of use, I wonder if IP systems wil get that. |
| 22:40.52 | delmar | pc500, the user is connecting how? |
| 22:41.09 | pc500 | JT - Because it was cheap as anything else on ebay, and they are well built. |
| 22:41.13 | delmar | JT, yup. im not interested in cisco. Polycom beat the pants off them. |
| 22:41.20 | JT | pc500: polycom are better phones |
| 22:41.21 | pc500 | delmar - phone > asterisk > calling out via voipjet to my cell. |
| 22:41.30 | delmar | I have a poly501 here... not a cheap phone .. speaker phone is just amaizing. |
| 22:41.31 | JT | cisco sip is junk |
| 22:41.45 | pc500 | their sip firmware isn't the greatest, no, but it was cheap for me. |
| 22:41.47 | delmar | pc500, and in that scenario .. whos on hold? |
| 22:41.59 | pc500 | delmar - the phone places the cell phone on hold. |
| 22:42.17 | pc500 | delmar - asterisk states: |
| 22:42.18 | delmar | pc500, i advise you replace the cell phone test user .. with a landline one. |
| 22:42.19 | pc500 | <PROTECTED> |
| 22:42.25 | delmar | pc500, for more accurate test results. |
| 22:42.31 | pc500 | delmar - Can I call you? |
| 22:42.35 | pc500 | I don't have a landline. |
| 22:42.43 | pc500 | I doubt it's the cell though. |
| 22:42.50 | pc500 | never know though |
| 22:43.24 | delmar | pc500, could do I spose... my wife is crashed out on the couch here tho and I dont wanna wake her. |
| 22:43.38 | pc500 | :) turn ringers off, hehe. |
| 22:43.40 | delmar | my constant typing is probably not good either haha |
| 22:43.51 | JT | pc500: you doubt it's the cell... on what grounds? |
| 22:44.01 | pc500 | JT - other music on hold on the cell has worked in the past. |
| 22:44.11 | pc500 | JT - And it happened before the upgrade too. |
| 22:44.16 | delmar | pc500, ok lets try direct .. your * box to mine .. as well as a route via voipjet to my NZ did. will pm some info in a minute |
| 22:44.22 | JT | music on hole to cellphones is hit and miss. |
| 22:44.22 | pc500 | ok. |
| 22:44.32 | JT | depends on the type of music and amplitude |
| 22:44.33 | pc500 | JT - sounds like #$@%, yes, but uslaly works. |
| 22:44.44 | delmar | pc500, im thinking the cell could be 'choppy' .. need to remove that from the picture. |
| 22:44.44 | JT | as well as anything your itsp does |
| 22:45.01 | JT | yeah, and not having a landline, how could you ;) |
| 22:45.37 | pc500 | ya, agreed |
| 22:45.43 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 22:45.43 | pc500 | JT - rofl, why have a landline? |
| 22:45.51 | JT | gee i wonder |
| 22:45.54 | JT | reliability |
| 22:49.04 | pc500 | JT - I got a cell phone. DOn't trust your asterisk box? |
| 22:50.23 | JT | definitely not |
| 22:50.32 | JT | cell phones are nowhere near as reliable as land lines |
| 22:50.49 | JT | i mean connected to a real phone |
| 22:51.15 | JT | and itsps are notoriously unreliable, especially considering you connect to them over the public Internet |
| 22:52.11 | JT | and most of them have no idea what they're doing |
| 22:56.17 | crimethinker | my cell phone was reliable on the 12th floor of the resort hotel I was in last week. but I would never trust it in my house. |
| 22:57.20 | crimethinker | ironically, the hotel's phone (in the room) was dead, and I had to call the front desk with my cell phone to ask them to fix it. |
| 22:58.03 | JT | maybe they were running asterisk ;) |
| 22:58.07 | *** join/#asterisk bmg505 (n=leon@196.209.176.104) |
| 22:58.13 | delmar | lol |
| 22:58.21 | delmar | they upgraded to 1.4 :P |
| 22:58.24 | delmar | and it broke |
| 22:58.43 | crimethinker | they're running the same Nortel horror as Ohio State - I recognize the voicemail's voice |
| 22:58.53 | Nugget | hah |
| 22:59.56 | crimethinker | anyway, room service wouldn't take orders from my cell, mostly because they didn't have a DID number, but also because they use the caller ID to bill, apparently. |
| 23:01.11 | crimethinker | funny you should mention upgrading to 1.4; I did today, and it's running just dandy. |
| 23:02.42 | lesouvage | JT: it's the custumers choice. People have to get used to the idea that telephony comes in different flavours from sometime up and running to enterprise class |
| 23:03.11 | JT | lesouvage: what? |
| 23:03.16 | JT | sentence does not parse |
| 23:05.09 | *** join/#asterisk ido (n=ido@unaffiliated/ido) |
| 23:06.03 | Rickalicious | Is there a way to have some console/dsp extensions auto answer, and others not? |
| 23:06.24 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:06.35 | Rickalicious | I am trying to use paging as well as ringing with the same hardware and through the same soundcard, ut through different extensions.. |
| 23:08.01 | JT | lesouvage: what were you trying to say? |
| 23:08.01 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) |
| 23:08.15 | *** join/#asterisk mltlnx (n=mltlnx@cpe-69-203-218-147.nyc.res.rr.com) |
| 23:09.57 | lesouvage | JT: That there are companies out there who are doing there very best with all the expertise availleble to offer high quality voip based on Asterisk but that this services has a price. In the past you just had one flavour: a landline offered by a monopilist |
| 23:10.13 | fujin | fail at english |
| 23:10.27 | JT | not everyone has monopolies |
| 23:10.33 | fujin | It's not really hard to create enterprise-grade telephony with Asterisk |
| 23:10.40 | fujin | in personal experience, anywa. |
| 23:10.43 | fujin | s/anywa/anyway/ |
| 23:10.51 | fujin | ^5 jbot |
| 23:11.02 | JT | and they can do their very best as much as they want, they're still not responsible for all the elements of your Internet connection |
| 23:11.10 | JT | fujin: it's pretty hit and miss with VoIPoI |
| 23:11.15 | fujin | that's very true |
| 23:11.24 | joaop | off.. there isnt sylpheed claws gtk2 in debian testing? |
| 23:11.26 | fujin | unless you can control your upstream providers, qos, etc |
| 23:11.43 | JT | also, your asterisk server needs to stay up during a power outage |
| 23:12.02 | fujin | heh |
| 23:12.07 | lesouvage | JT: there are compnies using free voip services like voipbuster or gizmo while using a cheap internet connection and discover that there are problems from time to time. f it is to good to be true most of the time it is not true. |
| 23:12.09 | fujin | HA asterisk :D |
| 23:12.27 | fujin | I'm going to do a writeup on ha-asterisk on voip-info at some stage, the information I found on there was very out of date |
| 23:12.33 | JT | haha, voipbuster or gizmo for business |
| 23:12.36 | fujin | lol. |
| 23:12.37 | delmar | pc500, its most certainly a silence suppression issue |
| 23:12.42 | JT | fujin: need the power backup systems |
| 23:13.03 | fujin | our datacentre has dual 500kVA feeds, 800kva UPS |
| 23:13.05 | fujin | and a big ass generator |
| 23:13.11 | pc500 | delmar - How to address? |
| 23:13.11 | fujin | all controlled by the building management system |
| 23:13.16 | pc500 | delmar - It's VoipJET |
| 23:13.19 | fujin | not really much to worry about, for power ;P |
| 23:13.47 | fujin | dual E1's, dual powersupplies in everything, dual asterisk boxes |
| 23:13.49 | fujin | winwin |
| 23:13.50 | pc500 | fujin - googld 365 n main |
| 23:13.51 | JT | fujin: are all your phones in the datacentre? |
| 23:13.52 | pc500 | :) |
| 23:13.56 | delmar | pc500, the symptoms were silence suppression for sure.. i blow into the phone constantly.. and the music is constant... if I stop.. the music will stop a couple'o secs later |
| 23:14.04 | fujin | JT: not in the DC, but on the desks |
| 23:14.12 | delmar | pc500, yet when i phoned my NZ DID directly.. which was via VJ.. it was fine... |
| 23:14.18 | JT | do they all have power backup too? |
| 23:14.21 | fujin | helpdesk callcentre, NOC, accounts receivable, sales, programming |
| 23:14.24 | delmar | pc500, seemed to affect the USA calling. |
| 23:14.47 | fujin | all are PoE, with adaptors available |
| 23:14.50 | pc500 | delmar - And am I beleive this only affects MoH operation? |
| 23:14.54 | fujin | have a stash of adapters in my drawer |
| 23:14.58 | delmar | pc500, lets try something else. you have a DID yourself? |
| 23:14.58 | JT | main 365, what clowns |
| 23:15.07 | pc500 | delmar - no |
| 23:15.10 | fujin | main 365? |
| 23:15.22 | pc500 | fujin - All backup systems in place, yet poewr still failed |
| 23:15.23 | JT | fujin: google them |
| 23:15.23 | pc500 | murphys law |
| 23:15.27 | delmar | pc500, you have dialout only.. no dialin? |
| 23:15.37 | fujin | They must really suck. |
| 23:15.41 | fujin | learn2infrastructure, imho |
| 23:15.42 | JT | more like idiot's law, someone clearlym screwed up |
| 23:15.46 | fujin | bingo |
| 23:15.47 | JT | yeah |
| 23:16.02 | pc500 | delmar - correct. |
| 23:16.17 | JT | fujin: so what's this about adapters? |
| 23:16.24 | fujin | all of my phones are PoE |
| 23:16.34 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 23:16.38 | fujin | haven't really thought about that stage |
| 23:16.44 | pc500 | fujin - http://radar.oreilly.com/archives/2007/07/365_main_datace.html |
| 23:16.46 | fujin | dual switches? so that if a switch fails |
| 23:17.10 | delmar | pc500, run it past VJ support i guess.. |
| 23:17.11 | JT | fujin: so the PoE switches aren't on power backup? |
| 23:17.14 | pc500 | delmar - Ya, I need an origination provider |
| 23:17.17 | pc500 | delmar - no response. |
| 23:17.18 | fujin | of course they are |
| 23:17.29 | fujin | the entire building is hooked up the the dual feeds, ups and generators |
| 23:17.49 | delmar | pc500, we been using Asterlink and les.net |
| 23:17.54 | delmar | but to be honest.. they are both hopeless. |
| 23:18.09 | delmar | we are switching a DID to voicepulse to test. |
| 23:18.09 | fujin | Telstraclear in NZ here are looking to provide a SIP product in the next six months |
| 23:18.12 | lesouvage | JT: If you do an implementation for say 250 phones and take all the precautions (poe, qos, redundentasterisk servers, different subnets for data and enterprised class itp) Asterisk is still a very attractive alternative for the Avayas and Ciscos of this world. |
| 23:18.24 | fujin | they told me that IAX2 was a proprietary protocol and they'd never support it, to which I told them they were idiots |
| 23:18.27 | delmar | fujin, they said that 6months ago |
| 23:18.52 | JT | iax2 is practically proprietary |
| 23:18.56 | JT | just use sip |
| 23:19.05 | delmar | fujin, slingshot dont support IAX either... dicks.. .yet they run an out dated SipExpress router on a Linux box for us to connect to.. quite hapily. |
| 23:19.21 | pc500 | delmar - Nice, $1/month did. |
| 23:19.29 | delmar | fujin, if you are in need of a IAX to SIP translater in NZ let me know.. my colo is doing exactly that for a few people in NZ |
| 23:19.58 | delmar | JT, Nat Traversal |
| 23:20.01 | delmar | ponder that |
| 23:20.10 | JT | lesouvage: sure, a pbx replacement... not a good argument for using VoIPoI though |
| 23:20.17 | JT | delmar: i have, and i have no issues |
| 23:20.23 | fujin | I assumed IAX2 was better for trunking, especially from someone carrier-grade. |
| 23:20.39 | fujin | delmar: I just wanted a backup for when my E1's are filled up (64~ concurrent) |
| 23:20.44 | JT | fujin: poor assumption, no-one in the real telecomms world uses IAX2 |
| 23:21.02 | JT | SIP and H.323 are the real VoIP protocols |
| 23:21.04 | pc500 | JT - I think even the majors provide VoIP termination, but it's SIP |
| 23:21.09 | pc500 | JT - like Qwest, etc. |
| 23:21.12 | fujin | mm |
| 23:21.14 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 23:21.23 | fujin | I suppose running iax2 would involve running Asterisk |
| 23:21.30 | fujin | or soem other licensed stuff from digium |
| 23:21.33 | delmar | there is nothing wrong with IAX2 |
| 23:21.37 | JT | no, it could be callweaver, yate or freeswitch |
| 23:21.41 | JT | it's not licensed |
| 23:21.43 | delmar | there are a number of VISP's out there using it. |
| 23:21.55 | JT | delmar: there's nothing that googd about it either |
| 23:21.57 | JT | good |
| 23:22.05 | JT | it's a moving target |
| 23:22.06 | pc500 | fujin - It's open source, you could do what you want with it, eh? |
| 23:22.10 | delmar | JT, theres lots I hate about SIP :P |
| 23:22.21 | pc500 | The only bad about SIP is NAT |
| 23:22.23 | JT | look how much trouble different versions of asterisk have with IAX2 interoperability |
| 23:22.39 | delmar | JT, never encountered an issue actually. |
| 23:22.42 | Techie-Micheal | I like SIP. SIP is teh <3 of my life. hehe. |
| 23:22.45 | JT | most people who can't get sip to work through nat have a setting set wrong |
| 23:23.33 | lesouvage | JT: PBX replacement is a natural moment for making a choice for the way to connect to the old telephony network. The hardest and most expensive part of an Asterisk migration is to connect to the old telephony like pstn or isdn. |
| 23:23.50 | pc500 | lesouvage - Or the rest of the world. |
| 23:23.52 | pc500 | :) |
| 23:23.53 | delmar | JT, SIP and Nat is easy.. if u use Stun servers and bounce shit around but... what does that achieve? another point of failure... security / privacy issues, increased latency. |
| 23:24.07 | JT | delmar: i never need stun |
| 23:24.15 | JT | delmar: again, you must have settings set wrong |
| 23:24.17 | JT | ~sipnat |
| 23:24.18 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 23:24.22 | JT | or a bad nat router |
| 23:25.03 | Techie-Micheal | JT: You'll need STUN for a phone connecting to a public asterisk, won't you? I know I did. |
| 23:25.15 | JT | lesouvage: quite necessary to connect to the real world unless you just expect people to only be talking to each other with |
| 23:25.22 | JT | Techie-Micheal: nope. |
| 23:25.28 | delmar | JT, Asterisk ===> [Router/Nat] ==== [interweb aka pr0n network] ======[Router/nat] === Asterisk --- OR .. make one of those Asterisk boxes a SIP phone device ... and that scenario.. needs Stun |
| 23:25.44 | JT | Techie-Micheal: something must be set wrong, we have hundreds of people been through here who can make sip work through nat |
| 23:26.29 | delmar | JT, of course, its much easier now I guess... thank christ for better SIP / uPnP routers. |
| 23:26.32 | JT | delmar: if you are acting as a sip server, you must be listening on a public ip (you can use port forwarding), iax2 is no different |
| 23:26.42 | JT | gar |
| 23:26.43 | JT | no |
| 23:26.47 | flenders | I don't have STUN |
| 23:26.48 | JT | you don't need upnp |
| 23:26.54 | JT | and you do not need a sip aware router |
| 23:26.58 | JT | half of them fuck it up |
| 23:27.00 | flenders | I use SIP on NAT as well |
| 23:27.05 | lesouvage | JT: my point is that if you do business with an enterprise class voip provider you just have to add sip and/or iax trunks to your system without the hasle of E1 interfaces. |
| 23:27.10 | JT | all you need is a nat device that does NAT sanely |
| 23:27.12 | JT | most do. |
| 23:27.24 | pc500 | or support SIP :) |
| 23:27.34 | JT | ... |
| 23:27.35 | Techie-Micheal | I'll give you that my DSL modem is kinda nub. :P |
| 23:27.50 | flenders | I don't get why people say they need a router that supports SIP |
| 23:27.58 | flenders | SIP is just a protocol |
| 23:27.58 | JT | lesouvage: using a dedicated connection to your voip provider? |
| 23:28.09 | JT | flenders: beats me |
| 23:28.09 | delmar | anyway.. all that crap aside.. i found the best way for me was to plop a DSL300 ethernet modem on a second interface on my linux box, so the linux box is the router and has a public IP. all sorts of SIP issues just went away. |
| 23:28.15 | flenders | if your router does nat properly, it doesn't need to support SIP |
| 23:28.17 | pc500 | flenders - There are routers that cna handle the RTP streams without any other modificaitons or nat enablement on asterisk |
| 23:28.32 | flenders | if it says it supports SIP, STAY AWAY FROM IT! it will fuck things up |
| 23:28.43 | delmar | flenders, a router that does nat properly.. supports SIP properly.. :P |
| 23:28.51 | lesouvage | JT: if telephony is mission critical I think that is a good suggestion. |
| 23:28.52 | JT | pc500: yes, it's called a nat device that is NOT broken |
| 23:29.04 | JT | lesouvage: it's almost always mission critical for business |
| 23:29.05 | delmar | flenders, its all about how well it helps when the media ports are negotiated in the SIP protocol |
| 23:29.07 | pc500 | Is there a command to show jitter in asterisk? |
| 23:29.14 | JT | delmar: gar |
| 23:29.20 | JT | delmar: you do not NEED help |
| 23:29.31 | JT | there does not need to be ANY sip helper on the router |
| 23:29.32 | flenders | IT'S NAT, for chrits sake |
| 23:29.39 | flenders | christs |
| 23:29.44 | JT | i don't think you understand how nat traversal works |
| 23:29.46 | pc500 | look what I started, lol |
| 23:29.47 | lesouvage | JT: Yes I know. |
| 23:30.01 | JT | lesouvage: E1s are reliable, a good choice for business |
| 23:30.06 | delmar | JT, ok run it past us.. whats the correct setup.. port forwarding.. the works.. |
| 23:30.16 | JT | delmar: give me a scenario first |
| 23:30.18 | JT | and read |
| 23:30.20 | JT | ~sipnat |
| 23:30.20 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 23:30.21 | flenders | I think the biggest mistake here is that people don't really know network devices and how they work |
| 23:30.38 | JT | especially aocomputing |
| 23:30.45 | JT | that's [tk]-defender's site |
| 23:30.54 | delmar | JT the most anoying scenario... SIP phone behind a NAT box... Asterisk also behind a NAT box... make them talk with NO one way audio :P |
| 23:30.56 | flenders | it was so easy for me to do it here... |
| 23:31.06 | pc500 | asterisk behind nat box is the failure. |
| 23:31.10 | pc500 | That's what you control and fix :) |
| 23:31.23 | delmar | yup. |
| 23:31.25 | JT | delmar: asterisk must have 5060, and 10000-20000 udp port forwarded to it from public ip |
| 23:31.35 | JT | externip= and localnet= in sip.conf must be set. |
| 23:31.59 | delmar | pc500, its why i have so many peope bouncing IAX to my Colo, then SIP to/from their tolls/did provider from that box. |
| 23:32.11 | JT | asterisk needs no port forwarding if it's only acting as sip client |
| 23:32.13 | pc500 | JT - I only has 50 media ports forwarded, but it works fine (I have to set SIP devices and asterisk accordingly). |
| 23:32.36 | JT | delmar: because they have no idea what they are doing? |
| 23:32.48 | delmar | JT, this could be true :P |
| 23:32.57 | *** join/#asterisk matt_ (n=matt@2001:770:168:1:220:edff:feb4:7c9d) |
| 23:33.25 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:33.55 | delmar | JT, the Asterisk box is behind a NAT router and the ISP does not allow Static IP's. |
| 23:33.56 | delmar | :P |
| 23:33.58 | delmar | solve that |
| 23:34.05 | delmar | switch ISP? |
| 23:34.05 | delmar | yup |
| 23:34.06 | delmar | :) |
| 23:34.13 | pc500 | i could, too lazy, no need. |
| 23:34.13 | pc500 | hehe |
| 23:34.18 | JT | delmar: umm |
| 23:34.30 | Qwell | dynamic DNS, forward ports. same deal |
| 23:34.44 | delmar | JT, can u set a dyndns as externip ? |
| 23:34.45 | JT | delmar: i have heaps of boxes behind nat on dynamic ip provider, no issues connecting to a sip server |
| 23:34.47 | delmar | oh yeah u can |
| 23:34.54 | JT | delmar: externhost |
| 23:34.58 | JT | not as reliable |
| 23:34.58 | delmar | Qwell, hey man |
| 23:35.03 | JT | but yeah |
| 23:35.14 | delmar | Qwell, how long have u been lurking eh? :P |
| 23:35.15 | JT | why the hell would you have a sip server on a dynamic ip? |
| 23:35.26 | Qwell | like... |
| 23:35.35 | Qwell | 10 hours? |
| 23:35.40 | Techie-Micheal | Home user wants to experiment with Asterisk? |
| 23:35.43 | delmar | JT, u wouldnt but .. anal Telco's like TelecomNZ haven't been allowing statics for years... |
| 23:36.03 | delmar | u can get statics on other ISP's now.. finally, but some want to charge $10 or more /month |
| 23:36.16 | JT | delmar: have to love new zealand |
| 23:36.26 | delmar | Qwell, omg u voyer :P |
| 23:36.26 | JT | still so backwards |
| 23:36.36 | delmar | yeah... |
| 23:36.53 | delmar | our government has to force the Telco to sort their shit out and do the right thing. |
| 23:37.05 | JT | pc500: also, why would you forward only 50 media ports, forward the full range specified in rtp.con |
| 23:37.09 | JT | rtp.conf |
| 23:37.23 | Techie-Micheal | The average home user isn't going to need a static IP, or may not even want a static IP. Sure, if they want to run a production Asterisk server from their home, they should get a static IP, but to play around with Asterisk? Seems overkill to me. |
| 23:37.53 | JT | Techie-Micheal: but you don't need a static to play around with it? |
| 23:38.12 | JT | you can easily connect to sip servers on the Internet with a dynamic ip |
| 23:38.13 | Techie-Micheal | JT: That's the point I'm trying to make. :P delmar suggested static IP. |
| 23:38.38 | JT | Techie-Micheal: i suggest it if you have sip or iax2 clients connecting to your asterisk from the outside |
| 23:39.02 | *** join/#asterisk ZX81 (n=matt@202.20.97.200) |
| 23:39.05 | delmar | as JT has pointed out, if Asterisk is setup right, I don't think a Static is needed for Asterisk to work as a server. |
| 23:39.06 | ZX81 | ~adn |
| 23:39.06 | jbot | rumour has it, adn is hmm... adn is is the Asterisk Daily News - http://www.venturevoip.com/news.php for HTML and http://feeds.feedburner.com/asterisknews for RSS |
| 23:40.12 | JT | delmar: to clairfy: asterisk connecting to sip servers on Internet = doesn't matter if static or dynamic. sip clients on the Internet connecting to asterisk = probably a very good idea to get a static ip. |
| 23:40.33 | *** join/#asterisk matt_ (n=matt@2001:770:168:1:220:edff:feb4:7c9d) |
| 23:40.41 | delmar | I know I certainly love my static |
| 23:41.40 | *** join/#asterisk mltlnx (n=mltlnx@cpe-69-203-218-147.nyc.res.rr.com) |
| 23:41.41 | flenders | I have asterisk on dynamic IP at home |
| 23:42.16 | flenders | the only thing is that I have to reload sip whenever the IP changes (I use externhost |
| 23:42.19 | flenders | ) |
| 23:43.33 | pc500 | When I do a conference on my phone, does it do it at the asterisk box level, or is it the phone doing it itself? |
| 23:44.01 | delmar | pc500 depends |
| 23:44.12 | delmar | pc500, are u pressing the conf button on the phone? |
| 23:44.17 | pc500 | yes' |
| 23:44.26 | pc500 | ip phone |
| 23:44.27 | delmar | pc500, probably a feature on the phone.. my Polycom does that |
| 23:44.54 | delmar | pc500, the call is hooked up within the phone.. and when i hangup the other two callers can continue as well. |
| 23:45.35 | flenders | it doesn't drop the call? |
| 23:45.46 | pc500 | it does |
| 23:45.47 | flenders | didn't know polycoms did that |
| 23:46.03 | flenders | < delmar> pc500, the call is hooked up within the phone.. and when i hangup the other two callers can continue as well. |
| 23:46.05 | pc500 | It appears that there's 2 sip channels. |
| 23:46.24 | pc500 | transfers are handled at the asterisk level |
| 23:46.34 | pc500 | but for conference, it just opens up more lines on the phone, strange. |
| 23:46.57 | JT | not strange |
| 23:47.03 | JT | completely normal sip behaviour |
| 23:47.10 | pc500 | conference is not part of SIP spec? |
| 23:47.23 | pc500 | It seems more logical for such a common function to be handled at the server level (like on hold, transfer, etc.) |
| 23:47.40 | JT | 3-way calling is a function of the sip endpoint |
| 23:47.48 | JT | conference rooms are different |
| 23:47.49 | JT | nup |
| 23:47.57 | JT | and transfers are signalled by the endpoints |
| 23:49.04 | pc500 | anyone else here use voipjet? |
| 23:49.20 | JT | the device the provides the dialtone is responsible for implementing 3-way calling and what not |
| 23:49.29 | JT | with sip phones, the phone provides the dialtone |
| 23:49.57 | pc500 | me no like :) |
| 23:50.08 | pc500 | JT - Is call forwarding a function of the phone picking up and transfering the call? |
| 23:50.42 | JT | usually phones can implement transfers themselves |
| 23:50.45 | JT | err |
| 23:50.48 | JT | call forwarding |
| 23:51.06 | pc500 | So they get the ring, and send a signal to the SIP server to immeidately transfer? |
| 23:51.24 | pc500 | Rather than registering a "shouldb e transfered to" with the server. |
| 23:51.58 | JT | yeah, nothing as fancy as registering those details to the server happens |
| 23:52.11 | pc500 | I sense sip v2.0 |
| 23:52.28 | Qwell | pc500: surely you mean 42.9.23 |
| 23:53.56 | pc500 | hehe |
| 23:55.50 | delmar | flenders, sorry was afk. yep. thats what my poly does. I had a support call, hooked in a supplier.. left them to it and hung up. she told me after that the guy was abusive, nasty, and a right prick after I hung up..... |
| 23:56.03 | delmar | naturally I attached the call recording to an email and sent it to the guys boss :P |
| 23:57.02 | delmar | dunno.. I could be wrong and it could be Asterisk doing it I guess... |
| 23:57.05 | delmar | Qwell, ? |
| 23:57.13 | anonymouz666 | what happens with two peers are talking with canreinvite=yes and then you try to perform a atxtransfer with #1 |
| 23:57.17 | delmar | Qwell, what do you think about this? |
| 23:57.35 | JT | delmar: the call was recorded in compliance with the legislation right? |
| 23:57.35 | flenders | again, didn't know polycoms could do that. I thought once you hangup, you drop both calls |
| 23:58.01 | delmar | JT, only 1 party needs to know the call is being recorded. |
| 23:58.13 | delmar | otherwise thats naughty. |
| 23:58.34 | delmar | flenders, yep. so did. it was a surprise. |
| 23:58.40 | JT | in most places, both parties need to know uless there's a warrant |
| 23:58.49 | delmar | JT, not in NZ. |
| 23:58.54 | JT | weird |
| 23:59.55 | delmar | JT, highly illegal if neither party is aware they are being recorded.. that goes for interception of phone calls, or an actual recording/listening device in a room where there are people speaking |
| 23:59.56 | anonymouz666 | hello? |
