| 00:19.22 | phix | How does Asterisk perform on OpenWRT devices? |
| 00:20.12 | WIMPy | What are OpenWRT devices? OpenWrt can run on anything. |
| 00:20.23 | WIMPy | Including desktop PCs. |
| 00:20.59 | WIMPy | Or to just answer the question: Probably the same as on any other flavour of Linux. |
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| 01:11.08 | phix | WIMPy: The particular device I have is 720MHz Qualcomm Atheros CPU, 64MB RAM, 8MB flash |
| 01:13.03 | phix | With a USB port so I can add in a ~64GB USB flash or a low powered 2.5" HDD |
| 01:14.45 | WIMPy | Sounds reasonable, but you know it's nearly impossible to give information about Asterisk performance do to the number of variables. |
| 01:15.01 | WIMPy | s/do/due/ |
| 01:15.33 | phix | WIMPy: I will be using alaw atm, maybe G729 or whatever that codec is, down the track |
| 01:16.11 | WIMPy | Transcoding might be a show stopper. |
| 01:16.40 | phix | my VoIP provider supports alaw and G729 |
| 01:16.49 | phix | so no transcoding there |
| 01:17.13 | phix | my SIP phones also support alaw and g729 |
| 01:17.37 | phix | Any other potential show stoppers? |
| 01:18.21 | WIMPy | As long as Asterisk fits in to RAM with te features you need, nothing obvious. |
| 01:19.20 | phix | hmmm, ok, I guess I need to get rid of stuff I don't need |
| 01:19.42 | WIMPy | That's always a good idea. |
| 01:20.11 | phix | Knowing what that is though :\ that is the trick hey |
| 01:28.01 | Katty | SPRING ROLLS. |
| 01:33.14 | phix | Use the power of spring rolls to resolve my asterisk issues? |
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| 08:25.39 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.5.0 (2015/08/07), 11.19.0 (2015/08/07), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 09:10.53 | *** join/#asterisk rox (~rox@BSN-142-168-6.static.siol.net) |
| 09:12.12 | rox | I have a weird thing going on in my queues; I have SIP agents in queues and in some cases, when asterisk calls an available SIP agent, who is a member of a queue, the call is immediately canceled |
| 09:12.19 | rox | that is |
| 09:12.28 | rox | asterisk sends INVITE to agent |
| 09:12.54 | rox | and immediately after INVITE asterisk sends SIP CANCEL message to the agent |
| 09:13.08 | rox | and the call is logged as "NO ANSWER" |
| 09:14.14 | rox | this happens when the agent in question is not busy, does not have other calls ringing or otherwise, but Asterisk for some reason cancels the call |
| 09:14.23 | rox | does anybody have an idea, what could be going on? |
| 09:14.55 | phix | hmmmm |
| 09:15.01 | rox | I have had a similar issue, when timeout parameter in queues.conf was longer than timout parameter of Dial command |
| 09:15.09 | phix | Is there a firewall or NAT between the server and the agents? |
| 09:15.21 | rox | in that case the next call was canceled due to queue timeout |
| 09:15.32 | rox | no, they are on the same network |
| 09:15.43 | phix | I have had issues where a connection has timed out but either end hadn't realised it until trying to send a call to it |
| 09:15.46 | phix | oh ok |
| 09:16.07 | rox | the connection doesn't time out, the CANCEL message is sent immediately after INVITE |
| 09:17.21 | rox | Agent's client usually responds with TRYING and sometimes even manages to squeeze the RINGING message, but CANCEL from PBX usually comes before any ringing starts or before the client's UI can react to the call in any detectable manner |
| 09:17.30 | rox | from the agent's point of view, nothing happens |
| 09:19.23 | rox | i know that if timeout in queues.conf is longer than Dial timeout, that the timeout is going to affect the next call, i.e. if queues.conf timeout is 10s and you dial with 8s timeout, if the call times out and is started again, after 2 seconds the queues.conf is going to come around and the PBX is going to send a CANCEL message to the agent during the next call |
| 09:19.37 | rox | my current issue looks very similar, but i can't figure out what could be trigering it |
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| 09:55.12 | phix | rox: *shrugs* maybe a timeout is set too low/ |
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| 10:07.43 | rox | phix: thank you, i'm checking all possible timeouts |
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| 11:19.07 | phix | rox: ok cool |
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| 13:22.08 | guepi | Hi everybody |
| 13:22.23 | phpboy | hi |
| 13:22.55 | guepi | I'm very nexw in asterisk and I use it @ home |
| 13:23.32 | guepi | asterisk is connected to my phone provider through a SIP account |
| 13:24.07 | phpboy | ok |
| 13:24.35 | pjensen00 | Asterisk and SIP are dating. |
| 13:24.43 | phpboy | correct |
| 13:24.57 | guepi | and when I try to call home, I've got a WARNING : check_auth: username mismatch, have <my-sip-account-I-defined>, digest has <anonymous> |
| 13:25.24 | guepi | and my call end to my "old analogic phone" |
| 13:25.45 | guepi | and I don't find where this anonymous is defined |
| 13:26.01 | pjensen00 | What version of asterisk are you using |
| 13:26.08 | guepi | I've just modified my sip.conf and my extensions.conf |
| 13:26.35 | guepi | Asterisk 11.7.0~dfsg-1ubuntu1 |
| 13:26.57 | pjensen00 | Use pastebin and paste your sip.conf and extensions.conf relevant portions. |
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| 13:31.44 | guepi | my sip.conf : http://pastebin.com/sSyFv5bU |
| 13:32.48 | guepi | my extensions.conf : http://pastebin.com/P2wwUdNv |
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| 13:35.55 | guepi | two questions : |
| 13:36.24 | *** join/#asterisk kolko (~kolko@46.48.58.17) |
| 13:37.25 | guepi | my phone provider is also my ADSL provider - I've got their ADSL box ; my SIP is connected to a router behind the box - May my provider prevent me to use SIP when I'm on a LAN ? |
| 13:38.12 | guepi | I've read my provider try to prevent his regular users to plug an asterisk on their SIP account ... may it be my issue ? |
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| 13:47.36 | guepi | pjensen00, any idea ? |
| 13:47.53 | pjensen00 | ah sorry, was in a different window. One second |
| 13:48.28 | guepi | ok thanks |
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| 13:50.34 | pjensen00 | Hrm. I'm much more familiar with pjsip, but I'll take a look in a few minutes. I am at work so my focus is split. |
| 13:50.42 | [TK]D-Fender | guepi: You have 2 peers for the same host. This is bad. |
| 13:51.19 | *** join/#asterisk mrmccrac (~Adium@2001:18e8:2:1009:f000::f7b) |
| 13:51.57 | guepi | [TK]D-Fender, ok. The first one is to send call and the second one is to receive calls |
| 13:52.45 | WIMPy | Why isn't that just one? |
| 13:52.58 | mrmccrac | i have tcpenable=yes and transport=tcp and tcpbindaddr=0.0.0.0 in my [general] section of sip.conf, and i can see my linux host is listening on the sip port with tcp, but when its sending out registration packets to the remote peer its sending UDP packets still (confirmed with tcpdump), any ideas what else i need to set to get it to register to remote peers with tcp? |
| 13:53.05 | [TK]D-Fender | guepi: Not good. |
| 13:53.12 | [TK]D-Fender | guepi: Incoming calls will ALWAYS hit the first |
| 13:53.32 | [TK]D-Fender | guepi: This is a design failure. You do not need 2 |
| 13:54.13 | guepi | [TK]D-Fender, ok. I'll remove the second one. |
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| 13:54.38 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:55.27 | mrmccrac | http://pastebin.com/dCw5t2Gy |
| 13:56.31 | mrmccrac | http://pastebin.com/xn13qTbD |
| 13:56.41 | mrmccrac | restarted asterisk but still sending UDP packets... |
| 13:57.08 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/branches/13/configs/samples/sip.conf.sample?revision=434654&view=markup |
| 13:57.18 | [TK]D-Fender | the sample config tells you the syntax for this |
| 13:57.28 | mrmccrac | ah |
| 14:00.06 | guepi | Thank you pjensen00 and [TK]D-Fender ! It works !!!! |
| 14:00.43 | [TK]D-Fender | guepi: You're welcome |
| 14:01.38 | *** join/#asterisk mccar (~chatzilla@195.208.45.210) |
| 14:01.50 | pjensen00 | guepi: Ha, you're welcome but D did all the work |
| 14:01.56 | mccar | Hi! I have a strange problem with attended transfer. When I press ** I get -- Remotely bridging SIP/line_73346619973-00000004 and SIP/105-00000005 After pressing "*" 4 or 5 times I get voice message "Transfer". Why this happens? |
| 14:02.31 | guepi | pjensen00, I receive call on my SIP phone. That is all I wanted to. |
| 14:03.43 | guepi | pjensen00, & [TK]D-Fender : Now I will redirect SPAM calls to the extension 201 : playing monkeys ! :D |
| 14:03.51 | [TK]D-Fender | mccar: because you told your dial to allow it |
| 14:05.34 | [TK]D-Fender | mccar: Look at your DIAL |
| 14:05.36 | pjensen00 | guepi: Re-direct to a dial plan that does silence detection and plays a generic recording along the lines of "Ah yes, but could you repeat that?" |
| 14:07.46 | mrmccrac | thinking i'm getting closer [TK]D-Fender |
| 14:08.17 | guepi | pjensen00, seems to be the stuff I'd like to do, but it seems to be quite difficult |
| 14:08.34 | guepi | pjensen00, maybe the next version :) |
| 14:10.52 | guepi | see you later guys ! Have a nice day ! |
| 14:11.53 | mccar | So why transfer is available only after 4-5 press of *? |
| 14:12.33 | mrmccrac | my register line is now register => SIP/<redacted>:<redacted>:::tcp@<redacted>:8060 |
| 14:14.22 | [TK]D-Fender | mccar: because you didn't press it fast enough in between the first few |
| 14:14.58 | mccar | I did as fast as I can |
| 14:15.15 | [TK]D-Fender | maybe it's losing digits. |
| 14:15.28 | [TK]D-Fender | What device are you doing that with? |
| 14:16.00 | mccar | Linksys SPA962 |
| 14:16.10 | *** join/#asterisk jkroon (~jkroon@197.96.224.13) |
| 14:16.32 | [TK]D-Fender | Then you shouldn't be using "t" anyway |
| 14:16.38 | [TK]D-Fender | It has REAL transfer functionality |
| 14:16.46 | [TK]D-Fender | use the phone the way it was meant to be used |
| 14:17.00 | [TK]D-Fender | DTMF call-handling = ass |
| 14:17.20 | mccar | It worked before I transfer asterisk on new server... |
| 14:17.24 | mrmccrac | [2015-08-20 14:17:04] WARNING[18651]: sip/config_parser.c:132 sip_parse_register_line: Format for registration is [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] at line 9 |
| 14:17.40 | mccar | All conf files unchanged.. |
| 14:18.17 | [TK]D-Fender | mrmccracmy register line is now register => SIP/<redacted>:<redacted>:::tcp@<redacted>:8060 |
| 14:18.22 | [TK]D-Fender | this does NOT look like that sample |
| 14:18.35 | mrmccrac | i saw this sample: |
| 14:18.37 | mrmccrac | SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port] |
| 14:18.53 | mrmccrac | anyhow its not complaining now |
| 14:19.30 | mrmccrac | this seems to work, maybe: |
| 14:20.03 | mrmccrac | register => tcp://<redacted>:<redacted>@<redacted>:8060 |
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| 14:26.04 | mccar | You're right! The reason in short wait time.. exten => _30XX,n,WaitExten(1) is not enough |
| 14:26.34 | mccar | I've changed it to 3 seconds and now its ok |
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| 14:41.30 | mrmccrac | hopefully just firewall issues now, i noticed with udp it tries to register shortly after restarting asterisk, with tcp now it doesn't do anything at all until i attempt a call |
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| 15:19.16 | Zogot | anyone use CEL? i have a wierd scenario where app_start doesn't trigger the 2nd Dial application if they are in a row |
| 15:19.25 | Zogot | but will if i add another application between the 2 dials |
| 15:19.28 | Zogot | like a NoOp |
| 15:20.01 | Zogot | exten => 105,1,Dial(phone) \n exten => 105,2,Dial(phone2) \n only 1 app_start in CEL |
| 15:20.30 | Zogot | exten => 105,1,Dial(phone) \n exten => 105,2,NoOp(nothing) \n exten => 105,3,Dial(phone2) \n 2 app_starts for Dial |
| 15:21.19 | WIMPy | What is app_start? |
| 15:21.26 | Zogot | from CEL |
| 15:21.33 | Zogot | https://wiki.asterisk.org/wiki/display/AST/CEL+Events+and+Fields |
| 15:21.48 | WIMPy | So the other way round? |
| 15:22.23 | Zogot | huh? |
| 15:22.51 | *** join/#asterisk CeBe (~CeBe@xd9be49a5.dyn.telefonica.de) |
| 15:23.03 | Zogot | if you mean, the even fires after the application has fired, thats what it does yep :), i can add more applications to the cel.conf to listen for (and it'll fire the app_start for those also) |
| 15:23.05 | WIMPy | it's not app_start that doesn't trigger, it's app_start that isn't being triggered. |
| 15:23.19 | Zogot | oh yes |
| 15:23.21 | Zogot | from my sentence |
| 15:23.22 | Zogot | ye |
| 15:23.35 | Zogot | app start isn't triggered by the 2nd dial application in a row* :) |
| 15:24.34 | WIMPy | Sounds suspect. |
| 15:28.04 | Cuzner | git status |
| 15:28.07 | Cuzner | oops |
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| 15:35.19 | SFJulie1 | hello |
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| 15:39.11 | SFJulie1 | I am trying to keep track of caller/callees inside queue_app application with AMI events, and I was wondering where I can find the Events send according to the states of the caller/callee |
| 15:40.12 | SFJulie1 | right now, after searching all through the wiki.asterisk.org I resorted to this https://github.com/asterisk/asterisk/blob/2d20379f8e5b3ca26678d2538104f0a35f87e12f/apps/app_queue.c#L6285 |
| 15:40.24 | SFJulie1 | am I getting overboard? |
| 15:43.46 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 15:44.13 | SFJulie1 | btw does L6485 says: if user has been hangedup by agent while being put in channel, then hold time will not be counted? |
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| 16:20.16 | Chainsaw | My ITSP just responded SIP 486 "PC Load Letter". |
| 16:20.20 | Chainsaw | Now I've seen everything... |
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| 16:25.04 | file | Chainsaw, :D |
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| 16:28.53 | WIMPy | Errr |
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| 16:33.06 | edong23 | hey guys, im having an issue... we have some atas (which i cannot provision, because they are provisioned by another company, please dont ask why. its crap) but... they use feature code *72 and send that immediately to my asterisk box. Now, i need call forwarding to work, but the digits dialed needs to be captured and then sent to the database. but it no worky. I tried waitexten but that dials the extension... lol. what is the application to |
| 16:33.06 | edong23 | <PROTECTED> |
| 16:34.27 | WIMPy | It dials *72, waits for an answer and then sends the destination inband? |
| 16:35.08 | edong23 | it dials *72 immediatly. The diaplan i can decode and anything *XX gets sent to asterisk |
| 16:35.12 | [TK]D-Fender | waitexten ... or read |
| 16:35.29 | edong23 | after that, i have built an exten for *72 but need to get the digits the user will dial inband |
| 16:35.34 | WIMPy | Yes, that would be Read() then. |
| 16:35.37 | edong23 | [TK]D-Fender: read i read about... |
| 16:35.38 | edong23 | ok |
| 16:35.40 | edong23 | let me try that |
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| 16:35.44 | WIMPy | Craptastic. |
| 16:35.51 | edong23 | waitexten doesnt seem to work... it dials the number after it gets it |
| 16:35.52 | edong23 | lol |
| 16:36.26 | edong23 | exten => _*72,n,WaitExten(20) |
| 16:36.26 | edong23 | exten => _*72,n,Set(DB(CFIM/${CALLERID(num)})=${EXTEN}) |
| 16:36.35 | [TK]D-Fender | edong23: No. |
| 16:36.39 | edong23 | waitexten dials the number it recieves, it doesnt set it... |
| 16:36.46 | WIMPy | Depends on what your context looks like, but, yes it's possible to use it, but not ideal. |
| 16:36.49 | [TK]D-Fender | edong23: Where is the PATTERN it is supposed to land on? |
| 16:36.54 | [TK]D-Fender | READ THE INSTRUCTIONS. |
| 16:37.16 | [TK]D-Fender | edong23: It jumps to the dialed extension based on patterns in that context |
| 16:37.44 | edong23 | other than answer and hangup, that is literally all that is in this context |
| 16:38.03 | WIMPy | Use Read. |
| 16:38.16 | edong23 | let me try read |
| 16:40.59 | edong23 | [TK]D-Fender: i read about waitexten. thats why i thought it would work. but.. i must hafe missed something |
| 16:42.25 | WIMPy | It can be made to work if you put it in its own context and make an extension to match all possible destinations. |
| 16:42.36 | WIMPy | But Read() is much better suited to this scenario. |
| 16:43.09 | [TK]D-Fender | edong23exten => _*72,n,Set(DB(CFIM/${CALLERID(num)})=${EXTEN}) <- this is NOT how it works |
| 16:43.18 | [TK]D-Fender | it JUMPS to the matching exten in the context you are in. |
| 16:43.24 | [TK]D-Fender | Read the instructions again |
| 16:43.37 | [TK]D-Fender | it does NOT continue in-line unless nothing was entered at all and it times out |
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| 16:44.43 | [TK]D-Fender | edong23exten => _*72,n,Set(DB(CFIM/${CALLERID(num)})=${EXTEN}) <- ${EXTEN] is also where you ARE in the dialplan ... and on this line it would be *72. Always. You should know just from putting it there that this is pointless |
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| 16:47.43 | edong23 | [TK]D-Fender: ok. i am reading that now. you are right. And i also see why it is calling out. This context is included in another. let me see |
| 16:48.24 | [TK]D-Fender | You let incoming calls from your provider land in a place that can dial OUT? This is a .... |
| 16:48.26 | [TK]D-Fender | ~soso |
| 16:48.26 | infobot | [~soso] Shoot-On-Sight Offense |
| 16:48.34 | edong23 | no |
| 16:48.44 | edong23 | imcoming? |
| 16:48.50 | edong23 | these are dialed digits from the user |
| 16:50.17 | [TK]D-Fender | Ok, so this is what a user is doing on a phone? |
| 16:50.22 | [TK]D-Fender | Then you need to jump out of there |
| 16:50.30 | edong23 | yes |
| 16:50.35 | edong23 | yeah, agreed |
| 16:55.34 | edong23 | however, without all that, read worked perfectly |
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| 18:28.48 | GlenK | hi there. Calls from a particular office, the quality is just fine on their end, meaning they can hear things just fine. but people on the other end, the voice is all choppy and weird. any ideas what might be going on, or how to go about diagnosing? |
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| 18:54.10 | jimmyz2 | Hi I hope this is the right place, i'm trying to compile 13.5.0 LTS on ubuntu 14.04 but when i run ./contrib/scripts/install_prereq install i get no output it just exits |
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| 20:39.10 | saint_ | hi all - can someone recommend a sip WoLAN phone ? My Yealinks DECT / SIP just crapped on me. |
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| 21:00.40 | Plixic | Anyone point in the right direction on an audio issue with an asterisk server hosted VPS? Calls initiating from an internal line forwarding to external using follow me has no audio .. I think it's a NAT issue but their is no firewall |
| 21:00.47 | Plixic | and it's on a public IP |
| 21:00.56 | Plixic | transfers and incoming/outgoing all work fine |
| 21:03.48 | *** join/#asterisk jms77 (80677c8d@gateway/web/cgi-irc/kiwiirc.com/ip.128.103.124.141) |
| 21:06.41 | jms77 | hey i have a question about detecting loss of a media stream, anyone got a minute to help me out? |
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| 21:27.00 | WIMPy | jms77: Ask your question |
| 21:27.05 | WIMPy | ~ask |
| 21:27.05 | 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. |
| 21:27.17 | jms77 | gotcha |
| 21:27.42 | jms77 | I'm trying to detect when there is no media stream as opposed to silence in an ARI app |
| 21:28.09 | jms77 | But based on what I've read there is no dinstinciton |
| 21:28.43 | jms77 | Is that the case? |
| 21:29.08 | jms77 | I want to know if a client has exited abnormally and attempt to reconnect the channel |
| 21:29.31 | jms77 | But the channel will persist for a quite awhile |
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| 21:30.07 | WIMPy | I don't know what ARI gives you, but on chan_sip that is handled by 'rtptimeout'/ |
| 21:30.09 | WIMPy | . |
| 21:30.34 | jms77 | thanks |
| 21:31.05 | babak | Hi,callers call queue(agents) and ask some people telephone numbers,agents search data base and fine phone number , how to automate playing telephone number for callers while agent became free for next call ? |
| 21:31.36 | WIMPy | SayDigits() |
| 21:32.56 | babak | WIMPy: agent found telephone number not Asterisk , we may send it by AMI but how connect it with call |
| 21:33.50 | WIMPy | If you use AMI, redirect the cannel to a piece of dialplan that uses SayDigits and maybe loops or whatever. |
| 21:35.20 | babak | WIMPy: if without AMI is possible Iprefer to do it, with AMI How to redirect a call that still controlled by queue application ? |
| 21:36.00 | WIMPy | "redirect" was th keyword. |
| 21:36.24 | WIMPy | You can either do it via AMI or you write an application with ARI. |
| 21:36.39 | robmal | AGI also can has dtmf. |
| 21:36.45 | WIMPy | You need to get the number to be said in to Asterisk. |
| 21:37.03 | robmal | It can has. |
| 21:37.14 | robmal | https://wiki.asterisk.org/wiki/display/AST/AGI+Commands |
| 21:37.17 | WIMPy | Yes, if you can ind the result in an AGI, you can simply use that. |
| 21:37.50 | WIMPy | Depends on how yu can make the relation between the channel and the number to be announced. |
| 21:38.00 | robmal | Yes. |
| 21:38.34 | babak | WIMPy: I should monitor all event to find appropriate channel ? |
| 21:39.03 | robmal | That's why i suggest AGI. |
| 21:39.14 | WIMPy | That depends entirely on your application / workflow how you might be able to find out. |
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| 22:14.43 | _-Jon-_ | Hello all |
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| 22:49.03 | dan_j | Hi. Whats the best way to replicate voicemail messages to a number of servers in more than one location? |
| 22:49.39 | dan_j | Assuming mailboxes could be read/written from any location. |
| 22:50.32 | robmal | drbd |
| 22:56.44 | dan_j | really? I didnt think about that option. but drbd can only be mounted on one server at a time? |
| 22:57.00 | robmal | Yes. |
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