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| 00:26.05 | aitiba | hi |
| 00:27.34 | aitiba | I have this extensions configuration http://pastebin.com/jhqdv3Lk but I can connect using linphone as sip:100@192.168.1.128 |
| 00:27.43 | aitiba | i get error with the password |
| 00:27.51 | aitiba | ¿someone can help me please? |
| 00:29.21 | [TK]D-Fender | aitiba: "sip set debug on" , "core set verbose 10" <- show us the failed call attempt from * CLI |
| 00:29.23 | [TK]D-Fender | ~pb |
| 00:29.24 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 00:29.31 | [TK]D-Fender | ^^^ PASTEBIN the attempts |
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| 00:31.58 | aitiba | [TK]D-Fender: I make a asterisk -rvvvv and then those two command. I try to connect usinf linphone but I dont get anything on the console |
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| 00:37.43 | [TK]D-Fender | aitiba: where is your * relative to your linphone? |
| 00:38.06 | aitiba | my what? |
| 00:38.17 | aitiba | i dont know what is * |
| 00:38.31 | [TK]D-Fender | ASTERISK |
| 00:41.54 | bchamberlain | aitiba what is the IP or your asterisk server? |
| 00:42.15 | aitiba | is a local one |
| 00:42.27 | bchamberlain | ifconfig will tell you the ip |
| 00:43.20 | bchamberlain | aitiba remove the host=line from the telefono1 peer. |
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| 01:30.11 | LeLion | hello everyone |
| 01:35.54 | LeLion | I am experiencing a problem where calls are slow to connect over an E1 PRI connected to a Cisco voice router resulting in a very high pdd. When the E1 is connected directly to the asterisk box using a te121 card, the pdd is much more acceptable. Has anyone here had the same experience? And, if so, were you able to correct it? |
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| 01:58.44 | aitiba | is there ast_tls_cert on asterisk installed from debian repositories? |
| 02:16.45 | phix | no idea |
| 02:16.51 | phix | have a look |
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| 03:48.55 | aitiba | could I gave ssl active on the asterisk installed from debian repositories? |
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| 05:53.00 | MangaKaDenza | oh, thats what this is... |
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| 07:47.07 | gerritfromsa | Haven't done irc in ages. Tried it again today after reading the latest LFX... |
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| 12:27.43 | Natureshadow | Hi! |
| 12:27.58 | Natureshadow | I am running (well, I would like to have it running...) Asterisk 11.5.1 on Debian sid |
| 12:28.30 | Natureshadow | My config and dialplan used to work before upgrading to 11.5, but now Asterisk segfaults on any incoming call, due to an issue obviously related to res_config_ldap |
| 12:28.36 | Natureshadow | http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=725925 |
| 12:29.19 | Natureshadow | What strikes me as odd is that outgoing calls work flawlessly, and reducing the dialplan to s => 1,Hangup() does not fix it either, so it is not related to my very complex dialplan |
| 12:29.41 | Natureshadow | I'd like help fixing this, because my asterisk has been offline for some weeks now already... |
| 12:32.24 | Natureshadow | Is it possible that I have to alloww Asterisk to update the LDAP entry with hte peer address upon registration? |
| 12:32.49 | Natureshadow | But, I do not even need to register or something to trigger this issue, as I said, it is completely unrelated to the dialplan |
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| 16:20.41 | Preatorian | Hey all |
| 16:22.56 | phix | hai Preatorian |
| 16:23.00 | phix | How are you buddy? |
| 16:23.29 | Preatorian | Tired, spended waaaaayyy to many hours working and so little on my home project (asterisk) :P |
| 16:24.14 | Preatorian | Yourself? |
| 16:28.55 | Preatorian | afk - Foods! |
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| 18:12.27 | Preatorian | Question, does anyone here have experience with a Digium TDM410P with 2x FxO channels not wanting to hangup after the calling party hung up (takes 8 seconds). |
| 18:13.55 | Preatorian | Caller(A) dials a number that is connected to the channel A/B ... rings fine ... Hangs up after few seconds, but the DAHDI channel does not seem to hang up straight away, but after 8 seconds. |
| 18:14.49 | Preatorian | busydetect is enabled, i've checked the timers as well 500,500 but none seem to make a difference. |
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| 18:20.31 | [TK]D-Fender | Preatorian: call your telco and make sure they enable CDS |
| 18:20.34 | [TK]D-Fender | ~cds |
| 18:20.34 | infobot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 18:22.49 | Preatorian | Is it a commen problem then on analog lines? |
| 18:22.52 | file | although even then it may take a few seconds |
| 18:24.55 | Preatorian | I'll call them first, my guess is that they dont even know what it is -.- |
| 18:29.00 | ghost75 | can i enable both t.38 and cng in * ? |
| 18:30.10 | file | Asterisk does not support CNG. |
| 18:30.35 | ghost75 | why it has then cng option in faxdetect |
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| 18:31.44 | file | oh, you mean CNG tone |
| 18:31.56 | ghost75 | fax on g.711 |
| 18:32.08 | file | then yes |
| 18:32.25 | ghost75 | i get t.38 refused message |
| 18:32.35 | ghost75 | or do i need to forward ports like 6004 '? |
| 18:33.00 | file | you need to enable T.38 and the remote side has to support T.38 |
| 18:33.05 | file | if it does not, then the T.38 negotiation will fail |
| 18:33.43 | ghost75 | just wonder if i need to forward any port on router for t.38 |
| 18:33.59 | file | UDPTL is carried over the ports configured in udptl.conf |
| 18:34.20 | file | that wouldn't cause a T.38 refused message though |
| 18:34.37 | ghost75 | let me try |
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| 18:51.43 | Preatorian | lol @ carrier, they directed me to 4 departments, in the end the engineer said "never heard of that feature in all 20 years of telephony" mmmm interesting (Disconnect Supervision) |
| 18:52.48 | Preatorian | I do wonder though, why does a regular old brick of a phone works fine, but pc + digium card with asterisk does not :P |
| 18:53.38 | [TK]D-Fender | Preatorian: pastebin your dahdi configs |
| 18:53.41 | [TK]D-Fender | ~pb |
| 18:53.42 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
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| 18:54.10 | Preatorian | i know about paste bin, which configs would you like to see or all? |
| 18:58.14 | [TK]D-Fender | your dahdi config, just like I asked. |
| 19:01.23 | Preatorian | Sended links in PM |
| 19:03.22 | [TK]D-Fender | Preatorian: seems ok, give "callprogress=yes" a try |
| 19:05.13 | Preatorian | Something i do wonder, since i read at "ringtimeout will not update on a reload." Does this apply to multiple Dahdi parameters. and ifso, what is the propar way to reload dahdi |
| 19:05.38 | Preatorian | Is that just with a "dahdi restart' or should i do a service dahdi restart to reload the modules too? |
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| 19:08.45 | apb1963 | Greetings all... I've been broken. |
| 19:08.56 | apb1963 | or rather my computer has |
| 19:09.12 | [TK]D-Fender | Preatorian: you need to reload the whole module, not just the config |
| 19:09.25 | Preatorian | service dahdi restart? |
| 19:10.17 | [TK]D-Fender | yes, including stopping * |
| 19:10.57 | Preatorian | Ok, stopped asterisk + dahdi and restarted dahdi + asterisk after (in that order) - checking :) |
| 19:12.09 | Preatorian | No, still the same, it seems to stop everytime exactly after 8 seconds though. |
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| 19:45.33 | Preatorian | [TK]D-Fender, if i capture the stream with Dahdi_monitor i can see a very clear and abrupt signal stopping and a sort of static noise returning. this same signal is being interpreted by the old brick phone as " Oh he hang up" and instantly stops the ringing proces. The static noise seems to be polarity related since no signal is being send nor received on the line itself. |
| 19:49.25 | [TK]D-Fender | CDS is usually either a cust or a polarity reversal.. not sure if there is an option to select which is used with DAHDI.. |
| 19:49.43 | Preatorian | i did some checking and capturing |
| 19:50.12 | Preatorian | With or without the options answeronpolarityswitch + hanguponpolarityswitch |
| 19:50.37 | Preatorian | No effect, even thought something is very clearly changing in the captures, and it aint audio or datatrigge |
| 19:50.48 | Preatorian | +r |
| 19:56.58 | Preatorian | How is this normally managed then? do i need different software for this particular part? |
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| 20:04.39 | [TK]D-Fender | Preatorian: Don't think it's necessary anything you're doive so much as DAHDI not having a good profile for your zone. |
| 20:05.40 | [TK]D-Fender | Preatorian: Sice you are in a position to do real debug, I'd recommend going to #asterisk-dev and bringing this to them tomorrow between 8-5 EST for the best offs of a dev getting their hands on this directly. Might get you a solid solution in the end |
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| 20:17.03 | gartral | ok, I feel stupid.. I can not for the life of me get incoming calls to go to the dial-by-name directory.. the line I feel in question is same => n,Dial(SIP/6501,20) where 6501 is my directory, cli complains that 6501 is not a peer and then says everyone is busy/congested.. |
| 20:17.07 | lvlinux | anybody ever get "Ending Skinny session from 172.17.0.123 (bad input)" when using a Cisco skinny phone w *??? |
| 20:18.16 | gartral | see http://paste.ubuntu.com/6314462/ for exact errors |
| 20:18.25 | lvlinux | gartral: never used directory but, whenever you do SIP/xxxx, then xxxx is supposed to be a peer or device |
| 20:19.02 | lvlinux | is 6501 the extension of your directory? |
| 20:19.27 | lvlinux | if so, then you can probably use Goto() to jump to it instead of Dial() |
| 20:20.24 | gartral | lvlinux: yes, ok I'll try that! |
| 20:22.06 | lvlinux | you may have to use the context and priority in the Goto() app, not sure. If so, you'd do "same => n,Goto(yourcontext,6501,1)" |
| 20:22.36 | gartral | lvlinux: strange, I get [Oct 27 16:21:07] WARNING[5439][C-000000b8]: pbx.c:6390 __ast_pbx_run: Channel 'Motif/+[scrubbed]' sent to invalid extension but no invalid handler: context,exten,priority=incoming-motif,SIP/6501,20 |
| 20:23.03 | [TK]D-Fender | gartral: SIP/6501 is NOT an extension |
| 20:23.08 | [TK]D-Fender | gartral: pastebin your actual dialplan |
| 20:23.50 | [TK]D-Fender | gartral: And I can't see where you get the idea of 20 being a proper priority number |
| 20:24.00 | lvlinux | gartral: have no clue about the Motif error stuff---are you using Motif for GVoice or something? |
| 20:24.18 | [TK]D-Fender | gartral: Sstop trying to treat a Goto as you would a Dial. They is no relationship between them |
| 20:24.27 | [TK]D-Fender | lvlinux: Don't worry about motif. |
| 20:24.32 | [TK]D-Fender | lvlinux: lets see the actual dialplan |
| 20:24.45 | gartral | [TK]D-Fender: http://paste.ubuntu.com/6314493/ |
| 20:24.46 | [TK]D-Fender | lvlinux: Because he's clearly missing a good chunk of dialplan basics |
| 20:24.51 | lvlinux | [TK]D-Fender: I think he meant 20 was the dialing timeout since he was thinking he could use Dial() |
| 20:25.06 | [TK]D-Fender | gartral: Where do we see anything concerning a Directory in there? |
| 20:25.08 | gartral | lvlinux: correct, I forgot to clear that |
| 20:25.20 | gartral | ? |
| 20:25.26 | gartral | stupid mouse... |
| 20:25.50 | [TK]D-Fender | gartral: you said you wanted to get to a "Directory" ... well that means calling the Directory() dialplan application. You aren't doing that anywhere in what you've shown us |
| 20:27.05 | lvlinux | I assume that he has the 6501 extension set up to call the Directory() app? but yes we do need to see the dialplan to be sure. |
| 20:27.20 | gartral | http://paste.ubuntu.com/6314510/ |
| 20:28.05 | gartral | lvlinux: indeed you are correct, but if that's a bad way of doing things then I have no issue changing it to the right way |
| 20:28.25 | [TK]D-Fender | gartral: same => n,Goto(users,6501,1) |
| 20:28.38 | [TK]D-Fender | gartral: Or just put Directory() RIGHT THERE and save the Goto |
| 20:28.59 | gartral | [TK]D-Fender: which is perfered? |
| 20:29.00 | lvlinux | yes that would be the simple thing to do |
| 20:29.25 | lvlinux | Goto's are kindof a necessary evil---you don't want to use them without a reason. |
| 20:29.34 | [TK]D-Fender | gartral: You aren't really saving yourself any trouble by using a GOTO. Ther is no extra bonus logic there to be saving effort in jumping to so you might as well jsut call Directory directly |
| 20:29.46 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.200) |
| 20:30.28 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 20:30.41 | gartral | [TK]D-Fender: sounds like I'll have less retardedness in the long-run with a Directory().. do I need to worry about and params/functions to pass to the application? |
| 20:30.55 | lvlinux | gartral: if you had to call the directory tons of times from different places in the dialplan, then it might make sense to use Goto, it depends. Usually you use Goto to jump around in IVRs. |
| 20:31.15 | [TK]D-Fender | gartral: Of course you do. that tells it WHAT directory to use, etc. That's what options are... you should read the apps instructions to understand what they are for. |
| 20:31.38 | gartral | lvlinux: I don't plan on using IVRs on this deployment |
| 20:34.06 | lvlinux | k |
| 20:34.32 | Preatorian | [TK]D-Fender: i'll do that thank you :) |
| 20:38.26 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 20:38.50 | danfromuk | Hi, is there any reason why ChanIsAvail wouldnt work? It always sets AVAILSTATUS to 0 |
| 20:39.03 | danfromuk | http://pastebin.com/uCTwYLig |
| 20:40.10 | [TK]D-Fender | danFirst, chanisavail is an APPLICATION, not a FUNCTION |
| 20:40.30 | [TK]D-Fender | danfromuk: You should definitely be reading its instructions.... |
| 20:40.52 | danfromuk | Sorry, Old code. 1sec |
| 20:41.34 | danfromuk | http://pastebin.com/ypdMZWyn |
| 20:42.58 | [TK]D-Fender | Preatorian: danNow go read the apps instructions. |
| 20:44.48 | danfromuk | [TK]D-Fender: if that comment is for me, I have and can't figure out what i'm doing wrong. |
| 20:45.03 | [TK]D-Fender | danWhat do the instructions say? |
| 20:45.28 | danfromuk | You want me to repeat the entire instructions? |
| 20:45.54 | [TK]D-Fender | danfromuk: how about the OPTIONS. |
| 20:46.02 | [TK]D-Fender | danWhat do THEY say about it? |
| 20:46.29 | danfromuk | Check all available channels, not just the first. Dont think that applies to my issue. |
| 20:46.44 | [TK]D-Fender | pastebin them... |
| 20:47.11 | danfromuk | http://pastebin.com/V93Rxg1k |
| 20:47.28 | danfromuk | None seem to indicate why AVAILSTATUS is always 0 |
| 20:47.49 | danfromuk | Could the SIP device state not be tracked? |
| 20:48.13 | [TK]D-Fender | sanNo, you semed to only look at teh FIRST option as if that was all that is important |
| 20:48.23 | [TK]D-Fender | "s" <---- |
| 20:48.28 | danfromuk | call-limit is set to 2 so it should be tracked |
| 20:49.00 | gartral | ok, so calling in now goes to directory, but the directory insists that none of my users exist.. |
| 20:49.07 | [TK]D-Fender | just because your device is being used for a call does NOT mean that it is incapable of accepting ANOTHER |
| 20:49.17 | [TK]D-Fender | gartral: Show us the directory you created. |
| 20:49.49 | danfromuk | [TK]D-Fender: I can accept another. However, I just want to know if its current on a call so i can play a different ringtone to the 2nd caller |
| 20:49.54 | danfromuk | It can* |
| 20:50.03 | danfromuk | currently* |
| 20:50.35 | [TK]D-Fender | [16:48][TK]D-Fender"s" <---- |
| 20:51.09 | gartral | [TK]D-Fender: http://paste.ubuntu.com/6314643/ |
| 20:51.23 | danfromuk | Sorry, didnt know what was for me. However, even when the peer is idle, it returned 0 |
| 20:51.34 | [TK]D-Fender | gartral: that is sip.conf. that has nothing to do with Directory() |
| 20:52.00 | [TK]D-Fender | danfromuk: PASTEBIN..... |
| 20:52.07 | [TK]D-Fender | danfromuk: do be complete.... |
| 20:52.37 | gartral | [TK]D-Fender: well I don't have a directory.conf so.. which file would that be in? |
| 20:54.09 | [TK]D-Fender | gartral: "core show application directory" <- before just using an app you should understand what it actual does, and requires |
| 20:54.41 | danfromuk | ok 's' seems to now return 1 for 'not in use' |
| 20:54.41 | gartral | [TK]D-Fender: the really odd thing is, if I call 6501 from a client unit, it works, on call-in, it does not |
| 20:54.47 | danfromuk | Wonder why it returns 0 without the 's' |
| 20:55.07 | danfromuk | 0 means "Unknown"; channel is valid, but unknown state. |
| 20:56.14 | [TK]D-Fender | gartral: you haven't shown us your updated configs or the failed call attempt.... |
| 21:05.03 | gartral | [TK]D-Fender: the only change in configs was adding exten=>n,Directory(family,users,ef) under [incoming-motif]... also looks like my voicemail.conf got a reversion somehow.. that explains the errors |
| 21:06.25 | *** join/#asterisk rbd_ (~rbd@cpe-076-182-043-018.nc.res.rr.com) |
| 21:06.28 | WIMPy | That's not a valid line for extensions.conf. |
| 21:06.42 | WIMPy | You should try to get in to the basics first. Try the |
| 21:06.47 | WIMPy | ~book |
| 21:06.47 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:07.04 | rbd_ | hey guys... is it possible to handle the autodialout "failed" extension from within an AGI/FastAGI script, or does that have to be handled in extensions.conf in the launched context? |
| 21:07.18 | [TK]D-Fender | well.. it COULD be valid, but I'm sure it isn't what he needs/wants :) |
| 21:07.47 | WIMPy | How could that be valid? |
| 21:08.01 | WIMPy | It's clearly missing a parameter. |
| 21:08.40 | [TK]D-Fender | WIMPy: ...oops :) |
| 21:09.01 | [TK]D-Fender | gartral: And indeed that line is not quite like it is supposed to be.... |
| 21:09.40 | gartral | [TK]D-Fender: what? |
| 21:10.02 | [TK]D-Fender | [17:04]gartral[TK]D-Fender: the only change in configs was adding exten=>n,Directory(family,users,ef) |
| 21:10.06 | *** join/#asterisk Natureshadow (~nik@shore.naturalnet.de) |
| 21:10.09 | [TK]D-Fender | gartral: that is NOT valid syntax |
| 21:10.38 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 21:10.57 | gartral | [TK]D-Fender: then why does exten => 6501,1,Directory(family,users,ef) work under [users] and same => n,Directory(family,users,ef) not work under [incoming-motif]? |
| 21:11.11 | [TK]D-Fender | [17:04]gartral[TK]D-Fender: the only change in configs was adding exten=>n,Directory(family,users,ef) <--- |
| 21:11.16 | [TK]D-Fender | gartral: taht does NOT say "same" |
| 21:11.26 | [TK]D-Fender | gartral: that says "exten" |
| 21:12.10 | Preatorian | :P |
| 21:12.13 | Penguin | extension n is a failure. |
| 21:12.44 | [TK]D-Fender | Penguin: Not necessarily :) |
| 21:13.03 | [TK]D-Fender | Penguin: If he's typing on an alphanumeric keyboard it can work.... |
| 21:13.07 | Penguin | Extension n. Priority D? or priority Directory? |
| 21:13.11 | gartral | [TK]D-Fender: http://paste.ubuntu.com/6314760/ |
| 21:13.16 | [TK]D-Fender | Penguin: But yeah ... that's just not what's really going on here |
| 21:13.49 | [TK]D-Fender | gartral: Good to know it supposedly made into the actual file in a sane manner |
| 21:14.12 | gartral | [TK]D-Fender: I've been working on his for quite a few hours, I apologize for my own confusion |
| 21:14.50 | gartral | this* |
| 21:22.13 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.74) |
| 21:22.19 | gartral | slams head into desk |
| 21:27.20 | *** join/#asterisk s7r (~s7r@openvpn/user/s7r) |
| 21:27.26 | s7r | anyone here? |
| 21:28.10 | [TK]D-Fender | looks around |
| 21:28.15 | [TK]D-Fender | ummm.. no? |
| 21:28.42 | s7r | so i downloaded an application on my blackberry which connects to a sip server (in order to use sip over internet and call) not via mobile carrier |
| 21:29.01 | s7r | i connect to port 5060, normally. but how can i know if the connection is secured by tls? |
| 21:29.08 | s7r | from the phone to my sip server i mean |
| 21:29.55 | [TK]D-Fender | s7r: It'd be clear in the documentation of the app and you'd see options specifically for this including certificates, etc |
| 21:30.21 | s7r | i don't see such information |
| 21:30.52 | s7r | i can only enter my server and port + username and password |
| 21:30.59 | s7r | and i have another option to select the codec |
| 21:31.03 | s7r | but nothing about SSL/TLS |
| 21:31.18 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 21:31.31 | lvlinux | then it doesn't support TLS---it would have a place to input certificate info if it did. |
| 21:32.03 | s7r | that sucks. so if i'm on wireless i am vulnerable |
| 21:32.07 | s7r | to interception |
| 21:32.32 | s7r | but what can be intercepted? the whole voice conversation? what about the username and password are those sent in plain text too? |
| 21:32.41 | *** join/#asterisk ghost75 (~quassel@dslb-092-075-058-180.pools.arcor-ip.net) |
| 21:32.56 | danfromuk | Hi. I'm trying to play a different tone when a caller dials an extension which is busy. But it doesnt seem to make a difference. Any idea what i'm doing wrong? |
| 21:32.57 | danfromuk | http://pastebin.com/ZdzWsygQ |
| 21:34.03 | [TK]D-Fender | danfromuk: Your expression syntax is wrong |
| 21:34.28 | [TK]D-Fender | danfromuk: -- Executing [202@kesher_phones:21] ExecIf("SIP/kesher_201-000013f3", "[2=2]?Playtones(callwaiting)") in new stack |
| 21:35.04 | danfromuk | Oops. Sorry, youre right as always |
| 21:35.12 | [TK]D-Fender | danfromuk: Would also be ncie to know if you actually learned anything from previous help give instead of just leaving. |
| 21:35.21 | [TK]D-Fender | given* |
| 21:36.12 | danfromuk | [TK]D-Fender: You don't usually summaries the help. I wondered why 's' causes the application to work |
| 21:36.20 | s7r | :( |
| 21:36.22 | lvlinux | s7r: your username is sent clear text but usually the password is hashed. |
| 21:36.45 | s7r | i thought so |
| 21:36.46 | [TK]D-Fender | [16:49][TK]D-Fenderjust because your device is being used for a call does NOT mean that it is incapable of accepting ANOTHER |
| 21:36.50 | ghost75 | is it a known issue that after application receivefax the dialplan will not continue? |
| 21:36.54 | [TK]D-Fender | danfromuk: that was pretty explicit |
| 21:36.57 | s7r | so .. bottom line... what am I doing is safe or not? |
| 21:37.25 | [TK]D-Fender | danfromuk: ChanIsAvail does not read like "IsItInjustAnyCallWhatsoever" |
| 21:37.36 | lvlinux | s7r: define "safe" lol---are you talking about confidential stuff?? Who do you need secrecy from? "safe" is relative... |
| 21:37.38 | [TK]D-Fender | s7r: Clearly interceptable |
| 21:38.03 | [TK]D-Fender | s7r: Voice is unencryped, and you could get rainbow hashed given enough time |
| 21:38.27 | lvlinux | s7r: and it doesn't take long these days, with GPU acceleration, etc. |
| 21:39.35 | s7r | thank you |
| 21:39.38 | danfromuk | [TK]D-Fender: Your statement didnt make sense. Chanisavail always returns 0. If what you says is correct, it should always return 'available' and not 'unknown' |
| 21:39.42 | s7r | for help |
| 21:39.45 | s7r | what can i do to get tls |
| 21:39.53 | lvlinux | s7r: when I use unencrypted SIP/RTP, I assume that certain persons/governments could gain access---normally it's not your general hackers, but persons at the ISPs, etc. |
| 21:40.20 | [TK]D-Fender | danfromuk: if there is no limit then it is unknown. * doesn't actually ASK the device if it can accept another call |
| 21:40.28 | [TK]D-Fender | danfromuk: that is another incorrect assumption. |
| 21:40.48 | [TK]D-Fender | danfromuk: it checks "is it there", and "is it maxed on a DEFINED limit". |
| 21:40.50 | lvlinux | s7r: use another application if you need security |
| 21:41.11 | s7r | it's the only one in f*cking blackberry app world |
| 21:41.25 | [TK]D-Fender | s7r: VPN it <- |
| 21:41.29 | lvlinux | s7R: what platform are you using? you said BB but which one? |
| 21:41.37 | s7r | Z10 |
| 21:42.11 | lvlinux | ok so BB10--- yes VPN will work, (that's what I do) |
| 21:42.15 | s7r | application called tera phone. it has 2 variants (lite - free and pro - which costs). i have the pro one, paid |
| 21:42.32 | s7r | i couldn't find another sip app to work with bb 10 os |
| 21:42.37 | lvlinux | what about Taki? |
| 21:43.10 | [TK]D-Fender | s7r: VPN <- |
| 21:43.27 | danfromuk | [TK]D-Fender: I already said, call-limit is set |
| 21:43.39 | s7r | lvlinux i didn't know about taki |
| 21:43.40 | *** join/#asterisk cusco (~tralala@2001:41d0:1:6caf::cafe) |
| 21:43.43 | cusco | hey ya |
| 21:43.47 | s7r | it didn't return to search results |
| 21:43.48 | cusco | recommended softphone in linux? |
| 21:43.53 | s7r | i will get a vpn |
| 21:44.12 | gartral | well it's working now, had to rebuild my voicemail.conf file |
| 21:44.17 | lvlinux | yes taki is very good---i use it on BB Playbook. Don't think it supports TLS though. |
| 21:44.43 | cusco | I don't need tls atm |
| 21:44.45 | lvlinux | VPN is the easy way though and in my experience works well. |
| 21:44.53 | cusco | I used to use twinkle |
| 21:45.02 | lvlinux | cusco: linphone, sflphone |
| 21:45.47 | lvlinux | cusco: i used to use twinkle but now I prefer linphone mostly |
| 21:45.58 | cusco | I was about to apt-get install it |
| 21:46.00 | lvlinux | cusco: there is also pjsip if you like text mode :-) |
| 21:46.03 | cusco | just seemed heavy |
| 21:46.08 | cusco | no, I would like a gui |
| 21:46.10 | gartral | cusco: My prefered softphone is jitsi |
| 21:46.39 | lvlinux | jitsi is good too, but does other stuff besides SIP and so sometimes it's confusing IMHO |
| 21:49.11 | *** join/#asterisk zamba (marius@flage.org) |
| 21:50.30 | danfromuk | Is it possible to use playtones to change the ringtone without answering the channel? |
| 21:50.38 | danfromuk | This is a SIP to SIP call. |
| 21:51.43 | [TK]D-Fender | danfromuk: clearly not |
| 21:52.04 | [TK]D-Fender | danfromuk: if you aren't answering the the ENDPOINT chooses what it actually does with the call progress |
| 21:53.03 | danfromuk | I assumed so. Just thought i'd check |
| 21:53.52 | WIMPy | Well, at that ppint there is the thing calles early media, that even SIP knows about. |
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| 22:04.20 | danfromuk | WIMPy: that way my thought when i asked the question |
| 22:05.01 | danfromuk | Does anything I set using 'playtones' get overridden when i use the Dial cmd? |
| 22:05.21 | danfromuk | I cant see an option in Dial that allows me to force a ringtone |
| 22:05.27 | danfromuk | custom ring ton |
| 22:05.29 | danfromuk | tone |
| 22:05.53 | [TK]D-Fender | Not "custom", just whatever the indication zone says |
| 22:05.57 | [TK]D-Fender | and as pure audio |
| 22:06.14 | danfromuk | Is there anyway to change it on a call by call basis? |
| 22:07.04 | danfromuk | I suppose I could use m(class) to use a recording of a different tone |
| 22:21.32 | WIMPy | That should be the safe bet. |
| 22:30.01 | *** join/#asterisk kresp0 (~kresp0@gateway/tor-sasl/kresp0) |
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| 23:05.39 | danfromuk | Works perfectly. Thanks |
| 23:38.07 | zendel | hi, Is there a way I could monitor the Jitter/Latency stats history. I know "sip show channels" & "Iax2 show channels" give me me some info for current channels. But tha's for current(present) traffic only. |
| 23:42.24 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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