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| 01:02.13 | smkelly | waits for it |
| 01:04.10 | smkelly | file: . |
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| 01:34.38 | ahklerner | http://pastebin.com/suw4iSSY |
| 01:35.20 | ahklerner | i am really new, does that tell anyone why i can not call out |
| 01:35.58 | ahklerner | or what i need to look at |
| 01:37.59 | Penguin | That extension appears to be doing exactly what it was programmed to do. |
| 01:38.43 | ahklerner | i am trying to get google voice setup on a raspberry pi, i am able to call my gv number and the sip phone rings, but the above is in the logs when i try to call out |
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| 01:42.27 | ahklerner | http://pastebin.com/RYTDPU2U <--- i would like this to be handling it |
| 01:42.31 | Penguin | The extension executed properly, so what did you expect it to do that it didn't do? |
| 01:44.22 | Penguin | I'm also not familiar with this gvoice command. What is your asterisk version? |
| 01:46.41 | ahklerner | asterisk is 11.11.0 |
| 01:48.41 | ahklerner | i am going to see if the freepbx might be famial with it |
| 01:48.59 | Penguin | Aside from what the command does, I can help you with extension matching and dialplan progression. |
| 01:49.34 | Penguin | What I saw didn't look like it involved FreePBX. |
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| 01:52.35 | gnudna | Hi Guys |
| 01:55.03 | gnudna | hey guys does this signify a problem when exiting the cli "Asterisk cleanly ending (0)." |
| 01:55.36 | gnudna | i used to set debug and verbose when doing testing and after exiting the cli debug would return to 0 and so did verbose |
| 01:56.02 | gnudna | i noticed since yesterday verbose goes back to 0 but debug stays at the level i set it at before i exited the cli |
| 01:57.53 | Penguin | Check asterisk.conf to see if you set your debug level to something other than 0. |
| 01:58.57 | gnudna | debug is commented out |
| 01:59.08 | gnudna | also i never actually touched this file |
| 01:59.51 | gnudna | maybe i just never paid attention to it but im pretty sure i just noticed this yesterday |
| 02:00.06 | gnudna | recreated the astdb in case it was corrupted |
| 02:00.09 | gnudna | but no change |
| 02:02.01 | gnudna | nothing we did yesterday triggered it is for sure as i tried running the config before adding the blacklist stuff and i see no errors just wondering why i am seeing it is all |
| 02:02.11 | gnudna | and by it i mean this behaviour |
| 02:02.34 | gnudna | brb in a few my show started ;) |
| 02:03.53 | Penguin | I don't know. If you don't want core debug on, turn it off. Hint: you probably won't ever need core debug enabled. |
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| 04:05.13 | gnudna | Hi Penguin yeah debug is disabled and be default set to off when i connect with cli was just thinking something was weird but it seems that is the behaviour |
| 04:05.38 | gnudna | thanks for the info and with that said back to my tv marathon ;) |
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| 04:51.17 | fling | Which usb modem to buy for a gsm bridge? |
| 04:51.24 | fling | is there something better than e1550? |
| 04:52.55 | fling | Which sip provider to use for cheap calls to +592 ? |
| 04:55.49 | Penguin | ~itsp |
| 04:55.49 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
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| 05:10.09 | gnudna | fling i use voip.ms there pretty good and you can check the rates before you signup |
| 05:12.38 | fling | gnudna: rate/minute = 0.2727 |
| 05:12.50 | fling | gnudna: is it in usd? |
| 05:13.47 | gnudna | yes |
| 05:13.52 | fling | not cheap :< |
| 05:14.04 | fling | 0.2 eur/minute with sipnet.ru |
| 05:14.40 | gnudna | but keep in mind they bill i 6 second increments and not per min |
| 05:15.22 | gnudna | so a 12 second call is 0.2727/12 and not the full 0.2727 |
| 05:15.44 | fling | gnudna: ok, thanks ;> |
| 05:15.56 | gnudna | best check different providers to get a decent rate for the area in question |
| 05:16.27 | gnudna | i for one call us/can/euro and im ok with the rates for now anyways;) |
| 05:17.31 | gnudna | i for one am new to voip as well so i am sure there might be better providers |
| 05:18.06 | gnudna | top of my head i have heard of anveo voip.ms and callcentric as propular choices in threads |
| 05:20.17 | fling | gnudna: I use zadarma.com and sipnet.ru |
| 05:20.33 | fling | gnudna: and domru.ru |
| 05:21.00 | gnudna | how much for a DID? |
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| 05:23.47 | gnudna | yeah i guess providers geographically closer makes more sense |
| 05:27.02 | fling | gnudna: zadarma.com gives a free did |
| 05:27.29 | fling | gnudna: and domru.ru too but there are restrictions⦠|
| 05:27.51 | gnudna | do they give european DID? |
| 05:28.10 | fling | idk but I have +7 from them |
| 05:28.21 | gnudna | will have to look into it |
| 05:28.53 | gnudna | could use a european maybe i could setup a sip -> sip and et free calls that way |
| 05:29.07 | fling | +7 3852 XXXXXX from domru.ru and +7 9XX XXX XXXX from zadarma.com |
| 05:29.11 | gnudna | wants to break his keyboard |
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| 05:29.33 | fling | try qgmlwb :> |
| 05:29.42 | fling | carpalx rules! |
| 05:30.15 | fling | gnudna: http://mkweb.bcgsc.ca/carpalx/?popular_alternatives |
| 05:30.34 | fling | What is the modern way to call skype users and receive calls from them? |
| 05:30.42 | fling | I don't like my running siptosis instance |
| 05:30.53 | fling | and I don't want to install freeswitch for using skype mod ;< |
| 05:32.06 | gnudna | would not know i do not use it |
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| 05:41.20 | gnudna | ok guys off for the night |
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| 05:44.31 | CRCinAU_ | hmmm - who do I bug about the asterisk binary packages being out of date? |
| 05:44.36 | CRCinAU_ | ie only having 11.1 vs 11.2 |
| 05:44.49 | CRCinAU_ | errr make that 11.11 vs 11.12 ;) |
| 05:45.06 | fling | CRCinAU_: your distro |
| 05:45.59 | CRCinAU_ | http://packages.asterisk.org/centos/6/ |
| 05:46.04 | CRCinAU_ | really? o_O |
| 05:46.34 | CRCinAU_ | wait - sorry - just understood how my question could be read two ways ;) |
| 05:46.49 | CRCinAU_ | hmmm - who do I bug about the asterisk provided binary packages being out of date? |
| 05:46.52 | CRCinAU_ | that's a bit better. |
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| 05:52.21 | CRCinAU_ | also looks like dahdi-firmware is missing - but required for those packages :( |
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| 06:41.37 | ruben23 | hi guys any idea what is the asterisk compatible version for a2billing |
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| 07:41.43 | fling | I'm about to run asterisk inside of an lxc container |
| 07:42.01 | fling | Is there a good networking solution without using nat? |
| 07:42.02 | ChannelZ | Quick, someone call a photographer! |
| 07:42.10 | fling | ChannelZ: I'm a photographer ;P |
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| 07:42.49 | shido6 | ruben23: im running 11.11.0 |
| 07:43.04 | shido6 | but im also running it with FreePBX 2.11.0.38 |
| 07:48.26 | fling | is rebooting to enable namespaces support ;< |
| 07:50.46 | fling | https://en.wikipedia.org/wiki/File:Hands-Fingers-Crossed.jpg |
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| 08:01.12 | fling | yay! |
| 08:01.37 | fling | ChannelZ: ok so now I want to know about preferred networking scheme |
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| 10:12.48 | grzew | Hi, how can I run script after someone hungs up in asterisk 1.8? |
| 10:13.31 | kleszcz | h,1,.. ? |
| 10:15.34 | grzew | kleszcz: When I am doing: h,1,NoOp("TEST") it seems to be not executed |
| 10:16.23 | grzew | I am trying to do this inside macro |
| 10:17.26 | kleszcz | dziwne u mnie dziala ;) |
| 10:18.00 | grzew | #SOA1 ? :D |
| 10:18.21 | grzew | Może dlatego że robiÄ to w macro? |
| 10:20.22 | grzew | wczeÅniej mam => s,n,coÅtam |
| 10:20.36 | grzew | a potem => h,1,NoOp("TEST") |
| 10:20.50 | grzew | I na konsoli widzÄ tylko: -- User disconnected |
| 10:21.25 | grzew | w |
| 10:21.44 | kleszcz | a czy masz na koncu s,n,Hangup() ? |
| 10:22.05 | grzew | Tak, ale mi chodzi o sytuacjÄ w której mam tak: |
| 10:22.18 | grzew | => s,n,MeetMe(asdasdadas) |
| 10:22.36 | grzew | no i jak siÄ rozÅÄ
czy odkÅadajÄ
c sÅuchawkÄ to chcÄ wywoÅaÄ skrypt |
| 10:29.55 | kleszcz | chwila jeszcze sprawdzam u siebie |
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| 10:36.17 | kleszcz | jak na razie dupa, nie ukrywam ze nie wykonywalem dla meetme akcji |
| 10:36.25 | kleszcz | zw ide zapalic |
| 10:38.19 | grzew | ok |
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| 11:36.39 | grzew | kleszcz: jesteÅ? |
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| 12:17.41 | Stefan27 | I've been looking at the Queue() dialoptions. Option C does indeed mark call as answered elsewhere always... but is there any way to mark the call as answered elsewhere only if the call stops ringing because of this ring-cycle's timeout (~30 seconds my case) but not when the user calling the queue chooses to cancel? |
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| 12:22.12 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:22.22 | crised | Normal GSM provider do charge on minute basis? (instead of some ITSP providers that do charge every 6 seconds) |
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| 12:54.12 | crised | When I measure latency to ITSP, this latency won't be the real call latency when I do a termination call |
| 12:54.39 | crised | the obvious path would be, from voip softphone, to the ITSP server, THEN to the termination gateway |
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| 12:56.02 | crised | so if I'm in Chile, I call to a chilean PSTN, and the server is located in the US, the bits of the call will travel to: 1) To the US ITSP Server 2)back from the ITSP server to the Chile gateway |
| 12:56.10 | crised | seems like an awful long trip |
| 12:56.38 | crised | if I already have 130 ms, pinging to the PSTN server, the real latency would be twice as that? |
| 12:57.10 | crised | In the other hand, would the call be directed from voip softphone to the termination gateway directly? |
| 12:57.20 | crised | [TK]D-Fender: :) |
| 12:57.28 | WIMPy | ping is not a measure of rtp latency. |
| 12:57.59 | WIMPy | No it wouldn't, as the NSA wants a copy of the data. |
| 12:58.23 | crised | WIMPy: the jitter? |
| 12:58.30 | crised | is that the measure of rtp latency? |
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| 12:58.55 | WIMPy | No, you have to use it to find out the latency. |
| 12:59.18 | crised | WIMPy: so an ITSP located in USA must route all their calls through their USA servers? |
| 12:59.27 | WIMPy | Jitter can add to the latency if jitterbuffers are in use. |
| 12:59.51 | crised | WIMPy: how do I find out the latency? just by making a phone call? |
| 12:59.57 | crised | that's a human way to do it |
| 13:00.42 | WIMPy | Yes, use it and you can measure it. |
| 13:00.59 | crised | WIMPy: NSA can still have a copy at the termination gateway of the digital data |
| 13:01.35 | crised | WIMPy: I want to have aan objective measure, not a human measure of the call, how is that done? |
| 13:03.07 | WIMPy | I don;t know how much rtcp tells you or how reliable it is. Google probably konws tools to measure latency. |
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| 13:04.08 | crised | WIMPy: How can I do a "traceroute" of the trip that the calls make? |
| 13:04.12 | crised | [TK]D-Fender: ^ |
| 13:05.04 | [TK]D-Fender | crised: GOOGLE <- |
| 13:06.13 | crised | [TK]D-Fender: (09:56:37) crised: In the other hand, would the call be directed from voip softphone to the termination gateway directly? |
| 13:08.34 | [TK]D-Fender | crised: You are making assumptions about where the termination gateway is even LOCATED |
| 13:09.26 | crised | [TK]D-Fender: I do make that assumption because their rate is super low, so the cal should be local in PSTN, Am I wrong? |
| 13:09.49 | [TK]D-Fender | crised: Still an assumption |
| 13:10.06 | [TK]D-Fender | crised: There is nothing in what you've said that proves anything |
| 13:10.22 | crised | [TK]D-Fender: What I'm asking is: Do calls NEED to pass through ITSP voip server? |
| 13:10.42 | crised | or it might give endpoints a more direct route between them |
| 13:11.14 | [TK]D-Fender | crised: What are you talkinga about? What endpoints? |
| 13:12.04 | crised | Voip Endpoint 1: me making a phone call, Endpoint 2: Termination Gateway |
| 13:12.38 | crised | in simplistic terms |
| 13:12.51 | crised | is VoIP peer to peer? or it must always pass through a server |
| 13:12.57 | crised | *the voip server |
| 13:13.05 | crised | *the ITSP server |
| 13:13.27 | crised | [TK]D-Fender: btw, I'm reading the book |
| 13:14.32 | [TK]D-Fender | crised: SIP can pass through SIP Routers before reaching a point where it will hit a gateway device. It DEPENDS. |
| 13:15.24 | [TK]D-Fender | crised: As for re-inviting endpoint s to communicate directly, pretty much no ITSP will have you go direct, usually the media will be told to go direct, but SIP will still go through their chain. |
| 13:16.20 | crised | voip.ms -> Â The calls will always pass through our server. |
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| 13:17.30 | [TK]D-Fender | What calls? You are not properly describing the entire chain that you're referring to. |
| 13:19.58 | crised | [TK]D-Fender: sorry, need to leave. |
| 13:20.04 | crised | :( |
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| 13:47.04 | c|oneman | k seriously, bria has 1 ringer. |
| 13:47.11 | c|oneman | the most awful sounding, intense one ever. |
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| 14:44.54 | crised | [TK]D-Fender: usually the "media" goes direct? I would imagine so, otherwise it will be making unnecessary trips |
| 14:45.19 | [TK]D-Fender | lok at your call. |
| 14:45.38 | crised | ? |
| 14:46.14 | [TK]D-Fender | Look at your call. |
| 14:46.58 | [TK]D-Fender | If you see them offering a different IP for media vs signalling then that pretty much confirms it |
| 14:47.33 | [TK]D-Fender | but their internal side shouldn't normally matter. |
| 14:47.57 | [TK]D-Fender | 'Chances are that wherever their SIP server is would be the same as where their PSTN conenctivity is. |
| 14:49.12 | file | it may or may not |
| 14:49.51 | crised | [TK]D-Fender: I imagine that "look at your call" means see what your ITSP offers |
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| 14:50.10 | [TK]D-Fender | I doubt any respectable ITSP would have a alrge amount of latency for their end of the link regardless |
| 14:50.30 | crised | [TK]D-Fender: yes, if the server is located in Tampa, then I think you're right, the PSTN termination would be right there |
| 14:52.25 | crised | [TK]D-Fender: Does the caller id of the phone number indicates where the PSTN termination is located? |
| 14:53.02 | [TK]D-Fender | no |
| 14:53.16 | [TK]D-Fender | crised: You are thinking too much for nothing |
| 14:53.23 | crised | [TK]D-Fender: that's me |
| 14:53.25 | crised | :) |
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| 15:08.58 | crised | Which SIP application would you advise for android? |
| 15:09.05 | anonymouz666 | zoiper |
| 15:09.13 | crised | ok |
| 15:09.24 | anonymouz666 | enjoy |
| 15:09.48 | crised | :-D |
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| 15:14.30 | crised | [TK]D-Fender: Please look at this map. Let's say I choose the server in Tampa, and let's assume that the PSTN termination is there. If I will be making calls to England, then voip.ms pstn carrier's bill will be larger if I choose the london server, Am I right? |
| 15:14.34 | crised | http://wiki.voip.ms/w/images/c/c1/Serverlocation3.png |
| 15:15.21 | crised | Perhaps, isn't there some sort of ip forwarding to the London server? |
| 15:17.30 | crised | *smaller |
| 15:17.42 | [TK]D-Fender | entirely possible |
| 15:18.10 | [TK]D-Fender | All of this is poijntless conjecture. just TRY them and see |
| 15:18.14 | crised | so the algorithm should be, use the PSTN termination as close as possible |
| 15:18.40 | crised | [TK]D-Fender: it's not pointless, I'm trying to understand the business |
| 15:18.55 | crised | [TK]D-Fender: I have tried it, but how am I supposed to view the route of the call? |
| 15:18.56 | [TK]D-Fender | fast, good, cheap. Pick TWO.\ |
| 15:19.19 | crised | good and cheap |
| 15:19.21 | crised | :) |
| 15:19.53 | [TK]D-Fender | Sure thing. Your expected deliver is in 20 years |
| 15:19.57 | [TK]D-Fender | +y |
| 15:20.29 | crised | [TK]D-Fender: Which would be the equivalent of "traceroute" in a voip call? |
| 15:20.47 | crised | or at least something similar |
| 15:22.40 | [TK]D-Fender | You may not even get to see the actual path. |
| 15:22.45 | [TK]D-Fender | Stop assuming this means anything |
| 15:22.56 | [TK]D-Fender | this is largely a waste of time neurosing about nothing |
| 15:23.01 | Penguin | I use mtr in udp mode for the best chance. |
| 15:23.05 | [TK]D-Fender | go try some calls and use your EARS |
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| 15:24.16 | anonymouz666 | anyone already saw this: [Sep 8 12:16:32] WARNING[2375][C-00000019]: res_srtp.c:406 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 110 |
| 15:24.17 | anonymouz666 | ? |
| 15:24.25 | anonymouz666 | SRTP with one way audio Asterisk 11.6 |
| 15:24.33 | anonymouz666 | SIP TLS handshake is ok |
| 15:25.00 | crised | Penguin: mtr in udp that's to try the ITSP server? |
| 15:25.07 | crised | [TK]D-Fender: yes, I understand your point. |
| 15:25.31 | Penguin | That's the best chance I have at seeing the path the voip packets MIGHT take. |
| 15:26.08 | crised | Penguin: assuming that the PSTN termination is there |
| 15:26.18 | Milenco | Hmm..when using Asterisk call-files, is it possible to create a failover somehow? |
| 15:26.33 | Penguin | You won't ever see where the call is actually terminated using packets. |
| 15:26.45 | Penguin | Once it hits their equipment, you can't see anything past that. |
| 15:26.46 | crised | Penguin: ok |
| 15:26.51 | crised | ok |
| 15:26.56 | crised | thanks |
| 15:26.58 | Milenco | I can set a Channel but would like to set it to a macro which handles failover provider (with either another SIP provider or a landline) |
| 15:27.06 | Penguin | "best chance" |
| 15:27.10 | crised | ok |
| 15:27.37 | crised | thanks, bye for now |
| 15:27.54 | [TK]D-Fender | [11:26]PenguinYou won't ever see where the call is actually terminated using packets. <- it could be possible depending on the scenario if you get passed the media address of the actual gateway. NO bets on this however. It's likely to be at least a media gateway just the same in front of the actual PSTN hardware |
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| 15:38.54 | WIMPy | Milenco: Use a local channel and dialplan. |
| 15:39.51 | Milenco | Thanks, I just found out whenever a call fails it jumps to 'failed' in the same context |
| 15:39.58 | Milenco | I can probably use that :) |
| 15:40.16 | tinklebear | asterisk should implement hierarchichal contexts...that'd be pretty cool. |
| 15:44.55 | [TK]D-Fender | tinklebear: as in? |
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| 15:48.21 | WIMPy | Sounds like includes to me. |
| 15:49.59 | Milenco | Hmm, just to be sure: s-ANSWER,1,Playback(hello-world) starts exactly when a call is picked up, right? |
| 15:50.52 | WIMPy | If you had an option in your dial that would make it do so, yes. |
| 15:51.18 | WIMPy | But not by default or anyhting the like. |
| 15:51.46 | [TK]D-Fender | Milenco: we have no idea what happens before that line. |
| 15:51.53 | [TK]D-Fender | 'mileSeeing it alone like that proves nothing |
| 15:52.30 | Milenco | No your right, let me rephrase |
| 15:52.42 | [TK]D-Fender | Milenco: PASTEBIN is your friend... |
| 15:52.43 | [TK]D-Fender | ~pb |
| 15:52.44 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:52.45 | [TK]D-Fender | ^^^ |
| 15:52.46 | Milenco | s-ANSWER works in the same way, for example, s-BUSY works, right? |
| 15:53.20 | Milenco | I wanna make a failover for a callfile. If that one fails it jumps to "failed" in the same context |
| 15:53.28 | WIMPy | Milenco: Same again: Inly if you configured something. |
| 15:53.44 | Milenco | There I want to try again and whenever the call _has_ worked I want to redirect to where it should be going in the first place |
| 15:54.05 | [TK]D-Fender | [11:52]Milencos-ANSWER works in the same way, for example, s-BUSY works, right? <- that is just the name you gave that extension. You could just as easily have called it FRED |
| 15:54.33 | Milenco | 'failed' will try to Dial() again and jump to s-${DAILSTATUS} |
| 15:54.33 | [TK]D-Fender | Milenco: The name there proves nothing about why it is called what it is necessarily, nor describes what may have been prcessed earlier |
| 15:54.44 | [TK]D-Fender | Milenco: Your method for thry to break this down is faulty |
| 15:54.53 | [TK]D-Fender | Milenco: PASTEBIN <--------- |
| 15:54.58 | [TK]D-Fender | trying* |
| 15:55.28 | Milenco | I don't have code yet, so there not much to paste :) |
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| 15:55.43 | [TK]D-Fender | Milenco: Well that one line means nothing |
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| 15:56.17 | [TK]D-Fender | Milenco: You can't show us a single bolt and then ask us about details about how the plane flies based on it. |
| 15:56.27 | Milenco | haha no your right |
| 15:57.01 | WIMPy | Milenco: If you don't have "code" yet, nothing of what you wrote will happen. |
| 15:57.30 | [TK]D-Fender | "Here's a paint chip off my passenger door, now why doesn't the eject button on my CD player work?" |
| 15:58.32 | Milenco | haha |
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| 16:00.40 | Milenco | Thanks anyway, I think I can figure out a working solution for my issue :) |
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| 16:07.07 | tinklebear | [TK]D-Fender [A][1][/1][2][a][/a][/2][/A] [B][1][/1][2][/2]][/B] Context: A.1,A.2.a Context: B.1,B.2 |
| 16:08.13 | [TK]D-Fender | tinklebear: Not sure I get the point... |
| 16:08.37 | WIMPy | I don't get it other than you can just name your contexts that way if you want to. |
| 16:10.50 | tinklebear | Inside contexts could have permissions in outer contexts but not vice versa--much like scope in many programming languages. So you could have several components within an organization that would be clearly defined for the whole organization and easily understand the permission structure. |
| 16:11.18 | [TK]D-Fender | contexts can already include other contexts.... |
| 16:11.31 | [TK]D-Fender | and are not bi-driectional |
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| 16:12.02 | tinklebear | The inside contexts have permissions in the outer contexts? |
| 16:12.41 | [TK]D-Fender | http://pastebin.com/v6j9qcn3 |
| 16:12.42 | tinklebear | ...anyhow just a thought. Likely it's shit. I didn't know you could have nested contexts. |
| 16:12.47 | WIMPy | Your wording is inappropriate. Contexts don;t have permissions. |
| 16:12.48 | [TK]D-Fender | a has access to 12 & a2 |
| 16:12.58 | tinklebear | wimpy...exactly |
| 16:13.00 | tinklebear | that si my point |
| 16:13.04 | [TK]D-Fender | contexts ARE permissions |
| 16:13.07 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 16:13.35 | WIMPy | And includes are hirachical. |
| 16:13.41 | [TK]D-Fender | yup |
| 16:14.03 | [TK]D-Fender | You could make a [b] that includes "a" ... and other stuff |
| 16:14.39 | tinklebear | Can't you just jump to another context completely without that include? |
| 16:15.20 | [TK]D-Fender | http://pastebin.com/kMWzFDPB |
| 16:15.34 | [TK]D-Fender | tinklebear: Where do you get this concept of "jumping" from? |
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| 16:17.01 | WIMPy | You can Goto whereever you want. |
| 16:17.09 | tinklebear | exactly |
| 16:17.18 | tinklebear | so it's not permission based at all |
| 16:17.22 | tinklebear | it's just a search thing |
| 16:17.30 | [TK]D-Fender | You have permissions based on where you jump to. |
| 16:17.53 | [TK]D-Fender | and its you code in where they ARE that can allow them to go somewhere else |
| 16:17.54 | WIMPy | I already said that contexts don't have permissions. |
| 16:18.14 | [TK]D-Fender | It's up to you to make heirachy's that make sense |
| 16:18.19 | tinklebear | "<[TK]D-Fender>contexts ARE permissions" |
| 16:18.51 | [TK]D-Fender | tinklebear: in terms of what you can do while in there. |
| 16:19.08 | tinklebear | gotcha...thanks |
| 16:19.17 | tinklebear | I told you it was probably a shit idea. :) |
| 16:19.30 | WIMPy | But not what the writer of the dialplan can do. He can do everything. |
| 16:19.32 | [TK]D-Fender | tinklebear: over-thinking. |
| 16:20.38 | tinklebear | Not at all...I'm just trying to turn something that works well into something that doesn't work as well. :) |
| 16:21.26 | jeev | anyone ever use a Polycom SoundStation2W with reviews? wireless/battery conference phone ? |
| 16:21.55 | [TK]D-Fender | jeev: Works well. Battery lasted several years. |
| 16:22.08 | [TK]D-Fender | jeev: I should have replaced mine long ago however |
| 16:22.28 | jeev | really, thanks fender |
| 16:22.42 | jeev | i have one somewhere in the garage but i don't have the other end for it. |
| 16:22.46 | jeev | i have the main unit |
| 16:23.17 | jeev | if i buy a base station separate, it'll match up? it's dect.. |
| 16:23.36 | [TK]D-Fender | I don't recall it being DECT... |
| 16:23.59 | WIMPy | And DECT only defines the radio part, no signalling. |
| 16:24.23 | WIMPy | GAP defines basic interoperability. |
| 16:24.28 | jeev | 1.8 GHz DECT, 1.9 GHz DECT 6.0, |
| 16:24.28 | jeev | or 2.4 GHz WDCT radio interface to |
| 16:24.28 | jeev | base station |
| 16:24.30 | jeev | wpoops sorry |
| 16:24.40 | WIMPy | But only as far as being able to dail, accept a call and hang up. |
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| 17:12.18 | marceloamorim | guys, anyone knows why I set on modules.conf noload => res_smdi.so and my /var/log/asterisk/messages keeping warning me that I can`t load the config smdi.conf? |
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| 18:47.01 | jwr__ | If I want to quickly empty a voicemail box, can I just rm /var/spool/asterisk/voicemail/default/111/INBOX/*? I tried that but apparently the mailbox is still full. Is there some other way to accomplish that? |
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| 19:13.36 | pecanha | Hello guys, I'm deploying asterisk here for the first time. I've installed using apt-get install asterisk 1.8.x ( onubuntu 12.04 server). Just wondering, anyone using it? Is that stable enough? Or should I be looking for the newer versions? |
| 19:14.29 | WIMPy | It's rather old. |
| 19:15.09 | pecanha | WIMPy: I see |
| 19:15.32 | pecanha | WIMPy: Do you recommend a distro for it? Or should I compile from source? |
| 19:16.05 | WIMPy | I prefer to compile. |
| 19:16.16 | TazzNZ | hey all |
| 19:18.14 | pecanha | I did create few sip/extension (nat = yes, all behind nat except the server) for testing purposes, I'm trying GSM codec (by now the one that worked with our softphones zoiper/ekiga), sometimes I can hear both side of voices, always can hear hold music, but sometimes it works and sometimes it don't. It can be some configuration, where should I look for? |
| 19:19.11 | pecanha | Note: can't see errors in cli, both are registering fine as expected, but I get that behaviour |
| 19:22.03 | lvlinux | sounds like NAT issues to me. |
| 19:22.33 | pecanha | lvlinux: that's my feeling too, however, I'm a bit lost where should I look for |
| 19:24.41 | lvlinux | pecanha: first check your router and make sure no form of SIP ALG is enabled. |
| 19:24.58 | *** join/#asterisk TazzNZ (~TazzNZ@office-nat1.ytechops.co.nz) |
| 19:28.45 | pecanha | lvlinux: I'll do that |
| 19:32.40 | *** join/#asterisk e4voip (uid13742@gateway/web/irccloud.com/x-uertgibhijbpxmpb) |
| 19:35.55 | *** join/#asterisk crised (~crised@186.67.181.203) |
| 19:39.53 | pecanha | lvlinux: indeed my router was with few options ALG enabled, however, even disabling it, still not working fine |
| 19:43.19 | crised | Does normally SIP gets transported in TCP? |
| 19:44.16 | malcolmd | udp, though tcp and tls are other common transports |
| 19:45.07 | crised | ok |
| 19:45.36 | lvlinux | pecanha: ok well leave it off for sure!!!!! it will only cause you problems. |
| 19:45.48 | lvlinux | pecanha: now that that is taken care of you can move to the next step. |
| 19:46.34 | lvlinux | pecanha: When you look on your console, does the call go through properly even when you hear no audio? |
| 19:46.54 | lvlinux | Check to make sure it does, then if so, run a SIP debug and see what you find there. |
| 19:47.32 | pecanha | lvlinux: yes it goes fine |
| 19:47.43 | pecanha | let me try |
| 19:47.46 | lvlinux | k |
| 19:51.08 | *** join/#asterisk puzzled_ (~patrick@puzzled.xs4all.nl) |
| 19:55.36 | pecanha | lvlinux: should I look for any specific error? |
| 19:59.20 | pecanha | lvlinux: pasted http://pastebin.com/AT0PN6Jt |
| 20:02.25 | lvlinux | i'll ck it---fixin to leave tho---so might not be able to respond till later maybe. |
| 20:04.04 | crised | [TK]D-Fender: from asterisk book: once phones have started the call, they communicate directly with each other. |
| 20:04.11 | crised | This was the missing piece in my mind puzzle |
| 20:04.46 | [TK]D-Fender | crised: that statement is not complete |
| 20:04.59 | [TK]D-Fender | crised: It depends on a LOT of things |
| 20:05.05 | lvlinux | pecanha: have you checked it with only ULAW on both ends? |
| 20:05.13 | lvlinux | gtg....bbl... |
| 20:05.18 | pecanha | lvlinux: yes |
| 20:05.37 | pecanha | the only I could hear on both sides was GSM |
| 20:05.44 | crised | [TK]D-Fender: (if possible) |
| 20:05.54 | crised | [TK]D-Fender: but I get the point |
| 20:05.58 | pecanha | ulaw could hear on one side |
| 20:06.16 | crised | [TK]D-Fender: "so that data doesn't have to tie up the resources of the proxy" |
| 20:06.44 | crised | [TK]D-Fender: so obviously, the route be as direct as possible, between 2 phones |
| 20:09.13 | lnb | how can the time be shortened for going from trunk0 to trunk1 on outbound calls |
| 20:12.36 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-ujsspsymfjxyhhak) |
| 20:12.36 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 20:13.08 | navaismo | what time? |
| 20:13.23 | navaismo | in what case? |
| 20:13.38 | lvlinux | pecanha: so gsm always worked both sides? |
| 20:14.27 | pecanha | not always |
| 20:15.55 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.hsi01.unitymediagroup.de) |
| 20:18.00 | lnb | navaismo: make a call, trunk 0 is unreachable, it looks like it retransmits 6 times and it takes about 30 or so seconds before it tries trunk 1, and that is enough time so the phone has CALL TIMEOUT and hangs up, but freepbx (asterisk) starts at that point using trunk1 and it completes the call, however the endpoint the caller used has hungup |
| 20:19.20 | navaismo | uh? usually after a busycongested i goes to the next priority inmediately |
| 20:19.21 | *** join/#asterisk huatou (~huatou@107.170.182.241) |
| 20:19.32 | lnb | one solution is to change the Session Timer T1 from 0.5 to 2 on the phone, but in an office with lots of phones its a real pain. Was hoping there is something I can do in asterisk to change it |
| 20:20.14 | lnb | navaismo: i am talking about when a itsp is down, it doesn't send any replies at all to the INVITE |
| 20:20.58 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 20:21.06 | lnb | navaismo: i have tested this out on yealink phones, and changing the Session Timer does work and the calls are completed |
| 20:21.12 | navaismo | did you qualify the trunk? |
| 20:21.21 | lnb | ? |
| 20:21.34 | navaismo | qualify=yes |
| 20:22.22 | lnb | there is no line like that in fbpx for a trunk |
| 20:22.51 | navaismo | amm ususally you wrote the settings right? so you can add it |
| 20:23.19 | lnb | or do you mean in the Peer Details? |
| 20:23.30 | navaismo | yep there |
| 20:23.31 | lnb | ok I think you mean to add it there |
| 20:23.45 | lnb | so qualify=yes will do ? |
| 20:24.47 | *** join/#asterisk jlnt (~lankthata@107-143-2-51.lightspeed.rcsntx.sbcglobal.net) |
| 20:24.57 | navaismo | "Check if client is reachable. If yes, the checks occur every 60 seconds. Default no." |
| 20:25.32 | lnb | ok changed host= to wrong ip address and now will try a call and see after adding qualify=yes |
| 20:25.54 | lnb | wow |
| 20:25.59 | lnb | it worked! |
| 20:26.05 | lnb | THANK YOU NAVAISMO |
| 20:26.41 | *** join/#asterisk jkroon (~jkroon@uls-154-73-32-14.wall.uls.co.za) |
| 20:27.01 | navaismo | usually you dont want to qualify trunks, but in some cases is good to know if its down |
| 20:27.44 | *** join/#asterisk seanzh (~sean.haye@wsip-70-184-118-104.ph.ph.cox.net) |
| 20:28.01 | jkroon | hi guys, using ast 11(.10.2) - in some scenarios I want to log the call before playing back a message to the caller, eg, a sequence like ResetCDR(w); NoCDR(); Playback(...); Hangup() |
| 20:28.50 | jkroon | the majority of cases where this happens it's because of an upstream error (eg, 404, so hangupcause 1 etc ...), however, if I do the above sequence it always logs as NO ANSWER |
| 20:29.43 | jkroon | I'd expect FAILED in most of the cases where I do that ... |
| 20:30.05 | lnb | navaismo: why would you not want to qualify trunk? |
| 20:30.16 | seanzh | Hi All, I am using res_fax_digium and I'm having an issue receiving faxes. Is there a special keyword I can search for in my debug logs to identify the potential issue? |
| 20:30.28 | jkroon | am I doing something wrong, or is this just the way it works? |
| 20:35.22 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 20:40.03 | seanzh | I see Event: SoftHangupRequest but I don't know what to look for above the end of the call.. (also, I am not well versed in Fax for Asterisk.. Please let me know if this is the wrong venue for these questions) |
| 20:40.48 | *** join/#asterisk calum_ (~calum_@cpc67428-harg5-2-0-cust142.7-1.cable.virginm.net) |
| 20:44.06 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 20:44.38 | seanzh | I am seeing the following debug message: Cause-txt: Normal Clearing which translatest to: AST_CAUSE_NORMAL_UNSPECIFIED 31 |
| 20:45.13 | crised | Is SCTP widely used as transport in voiceip? Or does UDP handle most of the tasks? |
| 20:47.28 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 20:48.53 | navaismo | lnb some providers doesnt want to be "bothered" every 60 seconds |
| 20:50.14 | *** join/#asterisk cyborg-one (~cyborg-on@212-178-3-60.broadband.tenet.odessa.ua) |
| 20:51.19 | lnb | well how often does register do that? |
| 20:51.37 | lnb | as in a trunk that registers with itsp |
| 20:51.57 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:55.55 | lvlinux | crised: usually UDP, sometimes TCP |
| 20:58.00 | file | sometimes TLS |
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| 21:09.19 | navaismo | default 60 seconds, you can change it by setting the miliseconds instead the word "yes" |
| 21:09.40 | lnb | ? |
| 21:10.43 | [TK]D-Fender | not 60... |
| 21:12.27 | lnb | navaismo: i was asking you, what is the difference between qualify=yes and registering trunk with itsp in terms of data transmissions to the itsp? |
| 21:12.42 | lnb | isnt it just about the same? |
| 21:13.37 | navaismo | hi, hmmm idk but i guess ine is like a keep alive and the other is actually a register method |
| 21:13.45 | lnb | right |
| 21:13.51 | lnb | but they both contact the itsp |
| 21:14.22 | lnb | the difference i see now is, clients will be able to make calls and not have to wait for trunk0 to time out |
| 21:14.44 | lnb | if the itsp trunk0 is pointing at is down temporarily |
| 21:16.04 | lnb | now i can change the session timer t1 back to 0.5 |
| 21:16.12 | lnb | in all the phones i changed |
| 21:16.20 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:16.23 | lnb | i wish you would have told me before :) |
| 21:16.33 | lnb | but again, thanks for that tip |
| 21:20.11 | *** join/#asterisk hfp_ (~hfp@MTRLPQ0736W-LP130-02-1279512586.dsl.bell.ca) |
| 21:23.51 | fury_ | is it possible to tell asterisk what network interface to use for SIP? I know that NAT can be tricky, luckily this particular box is both on a NAT and *is* the NAT, so has a completely public IP on another interface. Trouble is, looking at 'sip set debug on', it seems to be giving out my NAT address in places. I have 'bindaddr = 0.0.0.0' in [general] in sip.conf currently. |
| 21:24.45 | fury_ | ideally I'd be able to tell it to use eth0 instead of specifying an IP, as the IP may change |
| 21:25.17 | Penguin | The only way to choose one interface is to bind to a SINGLE address on the interface you want to use. |
| 21:25.38 | fury_ | ok, cool, so what I just said which isn't really ideal. :( that sucks, but alright, thanks! |
| 21:26.01 | fury_ | isn't there a bindhost so it resolves the domain? |
| 21:26.24 | Penguin | I think I would bind it to the private interface's addr, and treat it as if it is natted. |
| 21:27.06 | fury_ | I have many issues, since the box actually has a public IP I'd like to eliminate NAT as one of them |
| 21:28.03 | Penguin | But if your public addr changes, it's hard to bind to it in a static config file. |
| 21:28.10 | fury_ | yup I agree |
| 21:28.17 | Penguin | Hence my suggestion. |
| 21:28.17 | fury_ | iirc there was a way to get it to use a hostname |
| 21:28.36 | fury_ | (which would resolve to the public IP) |
| 21:29.01 | Penguin | Even if bindaddr/udpbindaddr/tcpbindaddr/tlsbindaddr will accept an FQDN, it will only resolve it when chan_sip loads. |
| 21:29.36 | Penguin | When the DNS resolution changes, there's nothing to trigger that in chan_sip. |
| 21:30.03 | fury_ | fair enough. this is mostly just temporary anyway, but I suppose what you're saying is I really should just set up NAT properly. |
| 21:31.49 | Penguin | Asterisk is actually pretty good at working well with /most/ NATs. |
| 21:32.40 | Penguin | Now when it is behind the NAT, you'll want to use externhost and the dynamic DNS hostname you were referring to a minute ago. |
| 21:33.11 | fury_ | I'm sure it is. like I said right now I'm having other issues (login timeout to SIP provider), and I wanted to eliminate NAT as a cause as sip.conf states "SIP operation behind a NAT is tricky" |
| 21:33.26 | fury_ | so nat=yes and externhost=my.host.name and that's basically it? |
| 21:33.36 | Penguin | And directmedia=no |
| 21:34.13 | fury_ | ok cool. if you don't mind me asking, this can go in [general] yes? |
| 21:34.22 | fury_ | eg. it doesn't have to be provider-specific |
| 21:34.31 | Penguin | Those things need to be in the general section. |
| 21:34.57 | Penguin | You can also define nat and directmedia in the individual peer sections. |
| 21:35.19 | file | and localnet. |
| 21:35.34 | Penguin | Ah, right. Define localnet in the general section, too. |
| 21:35.38 | Penguin | ~sipnat |
| 21:35.38 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 21:36.00 | Penguin | "nat, directmedia, externhost or externaddr, and localnet." |
| 21:36.49 | fury_ | thanks |
| 21:37.23 | fury_ | sip set debug on, I'm still seeing: Call-ID: 296725c602e091244a7456f26c16a176@10.0.0.1 |
| 21:37.27 | fury_ | which is my internal IP |
| 21:37.30 | fury_ | I wonder if that even matters |
| 21:37.41 | Penguin | What's the actual problem? |
| 21:37.56 | fury_ | I get: chan_sip.c:13656 sip_reg_timeout: -- Registration for 'uv-2133160@conus.ucallweconn.net' timed out, trying again (Attempt #4) |
| 21:38.04 | fury_ | it works with a softphone |
| 21:38.09 | fury_ | so I don't think it's the provider |
| 21:38.15 | fury_ | I can paste the sip.conf, sec I just have to anonymize it |
| 21:39.11 | fury_ | http://www.codexterous.com/paste/results/fmIXKy51.html |
| 21:39.21 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 21:39.27 | fury_ | oh wait, maybe realm= is a problem |
| 21:39.54 | fury_ | no that doesn't seem to matter according to the config docs |
| 21:40.50 | fury_ | aside from the NAT stuff, this is taken from old configs I did years ago, which used to work |
| 21:41.04 | Penguin | Are you using pfSense? |
| 21:41.17 | fury_ | it's freebsd, so sort of, but it's a standalone box with no pfsense |
| 21:41.38 | fury_ | I use pf as a firewall if that's what you were getting at |
| 21:42.14 | Penguin | I had a feeling. |
| 21:42.31 | fury_ | why? |
| 21:42.39 | Penguin | I've seen this a lot of times and it's often when someone is trying to use asterisk on their pfSense appliance. |
| 21:43.02 | fury_ | I mean keep in mind pfsense is only related to freebsd in that it uses the same kernel, and that's pretty much it |
| 21:43.08 | fury_ | otherwise this is in no way pfsense |
| 21:43.31 | fury_ | freebsd is a complete operating system (I apologize if you know this already) |
| 21:43.41 | fury_ | where pfsense is router software |
| 21:43.52 | fury_ | regardless I've got the port forwarding etc all set up |
| 21:44.18 | Penguin | I'm aware. I'm just saying that your problem is somehow related to pf. |
| 21:45.19 | Penguin | I've never seen anyone show up and say, "Hey, I finally fixed that problem on my pfSense box, and here's how..." |
| 21:45.28 | fury_ | well, possibly, but people using pfsense usually have no idea about the internals of pf, they just use a web UI to set up firewalls and generally don't really know what they're doing. my pf config is hand written by me and has all sorts of stuff going on. I've tried passing directly on in to the main interface, forwarding to the private one, etc. I actually know pf |
| 21:45.53 | fury_ | plus I've had asterisk running just fine on freebsd before |
| 21:46.16 | Penguin | I've used it on FreeBSD as well. NAT was elsewhere, though. |
| 21:47.07 | fury_ | my best clue is that asterisk is sending @10.0.0.1 somewhere, which is my nat IP. |
| 21:47.18 | fury_ | and I don't think it should be doing that. |
| 21:47.23 | fury_ | do you know how to fix that? baby steps :) |
| 21:47.41 | *** join/#asterisk oesteve (~oscar@145.Red-88-15-97.dynamicIP.rima-tde.net) |
| 21:50.22 | *** join/#asterisk spicyramen_ (~Adium@c-98-210-160-181.hsd1.ca.comcast.net) |
| 21:51.46 | *** join/#asterisk aross42 (~aross@50-204-179-34-static.hfc.comcastbusiness.net) |
| 21:53.31 | Penguin | When I make a call, my public IP address shows up on the Call-ID. This should be done by asterisk when you have set nat=yes, localnet=your-net, and externhost=your.host. Perhaps you didn't remember to reload sip? You might need to unload and load the whole module or even restart asterisk. |
| 21:54.28 | fury_ | I've actually been restarting asterisk every time to make sure, but I know about sip reload |
| 21:55.20 | Penguin | Look at the output from sip show settings. Be sure your external addr and localnet(s) are showing correctly. |
| 22:00.46 | fury_ | this is after a fresh restart: http://www.codexterous.com/paste/results/JQ1HrF86.html |
| 22:00.48 | fury_ | it appears they are |
| 22:00.52 | fury_ | yet call-id is still 10.0.0.1 |
| 22:01.38 | fury_ | perhaps I'm missing something else? the external addr and localnets are as they should be |
| 22:04.31 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 22:08.04 | Penguin | You're trying to make a call from a phone in your LAN? |
| 22:08.35 | fury_ | eventually I'll be using the asterisk console initially to dial a call, but first I'm trying to get SIP working |
| 22:09.06 | Penguin | Is that a yes or no? |
| 22:09.10 | fury_ | using the same NAT, as long as I configure the pf rules correctly (forward the right ports, etc), I can use a softphone (PhonerLite) to successfully make a call |
| 22:09.22 | fury_ | (from a different machine) |
| 22:09.36 | fury_ | well eventually, yes. but the goal currently is to get SIP to work with asterisk. but yes, I'll be making calls. |
| 22:09.46 | fury_ | not from a hardware phone if that's what you mean. |
| 22:10.03 | Penguin | I'm not asking about the future plans. I'm asking about this call right now that you just showed me. |
| 22:10.58 | fury_ | I don't think I showed you a call. I showed you the config, do you mean the -outbound or whatever that's commented out? or are you speaking of the debug info? seems to me that's just asterisk trying to register, as the method it's using is REGISTER. |
| 22:11.34 | fury_ | the call-id I mean. it isn't an actual call. |
| 22:11.57 | fury_ | so to clarify, the answer is no, I'm not trying to make a call currently. |
| 22:11.59 | Penguin | Ah, right. I was thinking of something else. |
| 22:12.06 | fury_ | no problem at all |
| 22:13.50 | Penguin | My Call-ID in a REGISTER packet shows 127.0.0.1. |
| 22:14.39 | fury_ | ok cool. so that doesn't matter then. I think I have it figured out |
| 22:14.39 | fury_ | <Penguin> I've never seen anyone show up and say, "Hey, I finally fixed that problem on my pfSense box, and here's how..." |
| 22:14.51 | fury_ | Hey, I finally fixed that problem on my pf FreeBSD box, and here's how :) |
| 22:15.18 | fury_ | everything you said about nat, proper firewall rules, and bindaddr=10.0.0.1 |
| 22:15.32 | fury_ | which is my nat ip |
| 22:15.33 | Penguin | Your Via, From, To, and Contact all have the appropriate addresses in them. |
| 22:15.55 | fury_ | I'm now getting sip reads |
| 22:15.58 | fury_ | and no timeouts anymore |
| 22:16.12 | fury_ | sip show subscriptions, however, shows 0 active SIP subscriptions |
| 22:16.15 | fury_ | I'm not sure if that matters |
| 22:16.19 | Penguin | (1626.23) <Penguin> I think I would bind it to the private interface's addr, and treat it as if it is natted. |
| 22:16.32 | Penguin | You finally got around to doing that? |
| 22:17.22 | fury_ | apologies, I either missed that amongst all the other chatter or was unclear on what that meant |
| 22:17.32 | Penguin | Subscriptions are something else. You'll want to check sip show registry to see if you registered successfully. |
| 22:18.17 | fury_ | ah. |
| 22:18.17 | fury_ | conus.ucallweconn.net:5060 N uv-2133160 105 Registered Mon, 08 Sep 2014 18:17:15 |
| 22:18.17 | fury_ | 1 SIP registrations. |
| 22:18.26 | Penguin | 105 Registered! |
| 22:18.33 | Penguin | You're good to go. |
| 22:18.51 | fury_ | awesome |
| 22:19.00 | fury_ | now to figure out how to place a call ;) thank you so much for your help. |
| 22:19.08 | fury_ | I really do appreciate your time. |
| 22:19.43 | Penguin | It's no problem at all. I'm glad it's solved. |
| 22:20.39 | fury_ | simple pf firewall config: http://www.codexterous.com/paste/results/yl2U3Q16.html |
| 22:20.43 | fury_ | just as an example |
| 22:20.54 | fury_ | probably the 10000:11000 is for directmedia which is not used anyway |
| 22:21.06 | fury_ | and it needs more work |
| 22:21.19 | fury_ | but essentially pf == correct port forwards, bindaddr = nat IP in config |
| 22:21.31 | fury_ | should work fine for pfsense |
| 22:21.55 | fury_ | probably could make it work without NAT but I'd rather not with a dynamic IP |
| 22:23.43 | Penguin | When asterisk is used as a client, it typically does not need any ports forwarded to it. |
| 22:23.49 | Penguin | And I have to leave. |
| 22:27.22 | *** join/#asterisk ggayan (~ggayan@190.215.47.74) |
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| 22:45.30 | ahklerner | penguin are you there |
| 22:50.40 | ahklerner | i am attempting to use a variation of this -> http://pastebin.com/wkX4bGKT to handle outbound calling by getting google to call me. |
| 22:53.38 | *** join/#asterisk bkruse (~Adium@64.89.97.113) |
| 22:58.18 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 23:12.45 | ahklerner | i have made some progress, but its probably wrong |
| 23:16.34 | lvlinux | ahklerner: what is your progress so far? what does it do or not do that is wrong? |
| 23:16.59 | ahklerner | i am able to call my gv number and it rings my sip phone |
| 23:18.38 | ahklerner | when i attempt to call out it was saying 'call cannot be completed as dialed' but i changed the context devices to from-internal, now it shows console 'Never received callback from google blah; |
| 23:23.08 | ahklerner | i also changed the wait to 60 seconds to wait longer for google to call, still not working |
| 23:23.53 | ahklerner | also if i run the gvoice command on command line it does call me |
| 23:25.01 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 23:25.44 | lvlinux | what happens on the console? |
| 23:27.17 | ahklerner | it seems to call the gvoice command then wait |
| 23:27.22 | ahklerner | then nothing happens |
| 23:27.33 | ahklerner | then it says Never received callback from google voice |
| 23:28.05 | lvlinux | but u can run the command on the terminal and it works? |
| 23:28.11 | ahklerner | yes |
| 23:28.37 | ahklerner | my sip phone rings with a call from google |
| 23:28.39 | lvlinux | try running it in the terminal so they will call you right when * is waiting for the call and see if everything else works as planned. |
| 23:28.53 | ahklerner | ok |
| 23:34.33 | ahklerner | lvlinux: http://pastebin.com/2QrrvHPJ |
| 23:35.08 | ahklerner | through line 7 my sip phone was not connected |
| 23:36.51 | lvlinux | i see that---that is probably why it didn't work |
| 23:37.22 | ahklerner | it did work when i tried again |
| 23:37.49 | ahklerner | starting at line 8 is the second run of the command on the command line |
| 23:38.22 | ahklerner | sorry 16 |
| 23:38.49 | ahklerner | i am just fumbling through this |
| 23:40.30 | lvlinux | ah ok i see |
| 23:41.31 | lvlinux | well if it worked running it manually, then chances are that's where your problem is--- the command is either not being run from within asterisk or it's being killed or something like that. |
| 23:41.51 | lvlinux | or possibly it requires a terminal rather than just a shell? |
| 23:42.04 | lvlinux | you can try taking the & off the end of the command--see if that works |
| 23:42.47 | ahklerner | i will give that a go |
| 23:43.25 | *** join/#asterisk [Outcast] (~outcast@pool-96-237-170-171.bstnma.fios.verizon.net) |
| 23:43.26 | lvlinux | did you have the & on there for a particular reason? |
| 23:43.58 | ahklerner | app_system.c:138 system_exec_helper: Unable to execute |
| 23:44.13 | ahklerner | http://hobbiesbytwinclouds.wordpress.com/2014/02/07/how-to-make-and-receive-call-using-google-voice-without-xmpp/ |
| 23:44.24 | ahklerner | just using that writeup and that was in there |
| 23:44.50 | ahklerner | so do i need to give asterisk user some permissions |
| 23:46.21 | lvlinux | ehh possibly---but I doubt that gvoice is doing anything that needs much---oh wait |
| 23:46.32 | lvlinux | i just thot of something---where is your gvoice program located? |
| 23:47.01 | lvlinux | you can try putting the full path in the command--i.e. /usr/bin/gvoice... |
| 23:47.28 | ahklerner | i did try that, |
| 23:47.32 | ahklerner | it still did not work |
| 23:47.40 | ahklerner | well wait |
| 23:50.23 | ahklerner | i tried /usr/bin but when i look in /usr/bin there is no gvoice there |
| 23:50.30 | ahklerner | sooo, i will need to find it |
| 23:50.43 | *** join/#asterisk Kerber0s (~Kerber0s@105.157.191.153) |
| 23:51.11 | lvlinux | yep that'll do it... |
| 23:51.21 | lvlinux | you can run "which gvoice" |
| 23:51.23 | ahklerner | ;) |
| 23:51.32 | ahklerner | awesome thanks! |
| 23:51.45 | lvlinux | find it? |
| 23:51.54 | ahklerner | /usr/locl/bin |
| 23:51.58 | lvlinux | ah yes |
| 23:52.03 | ahklerner | /usr/local/bin |
| 23:52.07 | lvlinux | k well try putting that in there |
| 23:53.38 | ahklerner | it is working now! |
| 23:53.44 | ahklerner | wow |
| 23:53.50 | ahklerner | so cool |
| 23:54.22 | lvlinux | yay! :-) |
| 23:54.54 | lvlinux | have fun---and now read the asterisk book and _learn what it does_ :-D |
| 23:56.33 | ahklerner | yes now i need to |
| 23:59.48 | ahklerner | lvlinux: is there are particular book you speak of |
