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| 02:46.07 | averythomas | Going to use a mac mini as a server, anyone else done that? |
| 02:48.30 | [TK]D-Fender | I'm sure someone has |
| 02:48.45 | [TK]D-Fender | ~polls |
| 02:48.45 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 02:52.42 | MaliutaLap | I prefer the Morris Minor over the Mini :P |
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| 03:39.17 | phix | averythomas: You installing Linux onto it? |
| 03:40.24 | phix | MaliutaLap: Yeah, better than those shitty feit's |
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| 04:04.33 | averythomas | phix: only OSX |
| 04:15.58 | phix | hmmmm |
| 04:16.38 | phix | I guess you can do that |
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| 07:08.00 | lkthomas | polycom phone is so stupid |
| 07:08.05 | lkthomas | the Digimaps is crazy |
| 07:08.16 | eirirs | all hardwares are stupid, they just do what you tell them to do |
| 07:08.47 | lkthomas | I don't understand why Cisco phone could dial 9 first, then keep punch phone numbers |
| 07:08.58 | lkthomas | while Polycom phone, you need to type all phone numbers and press send |
| 07:10.48 | lkthomas | http://apetec.com/voip/Polycom-DialPlan.htm |
| 07:10.54 | lkthomas | they think they are smart |
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| 07:22.09 | ChannelZ | that's not a lot different than Cisco dialplans |
| 07:26.08 | lkthomas | ChannelZ, I never setup Cisco dialplans, never needed |
| 07:26.21 | lkthomas | ChannelZ, I would just pickup the phone and keep punching numbers |
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| 07:29.36 | ChannelZ | Well, that the defaults are moderately useful or not I guess is your complaint |
| 07:29.48 | lkthomas | LOL |
| 07:29.54 | lkthomas | maybe ? |
| 07:34.03 | ChannelZ | dunno, not entirely sure what you are wanting to happen |
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| 11:40.54 | skrusty | does anyone know what the valid hangup reasons are for ARI? |
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| 12:00.09 | WIMPy | Q.850, as usual? |
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| 12:12.23 | skrusty | WIMPy: think i just got confused as it took a string, but i guess it's just the cause code as a string |
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| 13:08.45 | Gugge | What is a good/cheap pcie card just for use as a timing source? |
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| 13:12.32 | WIMPy | Do you ask because you think you need some external timing source or because you're having issues with internal timing? |
| 13:13.01 | Gugge | because im having issues with both internal timing and dahdi's internal timing, and dahdi_dummy timing, when under high load |
| 13:13.11 | Gugge | And i would like to test if hardware timing makes a difference :) |
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| 13:13.24 | topriddy | hi guys |
| 13:13.55 | Gugge | WIMPy: im in no way sure it will make a difference ... but an easy way to find out is to try :) |
| 13:14.12 | topriddy | i want to know what iâll need to build a telephony solution like twilio/tropo to be used in-house if I dont want to subscribe to any of these |
| 13:14.25 | topriddy | basically, i would be building IVR solutions around it. |
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| 13:15.02 | topriddy | pointers as to what kind of hardware infrastructure, software & software config, and other setup would be great |
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| 13:19.20 | skrusty | WIMPy: fyi, seems not. the hangup causes in ari are it seems named (in the swagger spec) as "normal", "busy" and "congestion". |
| 13:19.46 | topriddy | skrusty: Gugge WIMPy anyone have clues on above? i am desperate |
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| 13:20.28 | skrusty | why not use a service like that, out of interest? |
| 13:20.37 | skrusty | creating your own is a lot of work, i know. i created one :) |
| 13:21.17 | topriddy | skrusty: i want to deploy it in a closed network and manage it all by myselfâ¦a security requirement. plus iâm hoping in comes cheaper |
| 13:21.40 | skrusty | i doubt it, there's no "off the shelf" product that is twilio/tropo |
| 13:21.55 | skrusty | they are comlpex hosted services, a mix of hardware and software |
| 13:22.08 | topriddy | skrusty: so what exactly would I need? |
| 13:22.31 | topriddy | skrusty: also, if you can have a look at FoneNode.com, its a small telephony solution and emphasises that it isnt exactly rocket science |
| 13:22.50 | mjordan | hm. |
| 13:23.25 | skrusty | topriddy: the question your asking, you're assuming anyone would just give that information away |
| 13:23.37 | WIMPy | Gugge: Well, if you have a digital line, connect tthat. Telcos usually provide very acurate timing. |
| 13:23.45 | topriddy | skrusty: sigh. is this a closed source community? I am confused here |
| 13:23.54 | skrusty | no, asterisk is open source |
| 13:23.56 | WIMPy | skrusty: Ah, so it's like the DIALSTATUS. |
| 13:24.02 | skrusty | WIMPy: yeah |
| 13:25.00 | mjordan | WIMPy: If you're looking for hangup cause, it is conveyed in a ChannelDestroyed event: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+REST+Data+Models#Asterisk13RESTDataModels-ChannelDestroyed |
| 13:25.05 | skrusty | topriddy: open source doesn't mean everyone else will do something for you |
| 13:25.15 | skrusty | mjordan: no it was me |
| 13:25.20 | mjordan | ah |
| 13:25.21 | skrusty | and i wanted to set the cause in ARI Hangup |
| 13:25.22 | mjordan | sorry :-P |
| 13:25.25 | mjordan | came in half way |
| 13:25.42 | topriddy | skrusty: i didnt ask you do something for me. i asked how I could achieve something similar. you act like this is rocket science. |
| 13:25.51 | skrusty | took me a while (i should have read the sweagger spec) that it's restriced to those values |
| 13:25.51 | mjordan | skrusty: you won't normally see that event unless you subscribe for the entire lifetime of the channel, either by subscribing to the channel directly or to the endpoint that the channel is related to |
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| 13:26.42 | skrusty | topriddy: i've spent the last year developing my soltion, im not about to just hand over that information - sorry! :) |
| 13:26.43 | mjordan | topriddy: your question is very broad and vague. "How do I go make tropo/twilio" is ... well. Sort of a long answer. And there isn't a "buy this HW, put these things in, start it up, and tada - you have your own API service". |
| 13:26.50 | Gugge | WIMPy: i dont have a digital line, and i dont have any pstn/isdn hardware :) |
| 13:27.13 | topriddy | mjordan: i would like to ignore the API parts, i dont need that |
| 13:27.18 | mjordan | er |
| 13:27.20 | skrusty | lol |
| 13:27.33 | skrusty | so you just want to write apps for telephony? |
| 13:27.34 | topriddy | mjordan: what i am askinig is what would i need to terminate a call, needed for an IVR system. |
| 13:27.38 | mjordan | then why would you use tropo or twilio as an example? |
| 13:27.42 | skrusty | *sigh* |
| 13:27.46 | skrusty | that's a very different question |
| 13:27.50 | mjordan | topriddy: Why can't Asterisk do that for you? |
| 13:27.57 | skrusty | Asterisk and a SIP provider might be a good start :) |
| 13:28.01 | WIMPy | Gugge: Then you have to rely on the Oscilator on that card. How ever goot that might be. |
| 13:28.19 | topriddy | thats the exact question i have been trying to ask all along. |
| 13:28.26 | skrusty | mjordan: is there any plan to allow better hangup causes in ARI Hangup (channel delete) |
| 13:28.39 | skrusty | i want to send 480 for example |
| 13:28.56 | Gugge | WIMPy: what card? (that was the original question) |
| 13:29.05 | mjordan | skrusty: that's a SIP cause, not an Asterisk cause. :-) |
| 13:29.08 | WIMPy | Gugge: ANY card. |
| 13:29.16 | topriddy | how to have a âserviceâ that can make phone calls, accept user input, play voice etc. also, what network places the call would I be paying any telecoms in the process or my internet data would do. |
| 13:29.20 | skrusty | yeah, but asterisk only allows (via ARI) busy, congestion and normal |
| 13:29.30 | skrusty | from what i can see anyway |
| 13:29.32 | Gugge | WIMPy: so all cards have a timing source? |
| 13:29.42 | topriddy | mjordan: please do you follow? |
| 13:30.01 | mjordan | skrusty: hm. |
| 13:30.06 | skrusty | he's answered your question topriddy |
| 13:30.09 | skrusty | asterisk does that |
| 13:30.09 | mjordan | skrusty: are you talking about the reason query parameter? |
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| 13:30.16 | skrusty | mjordan: yes |
| 13:30.32 | mjordan | no answer should be a 480 |
| 13:30.33 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Channels+REST+API#Asterisk13ChannelsRESTAPI-hangup |
| 13:30.35 | WIMPy | Gugge: They need to. On POTS you don't get timing. On ISDN you do, but (almost) all cards can be used without a line to only connect phones. So they still need to generate the timing themselves. |
| 13:30.44 | wasanzy | Working on my OBD, this is an error am getting which I still don't know why: http://pastebin.com/kvjLDaq5 |
| 13:31.04 | Gugge | WIMPy: great, so any cheap pstn card with a dahdi driver is fine :) |
| 13:31.07 | wasanzy | the 404, am not sure where it is coming from and why |
| 13:31.07 | mjordan | er, nuts. |
| 13:31.12 | mjordan | no answer isn't one of the values |
| 13:31.17 | skrusty | indeed :) |
| 13:31.38 | mjordan | skrusty: those actually should just map to hangup cause codes internally. It wouldn't be hard to expand the optionos. |
| 13:31.42 | mjordan | options, even. |
| 13:31.44 | WIMPy | Gugge: As good as the oscilator on it. |
| 13:31.59 | skrusty | i assumed they did, and then i spotted the limited values in the spec |
| 13:32.06 | mjordan | that wouldn't be SIP cause codes, however. |
| 13:32.10 | skrusty | and, so i've no way of sending a hangup with no answer |
| 13:32.23 | mjordan | skrusty: should be an easy patch ;-) |
| 13:32.28 | file | mjordan, it probably wouldn't be hard to do both... a list for people unfamiliar and then if not in the list... see if it's an explicit cause code |
| 13:32.30 | skrusty | noooooooooooooooooooo :D |
| 13:32.42 | skrusty | see, file will do it for me! ;) |
| 13:32.44 | WIMPy | Gugge: But any bog standard quarz should be ok, unless you do fax via TDM, but if you did, you would already have the timing. |
| 13:32.46 | mjordan | file: yeah. The value type is a "string" |
| 13:33.00 | file | skrusty, you don't want me coding right now :P I'm raging |
| 13:33.03 | mjordan | it could always check for human readable first, then translate it to a numeric cause code or use the AST enum literals |
| 13:33.11 | skrusty | haha :D |
| 13:33.14 | mjordan | skrusty: open source ;-) |
| 13:33.19 | skrusty | damn you! :) |
| 13:33.35 | skrusty | "but you should do it for me...." ;) |
| 13:33.37 | file | skrusty, have faith in yourself! :P |
| 13:33.50 | skrusty | eats his own words |
| 13:36.00 | skrusty | "DO NOT EDIT" ... hmm |
| 13:36.20 | file | generated code |
| 13:36.27 | skrusty | where should i be looking? |
| 13:36.36 | skrusty | ari_channels.c seems to be generated |
| 13:36.40 | skrusty | re |
| 13:36.41 | skrusty | s |
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| 13:37.23 | file | the API is defined in rest-api/api-docs/ |
| 13:37.32 | file | the implementation is elsewhere |
| 13:38.19 | skrusty | can you give me an idea as to where? :) |
| 13:39.09 | mjordan | skrusty: https://wiki.asterisk.org/wiki/display/AST/Create+a+new+resource+with+ARI |
| 13:39.19 | mjordan | skrusty: that should help with modifying a resource as well |
| 13:39.28 | skrusty | ok, cheers. will have a read |
| 13:39.46 | mjordan | generally, once you've updated the JSON and re-built the autogenerated code, you should go update resource_channels.c |
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| 13:40.52 | file | and where would involve me looking and finding it, as there is no immediate answer |
| 13:40.55 | file | :P |
| 13:41.05 | skrusty | ;) |
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| 13:41.15 | skrusty | found it |
| 13:41.43 | skrusty | so, atm it just takes the string and if else's to the corresponding AST_CAUSE_* |
| 13:44.28 | skrusty | so, maybe for now, extend that list to add other common causes |
| 13:46.35 | skrusty | hmm, that code references AST_CAUSE_BUSY, which is not in https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings, (it's referenced as AST_CAUSE_USER_BUSY) |
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| 13:47.11 | file | they are the same |
| 13:47.21 | file | there's a few aliases from olden times |
| 13:47.31 | skrusty | which is the alias? |
| 13:48.19 | file | AST_CAUSE_BUSY is an alias to AST_CAUSE_USER_BUSY |
| 13:48.24 | skrusty | ok, cheers :) |
| 13:48.47 | file | they are at the bottom of include/asterisk/causes.h |
| 13:48.54 | skrusty | thanks file! |
| 13:49.16 | skrusty | got it |
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| 14:21.37 | wasanzy | Working on my OBD, this is an error am getting which I still don't know why: http://pastebin.com/kvjLDaq5 |
| 14:21.45 | wasanzy | the 404, am not sure where it is coming from and why |
| 14:21.59 | *** join/#asterisk callsign (~jonathan@68.235.238.148) |
| 14:24.07 | marceloamorim | guys, anyone knows about TE110P? I have this card but when I use the "#cat /proc/interrupts" I get '16: 75033627 73798074 IO-APIC-fasteoi wcte11xp' but when I check the irc with dahdi_scan, the irc shows is 0 |
| 14:27.42 | [TK]D-Fender | wasanzy: When you look at actual * SIP debug you'll see why... |
| 14:28.11 | [TK]D-Fender | wasanzy: As for WHERE.... you see it right here : OPTIONS sip:10.71.173.187:5060 SIP/2.0 |
| 14:31.31 | *** join/#asterisk quantryx (~jonathan@68.235.238.148) |
| 14:33.50 | *** part/#asterisk gunfree (~gunter@Core4-ALOG-RJ.infolink.com.br) |
| 14:35.31 | sruffell | marceloamorim: TE110P is not actively supported anymore, butâ¦.dahdi_scan always shows 0 for IRQ because the driver doesn't track that information in the span structure anymore. (since commit: http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commit;h=8d23f878) â¦. |
| 14:36.53 | sruffell | I'll give you that dahdi_scan probably should be changed to go look in device tree to find the actual IRQ. |
| 14:38.45 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 14:41.15 | *** join/#asterisk caveat- (hoax@shell.bshellz.net) |
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| 14:44.11 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:45.12 | *** join/#asterisk EachH (54c9f790@gateway/web/freenode/ip.84.201.247.144) |
| 14:50.44 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-yemvurycmcjlsmjz) |
| 14:51.49 | EachH | Hi! Does somebody know how to set SIP header on PJSIP callee channel when using Queue(eg callee is chosen by queue itself)? |
| 14:53.35 | [TK]D-Fender | EachH: Use Local channels as memebers and do it before dialing the device yourself in there |
| 14:54.50 | *** join/#asterisk Synthase_ (uid63346@gateway/web/irccloud.com/x-wvclarghwhpzrwkj) |
| 14:56.35 | *** join/#asterisk xochilpili (~xochilpil@unaffiliated/xochilpili) |
| 14:56.43 | *** join/#asterisk Kilo`byte (~kilobyte@5.231.51.78) |
| 14:56.52 | xochilpili | hi all, |
| 14:57.39 | xochilpili | i have some issues with sipml5 and asterisk, i can login but when i call to *43 (echo-demotest) i cant hear anything |
| 14:57.58 | xochilpili | is there a channel for sipml5 or i can be here? |
| 14:59.11 | EachH | [TK]D-Fender: is there any pitfalls when using these Local channels? there would not be any problems with queue member status handling, transfers? |
| 14:59.31 | [TK]D-Fender | No more than any other channel |
| 15:00.06 | [TK]D-Fender | And it's what you have to do. |
| 15:01.44 | wasanzy | [TK]D-Fender: the sip debug didn't tell much. |
| 15:02.07 | EachH | xochilpili: u're using pjsip or chan_sip? |
| 15:02.14 | [TK]D-Fender | I'm pretty sure it does |
| 15:05.07 | xochilpili | EachH, chan_sip |
| 15:05.46 | *** join/#asterisk Jasonwoot (~FearDoesN@12.133.48.130) |
| 15:06.34 | *** join/#asterisk mjordan (mjordan@nat/digium/x-mwjhbzhdpgpufhwe) |
| 15:06.34 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:10.03 | EachH | xochilpili: ure unlucky, im using pjsip :) You already set up DTLS keys? |
| 15:11.53 | xochilpili | EachH, yes, but i can use pjsip instead, no problem, lucky change |
| 15:12.23 | xochilpili | EachH, i can post all configs and debug if you can see them |
| 15:14.28 | *** join/#asterisk ayrjola (~ayrjola@80.248.109.192) |
| 15:15.15 | EachH | xochilpili: have you configured everything like described there https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 ? |
| 15:16.38 | *** join/#asterisk caveat- (hoax@shell.bshellz.net) |
| 15:18.23 | xochilpili | EachH, yes |
| 15:18.52 | xochilpili | EachH, firewall is off (centos) |
| 15:20.09 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 15:21.07 | *** join/#asterisk raymondhim (~raymondhi@c-66-41-216-194.hsd1.mn.comcast.net) |
| 15:21.16 | EachH | xochilpili: Now, just configure everything for pjsip, and disable chan_sip entirely. And configure endpoints without media_encryption and dtls_* options |
| 15:21.19 | wasanzy | [TK]D-Fender: http://pastebin.com/72aFtfUA This is the sip debug, it looks like the other sip server is sending known request to the Asterisk server. |
| 15:21.49 | [TK]D-Fender | wasanzy: Looking for s in public (domain 10.71.173.187) SIP/2.0 404 Not Found |
| 15:21.50 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 15:21.56 | [TK]D-Fender | Looking for ... not found. |
| 15:22.04 | [TK]D-Fender | Says VERY clearly what it's loking for. |
| 15:22.28 | [TK]D-Fender | And then there is a question of even understanding what the request is, and its purpose |
| 15:25.55 | wasanzy | [TK]D-Fender: the request am not understanding |
| 15:26.11 | [TK]D-Fender | wasanzy: taht is a SIP QUALIFY keep-alive test. |
| 15:26.20 | EachH | xochilpili: my pjsip.conf for webrtc http://pastebin.com/0YLVV2eG |
| 15:26.27 | wasanzy | ok |
| 15:26.49 | [TK]D-Fender | wasanzy: And it doesn't matter that you respond 404... all the other side case is that you answer at all. |
| 15:27.03 | [TK]D-Fender | cares* |
| 15:27.15 | *** join/#asterisk topriddy (~topriddy@41.58.51.195) |
| 15:27.19 | wasanzy | ok |
| 15:27.36 | wasanzy | let me sort out the s in the public |
| 15:27.51 | EachH | xochilpili: Because encryption is disabled you have to run Chromium(or chrome Dev channel) with following options: --disable-user-media-security --use-fake-ui-for-media-stream --disable-webrtc-encryption |
| 15:33.07 | xochilpili | EachH, in "Public Identity" sipml5 option do you type: pjsip:6001@yourealm or sip:6001@yourealm? |
| 15:36.06 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 15:36.32 | EachH | xochilpili: sip:355@192.168.0.1 |
| 15:36.41 | *** join/#asterisk PaybackTony (~Adium@2601:7:700:7fa:b09b:420f:d3f9:8cf8) |
| 15:37.09 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-kozlequwfndmibss) |
| 15:37.26 | EachH | xochilpili: Notice that only 354 & 355 are WebRtc peers |
| 15:37.55 | *** join/#asterisk caveat- (hoax@shell.bshellz.net) |
| 15:42.19 | xochilpili | EachH, where do you have the username and secret for 354 and 355? |
| 15:43.24 | EachH | xochilpili: Nowhere, yo can login with any password |
| 15:44.06 | wasanzy | I have a problem. before users can reach the system, they have to dial a short code like 745 on their cell phone, so I need to define that extension under the public context, now changing that to extension s, will not help. how can I go about it? |
| 15:44.17 | wasanzy | [TK]D-Fender: ^ |
| 15:45.00 | [TK]D-Fender | What re you talking about? |
| 15:45.33 | [TK]D-Fender | If they are dialing 745 ... then you need a match for 745 |
| 15:45.44 | [TK]D-Fender | "s" is not it... |
| 15:46.07 | [TK]D-Fender | And "befor they reach the system? Them dialing *is* an attempt to reach the system |
| 15:46.28 | *** join/#asterisk rrittgarn1 (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 15:47.06 | wasanzy | When users dial 745 on their cell phone, the telco forwards that request to my system @ that extension. but now that I see in the sip debug that the telco is trying to reach extension s , how do I handle that? |
| 15:47.51 | wasanzy | I already have extension 754 define, but changing that to s will produce another not found. |
| 15:48.19 | [TK]D-Fender | wasanzy: How does a cell telco know that 745 belongs to you in the first place? |
| 15:48.41 | xochilpili | EachH, i receive "wrong password" with 354 endpoint |
| 15:48.55 | xochilpili | i just copy and paste you'r pjsip conf |
| 15:49.16 | wasanzy | they have also configured the 745 in their sip server to forward the request to my system |
| 15:49.35 | [TK]D-Fender | wasanzy: do you register with them? |
| 15:49.43 | wasanzy | no |
| 15:49.54 | wasanzy | we don't register |
| 15:50.01 | [TK]D-Fender | wasanzy: Then they shouldn't be asking for "s" |
| 15:50.10 | [TK]D-Fender | wasanzy: Show me an actual call. |
| 15:50.19 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 15:50.27 | wasanzy | allowguest=yes |
| 15:50.28 | wasanzy | ok |
| 15:51.07 | wasanzy | [TK]D-Fender: with sip debug on? |
| 15:51.52 | *** join/#asterisk caveat- (hoax@shell.bshellz.net) |
| 15:52.27 | [TK]D-Fender | obviously |
| 15:52.43 | *** join/#asterisk cyford (~allen@76.122.73.37) |
| 15:54.17 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 15:54.39 | EachH | xochilpili: Try this endpoint configuration: http://pastebin.com/h7k84cTQ |
| 15:55.02 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 15:55.42 | wasanzy | ok |
| 15:56.58 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-jjlafmwvbgnexhcf) |
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| 15:57.17 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:58.14 | xochilpili | EachH, i receive the same output... chan_sip.c:27774 handle_request_register: Registration from '"354"<sip:354@192.168.0.6>' failed for '192.168.0.3:37685' - Wrong password |
| 15:58.36 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-ertltywgtjruahlv) |
| 15:58.57 | EachH | xochilpili:I said earlier that you have to disable chan_sip!!! |
| 15:59.44 | [TK]D-Fender | No you don't.... you just have to be talking to the right thing |
| 16:00.17 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:248:82fa:5bff:fe0a:dfef) |
| 16:01.19 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 16:01.21 | xochilpili | EachH, i get now, failed to connecto to server : module unload chan_sip |
| 16:02.30 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 16:03.36 | EachH | xochilpili: Disable it in modules.conf and restart, an you should be fine. Its happening because of this: https://issues.asterisk.org/jira/browse/ASTERISK-24725 |
| 16:05.42 | wasanzy | http://pastebin.com/0EBe6Dqg |
| 16:05.52 | wasanzy | [TK]D-Fender:^ |
| 16:06.05 | [TK]D-Fender | ... |
| 16:06.08 | [TK]D-Fender | THE WHOLE CALL |
| 16:06.29 | *** join/#asterisk jiuweigui (~jiuweigui@unaffiliated/jiuweigui) |
| 16:06.32 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
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| 16:07.31 | xochilpili | EachH, i noload the chan_sip module, and from sipml5 i type pjsip:354@192.168.0.6 and got: is not a valid Public identity |
| 16:08.02 | EachH | xochilpili: sip:354@192.168.0.6 |
| 16:08.12 | xochilpili | if i change it to: sip:354@192.168.0.6 i receive: Internal Server Error and Cli: res_pjsip/pjsip_configuration.c:1973 ast_sip_retrieve_auths: Couldn't find auth '354'. Cannot authenticate |
| 16:09.12 | EachH | xochilpili: Revert your config to first version that i provided |
| 16:09.46 | xochilpili | without aut*? |
| 16:09.46 | EachH | xochilpili: One without passwords |
| 16:11.37 | *** join/#asterisk volga629 (~bendersky@CPE4c5e0cb3c2a0-CM0c473dd46e50.cpe.net.cable.rogers.com) |
| 16:11.53 | *** join/#asterisk caveat- (hoax@shell.bshellz.net) |
| 16:12.32 | xochilpili | EachH, now i can connect, trying a call to 1000 (like the manual) and receive: Not Acceptable Here |
| 16:12.37 | volga629 | Hello Everyone, reviewboard.asterisk.org when filling new request what value should be for this Base Directory: ? |
| 16:13.04 | xochilpili | EachH, the manual set an extension 1000 to playback demo-congrats> https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 |
| 16:17.49 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
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| 16:28.08 | xochilpili | EachH, i can connect now, and i can make a call, but there's no audio (same problem before using sip) |
| 16:31.34 | file | xochilpili, use the latest release candidate - OpenSSL did a security fix which broke DTLS for a lot of programs |
| 16:32.55 | xochilpili | file, release candidate for sipml5? sorry i dont get it |
| 16:33.04 | file | for Asterisk |
| 16:33.08 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |
| 16:33.58 | xochilpili | file im using asterisk 13 |
| 16:34.13 | xochilpili | Asterisk 13.1.0 |
| 16:34.23 | file | if your Linux distribution is running a version of OpenSSL with the security fix in it then DTLS won't work and you won't get audio |
| 16:34.42 | file | the fix is in Asterisk 13.2.0-rc1 |
| 16:34.57 | mjordan | fix/hacky workaround to OpenSSL shenanigans |
| 16:35.23 | xochilpili | file, so do i have to uninstall this asterisk? |
| 16:35.27 | file | mjordan, supposedly it should have always been enabled and is ALWAYS enabled in latest versions |
| 16:35.53 | file | just nobody knew it was required because of the lack of documentation |
| 16:36.09 | file | xochilpili, I don't know how you installed it |
| 16:36.33 | xochilpili | file, i compile it from source |
| 16:36.49 | xochilpili | rpm -qa | grep openssl -> openssl-1.0.1e-30.el6_6.5.x86_64 |
| 16:36.53 | file | then you can most likely recompile 13.2.0-rc1 and install it over your existing install |
| 16:36.58 | file | erm compile |
| 16:37.39 | Xaviertoor | Iam try and res_hep.c:418 hep_queue_cb: Unable to send packet: Address Family mismatch between source/destination |
| 16:38.28 | xochilpili | file, do you have a download link for that version of asterisk? |
| 16:38.50 | file | no, but it is easily findable |
| 16:39.08 | xochilpili | http://fossies.org/linux/misc/asterisk-13.2.0-rc1.tar.gz/index_o.html << is this ok? |
| 16:40.04 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 16:40.08 | file | I can't vouch for that server, it is not the official Asterisk downloads site |
| 16:40.27 | xochilpili | http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13.2.0-rc1.tar.gz << that one |
| 16:40.54 | mjordan | yes, that is the RC |
| 16:41.47 | xochilpili | so, if i compile it and i dont uninstall asterisk (compiled before) doesnt matter it will overwrite? |
| 16:44.39 | file | yes, if some stuff isn't built and old remains it'll tell you when you do make install |
| 16:45.05 | file | there is also a "make uninstall" you can do to uninstall |
| 16:47.17 | *** join/#asterisk topriddy (~topriddy@41.58.51.195) |
| 16:47.55 | marceloamorim | sruffell: thx for the feedback, I don't know what happend, but now I got the other commands for pri |
| 16:49.09 | EachH | xochilpili: I recalling that i had same kind of problem when used default ice servers. And my config is without encryption, so your problem lies not in encryption problems |
| 16:49.19 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 16:52.02 | xochilpili | EachH, when i use your 'no encryption' i got : "Not acceptable here". then i put "dtls" and the certs, and i could make a call, without sound |
| 16:52.55 | EachH | xochilpili: Have you seen my message about chrome options& |
| 16:55.55 | EachH | xochilpili: But no-dtls is just for testing purposes, so yo should recompile Asterisk 13.2 anyway |
| 16:58.40 | xochilpili | EachH, yes, i did that, but i receive the same output: "Not acceptable here" |
| 17:00.18 | Xaviertoor | Do not listen to the playback by calling extension 1000 |
| 17:00.39 | Xaviertoor | and CLI return Unable to send packet: Address Family mismatch between source/destination |
| 17:02.07 | EachH | xochilpili: You did what? runned Chrome-dev with provided options? |
| 17:03.08 | xochilpili | yes |
| 17:03.32 | *** join/#asterisk caveat- (hoax@shell.bshellz.net) |
| 17:05.10 | EachH | xochilpili: U re using Chrome-dev channel version? |
| 17:08.10 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 17:08.26 | Kobaz | <PROTECTED> |
| 17:08.30 | AnonGirl | so um |
| 17:10.05 | xochilpili | EachH, no, but after install asterisk 13.2 rc1 transport-ws is not showing in pjsip show transports |
| 17:11.33 | EachH | xochilpili: Maybe you replaced all of you configs with make samples? |
| 17:13.43 | *** join/#asterisk riess82 (~riessma@80-121-2-124.adsl.highway.telekom.at) |
| 17:14.30 | xochilpili | EachH, noo! :D |
| 17:14.40 | xochilpili | i did not use make samples |
| 17:15.02 | xochilpili | i receive a new errror: ERROR[26229]: res_pjsip_config_wizard.c:826 object_type_loaded_observer: Unable to load config file 'pjsip_wizard.conf' |
| 17:16.19 | file | that's not a problem |
| 17:16.21 | EachH | Wtf is that?? Never seen it before. Try to read changelog :) My experience only related to Asterisk 13.1< |
| 17:17.03 | xochilpili | EachH, file, i just use module unload res_pjsip_config_wizard and thats all |
| 17:17.40 | xochilpili | EachH, the same transport you have in pjsip.conf, is not loading, just 0.0.0.0-udp |
| 17:19.57 | xochilpili | i dont get it, i also remove [0.0.0.0-udp] transport from pjsip.conf and still there no matter what: pjsip reload, core reload ... |
| 17:20.29 | EachH | check if chan_sip is running again, check for module res_pjsip_transport_websocket.so |
| 17:20.52 | EachH | it is res_pjsip_transport_websocket.so who responsible for ws transport |
| 17:21.55 | xochilpili | EachH, res_pjsip_transport_websocket.so running, not chan_sip loaded |
| 17:25.01 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 17:25.16 | *** join/#asterisk dork (rev@tard.crapitalism.net) |
| 17:25.25 | EachH | I really dunno where problem really is on your side... My config is working for me with 13.1, Chrome-dev --disable-user-media-security --use-fake-ui-for-media-stream --disable-webrtc-encryption |
| 17:26.52 | EachH | If you get not-acceptable here it means that there is encryption involved or unsupported codec or something else. Maybe encryprion enabled by default in 13.2? |
| 17:27.37 | xochilpili | EachH, im rolling back to 13.1 |
| 17:29.21 | EachH | What linux distro you using? |
| 17:30.32 | xochilpili | Centos |
| 17:31.35 | *** join/#asterisk caveat- (hoax@shell.bshellz.net) |
| 17:31.42 | xochilpili | i got the same problem with 13.1 [transport-ws] not found |
| 17:31.47 | *** part/#asterisk AnonGirl (janice@need.sleep.caffeinet.uk.to) |
| 17:34.46 | EachH | Uninstalled 13.2 properly? I cant help you with this 13.1 It looks like system/build related problem |
| 17:35.28 | xochilpili | EachH, yeah, i messed up :( |
| 17:36.18 | xochilpili | from zoiper i can login and call, no problem |
| 17:36.26 | EachH | Try to make samples and replace only pjsip.conf & edit http.conf |
| 17:39.11 | EachH | xochilpili: Only pjsip.conf & enabilng http in http.conf should be changed in default configs to make it work in my case(obviously you need to ajust extensions.conf) |
| 17:41.08 | EachH | xochilpili: BTW, im also changing ice servers list on sipml side to my local one (im using resiprocate-turn-server) |
| 17:42.31 | xochilpili | i make config, and place you pjsip.conf, and again there's no transport-ws |
| 17:42.34 | xochilpili | loaded |
| 17:42.38 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 17:44.07 | xochilpili | sorcery.c:1180 sorcery_wizard_load: Type 'transport' is not reloadable, maintaining previous values << what's that? |
| 17:45.17 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 17:45.26 | EachH | transports cant be reloaded, u should restart asterisk |
| 17:45.59 | EachH | xochilpili: make samples! make config is just installing init scripts |
| 17:46.45 | EachH | xochilpili: remove /etc/*, make samples, replace pjsip.conf & enable http in http.conf |
| 17:46.59 | EachH | xochilpili: AAW sorry |
| 17:47.08 | EachH | xochilpili: remove /etc/asterisk/* |
| 17:47.21 | EachH | xochilpili: not /etc/* :) |
| 17:47.26 | xochilpili | jaja |
| 17:49.55 | EachH | xochilpili: i forgot about disabling chan_sip, do it too |
| 17:50.38 | xochilpili | EachH, i did make samples and make conf, im right now rewriting extensions.conf also i replaced pjsip.conf <yours> |
| 17:52.20 | xochilpili | i didnot backup extensions.conf :/ |
| 17:54.17 | EachH | just copy sample shit from asterisk webrtc guide |
| 17:55.26 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 17:55.43 | EachH | xochilpili: and btw, cos context of 354 endpoint is default, you just can call demo: 1000 |
| 17:55.52 | *** join/#asterisk caveat- (hoax@shell.bshellz.net) |
| 18:01.07 | *** join/#asterisk EachH (5c37095b@gateway/web/freenode/ip.92.55.9.91) |
| 18:03.01 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 18:04.02 | xochilpili | EachH, same problem :/ |
| 18:04.20 | xochilpili | pjsip show transports : only show udp |
| 18:04.59 | EachH | What are in logs? |
| 18:07.06 | EachH | Yuo may have something listrning on 8088 port already |
| 18:07.33 | xochilpili | EachH, now i can login, and make the call response: Not accepted here |
| 18:07.42 | xochilpili | where do i download chrome-dev? |
| 18:10.31 | EachH | http://chromium.woolyss.com/, or just change channel in chrome settings |
| 18:11.12 | xochilpili | EachH, again, Failed to connet to the server |
| 18:11.17 | xochilpili | aggrrrrs! |
| 18:11.36 | *** join/#asterisk DocScrutinizer05 (~saturn@openmoko/engineers/joerg) |
| 18:11.44 | *** part/#asterisk DocScrutinizer05 (~saturn@openmoko/engineers/joerg) |
| 18:13.11 | EachH | xochilpili: Again after what? you checked for 8088 port? |
| 18:13.14 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 18:15.50 | xochilpili | <PROTECTED> |
| 18:16.20 | xochilpili | im such an idiot: enabled=yes ; commented line |
| 18:16.22 | xochilpili | sorry |
| 18:17.26 | xochilpili | EachH, now, testing with chrome-dev |
| 18:19.52 | qakhan | hi All, is there any way we can creat customized Table and add some columns. like, date_time, callerid, calledid, uniqueid, user, etc.... |
| 18:20.42 | xochilpili | EachH, im on fedora, there's no installation for this os |
| 18:23.38 | xochilpili | EachH, no make any sense, to install chromium, why i cant use with my google chrome browser installed 38.0.2125.111 on fedora? |
| 18:24.50 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 18:26.20 | xochilpili | failed to set remote answer sdp: Called with SDP without DTLS fingerprint << i got this |
| 18:27.51 | *** join/#asterisk caveat- (hoax@shell.bshellz.net) |
| 18:38.52 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 18:39.42 | [TK]D-Fender | qakhanhi All, is there any way we can creat customized Table and add some columns. like, date_time, callerid, calledid, uniqueid, user, etc.... What do you expect will actually put things in there? |
| 18:46.35 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 18:47.31 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 18:48.45 | xochilpili | [TK]D-Fender, i know you're very busy, but have a clue what's going on with sipml5 with no sound issue? |
| 18:49.23 | *** join/#asterisk ibercom (5320c26f@gateway/web/freenode/ip.83.32.194.111) |
| 18:49.30 | EachH | xochilpili: You can, but it's questionable that it supports those flags. If not, you will need to tinker with dtls, which is broken by latest OpenSSL update in aster 13.1 |
| 18:49.37 | qakhan | date_time = call date and time, callerid = 123-xxx-xxxx, calledid = 345-xxx-xxxx, uniqueid = call unique id, user = username |
| 18:50.21 | [TK]D-Fender | qakhan: I did ask what you were going to want in the fields.. the names were obvious... what ACTION are you expecting to actually SETR those values for you? |
| 18:50.59 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 18:51.56 | EachH | xochilpili: get chrome-dev from https://www.google.com/chrome/browser/desktop/index.html?platform=linux&extra=devchannel and try it, if it will work, then you may start tinkering with 13.2 and dtls |
| 18:52.59 | xochilpili | EachH, downloading... |
| 18:53.39 | qakhan | [TK]D-Fender i just want to have mentioned values in that table. |
| 18:53.48 | [TK]D-Fender | .. |
| 18:54.07 | [TK]D-Fender | qakhan: And I'll ask again How do you expect to PUT them there? |
| 18:54.23 | qakhan | through dialplan |
| 18:54.33 | [TK]D-Fender | Then that is just database work, and nothing Asterisk cares about. |
| 18:55.06 | [TK]D-Fender | qakhan: * can use it's DB-related apps & functions, or you can call your own script externally |
| 18:55.56 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 18:56.23 | qakhan | [TK]D-Fender can you please tell me in detail how i can put * record in table |
| 18:56.30 | [TK]D-Fender | ~book |
| 18:56.30 | 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 |
| 18:56.32 | [TK]D-Fender | ^^^^ |
| 18:56.43 | [TK]D-Fender | *'d DB functionality is well documented. |
| 18:56.48 | [TK]D-Fender | *'s* |
| 18:57.20 | qakhan | ok |
| 18:58.45 | xochilpili | EachH, i cant run google-chrome-unstable --disable-user-media-security --use-fake-ui-for-media-stream --disable-webrtc-encryption |
| 18:59.02 | xochilpili | kernel issue: kernel:[372114.696691] Process chrome (pid: 23745, ti=c372a000 task=e699f8e0 task.ti=c372a000) |
| 18:59.29 | *** join/#asterisk ayrjola (~ayrjola@80.248.109.192) |
| 19:00.13 | ibercom | Asterisk realtime: which mysql engine is better for asterisk ? Myisam or Innodb ? |
| 19:01.12 | ibercom | I search for sippeers. |
| 19:01.44 | *** join/#asterisk airjump (~Thunderbi@p5B0A0D11.dip0.t-ipconnect.de) |
| 19:03.06 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 19:03.18 | [TK]D-Fender | ibercom: Considering what the rest of load must looks like on a box I can't see how the backend stoarge format even matters... |
| 19:04.57 | ibercom | I look for performance... |
| 19:11.30 | *** join/#asterisk sparetire (~sparetire@unaffiliated/sparetire) |
| 19:12.03 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 19:12.11 | *** join/#asterisk danjenkins (~anonymous@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 19:15.47 | *** join/#asterisk lanning (~lanning@50-193-22-25-static.hfc.comcastbusiness.net) |
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| 19:32.01 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 19:32.27 | [TK]D-Fender | ibercom: When you consider the realistic size of * DB's this is probably the lowest level of ricing possible... |
| 19:38.31 | ibercom | Yes, you're right. I look for a good configuration for locks and threading. |
| 19:39.17 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 19:40.11 | ibercom | Another question about Asterisk Realtime: When I make a call, chan_sip looks for the destination number in the sippeers table, even before asterisk starts dialplan. Why ? |
| 19:41.29 | [TK]D-Fender | Because if has to find out if the caller is even KNOWN and allowed |
| 19:41.39 | [TK]D-Fender | And theree WHERE to look in the dialplan. |
| 19:41.49 | [TK]D-Fender | That magical thing known as the "context" |
| 19:42.23 | [TK]D-Fender | It doesn't just start doing what it says and then start asking who they are :) |
| 19:43.37 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 19:44.12 | *** join/#asterisk caveat- (hoax@shell.bshellz.net) |
| 19:44.55 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 19:45.47 | ibercom | This is not optimal, specially when calling external number that have nothing to do with my users or peers. |
| 19:46.25 | [TK]D-Fender | incorrect |
| 19:46.34 | ibercom | Caller are mine. Called no always. |
| 19:46.42 | [TK]D-Fender | Because * has to know that that is even a device that is permitted to call anywhere |
| 19:46.54 | [TK]D-Fender | Let alone deciding where it is allowed to call |
| 19:47.29 | [TK]D-Fender | Every device can be in a different context. Calls could arrive from anywhere. |
| 19:47.45 | [TK]D-Fender | And what you allw any given device to do may not match any other |
| 19:48.01 | *** join/#asterisk danjenkins (~anonymous@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 19:48.18 | [TK]D-Fender | * needs to know wht context, so that automatically requires a peer match |
| 19:49.36 | ibercom | From anywhere yes, but I mean TO: header ¿? |
| 19:51.21 | [TK]D-Fender | "To:" isn't necessarilty even important. The target is actually the INVITE line itself |
| 19:51.44 | [TK]D-Fender | And the "From:" determines who the callere is and thus what context their call falls in. |
| 19:53.38 | ibercom | Yes, but asterisk look for "target" in sippeers table. It is odd. |
| 19:54.12 | mjordan | not really. |
| 19:54.23 | mjordan | If it is a peer that is doing the outbound dialing, Asterisk has to know what codecs to offer. |
| 19:54.23 | ibercom | Why ? |
| 19:54.31 | mjordan | If it isn't a peer, then it has to use the codecs defined in the general section. |
| 19:54.38 | mjordan | Otherwise, how does it know what capabilities it should offer? |
| 19:55.07 | mjordan | Or whether the outbound "thing" - peer or 'general' section defined (which is always a little goofy) - can do direct media? Or support SRTP? Or should it connect using a secure transport? |
| 19:55.11 | [TK]D-Fender | Again this shouldn't even be a question. Proof of identity is REQUIRED to know what context to dump them in. |
| 19:55.14 | mjordan | The Dial statement doesn't provide any of that |
| 19:55.30 | mjordan | [TK]D-Fender: are we talking about inbound or outbound? |
| 19:55.34 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 19:55.40 | [TK]D-Fender | That is even before the need for codecs at all |
| 19:55.50 | mjordan | I was thinking it was outbound from Asterisk, but if its inbound, then yeah, there's nothing to do :-) |
| 19:55.54 | [TK]D-Fender | I could ahve it set so they will NOT agree on codecs period yet still run some dialplan... |
| 19:56.04 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 19:56.55 | *** join/#asterisk sruffell (sruffell@asterisk/the-kernel-guy/sruffell) |
| 19:56.55 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 19:57.00 | ibercom | Inbound to asterisk. After dialplan outbound. |
| 19:59.16 | ibercom | Inbound: it checks "from" but why "to / target " ? |
| 20:00.43 | *** join/#asterisk caveat- (hoax@shell.bshellz.net) |
| 20:01.03 | [TK]D-Fender | * shouldn't care about the "To:" |
| 20:01.17 | [TK]D-Fender | INVITE <-------- that defines the dialplan lookup |
| 20:01.30 | ibercom | This is the problem or bug... |
| 20:01.34 | [TK]D-Fender | Neither |
| 20:01.49 | [TK]D-Fender | SOP |
| 20:02.21 | [TK]D-Fender | If you need to gt it from the "To:" you can parse that out in the dialplan, but it will require beginning processing the call |
| 20:05.50 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 20:08.28 | ibercom | But asterisk does automatically that search "INVITE destination in sippeers", before processing the dialplan. |
| 20:11.09 | ibercom | The process of "from:" is correct. I understand it, but the other... |
| 20:11.16 | *** join/#asterisk P-G (~P-G@208-58-0-74.md.subnet.cable.rcn.com) |
| 20:12.24 | [TK]D-Fender | ibercom: You're missing the point. INTIVE is the number to lookup for the dialplan. It does not CARE about dialplan yet. It first cares who the CALLER is. |
| 20:12.44 | [TK]D-Fender | ibercom: as in "If I don't like you then I don't care what you want". |
| 20:13.12 | [TK]D-Fender | ibercom: It's "Who are you? Ok, what do you want?", not "What do you want? Who are you?" |
| 20:14.14 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 20:15.06 | P-G | Mind if I ask a few potentially stupid questions? |
| 20:15.27 | [TK]D-Fender | ~ask |
| 20:15.28 | 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. |
| 20:15.49 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 20:16.49 | P-G | Ok. Basically, I'm interested in deploying a PBX and I am vaguely familiar with Asterisk from working with stuff like FreePBX but I want to do a few things I haven't been able to figure out. |
| 20:16.51 | ibercom | Thank you <[TK]D-Fender> I will read something else :-) |
| 20:17.34 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 20:17.43 | [TK]D-Fender | ibercom: Just keep in mind the standard linear concept and it'll be fine :) |
| 20:18.17 | P-G | The first is whether there's a convenient way to configure phone number routing, like a firewall, seperately for each call group and device. |
| 20:18.38 | [TK]D-Fender | those terms are a little mixed |
| 20:18.55 | P-G | Ok, let me see if I can explain better... |
| 20:19.06 | [TK]D-Fender | Firewalling is a "device" concept not a "phone number" one |
| 20:19.16 | P-G | Sorry, I think of it as software. |
| 20:20.13 | P-G | Basically I want to drop or accept calls from specific numbers to specific call groups or numbers/devices. |
| 20:20.30 | [TK]D-Fender | P-G: that's what the dialplan is for... |
| 20:20.44 | [TK]D-Fender | P-G: Don't put things in their context that you don't want them to dial. |
| 20:21.10 | P-G | Imagine the case of incoming call routed to call group for department1 routed to call-group1 answered by user1 on phone1. |
| 20:21.32 | P-G | Is it possible for the call to be dropped at any of those transitionary phases? |
| 20:21.49 | P-G | Inbound is simple enough but that's three call groups. |
| 20:22.15 | P-G | And then a device/number. |
| 20:22.20 | [TK]D-Fender | What is the "call group" you're referring to? |
| 20:22.40 | P-G | Like a ring group. |
| 20:23.18 | [TK]D-Fender | Well call comes in targeting a number. That lands in a context and your dialplan starts its processing.... |
| 20:23.52 | P-G | The dialplan is destination based though, right? |
| 20:24.03 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 20:25.04 | P-G | I'm imagining source based call priority where calls with insufficient priority are not passed to the next hop unless the hop's threshold is lower. Something like that. |
| 20:26.34 | [TK]D-Fender | Contexts are what separate things in the dialplan. |
| 20:26.46 | [TK]D-Fender | And the Exten you land on is based on sorting rules |
| 20:27.04 | P-G | Ok so it is possible through those means? |
| 20:27.32 | P-G | How would you attribute the priority threshold of the next hop though? |
| 20:27.42 | [TK]D-Fender | What "next hop"? |
| 20:28.06 | P-G | The destination (e.g. next call group). |
| 20:28.23 | [TK]D-Fender | that is just more priorities in your exten as you're processing... |
| 20:28.46 | P-G | I don't understand. :( |
| 20:29.28 | P-G | Can a call group or forwarding number be attributed an indefinite priority threshold value? |
| 20:30.19 | [TK]D-Fender | "indefinite priority threshold value" <- this is not a "thing" as far as I can see. |
| 20:30.32 | P-G | Like if I were to log in and set department1's priority threshold to 7 and someone calls in with a priority of 6 the call is dropped or forwarded ensewhere. |
| 20:30.42 | P-G | Yeah, I'm making up terms... |
| 20:30.50 | [TK]D-Fender | It is feeling like you are trying to find a parallel to some other process you are used to vs how *'s dialplan actually works... and no such parallel exists directly |
| 20:31.18 | [TK]D-Fender | Doesn't mean you can't end up with the outcome you want |
| 20:31.19 | *** join/#asterisk caveat- (hoax@shell.bshellz.net) |
| 20:31.22 | [TK]D-Fender | but you need to break it down first |
| 20:31.32 | P-G | Yeah, ok. |
| 20:31.37 | P-G | Let's see. |
| 20:31.41 | [TK]D-Fender | Lets start with "priority". Where does this ranking come from in the first place? |
| 20:32.36 | P-G | Well it could be hard coded by the system admin if necessary but ideally it could also be set programatically. |
| 20:33.27 | P-G | It's an integer value that represents how important a call is. |
| 20:33.29 | [TK]D-Fender | That is more dialplan logic you need to code. |
| 20:33.45 | P-G | Ok so that would be coded into the dialplan? |
| 20:33.46 | [TK]D-Fender | in order to pull that value from "somewhere" |
| 20:34.07 | P-G | Yeah, I'm just trying to get some idea of how this would be done. |
| 20:34.36 | P-G | So this variable could be attributed to a call group or phone (number)? |
| 20:34.37 | [TK]D-Fender | Yes. having a value like that in mempor at all is stuff for you to code. Then determining FLOW based on that.. is MORE dialplan |
| 20:35.00 | P-G | Mempor? |
| 20:35.05 | [TK]D-Fender | memory |
| 20:35.35 | [TK]D-Fender | You would CHECK for this value in your dilplan and the branch to different parts of it depending on what you want to do. |
| 20:35.57 | P-G | Ok, that sounds like what I want. What language does that use? |
| 20:36.03 | [TK]D-Fender | Think of this a RAW programming. There is no automatic concept for this. YOU set these values and check for them yourself. |
| 20:36.09 | [TK]D-Fender | DIAL is the language. |
| 20:36.14 | [TK]D-Fender | DIALPLAN* |
| 20:36.25 | P-G | Dialplan is its own language? |
| 20:36.44 | [TK]D-Fender | yes |
| 20:36.47 | [TK]D-Fender | ~book |
| 20:36.47 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:36.48 | [TK]D-Fender | ^^^^ |
| 20:36.55 | [TK]D-Fender | You need to learn your * basics |
| 20:37.10 | [TK]D-Fender | And the dialplan is about 90% of Asterisk. |
| 20:37.15 | [TK]D-Fender | (possibly more) |
| 20:37.18 | P-G | Lol, I do. I'll pick up a copy next time I go to a store. |
| 20:37.27 | P-G | I thought asterisk used mostly C though. |
| 20:37.32 | P-G | Not anymore? |
| 20:37.41 | P-G | My info is years old. |
| 20:38.01 | [TK]D-Fender | And entirely mistaken if you thought this was the level you should be thinking about :) |
| 20:38.24 | P-G | What? |
| 20:38.28 | [TK]D-Fender | Your car is made with a certain amount of steel. It doesn't mean you need to FORGE your car yourself in order to drive it. |
| 20:38.53 | P-G | You think I can find a pre-made module or something? |
| 20:38.56 | [TK]D-Fender | Asterisk is written in "C". But you are writing processing rules FOR Asterisk ... not "executable machine code" |
| 20:39.01 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 20:39.13 | [TK]D-Fender | You need to understand the very bascs of *. You aren't there yet... |
| 20:39.21 | [TK]D-Fender | You seriously need to sit down with the books. |
| 20:39.26 | P-G | Well yeah but last time I used Asterist there were fields for C configuration. |
| 20:39.28 | [TK]D-Fender | running * doesn't require any "C". |
| 20:39.48 | P-G | Yeah, I will read the book. I just want to make sure Asterisk will actually do what I want first though. |
| 20:39.51 | [TK]D-Fender | No.... no normal user should know or care about "C" |
| 20:40.01 | P-G | But I love C. ;( |
| 20:40.15 | [TK]D-Fender | And "c" isn't the solution for everything |
| 20:40.42 | P-G | Well, I will read the book and use whatever language they want if it is possible to do what I want. |
| 20:40.57 | [TK]D-Fender | All call processing = dialplan. Always has been. |
| 20:41.19 | P-G | Ok, and that can handle the call priority thing. |
| 20:41.43 | [TK]D-Fender | Call comes in and I want to A, then B, then check for something, and either C or D depending. <- ALL dialplan. |
| 20:41.55 | [TK]D-Fender | ALL processing of the call = dialplan |
| 20:42.39 | P-G | And constants or variables can be attributed indefinitely to A, B, C, or D? |
| 20:43.03 | P-G | Even if the service restarts? |
| 20:43.11 | [TK]D-Fender | those letters are hypotheitcal ACTIONS... not "variables" |
| 20:43.35 | P-G | Ok so the targets then. |
| 20:43.49 | P-G | Source and destination. |
| 20:44.07 | [TK]D-Fender | Call comes in. They dialed 12345. Therefor I want to Answer, playback a sound, read a number from the user, if they dialed 1 go here... if they dialed 2, go there... etc |
| 20:44.30 | ayrjola | Hi, |
| 20:45.14 | P-G | What about, they called from 1 so priority is 2, destination is 3 with a priority threshold of 4 so drop call? |
| 20:45.37 | P-G | Is it possible to set 2 and 4 as persistent variables? |
| 20:46.18 | ayrjola | I have tried stress testing pjsip asterisk 13.1.0 with sipp but after 200 concurrent calls I get 488 or 503 answers from asterisk. |
| 20:46.38 | ayrjola | I'm missing some configuration parameter? |
| 20:46.56 | *** join/#asterisk caveat- (hoax@shell.bshellz.net) |
| 20:47.11 | ayrjola | with old chan_sip I could hit close to 2000 concurrent with same kind hardware |
| 20:47.21 | [TK]D-Fender | P-G: Yes. |
| 20:47.29 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 20:47.36 | P-G | Ok, nice. |
| 20:47.58 | P-G | Two more questions. I can hold off if these are too irritating... |
| 20:48.03 | [TK]D-Fender | P-G: You can have "global variabels", and other more sticky storage like AstDB to hold those |
| 20:48.19 | [TK]D-Fender | P-G: Let's git 'r' done |
| 20:48.23 | P-G | Ok but as long as it can be done it's worth learning. :) |
| 20:48.55 | P-G | Ok, is there a reliable and generally compatible method of implementing BLF in conjunction with software SIP clients? |
| 20:49.08 | P-G | Like for call parking. |
| 20:49.19 | [TK]D-Fender | * does BLF.... |
| 20:49.43 | P-G | Zoiper says it supports BLF, they are definitely compatible? |
| 20:49.52 | [TK]D-Fender | you don't "implement" it so much as define "hints" ath say what the device subscribes to and what * watches for that status |
| 20:50.09 | [TK]D-Fender | * BLF works with pretty much everything |
| 20:50.31 | P-G | Ok, I guess that's good enough for me. |
| 20:51.04 | P-G | Last is the issue of contacts synchronization. Is that normally done through Asterisk or some other service? |
| 20:51.22 | [TK]D-Fender | P-G: Now this is not the same as letting a device SAY that it's "busy" for instance. |
| 20:51.33 | [TK]D-Fender | P-G: Something else. |
| 20:51.37 | P-G | Yeah, I know. |
| 20:51.46 | [TK]D-Fender | P-G: that is never Asterisk based. |
| 20:51.46 | P-G | But I assume that is also supported... |
| 20:51.55 | P-G | Ok, how might it be done? |
| 20:51.56 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 20:52.02 | P-G | LDAP? |
| 20:52.06 | [TK]D-Fender | P-G: Some devices can usee a more common storage like LDAP. |
| 20:52.14 | [TK]D-Fender | One otopin... which depends on the device |
| 20:52.23 | P-G | Ok, that's another can of worms. |
| 20:52.27 | [TK]D-Fender | And the format et cwill likely be specific to that brand |
| 20:52.36 | [TK]D-Fender | can? BUCKET. |
| 20:52.49 | P-G | Yeah, I'll have to look for compatability charts. |
| 20:53.08 | P-G | That all sounds promising though. |
| 20:53.52 | P-G | I guess I should make sure Asterisk supports SIPS. |
| 20:53.57 | P-G | Or something like it. |
| 20:54.05 | P-G | For tunneling. |
| 20:54.20 | P-G | And hand shaking. |
| 20:54.43 | [TK]D-Fender | no SIPS. TLS + SRTP |
| 20:54.53 | [TK]D-Fender | (last I heard) |
| 20:55.15 | P-G | Ok, that's good enough for me too. |
| 20:55.29 | P-G | ZRTP seems popular but I guess that's not taking on quickly? |
| 20:56.28 | [TK]D-Fender | Nope |
| 20:56.34 | [TK]D-Fender | pretty non-existant |
| 20:56.46 | [TK]D-Fender | RFC ----> NO COMMENT |
| 20:57.02 | P-G | Yeah, I have read good things about it though. SIPS still looks best to me. |
| 20:57.14 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 20:57.30 | P-G | And I've read that SMS is not worth the effort of programming? |
| 20:57.45 | P-G | And MMS is even harder? |
| 20:58.44 | [TK]D-Fender | those are more "cell" concepts than SIP |
| 20:59.06 | [TK]D-Fender | SIP has it's own "message" methods and can be used to carry those from an ITSP for instance. |
| 20:59.09 | Penguin | [TK]D-Fender: its |
| 20:59.12 | [TK]D-Fender | Compliance may vary |
| 20:59.41 | P-G | Yeah but can Asterisk handle them to some degree or is it SIP/AIX specific? |
| 21:00.17 | P-G | Are there other services that might be used to act as SMS/MMS exchanges? |
| 21:00.36 | [TK]D-Fender | Asterisk is not a great messaging platform |
| 21:00.41 | [TK]D-Fender | it is meant as a voice platform |
| 21:00.49 | P-G | Ok. |
| 21:00.51 | [TK]D-Fender | But it does have some handling capabilities |
| 21:00.57 | P-G | Hm. |
| 21:01.12 | P-G | Enough to act as a gateway? |
| 21:01.22 | [TK]D-Fender | Go read hte WIKI for the "MESSAGE" dialplan function and related bits to get an idea |
| 21:01.30 | P-G | Like SMS/MMS --> Email? |
| 21:01.35 | [TK]D-Fender | Even message processing = dialplan. |
| 21:01.48 | P-G | Ok, let me see if I can find that. |
| 21:01.49 | [TK]D-Fender | All of that |
| 21:02.19 | P-G | http://www.voip-info.org/wiki/view/Asterisk+cmd+SendText |
| 21:02.19 | P-G | ? |
| 21:02.41 | [TK]D-Fender | no. |
| 21:02.45 | [TK]D-Fender | And that's the ANCIENT WIKI |
| 21:02.49 | [TK]D-Fender | ~asteriskwiki |
| 21:02.49 | infobot | hmm... asteriskwiki is http://wiki.asterisk.org |
| 21:02.52 | [TK]D-Fender | ^^^ |
| 21:03.47 | P-G | https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=4260046&src=search |
| 21:03.47 | P-G | ? |
| 21:05.18 | [TK]D-Fender | .... |
| 21:05.22 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 21:05.23 | [TK]D-Fender | [16:01][TK]D-FenderGo read hte WIKI for the "MESSAGE" dialplan function and related bits to get an idea |
| 21:05.41 | [TK]D-Fender | "MESSAGE" dialplan function <--------- |
| 21:05.48 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 21:05.48 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 21:06.44 | P-G | Yeah, I'm not finding that. |
| 21:07.04 | P-G | I know SIP has some IM provisions, are you sure message doesn't apply to that? |
| 21:07.21 | [TK]D-Fender | it DOES. |
| 21:07.30 | [TK]D-Fender | I gave you the specifi them to look for. |
| 21:07.40 | [TK]D-Fender | Dialplan functions <-------- |
| 21:07.44 | [TK]D-Fender | name : MESSAGE |
| 21:07.45 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 21:07.52 | [TK]D-Fender | not "sms", not "sendtext" |
| 21:07.57 | [TK]D-Fender | MESSAGE |
| 21:08.04 | P-G | I did search for that. |
| 21:08.22 | [TK]D-Fender | <PROTECTED> |
| 21:08.30 | P-G | https://wiki.asterisk.org/wiki/dosearchsite.action?queryString=dialplan+functions+message |
| 21:08.33 | [TK]D-Fender | I sad FUNCTION, not APPLICATION |
| 21:08.43 | [TK]D-Fender | night/day |
| 21:09.09 | P-G | https://wiki.asterisk.org/wiki/display/AST/Dialplan+Functions?src=search |
| 21:09.14 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_MESSAGE |
| 21:09.16 | P-G | No "message". |
| 21:09.22 | [TK]D-Fender | ight there |
| 21:09.31 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Dialplan+Functions <-- function list |
| 21:09.55 | P-G | Ok, that list was old? |
| 21:10.08 | [TK]D-Fender | No, you're just looking in the wrong place |
| 21:10.15 | P-G | I'm bad with websites. ;( |
| 21:11.10 | P-G | Ok so that could be used to manage SMS. |
| 21:11.27 | [TK]D-Fender | It's "SIP MESSGING" at this point |
| 21:11.38 | [TK]D-Fender | And is provider dependent |
| 21:12.32 | P-G | Well if the system can receive SMS through GSM or whatever, it should be able to send them out as well. |
| 21:13.07 | P-G | It would be nice to pass it through SIP as well but I want users to receive SMS as they would normally, through their SMS app. |
| 21:13.21 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 21:13.42 | P-G | If it can pass through SIP also, that would be a conveinent way to message from a P.C. |
| 21:13.55 | P-G | Am I making any sense? |
| 21:14.47 | [TK]D-Fender | You sorta skipp some along the way |
| 21:15.09 | [TK]D-Fender | Receiving on an "SMS app" .... well that doesn't say what ASTERISK sends to and how. |
| 21:15.37 | [TK]D-Fender | If you are using an ITSP.. then that is pretty much a SIP MESSAGE. TEHY send it to the PSTN and that eventually reaches your end device |
| 21:15.50 | P-G | Yeah, asterisk would send the SMS out through GSM rather than SIP. That works, right? |
| 21:16.06 | [TK]D-Fender | You need to keep your thinking linear in the sense of "what does Asterisk DIRECTLY talk to and how" |
| 21:16.24 | [TK]D-Fender | If you want * to talk "direct GSM", then there are interfaces for that. |
| 21:17.26 | P-G | Yeah, hopefully I could get a GSM modem but if I can't find one I could pass it out a software defined radio. |
| 21:18.05 | P-G | GSM modem would be easier, right? |
| 21:18.33 | *** join/#asterisk caveat- (hoax@shell.bshellz.net) |
| 21:18.34 | P-G | Asterisk could pass out the SMS fully formed rather than dealing with some API. |
| 21:18.41 | [TK]D-Fender | Actually.. once you have decided what you want to send (based on whatever).. you don't need that device to be one Asterisk talks to directly. It may have a system script, etc that you can use to do the dirty work. |
| 21:18.45 | [TK]D-Fender | Which could be up to you |
| 21:19.07 | [TK]D-Fender | Depends if the interface is something * knows how to talk to. |
| 21:19.11 | [TK]D-Fender | to do that |
| 21:20.01 | P-G | Ok, well I'll definitely look into that eventually but what about passing through SIP? Do you know if that's possible to configure reliably? |
| 21:21.42 | [TK]D-Fender | if it's SUPPORTED by the other end... it's as reliable as any other SIP comm... |
| 21:21.55 | [TK]D-Fender | And needs to be supported by th4e provider |
| 21:22.05 | ayrjola | Hi, |
| 21:22.24 | P-G | Yeah, ok, that makes sense. But any support would be the right support? |
| 21:22.34 | P-G | No weird compatability issues to be warry of? |
| 21:23.33 | [TK]D-Fender | This is SIP.....YMMV |
| 21:23.52 | [TK]D-Fender | Wierd comes bundled with the solution ;) |
| 21:25.31 | P-G | Yeah, I was afraid of that. :p |
| 21:25.46 | [TK]D-Fender | That's what we call "testing" |
| 21:25.59 | P-G | Ok, I'll pick up the book and go for it. :D |
| 21:26.00 | [TK]D-Fender | Displace ass. Test :) |
| 21:26.13 | P-G | Thanks for all your help. |
| 21:26.32 | [TK]D-Fender | You're just going to have to get your hands dirty to see how the pieces can work for you |
| 21:26.32 | ayrjola | Is there somewhere in asterisk wiki documentation how to compile pjsip to support more concurrent calls? Or is there some reason to limit pjsip with asterisk? |
| 21:27.11 | [TK]D-Fender | ayrjola: it'd be more of an * efficiency issue with the stack than a hard "limit" |
| 21:27.12 | P-G | Yeah, I just wanted to make sure I wouldn't be investing a couple weeks of my life into something that wouldn't even be viable. I think this will work out well. |
| 21:27.36 | [TK]D-Fender | P-G: We don't knwo your full big picture so it's harder to tell from these sparse bits |
| 21:28.33 | P-G | I mean, nobody ever knows the FULL big picture. I think my questions were the important ones, the other problems that crop up should be manageable. |
| 21:28.59 | P-G | At the very least, next time I come here looking for help I'll have read that book. :p |
| 21:29.09 | *** join/#asterisk frek818 (~frek818@172.56.14.15) |
| 21:29.11 | [TK]D-Fender | Don't bet on that. This is Asterisk ... details KILL ;) |
| 21:29.23 | ayrjola | [TK]D-Fender thanks, huoh.. so back to chan_sip if I need test more than couple of hundred concurrent calls? |
| 21:29.36 | P-G | Ugh, we'll see. |
| 21:29.48 | P-G | Anyway, I've got to go. Thanks a lot! |
| 21:29.57 | [TK]D-Fender | ayrjola: Well I don't have practical experiencee with PJSIP but if you're running into load issues, it should be jsut that, and not a "limit". |
| 21:30.13 | P-G | I definitely owe you one. |
| 21:30.34 | [TK]D-Fender | ayrjola: if you've got some "real world" suage issues and want to see what the rest of the community is experiencing and working on check the user-lists |
| 21:30.45 | [TK]D-Fender | P-G: You're welcome. |
| 21:33.37 | [TK]D-Fender | and .... it's checkout time, BBIAB. |
| 21:33.42 | [TK]D-Fender | heads home |
| 21:33.50 | ayrjola | [TK]D-Fender I keep hittng pretty solid wall on 200 (+-2) concurrent calls. But for now its just for testing with other sip system and I have done this previously with chan_sip and wanted to test new pjsip stack. I will have to play more with pjsip some other time, need to get this test finished in couple of days. Thanks anyway. |
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| 21:59.44 | linagee | what is the highest level of (peering?) with SMS? Is it too cost prohibitive to try and do this myself? |
| 22:00.09 | *** join/#asterisk caveat- (hoax@shell.bshellz.net) |
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| 22:11.18 | linagee | ah. SMPP... |
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| 22:26.46 | xochilpili | anyone familiar with sipml5 and asterisk 13.1? |
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| 23:03.44 | johnny_|_ | i used simpl5 but never with 13.1 |
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| 23:22.07 | lf94 | Hey, I have a device that calls a japanese phone number and want to intercept the call with my computer's modem. Is this possible with asterisk? |
| 23:22.33 | lf94 | To be more specific the device is a 64 disk dive modem cartridge that calls out. |
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