| 00:02.30 | Katty | peeks in |
| 00:03.07 | jblack | Hi katty. I'm on my way out. |
| 00:03.15 | Katty | mkay |
| 00:18.27 | *** join/#asterisk WindBack (n=quassel@200-122-74-15.cab.prima.net.ar) |
| 00:20.28 | WindBack | In * 1.6 it's possible to create templates in the sip.conf file |
| 00:21.17 | WindBack | I want to set up the mailbox in every extension in the way mailbox=exte@voicemailContex |
| 00:21.41 | WindBack | to send the notifys to the phones |
| 00:22.01 | WindBack | there is a way to do it in a generic way |
| 00:22.19 | WindBack | to avoid writting it on every extension? |
| 00:33.20 | drmessano | HAHAH |
| 00:33.37 | drmessano | jblack: "eating pistacios" |
| 00:33.39 | *** join/#asterisk Malkor (n=marco@hlle-d9ba03e9.pool.mediaWays.net) |
| 00:33.40 | drmessano | I love you man |
| 00:33.43 | *** join/#asterisk xanderp (n=xanderp@c-98-220-167-76.hsd1.in.comcast.net) |
| 00:33.49 | Katty | and not sharing?! |
| 00:34.32 | drmessano | I was amused at the conecpet of the completely useless status update that jblack seems to have embraced |
| 00:34.52 | drmessano | concept too |
| 00:35.16 | [TK]D-Fender | WindBack: No |
| 00:36.48 | xanderp | I setup a successfully tested a softphone sip client from my LAN. I'd like to setup a secure way to have it work from the Internet. I currently don't have my asterisk allowing clients to connect from internet, can someone point me to good docs on this? |
| 00:37.04 | WindBack | [TK]D-Fender: another question please, Can I have different subscribecontext per UA? |
| 00:37.49 | [TK]D-Fender | WindBack: Per peer |
| 00:37.57 | WindBack | [TK]D-Fender: ok |
| 00:38.31 | WindBack | So, can I write this option in each peer? Is not a neccesary global option, ok? |
| 00:39.06 | [TK]D-Fender | WindBack: Clearly |
| 00:39.35 | WindBack | [TK]D-Fender: the question is: Can i define it on each peer? |
| 00:39.41 | WindBack | in the sip.conf? |
| 00:39.57 | [TK]D-Fender | WindBack: How many more times do I have to say it? |
| 00:40.18 | [TK]D-Fender | WindBack: YES YOU CAN SET IT IN EACH PEER SPECIFICALLY. |
| 00:40.23 | [TK]D-Fender | WindBack: How about now? |
| 00:41.21 | WindBack | [TK]D-Fender: ok, thank you |
| 00:42.27 | WindBack | [TK]D-Fender: sorry if i bother you, it wasn't my intention |
| 00:43.47 | ChannelZ | it never is |
| 00:43.55 | [TK]D-Fender | WindBack: It'd simply be nice not to have to answer the same question 3 times sequentially... |
| 00:45.35 | WindBack | [TK]D-Fender: I red in some pleace that this is just global option. This is why I wanted to be sure |
| 00:46.07 | [TK]D-Fender | WindBack: Then perhaps you should ask me another 5-6 times. |
| 00:46.17 | [TK]D-Fender | WindBack: You know... juwt to make sure I'm REALLY sure |
| 00:47.09 | ChannelZ | the sample configs show it per peer.... |
| 00:48.20 | WindBack | [TK]D-Fender: why are you so aggressive? |
| 00:50.14 | ChannelZ | roids |
| 00:51.22 | [TK]D-Fender | I am NOT agreessive .... and I'll KILL the next fucker who says I am! |
| 00:52.05 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
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| 01:03.49 | xanderp | The thing that made him so worried was the fact that everyone kept asking him what he was looking so worried about all the time... |
| 01:04.20 | *** join/#asterisk Whtsup (n=sssi@WimaxUser375-13.wateen.net) |
| 01:07.27 | Whtsup | hello |
| 01:16.01 | *** join/#asterisk chendy (n=chatzill@119.137.95.76) |
| 01:16.25 | Katty | ello |
| 01:48.00 | *** join/#asterisk dgoner (n=david@mx1.repairpc.net) |
| 01:53.24 | carrar | HELLO KATTY |
| 01:54.34 | *** join/#asterisk arcanine18_2000 (n=saxon_m2@121.58.209.130) |
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| 02:19.40 | Katty | carrar: hi |
| 02:25.40 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:25.40 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 02:31.50 | *** join/#asterisk voipmonk (n=voipmonk@67.204.57.187) |
| 02:32.14 | carrar | I saw your future car katty |
| 02:32.14 | carrar | http://www.monthlyjoongang.com/wp-content/uploads/2009/07/hello-kitty-ferrari.jpg |
| 02:33.06 | [TK]D-Fender | carrar: Proof that money can't buy taste.... |
| 02:33.15 | carrar | haha |
| 02:34.52 | [TK]D-Fender | Ok this is simply not working... no way to compress the lyrics for "Semi-Charmed Life" to under 3 pages..... GAH |
| 02:35.10 | [TK]D-Fender | stare at his sheets some more.. |
| 02:35.14 | carrar | remove every 3rd letter |
| 02:38.07 | [TK]D-Fender | Ooooh I think I've almost done it... make it harder to follow, but I should only need it as a guide... |
| 02:39.02 | [TK]D-Fender | \o/ |
| 02:39.04 | [TK]D-Fender | I rock |
| 02:39.17 | [TK]D-Fender | (in more ways than 4) |
| 02:40.02 | *** join/#asterisk asteriskmonkey (n=philip@TOROON63-1168099548.sdsl.bell.ca) |
| 02:40.37 | asteriskmonkey | anyone use the audiocodes mp-118 with 5.6? |
| 02:50.51 | Katty | carrar: gosh. |
| 02:52.10 | Katty | carrar: http://gtcarlot.com/gallery/photo.php?id=409268 <- my next car. |
| 03:00.02 | Katty | http://i.imgur.com/GKAMW.jpg <- 1917 |
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| 03:53.10 | Katty | hi jaytee |
| 03:53.17 | jaytee | hi Katty |
| 03:54.05 | *** join/#asterisk bobsaccamano (i=ca839b14@gateway/web/freenode/x-ralakrzspkvlgvou) |
| 03:54.49 | bobsaccamano | hi..i have a doubt related to digit maps for a business group: Suppose the dialed digits do not match the digit strings in the map, then what happens? |
| 03:54.56 | bobsaccamano | is the call allowed to go through? |
| 03:56.11 | [TK]D-Fender | bobsaccamano: What map? |
| 03:56.44 | bobsaccamano | [TK]D-Fender: a Digit Map..like so (0T|00T|[1-7]xxx|8xxxxxxx|#xxxxxxx|*xx|91xxxxxxxxxx|9011x.T) |
| 03:56.59 | bobsaccamano | just an example.. |
| 03:57.19 | [TK]D-Fender | bobsaccamano: That clearly is not an ASTERISK dialplan issue. Maybe you should tell us what device you're talking about so we know what to answer you <- |
| 03:57.48 | Katty | jblack: interesting. my kidney is also worthless |
| 03:57.51 | [TK]D-Fender | "Hi my system is running hot... what should I do?" |
| 03:58.02 | [TK]D-Fender | "Oh I forgot to tell you.. I'm talking about my OVEN" |
| 03:58.08 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 03:58.17 | bobsaccamano | [TK]D-Fender: yeah..its not asterisk..this is a softswitch by NSN |
| 03:58.33 | [TK]D-Fender | bobsaccamano: Maybe you should read its MANUAL to see how it will react <- |
| 03:59.02 | bobsaccamano | [TK]D-Fender: thanks.. |
| 03:59.20 | [TK]D-Fender | bobsaccamano: I have never heard mention of that system in here ever |
| 03:59.24 | ian6 | [TK]D-Fender: ... you can help me with my oven? |
| 03:59.45 | [TK]D-Fender | ian6: I only support EZ-Bake v3 ovens.... |
| 04:00.30 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 04:01.44 | Katty | jblack: go figure. ryan's would sell for a lot |
| 04:06.17 | eppigy | KIMCHI |
| 04:07.30 | bobsaccamano | [TK]D-Fender: i wanted to know if there's a standard way of dealing with digit maps |
| 04:13.13 | *** join/#asterisk coldk1ller (n=coldkill@74.208.13.224) |
| 04:13.22 | *** part/#asterisk coldk1ller (n=coldkill@74.208.13.224) |
| 04:15.02 | jblack | Katty: Heh |
| 04:17.40 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 04:24.00 | [TK]D-Fender | bobsaccamano: Every device can be different. Polycom has a flag with options on what to do when the pattern fails. |
| 04:24.25 | [TK]D-Fender | bobsaccamano: So no, there is no "easy answer" for you. You're jsut going to have to read the dosc on how yours reacts and if in doubt go test it |
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| 04:36.33 | rdude99 | Is it possible to use the say/play commands but to instead route audio to the console / soundcard vs. a phone line? |
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| 05:42.35 | Defraz | exit |
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| 06:42.01 | *** join/#asterisk p1mrx (n=paul@mediabox.apt.pmarks.net) |
| 06:42.49 | p1mrx | what does "DISCO" mean, in a voip context? |
| 06:43.25 | *** join/#asterisk Failrar (n=Failrar@2001:470:1f15:316:2a0:d1ff:fe4e:e802) |
| 06:43.30 | p1mrx | people seem to use it to talk about client VoIP hardware, but what does it stand for? |
| 06:44.22 | p1mrx | oh, never mind, I just found it: "Device Is Supplied by the COmpany" |
| 06:46.32 | snadge | disco stu.. loves.. disco music |
| 06:48.12 | Tim_Toady | openvox analog cards are any good? anyone has experience with this hardware? |
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| 07:10.38 | drmessano | Tim_Toady: Limited use, but they seem solid to me |
| 07:12.07 | Tim_Toady | i see, thx |
| 07:14.06 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 07:17.03 | arossouw | anyone used Snom 300's , these phones keep losing their password setting |
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| 07:37.16 | tzafrir | stupid little trick: as we use analog phones here, and each has his own PC, I wrote a simple dialer CGI script |
| 07:37.55 | tzafrir | It has a simple mapping of IP address to phone (originating device, in Asterisk) |
| 07:38.38 | tzafrir | You feed a number, and it originates a call from your device. |
| 07:39.16 | tzafrir | Added one extra apache rewrite rule, and I can dial using http://pbx/dial/<number> |
| 07:44.31 | tzafrir | No authentication needed. |
| 07:45.18 | jblack | I nearly wrote something like that for sip phones. |
| 07:45.26 | jblack | a click-to-dial |
| 07:49.44 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 07:49.49 | ChannelZ | I wanted to do something like that from a client database |
| 07:50.18 | *** join/#asterisk Malkor (n=marco@hlle-d9ba01a0.pool.mediaWays.net) |
| 07:53.35 | creativx | click2dial is a must |
| 07:54.30 | mort_gib | tzafrir: What was that for?? |
| 07:55.02 | tzafrir | well, the point is that you don't have a local phone client |
| 07:55.41 | tzafrir | most other dialers that I know require giving every user manager access to the system |
| 07:56.02 | tzafrir | or a much more complex software on the Asterisk box |
| 07:56.03 | jblack | shudders |
| 07:56.22 | tzafrir | Here I have something that is much simpler |
| 07:56.36 | mort_gib | http://www.voip.com.sg/voip-products/asterisk-tools/asterisk-outlook-dialer.html |
| 07:58.29 | jblack | yuck. michael jackson won an award at the AMA. Even dead, he's unstoppable. |
| 07:59.33 | ChannelZ | Hurray! |
| 08:03.08 | tzafrir | mort_gib, my dial.cgi is practically the same as originate.php in the place where you pointed me, |
| 08:03.33 | tzafrir | Only it provides the originating device in a much safer way |
| 08:03.40 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 08:05.07 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 08:05.25 | tzafrir | I take my words back |
| 08:06.14 | *** join/#asterisk mpe (n=mpe@gate.ipvision.dk) |
| 08:07.48 | tzafrir | This originate.php allows you to originate a call from any channel |
| 08:08.11 | *** join/#asterisk icyValk77 (n=icyValk7@213.129.64.4) |
| 08:09.13 | tzafrir | It also does no input sanitation, and thus I believe you can make it dial to any application you want, with a properly-crafted channel argument |
| 08:09.21 | creativx | astmanproxy gives you the same |
| 08:09.59 | tzafrir | my point is that I don't trust the users on the LAN. I don't want to. |
| 08:10.37 | creativx | what, you dont have a bat hanging on the wall? |
| 08:10.58 | tzafrir | In my case calls can be originated only from a select set of devices. Even if you manage to fake the IP address (which would ring some alarm bells, as I use arpwatch) |
| 08:12.25 | tzafrir | I realise mapping IP addresses to devices isn't applicable in many (most?) cases. But here it works |
| 08:22.56 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 08:23.44 | *** join/#asterisk cjk (n=cjk@vodsl-8293.vo.lu) |
| 08:24.06 | lordmortis | how can I do a CDR lookup via an external application? (or can * do a web request and stash the result in a variable?) |
| 08:24.40 | cjk | hi, how can i identify iax signalling traffic using iptabels (poke, authentication etc...) |
| 08:29.06 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
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| 08:48.50 | Sargun | Anyone here mucked with SS7? |
| 08:50.29 | creativx | lordmortis: curl |
| 08:50.45 | lordmortis | creativx: aah okay. Why not! :) |
| 08:50.50 | lordmortis | thanks for that |
| 08:50.54 | creativx | thats what I use |
| 08:51.07 | creativx | i think |
| 08:51.09 | creativx | hehe. let me check |
| 08:51.34 | lordmortis | curl function, and if i've got the libs it'll be built into the system |
| 08:51.37 | lordmortis | ? |
| 08:51.51 | creativx | exten => s,n,set(foo=${CURL(${myurl}${qstring})}) |
| 08:52.00 | lordmortis | cool |
| 08:52.01 | creativx | i suppose, not sure |
| 08:52.16 | creativx | it works here, and i have no idea what libs i have or not :) |
| 08:54.04 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:55.00 | tzafrir | infobot, tell Sargun about ask |
| 08:55.26 | Sargun | Anyone here have an SS7 trunk, with HLR? how did you get it? |
| 08:56.14 | tzafrir | Sargun, you want the exact names of the persons you need to pay off? |
| 08:56.19 | Sargun | ahhaha |
| 08:56.22 | Sargun | essentially, yes |
| 08:56.40 | tzafrir | has no business with SS7 |
| 08:57.04 | gr0mit | Sargun, once, a while back yes |
| 08:57.12 | Sargun | gr0mit, in the US? |
| 08:57.23 | gr0mit | <PROTECTED> |
| 08:57.31 | gr0mit | but it was connected to a GSM switch in our lab |
| 08:57.41 | Sargun | you didn't connect it to the PSTN? |
| 08:58.00 | gr0mit | <PROTECTED> |
| 08:58.13 | gr0mit | I used Chan_SS7 |
| 08:58.19 | gr0mit | if I recall correctly |
| 08:58.30 | Sargun | needs connectivity to the PSTN |
| 08:58.31 | gr0mit | my only recollection really is that it was not for the fainthearted! |
| 08:58.43 | *** join/#asterisk mortsmel (n=andrew@65.183.179.53) |
| 08:59.25 | gr0mit | I did get calls between the two switches, which was my real goal |
| 08:59.50 | Sargun | hehe |
| 09:01.01 | gr0mit | it was only a test environment |
| 09:01.10 | gr0mit | when I worked for Motorola |
| 09:01.48 | Sargun | ah |
| 09:05.02 | creativx | sigh |
| 09:05.05 | creativx | when is my n900 arriving |
| 09:05.08 | gr0mit | but that, as they say is history |
| 09:05.15 | creativx | mobile phones and delivery dates == always success |
| 09:05.25 | gr0mit | pats his G1 |
| 09:06.00 | gr0mit | Sargun, first find your pet telco |
| 09:06.42 | Sargun | creativx, I'm wondering as well |
| 09:07.06 | creativx | they always hype the delivery dates |
| 09:07.18 | Sargun | gr0mit, That would be t-mobile, or AT&T I think |
| 09:08.42 | Sargun | gr0mit, step 2? |
| 09:14.26 | gr0mit | get a price from them for your T1 |
| 09:14.33 | *** join/#asterisk icyValk77 (n=icyValk7@213.129.64.4) |
| 09:15.50 | gr0mit | Sargun, what are tou trying to achieve? |
| 09:16.49 | Sargun | Well, basically we have an application server that communicates over SS7. We were using a provider earlier, but they're being a PITA (we don't have low level access). We need to make an SMSC and do our own SMS. |
| 09:19.02 | Sargun | gr0mit, We have a T1, but they apparently wont sell us a block or SS7 PRI on the T1 |
| 09:20.18 | gr0mit | okay |
| 09:20.54 | gr0mit | have you tried talking to one of the European providers? |
| 09:21.07 | gr0mit | I think there are a lot more flexible |
| 09:21.19 | gr0mit | for example, Jersey Telecom or Guernsey Telecom |
| 09:23.37 | Sargun | can't, need a US block of numbers |
| 09:28.12 | Sargun | Alternatively, do you know of any SMSCs which will give me full MAP access? |
| 09:28.20 | Sargun | or a significant amount of MAP access? |
| 09:28.21 | *** join/#asterisk Faustov (i=user@gentoo/user/faustov) |
| 09:33.54 | gr0mit | hello Faustov |
| 09:35.12 | Faustov | morning gr0mit |
| 09:35.40 | Faustov | I'm lurking for mr Chainsaw :> |
| 09:35.53 | *** join/#asterisk bmg505 (n=leon@196.209.8.178) |
| 09:36.07 | gr0mit | sounds dangerous! |
| 09:36.34 | Faustov | I can't reproduce the init.d problem and I can't debug it on my production box |
| 09:36.40 | Faustov | I'll need him to have a look |
| 09:38.03 | gr0mit | very strange |
| 09:38.13 | gr0mit | do you normally restart*every night? |
| 09:41.18 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:42.23 | tzafrir | Faustov, what problem? |
| 09:42.56 | Faustov | gr0mit: no, I don't |
| 09:43.14 | Faustov | tzafrir: /etc/init.d/asterisk start does not do anything and no errors. |
| 09:43.53 | tzafrir | Faustov, if a certain program "does not do anything", I apply strace -f |
| 09:44.44 | tzafrir | Reading this makes it look as if '-f' is for '--force' |
| 09:44.49 | Faustov | tzafrir: init.d is a bash script, I'd put some assertions inside to test where it has a problem, but unfortunately it is in production :< |
| 09:45.28 | tzafrir | How have you started asterisk? |
| 09:45.52 | tzafrir | if it is a bash script, just use: bash -x /etc/init.d/asterisk start |
| 09:46.54 | Sargun | gr0mit, So, 10 steps to establish your own SMS endpoint? |
| 09:47.13 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:47.33 | gr0mit | Sargun, no idea! |
| 09:49.17 | Faustov | tzafrir: there's a safe_asterisk script, I think shipped with wanpipe |
| 09:49.26 | Faustov | so I used this in the meantime |
| 09:50.02 | tzafrir | it's shipped with asterisk, actually |
| 09:51.03 | Faustov | sorry for that :P |
| 10:01.30 | *** join/#asterisk stephbul (n=stephbul@bulot.org) |
| 10:07.32 | memph | hi |
| 10:08.04 | gr0mit | Sargun, I have normally used SMS providers to do everything for me |
| 10:12.16 | *** join/#asterisk datacompboy (n=datacomp@l64-93-216.static.cn.ru) |
| 10:13.34 | datacompboy | Hi everybody:) Does anybody knows any good way to record call statistics (hystogram, echo detection, etc) without call recording? I can't enable call recording, since it prohibited by law; but i need to have tech info about call quality, not only words of callers... |
| 10:14.19 | *** join/#asterisk dkirker (n=dkirker@gateway0.openmobl.com) |
| 10:16.06 | memph | I've upgraded to asterisk 1.6.0.9 thru trixbox, recreated all contacts, and then in my sip client I see all contacts available |
| 10:16.30 | memph | even offline contacts |
| 10:16.46 | memph | I don't understand why |
| 10:17.28 | memph | in my trixbox admin panel, I see the contacts that are really online, but not in the sip client |
| 10:26.12 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 10:33.20 | *** join/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 10:33.38 | madduck | how can I support SIP clients that can only do solicited MWI voicemail? |
| 10:43.12 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:43.51 | *** join/#asterisk PT_LAmb (n=asdf@89.214.17.178) |
| 10:57.44 | *** part/#asterisk ming_zym (n=ming_zym@121.0.29.237) |
| 11:05.27 | *** join/#asterisk thenthenio (n=mauro@217.221.242.43) |
| 11:05.54 | thenthenio | hello! |
| 11:06.18 | thenthenio | Are there any Italians here? |
| 11:07.58 | jblack | heh |
| 11:08.16 | jblack | Probably. |
| 11:08.39 | PT_LAmb | Hello all. I don't know if this is a support channel. But I'm trying to find out, if one can auto-configure SIP clients through options on sip.conf. on the [phones] context or on each individual phone context. |
| 11:09.35 | PT_LAmb | If that option is available for AIX phones it's an option for me. |
| 11:10.33 | thenthenio | jackal: I would like to have some advice for VOIP providers in Italy.... |
| 11:12.24 | madduck | is there any way to turn on SIP debugging for a peer before that peer comes online (i.e. before the ip/port are known)? |
| 11:12.54 | datacompboy | madduck: you can enable globally |
| 11:12.56 | jblack | pt_lamb: Most settings an be set on a per host basis, yes. |
| 11:13.13 | datacompboy | madduck: "sip set debug" and filter after |
| 11:13.57 | madduck | datacompboy: right, but the filtering after is hard with 1000+ clients. |
| 11:15.15 | datacompboy | madduck: yep. but enabled debug before peer logged in can be useful only if you want to debug registration. |
| 11:15.29 | PT_LAmb | jblack, k. searching for a reference guide on sip.conf |
| 11:15.36 | PT_LAmb | jblack, thx |
| 11:15.48 | madduck | datacompboy: right, I am debugging MWI stuff |
| 11:16.36 | madduck | we cannot get the nokia E phones to work with asterisk voicemail |
| 11:18.00 | datacompboy | madduck: may be that time to come to debug system with only one client? |
| 11:18.31 | madduck | i actually found out that most of the time, the client will re-register with the same port. ;) |
| 11:18.35 | madduck | so this works, but MWI does not. |
| 11:21.25 | madduck | http://discussion.forum.nokia.com/forum/showthread.php?s=e5dd3c798a889da66bedb0ae1fe0533e&t=109345 |
| 11:37.36 | *** join/#asterisk Malkor (n=marco@hlle-d9ba01a0.pool.mediaWays.net) |
| 11:49.53 | *** join/#asterisk mort_gib (n=mjensen@195.244.210.177) |
| 11:50.04 | *** join/#asterisk dkirker (n=dkirker@gateway0.openmobl.com) |
| 11:51.34 | mort_gib | Hi, I need some wireless phones, WiFi or dect, but good quality -Any recommendations?? |
| 11:56.43 | jblack | Go with wireless sip phones. |
| 11:56.48 | jblack | Polycom sells some, I think. |
| 11:59.03 | mort_gib | jblack: Yeah, I had a look at some Siemens ones, they looked nice and since they are Dect they have better cover, but I'm open for suggestions.... |
| 11:59.18 | mort_gib | Polycoms Kirk series look nice too |
| 12:17.58 | *** join/#asterisk levity (n=levity@unaffiliated/canuck) |
| 12:37.16 | *** join/#asterisk farkus_ (i=chatzill@72.225.212.219) |
| 12:47.19 | *** join/#asterisk waa (n=waa@201.94.132.6) |
| 12:54.04 | *** join/#asterisk balaji (n=balajibh@96-10.southernonline.net) |
| 12:54.17 | *** join/#asterisk tgunr (n=tgunr@host128-56-dynamic.52-79-r.retail.telecomitalia.it) |
| 12:54.18 | balaji | iam installing zaptel with x100p |
| 12:54.27 | balaji | i can see the card with lspci |
| 12:54.36 | balaji | but when i give ztcfg -vvv i dont see them |
| 12:55.11 | balaji | when i do make config |
| 12:55.12 | balaji | I think that the zaptel hardware you have on your system is: |
| 12:55.12 | balaji | pci:0000:05:00.0 wcfxo- 1057:5608 Wildcard X100P |
| 12:55.37 | *** part/#asterisk AdvoWork (n=AdvoWork@unaffiliated/advowork) |
| 13:10.17 | balaji | NOTICE-wcfxo: WCFXO/0: Unknown DAA chip revision: REVB=0 |
| 13:11.52 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 13:12.29 | *** join/#asterisk ghenry (n=ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 13:13.07 | ghenry | Hi, What's the best way to update a users status via jabber to inform other extension that they are on the phone? We already have BLF on the phones but would like the Jabber status changes too |
| 13:13.25 | ghenry | I'm looking at http://www.voip-info.org/wiki/view/Asterisk+Jabber but it's all about rounting depending on status. |
| 13:18.21 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 13:18.23 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 13:20.12 | *** join/#asterisk Skeeter- (i=Skeeter@c216.218.2-65.clta.globetrotter.net) |
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| 13:28.09 | *** part/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 13:30.15 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 13:30.48 | *** join/#asterisk voipmonk (n=voipmonk@67.204.57.187) |
| 13:31.51 | *** join/#asterisk TheDavidFactor (n=chatzill@fw1.safedataisp.net) |
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| 13:35.35 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 13:41.17 | *** join/#asterisk voipmonk (n=voipmonk@67.204.57.187) |
| 13:43.45 | ManxPower-work | ~answers |
| 13:43.46 | infobot | i guess answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 13:47.47 | PY8AZT | lk |
| 13:48.30 | thomas | hiho |
| 13:48.44 | thomas | is it posible on the asterisk console to show the voip client? |
| 13:49.08 | Corydon76-dig | thomas: sip show peers |
| 13:49.25 | thomas | Corydon76-dig: 53002/53002 172.16.52.50 D N 5060 Unmonitored Cached RT |
| 13:49.52 | Corydon76-dig | There you go |
| 13:49.53 | *** join/#asterisk malaiwah (n=mbelleau@64.47.115.5) |
| 13:50.02 | thomas | hm? |
| 13:50.51 | thomas | Corydon76-dig: i mean the voip client.. snom.. zoiper... |
| 13:51.10 | *** join/#asterisk ariel_ (n=chatzill@216.7.145.202) |
| 13:51.44 | ariel_ | Morning |
| 13:51.45 | thomas | aha |
| 13:51.50 | Corydon76-dig | There's no differentiation between different types |
| 13:51.52 | thomas | Corydon76-dig: sip show peer 53002 |
| 13:51.54 | thomas | <PROTECTED> |
| 13:51.55 | thomas | :-)= |
| 13:51.59 | thomas | muy bien, gracias! |
| 13:55.50 | *** join/#asterisk muiro (n=muiro@cpe-173-89-177-15.neo.res.rr.com) |
| 13:57.05 | *** join/#asterisk Tim_Toady (n=moi@77.49.8.2.dsl.dyn.forthnet.gr) |
| 14:02.15 | *** join/#asterisk _ShrikE (n=_ShrikE@74.185.215.60) |
| 14:03.37 | *** join/#asterisk sulex (n=sulex@88-149-154-95.static.ngi.it) |
| 14:03.54 | ariel_ | <PROTECTED> |
| 14:05.32 | snadge | ssshh... im hunting wabbits |
| 14:06.27 | ariel_ | and I am in the high seas watching bad weather.... |
| 14:06.54 | snadge | really.. where abouts |
| 14:06.57 | ManxPower-work | I'm hiding from the Monday People |
| 14:07.16 | snadge | ManxPower-work: isle of man? |
| 14:07.29 | ManxPower-work | snadge: Simpsons |
| 14:07.52 | snadge | maxxx power or something isnt it ? |
| 14:08.05 | ManxPower-work | Max Power, yes |
| 14:08.07 | ariel_ | I am just waiting for us to get to port so I can get off this ship and go back home.... |
| 14:08.21 | thenthenio | Hello people! |
| 14:08.28 | *** join/#asterisk cesar_CR (n=cesar@201.196.51.10) |
| 14:08.38 | ariel_ | shhh there are people watching |
| 14:10.03 | thenthenio | If I have 3 geographic numbers suplied by my ISP on VOIP do I need any special hardware on a PC with Asterisk? |
| 14:10.47 | ManxPower-work | ariel_: See what I mean? Monday People. |
| 14:10.50 | ariel_ | why special hardware? |
| 14:10.56 | ariel_ | yes |
| 14:11.05 | cjk | hi, how can i identify iax signalling traffic using iptabels (poke, authentication etc...) |
| 14:11.19 | ariel_ | ethereal |
| 14:11.27 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 14:11.40 | ManxPower-work | cjk: You can't. IAX2 uses the same ports for signalling and for audio |
| 14:12.10 | thenthenio | ariel_: I mean special telephony boards... |
| 14:12.22 | cjk | ManxPower-work, come on this is possible using string matching or things like this on iptables |
| 14:13.54 | ManxPower-work | cjk: try reading http://www.rfc-editor.org/authors/rfc5456.txt |
| 14:14.13 | ariel_ | thenthenio: asterisk to voip service via IP sip/iax2 etc to your phones either softphones or hard. Why would you need special hardware? |
| 14:15.04 | cjk | ManxPower-work, i know it uses the same port, but with iptables you can grep regexes inside packtes..... i through someone would have done something similar here |
| 14:15.27 | ManxPower-work | cjk: I wish you the best of luck. |
| 14:15.30 | voipmonk | uhmmm |
| 14:15.38 | voipmonk | you may want to look at ngrep for that |
| 14:16.00 | voipmonk | http://ngrep.sourceforge.net/ |
| 14:16.42 | thenthenio | I have good knowledge on Unix/Linux administration but I'm a newbie on VOIP... I suppesed nothing else than the PC was needed but I was not sure... |
| 14:17.01 | ManxPower-work | thenthenio: read The Book |
| 14:17.02 | ManxPower-work | ~book |
| 14:17.03 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 14:17.21 | thenthenio | Thanks ManxPower-work! |
| 14:17.25 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:17.36 | thenthenio | This is what I need! |
| 14:18.32 | thenthenio | An asterisk PC needs 2 erhernet connections at least or just one? |
| 14:19.08 | ariel_ | depends on what you want it to do, one at least |
| 14:19.10 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:19.47 | Katty | stretches |
| 14:20.38 | thenthenio | So, in the simpliest way an Asterisk server is just a common PC with Asterisk on it, right? |
| 14:20.48 | ManxPower-work | thenthenio: less talk, more reading of the book |
| 14:21.01 | ariel_ | rofl |
| 14:21.25 | [TK]D-Fender | thenthenio: that is all an * server is |
| 14:21.50 | ariel_ | wow, seems customs is looking for me.... need to get my passport and doc's ready....forgot about them. |
| 14:22.29 | thenthenio | ManxPower-work: I understand, it's just to try to have a rough idea of what I need! |
| 14:22.56 | thenthenio | [TK]D-Fender: sorry, I did not catch you... |
| 14:23.14 | ManxPower-work | thenthenio: Most of here have no interest at all in teaching people basic VoIP and Asterisk. |
| 14:23.33 | ManxPower-work | That is why many of us tell the n00bs to read the Asterisk book |
| 14:23.53 | thenthenio | I will do! |
| 14:24.00 | [TK]D-Fender | thenthenio: You only need special hardware to do special things... like plugging a physical telephony line in. |
| 14:24.47 | thenthenio | Very good, that's what I wanted somebody to tell me! |
| 14:24.50 | voipmonk | like a T1 or E1, or RJ-11, or your finger |
| 14:24.53 | ariel_ | what is special about that? to me special is plugging into an x-10 network or serial controllers to robots or |
| 14:24.56 | *** join/#asterisk fskrotzki_ (n=fskrotzk@74.74.245.250) |
| 14:25.25 | ariel_ | ananlog cards, t1 e1's are normal use items |
| 14:25.34 | ManxPower-work | Asterisk 2.0: Now with SpencerBot! |
| 14:26.35 | *** join/#asterisk jkroon (n=jkroon@uriel.interexcel.co.za) |
| 14:26.47 | thenthenio | ariel_: You are right, for you dealing with telephony they are normal! For me a normal PC includes just VGA, erhernet and USB connectivity... |
| 14:29.24 | [TK]D-Fender | thenthenio: No need for VGA... go headless.... |
| 14:29.25 | jkroon | I've got a situation where a user is allowed to make X minutes worth of calls, however, he's allowed to make this at any concurrency of his choosing. The L() option for Dial() helps with this but leaves a lot of loopholes open for exceeding that limit. |
| 14:30.20 | *** join/#asterisk Godfather_ (n=Godfathe@187.Red-83-38-229.dynamicIP.rima-tde.net) |
| 14:33.11 | thenthenio | So, as long as calls are on VOIP nothing special is needed, If I want to connect to PSTN or ISDN I need a dedicated board and if I want to route calls from residential to GSM I need a GSM board and let Asterisk do the job! |
| 14:33.38 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 14:34.41 | [TK]D-Fender | TheSure |
| 14:34.43 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:34.43 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:34.45 | [TK]D-Fender | thenthenio: Sure |
| 14:34.58 | *** join/#asterisk Stese (n=Someone@adsl.ntsols.com) |
| 14:34.59 | voipmonk | thenthenio: thats the general idea. a gsm board... maybe like the ATEUS voiceblue or similar.... how many calls did you want to send out over the gsm cell network? the voiceblue may be too underpowered... |
| 14:35.04 | thenthenio | Veeeery good! |
| 14:35.45 | ariel_ | gsm card to me = special... |
| 14:35.55 | ariel_ | well I am off to deal with US Custom see you all later. |
| 14:37.54 | Stese | Hi all |
| 14:38.01 | ManxPower-work | US Customs? We'll never see ariel again |
| 14:38.33 | [TK]D-Fender | Nothing says "I love you" like an invasive cavity search |
| 14:39.16 | Stese | Hopefully a quick question... can anyone point me in the direction of the information that should be passed in a SIP transaction for a call, from the Invite onwards |
| 14:39.44 | [TK]D-Fender | ~siprfc |
| 14:39.45 | infobot | hmm... siprfc is http://www.faqs.org/rfcs/rfc3261.html |
| 14:39.48 | [TK]D-Fender | ^^^^^^^^^ |
| 14:40.00 | Stese | Thanks... I hope I can understand it :P |
| 14:40.11 | ManxPower-work | Stese: You would be the first. |
| 14:41.47 | *** join/#asterisk kudos4421 (n=kudos442@65.88.8.2) |
| 14:41.50 | Stese | Well, I've got call transfering and non connecting Audio issues, and I want to be able to understand what shuold be in a sip debug |
| 14:42.04 | voipmonk | ~sipnat |
| 14:42.05 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:42.25 | kudos4421 | hello all. Can someone help me with voicemail configuration? |
| 14:42.30 | Stese | I hoping I don't have NAT issues... but I'll check those firstr |
| 14:43.01 | [TK]D-Fender | kudos4421: Ask a specific question, get a specific answer... |
| 14:43.03 | kudos4421 | I'm trying to disable users from being dumped into VM (from a queue) and them being able to get out of it by hitting # or * or just going crazy on their keys... |
| 14:43.20 | ManxPower-work | kudos4421: change your dialplan |
| 14:43.45 | *** join/#asterisk cuco (n=Diego@local.xorcom.com) |
| 14:43.52 | [TK]D-Fender | kudos4421: If your queues lead to VM, then thats your poor dialplan design. It should not be calling queue members that can lead to it. |
| 14:43.54 | voipmonk | kudos4421: featuremaps and application maps should help you go check 'em out |
| 14:44.26 | [TK]D-Fender | kudos4421: Also * & 0 should only excape if you allowed them to and have a matching exten in your dialplan for their respective exit points |
| 14:44.48 | [TK]D-Fender | [09:43]<voipmonk>kudos4421: featuremaps and application maps should help you go check 'em out <- WTF? |
| 14:45.18 | kudos4421 | D-Fender: will look @ dialplan |
| 14:45.42 | ManxPower-work | kudos4421: hope you are not using a GUI |
| 14:45.59 | kudos4421 | D-Fender: The queue does lead to VM only as a timeout. Is that bad somehow? |
| 14:46.32 | kudos4421 | Manx-Power: why? |
| 14:46.48 | ManxPower-work | kudos4421: you'll never figure out your dialplan if you are using a gui |
| 14:47.40 | kudos4421 | Manx-Power: I'm still using trix for most stuff, but have added custom apps and routines in the confs |
| 14:48.53 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:49.02 | [TK]D-Fender | kudos4421: Move along then.... your dialplan is not your own... #freepbx <- if they'll even support your setup |
| 14:49.34 | [TK]D-Fender | kudos4421: Or to some other trixbox specific channel/resource |
| 14:50.47 | kudos4421 | D-Fender: So, you're asuming everyone needs to be as good as you in order to be on this channel? What's wrong with learning from the trix dialplan? |
| 14:51.06 | *** join/#asterisk TommyBotten (n=noosjent@fcnoos-fw03.freecode.no) |
| 14:52.14 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 14:52.58 | *** join/#asterisk UQlev (n=yuriy@87.228.199.125) |
| 14:53.03 | jblack | kudos4421: no, it has to do with you having loads of stuff that dont come with asterisk |
| 14:53.06 | [TK]D-Fender | kudos4421: Your chages get blown away by the GUI and nobody here want's to fight with it for you. It is a cookie-cutter system that has you do things its way |
| 14:53.31 | *** join/#asterisk ehsjoar (n=ehsjoar@98.245.155.132) |
| 14:54.09 | kudos4421 | D-Fender: Can you suggest a better way short of starting from the base asterisk sample configs? |
| 14:54.24 | kudos4421 | D-Fender: I mean, is there a better gui/distro? |
| 14:56.28 | jblack | that "gui/distro" always makes learning asterisk harder |
| 14:59.03 | kudos4421 | k thx |
| 14:59.28 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:05.26 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:06.47 | ManxPower-work | ~freePBX |
| 15:06.48 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 15:07.14 | ManxPower-work | People using GUIs seem to think they are entitled to people here helping them. |
| 15:07.14 | [TK]D-Fender | kudos4421: You may jsut be configuring your setup via the GUI wrong |
| 15:07.24 | [TK]D-Fender | kudos4421: But this is not the place for supoprt on that. |
| 15:08.25 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:9c19:e836:37c1:12e0) |
| 15:08.42 | *** join/#asterisk KavanS (n=KavanS@173.50.141.22) |
| 15:11.52 | ManxPower-work | It's like someone bringing their Mac Truck to an Audi dealership for repairs |
| 15:13.11 | *** join/#asterisk Skeeter- (i=Skeeter@24.226.190.141) |
| 15:14.42 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:14.58 | *** join/#asterisk ruyo (n=psantos@195.23.253.223) |
| 15:18.35 | snadge | i use asterisk now and freepbx.. not for any particular reason.. but its my understanding that they generate asterisk configuration files |
| 15:18.55 | [TK]D-Fender | snadge: They do |
| 15:19.17 | snadge | that dont work that you cant get support in here from :p |
| 15:19.33 | [TK]D-Fender | snadge: They work. you don't |
| 15:19.57 | snadge | i still havn't figured out my inbound routing problem |
| 15:20.28 | snadge | i can place calls through it though.. so thats a start.. kind of need to do more.. but more worried about moving house and stuff.. not really work related, but still |
| 15:23.01 | snadge | its just a sip trunk and a couple of extensions.. should be relatively straight forward |
| 15:24.04 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 15:24.11 | *** join/#asterisk nfi|ermes (n=ErMeS@host54-26-static.93-94-b.business.telecomitalia.it) |
| 15:26.09 | *** join/#asterisk ddickenson_ (n=chatzill@67-198-0-5.static.grandenetworks.net) |
| 15:28.19 | ddickenson_ | I need to do some call routing based on which channel on a channelized t1 a call comes in on... I have a lineside t1 that won't pass any digits and need to be able to say something like exten => DAHDI/1-1,1,Dial(SIP/Testphone,20) but that doesn't seem to work. Any ideas on how to route a certain channel to a certain sip device? I know how to do it calling outbound but not coming in. |
| 15:29.37 | [TK]D-Fender | ddickenson_: DAHDI/1-1, is not a valid EXTEnsioN |
| 15:29.51 | [TK]D-Fender | ddickenson_: Time to read your dialplan basics... |
| 15:29.53 | [TK]D-Fender | ~stdextens |
| 15:29.54 | infobot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 15:29.56 | [TK]D-Fender | ^^^^^^^^^^ |
| 15:30.00 | ddickenson_ | I realize that, that was just the best way to explain what I was trying to do |
| 15:31.28 | ddickenson_ | if I use the "s" extension there will be no way of determining which call goes where. Every extension from that context will go to "s" and since I have no digits passing to tell it where to route from there the best I could do is send it to an IVR or something which won't work in this case |
| 15:31.31 | *** join/#asterisk Skeeter- (i=Skeeter@c216.218.2-65.clta.globetrotter.net) |
| 15:35.19 | ddickenson_ | If I call the extension associated with channel 1 on the t1 it will show up in * as DAHDI/1-1, channel 2 as DAHDI/2-1 etc. so I know that there should be a way to tell asterisk to route this according to what channel it comes in on basically a 1 to 1 kind of thing like setting up a pots line but over a t1 |
| 15:35.22 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:35.22 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:35.25 | [TK]D-Fender | ddickenson_: You should ahve relized that you should set a unique CONTEXT for each CHANNEL on your card |
| 15:36.04 | ddickenson_ | ok, so thats the only way to do it... bummer |
| 15:36.28 | ddickenson_ | thanks |
| 15:36.48 | [TK]D-Fender | ddickenson_: Or in your single "s" exten do a Goto() to an extension cut from the channel-name |
| 15:37.06 | ManxPower-work | ddickenson_: If your PSTN T-1 is 1-24 and and your PBX T-1 is channels 25-48 then you could take the channel number, add 24 and Dial the resulting channel number |
| 15:37.28 | [TK]D-Fender | ddickenson_: DAHDI/1-1 <-- chop it up |
| 15:38.13 | [TK]D-Fender | ManxPower-work: He's not pumping it through 1-1 with another PBX.... |
| 15:38.48 | ddickenson_ | actually it's a line side t1 from a nortel switch |
| 15:39.21 | [TK]D-Fender | ddickenson_: Doesn't amtter what its from |
| 15:39.31 | ddickenson_ | ok |
| 15:40.24 | *** join/#asterisk icyValk77 (n=icyValk7@213.129.64.4) |
| 15:46.10 | *** join/#asterisk ruyo (n=psantos@195.23.253.223) |
| 15:46.53 | *** join/#asterisk krdian (n=krdian@killer.radom.net) |
| 15:47.07 | *** join/#asterisk Skeeter- (i=Skeeter@190-141.cgocable.ca) |
| 15:47.11 | krdian | hi |
| 15:50.57 | madduck | mort_gib: DECT phones have *way* longer battery life. I am happy with the Gigaset C450IP but it is a bit clunky |
| 15:51.06 | madduck | just ordered my first S685IP |
| 15:51.41 | madduck | the C450IP just works though, but it has pretty much no features other than being a phone. |
| 15:51.49 | mort_gib | madduck: That was the one I was looking at |
| 15:51.53 | madduck | a PSTN one which just happens to do SIP instead |
| 15:52.12 | madduck | in that price range, it's probably second to none |
| 15:52.16 | mort_gib | madduck: It takes a bluetooth headset too |
| 15:52.22 | madduck | and you can get them <$50 on ebay |
| 15:52.26 | madduck | mort_gib: not the C450IP |
| 15:52.34 | madduck | the S685IP |
| 15:52.38 | mort_gib | madduck: No the S685 |
| 15:52.50 | madduck | right. ask me again in about 10 days. |
| 15:52.53 | mort_gib | budget is up to £250 each |
| 15:53.00 | mort_gib | Haven't got 10 days :-) |
| 15:53.04 | madduck | i am getting it on 3 dec but won't be able to play until 7 dec |
| 15:55.04 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:59.28 | *** part/#asterisk Stese (n=Someone@adsl.ntsols.com) |
| 16:04.04 | *** join/#asterisk kaldemar (n=kaldemar@unaffiliated/kaldemar) |
| 16:05.27 | balaji | x100p with zaptel having problem in my server |
| 16:05.40 | balaji | i can see the card in lspci |
| 16:05.57 | balaji | but zaptel not able to load any one here suggest what is wrong |
| 16:07.56 | madduck | did you compile the drivers? |
| 16:08.18 | [TK]D-Fender | Did you compile * AFTER having compiled Zaptel/DAHDI? |
| 16:08.58 | balaji | yes i did |
| 16:09.04 | balaji | here the errors in log "http://pastebin.ca/1683837" |
| 16:09.27 | balaji | no iam not gone till that steps of Asterisk |
| 16:09.34 | balaji | just loading zaptel first |
| 16:09.37 | ManxPower-work | balaji: you either have a bad card, have the card in an incompatible slot or you have a bad card. |
| 16:09.48 | balaji | i dont see ztcfg -vvvvv |
| 16:09.49 | ManxPower-work | sorry or an unsupported card. |
| 16:10.11 | *** join/#asterisk cnu (i=cnu@113.80-203-44.nextgentel.com) [NETSPLIT VICTIM] |
| 16:10.22 | balaji | it was working card i can say, i pulled from other server to testin in the new server |
| 16:10.28 | ManxPower-work | you won't since the driver is detecting the card, but is unable to initialize the card |
| 16:10.42 | ManxPower-work | balaji: try the card in a different slot. |
| 16:10.44 | *** join/#asterisk Svedrin (i=svedrin@glint.funzt-halt.net) |
| 16:10.49 | *** join/#asterisk Woody2143 (n=Woody214@machine76.Level3.com) [NETSPLIT VICTIM] |
| 16:11.11 | balaji | ok we will do that and get back here in 15min, thanks for the suggestion |
| 16:11.21 | ManxPower-work | remember those cards had significant compat issues and have not been manufactured in five(?) years. |
| 16:11.36 | gr0mit | mort_gib, the siemens phones are good |
| 16:12.14 | mort_gib | gr0mit: Thanks! |
| 16:12.30 | gr0mit | I have one - works well |
| 16:12.40 | mort_gib | but do I go one handset per base or am I ok to have more than one handset per base?? |
| 16:12.43 | gr0mit | are you in .uk? |
| 16:13.04 | mort_gib | Not quite... |
| 16:13.10 | mort_gib | Gibraltar/Spain |
| 16:13.37 | gr0mit | ok well if u want cheap, then go for the A58oIP |
| 16:13.45 | gr0mit | i mean A580IP |
| 16:14.04 | gr0mit | £53.04 plus VAT and shipping |
| 16:14.18 | mort_gib | I don't want cheap, I want pro quality |
| 16:14.41 | gr0mit | i can charge you more if you like ! |
| 16:14.54 | mort_gib | :-) I'm sure you can |
| 16:15.00 | gr0mit | what is the application? |
| 16:15.05 | balaji | ManxPower-work: ok just for testing iam using one line for IVR testing |
| 16:15.08 | mort_gib | A shop |
| 16:15.12 | mort_gib | M&S |
| 16:16.18 | Katty | dobedo |
| 16:16.51 | balaji | changed the slot still have the same problem |
| 16:18.09 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:18.21 | [TK]D-Fender | balaji: I did not see an actual error anywhere. |
| 16:18.29 | [TK]D-Fender | balaji: "ztcfg -vvvv" <- PB |
| 16:18.34 | ManxPower-work | balaji: It sucks to be you. |
| 16:19.04 | ManxPower-work | [TK]D-Fender: the actual error message is |
| 16:19.04 | ManxPower-work |  kernel: [11996.380042] Failed to initailize DAA, giving up...  kernel: [11996.380120] wcfxo: probe of 0000:05:00.0 failed with error -5 |
| 16:19.34 | [TK]D-Fender | \o/ |
| 16:19.40 | [TK]D-Fender | yup... could be a flakey card |
| 16:19.49 | [TK]D-Fender | Go ask for a warranty replacement! |
| 16:20.04 | balaji | [TK]D-Fender: http://pastebin.ca/1683855 |
| 16:20.13 | ManxPower-work | [TK]D-Fender: he's going to say "It worked when I removed it from the other server a few mins ago" |
| 16:20.27 | balaji | [TK]D-Fender: it was working card i pulled from other server |
| 16:20.45 | [TK]D-Fender | ManxPower-work: Open the covers and that's when static strike... *BAM*! |
| 16:21.41 | gr0mit | mort_gib, well it will be fine then |
| 16:22.12 | mort_gib | I think so, just ordering now... |
| 16:22.17 | gr0mit | you can get up to 6 handsets registered with a base |
| 16:22.26 | gr0mit | where are u ordering from? |
| 16:22.45 | mort_gib | gr0mit: Yeah, but will it work nicely?? www.wildix.com |
| 16:23.22 | mort_gib | Used to use www.voipon.co.uk and they are good, but their post sale support sucks! |
| 16:23.32 | gr0mit | wot price are they quoting? |
| 16:24.05 | mort_gib | RMA a joke and their VOIP is fine, but DON'T ask why a certain country sounds like they are on Mars |
| 16:24.22 | mort_gib | Wildix are quoting EUR 111 for one handset + base station |
| 16:24.29 | gr0mit | ouch! |
| 16:24.39 | gr0mit | pm |
| 16:35.00 | *** part/#asterisk balaji (n=balajibh@96-10.southernonline.net) |
| 16:36.00 | *** join/#asterisk ryduh_ (n=ryduh@204.16.143.186) |
| 16:36.27 | *** part/#asterisk ryduh_ (n=ryduh@204.16.143.186) |
| 16:37.46 | *** join/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 16:37.48 | drfreeze | Hello |
| 16:38.12 | drfreeze | Can someone help me with debugging a tftp boot |
| 16:38.22 | drfreeze | I'm not seeing any log files generated |
| 16:38.30 | angryuser | hi all , can someone tell me if TE420B is working with asterisk 1.2 |
| 16:38.33 | angryuser | ? |
| 16:39.25 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:40.02 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:42.48 | eppigy | HOLA |
| 16:44.01 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 16:44.35 | *** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1096762451.dsl.bell.ca) |
| 16:45.11 | dlynes | Is there a way to force spandsp to use t.38, instead of straight ulaw? |
| 16:46.16 | Chainsaw | dlynes: It will always start out on ulaw. |
| 16:46.28 | Chainsaw | dlynes: Once the CNG tone is heard, it will swap over to T.38 |
| 16:46.37 | Chainsaw | dlynes: Attempts to suppress ulaw entirely will break the negotiation. |
| 16:46.43 | dlynes | Chainsaw: comfort noise generation? |
| 16:46.52 | Chainsaw | dlynes: CalliNG tone. |
| 16:46.58 | dlynes | Chainsaw: oh |
| 16:47.12 | Chainsaw | dlynes: You know the two high-pitched tones at the start? High, higher *dual tones from here* |
| 16:47.17 | [TK]D-Fender | CNG = Comfort Noise Generation. Don't overlap standard acronyms |
| 16:47.22 | Chainsaw | dlynes: The high tone is CNG (calling fax), it answers with CED. |
| 16:47.25 | dlynes | Ok...yeah...I know what fax tones are |
| 16:47.41 | dlynes | Just never heard it called CNG before...CNG is usually comfort noise generation |
| 16:47.43 | Chainsaw | [TK]D-Fender: I didn't invent fax tone terminology. |
| 16:48.05 | Chainsaw | dlynes: *nod* An unfortunate overlap. |
| 16:48.50 | dlynes | Chainsaw: Yeah..just noticed the digium fax for asterisk has a ',z' for t.38, and the spandsp faxing doesn't |
| 16:49.31 | dlynes | On that note, is there anything fax for asterisk can do that spandsp cannot? |
| 16:49.31 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 16:51.19 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 16:52.28 | Chainsaw | dlynes: I wouldn't know, I use the spandsp infrastructure. |
| 16:52.40 | Chainsaw | dlynes: Which appears to have gained T.30 ECM support since I last used it in Asterisk 1.2 |
| 16:55.01 | *** join/#asterisk felipe_ (n=felipe@my.nada.kth.se) |
| 16:55.03 | dlynes | Chainsaw: ecm == echo cancellation module? |
| 16:55.16 | dlynes | erm error correction modulation, I mean? |
| 16:55.45 | Chainsaw | dlynes: Error correction mode, yes. |
| 16:56.16 | dlynes | Chainsaw: ah...so it might actually help on connections that are ulaw only (no t.38) |
| 16:56.27 | dlynes | Chainsaw: and on buggy timing code :) |
| 17:01.46 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:02.05 | *** join/#asterisk mnt_real (n=sinan@bas12-montrealak-1167976377.dsl.bell.ca) |
| 17:05.29 | Katty | hm. |
| 17:05.34 | Katty | i just renewed my WoW subscription |
| 17:07.21 | dlynes | Wow! |
| 17:07.32 | dlynes | What is WoW? |
| 17:07.47 | thehar | Katty: oh noes |
| 17:07.53 | thehar | another one bites the dust |
| 17:08.30 | Chainsaw | dlynes: It's like... an electronic social life. |
| 17:08.37 | Qwell | ~wow |
| 17:08.38 | dlynes | oh...weird concept |
| 17:08.47 | Qwell | Chainsaw: quite the opposite indeed |
| 17:08.54 | dlynes | like twitter on speed? |
| 17:09.15 | Chainsaw | dlynes: It's an online role-playing game that requires a monthly subscription fee. |
| 17:09.41 | dlynes | ah....not interested...but thanks for the info |
| 17:10.14 | dlynes | mud was fun, but i wouldn't pay for it...too addictive and wastes entirely way too much time |
| 17:11.12 | Chainsaw | dlynes: I don't judge. I used to be part of a Counter-Strike clan. |
| 17:13.38 | ManxPower-work | The only roleplaying I do involves leather. |
| 17:13.44 | dlynes | Chainsaw: i wasn't judging...i used to be majorly addicted to mud |
| 17:13.55 | dlynes | Chainsaw: mud == multiuser dungeon |
| 17:14.30 | dlynes | Chainsaw: wasted waaaaay too many hours on it...if i still was, i'd be divorced...chinese wives don't have much tolerance for stuff like that :) |
| 17:14.49 | Qwell | ManxPower-work: Too far. It's only Monday morning. |
| 17:15.40 | ManxPower-work | Qwell: "too far" would have included details |
| 17:15.47 | *** join/#asterisk sun28 (n=light@sun28.ipfw.su) |
| 17:15.56 | dlynes | ManxPower-work: tmi |
| 17:16.21 | ManxPower-work | at least it stopped the WOW/MUD talk |
| 17:18.26 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:19.15 | drfreeze | Ok, I am setting up a new system, and can't get the first sip phone to register |
| 17:19.35 | *** join/#asterisk bbt (n=sam@180.189.139.92) |
| 17:19.37 | drfreeze | It loads the config files, but it still unspecified |
| 17:19.49 | drfreeze | What should I be looking at to debug this phone? |
| 17:20.34 | *** join/#asterisk Assuero (n=Assuero@189.115.228.245.dynamic.adsl.gvt.net.br) |
| 17:20.36 | drmessano | ManxPower-work: I am interested in this leather of yours. I have 300 gold and a silk tunic for trade |
| 17:21.19 | Assuero | hi all |
| 17:21.36 | dlynes | drmessano: *snicker* |
| 17:22.25 | dlynes | drfreeze: start by telling us what kind of phone it is |
| 17:24.31 | Katty | thehar: yeah i know. |
| 17:24.50 | Katty | thehar: but it's cheap entertainment (= |
| 17:25.29 | *** join/#asterisk Malkor (n=marco@hlle-d9ba01a0.pool.mediaWays.net) |
| 17:25.38 | Assuero | anybody use asterisk 1.6.0.x with freepbx? |
| 17:25.46 | [TK]D-Fender | drmessano: Thems dangerous word thatr! |
| 17:25.51 | Katty | thehar: and i don't really have time to raid anymore, so i'll just be poking about |
| 17:25.59 | [TK]D-Fender | Assuero: Plenty of people |
| 17:26.06 | thehar | Katty: hehe |
| 17:26.57 | eppigy | < Katty> i just renewed my WoW subscription |
| 17:26.58 | eppigy | :< |
| 17:27.03 | ManxPower-work | Assuero: I suspect there are people that use Asterisk 1.6.0 with FreePBX, but I bet they are all on the #FreePBX channel. |
| 17:27.07 | Katty | pats eppigy |
| 17:27.11 | Katty | everything will be okay |
| 17:27.18 | eppigy | i fear the worst |
| 17:27.24 | Katty | ryan's not raiding |
| 17:27.29 | eppigy | DISCORDIA |
| 17:27.30 | Katty | or playing |
| 17:27.30 | Assuero | thanks |
| 17:27.32 | Assuero | but |
| 17:27.33 | Katty | so..you know... |
| 17:27.36 | Katty | it won't be nearly as fun |
| 17:27.39 | eppigy | oh well that is good |
| 17:27.42 | eppigy | oh |
| 17:27.43 | Assuero | the problema appear with asterisk |
| 17:27.45 | Katty | he's back into recording music |
| 17:27.59 | drmessano | drfreeze: You're in a cold, poorly lit telco closet. The humming of electricity and the stentch of fat AT&T tech permeates the air. Your phones ring with dialtone when the handsets are lifted from the base, yet your calls do not complete. You have: SIP Phone, Can of Red Bull, a Kit Kat bar, and a stale Cheetoh |
| 17:27.59 | Assuero | when play for voicemail, the CLI ends |
| 17:28.01 | Katty | he got a paul reed double something or other |
| 17:28.01 | drmessano | >> |
| 17:28.12 | Katty | and some effects peddle |
| 17:28.37 | drmessano | SOMEONE GO NORTH PLZ |
| 17:29.37 | Katty | i don't know much about music recording |
| 17:29.42 | Katty | but he's enjoying it. the neighbors are not. |
| 17:30.16 | Katty | he's been looking for something to sound proof one of the spare bedrooms. |
| 17:30.36 | Katty | anywho. i must go get my oil changed. |
| 17:30.37 | Katty | leafs |
| 17:31.53 | eppigy | im back into ragin 24x7 |
| 17:32.05 | drmessano | SOMEONE GO NORTH PLZ |
| 17:32.12 | eppigy | GOIN LONG |
| 17:32.23 | [TK]D-Fender | [12:30]<Katty>he's been looking for something to sound proof one of the spare bedrooms. <- That's what she said |
| 17:32.35 | drmessano | HAHAHAHH!!! |
| 17:32.40 | drmessano | WIN |
| 17:33.08 | raden_work | morning |
| 17:33.36 | raden_work | Katty, want to sound proof a rooom cheap and improve the harmonics ? |
| 17:34.17 | drmessano | Carpet! |
| 17:36.57 | pif | hi, what is the best option use * 1.4 with a beronet/junghanns 4BRI card? misdn, dahdi, lcr, other? |
| 17:37.23 | hardwire | ok.. asterisk 1.2.24.. if I specify a localnet to a routed subnet not a local subnet will it attempt to bind to the correct interface when bringing up the call? |
| 17:39.29 | ManxPower-work | hardwire: the OS decides the source IP of the packet |
| 17:41.05 | Katty | raden_work: ryan doesn't usually do Cheap(tm) |
| 17:42.05 | raden_work | Katty, my dads been in recording business many years |
| 17:42.17 | raden_work | Katty, thing that works best for home recording study |
| 17:42.41 | raden_work | goto walmark get the eggshell crates for beds the foam ones cover room with it :) |
| 17:43.11 | Katty | i will pass the information along. |
| 17:43.28 | hardwire | ManxPower-work: unfortunately asterisk was sending the external IP as a return address. |
| 17:43.40 | hardwire | I added it to localnets and all is well |
| 17:43.47 | [TK]D-Fender | hardwire: then you should include their localnet |
| 17:43.53 | eppigy | sup raden_work |
| 17:43.54 | hardwire | did and done |
| 17:44.18 | raden_work | eppigy, not much man got my rack built for all my equipment finally got all my machines setup and going to start on my ccna stuff soon |
| 17:44.25 | eppigy | NICE |
| 17:44.53 | raden_work | how u been ? |
| 17:45.01 | hardwire | does the happy dance |
| 17:45.11 | drmessano | CCNA: Can't Comprehend Network Administration |
| 17:45.20 | hardwire | now all our voip traffic uses our metro link instead of the 512k dsl we are using as a backup |
| 17:45.25 | hardwire | lol |
| 17:45.25 | hardwire | no more "hey.. my phone sucks" |
| 17:47.03 | *** join/#asterisk ryduh_ (n=ryduh@204.16.143.186) |
| 17:47.18 | ryduh_ | Is it possible to Dial while playing a sound file? |
| 17:47.39 | Katty | with a call file |
| 17:48.03 | voipmonk | ryduh_: running a campaign? |
| 17:48.13 | ManxPower-work | ryduh_: Yes. |
| 17:48.33 | Katty | k, oil change for reals. |
| 17:48.35 | Katty | leafs |
| 17:49.09 | eppigy | raden_work: pretty good |
| 17:49.16 | raden_work | good to hear |
| 17:49.39 | ryduh_ | I've got an asterisk system set up. when I receive a call, I play a sound and then dial, it takes too long though so I want to Dial while playing the sound |
| 17:50.01 | atis_work | ryduh_: see music on hold and M flag |
| 17:50.31 | atis_work | uhh, sorry - it's lower m flag |
| 17:50.58 | ryduh_ | The sound I'm playing is a greeting. I use the m flag already on the Dial command but it plays the hold music. |
| 17:51.14 | atis_work | ryduh_: so set the greeting as music-on-hold |
| 17:51.31 | ryduh_ | atis_work: right but what about when the greeting ends? |
| 17:51.52 | atis_work | ryduh_: make it longer.. fill end with music or just decrease dial timeout |
| 17:52.39 | atis_work | or just make greeting to be first and set your moh class for alpha-numeric order |
| 17:53.19 | ryduh_ | http://pastebin.com/d1567471b |
| 17:53.38 | ryduh_ | atis_work I'm not sure what you mean by your last post |
| 17:53.50 | atis_work | ryduh_: see sample musiconhold.conf |
| 17:54.17 | atis_work | ryduh_: you can put several files in one music class and make them always start with one that's greeting |
| 17:54.22 | *** join/#asterisk wimt (i=wimt@freenode/staff/wikipedia.wimt) |
| 17:54.27 | ryduh_ | awesome |
| 17:54.32 | ryduh_ | hmm |
| 17:54.56 | ryduh_ | I loop around though. I'd like not to have to play the message again |
| 17:55.02 | ryduh_ | any possibility of doing that? |
| 17:55.24 | atis_work | will your music end in 30 seconds which is ring time? |
| 17:55.49 | atis_work | just copy original music many times, so it never ends |
| 17:56.04 | ryduh_ | No, I put the caller onhold and loop between dialing 3 numbers until someone picks up or caller hangsup |
| 17:56.49 | atis_work | well, you can switch class when first dial is over |
| 17:57.23 | ryduh_ | but then I would loop back into it after the third call. |
| 17:57.24 | *** join/#asterisk errotan (n=errotan@81.0.115.3) |
| 17:57.45 | ryduh_ | I guess I can take the first call out of the loop and add it onto the bottom without the greeting |
| 17:57.49 | atis_work | nope.. set class to greeting before entering loop, and default-random after first |
| 18:03.29 | AndyGraybeal | so i'm sorry, i realize i'm an idiot.. but quickly.. hypethatically, i have an asterisk server, it's got a static ip from say verizon, when someone calls my phone number, how exactly is it routed onto the internet? again, i'm sorry for this question. |
| 18:03.57 | AndyGraybeal | how does my asterisk server know that someone has called my number .... does verizon handle this ? |
| 18:04.23 | voipmonk | verizon sends u the call via sip and you setup asterisk to look for it and do something about it |
| 18:04.45 | [TK]D-Fender | AndyGraybeal: What does a "phone number" have to do with your internet connection? |
| 18:04.49 | voipmonk | if verizon is where u get you r# and voip service from |
| 18:05.03 | [TK]D-Fender | AndyGraybeal: And what is this "routed onto the internet" thing you're going on about? |
| 18:06.50 | AndyGraybeal | voipmonk: awesome thank you |
| 18:07.00 | AndyGraybeal | [TK]D-Fender: i'm too ignorant to know what your asking :) |
| 18:07.56 | [TK]D-Fender | AndyGraybeal: [13:04]<[TK]D-Fender>AndyGraybeal: What does a "phone number" have to do with your internet connection? <--- this ain't Raw-Cat Sigh-Hence |
| 18:07.56 | AndyGraybeal | voipmonk: yea, we plan on getting our service from verizon |
| 18:08.20 | *** join/#asterisk moy (n=moy@74.12.130.190) |
| 18:08.24 | voipmonk | verizon has an interesting testing phase you'll love |
| 18:08.28 | AndyGraybeal | [TK]D-Fender: well you can stop making fun of me now, i think voipmonk helped me out |
| 18:09.13 | [TK]D-Fender | AndyGraybeal: Hey I asked you to clarify what it is you're asking about..... |
| 18:09.30 | [TK]D-Fender | AndyGraybeal: AndyGraybeal It was not a difficult question |
| 18:09.53 | AndyGraybeal | [TK]D-Fender: i responded with i don't know enoguh to answer, and now your making fun of me, i'm going to stop talking with you for now. |
| 18:10.22 | voipmonk | next time, AndyGraybeal , just clarify - TK doesnt assume like I just did. |
| 18:10.49 | [TK]D-Fender | AndyGraybeal: You can't describe anything more that I have a phone number (its not delivered over any tech you can describe? Or from a vendor you can name?), and what relationship should it have to your internet connection? |
| 18:10.57 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 18:11.08 | [TK]D-Fender | ~assume |
| 18:11.09 | infobot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav It makes an (ass) out of (u) and (me) |
| 18:11.13 | [TK]D-Fender | voipmonk: ^^^ correct |
| 18:11.42 | AndyGraybeal | okay anyway - i don't know enough to know what i'm talking about so i'm glad that voipmonk was helpful. |
| 18:12.00 | AndyGraybeal | i explaine that i was sorry for the question in the first place. |
| 18:12.07 | *** join/#asterisk levity (n=levity@unaffiliated/canuck) |
| 18:12.27 | ManxPower-work | I suspect most of us assume "version" means POTS or PRI |
| 18:12.30 | [TK]D-Fender | AndyGraybeal: You seem to know you have a phone number, yet not enough to know what its supposed to be delivered over? |
| 18:12.33 | ManxPower-work | ..er..VeriZon. |
| 18:12.45 | ManxPower-work | [TK]D-Fender: he doesn't want our help. |
| 18:12.58 | [TK]D-Fender | ManxPower-work: I might start believing that... |
| 18:14.30 | ManxPower-work | [TK]D-Fender: he doesn't know enough to be successful at most anything with Asterisk. I figure that is punishment enough for wasting everyone's time. |
| 18:15.02 | [TK]D-Fender | ManxPower-work: Well I said nothing of the sort.... I'm just trying to validate what I'm being told... |
| 18:15.44 | ManxPower-work | [TK]D-Fender: but he didn't actually tell us anythinh |
| 18:16.05 | [TK]D-Fender | ManxPower-work: So far he has a phone number and and internet connection :) |
| 18:17.15 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:17.25 | [TK]D-Fender | ManxPower-work: Somehow a call to his number is supposed to be 'routed onto the internet". Is that via his setup? Or is that number supposed to be sent to his server via the provider being an ITsP? I'm not about to guess. |
| 18:19.29 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 18:20.36 | ManxPower-work | I'm starting to think SugarCRM is actually a Microsoft product. |
| 18:21.14 | *** join/#asterisk bbt (n=sam@180.189.139.92) |
| 18:22.10 | drmessano | Nope |
| 18:22.25 | drmessano | It didnt tell me I needed to have a genuine copy of windows to run it |
| 18:26.22 | ryduh_ | lol |
| 18:30.25 | jblack | Katty gave me facebook herpes. |
| 18:31.02 | jblack | A truly evil individual. |
| 18:31.58 | ryduh_ | hmm. So I have two files in my /var/lib/asterisk/moh directory however whenever I put myself on hold, One of two things will happen. No music will play, or one of the files will repeat over and over |
| 18:32.30 | jblack | before we go anywhere, how closely did you read the docs on moh? |
| 18:32.37 | jblack | In particular, the part about extensions |
| 18:36.20 | jblack | he must be reading it now. ;) |
| 18:36.46 | ryduh_ | jblack: I just took a look at that part again. I moved the files into a new directory and switched my musiconhold.conf file to this: http://pastebin.com/d58e1836c |
| 18:37.02 | ryduh_ | and now, the other file will play, or no music will play |
| 18:37.20 | jblack | what about the extensions? |
| 18:37.41 | jblack | can you pb an ls of that directory? |
| 18:38.26 | ryduh_ | http://pastebin.com/d7c54224c |
| 18:38.47 | ryduh_ | Did I miss something about extensions? |
| 18:39.37 | *** join/#asterisk qdk (n=qdk@0x573c2220.bynqu1.dynamic.dsl.tele.dk) |
| 18:45.30 | ryduh_ | jblack: Am I totally missing the point here? |
| 18:46.35 | jblack | Well, asterisk, when presented a pile of files with the same name, excepting the extension, picks the "best" one. |
| 18:46.50 | jblack | is looking at your pastebin now |
| 18:47.03 | jblack | Ok. Sometimes one plays, sometimes the other? |
| 18:47.09 | ryduh_ | sometimes one plays |
| 18:47.15 | ryduh_ | sometimes neither |
| 18:47.32 | jblack | which one plays? |
| 18:47.36 | ryduh_ | reno |
| 18:48.03 | jblack | can you put mc_thanks.gsm up on the intarweb, perhaps filebin.ca ? |
| 18:49.30 | ryduh_ | http://filebin.ca/qsmxrt |
| 18:50.00 | *** join/#asterisk Godfather_ (n=Godfathe@245.Red-88-12-204.dynamicIP.rima-tde.net) |
| 18:53.07 | jblack | yeah, that's a gsm, 8khz, 16kbit |
| 18:53.11 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:54.03 | ryduh_ | I used Record() to record that |
| 18:54.22 | jblack | Yeah, it's perfectly fine. |
| 18:54.52 | jblack | I don't suppose you have a small pile of mp3s laying around, that we can use for testing purposes? |
| 18:56.18 | jblack | I can presume that you do -not- have a [default] moh stanza, and that you _are_ setting musiconhold to being ... native-random ? |
| 18:56.30 | ryduh_ | jblack: I've got a 1 minute mp3 from an outside source |
| 18:56.51 | jblack | and that you know how to reload the musiconholdconfig? |
| 18:57.01 | jblack | I.E. can I safely assume no week 1 mistakes? |
| 18:57.24 | ryduh_ | jblack: I do have a default but I am using m(native-random) on dial |
| 18:57.44 | ryduh_ | I've reloaded moh config and restarted * as well to no avail |
| 18:57.49 | *** join/#asterisk qdk (n=qdk@87.61.141.249) |
| 18:59.17 | jblack | I'd test with an actual musiconhold, as in setting moh to native-random and putting someone on hold. |
| 18:59.24 | jblack | just to double check. |
| 18:59.31 | jblack | cuase otherwise, I'm out of ideas. |
| 18:59.55 | jblack | It _could_ be that your phone doesn't like out of band audio -- audio before an answer. |
| 19:02.09 | ryduh_ | using musiconhold() plays the reno file 3 out of 3 times |
| 19:02.29 | ryduh_ | well, I answer it and then do another dial. Is that what you mean? |
| 19:03.16 | jblack | Well, depending on circumstance, calling answer may or may not be approprirate. |
| 19:03.33 | jblack | did you put a wait before answer to let the line settle? |
| 19:03.40 | ryduh_ | Ah, nope the 4th time it play no mus |
| 19:03.43 | ryduh_ | music |
| 19:03.44 | jblack | pardon, after answer. |
| 19:03.53 | ryduh_ | jblack: no. I will right now |
| 19:04.13 | jblack | I don't think that's your problem. t this pint, I don't know whwat it is |
| 19:05.13 | ryduh_ | yeah adding a wait after answer didn't change anything |
| 19:06.01 | jblack | good luck |
| 19:06.17 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
| 19:10.02 | Katty | returns |
| 19:10.10 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 19:11.11 | Katty | jblack: i did not make scary faces at the kid. |
| 19:11.18 | Katty | jblack: that's the last thing the poor child needs :P |
| 19:11.30 | Katty | jblack: seeing burly old men at the automotive place is probably traumatizing enough! |
| 19:11.59 | jblack | You should have. |
| 19:12.35 | jblack | a big ugly face and wide sweeping motions of the arms. THen mommie would have to take care of him! |
| 19:12.47 | jblack | By traumatizing him, you'd be saving him! |
| 19:13.08 | Katty | mother probably can't afford day care. |
| 19:13.18 | Katty | and mother staying at home with him would probably do more damage. |
| 19:13.27 | Katty | it's just bad juju |
| 19:13.37 | jblack | The word "desperate" seems to ring a bell in my head... |
| 19:14.46 | jblack | I want to make the whole intarweb my friend. |
| 19:16.23 | *** join/#asterisk retentiveboy (n=pdugas@rrcs-70-63-227-166.midsouth.biz.rr.com) |
| 19:19.10 | *** join/#asterisk goodjoke (i=1827a8fa@gateway/web/freenode/x-iwcmafgpmzsauqwd) |
| 19:29.53 | Katty | finally |
| 19:29.56 | Katty | i get to eat my banana. |
| 19:30.32 | muiro | that's what she said? |
| 19:30.59 | Katty | no |
| 19:31.02 | muiro | oh... |
| 19:31.04 | muiro | okay |
| 19:31.50 | ManxPower-work | "FreePBX/Trixbox is to Asterisk as Windows 95 is to DOS. " |
| 19:32.08 | muiro | lol |
| 19:34.40 | *** join/#asterisk haryv (i=lanny@174.1.114.16) |
| 19:36.28 | *** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1242444445.dsl.bell.ca) |
| 19:36.45 | dlynes | Anyone know what module addqueuemember is in? |
| 19:37.28 | dlynes | I've tried loading app_queue.so, but it doesn't seem to give me that application |
| 19:37.49 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 19:37.50 | haryv | Found or at least identified why asterisk does not respond when the external firewall is off. Why in the world, would a failed sip registration cause significant huge delays in asterisk responce times or, no responce from asterisk at all when wanting to make a internal pots call? |
| 19:40.02 | ManxPower-work | haryv: known issue where Asterisk blocks when it can't do a reverse lookup of the hostnames associated with the local interface IPs. |
| 19:40.24 | ManxPower-work | put all IPs on your Asterisk server in /etc/hosts and make sure the OS consults hosts before DNS |
| 19:41.05 | haryv | thats the antaganizing issue I have had for a very long time! |
| 19:41.36 | haryv | seems no one had a answer till now! |
| 19:42.19 | haryv | I wonder how many other network may not have this configuration untill there dsl goes down :) |
| 19:42.56 | haryv | mmm everything is dhcp |
| 19:43.38 | ManxPower-work | haryv: This has been an issue since Asterisk 0.65 (where I first discovered the issue) |
| 19:44.01 | haryv | if that was the case, no one has provided a answer even here. |
| 19:44.25 | ManxPower-work | haryv: it's been talked about on the mailing lists |
| 19:51.18 | *** join/#asterisk haryv (i=lanny@174.1.114.16) |
| 19:52.01 | haryv | Manx, found one link so..will see if it is the answer to this issue. |
| 19:59.33 | ManxPower-work | haryv: I gave you the answer. 8-) |
| 20:04.05 | haryv | yes I know. |
| 20:04.33 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.227.159.dsl.dyn.forthnet.gr) |
| 20:04.38 | haryv | but, how do you make it point to localhost vs resolv.conf? |
| 20:04.44 | *** join/#asterisk Alagar (n=Administ@122.164.104.235) |
| 20:05.04 | ManxPower-work | <PROTECTED> |
| 20:05.42 | ManxPower-work | your DHCP client should be able to update /etc/hosts for IPs you get via DHCP |
| 20:06.56 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 20:07.59 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 20:08.49 | haryv | http://pastebin.com/m27c06d18 what I have that has not been changed. |
| 20:11.51 | haryv | discussion on it http://www.trixbox.org/forums/trixbox-forums/open-discussion/asterisk-and-loss-internet |
| 20:11.53 | haryv | :) |
| 20:14.50 | *** join/#asterisk sulex (n=sulex@host-78-14-173-96.cust-adsl.tiscali.it) |
| 20:18.01 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:22.01 | *** join/#asterisk yosi1234 (i=Bmw_Toro@99.226.209.33) |
| 20:22.13 | yosi1234 | hi all |
| 20:22.40 | yosi1234 | I have Accidently DELETED a Queue from my list of Queues...Yikes! I didn't "Apply Configuration" yet, is there anyway to bring it back???? pleaze help! |
| 20:22.43 | Katty | omnomnomnoms sunflower kernels |
| 20:22.51 | yosi1234 | any ideas? |
| 20:22.58 | Katty | yosi1234: there is no Apply Configuration button |
| 20:23.07 | *** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com) |
| 20:23.10 | Katty | yosi1234: maybe you meant to be in #trixbox or #freepbx |
| 20:23.11 | yosi1234 | sorry i'm in pbx in a flash |
| 20:23.17 | yosi1234 | butno one is around in their channel |
| 20:23.35 | yosi1234 | i will try trixbox... |
| 20:26.10 | *** part/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:26.15 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:26.18 | ManxPower-work | haryv: and those 4 IPs are the ONLY ones on that host (verify with ifconfig)? |
| 20:27.15 | ManxPower-work | haryv: try searching the mailing lists: http://www.google.com/search?hl=en&client=firefox-a&rls=org.mozilla%3Aen-US%3Aofficial&hs=Me0&q=site%3Alists.digium.com+%22internet+down%22+sip&aq=f&oq=&aqi= |
| 20:27.19 | ManxPower-work | ~mailinglist |
| 20:27.20 | infobot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 20:33.40 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 20:39.51 | Katty | frowns |
| 20:40.17 | Katty | i can deal with people forwarding me spam inside this office |
| 20:40.38 | Katty | but now they are forwarding their prayer requests for thanksgiving to the global list. |
| 20:41.41 | hardwire | yeh |
| 20:41.45 | hardwire | a friends workplace is doing that as well |
| 20:41.52 | hardwire | he sent in "I'm thankfull she was 18 after all.. phew" |
| 20:42.58 | *** join/#asterisk thenthenio (n=thenthen@93-36-213-81.ip62.fastwebnet.it) |
| 20:43.16 | thenthenio | Hello! |
| 20:43.28 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 20:44.30 | thenthenio | On the AsteriskNOW page I did not find any requirements for the PC, what are the minimum PC reqirements? |
| 20:45.31 | haryv | back |
| 20:46.09 | *** part/#asterisk haryv (i=lanny@174.1.114.16) |
| 20:46.11 | Katty | thenthenio: two tin cans, and a string. |
| 20:46.30 | ManxPower-work | I wish pidgin had a /ignore option |
| 20:47.02 | Katty | ManxPower-work: sounds serious. |
| 20:47.35 | ManxPower-work | Katty: just all these people that KNOW this is not the right place for GUI questions and still come back asking them. |
| 20:47.50 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 20:48.05 | Katty | ManxPower-work: i imagine they come back because they get answers. |
| 20:48.20 | ManxPower-work | Katty: and THAT is the saddest thing of all. |
| 20:48.22 | Katty | ManxPower-work: if the answers stopped, they'd probably stop coming back |
| 20:48.24 | thenthenio | Just to have take a trip on it, is it possible not to have a VOIP connection and try it as a switchboard for internal stations only (PCs with softphone)? |
| 20:48.35 | ManxPower-work | Katty: Belive me, they would still be coming back. |
| 20:49.16 | Katty | ManxPower-work: thta doesn't seem very logical. |
| 20:49.23 | ManxPower-work | "But nobody is answering on TrixBox!". "Well maybe that should tell you something about trixbox" |
| 20:49.41 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 20:49.41 | ManxPower-work | Katty: They are GUI users, you can't expect them to be logical. |
| 20:50.14 | ManxPower-work | Katty: if I had my way we'd kick/ban them all |
| 20:50.37 | Katty | ManxPower-work: i'm sure the requiring registration helps |
| 20:50.51 | *** join/#asterisk Buklov (n=buklov@213.138.71.254) |
| 20:52.55 | ManxPower-work | "But FreePBX is Asterisk". "Windows 95 is DOS. What is your point?" |
| 20:53.36 | Katty | hugs ManxPower-work |
| 20:53.48 | *** join/#asterisk PabloM (n=user@190.245.131.114) |
| 20:54.03 | PabloM | hi |
| 20:54.06 | Katty | hi |
| 20:54.28 | PabloM | does anyone know how to do this? Dahdi/1-1/*31#55555555 |
| 20:55.23 | ManxPower-work | PabloM: remove the -1 |
| 20:55.45 | ManxPower-work | I assume by "do this" you means "dial this". |
| 20:56.10 | PabloM | ManxPower-work: yes, Dahdi/1/*31#55555555 doesn't work either |
| 20:56.39 | PabloM | ManxPower-work: the problem is the *31# prefix |
| 20:56.43 | Katty | ManxPower-work: not only does it Not Work |
| 20:56.49 | Katty | ManxPower-work: it Doesn't Work At All |
| 20:57.13 | Katty | PabloM: then take it off |
| 20:57.17 | russellb | and by doesn't work at all, he means that the server burst into flames |
| 20:57.30 | Katty | i'll get the marshmallows. |
| 20:57.33 | ManxPower-work | PabloM: If that doesn't work then your telco is not accepting it. |
| 20:57.51 | ManxPower-work | Remember you can't just dial the analog codes on PRI or SIP and expect them to work |
| 20:59.50 | PabloM | ManxPower-work: thx, I didn't know that |
| 21:07.26 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 21:16.01 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 21:16.45 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 21:16.46 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 21:17.36 | Katty | yawns |
| 21:25.39 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 21:26.12 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 21:28.59 | *** join/#asterisk ChUbB (n=IceChat7@62-31-213-230.cable.ubr12.aztw.blueyonder.co.uk) |
| 21:29.46 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
| 21:29.52 | iq | Hi |
| 21:31.34 | lesouvage | iq: good evening |
| 21:35.23 | *** join/#asterisk estranger (i=russ@russ.trifecta.com) |
| 21:37.44 | estranger | any suggestions for an Indian DID provider? the two companies I use dont offer india (link2voip / flowroute) |
| 21:40.08 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:40.08 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:40.11 | *** part/#asterisk iq (n=iq@unaffiliated/iq) |
| 21:42.34 | *** join/#asterisk delvin (n=delvin@200-251-138-94.poolip.BHE.embratel.net.br) |
| 21:47.29 | Katty | infobot: itsp-uk? |
| 21:47.38 | Katty | infobot: itsp? |
| 21:47.39 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 21:48.10 | Katty | estranger: you might try either of those lists...some of the providors might service India..not sure |
| 21:48.38 | estranger | ill click around.. i guess there used to be laws in india forbidding this, but was lifted last year? figure thats why its hard to find them |
| 21:48.46 | estranger | ~itsplist-us |
| 21:48.47 | infobot | extra, extra, read all about it, itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 21:50.10 | *** join/#asterisk Whtsup (n=sssi@WimaxUser372-235.wateen.net) |
| 21:50.18 | Whtsup | hello |
| 21:50.22 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 21:50.36 | Whtsup | how is everyone |
| 21:51.06 | estranger | im swell, and dandy |
| 21:52.41 | Whtsup | res_agi.c:2203 deadagi_exec: Running DeadAGI on a live channel will cause problems, please use AGI |
| 21:52.54 | Whtsup | how can i solve this problem |
| 21:54.20 | jblack | Don't use DeadAGI in your dialplan. |
| 21:54.36 | Whtsup | im using a2billing |
| 21:54.45 | Whtsup | and dialplan is using deadagi |
| 21:54.54 | jblack | There you go, you know what to fix. |
| 21:55.22 | jblack | Lucky you that there's a big neon sign pointing at what needs your attention. =) |
| 21:57.36 | phix | Morning |
| 21:58.29 | Katty | estranger: no idea. don't live there. |
| 21:58.50 | Katty | i have a very acute case of the sleepies. |
| 21:58.53 | *** join/#asterisk raden_work (n=jon@69.179.99.17) |
| 21:59.30 | Katty | hi raden |
| 22:01.58 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 22:02.39 | Katty | hi fender |
| 22:06.53 | [TK]D-Fender | Katty: Mew. |
| 22:07.57 | beek | Afternoon [TK]D-Fender ... Helloooooo Katty! |
| 22:09.25 | Katty | hi beek :> |
| 22:09.42 | Katty | jblack: mushroom and chicken stirfry tonight |
| 22:10.39 | beek | Katty: sounds tasty. |
| 22:10.58 | Katty | beek: hopefully it will be (= |
| 22:13.11 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:15.42 | *** join/#asterisk bpgoldsb (n=bpgoldsb@209.208.68.1) |
| 22:15.48 | *** join/#asterisk bannabob (n=bannabob@dsl081-049-016.sfo1.dsl.speakeasy.net) |
| 22:16.50 | *** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1242444445.dsl.bell.ca) |
| 22:18.19 | raden_work | hi Katty |
| 22:18.28 | raden_work | hugs Katty |
| 22:19.00 | Katty | hugs raden_work |
| 22:20.45 | raden_work | eppigy, how hard is it to make it so i can access a internal FTP server via external IP through ASA ? |
| 22:21.15 | [TK]D-Fender | raden_work: Sounds like a single firewall rule on just about any router |
| 22:22.50 | raden_work | well i suppose it would just be port route |
| 22:25.56 | Godfather_ | I'm trying to register with my mobile using Fring to *. I enabled the debug on sip. (sip.conf -> http://pastebin.com/m3f18ee40 ) I enter to fring username: 105@10.1.1.3 and proxy direction: SIP:10.1.1.3, but i get nothing into the debug |
| 22:26.34 | Godfather_ | i can ping the phone |
| 22:26.47 | *** join/#asterisk ricdanger (n=bla@193.137.26.210) |
| 22:26.50 | ricdanger | hi |
| 22:26.55 | Katty | hi |
| 22:27.00 | ricdanger | I'm having a problem. For some reason, asterisk shows wrong hints |
| 22:27.01 | Katty | how're you |
| 22:27.10 | ricdanger | for example: one extension is idle but asterisk says "InUse&Ringing" |
| 22:27.18 | ricdanger | any idea what can be the problem? |
| 22:28.23 | Katty | eppigy: |
| 22:28.31 | Katty | ^- http://www.philosophy.com/images/products/lg/prodlg_00650510.jpg |
| 22:28.37 | Katty | eppigy: DO YOU SEE IT?! |
| 22:30.20 | Katty | eppigy: they don't make it in 16 oz :< |
| 22:32.45 | *** join/#asterisk voipmonk (n=voipmonk@67.204.57.187) |
| 22:32.56 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:35.00 | *** join/#asterisk lanning (n=lanning@208.87.235.224) |
| 22:35.11 | bannabob | I have a DID provider that wants me to authenticate by IP address. However their calls originate from a whole class C. Whats the best way to solve this? |
| 22:35.47 | Chainsaw | bannabob: Will they not accept CIDR masks? |
| 22:37.35 | [TK]D-Fender | bannabob: You authenticating to them via IP has nothing to do with the IP they send YOU call from |
| 22:38.14 | bannabob | They are authenticating to me via IP. |
| 22:38.21 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) |
| 22:38.40 | Katty | bannabob: please tell me you have a public, static, ip address |
| 22:38.52 | [TK]D-Fender | bannabob: therefor your IP should be fixed. Doesn't mean the server sending you calls has to be |
| 22:39.57 | Katty | [TK]D-Fender: come make me dinner. i think i'm too tired to cook |
| 22:40.12 | [TK]D-Fender | Katty: I'm already done with mine. |
| 22:40.14 | bannabob | Public, static. |
| 22:40.26 | Katty | [TK]D-Fender: good. now you can start on mine. |
| 22:40.39 | [TK]D-Fender | Katty: Grab pan. Oil. Add steak. Apply heat. Flip twice. Salt. Pepper. Eat |
| 22:40.42 | Katty | [TK]D-Fender: i'll be home in 20 minutes :P |
| 22:40.56 | Katty | i don't like steak. much. |
| 22:41.23 | Katty | but i might be able to manage to start some rice |
| 22:41.42 | [TK]D-Fender | Katty: the rest of my diet is even easier :) |
| 22:44.38 | bannabob | My layout is this. I have a provider (vitelity) that provides several hundred DID's. I set an IP for them to send the calls to. The calls go to an opensips server that loadbalances to several asterisk servers. When I get an inbound call at the asterisk server it has an any ip from within a /24. So other then adding an entry for each ip I cant see a better solution. 254 entries seems to be a management mess. |
| 22:45.43 | war9407 | /j #debian |
| 22:45.48 | war9407 | er, oops |
| 22:46.13 | [TK]D-Fender | bannabob: "host=dynamic" |
| 22:46.31 | bannabob | Then I have to register with them dont I? |
| 22:46.44 | jblack | bannabob: If it's something you do only once... why not bite the bullet? |
| 22:47.10 | jblack | brute force may be an inelegant approach, but it's a viable, useful one from time to time |
| 22:47.56 | bannabob | jblack, cause when they add a new range its a pain and add a potential downtime issue. |
| 22:48.33 | jblack | maybe you should bite the bullet now and migrate to an internal class b. |
| 22:49.01 | jblack | I'm sure you can make a nifty agi to automate things and forgo registration |
| 22:49.04 | *** join/#asterisk lordmortis (n=lordmort@124.169.106.150) |
| 22:49.06 | [TK]D-Fender | bannabob: What did you think "authenticating by IP meant? That's precisely the point. So they can send calls to your IP without have to have you register |
| 22:49.19 | jblack | there's no such thing as authentication by ip. |
| 22:49.28 | [TK]D-Fender | jblack: AGI? ..... silly ... |
| 22:49.37 | voipmonk | chuckles |
| 22:49.40 | bannabob | jblack, agi would be over kill |
| 22:49.43 | jblack | AgI = the duct tape of the world. :) |
| 22:50.06 | jblack | stretches out a few feet^Wlines of code to prove his point |
| 22:50.08 | russellb | if I actually wrote a lot of dialplans, I'd probably do them all in AGI. |
| 22:50.59 | jblack | I keep dead simple stuff in dialplans, and jump to agi at the first hint of complexity. |
| 22:51.32 | bannabob | [TK]D-Fender, sure but i have to have a host= line in the sip.conf to accept calls from. Thats the issue I cant put host=xxx.xxx.xxx.xxx/24 in the sip.conf |
| 22:51.57 | [TK]D-Fender | [17:46]<[TK]D-Fender>bannabob: "host=dynamic" <-------------------------- |
| 22:52.30 | Chainsaw | [TK]D-Fender: If only IRC had marquee tags. |
| 22:52.39 | bannabob | jblack, actually I do use alot of heavy AGI but keeping it in the extensions.conf makes it much faster and uses less resources on the server. |
| 22:53.00 | jblack | That's usually true |
| 22:53.02 | [TK]D-Fender | Chainsaw: module load res_bigfuckingneonarrow.so :p |
| 22:53.06 | jblack | I'll give you that. |
| 22:53.20 | jblack | Of course... cpu cycles are practicly free these days. =) |
| 22:53.38 | jblack | I found a megahertz in my box of cheerios this morning. |
| 22:53.44 | [TK]D-Fender | jblack: AGI is for call processing... what would a call have to do with replacing a REGISTER? |
| 22:54.09 | jblack | You'd agree that part of call processing is call routing, correct? |
| 22:54.28 | bannabob | [TK]D-Fender, but I have to use 5 a registration line. And it seems that the vitelity reg. server seems to have a fit if I try to reg more then 5 servers at once. |
| 22:54.32 | Katty | wants to know where jblack gets his cheerios |
| 22:54.59 | jblack | katty: From the Booleans down the store. The cheri-0s, and cheri-1s. |
| 22:55.12 | jblack | s/store/street |
| 22:55.15 | Katty | you're so funny |
| 22:55.19 | *** join/#asterisk kleofas (n=kleofas@chello089079030123.chello.pl) |
| 22:55.30 | jblack | You should see me when I have my clown nose on. |
| 22:56.43 | [TK]D-Fender | bannabob: you shouldn't be registering from the sound of things. |
| 22:56.50 | jblack | What sort of phones do you have? Are they the ones that are smart enough to download their configuration according to their mac address? |
| 22:57.02 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 22:58.04 | Katty | fischer price |
| 22:58.16 | bannabob | [TK]D-Fender, i dont want to register. I was under the impression that if I put dynamic then I had to register. But are you saying that I can put dynamic without a register line? If thats the case I can jsut use an ACL then. |
| 22:58.38 | Katty | time to scoot. later folks. |
| 22:58.42 | jblack | Have fun. |
| 22:58.50 | jblack | I think i need to go kill some zombies. |
| 22:59.06 | bannabob | jblack, what phones do I use? |
| 23:00.58 | bannabob | BTW I am using 1.4 not 1.6 |
| 23:01.58 | [TK]D-Fender | bannabob: Correct |
| 23:02.20 | jblack | aaahhh... FB is broken. |
| 23:02.24 | bannabob | ok that makes it easy then. |
| 23:02.36 | bannabob | FB should always be broken.... |
| 23:03.16 | jblack | if FB dies, I'm going to live on your couch and eat all your food until I gas up your apartment so bad that everyone dies. |
| 23:04.31 | bannabob | That would be different from now, how? |
| 23:05.12 | jblack | look at your couch. Notice that I'm not on it. |
| 23:05.40 | jblack | Now imagine your couch with me on it. eating all your food. converting all of your oxygen into co2 and methane. |
| 23:07.06 | [TK]D-Fender | jblack: You suck at phone sex.... |
| 23:07.07 | bannabob | Sorry, my mistake thats my Bull Mastif on the couch.... |
| 23:07.12 | [TK]D-Fender | goes to play another game... |
| 23:07.32 | bannabob | BTW, thanks for the help [TK]D-Fender |
| 23:07.53 | [TK]D-Fender | bannabob: You're welcome |
| 23:12.21 | *** join/#asterisk aiksa[LV] (n=aiksa[LV@mx.fiveplus.lv) |
| 23:12.25 | aiksa[LV] | Hi everybody |
| 23:12.58 | aiksa[LV] | I have a long time ago seen a sample of sip configuration file where entries were based on a templates |
| 23:13.17 | aiksa[LV] | like there was a template for generic Snom, ... zoiper, whatever |
| 23:13.31 | aiksa[LV] | can anyone nudge me in a right direction for the documentation of this |
| 23:13.42 | aiksa[LV] | kind of inheritence |
| 23:15.26 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 23:15.58 | aiksa[LV] | no need anymore. |
| 23:16.13 | aiksa[LV] | foun it. and it appears that it is asterisk wide option - http://www.voip-info.org/wiki/view/Asterisk+config+template |
| 23:16.16 | aiksa[LV] | oh joy |
| 23:24.26 | hardwire | anybody ever heard of asterisk sending the same packet twice? |
| 23:24.50 | hardwire | rtp |
| 23:24.52 | hardwire | every send |
| 23:25.09 | ecrane | Can anyone recommend a good online guide to SIP? I'm hoping for something friendlier then the RFC. |
| 23:25.36 | aiksa[LV] | hardwire: under what circumstances, what is used for signaling? |
| 23:26.47 | hardwire | aiksa[LV]: no worries.. I just verified that it's a tcpdump issue with vlan offloading |
| 23:27.55 | hardwire | I was seeing a combination of asterisk sending more often than the phone was as well as each asterisk packet being duplicated against the vlan root device and the vlan device. |
| 23:27.56 | aiksa[LV] | hardwire: ok. fine then :) |
| 23:27.59 | hardwire | so.. it was confusing |
| 23:28.15 | hardwire | now I want to know why asterisk is sending smaller packets more often |
| 23:28.16 | hardwire | digs |
| 23:28.19 | aiksa[LV] | ecrane: I cant recommend any definitive guide |
| 23:28.53 | aiksa[LV] | although i suppose that google search for "SIP basics" or "SIP 101 " or the corresponding wikipedia entry should be good enough to start |
| 23:29.27 | aiksa[LV] | nevertheless as you will move forward this dark forest, you cant avoid rfc |
| 23:29.47 | ecrane | Ok, thanks. I'll try google after I'm done with the bing results. |
| 23:30.03 | aiksa[LV] | hardwire: if i am not mistaken it can be set as an rtp parameter |
| 23:30.11 | aiksa[LV] | how many voice frames to send in a packet |
| 23:30.24 | aiksa[LV] | but I may be wrong |
| 23:30.42 | hardwire | indeed |
| 23:33.38 | aiksa[LV] | indeed it does or - indeed I was wrong |
| 23:33.39 | aiksa[LV] | ?:) |
| 23:34.32 | hardwire | indeed |
| 23:36.07 | *** part/#asterisk ryduh_ (n=ryduh@204.16.143.186) |
| 23:39.27 | aiksa[LV] | :) |
| 23:41.44 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-74-219-221-11.central.biz.rr.com) |
| 23:42.16 | *** join/#asterisk voipmonk (n=voipmonk@67.204.57.187) |
| 23:42.30 | aiksa[LV] | hardwire: I on the other hand am having a whole lot more interesting thing going on at the moment |
| 23:42.33 | aiksa[LV] | :)) |
| 23:43.07 | hardwire | lemme hear it |
| 23:43.27 | aiksa[LV] | I am trying to flood our dedicated vlan with a telco to see if there are any caps/limitation set on the connection |
| 23:43.58 | aiksa[LV] | I kinda suspect there are - they are saying there is not |
| 23:44.15 | hardwire | iperf? |
| 23:44.16 | hardwire | mz? |
| 23:44.20 | aiksa[LV] | nope |
| 23:44.24 | hardwire | why not? |
| 23:44.28 | aiksa[LV] | two nc instances piped togeather |
| 23:44.41 | aiksa[LV] | on both ends, forming the loop |
| 23:44.42 | hardwire | udp? |
| 23:44.50 | aiksa[LV] | mhm |
| 23:44.55 | aiksa[LV] | that was -yes |
| 23:45.12 | hardwire | no.. no it wasn't |
| 23:45.22 | hardwire | I'd use mz :) |
| 23:45.38 | aiksa[LV] | ok |
| 23:45.41 | hardwire | that way you can at least measure jitter |
| 23:45.47 | hardwire | and completely waste the pipe |
| 23:45.53 | aiksa[LV] | the problem is that i have tdmoe running over that line |
| 23:46.10 | hardwire | that doesn't sound like a problem if you're trying to kill it |
| 23:46.17 | aiksa[LV] | and as soon as I try to squeeze more that 4 E1 |
| 23:46.24 | hardwire | ah |
| 23:46.27 | aiksa[LV] | more than 4 E1 |
| 23:46.32 | hardwire | using redfone stuff? |
| 23:46.36 | aiksa[LV] | yup |
| 23:47.06 | hardwire | othre than extra machine.. any reason why asterisk isn't on the redphone side forming an IAX2 trunk? |
| 23:47.31 | hardwire | tdmoe is better served on local ethernet without a lot of distruption |
| 23:47.32 | aiksa[LV] | #1 I cant put a machine there. |
| 23:47.46 | aiksa[LV] | hardwire: it serves fine up to 3 PRIs |
| 23:47.51 | ricdanger | anyone using BLF without problem on 1.6 and Grandstream? |
| 23:48.01 | aiksa[LV] | 4 and more are starting to cause trouble |
| 23:48.05 | hardwire | ah |
| 23:48.15 | aiksa[LV] | in theory I should get 1Gbps |
| 23:48.21 | hardwire | that would be nice. |
| 23:48.27 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:48.27 | ricdanger | I got stuck extension in "InUse&Ringing" state |
| 23:48.30 | aiksa[LV] | i kinda doubt i get it |
| 23:48.58 | aiksa[LV] | it is fibre all the way to the other side |
| 23:49.28 | aiksa[LV] | *BUT* ... there are three switches on the way there which dont belong neither to us nor to the telco |
| 23:49.35 | hardwire | ah |
| 23:49.38 | hardwire | I was wondering about that |
| 23:51.10 | aiksa[LV] | and the line provider wouldnt allow us to examine their configuration, to spot any caps/shaves/incompatible qos rules |
| 23:52.29 | aiksa[LV] | while on the other screen, I am doing evaluation of statistical model for estimation of the probability of the defaults of the individuals :)) and its 2 in the morning here. oh the joy :) |
| 23:53.17 | hardwire | sleep now |
| 23:53.51 | aiksa[LV] | not yet |
| 23:54.29 | hardwire | attempts to figure out how to get asterisk to spit out rtp port information on initiated calls |
| 23:55.35 | hardwire | http://www.explodingdog.com/dumbpict51/sleepnownomoretalking.gif |
| 23:57.38 | aiksa[LV] | :) |
| 23:57.54 | aiksa[LV] | my newest fav. http://xkcd.com/664/ |
