| 00:01.57 | tm1000 | vite: you can do that |
| 00:02.44 | vite | will it look like my user @ myip addy? |
| 00:02.55 | tm1000 | no |
| 00:03.13 | tm1000 | you'd have to set it up like a trunk |
| 00:03.28 | tm1000 | thats if you want mapping |
| 00:03.40 | tm1000 | if it was phone -> sip address then it would be user@ addy |
| 00:05.09 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
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| 00:09.14 | vite | tm1000: but I don't have to pay any fee to create a sip server that can call other sip providers? |
| 00:09.27 | tm1000 | no not providers |
| 00:09.35 | vite | sip end users |
| 00:09.36 | tm1000 | if you wanted to call friends with other asterisk servers |
| 00:09.41 | tm1000 | yes |
| 00:09.51 | tm1000 | they'd have to setup a trunk for you to connect to |
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| 00:29.35 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 01:09.56 | snowhite | Hey everyone; I am trying Asterisk 1.8 CURL function and trying to send CLID Name but name consists of "Firstname Lastname" so there is a space |
| 01:10.08 | snowhite | how can I send the space charecter? |
| 01:10.13 | snowhite | or maybe replace it with "-"? |
| 01:10.19 | Chainsaw | Presumably you HTTP-encode it with a %20? |
| 01:11.33 | snowhite | but callerid variable has first name / lastname in one variable already |
| 01:11.37 | Chainsaw | (Which is the hexadecimal encoding for ASCII character 32, a space) |
| 01:12.02 | Chainsaw | So http-encode the variable contents before you send them? |
| 01:12.12 | snowhite | so some string manipulation should happen |
| 01:12.15 | snowhite | is there a function for that? |
| 01:12.28 | Chainsaw | Try core show functions |
| 01:12.32 | Chainsaw | And see if anything stands out to you. |
| 01:12.32 | WIMPy | Or just quote the string and let CURL handle it? |
| 01:12.43 | Chainsaw | Or that, yes. |
| 01:13.15 | snowhite | WIMPy - can you please explain what you mean by quote? |
| 01:13.31 | Chainsaw | "quote variable with spaces and other fiddly things" |
| 01:13.42 | *** join/#asterisk micols (~t@rlogin.dk) |
| 01:13.47 | WIMPy | that |
| 01:13.50 | Chainsaw | Instead of quote%20variable%20with%20spaces%20and%20other%20fiddly%20things |
| 01:14.01 | Chainsaw | As WIMPy's version is easier to write, and to read. |
| 01:14.14 | Chainsaw | (Although the functional result should be identical after decoding) |
| 01:14.49 | snowhite | let me show you my line and you will probably understand my issue better: |
| 01:14.52 | snowhite | exten => s,n,Set(gcm_string=${CURL(http://www,mydomain.com/receive.php?caller=${CALLERID(name))}) |
| 01:15.17 | snowhite | ${CALLERID(name)} output is "Firstname Lastname" |
| 01:15.28 | snowhite | so there is already a space in that variable |
| 01:15.36 | WIMPy | That comma looks bad. |
| 01:15.56 | snowhite | if Firstname was in a separate variable from Lastname then I could use what you are suggesting with %20 |
| 01:16.10 | snowhite | what comma? |
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| 01:16.25 | WIMPy | And fails if someone has three names. |
| 01:16.33 | snowhite | exactly |
| 01:16.35 | WIMPy | The one between www and mydomain. |
| 01:16.45 | snowhite | oh ; that's a typo |
| 01:16.54 | WIMPy | thinks that func_curl should handle that. |
| 01:17.07 | snowhite | doesn't |
| 01:17.12 | snowhite | in 1.8.15 |
| 01:17.18 | snowhite | it only sends the lastname |
| 01:17.21 | WIMPy | But many thinks don't work as I expect them to. |
| 01:17.49 | WIMPy | Oh. |
| 01:17.52 | snowhite | so can you think of another function that allows me to manipulate Callerid(name) string to remove spaces? |
| 01:18.06 | WIMPy | Only the first would have seemed more likely to me. |
| 01:18.28 | snowhite | yeah, I don't know why lastname comes but it's clearly the space issue |
| 01:18.54 | snowhite | I'd rather not go out of dialplan to AGI or php scripts for this |
| 01:20.09 | WIMPy | Use the shell version of CURL or use REPLACE. |
| 01:20.57 | snowhite | SHELL version creates other problems; like returns, timeout, etc...rather stay away from it |
| 01:23.12 | *** join/#asterisk micols (~t@rlogin.dk) |
| 01:25.05 | snowhite | seems like not possible |
| 01:25.55 | WIMPy | Ok, so I guess you have to use CUT. |
| 01:26.10 | WIMPy | would just go and use AGI. |
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| 01:31.01 | [TK]D-Fender | <PROTECTED> |
| 01:31.14 | snowhite | just reading that |
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| 01:40.33 | snowhite | space character is not %20 in asterisk dialplan |
| 01:40.39 | snowhite | what would it be? |
| 01:40.55 | WIMPy | dialplan? |
| 01:40.57 | voxter | what do you mean, space character in dialplan? |
| 01:42.04 | [TK]D-Fender | snowhite: put a literal space |
| 01:42.21 | snowhite | I tried to no avail |
| 01:42.45 | [TK]D-Fender | snowhite: And you should probably show us.... |
| 01:43.05 | snowhite | one sec; trying with " " now |
| 01:43.31 | *** join/#asterisk droemel (~droemel@p4FCAD5DA.dip.t-dialin.net) |
| 01:43.34 | [TK]D-Fender | No quotes |
| 01:43.54 | [TK]D-Fender | The dialplan does not have "data types" |
| 01:44.06 | WIMPy | Dialplan usually doesn;t like quotes very much, but I still don't understand the question. |
| 01:44.24 | [TK]D-Fender | WIMPy: He's screwing up his latest CUT attempt no doubt. |
| 01:44.30 | [TK]D-Fender | awaits the backup |
| 01:44.49 | [TK]D-Fender | already has the list of usual suspects ready.... |
| 01:46.07 | *** join/#asterisk micols (~t@rlogin.dk) |
| 01:48.04 | snowhite | exten => s,n,Set(FirstName=${CUT($CALLERID(name), ,1)}) exten => s,n,Verbose(${FirstName}) |
| 01:48.29 | [TK]D-Fender | $CALLERID(name) <-- no $ |
| 01:48.56 | WIMPy | And no $ without {} anyway. |
| 01:49.04 | [TK]D-Fender | That's not actually a valid format for any use.... |
| 01:50.27 | snowhite | ops missed {} |
| 01:51.40 | snowhite | and now: exten => s,n,Set(FirstName=${CUT(${CALLERID(name)}, ,1)}) |
| 01:51.41 | snowhite | ? |
| 01:52.45 | [TK]D-Fender | NO $ <---------------- |
| 01:52.50 | [TK]D-Fender | AND no {} |
| 01:53.18 | [TK]D-Fender | It askes you to pass a varbale name (or function), not a "value" |
| 01:53.30 | [TK]D-Fender | variable* |
| 01:53.35 | snowhite | right before CUT ? |
| 01:56.13 | [TK]D-Fender | [21:48][TK]D-Fender$CALLERID(name) <-- no $ |
| 01:56.23 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 01:57.26 | snowhite | ${ is needed before CUT |
| 01:57.33 | [TK]D-Fender | yes |
| 01:57.38 | snowhite | or else variable sets to = CUT(................ |
| 01:58.04 | [TK]D-Fender | because you are getting the result of the function as what to set the variable on the left of the "=" as |
| 02:04.14 | [TK]D-Fender | AND THEY'RE GONE |
| 02:04.18 | [TK]D-Fender | </meme> |
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| 02:37.54 | SeRi | wow nyc does not allow gas powered on any kind of gas generating back up generator? |
| 02:38.08 | SeRi | s/on/or/ |
| 02:39.25 | SeRi | CC claims that due to sandy thei DC shut down because tehy do not have back up generators..... NYC City regulation... |
| 02:39.40 | SeRi | s/thei/their/ |
| 02:40.07 | SeRi | That makes no sense |
| 02:43.04 | SeRi | well I hope they all come back online ok. |
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| 02:54.40 | maxhbp204 | Hi, i have installed asterisk 1.8-cert version, i have created 250 iax users, and i have done port forwarding 8080 to 4569 and rules are also ok, but i am not able register iax phone on server and it is also not showing any error, i have also put calltokenoptional=0.0.0.0/0.0.0.0 and requirecalltoken=no in general section as well, can anybody help me how can i enable registeration for iax? please |
| 02:55.25 | WIMPy | iax set debug on |
| 02:55.58 | jpsharp | Why did you forward 8080 to 4569? And did you tell your clients to register via port 8080? |
| 02:56.45 | maxhbp204 | yes |
| 02:56.53 | maxhbp204 | all requests are coming on 8080 |
| 02:57.34 | WIMPy | So what dos Asterisk have to tell when you receive a registration attempt? |
| 02:57.34 | jpsharp | And you did forward UDP port 8080 and not just TCP port 8080? |
| 02:57.56 | maxhbp204 | i have forwarded udp port only |
| 02:58.12 | jpsharp | just checking the "is it plugged in" obvious questions. |
| 02:58.32 | maxhbp204 | 3 REDIRECT udp -- 0.0.0.0/0 0.0.0.0/0 udp dpt:8080 redir ports 4569 |
| 02:58.48 | maxhbp204 | this way i have configured redirection for ports |
| 02:59.34 | jpsharp | Well, now is the time to kick on IAX debug and see what it sees coming in. |
| 03:01.13 | maxhbp204 | is there any limit like on same time one registeration can happen? |
| 03:01.28 | maxhbp204 | because i have seen one register request only on 8080 and then no any other came |
| 03:01.51 | maxhbp204 | and in iax debug also it is having same way, one request logs are there only |
| 03:01.52 | jpsharp | No, not really. It'll process registrations as fast as it can. |
| 03:03.57 | maxhbp204 | ok let me try iax debug and check it |
| 03:05.06 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 03:08.32 | maxhbp204 | one phone is registered now with 4569, although port forwarding is there, and i am getting voice cutting problem on iax channel, what might be the issue for voicecutting |
| 03:09.02 | maxhbp204 | robotting sound and cutting voice occured |
| 03:10.22 | jpsharp | That's usually indicative of network congestion or server lag. |
| 03:11.08 | maxhbp204 | ok that might be network issue possibly |
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| 04:04.14 | dandate2 | did anyone elses asterisk die when callcentric went down today? |
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| 04:07.14 | jpsharp | Not that I'm aware of. |
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| 04:54.09 | rahail | any one have implent asterisk with mvne provider |
| 04:54.17 | rahail | implome |
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| 05:00.36 | maxhbp204 | hi, we are using iax on asterisk 1.8-cert, but we are getting robotic sound on playback of files and also in calls, can anybody help me for fixing this iax issue |
| 05:07.47 | ChannelZ | what codec are the files in? |
| 05:08.23 | ChannelZ | and what codec is being used in the call(s) in question? |
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| 05:16.21 | maxhbp204 | ulaw, ilbc, g729 |
| 05:16.43 | maxhbp204 | does it having any isp kind issues |
| 05:18.36 | ChannelZ | well it could but there's a lot of variables |
| 05:19.27 | maxhbp204 | ok just for checking can you tell me some of them, so i can try it now |
| 05:19.56 | ChannelZ | well get rid of all the codecs and try just ulaw |
| 05:20.13 | ChannelZ | But your prompts might still sound like crap if you've only got gsm installed or something |
| 05:21.30 | *** join/#asterisk oleslav (~astray@178.76.237.142) |
| 05:21.43 | oleslav | hi everybody! |
| 05:22.14 | ChannelZ | PEOPLE HI! |
| 05:22.40 | WIMPy | Free dope? |
| 05:23.25 | ChannelZ | Nope. Name the reference! |
| 05:23.35 | maxhbp204 | ok i will try with that thanks ChannelZ |
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| 07:51.54 | krotos | hi all guy |
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| 08:08.56 | krotos | hi all |
| 08:11.24 | pppingme | ? |
| 08:15.07 | krotos | goood morning |
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| 08:33.12 | krotos | i'm having some trouble with SPA3102, it does not sent the CID (from PSTN to VoIP) , but show me the DID |
| 08:36.48 | *** join/#asterisk chris- (~chris_not@p5DD18561.dip0.t-ipconnect.de) |
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| 08:39.04 | bulkorok | hi |
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| 08:42.26 | chris- | hi |
| 08:43.59 | chris- | is there someone who was working with mysql or postsql on asterisk? |
| 08:45.10 | bulkorok | ~ask |
| 08:45.10 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 08:45.38 | chris- | i have some problems to configure the voicemail |
| 08:45.55 | chris- | i get the following error if i start asterisk |
| 08:46.00 | chris- | [Oct 29 16:13:15] WARNING[16720] config.c: Realtime mapping for 'voicemail' found to engine 'pgsql', but the engine is not available |
| 08:46.01 | chris- | [Oct 29 16:13:15] NOTICE[16720] config.c: Registered Config Engine pgsql |
| 08:46.35 | bulkorok | odbc voicemail must be selected in make menuselect |
| 08:47.10 | bulkorok | https://wiki.asterisk.org/wiki/display/AST/Storing+Voicemail+in+PostgreSQL+via+ODBC |
| 08:47.34 | chris- | thx, i will try it |
| 08:47.45 | bulkorok | I tried with mysql and it worked without problem |
| 08:48.07 | chris- | ok thx |
| 08:48.46 | bulkorok | I'd say the odbc dsn is not configured correctly |
| 08:52.41 | kaldemar | chris-: your warning is specifically because the config engine is not available yet. put "preload => res_config_pgsql.so" in modules.conf. |
| 08:57.25 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 09:00.23 | chris- | i am afk now, thx for helping me |
| 09:01.01 | tonikasch | Qwell, I think perhaps topic has an error.... Asterisk 10.9.0 (2012/10/08) <--- Shouldn't it be Asterisk 1.9.0 ? |
| 09:03.10 | carrar | Why do you think that? |
| 09:05.08 | carrar | Someone should let Digium know this |
| 09:05.15 | carrar | They got the versioning all wrong |
| 09:05.23 | wdoekes | ohnoes! |
| 09:05.44 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 09:05.49 | wdoekes | let's tell the thunderbird team too |
| 09:06.16 | wdoekes | http://www.mozilla.org/en-US/thunderbird/all.html <-- can't be right.. I used version 3 this year |
| 09:06.42 | carrar | People still use thunderbird? |
| 09:06.57 | wdoekes | those that are not using gmail |
| 09:07.01 | *** join/#asterisk Neptu (~Neptu@mail.avtech.aero) |
| 09:07.07 | carrar | People still use google!!>!> |
| 09:07.08 | carrar | ? |
| 09:07.35 | carrar | <-- PINE |
| 09:07.44 | wdoekes | haha |
| 09:07.50 | carrar | never had a virus |
| 09:07.51 | carrar | :) |
| 09:08.00 | carrar | in 20 some years |
| 09:08.17 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 09:08.41 | carrar | Those pine viruses are hard to get :) |
| 09:08.45 | carrar | heh |
| 09:09.47 | Maliuta | Alpine |
| 09:10.20 | carrar | one day I'll upgrade |
| 09:10.24 | carrar | not in a hurry |
| 09:11.05 | carrar | So far nothing in it I need |
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| 09:36.14 | tonikasch | Hi, do you know how to dial an extension if phone is hang off for a certain amount of time? |
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| 09:51.15 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 09:51.30 | ectospasm | tonikasch: you mean like an immediate dial? or if you dial a certain digit string you want the phone to send it onto Asterisk after a timeout? |
| 09:52.10 | tonikasch | ectospasm, the first, to make a call when user has had the phone off for some seconds |
| 09:53.21 | tonikasch | when making a call, if user has not pressed any digit, to make call after some seconds |
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| 09:57.44 | tonikasch | so i guess a call upon some seconds from hookoff |
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| 09:59.14 | kaldemar | tonikasch: "core show application RetryDial" <- is this what you're after? |
| 09:59.23 | tonikasch | i'll see |
| 09:59.55 | tonikasch | no, i guess not |
| 10:00.25 | kaldemar | i'm having a hard time understanding what you want. |
| 10:00.32 | *** part/#asterisk einstein123 (~msa@80.146.245.99) |
| 10:01.40 | tonikasch | i have an fxs-fxo interface attached to my router, so when i hang up the phone, without dialing any number in the phone, i want asterisk to dial the voicemail number if some seconds are passing |
| 10:02.39 | kaldemar | is that a real FXS or an ATA? |
| 10:03.12 | tonikasch | i guess it is an ata, i'me using channel lantiq |
| 10:03.35 | kaldemar | from asterisk's point of view it is then a VoIP device. |
| 10:04.01 | tonikasch | its device name is TAPI/1 |
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| 10:07.52 | tonikasch | oh, so it's up to the "voip" device (the phone) to make it work |
| 10:07.55 | tonikasch | ? |
| 10:08.06 | kaldemar | if it is a VoIP device. |
| 10:08.23 | tonikasch | it is an analog phone :$ |
| 10:09.08 | kaldemar | if asterisk saw it as an FXS, you could configure immediate=yes for it in chan_dahdi.conf and make it enter the dialplan when the phone is put off-hook. then you could do what ever you want. |
| 10:09.35 | kaldemar | just to make sure, how does the device communicate with asterisk? |
| 10:09.52 | tonikasch | through chan_lantiq |
| 10:11.59 | chris- | kaldemar: thx, its working now |
| 10:13.52 | kaldemar | tonikasch: i have no idea about that one. |
| 10:14.26 | tonikasch | yes... that's the point, this channel is under heavy wip from openwrt devs |
| 10:14.37 | tonikasch | i guess i'll have to wait :) |
| 10:14.42 | tonikasch | thanks anyway |
| 10:17.30 | WIMPy | tonikasch: Do you have any pointer where we can find out aobout it? |
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| 10:25.26 | tonikasch | WIMPy, sorry, about what? |
| 10:25.50 | tonikasch | about channel lantiq? |
| 10:27.11 | tonikasch | in https://dev.openwrt.org/browser/packages/net/asterisk-1.8.x/src-lantiq/channels/chan_lantiq.c |
| 10:27.32 | tonikasch | but I don't know if that is what you are searching for |
| 10:27.45 | tonikasch | and for now dev.openwrt.org is down |
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| 10:38.22 | hurdman | hi |
| 10:38.31 | hurdman | have anyone ever had that sort of msg ? |
| 10:38.32 | hurdman | [Oct 29 11:39:22] WARNING[2146] chan_dahdi.c: Can't fix up channel from 148 to 151 because 151 is already in use |
| 10:38.35 | hurdman | [Oct 29 11:39:22] WARNING[2146] chan_dahdi.c: Hangup REQ on bad channel 0/27 on span 5 |
| 10:39.00 | hurdman | i can"'t understand what does it mean, and it's the first time i have got that |
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| 10:49.50 | tonikasch | hurdman, perhaps take a look at http://www.vicidial.org/VICIDIALforum/viewtopic.php?t=19633 |
| 10:50.53 | tonikasch | last post |
| 10:50.59 | tonikasch | not, the pre-last one |
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| 11:18.10 | fredericve | Hi, my asterisk installation crashes (segfault) when I use the dial command on the console. I guess I'm missing some dependency. Anyone got an idea? |
| 11:18.58 | Faustov | fredericve: I suggest strace |
| 11:19.05 | bulkorok | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 11:19.07 | Faustov | it will help you find out why |
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| 11:24.20 | fredericve | hmm any command that involves "console" seems to crash it |
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| 11:31.29 | hurdman | tonikasch: i'll test thanks ! |
| 11:36.46 | tonikasch | hurdman, you are welcome |
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| 14:30.32 | NovceGuru | Nice that Callcentric runs entirely out of one datacenter without a generator |
| 14:30.33 | NovceGuru | haha |
| 14:31.51 | NovceGuru | Ok, so that was possibly dslreports FUD, they do have a generator at LGA11, but flooding has caused an evacuation and fuel supply to disappear |
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| 15:16.16 | fredericve | Does anyone know which channel drivers or modules are required to be able to do testcalls from the console on asterisk 1.8? |
| 15:17.48 | jpsharp | chan_alsa |
| 15:18.26 | wdoekes | res_clioriginate |
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| 15:39.43 | gain | hi all |
| 15:40.53 | gain | I've asterisk Asterisk 1.8 and I'm wondering if I need t38 if it's already in or if I've to install something more |
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| 15:45.39 | [TK]D-Fender | gain, * 1.8 supports T.38 in PASSTHROUGH only. * 10 supports Gateway mode |
| 15:46.19 | *** join/#asterisk moy_ (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 15:46.31 | gain | [TK]D-Fender: I need to pass it to iaxmodem... is 1.8 enough? |
| 15:47.12 | [TK]D-Fender | IAXModem != T.38 That is IAX2. |
| 15:47.24 | [TK]D-Fender | T38modem is another product to look at |
| 15:48.00 | gain | [TK]D-Fender: thanks |
| 15:48.38 | gain | [TK]D-Fender: so 1.8 + T38modem + hylafax could be a solution? |
| 15:48.53 | [TK]D-Fender | yes |
| 15:49.58 | gain | [TK]D-Fender: I've already connected my asterisk 1.8 to a innovaphone ip6000 voip gateway, and I can see it with sip show peers as registered |
| 15:50.22 | gain | but I can't see any call incoming in cli with maximum verbosity... |
| 15:50.30 | [TK]D-Fender | "sip set debug on" |
| 15:50.31 | gain | can be a t38 problem? |
| 15:50.31 | [TK]D-Fender | ^^^ |
| 15:51.56 | *** join/#asterisk mathi (~Matthew@ip-62-235-194-110.dsl.scarlet.be) |
| 15:52.00 | gain | core show codecs donesn't list any t38... can be a problem? |
| 15:52.10 | gain | or I'm searching the wrong name? |
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| 15:52.15 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:52.26 | [TK]D-Fender | You're looking for things that don't matter. |
| 15:52.31 | [TK]D-Fender | "sip set debug on" <------------- |
| 15:52.44 | [TK]D-Fender | T.38 isn't a codec ... it's a protocol. |
| 15:53.00 | mathi | hi |
| 15:53.07 | mathi | how can I set LONG DTMF tones ? |
| 15:59.13 | tm1000 | file: there was something I was going to ask you. For motif. On the inbound context. If I don't want to have the extension be a catch all "s" what can I have it be to match? I tried my email address (user account for google voice) but that didn't match. I also tried phone number but that also didn't match |
| 16:00.40 | gain | [TK]D-Fender: can I manage different protocols? I need t38 and g711 too... is g711 a protocol for fax too? people I'm workin with make me confusing with codecs... :( |
| 16:00.44 | *** join/#asterisk vlad_sta_ (~vlad_star@wn2nat44.beelinegprs.ru) |
| 16:01.57 | [TK]D-Fender | gain, G.711 *is* a codec. |
| 16:02.05 | [TK]D-Fender | gain, You keep mixing apples & oranges |
| 16:05.53 | fredericve | gain: There is a T38 gateway patch available for asterisk 1.8 if you don't want to go to asterisk 10 or 11 |
| 16:06.09 | fredericve | I would recommend asterisk 11 though |
| 16:07.33 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 16:09.10 | gain | [TK]D-Fender: here the debug http://pastebin.com/mn4wpmtf |
| 16:09.23 | [TK]D-Fender | gain, what you haven't done is tell us the details about your inbound call.... |
| 16:10.02 | [TK]D-Fender | gain, there is no call attempt in that pastebin |
| 16:11.25 | gain | uh... so the problem is from the innovaphone? |
| 16:11.43 | gain | that's all that appears when I try to call... |
| 16:11.54 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 16:13.00 | [TK]D-Fender | then you've set it up wrnog |
| 16:13.06 | [TK]D-Fender | wrong* |
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| 16:19.26 | gain | my asterisk has a sip trunk that receive calls from an innovaphone, that result registered |
| 16:19.47 | gain | i see it in sip show peers |
| 16:20.14 | [TK]D-Fender | seeing it there means nothing |
| 16:20.14 | gain | when I make a call the innovaphone forwards it to me but all I can see is the log above |
| 16:20.25 | [TK]D-Fender | Thre is no acll FROM it to your server |
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| 16:20.54 | [TK]D-Fender | That is registration. That is not a call. |
| 16:22.25 | gain | thanks I got it, I just want to know if there's anything I can check on my side (the asterisk side) |
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| 16:35.37 | navaismo | hope people afected by Sandy are fine |
| 16:36.06 | gain | bob from other side (innovaphone side) say that could arrive faxes both from t38 and g711... |
| 16:36.58 | [TK]D-Fender | T.38 is a PROTOCOL, G.711 is a CODEC. Stop mixing the two up. And BOTH are negotiated over SIP. |
| 16:37.09 | jblack | gain: So, about codecs versus protocols. |
| 16:37.29 | [TK]D-Fender | And your Innovaphone ... IS NOT SENDING ANY CALL TO YOUR SERVER. |
| 16:38.16 | *** join/#asterisk WebSprocket (~WebSprock@static-159-253-76-34.karoo.kcom.com) |
| 16:38.20 | WebSprocket | #freepbx |
| 16:38.23 | jblack | gain: actually, you'd be better just googling the difference. I don't want to tell you something almost right. |
| 16:38.53 | *** join/#asterisk Rokfan (~Rokfan@D522448D.static.ziggozakelijk.nl) |
| 16:38.55 | jblack | gain: http://ofps.oreilly.com/titles/9780596517342/asterisk-UnderstandingVoIP.html |
| 16:39.31 | gain | <PROTECTED> |
| 16:40.47 | WebSprocket | Wondering if anyone has had an exposure to a2billing, Im wondering how i turn off the balance and asking to dial number when calling from freepbx |
| 16:41.13 | [TK]D-Fender | ~freepbx |
| 16:41.13 | 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 |
| 16:41.15 | [TK]D-Fender | ^ |
| 16:41.54 | jblack | A protocol like t.38 handles one part of your , while a codec handles a different part of your program. On defines the process of communication, and the other defines the actual data format. That make sense to you? |
| 16:42.29 | jblack | bleh. A protocol like t.38 handles one part of your problem, while a codec handles a different part of your problem. One defines the process of communication, and the other defines the data format. That make sense to you? |
| 16:44.43 | *** part/#asterisk WebSprocket (~WebSprock@static-159-253-76-34.karoo.kcom.com) |
| 16:45.57 | gain | jblack: y |
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| 17:22.48 | danfromuk | Hi. Why doesn't _9NNNNNNN match with 96603416 ? Is there a bug? |
| 17:23.02 | jmetro | do 0's match n's |
| 17:23.04 | Qwell | Because N doesn't match 0 |
| 17:23.09 | Qwell | (or 1) |
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| 17:23.18 | danfromuk | Ah. well spotted! |
| 17:23.19 | jmetro | 9XXXX etc etc |
| 17:23.29 | Rumbles | Hi guys, I'm having some headaches with faxing, basically, 95% of inbound faxes fail and produce an almost empty tiff file (130-390 bits) |
| 17:23.42 | Rumbles | I have tried logging a fax receipt to see if I can find anything more, but the file generated with verbose etc set to 10 was around 23MB (500k+ lines) |
| 17:23.45 | Qwell | Rumbles: welcome to the wonderful world of fax. |
| 17:23.57 | Rumbles | could kill the inventor of fax machines today |
| 17:24.03 | Rumbles | anyone know any good web pages or info for bug fixing fax? |
| 17:24.14 | jmetro | deprecate fax, upgrade to email |
| 17:24.21 | Rumbles | not an option :'( |
| 17:24.29 | jmetro | i know ._. |
| 17:24.33 | Qwell | I replaced Fax with Google Wave. |
| 17:24.51 | *** topic/#asterisk by Qwell -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.9.0 (2012/10/08), 1.8.17.0 (2012/10/08), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.13 (2012/10/09) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 17:24.57 | Rumbles | unfortunately, there are some people who are still living in the 80s |
| 17:24.59 | Qwell | looks at mjordan |
| 17:25.18 | *** topic/#asterisk by Qwell -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.0.0 (2012/10/30), 10.9.0 (2012/10/08), 1.8.17.0 (2012/10/08), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.13 (2012/10/09) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 17:26.43 | Rumbles | if anyone has any tips r.e. fax (I know, long shot) let me know via PM please, I'm going to go home and find a hole to crawl in to |
| 17:27.04 | Qwell | ~fax |
| 17:27.04 | infobot | rumour has it, fax is The honor of designing the first fax *service* in actual use goes to Giovanni Caselli, an Italian abbot, born in Siena in 1815, who turned his hand to science and was, by 1849, editing a scientific magazine. In 1856 he claimed that he had developed a device, which he called a "pantelegraph," that could send facsimiles of images and text. Napoleon III did not come up with the idea, he merely backed it. |
| 17:27.14 | Qwell | ~faxforasterisk |
| 17:27.14 | infobot | i guess faxforasterisk is Digium's commercial Fax For Asterisk module is available at http://www.digium.com/en/products/software/faxforasterisk.php |
| 17:27.17 | Qwell | close enough |
| 17:28.54 | Rumbles | thanks Qwell |
| 17:28.58 | _Corey_ | Well, that bit of history certainly made my day |
| 17:28.58 | Rumbles | made me smile at least |
| 17:29.15 | Rumbles | so it's not an 8-s technology, it's a 1800's technology |
| 17:29.26 | file | Death to Fax! |
| 17:29.41 | hardwire | speaking of which. |
| 17:29.52 | hardwire | Just disabled the fax exten on my ivr trees.. wtf was it doing there |
| 17:30.34 | hardwire | also.. grr.. there was an exten exten => _[a-z].,1,Goto(${EXTEN},1) in there.. and I have no idea why |
| 17:30.48 | hardwire | may have been the day I decided to just have crack for lunch. |
| 17:31.26 | hardwire | all I know is.. a fax was detected. matched it.. looped it. pooped it. |
| 17:33.28 | *** join/#asterisk jblack (~jblack@230.sub-70-192-145.myvzw.com) |
| 17:36.10 | *** join/#asterisk friend (~kvirc@bzq-79-177-230-175.red.bezeqint.net) |
| 17:37.20 | *** join/#asterisk Galen (~Galen@rrcs-24-43-17-235.west.biz.rr.com) |
| 17:38.41 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:39.14 | friend | hi. I'm new on Asterisk |
| 17:39.16 | friend | I'd like to play text from a remote URL using Festival TTS. How can I set it at the dialplan? |
| 17:40.03 | kresp0 | friend: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#ExternalServices_id36056008 |
| 17:40.41 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 17:40.57 | friend | thank you I'm looking |
| 17:41.12 | kresp0 | and http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#IVR_id246940 |
| 17:41.33 | kresp0 | short answer: |
| 17:41.34 | kresp0 | ~book |
| 17:41.34 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:49.17 | *** join/#asterisk Rokfan (~Rokfan@D522448D.static.ziggozakelijk.nl) |
| 17:49.46 | *** join/#asterisk Rokfan (~Rokfan@D522448D.static.ziggozakelijk.nl) |
| 17:49.48 | *** join/#asterisk cklimos (~Claude@209.5.121.227) |
| 17:50.26 | *** join/#asterisk elico (~Thunderbi@109.64.221.76) |
| 17:50.48 | friend | Many thanks kresp0 |
| 17:50.51 | friend | I didn't know all these books are available online as HTML |
| 17:50.57 | friend | very useful |
| 17:51.49 | kresp0 | nice to know :) |
| 17:53.08 | kresp0 | btw, all the links I write before are part of the same book: The Book. |
| 17:55.11 | jmetro | the definitive book - rawr |
| 17:55.39 | [TK]D-Fender | Rumbles, You have not told us one thing about what you're using ... what is the call arriving on? Precisely what are you using to receive it? |
| 17:57.57 | *** join/#asterisk vlad_starkov (~vlad_star@ppp91-79-92-206.pppoe.mtu-net.ru) |
| 18:00.39 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
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| 18:11.19 | *** join/#asterisk Russ (~russ@pool-74-111-207-81.lsanca.fios.verizon.net) |
| 18:11.31 | Russ | asterisk.org, sandy? |
| 18:12.01 | Russ | er, wait |
| 18:12.10 | Russ | nm |
| 18:12.28 | jpsharp | sandy's not here, man. |
| 18:13.00 | *** join/#asterisk triode3 (~number9@76.72.93.193) |
| 18:13.12 | *** join/#asterisk sertaconay (~sertacona@unaffiliated/sertaconay) |
| 18:13.12 | triode3 | hello? |
| 18:13.24 | Qwell | hi |
| 18:14.43 | *** join/#asterisk mzb_ (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 18:30.45 | *** join/#asterisk chris_n (~Chris@184.7.30.18) |
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| 18:48.16 | rrittgarn | Looking for an ITSP, specifically SIP trunking provider, anyone have any recommendations? |
| 18:49.52 | rrittgarn | everybody busy with the hurricane today? |
| 18:50.08 | *** join/#asterisk jdoe (jdoe@falseprophet.ca) |
| 18:50.59 | jdoe | how do I call an ael macro from a feature code? I tried "gosub(macroname,~~s~~,1)" but it doesn't seem to work (feature code triggers, but the sub never gets called) |
| 18:51.42 | _Corey_ | rrittgarn: I'm in Philly and we escaped the majority of the precipitation, but the region has widespread power outages |
| 18:52.13 | kresp0 | hi all, |
| 18:52.14 | kresp0 | doing a dundi lookup, I get 2 diferent results from the same dundi UID when I should receive one. |
| 18:52.14 | kresp0 | this is the response that I get to dundi lookup: http://pastebin.com/neY3aF7V |
| 18:52.14 | kresp0 | and this is what the dundi debug says on the node that answer that query. |
| 18:52.28 | kresp0 | http://pastebin.com/zQa9iNyK |
| 18:53.22 | *** join/#asterisk triode3 (~number9@zeus.allcom-inc.com) |
| 18:53.34 | triode3 | anyone here using digium AEX808 cards on pots? |
| 18:53.42 | kresp0 | i've double-checked dundi.conf and it should respond only with IP 10 (as seen on dundi debug), but for some reason I get 2 diferent responses from that dundi node |
| 18:53.59 | Qwell | ~polls |
| 18:53.59 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 18:54.09 | *** join/#asterisk sertaconay (~sertacona@unaffiliated/sertaconay) |
| 18:54.24 | bchia | jdoe - macro and gosub are 2 different apps, if you post a dialplan snippet / cli output on pastebin we can help you debug |
| 18:54.56 | kresp0 | ok, now i see that they are different dundi UID, they start and end with the same digits.. -.- |
| 18:55.23 | WIMPy | kresp0: Did you build a loop? |
| 18:55.45 | kresp0 | no, someone didnt configure their dundi.conf properly |
| 18:56.09 | kresp0 | and they are using the example IP I put on the config templates |
| 18:57.20 | kresp0 | I mean, the sample numbers, not the IP |
| 18:57.34 | triode3 | I am using an AEX808 on pots. I can dial in, but when I dial out asterisk gives chanunavail. When I hook up a buttset I can dial out no problem. when I hook up the buttset and have asterisk dial, I hear a dialtone, asterisk dials, I get ringing, then asterisk hangs up. Any ideas? |
| 18:58.13 | jdoe | bchia: not in ael they're not. |
| 18:59.33 | jdoe | bchia: in a dialplan generated from ael macros get generated as gosubs. |
| 18:59.46 | bchia | good call, I remember once upon a time gosub was implemented under the hood when you call macro in ael, but I'm not familiar with the current implementation |
| 19:00.55 | jdoe | me neither, I switched to this last night as an experiment... the only problem I've had so far is that my feature code doesn't work anymore. |
| 19:01.09 | jdoe | I could ... just write it in extensions.conf I guess, but that feels wrong. |
| 19:01.56 | bchia | are you preceding the name with an ampersand? https://wiki.asterisk.org/wiki/display/AST/AEL+Macros |
| 19:02.57 | jdoe | no, it's being called from features.conf |
| 19:03.07 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 19:03.11 | jdoe | old behaviour was "flagcall => #9,self/both,Macro,flagcall" |
| 19:03.24 | jdoe | which obviously doesn't work |
| 19:03.26 | jdoe | (anymore) |
| 19:03.30 | jdoe | new attempt is "flagcall => #9,self/both,Gosub(flagcall,~~s~~,1)" |
| 19:03.35 | jdoe | which isn't calling the sub |
| 19:06.56 | Qwell | Why the switch from , to ()? |
| 19:08.19 | jdoe | Qwell: because as I understand it, otherwise I can only pass one argument to the app. |
| 19:08.48 | jdoe | I would assume the moh_class conflicts otherwise, but maybe it doesn't. |
| 19:09.37 | Qwell | shrugs |
| 19:11.12 | *** join/#asterisk acidrye0n (~acidryeon@250.sub-70-192-68.myvzw.com) |
| 19:11.18 | Qwell | Do you have a context called flagcall? |
| 19:13.37 | jdoe | yeah |
| 19:13.38 | jdoe | http://pastebin.com/mwKrS3Md |
| 19:14.01 | Qwell | and the log says? |
| 19:14.21 | jdoe | [2012-10-30 18:46:41] -- Feature Found: flagcall exten: flagcall |
| 19:14.21 | jdoe | that. |
| 19:14.24 | jdoe | nothing else related. |
| 19:15.03 | jdoe | which seems wrong, the exten is ~~s~~, the context is flagcall. |
| 19:16.52 | Qwell | and with debug enabled? |
| 19:17.00 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 19:18.03 | jdoe | as in "core set debug"? That's the only related line. |
| 19:18.23 | Qwell | There will be at least one more line with debug on. |
| 19:18.26 | Qwell | ~asterisk debugging |
| 19:18.33 | Qwell | ~debugging asterisk |
| 19:18.36 | Qwell | infobot: I hate you. |
| 19:18.36 | infobot | You hate you.? |
| 19:19.24 | WIMPy | ~collectdebug |
| 19:19.24 | infobot | collectdebug is, like, a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 19:19.32 | jdoe | sec. |
| 19:21.18 | jdoe | verbosity is already 10, debug is set to 99. I didn't do a logger reload before, but did this time for fun. |
| 19:21.21 | jdoe | there was no extra line. |
| 19:21.38 | jdoe | lemme change logger.conf too I guess. |
| 19:21.42 | Qwell | To what location is debug sent? |
| 19:22.30 | jdoe | before? I was looking at the console. |
| 19:23.12 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 19:24.34 | jdoe | tailing the logfile created by the link above, I do get an extra line. |
| 19:24.35 | jdoe | [2012-10-30 19:23:04] DEBUG[26812] features.c: Feature interpret: chan=SIP/bbbb-00000027, peer=SIP/aaaa-00000028, code=#9, sense=2, features=18, dynamic=#flagcall |
| 19:24.38 | jdoe | [2012-10-30 19:23:04] VERBOSE[26812] features.c: [2012-10-30 19:23:04] -- Feature Found: flagcall exten: flagcall |
| 19:24.43 | jdoe | ... but nothing more about flagcall. |
| 19:25.14 | Qwell | Did you read the giant IMPORTANT NOTE in features.conf? |
| 19:27.58 | *** join/#asterisk bmg505 (~leon@196-209-7-244.dynamic.isadsl.co.za) |
| 19:30.27 | *** join/#asterisk jblack (~jblack@251.sub-70-192-129.myvzw.com) |
| 19:31.42 | rdegges | Sup everyone. |
| 19:32.12 | *** join/#asterisk bchia (~Adium@nat/digium/x-tbhtjaqmidbypeld) |
| 19:35.17 | jdoe | hmm. |
| 19:36.25 | jdoe | wonder why it was working before. And I guess I wonder if it's possible to do this now. |
| 19:44.45 | *** join/#asterisk raub (~raub@ip70-171-42-89.ga.at.cox.net) |
| 19:47.01 | raub | Trying to compile wanpipe in centos with astrisk already installed (and dahdi-linux dahdi-tools libpri). What is the path for dahdi that you need to feed wanpipe's config file? |
| 19:47.10 | raub | It seems to want the source |
| 19:47.40 | jdoe | Qwell: do you know when that might have changed? Poking through old issues (and people's config snippets), calling a macro (a real one, not a sub) appears to have been legit behaviour from feature codes before. |
| 19:48.05 | raub | (or zaptel path) |
| 19:52.49 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:52.49 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:07.24 | *** join/#asterisk MarKsaitis (~MarKsaiti@b019c35b.bb.sky.com) |
| 20:13.45 | *** join/#asterisk natschil (~nathanael@stgt-5f7095de.pool.mediaWays.net) |
| 20:17.14 | natschil | Hello. I'm tying to setup a system that allows people to call each other encryptedly from one computer to another. However, I've run into so many issues on the way, I thought to ask here what the simplest way to achieve this is. Most respectable sip clients support zrtp. Asterisk does not. Asterisk supports encrypted sip. Most sip clients do not, and I haven't gotten those who claim to support the feature to work (yet). Asterisk |
| 20:17.15 | natschil | does encrypted IAX2. No siphone that is open source that I know of does. Clearly, there is a problem here. Ideally, I would like to simply use zrtp, and I cannot understand why asterisk blocks zrtp messages from passing through. Therefore: anybody have any thoughts on this? |
| 20:17.16 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 20:18.48 | drmessano | Yeah, Asterisk supports SRTP and not ZRTP. Either someone needs to add SRTP support to an open source client, or someone needs to contribute code to asterisk for ZRTP |
| 20:18.57 | drmessano | It's really as simple as that |
| 20:19.25 | Qwell | https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial |
| 20:19.45 | drmessano | The SRTP in Asterisk uses libsrtp and shouldn't be burden for anyone to implement. Its really up to them |
| 20:19.47 | mjordan | most SIP clients support SDES-SRTP |
| 20:21.25 | drmessano | Actually, linphone does SRTP.. and it's completely open sores |
| 20:21.28 | drmessano | So there ya go |
| 20:21.44 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 20:23.15 | natschil | drmessano: thanks for the link to linphone, I dunno why searching for srtp in the ubuntu repos didn't come up with that. |
| 20:23.29 | drmessano | I searched for open source sip srtp |
| 20:23.32 | drmessano | On google |
| 20:23.58 | Qwell | natschil: Try searching for DNS. Does that show every app that uses DNS? |
| 20:24.04 | drmessano | You're not going to find srtp in a search of the Ubuntu repos unless they advertise it in the tiny package description |
| 20:24.11 | natschil | drmessano: I tried searching for something similar, most of the hits I got was for software that was no longer being maintained |
| 20:24.31 | *** join/#asterisk engrxyz (~retr@host81-150-217-167.in-addr.btopenworld.com) |
| 20:25.05 | natschil | btw, anybody know why asterisk doesn't support pass through zrtp? |
| 20:25.20 | natschil | Qwell: good point, though srtp is a lot more specific, given that lot of software phones do not support it. |
| 20:25.31 | natschil | drmessano: thanks for the link to linphone btw |
| 20:25.38 | Qwell | natschil: drmessano gave you a very good reason why. |
| 20:25.44 | drmessano | srtp is extremely common |
| 20:26.00 | Qwell | <drmessano> ... or someone needs to contribute code to asterisk for ZRTP |
| 20:26.11 | natschil | Qwell: I believe that code exists, just not in the asterisk trunk |
| 20:26.38 | Qwell | Nobody said it didn't exist. |
| 20:27.53 | natschil | Qwell: I never said anybody said it didn't exists |
| 20:28.11 | drmessano | I dont ever see ZRTP as something Digium themselves would maintain, as its use with Asterisk is of a very limited and narrow scope. |
| 20:28.33 | drmessano | So someone would need to actively maintain it |
| 20:28.34 | natschil | drmessano: the nice thing with zrtp is that is a protocol for end to end encryption, and hence the server cannot see what is passing through |
| 20:29.04 | natschil | drmessano: with end to end encryption, I don't believe the server needs to do much other than pass on the packets. And asterisk is extremely good at passing on packets. |
| 20:29.40 | drmessano | natschil: I am very familiar with ZRTP. I spent a good bit of time experimenting with PRZ's beta client years ago. But ZRTP is not something I see being a good fit in Asterisk |
| 20:29.49 | natschil | drmessano: why? |
| 20:30.27 | drmessano | Because it doesnt have any sort of mass appeal. It's a very niche feature that is likely to become very out of date as it passes from maintainer to maintainer |
| 20:31.13 | natschil | drmessano: If zrtp has no potential for widespread appeal, why do so many sip software clients support it? |
| 20:31.28 | Qwell | Because so many sip software clients support it. |
| 20:31.30 | drmessano | .... |
| 20:33.29 | drmessano | So if someone spent the time implementing ZRTP in Asterisk 12, you would see thousands and thousands implementing it? Thousands and thousands of ASTERISK ADMINS would be out there implementing ZRTP for their end users? |
| 20:34.10 | natschil | drmessano: I cannot speak for anybody but myself, I would. But there's no point in having an argument. Thanks a lot for the pointer to linphone. |
| 20:34.43 | drmessano | FreeSWITCH supports ZRTP. Its more of a toy than Asterisk. Maybe there should be a pointer to FS for ZRTP support |
| 20:35.06 | natschil | drmessano: maybe. I'll have a look at it. |
| 20:39.31 | danfromuk | Hi, we had a power failure in at our datacentre. There are two asterisk boxes for redundancy, however that didnt help because the power failure killed both. If I want to make asterisk redundant by installing a third in a separate datacentre, how can I failover calls to the third IP? Is srv dns any good for this purpose? |
| 20:41.42 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 20:41.53 | jmetro | would you really need three over two? considering one failure killed both? |
| 20:42.14 | drmessano | SRV DNS would work fine |
| 20:42.49 | natschil | drmessano: freeSWITCH and openbsd don't work well together. shit, just as I thought I had a solution. |
| 20:43.05 | drmessano | natschil: http://zfone.com/prod_asterisk.html |
| 20:44.32 | drmessano | It looks as though there's a licensing agreement that would need to be in place for someone to even implement it |
| 20:45.43 | drmessano | Plus, the guy on the home page at http://zfone.com/ is whispering something into that cute girls ear, and its probably about his genitals. I dont support that sort of behavior |
| 20:48.31 | _Corey_ | haha |
| 20:48.40 | danfromuk | jmetro: 2 in the current datacentre which allows for maintenance and general onsite failures. a 3rd would be in a separate datacentre for major outages. |
| 20:48.55 | _Corey_ | In think the VON Magazine shot also dates things a bit |
| 20:49.14 | drmessano | _Corey_: I was going to say that too, but I didnt want to show my age around the 17 yr olds |
| 20:49.26 | _Corey_ | lol |
| 20:49.50 | natschil | drmessano: I saw the page, don't want to recompile asterisk. don't see how the picture is relevant. |
| 20:50.16 | drmessano | Oh, youre one of those then |
| 20:50.41 | drmessano | try ./configure --with-sense-of-humor |
| 20:51.01 | _Corey_ | drmessano: I am very much amused... |
| 20:51.48 | drmessano | natschil Then I think if you want full ZRTP support in the next couple of years you should try FreeSWITCH. The page there even points out that the patch they provide for Asterisk (1.4.23 GASP) only partially supports the feature set |
| 20:52.11 | drmessano | or you could pay someone out the wazoo to code it. Im sure someone would |
| 20:53.18 | drmessano | I had a look at the patchfile and it's 10MB.. and only supports PART of the feature set? I think someone is going to be writing a lot of code. |
| 20:53.38 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 20:54.53 | natschil | drmessano: If I wanted it that bad I would code it myself. I would use FreeSWITCH if I could, but quite frankly I am too frustrated with this whole thing even attempt to compile the thing on freebsd. I have no idea why the patch is 10MB. However, I feel that the fact that asterisk doesn't even support pass-through zrtp (which should be trivial to implement, though I could be wrong) shows that it doesn't really care about privacy |
| 20:54.54 | natschil | and security. Of course, asterisk may not be about those things, so that's okay with me, just a bit annoying. |
| 20:55.22 | natschil | but before I start a flamewar I'm going. |
| 20:55.23 | *** part/#asterisk natschil (~nathanael@stgt-5f7095de.pool.mediaWays.net) |
| 20:55.33 | drmessano | Nice troll and a part |
| 20:55.54 | fprior | Hi all; I'd like test WebRTC in Asterisk 11, following Wiki at http://code.google.com/p/sipml5/wiki/Asterisk I receive this error: http://pastebin.com/8GfbDLU9 ; any suggestion ? |
| 20:56.00 | *** join/#asterisk enki (jbabiak@kahuna.routed.com) |
| 20:56.05 | drmessano | I guess Asterisk doesnt care about security at all if it doesnt support ZRTP. I almost couldnt type that sentence I was laughing so hard. |
| 20:56.11 | enki | hi |
| 20:57.04 | enki | Does Joshua Colp (Digium) hang out in here? |
| 20:57.44 | newtonr | -> file |
| 20:58.48 | file | enki, hi |
| 20:58.48 | _Corey_ | drmessano: Any response would have been pointless... obviously supporting mainline security standards isn't enough to prove your security bona fides... you need to implement exotic shit |
| 20:59.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:59.27 | file | fprior, can you ensure that res_http_websocket.so loaded before chan_sip.so and tell me what distro version this is on? |
| 20:59.49 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 21:00.26 | drmessano | _Corey_: Obviously McDonalds doesn't care about staying in business because they dont offer meals with Iguana |
| 21:00.36 | _Corey_ | exactly |
| 21:00.38 | _Corey_ | lol |
| 21:01.11 | newtonr | mmmm McIguana |
| 21:01.57 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 21:02.18 | file | enki, please speak in here |
| 21:03.57 | pabelanger | SPEAK |
| 21:05.52 | fprior | @file , reloaded both manually in that order ; Linux debian000 2.6.32-5-686 #1 SMP Sun May 6 04:01:19 UTC 2012 i686 GNU/Linux; |
| 21:06.22 | file | fprior, by reload did you mean "unload" and then "load" of each? |
| 21:06.30 | enki | file: I noticed you were working on my bug report (20216) and were running into an issue replicating it, so thought I would see if I could help. |
| 21:06.51 | fprior | @file , yes |
| 21:08.05 | file | enki, I don't know of anything you could really help me with besides confirming the firewall question... it really does seem both from my code reading, testing, and feeling that it's environmental |
| 21:08.54 | enki | would it help if I let you on the box to see the problem in realtime where it's manifesting itself on? |
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| 21:09.28 | file | enki, yes |
| 21:09.55 | enki | there is no hardware firewall in front of the box, as it's connected directly to the ISP demarc. There is an iptables based firewall running on the box itself, but I don't believe it's causing the issue. I emailed a developer who maintains it to see what he thought. |
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| 21:10.56 | file | the only way Asterisk wouldn't be getting the media is 1. If it deadlocked or 2. Something is preventing it from being read in |
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| 21:11.10 | file | well, the only way it wouldn't show up if rtp set debug on was done |
| 21:12.06 | file | fprior, there seems to be some sort of issue with some systems where the code that makes the websocket support optional is not getting overridden by the real thing, so it doesn't work as expected |
| 21:12.56 | wdoekes | enki: do you have a timing source loaded? |
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| 21:21.12 | fprior | @file , any suggestion ? test with other distro ? try another Howto ? |
| 21:21.26 | file | fprior, a Howto won't fix the bug - I personally use Ubuntu |
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| 21:22.39 | file | if you could please create an issue that would be great |
| 21:23.02 | file | specify the exact version of Debian and the output of "gcc -v" |
| 21:23.12 | fprior | @file desktop or server ? have you test succesfully webrtc ? |
| 21:23.21 | file | fprior, server and yes |
| 21:23.41 | file | I can call from my Windows laptop to Asterisk in Chrome Canary using WebRTC and sipml5, with no patches to Asterisk |
| 21:26.22 | fprior | @file, great. Ok, I'll try on Ubuntu server. were you at AstriCon ? several came back wanting to try WebRTC |
| 21:26.51 | file | fprior, I was - I gave a talk about it with Tim Panton and also spoke about it at the Asterisk 11 thingy |
| 21:26.54 | mjordan | fprior: he's the guy in the white hat. |
| 21:27.11 | mjordan | aka: the bowler. |
| 21:28.07 | mjordan | http://www.youtube.com/watch?v=x2-KNK4zJGg |
| 21:28.28 | file | wait what youtube |
| 21:28.37 | mjordan | bwahaha |
| 21:28.38 | file | oh, it's me! |
| 21:28.45 | file | I don't want to watch that. |
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| 21:28.54 | mjordan | its on the internets now |
| 21:29.35 | fprior | ah yes, I understand now. I'm the guy slightly taller than Matt |
| 21:30.31 | mjordan | given my relative height, you've narrowed yourself to about 25% of the attendees :-) |
| 21:31.07 | file | I describe myself as "awesome" |
| 21:31.56 | fprior | only one of them spoke about chan_dongle |
| 21:33.37 | fprior | @mjordan I think seeing you among attendees |
| 21:33.59 | mjordan | fprior: ah ha! You would be much taller than me :-) |
| 21:34.19 | mjordan | nice talk btw - did you catch David's talk about libwat? |
| 21:38.51 | fprior | @mjordan unfortunately not, but I marked it; I'll see it online. content was something similar ? |
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| 21:55.50 | jeffspeff | just a friendly reminder that tomorrow Codeweavers will be giving away their Cross-Over for linux and mac for 24hrs only |
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| 21:57.00 | jeffspeff | http://flock.codeweavers.com/ and http://www.codeweavers.com/products/#cxlinux |
| 21:57.04 | drmessano | Whats funny is that when I follow the YouTube link, there's a sponsored video for "Hearty Beef Stew" at the top of the list |
| 21:57.49 | slav3_kitten | anyone here have a set of working configs for cisco 7940/7960 phones with sip images? |
| 21:58.08 | mjordan | fprior: somewhat, yes. They developed a generic library for interfacing with GSM cards |
| 21:58.10 | jeffspeff | slav3_kitten, i think those configs are on voip-info |
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| 21:58.43 | slav3_kitten | hmm i must have missed them. now i feel like a moron lol |
| 21:59.07 | slav3_kitten | thanks though jeffspeff |
| 21:59.12 | jeffspeff | lol, they're generic for the 7900 series i believe |
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| 21:59.42 | jmetro | hm.. any tips on installing app_konference for 1.8 |
| 21:59.48 | slav3_kitten | you'd think that... i still can't manage to get the 7911 i have working for the love of me. i have gotten the 7960 to kind of work |
| 22:00.06 | jeffspeff | slav3_kitten, http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx |
| 22:00.41 | jeffspeff | slav3_kitten, cisco has been really pushing for their Unified Communications "experience" |
| 22:00.53 | jeffspeff | they don't release working config files either |
| 22:01.45 | slav3_kitten | yea that's been my main problem which led to me getting frustrated enough to give up. but i already have the cisco phones, and inline power switch so if it's going to happen it's going to happen with those phones. |
| 22:01.59 | jeffspeff | a while back i was in the same boat with cisco phones, but actually had the money to pay for support. They kept transferring me between the UC TAC team and the Phone TAC team. both teams telling me that they didn't support what i was doing; which is fine and dandy, i just wanted more config files. |
| 22:02.34 | jeffspeff | i ended up emailing one of their supervisors the config files i was working with, and she called me back asking how i had obtained those files, because cisco doesn't release them to the public |
| 22:03.03 | slav3_kitten | nice... |
| 22:03.24 | jeffspeff | oh, and they pretended that those files didn't exist (until i provided them my copy) and then they pretended that nobody could provide the information i was asking for |
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| 22:03.36 | slav3_kitten | i like cisco products, but their support is a nightmare especially if you don't work for a large company |
| 22:03.41 | *** part/#asterisk natschil (~nathanael@stgt-5f7095de.pool.mediaWays.net) |
| 22:04.12 | fprior | @mjordan so, was an honor to have you among the participants |
| 22:04.33 | jeffspeff | slav3_kitten, that's just it. i do work for a large company, i'm my cisco id is registered as a channel partner and reseller and i was still treated like that |
| 22:04.47 | slav3_kitten | wow |
| 22:05.19 | jeffspeff | i user their ASA devices for my network, but for phones i go with polycom. documented config files, and you don't have to pay subscriptions or anything to download newer firmware |
| 22:08.31 | slav3_kitten | yea if didn't have phones and i had a 802.3af poe switch i'd likely have gone polycom |
| 22:08.56 | drmessano | The cisco phones are great hardware. It's getting them to actually work and supported well without using a Cisco Call Manager that's the issue |
| 22:09.07 | slav3_kitten | but i've got 4 7940's 1 7960, a 7925(iirc wifi phone), and a 7911 or 7912 can't remember |
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| 22:10.03 | drmessano | We are a big Cisco partner, and I could have gotten Cisco phones for 50% of what I pay for Polycom... but not worth the hassle |
| 22:10.06 | slav3_kitten | and it's a home network so i don't really have the capital to do things right, i think i'm going to run asterisk form a netbook or old laptop in the network closet |
| 22:10.17 | NovceGuru | haha voip.ms lowering their porting fee to $10, wonder if it has anything to do with callcentric |
| 22:10.22 | jmetro | oh polycoms.. |
| 22:10.25 | drmessano | or Flowroute |
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| 22:10.43 | NovceGuru | Did they crash out too with the nyc power issues? |
| 22:10.48 | NovceGuru | I don't have any clients with them |
| 22:10.52 | drmessano | Who? |
| 22:10.56 | NovceGuru | flowroute |
| 22:10.59 | drmessano | Nope |
| 22:10.59 | NovceGuru | or did they lower their fee too? |
| 22:11.13 | drmessano | Flowroute lowered the fee a week or so ago |
| 22:11.18 | drmessano | I suspect everyone else has |
| 22:11.20 | NovceGuru | I can't find anybody besides callcentric that can/will port out a 740-694-xxx number |
| 22:11.36 | drmessano | Callcentric sucks balls |
| 22:13.16 | drmessano | I heard that PBX In A Flash boxes were dying left and right during reboots and reload because they have iptables rules with Callcentric IPs in them, and iptables wouldn't load with them down |
| 22:13.29 | drmessano | That was pretty hilarious |
| 22:21.30 | slav3_kitten | speaking of nyc. i've not heard from a number of friends that lived there |
| 22:28.22 | jmetro | our sip provider is in nyc..now looking for a sip provider in the midwest suffice it to say. |
| 22:29.32 | doulos1 | exit |
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| 22:33.06 | [TK]D-Fender | heads out for the evening |
| 22:33.38 | natschil | Does anybody know why I'm getting == Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) every time my client (sflphone) tries to connect to asterisk over encrypted sip? |
| 22:33.44 | slav3_kitten | jmetro, i'm sure they will have it fixed in a bit |
| 22:34.13 | natschil | I've looked at the relevant source code, and it seems that the SSL_accept function of openssl isn't making the connection |
| 22:34.47 | jmetro | slav3_kiten well we didnt even know they were in the east coast until last night. and their backup system on the west coast is awful. |
| 22:35.17 | slav3_kitten | ohhhh |
| 22:35.35 | natschil | looks like it might be a bug in the way asterisk reports ssl bugs. |
| 22:35.44 | natschil | as that error string isn't very informative. |
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| 22:48.58 | NovceGuru | jmetro: who are they? |
| 22:49.02 | NovceGuru | drmessano: that is quite hilarious |
| 22:49.03 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 22:49.18 | NovceGuru | apparently callcentric is not even in a datacenter? |
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| 23:10.52 | paulc | A friend of mine ported from their local cable co to Callcentric, then apparently they got hit with a denial of service attack and it didn't look too good |
| 23:11.01 | paulc | "Isn't yours down as well?" she asks |
| 23:11.27 | paulc | "uh, no - I don't use them" I reply... (because I didn't need 911 and she did) |
| 23:11.51 | paulc | this was before it was mandated.. my current carrier now provides it so we're all good |
| 23:13.11 | NovceGuru | nie |
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| 23:34.09 | jeffspeff | anybody else using ver 11 ? |
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| 23:41.55 | tonikasch | nope, here using 1.8.10.1 |
| 23:42.10 | WIMPy | "else"? Are you using it? |
| 23:42.21 | jkates | using 11.0 does anybody know how to debug motif? |
| 23:43.05 | jkates | Ie I have not motif commands like I have for other channels like sip |
| 23:43.10 | WIMPy | Oh, so jeffspeff is not the only one. |
| 23:43.42 | jeffspeff | lol |
| 23:43.50 | jeffspeff | i was beginning to feel lonely |
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| 23:44.30 | jeffspeff | WIMPy, you toyed with 11 yet? |
| 23:45.21 | jkates | oops no motif commands |
| 23:45.35 | jkates | should I have any motif commands? |
| 23:45.53 | WIMPy | jeffspeff: I've been on TRUNk for several months. |
| 23:45.58 | file | I don't think I wrote any |
| 23:46.01 | jeffspeff | i migrated server hardware on saturday and decided to go ahead and upgrade from 10.7 to 11 while i was a it... whenever you start * or restart * it seems to take a bit for everything to load before it finally says * is ready |
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| 23:46.33 | jeffspeff | didn't have this "experience" on previous versions |
| 23:47.32 | file | jkates, that was to you |
| 23:47.42 | WIMPy | I find that it takes ling to stop. |
| 23:48.00 | jeffspeff | running 16gb ram, dual socket quad core xeon processors; 15k rpm scsci drives raid 5 |
| 23:48.18 | jeffspeff | stopping doesn't seem to take as long as starting |
| 23:48.30 | file | jeffspeff, there's nothing explicit that would do that... you'd have to look at the log and which modules are taking awhile |
| 23:48.48 | jeffspeff | file, good idea. i'll check that out |
| 23:49.05 | jeffspeff | thanks |
| 23:49.15 | jeffspeff | once it's running, all is good |
| 23:49.16 | file | like right now some ITSPs are down, so if you are using any that could contribute |
| 23:50.03 | jeffspeff | mine are hosted in dallas |
| 23:50.47 | jeffspeff | not a big issue, just one of those things you notice different when you do an upgrade |
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| 23:54.30 | jeffspeff | what kind of market do you think there is in the US for a multi-tenant * gui (windows app or web based)? like for service providers or something? |
| 23:55.06 | jeffspeff | and by gui i mean full fledged system that handles installation of server, configuration and end-user features |
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