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| 00:39.02 | snadge | will an iax2 reload hang up calls? |
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| 01:00.37 | JerJer | snadge: no |
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| 01:35.31 | lvlinux | ~book |
| 01:35.32 | 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 |
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| 04:08.11 | fling | What do I need for enabling opus support? |
| 04:08.55 | fling | and which dongle to buy for gsm bridge? |
| 04:09.55 | fling | https://github.com/meetecho/asterisk-opus ? |
| 04:15.52 | fling | how to check for a supported codec list? |
| 04:16.17 | fling | discovered 'core show codecs' |
| 04:19.47 | fling | http://dpaste.com/2WA7CAJ |
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| 04:33.26 | Penguin | If you like that, you might also enjoy 'core show translation recalc 10' |
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| 04:52.05 | fling | Penguin: thanks :> |
| 04:52.08 | fling | pcm_route.c:947:(find_matching_chmap) Found no matching channel map http://dpaste.com/3BCBEC4 |
| 04:52.11 | fling | How to fix this? otoh I hear ringing and my own dtmf |
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| 09:42.55 | fling | what suff/prefix scheme is better to use for specifying protocol in dids? eg is skype#echo123 look good? |
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| 10:02.19 | dym | Hey everyone. From time to time i will have an odd problem. Within a phone call (SIP) i will stop hearing the other party, while for the other party the call continues normal. while that happens the network output of said asterisk instance spikes up and re-normalizes after a bit. |
| 10:02.24 | dym | What could cause this? |
| 10:02.30 | fling | Don't I need to allow sip guest for the default context to work? |
| 10:02.46 | fling | dym: nat? |
| 10:03.09 | dym | fling: one party is behind nat, yeah |
| 10:03.19 | dym | fling: http://drop.openroot.de/XiXU/Har6Sczc happened just a minute ago |
| 10:03.40 | dym | OH |
| 10:03.43 | dym | wrong graph! |
| 10:03.45 | dym | http://drop.openroot.de/LuG3/iM2ePTPO |
| 10:03.48 | dym | okay, this is odd |
| 10:05.53 | dym | i only have a few calls on there, per day. |
| 10:06.04 | dym | ~20/30 and at max 3 concurrent |
| 10:07.20 | dym | fling: would the RTP stream just simply drop out every now and then cause of NAT? |
| 10:07.34 | fling | idk much about that |
| 10:08.48 | dym | but you thought you might just suggest it? :D |
| 10:14.41 | fling | dym: I had the same issue because of nat |
| 10:14.50 | dym | right |
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| 12:43.47 | ovoshlool | Hello/ Have a troube with bridging a channel created by me through ARI. Channel creates successfull butwhen try to add it to bridge - STATUS: 400 channel not found |
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| 12:44.45 | ovoshlool | this is what i see at json response |
| 12:44.47 | ovoshlool | http://pastebin.com/jHaLsZab |
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| 12:53.27 | file | you can't act on a channel currently until it has been answered |
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| 12:54.43 | file | (for originated channels) |
| 12:56.29 | ovoshlool | Channel is answering. But how I cat identificate answer with ARI? |
| 12:56.44 | file | your events don't show it answering. |
| 12:57.04 | file | after you attempted to add it to a bridge it showed the channel ringing |
| 12:57.20 | ovoshlool | I know. But It is. |
| 12:57.36 | file | Asterisk doesn't think so |
| 12:57.46 | file | unless it does you won't be able to do anything with it |
| 12:58.54 | ovoshlool | hm// You right/ When channel answers ARI recieves event about this? |
| 12:59.06 | file | yes. |
| 12:59.56 | ovoshlool | еÑÑÑÐ»Ñ |
| 12:59.59 | ovoshlool | Thanks |
| 13:00.18 | ovoshlool | very helpfull |
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| 15:09.04 | Ast001 | Hello, I am trying to achieve to catch extension typed by user on his phone. I tried to do that with Background application and WaitExten(30) after it. It should wait 30 seconds. Everything goes fine if user enter extension without few seconds of pause between digits, but if he type few digits and wait few secs Asterisk is sending him into invalid extension before 30 seconds pass. Is there a way I can set how many seconds user can wait before typing next num |
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| 15:11.55 | Ast001 | to be more clear I don't want user to go into invalid extension before 30 seconds pass. |
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| 15:14.07 | jameswf | soooooo.... held calls in 12 go in to bridgewait? If I understand that correctly are all calls in a single resource so if I add a announcer it talks to all? |
| 15:14.58 | [TK]D-Fender | Ast001: "core show function TIMEOUT" <- |
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| 15:15.59 | [TK]D-Fender | Ast001: And what you have asked for would require some ugly hackery to make possible as described. |
| 15:16.06 | anonymouz666 | SRTP is a pain |
| 15:18.14 | Ast001 | ok thanks [TK]D-Fender I will try that. |
| 15:23.18 | file | jameswf, no - if you have two channels talking to eachother the act of one putting the other on hold does not make it go into bridgewait or a holding bridge |
| 15:27.28 | Ast001 | <PROTECTED> |
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| 15:31.36 | [TK]D-Fender | Ast001: Still doesn't give 30 seconds "total", but closer I guess |
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| 15:31.43 | anonymouz666 | so iphone 6 is out there |
| 15:32.06 | mjordan | file: the task of updating the bridges page on the wiki is now more apropos |
| 15:32.49 | mjordan | jameswf: A holding bridge can generally be accessed either through the dialplan or ARI: https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_BridgeWait |
| 15:33.20 | mjordan | jameswf: there are other things that may use the mixing technology of a holding bridge, but that's an implementation detail internally. You don't get to announce to things unless those applications support it :-) |
| 15:38.04 | jameswf | thinks we should be able to announce to all the things.... |
| 15:39.35 | jameswf | Actually one of my little BS projects is to "YO" a random channel whenever someone sends a YO to a specified user.... it really is one of the coolest stupid API's ever |
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| 16:09.07 | adsc | i am in the asterisk gui cli console window, and "core show help" tells me that there is a command "logger set level {DEBUG|NOTICE}", but if I put "logger set level DEBUG" I get an unknown command error...anyone know why? |
| 16:10.35 | adsc | my goal is to see real time debug output in this window for calls I make, because I think my dial plan has errors |
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| 16:11.50 | mjordan | jameswf: So... when you say "Yo", you want to actually play a sound file to them? |
| 16:12.27 | mjordan | and by "send a Yo"... how are you detecting the user wants to "Yo" someone? |
| 16:12.39 | jameswf | independent of my hold question... yes I thought I would use whisper to play a You sound |
| 16:13.23 | mjordan | jameswf: do you like and/or know Python? |
| 16:13.43 | jameswf | the YO api uses webhooks. I have a server that collects these calls so I can poll it. When I poll the server I then act on the "yo's" |
| 16:13.52 | jameswf | <3 python |
| 16:15.30 | adsc | disregard my question, I just launched the cli using asterisk -rvvv |
| 16:16.04 | adsc | however, nothing happens when I place an outgoing call...shouldn't it print something in the CLI? The device is registered and can receive calls... |
| 16:16.06 | mjordan | jameswf: so here is a python app I wrote (and just hacked apart, as there was some stuff that had to get ripped out of there) that does something similar with ARI |
| 16:16.09 | jameswf | my second project idea was to keep a list of YO users.. when a user sends the PBX a yo it originates a call to the user (likely a cell phone) with a disa channel |
| 16:16.10 | mjordan | http://pastebin.com/EqUVe83R |
| 16:16.21 | mjordan | It is a 'prisoner's dilemma' conference room |
| 16:17.00 | mjordan | if anyone hits a '0', then a random channel is chosen and gets howler monkeys played to them |
| 16:17.26 | mjordan | similar concept |
| 16:17.30 | file | (there's a few cool things in there) |
| 16:17.42 | mjordan | I ripped out the IRC bot :-) |
| 16:17.51 | mjordan | And I'm pretty sure that device state shouldn't be in there |
| 16:17.53 | mjordan | BUT WHATEVER |
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| 16:30.10 | jameswf | https://gist.github.com/jfinstrom/97c4d151c259d9d26d28 is what I use as I poke around the ARI... |
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| 16:37.29 | mjordan | jameswf: ooo, going low level |
| 16:37.41 | mjordan | I personally think ari-py is a bit easier, but that's just me :-) |
| 16:37.50 | mjordan | https://github.com/asterisk/ari-py |
| 16:38.04 | mjordan | it is still a thin wrapper, so it doesn't do too much for you, but it keeps you from having to form all of the HTTP requests yourself |
| 16:38.10 | mjordan | and it does a bit of nice event callback handling |
| 16:38.59 | mjordan | i.e., channel.play(media='sound:tt-monkeys') instead of a POST /channels/{id}/play?media=sound:tt-monkeys |
| 16:41.16 | drmessano | jameswf, i'm following your YO experiments. Love the whole concept |
| 16:42.53 | mjordan | It'd be fun |
| 16:43.18 | mjordan | if they're on a call, you could /snoop them; if they aren't, you could originate a call to them with an auto-answer header :-P |
| 16:43.34 | mjordan | if they're on jabber, you could jabber them a message ... |
| 16:43.36 | mjordan | IT GOES ON |
| 16:43.42 | file | ARI! ARI! ARI! |
| 16:43.52 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
| 16:43.59 | mjordan | cough hackathon cough |
| 16:44.12 | file | mjordan, hackathon? what hackathon? |
| 16:44.33 | mjordan | file: why, this one! http://astriconhackathon.challengepost.com/ |
| 16:44.40 | file | #shamelessplug |
| 16:44.54 | mjordan | lost his shame somewhere after Asterisk 10 |
| 16:47.28 | *** join/#asterisk LiohAu (~LiohAu@lse83-h04-89-86-22-242.dsl.sta.abo.bbox.fr) |
| 16:50.24 | jameswf | isn't a real developer, just plays one on the internet |
| 16:53.43 | drmessano | I'm not a developer, I just cobble stuff together and sometimes make neat stuff |
| 16:53.47 | mjordan | are you writing code? Congratulations! You're developing :-) |
| 16:54.17 | mjordan | is pretty sure he misplaced his "Certificate of Real Developer" awhile back |
| 16:57.16 | drmessano | I'm more of the guy that invented the Flowbee |
| 16:57.30 | drmessano | Vacuum + Razor = infomerical |
| 16:57.44 | lnb | [TK]D-Fender: who else did you say besides scopserv? |
| 16:58.47 | [TK]D-Fender | lnb: Go search on your own. |
| 16:59.53 | mjordan | drmessano: I'll take a dozen |
| 17:00.07 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 17:01.14 | lnb | if scopserv would either reply to email, answer phone or return voice mail, it might be a way to go |
| 17:01.42 | drmessano | I remember when I developed my Asterisk 8-Ball, which was a faithful magic 8-ball developed in dialplan that tweeted, facebooked, and returned the results with TTS.. I had job offers, companies with neat TTS APIs wanting to offer my unpublished API access if I would share code with them. etc.. I was all like "I am not a developed.. I'm just good with GLUE" |
| 17:01.57 | drmessano | developer* |
| 17:02.03 | drmessano | I can't type either, apparently |
| 17:02.41 | drmessano | Just goes to show you though, if you build something and it's neat, regardless of how it's put together, people will flock to it |
| 17:03.50 | mjordan | I think you'll like ARI. |
| 17:03.57 | jameswf | drmessano: I think you just described a certain project that shall not be named |
| 17:04.03 | mjordan | Hidden in that pastebin is a geo-lookup that dumps out the location of the caller as they join the conference. |
| 17:04.11 | mjordan | Just sayin. Things being in other languages makes it easy to mash up APIs. |
| 17:05.18 | file | Two words: Visual Basic. |
| 17:05.24 | drmessano | ....... |
| 17:05.30 | mjordan | funny story. |
| 17:05.36 | mjordan | I once wrote a message switch in VB. |
| 17:05.48 | file | mjordan, get back to work YOU |
| 17:05.50 | mjordan | Granted, it was VB.NET, so at least it was OO. |
| 17:05.56 | mjordan | VB 6. Thar be the devil. |
| 17:06.02 | drmessano | $ file /usr/lib/vbrun300.dll |
| 17:06.12 | drmessano | <PROTECTED> |
| 17:06.25 | drmessano | :( |
| 17:06.49 | jameswf | I can't do anything on my linux box there doesn't seem to be a C: drive |
| 17:07.10 | drmessano | I THINK MY C DRIVE FAILED I CANT FIND IT |
| 17:07.23 | drmessano | Whatever you do, DONT REBOOT.. It wont come back up!! |
| 17:07.28 | drmessano | Ever? Ever! |
| 17:09.18 | lnb | jameswf: there is no c: drive in linux |
| 17:09.56 | *** join/#asterisk timahvo1 (~rogue@197.237.134.227) |
| 17:10.12 | mjordan | sarcasm is hard to convey in text |
| 17:10.34 | drmessano | Extremely |
| 17:10.41 | *** join/#asterisk k5673 (~jcapurro@190.52.189.6) |
| 17:11.01 | drmessano | There's no A: drive on my desktop |
| 17:11.06 | lnb | the only linux with c: is dos 6.22 |
| 17:11.20 | drmessano | FAIL |
| 17:11.21 | lnb | enhanced version of course |
| 17:11.48 | drmessano | DOS was a beautiful sexy beast of an OS.. calling it a Linux is sacreligious |
| 17:18.13 | [TK]D-Fender | misses CP/M |
| 17:21.55 | *** join/#asterisk spicyramen_ (~Adium@c-98-210-160-181.hsd1.ca.comcast.net) |
| 17:25.07 | *** join/#asterisk monsterco (46333431@gateway/web/freenode/ip.70.51.52.49) |
| 17:25.57 | monsterco | Is there anyway to lock Aastra phones? By means of provisioning or some other way which doesn't allow a user with physical access to the phone to reset it or change it's settings |
| 17:26.23 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:37.52 | monsterco | Is there anyway to lock Aastra phones? By means of provisioning or some other way which doesn't allow a user with physical access to the phone to reset it or change it's settings |
| 17:39.55 | [TK]D-Fender | monsterco: Have you downloaded their admin guide? |
| 17:43.42 | *** join/#asterisk monsterco (4c476c49@gateway/web/freenode/ip.76.71.108.73) |
| 17:44.09 | *** join/#asterisk darkdrgn2k (~darkdrgn2@69-165-131-20.dsl.teksavvy.com) |
| 17:44.21 | monsterco | yeah, I downloaded Aastra Admin Guide - I am afraid if after all work in provisioning a user might simply be able to use some combination keys on the phone set and reset everything - hence I am asking here to see if anyone has experience with this |
| 17:44.41 | darkdrgn2k | ok, FAX G711 tranmissions.... im supposed to hear communications on BOTH sides of the call... right? |
| 17:45.02 | monsterco | Does Aastra 6753i (for example) have the ability to lock for good with provisioning? or someone with physical can bypass provisioning? |
| 17:45.34 | darkdrgn2k | cause one way i hear the fax "BEEP" then a very short fax like sound and then silence |
| 17:45.40 | darkdrgn2k | while the other side is trying hard to connect |
| 17:47.51 | lnb | [TK]D-Fender: finally got hold of owner of scopserv gave me pricing etc. |
| 17:51.32 | monsterco | anyone on provisioning of aastra phones? |
| 17:52.57 | [TK]D-Fender | monsterco: Is what they say in their guide unclear? |
| 17:53.21 | monsterco | they don't mention this at all |
| 17:53.27 | monsterco | the guide is only about provisioning |
| 17:53.40 | [TK]D-Fender | monsterco: Most manuals leave out things that don't exist |
| 17:54.03 | monsterco | this would be strange because it exists on Cisco and Polycom phones |
| 17:54.03 | [TK]D-Fender | monsterco: Does it say you can LOCK lock changes in there? |
| 17:54.13 | [TK]D-Fender | local* |
| 17:54.17 | monsterco | it's just not there at all |
| 17:54.23 | Penguin | My speculation is that you can set a password to prevent changes from the phone menu. |
| 17:55.18 | monsterco | that is there but that is not what I need - I want to lock the phone totally including firmware resett |
| 17:55.57 | Penguin | A password isn't good enough? You want to install the phones inside Fort Knox as well? |
| 17:57.30 | [TK]D-Fender | factory reset seems to list having to enter the admin PW.... |
| 17:57.36 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.hsi01.unitymediagroup.de) |
| 17:57.57 | monsterco | A password that can be reset by holding down 123 and # is useless |
| 17:58.32 | [TK]D-Fender | Where do you see this? |
| 17:58.41 | monsterco | that is no brainier and already part of the portal (you can change the default 22222 password to anything you want) - but any user can hold down keys and reset the whole firmware |
| 17:58.47 | monsterco | that's what I want to prevent |
| 17:59.08 | [TK]D-Fender | You know that would brick a phone if a PW was ever lost.... |
| 17:59.12 | monsterco | here: http://www.noahlh.com/blog/2012/03/how-to-hard-reset-an-aastra-6757i57i/ |
| 17:59.39 | monsterco | I am fine with that - I can have a provisioning server with backup and redundancy like I do for other phones |
| 17:59.40 | *** part/#asterisk k5673 (~jcapurro@190.52.189.6) |
| 18:01.10 | darkdrgn2k | does this fax handshake sound wrong? http://www.networkedserver.net/123.wav |
| 18:03.00 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 18:03.35 | [TK]D-Fender | audio sounds fine. |
| 18:04.49 | Penguin | If the password does not protect against the keyed reset, then it's not the password I'm talking about. |
| 18:05.28 | Penguin | If no other password exists, then those phones clearly must not have the feature you're after. |
| 18:06.42 | monsterco | Well, Polycom and Cisco can also be hard reset and their passwords will default but they also have Provisioning which locks them by a large hash key which no one can remove - everytime someone tries to reset them they point to provisioning server and download the same local settings |
| 18:06.47 | monsterco | I am looking for the same on Aastra |
| 18:06.59 | monsterco | maybe better question - Anyone here uses Aastra phones? |
| 18:07.04 | *** join/#asterisk its_jeremy_ (~omghax@gateway/tor-sasl/itsjeremy/x-75806909) |
| 18:07.28 | Penguin | I think ideally the password would be required to reset the phone. If you ever forgot the password and needed to change the phone settings, you would be required to do it by provisioning server. If you didn't have a provisioning server to make the necessary changes or unlock the phone, you'd have to set up one. |
| 18:08.29 | Penguin | Similar to how Vonage locked down the PAP2T ATA. |
| 18:09.00 | monsterco | That's exactly what I want - is that available on Aastra? |
| 18:09.17 | Penguin | It would take intelligence and determination to circumvent your choice of security. |
| 18:09.26 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:09.42 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 18:10.17 | Penguin | When I had a Vonage adapter that I wanted to get into, I had to set up my provisioning server and override vonage's provisioning hostname with local DNS. |
| 18:11.00 | Penguin | It's not something that the average phone user would do in the office if they didn't like the settings you enforced. |
| 18:11.30 | *** join/#asterisk e4voip (uid13742@gateway/web/irccloud.com/x-iiuyqrgeavlhdffu) |
| 18:11.56 | monsterco | Penguin - exactly and I want to do the same but Aastra admin is so vague |
| 18:12.00 | monsterco | admin guide |
| 18:12.14 | Penguin | I'm not an Aastra owner/user, so I don't know what's available. |
| 18:12.42 | Penguin | If it's not in the admim guide, though, it's probably not possible. This assumes the admin guide is complete. |
| 18:13.55 | marceloamorim | guys, when I set (Channel(language)=fr) and the ast_func_write: Function Channel not registered appears at the CLI, so to test I used core show functions and the funcion Channel(item) is there. any tip for me? |
| 18:14.30 | Penguin | CHANNEL(), not Channel() |
| 18:14.40 | Penguin | HUGE difference. |
| 18:15.06 | Penguin | Fuctions are always all caps. |
| 18:15.15 | Penguin | Functions, as well. |
| 18:15.52 | marceloamorim | oh, thx dude, I used to ael, I`m trying to learn at normal language |
| 18:16.51 | Penguin | For future reference: Application(), FUNCTION() |
| 18:20.00 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 18:20.07 | [TK]D-Fender | AEl still requires diaplan functions to be properly capitalized |
| 18:30.00 | *** join/#asterisk crised (~crised@186.67.181.203) |
| 18:30.12 | crised | Any good voip softphone for android and ios? |
| 18:30.19 | crised | That is in Spanish? |
| 18:31.21 | *** join/#asterisk trdillon1 (~tdillon@208.85.166.200) |
| 18:31.40 | trdillon1 | asterisk is randomly reloading on me |
| 18:31.46 | trdillon1 | it is usually like a week apart |
| 18:32.01 | trdillon1 | asterisk version 11.6-cert4 |
| 18:33.16 | trdillon1 | is there anything in asterisk that causes it to reload on its own? |
| 18:33.43 | Qwell | no |
| 18:33.55 | trdillon1 | what should i look for as the cause? |
| 18:36.26 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 18:41.07 | dym | Hey everyone. From time to time i will have an odd problem. Within a phone call (SIP) i will stop hearing the other party, while for the other party the call continues normal. While that happens the network output of said asterisk instance spikes up and re-normalizes after a bit. |
| 18:41.12 | dym | Any idea what could cause this? |
| 18:41.40 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 18:43.12 | Penguin | I'm sure it's the spike in network activity that is causing the hiccup in the media stream. |
| 18:44.29 | dym | :D |
| 18:45.01 | dym | it's gotta be the asterisk machine |
| 18:45.41 | Penguin | How are you monitoring the network activity? |
| 18:55.30 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 18:55.36 | *** part/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 19:01.06 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 19:02.29 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 19:20.14 | *** join/#asterisk tris (tristan@2001:1868:a00a::4) |
| 19:24.50 | *** join/#asterisk theron (~theron@173.252.71.2) |
| 19:25.36 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 19:42.18 | *** join/#asterisk nh-82 (bad5692c@gateway/web/freenode/ip.186.213.105.44) |
| 19:45.20 | nh-82 | any lol hacks? |
| 19:49.09 | *** part/#asterisk nh-82 (bad5692c@gateway/web/freenode/ip.186.213.105.44) |
| 19:56.57 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 19:59.25 | *** join/#asterisk mokmeister (~mokmeiste@95.45.41.60) |
| 20:06.46 | *** join/#asterisk spicyramen_ (~Adium@c-98-210-160-181.hsd1.ca.comcast.net) |
| 20:07.01 | spicyramen_ | hi all anyone have used Asterisk for Wbertc calls to pstn gateways ? |
| 20:07.12 | spicyramen_ | Im loooking for a large scale DTLS/SRTP to RTP transcoder |
| 20:07.20 | spicyramen_ | *webRTC |
| 20:07.30 | spicyramen_ | to enable web browsers call to PSTN |
| 20:07.42 | spicyramen_ | And dont pay Twilio |
| 20:09.40 | [TK]D-Fender | web browsers can't call the pstn. |
| 20:09.48 | [TK]D-Fender | STN != VoIP |
| 20:09.50 | [TK]D-Fender | +P |
| 20:10.04 | [TK]D-Fender | ASTERISK bridges calls |
| 20:10.39 | [TK]D-Fender | And transcodes |
| 20:11.00 | [TK]D-Fender | transcode implies B2BUA realistically speaking. |
| 20:14.02 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-64-136.dynamic.qsc.de) |
| 20:23.45 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 20:31.41 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 20:33.50 | *** join/#asterisk LiohAu (~LiohAu@lse83-h04-89-86-22-242.dsl.sta.abo.bbox.fr) |
| 20:38.24 | spicyramen_ | Browsers can call the PSTN using webrtc and asterisk, what I was referring is this Browser â WS â SIp Registrar (Kamailio) - Sip â Asterisk â> PSTN |
| 20:38.48 | spicyramen_ | but since WebRTC uses DTLS-SRTP/SDES-SRTP and my SIP provider do not support this |
| 20:38.56 | spicyramen_ | I need this secure to non-secure conversion |
| 20:39.05 | spicyramen_ | and was looking for feedback about using asterisk for this |
| 20:39.16 | spicyramen_ | I know doubagngo offers webrtc2sip |
| 20:39.33 | spicyramen_ | but looking for a more robust and active dev solution |
| 20:44.31 | mjordan | that you don't want to pay for. |
| 20:45.27 | mjordan | at the end of the day, Asterisk 11+ has the ability to take SIP over WS/WSS with DTLS for the media, and - as [TK]D-Fender said - bridge that with something else. Could be a SIP/PJSIP channel, could be a DAHDI channel. |
| 20:45.36 | mjordan | (PJSIP being 12+). |
| 20:45.46 | mjordan | You can take Asterisk and make it the gateway. |
| 20:45.54 | *** join/#asterisk tris (tristan@2001:1868:a00a::4) |
| 20:48.13 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:48.40 | spicyramen_ | thanks mjordan |
| 20:48.51 | spicyramen_ | Im willing to pay but havent found a bridge offering this |
| 20:49.07 | spicyramen_ | not sure if you guys know about one |
| 20:49.15 | spicyramen_ | Cisco xcoders dont offer this |
| 20:54.46 | Qwell | spicyramen_: For $500 I'll tell you the name of software that can bridge the two. |
| 20:54.50 | Qwell | Asterisk. kthx, payme |
| 20:55.33 | mjordan | oh well |
| 20:55.39 | mjordan | Qwell: you didn't get paid :-( |
| 20:55.47 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 20:55.49 | Qwell | next time |
| 20:56.10 | Qwell | I don't quite get what his hangup was. |
| 20:56.27 | Qwell | "If only there were some software that could do exactly what I'm asking for, in #asterisk!" |
| 20:56.52 | drmessano | He wants an app with a GUI, ready to go.. two boxes |
| 20:57.12 | drmessano | WebRTC Server: |
| 20:57.15 | drmessano | Other Server: |
| 20:57.26 | drmessano | [OK] [Cancel] |
| 20:57.32 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 20:57.38 | drmessano | 2014 Cisco Inc. |
| 20:57.44 | drmessano | There, GUI is done |
| 20:59.00 | drmessano | First patch, add [Reboot] button if running on top of Windows |
| 21:03.16 | mjordan | drmessano: you should sell that. Make some quick bucks, yo. |
| 21:10.11 | anonymouz666 | mjordan: 11.12.0 |
| 21:10.19 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 21:10.28 | mjordan | anonymouz666: re: your SRTP issues. File a bug; get a SIP trace along with a trace with RTP debugging enabled |
| 21:11.10 | mjordan | I can't guess as to what the problem might be. Either MS Lync is sending something we don't like (and it is on their end or an incompatibility), or there is something else that is mucked up. |
| 21:11.27 | mjordan | If the error happens periodically, I'd guess it's an RTCP crypto problem (again). |
| 21:12.17 | anonymouz666 | yes, it happens periodically. you see the SRTP being sent from Lync and the error starts on CLI |
| 21:14.29 | anonymouz666 | so there's nothing to do with SDES handshake and multiple crypto's. I was thinking that somehow two crypto's line was producing a AES key in SRTP that asterisk supporting just one was unable to understand |
| 21:15.36 | anonymouz666 | I just realize that I misunderstand how SDES works |
| 21:16.38 | mjordan | yup, it only needs one crypto line. The other is supposed to be in case you don't support the key family offered. |
| 21:17.00 | mjordan | the number in the attribute indicates the priority. |
| 21:17.22 | mjordan | My guess is that RTCP isn't getting decoded and we're dropping that on the floor. That doesn't always end up with one way audio. |
| 21:17.28 | mjordan | So you may have other problems then just that. |
| 21:17.56 | anonymouz666 | the priority you mean the MKI |
| 21:18.02 | *** join/#asterisk przerull (~root@4.71.171.175) |
| 21:19.14 | anonymouz666 | oh no, wait, the crypto attribute |
| 21:21.27 | przerull | Is it possible for a sip channel to receive dtmf using RFC2833 and send dtmf using SIP INFO? |
| 21:21.59 | mjordan | the tag, actually |
| 21:22.39 | mjordan | http://tools.ietf.org/html/rfc4568#section-4.1 |
| 21:23.03 | mjordan | btw, some browsers in the past (/cough chrome /cough) would use a tag value of 0, which is invalid |
| 21:23.13 | mjordan | (might have been firefox, now that I think about it) |
| 21:23.25 | mjordan | you may want to double check that Lync is starting with a tag of 1 |
| 21:23.31 | mjordan | starting with a tag of 0 will probably muck something up. |
| 21:24.14 | mjordan | anonymouz666: this will also answer your question about which crypto attribute Asterisk picks and why: http://tools.ietf.org/html/rfc4568#section-5.1.1 |
| 21:26.33 | mjordan | btw, looking at Lync, what they offer wouldn't be accepted without hacking on Asterisk a bit, since we don't currently support key lifetime |
| 21:27.04 | anonymouz666 | yeap, but we can accept that |
| 21:27.08 | anonymouz666 | we can't just honor |
| 21:28.10 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 21:28.17 | *** join/#asterisk Chinorro (~Chino@89.17.194.254) |
| 21:28.37 | anonymouz666 | i just asked file about chan_pjsip and also we don't support (yet :-) |
| 21:28.37 | mjordan | oh goody. They also made up their own crypto attribute. |
| 21:28.49 | mjordan | cryptoscale. |
| 21:29.20 | mjordan | if that's in the offer, I'm not sure what would happen, as that consumes a tag in the sequence of integers that would normally be used by a crypto attribute |
| 21:29.44 | anonymouz666 | I am lucky then :-) can't see this sdp attribute |
| 21:33.44 | anonymouz666 | mjordan: if you don't care, please take a look: http://pastebin.com/Qif3tAFL |
| 21:34.13 | anonymouz666 | the first crypto lifetime, MKI |
| 21:34.18 | anonymouz666 | the second crypt just lifetime |
| 21:34.18 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 21:34.55 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 21:38.27 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 21:39.32 | anonymouz666 | przerull: hello. can't you explain ? if you configure a device, a peer, for a specific DTMF mode, you can't just change the endpoint on the fly |
| 21:40.30 | *** join/#asterisk spicyramen_ (~Adium@c-98-210-160-181.hsd1.ca.comcast.net) |
| 21:42.01 | anonymouz666 | I am guessing you are trying to do a DTMF transcode :-) |
| 21:42.14 | przerull | thanks anonymouz666 |
| 21:42.30 | anonymouz666 | if you want to sent out a INFO why you just configure the endpoint for sending the SIP INFO? |
| 21:42.30 | przerull | yeah, my customer's wanted to "silence" the dtmf tones |
| 21:43.07 | przerull | since the audible tones are generally generated by the terminating trunk (PSTN) |
| 21:43.44 | przerull | I figued that by passing dtmf invalidly (my carrier only supports RFC2833) that it would |
| 21:44.15 | anonymouz666 | but you will invalidade all the dtmf's that could be valid in other situation |
| 21:44.15 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 21:44.17 | przerull | effectively silence the dtmf. It worked, unfortunately, it caused asterisk to no longer receive dtmf on that channel either |
| 21:44.21 | przerull | yeah |
| 21:44.45 | przerull | I was hoping that I might be able to continue to receive RFC2833 but send SIP INFO to the channel |
| 21:45.09 | przerull | but I'm thinking that I'd have to basically patch asterisk to do that (or even the SIP stack) |
| 21:45.22 | anonymouz666 | i might be wrong but i don't see a way to do that |
| 21:45.30 | przerull | I agree |
| 21:46.17 | anonymouz666 | mjordan: with that offer I pasted, can I go ahead to open an issue? |
| 21:46.19 | przerull | I don't see a way to do that either |
| 21:46.29 | przerull | thanks though :) |
| 21:47.08 | anonymouz666 | przerull: sometimes is easier to say to the customer that he desire is not possible |
| 21:47.19 | anonymouz666 | :) |
| 21:51.06 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 21:51.55 | przerull | lol yes. :). Thanks again. |
| 21:54.04 | mjordan | anonymouz666: we won't ever put in a lifetime. |
| 21:54.28 | mjordan | anonymouz666: since you're running a patched version of Asterisk, I'm not sure what we could do at this point. Whatever patch you've done to make it support lifetime probably wasn't sufficient |
| 21:54.57 | mjordan | it is interesting that it is offering two crypto attributes, both with the same suite. |
| 21:55.02 | mjordan | Not sure why it would bother doing that. |
| 21:55.25 | mjordan | Although, actually, Lync only supports that suite, so that makes some sense I guess |
| 21:55.42 | mjordan | anonymouz666: since you've patched Asterisk already, you probably want to patch it further and make it so that Asterisk fakes out a lifetime |
| 21:55.53 | mjordan | I'm fairly sure oej has some patches for this |
| 22:03.35 | anonymouz666 | yes, I am with his branch |
| 22:04.01 | anonymouz666 | makes no sense to open a issue about this then |
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| 22:05.46 | anonymouz666 | It could be related to RTCP like you said, even faking the SDP answer with lifetime the result is the same |
| 22:08.00 | anonymouz666 | with or without sending lifetime in the SDP answer, lync always send the SRTPs to asterisk (and then decode error). |
| 22:10.43 | mjordan | may want to ask oej about it |
| 22:10.48 | mjordan | I know he certified his branch with Lync |
| 22:10.56 | mjordan | so he'd probably be very interested in any issues that popped up |
| 22:12.31 | anonymouz666 | sure, thank you very much for your help. |
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