| 00:20.00 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:20.00 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 00:37.18 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 01:32.21 | drmessano | ~happyclownpbx |
| 01:32.22 | infobot | currently in closed beta, approaching 12GB in size, uses Asterisk for its core, it pwns, eats your children (seriously), and is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone |
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| 05:16.24 | tuxd00d | drmessano: Whatâs the story with HappyClownPBX? |
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| 09:41.01 | *** join/#asterisk polysics (~polysics@95.85.20.146) |
| 09:41.10 | polysics | hello y'all! |
| 09:41.32 | polysics | I am trying to get Asterisk to playback a Watson TTS URL |
| 09:41.43 | polysics | but I think I have hit some kind of byte limit |
| 09:42.18 | polysics | Playback('USER:PASS@stream.watsonplatform.net/text-to-speech/api/v1/synthesize?accept=audio/wav&text=Hello%20world&voice=en-US_AllisonVoice') is simple enough |
| 09:43.59 | polysics | or maybe it is those ampersands! |
| 09:51.01 | polysics | I just need to figure out "why" the above does not work and we should be set |
| 09:59.18 | polysics | but I actually have a simpler question |
| 09:59.43 | polysics | does Asterisk play an HTTP URL while it is downloading, or is the file saved THEN played? |
| 10:02.52 | file | it saves it and then plays currently |
| 10:04.12 | polysics | so if I have another application fronting this, I might as well download and manage those files from there and just tell Asterisk to play them |
| 10:05.00 | file | that is an option, yes |
| 10:05.18 | polysics | is there any others? |
| 10:05.34 | polysics | out of curiosity, that would work fine as far as I am concerned :) |
| 10:05.39 | file | no? |
| 10:05.45 | *** part/#asterisk betz (~betz@ptr-91b0rf2cdydiv2t3yos.18120a2.ip6.access.telenet.be) |
| 10:06.09 | polysics | I guess it can either be the web app downloading the files or a bash script called from AGI but it does not substantially change much |
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| 11:45.10 | *** join/#asterisk xochilpili (~xochilpil@unaffiliated/xochilpili) |
| 11:45.29 | xochilpili | hi all |
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| 12:02.29 | *** join/#asterisk tcpdump (sid47591@gateway/web/irccloud.com/x-peblzluxyykpesfb) |
| 12:02.35 | tcpdump | hello everyone. |
| 12:05.32 | tcpdump | I am planning to use Asterisk for a SIP/STUN proxy/server. I am trying to wrap my head around one more concept. So, I have an Asterisk server and two SIP clients, on different LANs (divided by the Internet), trying to call each other. They're both connected to the SIP server. One originates a call to the other. Obviously the Asterisk server sends to second client a "ring". When they pick up the second client does a STUN |
| 12:05.32 | tcpdump | binding request, then I assume the Asterisk server relays the public IP, port, and NAT strategy to the calling Client. |
| 12:05.36 | tcpdump | That looks right so far? |
| 12:06.32 | tcpdump | So, at that point does Client A originate a secondary SIP (point to point) connection with Client B, or is that just straight up UDP, and both use the original SIP connection for call control? |
| 12:06.42 | file | Asterisk isn't a SIP/STUN proxy - each leg is independent, and the connection is between the client and Asterisk - generally Asterisk forwards media |
| 12:07.04 | file | it can be configured to go directly but it provides the IP address/port as given to it by the client |
| 12:07.33 | wabbits | tcpdump google B2BUA = back to back user agent |
| 12:10.03 | tcpdump | file: so I was planning an using it to transport RTSP between the two clients via direct connect (using STUN). |
| 12:10.08 | tcpdump | wabbits: thx |
| 12:10.11 | tcpdump | googles |
| 12:10.12 | file | Asterisk does not support RTSP |
| 12:10.44 | file | if you are referring to RTP it won't forward STUN traffic between both sides, it'd be through Asterisk and not directly |
| 12:11.00 | tcpdump | file: yes, I apologize - RTP |
| 12:12.10 | tcpdump | file: so all of my RTP traffic would be proxied via the server then? |
| 12:12.18 | file | yes. |
| 12:12.35 | tcpdump | https://image.slidesharecdn.com/sip-intro-101103170955-phpapp01/95/sip-for-geeks-25-638.jpg?cb=1422649144 |
| 12:12.49 | file | Asterisk isn't a proxy. |
| 12:13.01 | tcpdump | This diagram depicts a P2P session for RTP - Is that a different technology than Asterisk? |
| 12:13.18 | file | it's not a different technology, it's a different deployment model |
| 12:14.21 | file | as wabbits mentioned, Asterisk is a B2BUA |
| 12:15.06 | wabbits | tcpdump I suggest we could help you better if you described what you want to do in non technical terms. |
| 12:17.51 | tcpdump | wabbits: thats a fair idea - I want to make a linux based media streaming device/server that serves up video over RTSP/RTP. I want to allow clients using an app anywhere in the world access it at any time. The thought was to use SIP and it's control protocols to initiate streaming sessions by "calling" each other. |
| 12:18.33 | wabbits | Do you really want RTSP? |
| 12:19.21 | wabbits | you can serve video over rtp |
| 12:20.00 | wabbits | tcpdump ^ |
| 12:20.39 | Samot | Why do you even need Asterisk for this? |
| 12:21.15 | tcpdump | wabbits: i probably do not want RTSP, just h.264 over RTP |
| 12:21.19 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 12:22.13 | wabbits | sound pretty easy if I understand the usage model. user A calls the server and views video. Is that it? |
| 12:22.22 | Samot | You also probably don't want the media to be P2P. |
| 12:23.18 | *** join/#asterisk pdugas (~retentive@2001:558:6011:71:3d78:26dd:beb:dc2) |
| 12:23.30 | tcpdump | Samot: the biggest reason I was looking at Asterisk is because I was told (by someone whose done it) that Asterisk was an all in one SIP server that could facilitate STUN (P2P) sessions between two clients, and then handle transporting it if the P2P session failed for some reason. |
| 12:23.40 | Samot | No |
| 12:23.45 | tcpdump | wabbits: yea, at a rudimentary level thats it. |
| 12:23.46 | Samot | They are wrong. |
| 12:23.59 | Samot | Asterisk is a Telephony Engine and a B2BAU |
| 12:24.29 | Samot | You tell Asterisk to use a STUN server some where else. |
| 12:24.36 | Samot | Asterisk does not do STUN or TURN |
| 12:24.46 | tcpdump | First off, let me say thanks for the patients. Im a sysamin/network admin and Im trying to wrap my head around these protocols/products that Ive never used before. |
| 12:25.26 | sekil | hello |
| 12:25.39 | tcpdump | Ok, so lets say we take the STUN factor out of the equation Asterisk could absolutely facilitate SIP/RTP calls between two clients it sounds like. |
| 12:25.49 | Samot | Yes. |
| 12:25.55 | Samot | It's a B2BUA |
| 12:25.57 | tcpdump | And all of the data would traverse my SIP server, basically. |
| 12:26.01 | Samot | Yes. |
| 12:26.09 | Samot | Google: Back to Back User Agent |
| 12:26.16 | Samot | That's what Asterisk does. |
| 12:26.28 | Samot | The calls are between Asterisk and the endpoint. |
| 12:26.29 | tcpdump | Samot: yea, wabbits sent that earlier - Im reading that now. |
| 12:26.37 | Samot | Asterisk bridges the channels together. |
| 12:26.43 | *** join/#asterisk pdugas (~retentive@2001:558:6011:71:3d78:26dd:beb:dc2) |
| 12:27.00 | file | if it's just serving up video files then there are better options out there really... |
| 12:27.12 | tcpdump | so each device would have a "phone number" and more or less register at start time, and remain connected via TCP/SIP. |
| 12:27.13 | *** join/#asterisk retentiveboy (~retentive@c-73-82-30-193.hsd1.ga.comcast.net) |
| 12:27.22 | tcpdump | file: its real time streaming, actually. |
| 12:27.26 | tcpdump | Its a camera. |
| 12:27.28 | Samot | Well RTP is 100% UDP |
| 12:27.34 | file | even then, there are better options |
| 12:27.40 | Samot | You can do your SIP Signaling over TCP |
| 12:27.45 | tcpdump | file: what comes to mind? |
| 12:27.51 | Samot | But RTP is _always_ UDP |
| 12:27.54 | file | any streaming software? |
| 12:28.07 | Samot | Well what is the camera streaming? |
| 12:28.26 | Samot | And is it streaming all the time or only during a call? |
| 12:28.37 | tcpdump | Just live video Samot - imagine a security camera in your home, and you want to watch it in your app. |
| 12:28.40 | tcpdump | only during the call. |
| 12:28.45 | sekil | vlc? |
| 12:28.47 | Samot | Well there is a different. |
| 12:28.51 | Samot | Difference. |
| 12:28.53 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 12:29.03 | Samot | One is just stream video over IP |
| 12:29.11 | Samot | The camera will have software mostly to remote watch that stream. |
| 12:29.15 | wabbits | veal lettuce and cucumber. delicious! |
| 12:29.50 | Samot | There is a difference between an IP camera |
| 12:29.58 | Samot | And a phone doing IP over SIP |
| 12:30.05 | Samot | With video |
| 12:30.51 | Samot | If it is just a 24/7 live video stream, then it has nothing to do with Asterisk or SIP/VoIP |
| 12:30.58 | Samot | They would be two separate things. |
| 12:31.30 | tcpdump | Samot: its not 24/7 only when the client wants to view it. |
| 12:31.42 | Samot | There is still a difference. |
| 12:32.06 | Samot | There is a difference from remotely accessing the camera and turn the video on... |
| 12:32.17 | Samot | That's mainly HTTP requests... |
| 12:32.31 | wabbits | sound like a client->server deal and no pear protocol required. |
| 12:32.38 | Samot | Vs. activating the video stream over SIP/RTP |
| 12:32.45 | wabbits | s/pear/peer |
| 12:33.06 | Samot | So far the IP camera doesn't sound like anything that would require voice. |
| 12:33.08 | Samot | It's just a camera. |
| 12:33.18 | Samot | It can be accessed remotely and video stream turned on |
| 12:33.18 | tcpdump | It will do audio as well. |
| 12:33.21 | Samot | OK |
| 12:33.25 | tcpdump | audio/video. |
| 12:33.26 | Samot | That has nothing to do with a CALL |
| 12:33.37 | Samot | It's an IP Camera. |
| 12:33.45 | Samot | I had those at my data center. |
| 12:33.53 | Samot | I could log in remotely, watch and listen |
| 12:34.00 | Samot | Nothing to do with a call |
| 12:34.49 | *** join/#asterisk fblackburn (~fblackbur@modemcable094.94-70-69.static.videotron.ca) |
| 12:36.19 | Samot | Do you understand what you are confusing? |
| 12:36.45 | Samot | IP Camera that does audio/video over HTTP/HTTPS |
| 12:37.00 | Samot | Vs an IP Phone with a Camera that does audio/video over SIP/RTP. |
| 12:37.35 | wabbits | tcpdump which does your camera support? |
| 12:37.52 | tcpdump | Yes and no. I understand that those cameras use some protocol (varying from camera to camera) to facilitate connectivity within a LAN to an outside device. |
| 12:37.57 | tcpdump | wabbits: Whatever I impliment. :D |
| 12:38.04 | Samot | IP Camera |
| 12:38.07 | tcpdump | Thus this discussion. |
| 12:38.07 | Samot | VERSUS |
| 12:38.11 | Samot | IP Phone with Video |
| 12:38.17 | Samot | TWO DIFFERENT THINGS |
| 12:38.17 | wabbits | so you are making the camera? |
| 12:38.22 | tcpdump | I am |
| 12:38.35 | Samot | Wait. |
| 12:38.38 | wabbits | then you are out of your depth |
| 12:38.41 | tcpdump | I dont see why my camera cant just be a phone? |
| 12:38.43 | Samot | You're "building the device"? |
| 12:38.50 | Samot | Because it needs to support SIP |
| 12:39.12 | Samot | It needs to support Video OVER SIP |
| 12:39.50 | tcpdump | I mean it doesnt have to , but it does need to work in most any network environment, even in low bandwidth, which I understand SIP handles quite well. |
| 12:39.50 | wabbits | you can buy cameras that do sip |
| 12:40.03 | sekil | there are cameras with support for RSTP/RTP stacks |
| 12:40.05 | Samot | tcpdump: Stop |
| 12:40.14 | Samot | You do not understand what you are trying to do. |
| 12:40.22 | Samot | SIP is not an alternative to HTTP for a Camera. |
| 12:40.56 | Samot | A video phone will initiate video only when there is a CALL |
| 12:41.07 | tcpdump | k |
| 12:41.11 | Samot | So the camera has to call an endpoint |
| 12:41.16 | Samot | or the endpoint has to call the camera. |
| 12:41.56 | *** part/#asterisk fblackburn (~fblackbur@modemcable094.94-70-69.static.videotron.ca) |
| 12:42.12 | Samot | tcpdump; What is the actual end goal? |
| 12:42.16 | Samot | What is the purpose of this? |
| 12:42.34 | Samot | How is it supposed to function from the user perspective? |
| 12:44.40 | Samot | You said before this is for a "security" camera. |
| 12:44.51 | Samot | But it would only be one when the user wants to view it. |
| 12:44.57 | Samot | That's not really a "security" camera. |
| 12:45.37 | Samot | This is sounding more like a "door" camera. |
| 12:45.46 | Samot | Just so you can see who is at the door and buzzing you. |
| 12:47.52 | Samot | tcpdump: ^^ |
| 12:48.08 | tcpdump | reads |
| 12:48.46 | tcpdump | Samot: im uisng a different protocol to handle the full time recording. |
| 12:49.00 | tcpdump | so yea, in this case it would be on demand only for this piece. |
| 12:49.03 | Samot | So then the video has nothing to do with the call. |
| 12:49.06 | Samot | No. |
| 12:49.12 | Samot | The video has nothing do with the call |
| 12:49.18 | Samot | They are separate. |
| 12:49.35 | Samot | The call just can't jump in and grab the video stream.. |
| 12:49.42 | Samot | From a device not doing SIP/RTP |
| 12:50.43 | Samot | tcpdump: Let me explain this another way.. |
| 12:50.53 | tcpdump | please |
| 12:51.04 | Samot | Either the video is over SIP/RTP and thus connected to a call... |
| 12:51.23 | Samot | Or it's over another protocol and is completely separate from the SIP call |
| 12:51.35 | Samot | If you are recording the video 24/7 |
| 12:51.43 | xochilpili | i have a cisco spa3102, and an asterisk server and im able to make internal calls and outgoing calls, but when i do an outgoing call it disconnects me from internet, im using ADSL (ISP) |
| 12:51.48 | Samot | Or streaming that video 24/7 to someplace to be viewed... |
| 12:51.51 | Samot | It's separate. |
| 12:51.53 | Samot | Completely |
| 12:51.59 | xochilpili | does any one know what cause this? |
| 12:52.17 | Samot | Why would it disconnect you from the Internet? |
| 12:52.26 | xochilpili | Samot, no idea :D |
| 12:52.28 | Samot | Are you using the 3102 as a router as well? |
| 12:53.04 | xochilpili | i have set the 3102 as a "bridge" but no idea why it disconnects from internet |
| 12:53.14 | Samot | What do you mean "bridge" |
| 12:54.22 | xochilpili | Samot, http://www.cisco.com/c/es_mx/support/docs/unified-communications/spa3102-voice-gateway-router/108733-pqa-108733.html |
| 12:54.51 | xochilpili | my english is not good enough to explain "what i meant for bridge" |
| 12:55.51 | Samot | OK well |
| 12:56.07 | Samot | How is the network setup? |
| 12:56.20 | Samot | Modem -> SPA3102 -> Router/Switch? |
| 12:56.37 | tcpdump | Samot: Thanks for helping me wrap my head around it. |
| 12:57.25 | xochilpili | Samot, not like that: Modem -> FortiGate -> switch -> SPA3102 |
| 12:57.39 | *** join/#asterisk retentiveboy (~retentive@2601:cf:4400:d8e4:b1df:ea85:cb75:3a6c) |
| 12:57.41 | Samot | So why is it in Bridge mode? |
| 12:57.50 | Samot | Bridge mode is for the router side of the 3102 |
| 12:58.10 | Samot | You probably have it configured wrong. |
| 12:58.27 | Samot | Why it would take down the whole network... |
| 12:58.29 | Samot | Don't know. |
| 12:58.33 | Samot | Never seen that happen. |
| 12:58.35 | xochilpili | using just LAN i cant make outgoing calls |
| 12:59.02 | Samot | Factory default it |
| 12:59.05 | Samot | Start over. |
| 12:59.14 | xochilpili | i did it 3 times :D |
| 12:59.44 | Samot | There is no reason the SPA3102 should take your network down when it drops a calls |
| 12:59.48 | Samot | Or anything. |
| 12:59.57 | Samot | So it's either your SPA3102 or your network. |
| 13:00.39 | xochilpili | Samot, just disconnects from internet, i have network, then public ip changes and for almost 4-5 secs i have no internet |
| 13:00.52 | Samot | OK |
| 13:00.59 | Samot | This is not an Asterisk issue. |
| 13:01.07 | Samot | This is an issue with your SPA3102 and/or your network |
| 13:01.14 | Samot | Including your modem and DSL connection. |
| 13:01.27 | Samot | Nothing should cause your DSL to drop the connection. |
| 13:01.39 | Samot | In regards to a call over the SPA3102. |
| 13:01.58 | xochilpili | oks |
| 13:02.00 | xochilpili | thanks |
| 13:02.04 | Samot | Your PPPoE connection for your DSL is dropping. |
| 13:02.12 | Samot | It's why you're getting a new IP for the WAN |
| 13:02.37 | Samot | Again, the SPA3102 or network issue. |
| 13:02.38 | [TK]D-Fender | <xochilpili> Samot, not like that: Modem -> FortiGate -> switch -> SPA3102 <---- are there other devices plugged into the OTHER side of the SPA? |
| 13:02.58 | xochilpili | yes, i know that, but it's in the fortigate that PPPoE |
| 13:03.09 | [TK]D-Fender | <xochilpili> Samot, not like that: Modem -> FortiGate -> switch -> SPA3102 ::: -> other stuff? |
| 13:03.22 | Samot | Also, there should be no reason the SPA3102 should be in "Bridge" mode.. |
| 13:03.24 | xochilpili | [TK]D-Fender, just the phone (wireless) |
| 13:03.35 | [TK]D-Fender | then son't use Bridge |
| 13:03.36 | Samot | Is it plugged into the FXS port? |
| 13:03.38 | [TK]D-Fender | don't |
| 13:03.45 | Samot | Or the LAN port? |
| 13:03.52 | Samot | What is plugged into the LAN port? |
| 13:03.58 | [TK]D-Fender | I was asking about the LAN network jack |
| 13:04.07 | Samot | Because there shouldn't be anything in it |
| 13:04.09 | Samot | Really. |
| 13:04.21 | xochilpili | ah, no in the LAN there's nothing, just in the WAN -> to the switch |
| 13:04.33 | Samot | Then there is no need for this thing to be in Bridge mode. |
| 13:04.37 | Samot | Factory default |
| 13:04.39 | Samot | Restart |
| 13:04.47 | Samot | Don't mess with the Router/WAN/LAN stuff |
| 13:04.55 | Samot | It's not something that needs to be messed with |
| 13:05.03 | xochilpili | ok, will try again |
| 13:05.06 | xochilpili | thanks! |
| 13:05.11 | *** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-vopuqtufgoovumzo) |
| 13:06.58 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 13:21.01 | skrusty | anyone know why, using realtime in pjsip, i have no transports? Enabled transport-udp-nat in pjsip.conf, but not showing up in console :/ |
| 13:22.23 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 13:22.40 | skrusty | found it! |
| 13:22.51 | skrusty | was an issue further up in the config :/ |
| 13:24.31 | *** join/#asterisk xochilpili (~xochilpil@unaffiliated/xochilpili) |
| 13:29.53 | *** join/#asterisk xochilpili (~xochilpil@unaffiliated/xochilpili) |
| 13:30.04 | xochilpili | i have use this spa config: https://wiki.freepbx.org/pages/viewpage.action?pageId=55476525 |
| 13:30.08 | xochilpili | im not using freepbx, but still... |
| 13:30.13 | xochilpili | i did not receive the incomming calls or outgoing calls from asterisk, but when i do a outgoing call from my phone house, i got disconnected from PPPoE |
| 13:32.07 | *** join/#asterisk skywayskase (~skywayska@163.182.162.226) |
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| 13:52.30 | *** join/#asterisk captain118 (uid167508@gateway/web/irccloud.com/x-emyefxasgdpvhcds) |
| 13:55.22 | *** join/#asterisk xochilpili (~xochilpil@unaffiliated/xochilpili) |
| 13:56.31 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-yevnssfyerubkdah) |
| 13:56.31 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 13:59.41 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 13:59.41 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 14:05.18 | *** join/#asterisk kharwell (kharwell@nat/digium/x-aidqzznnrcmnunne) |
| 14:05.18 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:16.55 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 14:24.43 | Dovid | What would cause this issue? The box has limited memory but it looks OK |
| 14:24.44 | Dovid | [2017-06-15 14:17:48] WARNING[17220]: res_musiconhold.c:644 spawn_mp3: Fork failed: Cannot allocate memory |
| 14:24.44 | Dovid | [2017-06-15 14:17:48] WARNING[17220]: res_musiconhold.c:703 monmp3thread: Unable to spawn mp3player |
| 14:30.14 | [TK]D-Fender | That looks very "not OK" to me... |
| 14:40.33 | tcpdump | I am running the asterisk vm - is there a way to rerun the firwall config wizard from the first login, if you happened to miss it? |
| 14:40.34 | Samot | Cannot allocate memory <-- Generally means you is outta memory |
| 14:40.53 | Samot | What firewall config? |
| 14:41.01 | Samot | Asterisk doesn't have a firewall. |
| 14:41.29 | Samot | Linux has iptables. |
| 14:41.35 | cresl1n | tcpdump: usually the operating system has a firewall builtin |
| 14:41.58 | cresl1n | It's a separate layer from Asterisk |
| 14:42.01 | tcpdump | cresl1n: so do you recommend using the os firewall and not using the responsive firewall included with the vm? |
| 14:42.01 | Samot | People use iptables for their firewall. |
| 14:42.13 | cresl1n | tcpdump: I always use iptables |
| 14:42.13 | Samot | tcpdump: What OS? |
| 14:42.27 | cresl1n | tcpdump: I don't know what responsive firewall you're referring to |
| 14:42.30 | tcpdump | Samot: the Astrisk offical VM |
| 14:42.39 | Samot | There is not Asterisk Official VM |
| 14:42.43 | Samot | What are you talking about? |
| 14:42.53 | cresl1n | tcpdump: links or it didn't happen :-P |
| 14:42.56 | tcpdump | http://www.asterisk.org/downloads/asterisknow |
| 14:42.58 | *** join/#asterisk niko1990 (~sascha@p5B33E0E9.dip0.t-ipconnect.de) |
| 14:42.59 | Samot | OK |
| 14:43.02 | *** part/#asterisk niko1990 (~sascha@p5B33E0E9.dip0.t-ipconnect.de) |
| 14:43.09 | Samot | That is a GUI distro. |
| 14:43.10 | cresl1n | tcpdump: ahhhhhhhhhh |
| 14:43.20 | Samot | Which is maintained by FreePBX now I believe. |
| 14:43.24 | Samot | Since it IS FreePBX |
| 14:43.32 | *** join/#asterisk niko1990 (~sascha@p5B33E0E9.dip0.t-ipconnect.de) |
| 14:43.37 | tcpdump | yea, looks like it has freepbx built on other things. |
| 14:43.41 | niko1990 | Hello everyone = |
| 14:43.43 | niko1990 | =) |
| 14:43.52 | Samot | tcpdump: AsteriskNOW = FreePBX |
| 14:43.59 | Samot | Maintained by the same people. |
| 14:44.00 | Samot | Same ISO |
| 14:44.07 | Samot | Different branding |
| 14:44.19 | tcpdump | Ah, i see Samot . |
| 14:44.26 | Samot | It's kind of a legacy thing now. |
| 14:44.50 | Samot | And since it is a GUI |
| 14:44.58 | Samot | It's not really supported here. |
| 14:45.13 | Samot | Because FreePBX *owns* the box. |
| 14:45.22 | Samot | Preconfigured dialplan generation.. |
| 14:45.36 | Samot | configuration files are structured differently |
| 14:45.41 | Samot | Modified differently.. |
| 14:46.09 | TandyUK | hey guys, looking for an ATA type device i think.... We installed Draytek 2860 Voip routers on our sites, btu these dont behave how the customer wants |
| 14:46.20 | Samot | Asterisk is like making a meal from scratch. |
| 14:46.28 | Samot | FreePBX is Stouffers. |
| 14:46.29 | TandyUK | these allow the router to regster a sip account, and connect 2 analog phones to the router |
| 14:46.31 | Samot | Already made |
| 14:46.33 | Samot | Already to go |
| 14:46.41 | Samot | Pop seal and use. |
| 14:46.47 | TandyUK | incoming voip call goes to analog phone, incoming PSTN call goes to anaog phone |
| 14:46.55 | TandyUK | behind these routers we have a bunch of IP phones |
| 14:47.10 | TandyUK | I need a way to take the customers incoming PSTN line, and present this to the sip phones |
| 14:47.18 | Samot | Yeah.. |
| 14:47.20 | Samot | You need a PBX |
| 14:47.28 | Samot | Because the Draytek 2860 is not that |
| 14:47.36 | TandyUK | i had assumed we just log the sip phone onto the router, and it acts as sip proxy for the sip accounts, and like an ata for the analog |
| 14:47.52 | Samot | No |
| 14:47.53 | TandyUK | so eg, if internet goes down, and you dial out using the ip phone, this falls back to using the pstn |
| 14:48.01 | Samot | The Draytek is an FXS device. |
| 14:48.02 | TandyUK | yeah im only finding this out now |
| 14:48.04 | Samot | Not an FXO device. |
| 14:48.12 | Samot | The documents are pretty clear. |
| 14:48.20 | tcpdump | Samot: that makes sense. i think FreePBX is my best bet til I learn this. Even if I dont use it as originally stated this AM, I still want to understand how it works. |
| 14:48.27 | TandyUK | theyre clear as mud tbf, even their knowledgebase doesnt make it very clear |
| 14:48.56 | TandyUK | im on the ohone with their tech support atm, who also seem pretty unaware that how I thought it worked is not how it actually works |
| 14:49.01 | Samot | TandyUK: It's a router with an ATA |
| 14:49.08 | Samot | A FXS ATA |
| 14:49.12 | Samot | Two FXS ports |
| 14:49.15 | Samot | That's pretty clear. |
| 14:49.22 | TandyUK | yes, but it must have some FXO support, as it takes ana alalog phone line in |
| 14:49.41 | cresl1n | no |
| 14:49.43 | TandyUK | eg, it adds cal lwaiting support for the analog phones |
| 14:49.47 | cresl1n | oh |
| 14:50.01 | TandyUK | if im on a voip call, and the analog line rings, it call-waits it for me |
| 14:50.03 | Samot | Two 'FXS' Phone Ports |
| 14:50.12 | TandyUK | it cant do that if nits not able to pick up the incoming call and handle it |
| 14:50.30 | Samot | It's an ATA |
| 14:50.37 | Samot | With some vertical features |
| 14:50.40 | TandyUK | Samot: i get the 2 fxs ports for analog phones |
| 14:50.42 | Samot | It does not do FXO |
| 14:50.48 | Samot | FXS is for phones. |
| 14:50.54 | TandyUK | what exactly is the PSTN line port connected to then? |
| 14:51.03 | TandyUK | thats FXO not FXS |
| 14:51.04 | Samot | It DOESN'T |
| 14:51.26 | Samot | The Draytek would connect to a PBX or VoIP server.. |
| 14:51.26 | TandyUK | in the front of the router there are 2/3 ports. |
| 14:51.30 | TandyUK | 1 PSTN LINE port |
| 14:51.46 | TandyUK | 1 Phones port (with a rj11 to 2x bt adaptor) for the 2 analog phones |
| 14:51.59 | niko1990 | I'm nearly brand new to asterisk. I have a question: I'm developing right now my own doorCom (Raspberry Pi). What I need: Asterisk running in the background as a service, asterisk should be connected as a sip extension on my fritzbox, and I need asterisk to place calls over the command line (bash scripts - something like "asterisk dial NUMBER"). Is this possible? (Later on i would like to use some more functions of asterisk (that i already |
| 14:52.07 | Samot | OK |
| 14:52.19 | TandyUK | and completely seperate to that, is the ADSL/VDSL rj11 port for wan |
| 14:52.38 | Samot | so you need to connect it ot a PBX |
| 14:52.47 | Samot | with a SIP account. |
| 14:52.55 | Samot | Like I said, it's in the docs. |
| 14:52.56 | Samot | I missed it |
| 14:52.59 | Samot | Now I see it |
| 14:53.02 | TandyUK | it is connected to a pbx, we have 2 sip accounts logged in on the router |
| 14:53.16 | TandyUK | these can ring either of the analog phones |
| 14:53.22 | TandyUK | but not an IP phone behind it |
| 14:53.24 | Samot | Where are the SIP phones connected to? |
| 14:53.42 | TandyUK | theyre plugged into the LAN ports, as for the sip registration, thats where im stuck |
| 14:53.47 | Samot | Right |
| 14:53.55 | Samot | There needs to be a PBX for THOSE phones |
| 14:54.02 | TandyUK | i would expect the sip phones to register with the router |
| 14:54.06 | Samot | Then the FXO connects to THAT PBX |
| 14:54.10 | Samot | No. |
| 14:54.10 | [TK]D-Fender | TandyUK> but not an IP phone behind it <- BECAUSE IT ISN'T ACTING LIKE A pbx |
| 14:54.13 | TandyUK | so when a PSTN call comes in, it forwards to the IP phones |
| 14:54.35 | TandyUK | exactly, so back to what i first asked: |
| 14:54.36 | [TK]D-Fender | TandyUK, the line port there is DUMB. Very little logic in its processing |
| 14:54.43 | [TK]D-Fender | it switches the LOCAL ports only |
| 14:54.46 | TandyUK | 15:46] <TandyUK> hey guys, looking for an ATA type device i think.... |
| 14:54.48 | Samot | This then gives the telephones access to your analogue line to allow you to make calls as well as your VoIP facility (you can select the PSTN line instead of VoIP by dialling #0) |
| 14:54.57 | [TK]D-Fender | TandyUK, that IS an ATA |
| 14:55.10 | [TK]D-Fender | TandyUK, it is NOT an FXO-> SIp gateway |
| 14:55.27 | TandyUK | ok, so an FXO>SIP gateway is what i need then |
| 14:55.31 | [TK]D-Fender | tha has a DUMB SWITCH between the PSTN port & the attached phones |
| 14:55.57 | [TK]D-Fender | " |
| 14:55.57 | [TK]D-Fender | Automatic phone switch-over for incoming calls on either PSTN or VoIP" |
| 14:56.00 | [TK]D-Fender | http://www.draytek.co.uk/products/business/vigor-2860#voip |
| 14:56.01 | Samot | Where are the local IP phones going to REGISTER? |
| 14:56.15 | TandyUK | Samot: with the gateway |
| 14:56.28 | Samot | So you need a gateway that supports multiple SIP accounts |
| 14:56.32 | TandyUK | which according to its dialplan will then direct calls either over an upstream sip account, or out over PSTN |
| 14:56.55 | TandyUK | right, so what hardware exists that can do that? |
| 14:57.03 | Samot | Well.. |
| 14:57.16 | Samot | Normal people would get a FXO to SIP device and register it to a PBX |
| 14:57.21 | Samot | Then register the phones to the PBX |
| 14:57.28 | Samot | Since the FXO is the PSTN gateway |
| 14:57.32 | TandyUK | yeah problem here is the customers PBX is in office #1 |
| 14:57.48 | TandyUK | in office #3, they have sip phones which register ove vpn to that pbx, plus analog phone lines |
| 14:58.06 | [TK]D-Fender | And the problem is ...? |
| 14:58.07 | Samot | Did you give them this solution? |
| 14:58.10 | TandyUK | the point of this is if internet goes down, sip phones can still make/recieve calls fro mthe analog line |
| 14:58.18 | TandyUK | no internet == no vpn == no sip server |
| 14:58.29 | TandyUK | no i didnt lol |
| 14:58.32 | [TK]D-Fender | those phone will have to have the BRAINS to use the local device |
| 14:58.33 | [TK]D-Fender | ^ |
| 14:58.42 | [TK]D-Fender | They need to be configured to fail over |
| 14:58.56 | Samot | So what are you trying to do then? |
| 14:58.58 | [TK]D-Fender | And the device there needs enough brains to accept & process those calls |
| 14:59.12 | TandyUK | Samot: eset this up |
| 14:59.13 | [TK]D-Fender | this is the point where you start putting PBX's at EACH location |
| 14:59.31 | TandyUK | theyve asked me to replace their existing siemens phones (which have sip AND pstn ports |
| 14:59.40 | TandyUK | with modern sip phones (which dont have the pstn part) |
| 15:00.25 | Samot | So they want POTS at each location for backup? |
| 15:00.27 | TandyUK | siemens n300ip i think it is, discontinued, phones are badly wearing out, they just want a modern replacement for 10 year old phones |
| 15:00.32 | TandyUK | exactly |
| 15:00.42 | Samot | 10:59:24 AMÂ <[TK]D-Fender>Â this is the point where you start putting PBX's at EACH location |
| 15:01.00 | TandyUK | so there are NO phones on the market now with these features? |
| 15:01.18 | [TK]D-Fender | <TandyUK> siemens n300ip i think it is, discontinued, phones are badly wearing out, they just want a modern replacement for 10 year old phones <- those ancient ideas of putting analog lines directly in the phones and that kind of fail-over is DEAD |
| 15:01.19 | Samot | Well you can get an FXO gateway for each location.. |
| 15:01.19 | lvlinux | Gigaset cordless...lol |
| 15:01.24 | [TK]D-Fender | Modern things don't think that way |
| 15:01.31 | [TK]D-Fender | you have to plan for something else |
| 15:01.36 | TandyUK | n300 is the wrong number, thats completely different |
| 15:01.36 | Samot | The phones have to be smart enough to failover to it |
| 15:01.57 | Samot | And then that gateway has to support a SIP account for EACH phone. |
| 15:02.28 | Samot | But then they have no voicemail |
| 15:02.30 | Samot | No IVR |
| 15:02.33 | Samot | No anything |
| 15:02.41 | Samot | They have a dumb gateway that sends calls out |
| 15:02.45 | Samot | and takes calls in. |
| 15:03.03 | Samot | Their entire call flow structure is gone. |
| 15:03.47 | Samot | If there *needs* to be a central PBX |
| 15:03.53 | Samot | And they want failover.. |
| 15:04.14 | Samot | Get a second DSL line and have the Drayteks do WAN failover. |
| 15:08.01 | Samot | If it's your job to replace their entire voice system at all their locations... |
| 15:08.21 | Samot | Look at what they want and what the best options to provide that solution are. |
| 15:08.58 | TandyUK | Im looking at gigaset N300 IP atm, whihc seems to replicate exactly what they want |
| 15:09.11 | TandyUK | 4 dect handsets, 4 sip accounts, plus PSTN |
| 15:09.52 | Samot | How many phones are at each location? |
| 15:10.44 | Samot | And are these desk phones? |
| 15:10.51 | Samot | That they currently have? |
| 15:12.58 | TandyUK | 2-3 handsets per location |
| 15:13.04 | TandyUK | all dect portables |
| 15:13.17 | TandyUK | Id ordered yealink W52's to replace them with |
| 15:13.27 | TandyUK | but then oviously hit the lack of pstn snag |
| 15:13.29 | *** join/#asterisk Kunsi (felix@unaffiliated/kunsi) |
| 15:13.46 | TandyUK | they have this setup on 7 different sites |
| 15:14.11 | TandyUK | fortunately we were only doing 2 sites to begin with! |
| 15:15.01 | Samot | But they have a central PBX? |
| 15:15.55 | TandyUK | at their head office, yes |
| 15:16.13 | TandyUK | each site has several sip extensions, with PSTN fallback |
| 15:16.28 | TandyUK | these numbers are published too, so most incoming calls come over the pstn |
| 15:16.33 | TandyUK | its only internal stuff that uses sip |
| 15:16.33 | Samot | So.. |
| 15:17.04 | Samot | You're going to register each extension with the PBX as a "SIP provider" on this DECT system.. |
| 15:17.16 | Samot | And then each phone registers to the DECT system.. |
| 15:17.19 | TandyUK | ok stop, lets start again |
| 15:17.29 | TandyUK | head office: PBX with outgoing trunks |
| 15:17.46 | TandyUK | shop #1: SIP Extensions 201, 202, 203 (and 200 is a hunt group for the shop) |
| 15:17.50 | [TK]D-Fender | so far your incoming processing seems to have no brain |
| 15:17.59 | TandyUK | shop #1 PSTN: 01903123456 |
| 15:18.08 | TandyUK | incoming PSTN calls ring to all 3 handsets at that sho |
| 15:18.15 | TandyUK | there is no logic to it, I agree |
| 15:18.29 | TandyUK | Ive been trying to sell this customer a hosted pbx for years withotu any sucess |
| 15:18.32 | [TK]D-Fender | And hopefully it' |
| 15:18.35 | Samot | OK |
| 15:18.37 | TandyUK | theyre too fixed in the analog world |
| 15:18.43 | [TK]D-Fender | And hopefully it's smart enough in ringing those DECT devices attached |
| 15:18.48 | Samot | So how does Shop #1 call Shop #2? |
| 15:18.54 | TandyUK | yes, the gigaset N300 IP does this |
| 15:18.55 | [TK]D-Fender | This plan is low-featured crap |
| 15:19.04 | Samot | So how does Shop #1 call Shop #2? |
| 15:19.08 | [TK]D-Fender | no voicemail. No IVR, not corporate messaging, nothing |
| 15:19.12 | TandyUK | internally, on ext 301 |
| 15:19.16 | Samot | OK |
| 15:19.17 | TandyUK | no ntohing whatsoever |
| 15:19.20 | Samot | So stop. |
| 15:19.22 | [TK]D-Fender | This makes it look like a Mickey Mouse operation |
| 15:19.37 | Samot | How does the DECT connect to the central PBX |
| 15:19.42 | [TK]D-Fender | Actually.... Disney HAS a huge phone system |
| 15:19.47 | TandyUK | SIP registartion via their internal VPN |
| 15:19.48 | Samot | So it can route that call from 301 to201? |
| 15:19.57 | Samot | So like I said.. |
| 15:19.58 | [TK]D-Fender | Some cheap Chinese Mickey Mouse rip-off operation :p |
| 15:20.07 | TandyUK | all the sip logins go over their vpn, and is purely for internal stuff |
| 15:20.10 | Samot | The DECT needs a "SIP provider" account for each extensions |
| 15:20.15 | Samot | That needs to route back to the handset. |
| 15:20.35 | TandyUK | yes it has lol |
| 15:20.43 | TandyUK | im not sure whats unclear here tbh |
| 15:20.49 | Samot | I understand that |
| 15:20.52 | TandyUK | central PBX has 1 sip extension per dect handset |
| 15:21.01 | Samot | HANDSET |
| 15:21.14 | Samot | But the DECT base is IN BETWEEN |
| 15:21.20 | TandyUK | yes, so the eg, N300ip base station registeres with 3 sip accounts, 1 per handset |
| 15:21.23 | Samot | The handset does not register with the PBX |
| 15:21.26 | TandyUK | correct |
| 15:21.28 | Samot | It registers with the n300 |
| 15:21.34 | Samot | So the n300 needs to register with the PBX |
| 15:21.42 | TandyUK | like i just said lol |
| 15:21.44 | Samot | Via it's "SIP Provider" accounts. |
| 15:21.55 | Samot | That you have to map back to the DECT devices |
| 15:22.11 | Samot | And hope your DECT base understands when the RG is hitting it |
| 15:22.11 | TandyUK | is waitign to see a point to this convo tbh |
| 15:22.17 | *** join/#asterisk Kunsi (franzi@unaffiliated/kunsi) |
| 15:22.28 | TandyUK | the ring group, eg "200" just rings 201,202,203 in unison |
| 15:22.51 | TandyUK | theres nothing for the base station to understand |
| 15:22.58 | TandyUK | it jus thas 3 calls coming in over sip |
| 15:23.07 | Samot | Except for it's ring time |
| 15:23.13 | Samot | How it handles CW |
| 15:23.17 | Samot | VM |
| 15:23.23 | TandyUK | there IS NO VM |
| 15:23.28 | TandyUK | there IS NO CW |
| 15:23.29 | Samot | OK. |
| 15:23.53 | TandyUK | theres not even a "fuck off we're shut" message out of hours |
| 15:23.59 | [TK]D-Fender | In this wonderful modern age this solution sounds like shit. But they are entirely welcome to it... |
| 15:24.12 | [TK]D-Fender | Have fun... |
| 15:27.04 | *** part/#asterisk niko1990 (~sascha@p5B33E0E9.dip0.t-ipconnect.de) |
| 15:28.42 | Samot | 11:18:48 AMÂ <TandyUK>Â theyre too fixed in the analog world << That's crap. I convert hotels from analog to IP all the time. |
| 15:29.10 | Samot | They are AMAZED at what can be done on IP vs analog that IMPROVES their business. |
| 15:29.41 | Samot | They where "fixed" in the analog world for the past 25+ years. |
| 15:30.13 | Samot | Just did an auto shop |
| 15:30.26 | Samot | Again, they are amazed at what they can do now. |
| 15:31.02 | Samot | They can actually reach mechanics out on the other side of the yard because of softphones |
| 15:32.11 | Samot | Or transfer calls to the two trunk drivers without worrying about cell phone voicemails, etc. |
| 15:32.14 | Samot | tow |
| 15:32.26 | *** join/#asterisk malcolmd (malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:32.26 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 15:33.33 | *** join/#asterisk samwierema (~samwierem@095-097-255-066.static.chello.nl) |
| 15:33.34 | lvlinux | tow trunks? Is that a SIP in SIP tunnel? lol |
| 15:33.54 | Samot | Bria. |
| 15:34.16 | samwierema | How do I confirm that a module for Asterisk is available? If the status is "Running" and the module is loaded? |
| 15:34.30 | Samot | Running means it's running |
| 15:34.33 | Samot | therefore loaded |
| 15:34.41 | lvlinux | and therefore "available" |
| 15:35.04 | tcpdump | One more odd question - since asterisk doesnt support RTSP, but rather video of RTP, does the SIP channel handle all of the QoS things that RTSP would normally handle? |
| 15:35.20 | lvlinux | You mean RTCP? |
| 15:36.39 | Samot | Is this over the Internet? |
| 15:36.45 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 15:36.58 | Samot | If it is, QoS is not guaranteed. |
| 15:37.20 | samwierema | Ok, so I couldn't resolve this yesterday: the res_statsd.so is running, but, when I call the StatsD dialplan application I get "No application 'StatsD' for extension". Is there anything needed beyond loading the module to use this application? |
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| 15:38.47 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:45.56 | lvlinux | samwierema: https://wiki.asterisk.org/wiki/display/AST/Utilizing+the+StatsD+Dialplan+Application |
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| 17:40.18 | qakhan | hi all,is there any way to setup asterisk failover? |
| 17:45.38 | Samot | In what way? |
| 17:45.47 | Samot | I mean the answer is yes but it's not cut and dry. |
| 17:46.21 | Samot | Asterisk, itself, has no method for this. |
| 17:46.28 | Samot | So you have to implement a method. |
| 17:47.43 | qakhan | is there any best prectice. |
| 17:48.00 | Samot | It all depends on the method you want to do it in. |
| 17:48.08 | Samot | There there is the best practice for that method. |
| 17:48.34 | Samot | Are you talking an HA failover with a floating IP, a heartbeat, etc? |
| 17:48.54 | Samot | Or are you talking having two servers and the phones use failover options to register to the backup |
| 17:49.02 | qakhan | my main goal is if server-A goes down then Server-B take over |
| 17:49.11 | Samot | I get your goal. |
| 17:49.18 | Samot | But there are various ways to do it |
| 17:49.30 | Samot | How you want to do it is going to determine the steps you take. |
| 17:49.53 | Samot | How will you monitor A and B? |
| 17:50.05 | qakhan | ok, can you please share some examples |
| 17:50.06 | Samot | Will there be something to redirect the traffic? |
| 17:50.10 | Samot | I did |
| 17:50.34 | Samot | 1:48:45 PM <Samot> Are you talking an HA failover with a floating IP, a heartbeat, etc? |
| 17:50.34 | Samot | 1:49:04 PM <Samot> Or are you talking having two servers and the phones use failover options to register to the backup |
| 17:51.22 | Samot | Those are basically the two extreme sides of the spectrum. |
| 17:51.37 | Samot | From the advanced to the very basic. |
| 17:51.44 | qakhan | HA failover with a floating IP, a heartbeat, will be over the OS (Linux) |
| 17:52.04 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
| 17:52.10 | Samot | So you have this already? |
| 17:52.43 | qakhan | does asterisk replicate its config with other server? |
| 17:52.57 | qakhan | no i dont have HA yet |
| 17:53.00 | Samot | You have to tell it to |
| 17:53.06 | Samot | rsync or something |
| 17:54.15 | Samot | Asterisk is just a service running on the server. |
| 17:54.17 | Samot | Like Apache |
| 17:54.19 | Samot | MySQL |
| 17:54.54 | Samot | How you replicate the data from Server A to Server B and vice versa is dealers choice. |
| 17:55.18 | qakhan | ok Thanks, |
| 17:55.32 | *** join/#asterisk skywayskase (~skywayska@163.182.162.226) |
| 17:55.49 | qakhan | but if servers swicthes current calls will be drop |
| 17:56.00 | Samot | Yes. |
| 17:56.15 | qakhan | can we avoid it? |
| 17:56.38 | Samot | Well how is the other Asterisk server know there is a call there? |
| 17:56.55 | Samot | Can it be avoided? Yes. |
| 17:57.16 | qakhan | how |
| 17:57.22 | Samot | Dude. |
| 17:57.28 | drmessano | Not using Asterisk |
| 17:57.33 | Samot | ^^^ |
| 17:57.38 | Samot | One way |
| 17:57.38 | drmessano | Asterisk is a B2BUA, not a SIP proxy |
| 17:57.53 | Samot | There is going to be a lot of moving parts for this |
| 17:58.08 | Samot | A SIP Proxy/SBC is gong to be one of them. |
| 17:58.37 | drmessano | In which case, Asterisk wont be handling the call |
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| 18:00.54 | Samot | Well I said it's possible to avoid the call dropping |
| 18:01.08 | Samot | I didn't say it was going to be a seamless transition.. |
| 18:01.49 | lvlinux | qakhan: Kamailio is very commonly used for this, as a SIP server in front of Asterisk. |
| 18:02.03 | Samot | It's not going to avoid the call dropping. |
| 18:03.15 | Samot | There's going to be a delay in the failover.. |
| 18:03.19 | Samot | Just accept it. |
| 18:03.38 | Samot | But at least if PBX A craps itself, you have PBX B |
| 18:09.08 | lvlinux | The call won't drop if the RTP media is direct will it? |
| 18:10.08 | Samot | Signaling. |
| 18:10.11 | Samot | So yes. |
| 18:10.17 | *** join/#asterisk retentiveboy_ (~retentive@c-73-82-30-193.hsd1.ga.comcast.net) |
| 18:10.22 | Samot | No signaling not way to send RTP |
| 18:10.31 | Samot | s/not/no/ |
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| 18:26.13 | lvlinux | I mean after RTP is already flowing between endpoints. |
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| 18:27.52 | lvlinux | Wouldn't the call continue at least until another SIP message was sent, like a re-invite or BYE etc. |
| 18:27.57 | lvlinux | ? |
| 18:29.48 | drmessano | Have you tried direct media on endpoints? |
| 18:30.36 | drmessano | Works great internally |
| 18:30.42 | drmessano | Doesnt externally |
| 18:31.42 | Samot | lvlinux: My point is there would need to be a SBC between the endpoints and the Asterisk server. |
| 18:31.56 | Samot | In which the endpoints know nothing about Asterisk. |
| 18:32.10 | Samot | And the SBC is sending calls to the needed Asterisk server for whatever reason.. |
| 18:33.45 | Samot | And even if the floating IP sent the RTP over to PBX B, it doesn't have any active channels.. |
| 18:33.52 | Samot | It's not using the RTP ports at the moment... |
| 18:34.02 | Samot | How is it going to know what to do with the audio? |
| 18:34.43 | Samot | What about that poor dude in the middle of the IVR tree |
| 18:34.52 | Samot | Is he going to be pushed back right into the same spot? |
| 18:36.14 | drmessano | If you want Box B in case Box A goes down, super easy to do |
| 18:36.24 | drmessano | if you want "Calls never dropping", plan to spend money |
| 18:36.32 | Samot | A lot. |
| 18:36.38 | drmessano | Most people dont need that level of redundancy |
| 18:36.49 | drmessano | If they do, theyre not on IRC looking for it |
| 18:36.55 | Samot | And just wave bye to your Asterisk servers.. |
| 18:37.08 | Samot | Because any solution isn't going to be needing them. |
| 18:37.12 | Samot | At that level. |
| 18:37.54 | Samot | A call is not two endpoints aka phones. |
| 18:38.03 | Samot | A call is between an endpoint and Asterisk. |
| 18:38.31 | Samot | If PBX A craps out.. |
| 18:38.57 | Samot | How are you going to move 5 users from a conf bridge over to PBX B on the fly without dropping anything? |
| 18:39.53 | drmessano | Doesnt PJSIP do that with the never_drop_calls_when_HA option? |
| 18:40.17 | Samot | Well really.. |
| 18:40.24 | Samot | At this level it's not even about Asterisk.. |
| 18:40.32 | Samot | It's about logic of how things would be done. |
| 18:40.44 | Samot | Like I said, the guy in the middle of the IVR |
| 18:40.58 | Samot | Will the data he's already entered be known on the new PBX |
| 18:41.03 | drmessano | The context of the question was Asterisk |
| 18:41.03 | Samot | Will it know what spot he was in? |
| 18:41.07 | drmessano | So the short answer is |
| 18:41.08 | drmessano | No |
| 18:41.08 | Samot | OK |
| 18:41.13 | Samot | So with Asterisk.. |
| 18:41.16 | Samot | Right |
| 18:41.20 | Samot | So calls are going to drop. |
| 18:41.52 | Samot | The point of HA is that when A craps, 30 seconds later B can take over. |
| 18:41.58 | Samot | Or whatever your timeframe is.. |
| 18:42.06 | Samot | So when they DO call back, you're getting the calls. |
| 18:42.13 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-170.cust.bezeqint.net) |
| 18:42.20 | Samot | Instead of forwarding them to cell phones because you're down. |
| 18:43.26 | *** join/#asterisk showitmedia (026bd9dd@gateway/web/freenode/ip.2.107.217.221) |
| 18:43.32 | Samot | You need to make a list |
| 18:43.46 | Samot | And break it down with "What I REALLY REALLY WANT" |
| 18:44.02 | Samot | And "What I Would Really Like" |
| 18:44.13 | Samot | And "What I Actually Need" |
| 18:45.07 | Samot | Because this is something that people either A) Underbuild or B) Overkill build. |
| 18:45.40 | showitmedia | Hi, I have a problem with phones dropping of an Asterisk 13 they keep changing state from unreachable to reachable and back in random order and constantly. My gues is some sort of session timeout or NAT issue, anyone have any Ideas? |
| 18:45.52 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 18:45.54 | Samot | Are they remote phones to the PBX? |
| 18:46.11 | Samot | Like over the Internet type connection? |
| 18:46.20 | showitmedia | the PBX is on the internet and the phones are on a local network |
| 18:46.26 | Samot | So yes. |
| 18:46.29 | Samot | Then, NAT. |
| 18:47.03 | showitmedia | That was my conclusion aswell, any Idea how to fix it? |
| 18:47.20 | drmessano | What router are you using? |
| 18:47.38 | showitmedia | I have the same setup running in 2 locations one works fine the other one keeps doing this ping pong |
| 18:47.42 | drmessano | What router are you using? |
| 18:48.21 | showitmedia | the location with issues have a Zyxel USG 100 between them and the internet |
| 18:48.50 | drmessano | That would do it |
| 18:48.52 | showitmedia | I have disabled the ALG SIP function as I know the Cisco equvalant can cause trouble |
| 18:49.07 | drmessano | What do you have in front of the Asterisk box? |
| 18:49.18 | showitmedia | iptables |
| 18:49.22 | showitmedia | :-) |
| 18:49.25 | Samot | So it's public Internet? |
| 18:49.32 | showitmedia | yes |
| 18:49.48 | Samot | Zyxel. |
| 18:49.54 | drmessano | The Zyxel is the problem |
| 18:49.55 | Samot | Those are the bad guys here. |
| 18:50.05 | drmessano | Ditch it |
| 18:50.13 | drmessano | Get a Mikrotik for 1/10 the price |
| 18:50.36 | showitmedia | that was also my toughts. but I'm no Zyxel expert and I don't recognise the interface on the box. |
| 18:50.38 | Samot | Shill. |
| 18:50.47 | drmessano | RB750Gr3 |
| 18:50.48 | Samot | But seriously.. |
| 18:50.50 | drmessano | ^ $55 |
| 18:50.50 | Samot | Get that. |
| 18:51.25 | showitmedia | The system goes live after midnight, so you guys are my last resort. |
| 18:51.35 | drmessano | We just told you what to do |
| 18:51.43 | drmessano | Get a Mikrotik for 1/10 the price |
| 18:51.44 | drmessano | RB750Gr3 |
| 18:51.46 | drmessano | ^ $55 |
| 18:51.52 | drmessano | Scrap the Zyxel |
| 18:52.02 | showitmedia | I've tried beating it but even creating a simple firwall rule on the box is hard work |
| 18:52.18 | drmessano | Its literally the problem |
| 18:52.24 | drmessano | Cant help you any further |
| 18:54.05 | showitmedia | wish I could scrap it. right now. Been trying to see if there was some sort of session timeout on the thing |
| 19:02.11 | drmessano | Its a ZyXel |
| 19:02.21 | drmessano | They generally suck with SIP |
| 19:02.37 | drmessano | Nothing to fix |
| 19:52.36 | salviadud | I want to know how to solve the dependency on compiling asterisk with imap |
| 19:52.47 | salviadud | On centos specifically |
| 19:53.20 | salviadud | I can't highlight the option when I do make menuselect |
| 19:55.53 | salviadud | Well, I just read I need some pakcages, forgetaboutit |
| 19:57.01 | drmessano | Install the needed packages and you wont need them |
| 19:57.06 | drmessano | Good job |
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