| 00:00.06 | *** part/#asterisk kharwell (kharwell@nat/digium/x-oqldwtoxbjbufkdu) |
| 00:13.34 | igcewieling | yorick: I just checked and you are correct for Ast 11 |
| 00:13.56 | *** join/#asterisk deweydb_ (~deweydb@S0106689e199caaf4.no.shawcable.net) |
| 00:14.19 | deweydb_ | ok. bria is driving me crazy. it crashes constantly, there are so many bugs. is there anything better for android? |
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| 02:41.47 | igcewieling | all softphones suck. |
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| 02:44.41 | igcewieling | deweydb_: I suspect some of the suckage is the phone or cellular internet |
| 02:45.13 | deweydb_ | nah |
| 02:45.26 | deweydb_ | i'm only using it on wifi. like 3 feet from AP |
| 02:45.43 | deweydb_ | AP is high quality, ubiquity, connected to 250 mbit internet service. |
| 02:45.48 | deweydb_ | thats not the issue |
| 02:45.53 | deweydb_ | the issue is that the app just fucking sucks |
| 02:45.56 | deweydb_ | crashes constantly |
| 02:46.08 | deweydb_ | like, calls come in, and my whole phone locks up and i can't answer them |
| 02:46.10 | igcewieling | deweydb_: CSIPSimple seems to suck less than most. |
| 02:46.22 | deweydb_ | or i'm on a call. and i try to go to another app, and the phone crashes. |
| 02:46.32 | deweydb_ | or ... oh man, i could list the suckyness for hours |
| 02:46.47 | deweydb_ | thanks, i'll try that |
| 02:46.55 | igcewieling | I suspect until recently, hardware didn't have to deal with low latency audio, and that might account for part of the issue. |
| 03:08.49 | *** join/#asterisk mikekane (~Mike@adsl-108-212-224-81.dsl.klmzmi.sbcglobal.net) |
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| 03:14.49 | mikekane | So, I'm very confused. When I left my office today everything was working great on my dev server. Logged in to rasterisk from home, and I'm being slammed with this message 14x/second: [Feb 1 22:07:06] NOTICE[2991]: chan_sip.c:28475 handle_request_register: Registration from '"1610398482" <sip:1610398482@my.server.ip.address>' failed for '128.199.69.200:5860' - Wrong password |
| 03:15.36 | mikekane | My first question is: Does that look like I'm being brute forced? |
| 03:15.56 | drmessano | Looks like a standard SIP attack |
| 03:16.13 | drmessano | Like literally every other people has experience |
| 03:16.14 | drmessano | Like literally every other people has experienced |
| 03:16.57 | mikekane | Fair enough. I only allow IP Auth anyway, so it's probably not vulnerable, correct? |
| 03:18.44 | mikekane | It looked like an attack to me, but I didn't want to overreact. |
| 03:30.04 | drmessano | If you're ACLing, as you say, you're fine |
| 03:31.23 | mikekane | I'm guessing there's an ACL under the hood. I only hit "Disable SIP Registration" in FlowRoute. |
| 03:31.36 | mikekane | I'm showing my asterisk noob colors today :/ |
| 03:36.51 | mikekane | meh, I just blocked the IP at the firewall. |
| 03:39.17 | mikekane | drmessano, Is there a better way? Like, after X failed attempts, block IP, or something? |
| 03:41.53 | drmessano | mikekane: |
| 03:42.12 | drmessano | That Flowroute setting has to do with calls going TO FLOWROUTE |
| 03:42.20 | drmessano | That doesnt at all protect your box |
| 03:42.33 | drmessano | Google for iptables SIP rate limiting |
| 03:42.42 | drmessano | Should fine some brute force protection stuff |
| 03:42.44 | drmessano | find* |
| 03:44.38 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-24.ph.ph.cox.net) |
| 03:49.34 | mikekane | Thanks |
| 04:34.03 | ChannelZ | I use |
| 04:34.07 | ChannelZ | iptables -A INPUT --protocol UDP --dport sip -m recent --set |
| 04:34.15 | ChannelZ | iptables -A INPUT --protocol UDP --dport sip -m recent --rcheck --seconds 30 --hitcount 5 --jump DROP |
| 04:34.46 | ChannelZ | but YMMV |
| 04:37.06 | ChannelZ | (I also exempt all my LAN devices and ITSP IPs prior to this..) |
| 04:39.59 | *** join/#asterisk UncleKiwi (~UncleKiwi@unaffiliated/unclekiwi) |
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| 05:19.10 | *** join/#asterisk MRH2 (~Thunderbi@233.47.187.81.in-addr.arpa) |
| 05:20.11 | MRH2 | hi trying to set callerid on outbound iax trunk, the provider says "IAX support currently only supports one caller Line Identity which will be used as a network number." |
| 05:20.30 | MRH2 | is there a setting for that in iax.conf maybe i am missing |
| 05:30.55 | [TK]D-Fender | You can pass CID with IAX2 just fine |
| 05:34.23 | MRH2 | is there anything in iax.conf that i need to watch out for? |
| 05:35.43 | MRH2 | like in sip it doesn;t like fromuser or authuser id you set callerid num |
| 05:35.49 | MRH2 | *if |
| 05:35.51 | [TK]D-Fender | no |
| 05:35.59 | *** join/#asterisk wasanzy (~wasanzy@197.159.129.10) |
| 05:36.02 | [TK]D-Fender | IAX doesn't use the CID as identification |
| 05:36.20 | [TK]D-Fender | there are no such thing as those headers |
| 05:36.35 | WIMPy | Sounds to me like you're just not allowed to set it. |
| 05:37.19 | MRH2 | is there any way to set it in iax.conf for all calls going out via a peer? |
| 05:38.23 | WIMPy | no |
| 05:38.54 | MRH2 | well sip works fine... i guess yeah maybe the supplier has to config iax on a user by user basis? |
| 05:39.47 | WIMPy | Only tehy know what they are doing. |
| 05:41.12 | MRH2 | lol true but is that normal for iax providers? not allowing individual setcalleridnum |
| 05:41.58 | drmessano | They're not claiming that's how *IAX* works |
| 05:42.13 | drmessano | "IAX Support" sounds like *their* IAX support |
| 05:42.18 | drmessano | Specifically |
| 05:42.19 | MRH2 | yes it is not very detailed |
| 05:42.24 | drmessano | yes it is |
| 05:42.36 | drmessano | That sounded very specific |
| 05:43.06 | drmessano | "IAX support" would indicate THEIR IAX support, not "IAX only allows.." |
| 05:43.09 | WIMPy | Providers offering IAX have becomve very rare. |
| 05:43.25 | drmessano | So basically, you have to use their tools to set one CID.. and that's all you get |
| 05:44.02 | MRH2 | ok thanks |
| 05:44.17 | drmessano | I would guess their IAX implementation is Asterisk or maybe even FreeSWITCH, and their SIP is using one of the well known SIP Proxies |
| 05:44.28 | drmessano | So they are somewhat limited in what they allow |
| 05:44.33 | [TK]D-Fender | Show the actual call |
| 05:44.35 | MRH2 | thought maybe i could set 1 but via my asterisk config |
| 05:44.52 | MRH2 | i am going to do more digging.. |
| 05:45.13 | drmessano | Rather than digging |
| 05:45.15 | drmessano | Don't use IAX |
| 05:45.20 | MRH2 | i couldn;t find anything in their api but it's so late hereit's early :P |
| 05:45.29 | drmessano | That's insanely simple |
| 05:45.50 | MRH2 | what is? |
| 05:46.14 | drmessano | Uh, what I said 2 lines above |
| 05:46.25 | drmessano | You must be using a PalmOS IRC Client |
| 05:46.36 | drmessano | 00:45:13 <drmessano> Rather than digging |
| 05:46.36 | drmessano | 00:45:16 <drmessano> Don't use IAX |
| 05:46.53 | MRH2 | i use iax as stage 2 call if there are sip issues |
| 05:47.16 | drmessano | What SIP issues could possibly exist that would necessitate using IAX? |
| 05:47.30 | MRH2 | servers, routing whatever |
| 05:47.40 | MRH2 | overload |
| 05:48.03 | drmessano | How often do they lose their whole SIP infrastructure that would require switching to the mostly unsupported IAX? |
| 05:48.22 | MRH2 | well there is a lot of network between me and them |
| 05:48.27 | drmessano | This is like justifying dialup backup for cable internet.. |
| 05:48.30 | MRH2 | it doesn;t have to be them that is the issue |
| 05:48.43 | drmessano | Lot of network issues would affect IAX EXACTLY THE SAME |
| 05:48.58 | drmessano | So your SIP just randomly stops? |
| 05:49.28 | MRH2 | it can qualify out |
| 05:49.34 | drmessano | ........ |
| 05:50.10 | drmessano | So you're fighting to get IAX working as a backup because you don't have SIP set up properly |
| 05:50.21 | MRH2 | lol no sip works properly |
| 05:50.28 | MRH2 | iax just in case |
| 05:50.28 | drmessano | "it can qualify out" |
| 05:50.41 | drmessano | You're actually literally making up failure scenarios |
| 05:50.44 | MRH2 | heck we all don;t have fibre..... |
| 05:51.00 | drmessano | How is IAX going to work over SIP in ANY of those scenarios |
| 05:51.08 | drmessano | and what does fibre have to do with tit? |
| 05:51.10 | drmessano | and what does fibre have to do with it? |
| 05:51.38 | drmessano | Im not sure I understand the issues you're having with SIP |
| 05:52.05 | drmessano | because none of these even remotely explain this made-up "IAX as a backup" thing |
| 05:52.13 | MRH2 | no my issue is only caller id with iax |
| 05:52.28 | drmessano | and your reasons for setting up IAX are insane |
| 05:52.33 | drmessano | Skip the IAX |
| 05:52.44 | drmessano | Your provider obviously doesn't well support it |
| 05:52.47 | drmessano | SIP works fine |
| 05:53.18 | MRH2 | lol |
| 05:54.14 | drmessano | lol? |
| 05:54.47 | drmessano | There is literally NOTHING about using IAX over the SAME network path that supports its setup as a "backup" |
| 05:55.05 | MRH2 | it's not like they are on the same subnet |
| 05:55.58 | drmessano | Different internet connection? |
| 05:56.13 | MRH2 | some overlap of datacentres |
| 05:56.32 | MRH2 | prob redundant connections |
| 05:56.56 | drmessano | So you've implemented this in multiple locations |
| 05:57.02 | MRH2 | not me |
| 05:57.08 | MRH2 | supplpier |
| 05:57.10 | MRH2 | supplier |
| 05:57.16 | drmessano | supplier? |
| 05:57.40 | drmessano | That's not an applicable term. What is a supplier? Your grocer? |
| 05:57.41 | [TK]D-Fender | We don't know anything |
| 05:57.45 | MRH2 | i dial out using sip registration and if that is offline or fails then to an iax server |
| 05:57.48 | [TK]D-Fender | we have no debug to look at |
| 05:57.56 | [TK]D-Fender | Everything is nothing more than a guess |
| 05:58.03 | [TK]D-Fender | and guessing is a waste of time |
| 05:58.07 | drmessano | MRH2: Are you referring to your PROVIDER? |
| 05:58.27 | MRH2 | yes like the P in ISP |
| 05:58.35 | drmessano | ..... |
| 05:58.47 | drmessano | Thats not and never will be a "Supplier" |
| 05:59.11 | drmessano | But YOU have the SAME internet connection that you're using for IAX or SIP, RIGHT? |
| 05:59.51 | MRH2 | on my send i have redundant connection |
| 05:59.54 | MRH2 | *my side |
| 06:00.00 | drmessano | Correct |
| 06:00.04 | drmessano | YOUR end |
| 06:00.08 | *** join/#asterisk fbnts (~fbnts@s099.spireinns.co.uk) |
| 06:00.24 | drmessano | SAME connection |
| 06:00.30 | drmessano | 00:54:47 <drmessano> There is literally NOTHING about using IAX over the SAME network path that supports its setup as a "backup" |
| 06:00.45 | [TK]D-Fender | yup |
| 06:00.46 | drmessano | This is nonsensical jibberish at this point |
| 06:00.51 | drmessano | So im done |
| 06:00.56 | [TK]D-Fender | UDP that is equally vulnerable |
| 06:01.04 | drmessano | Same fucking connection |
| 06:01.05 | MRH2 | ? |
| 06:01.11 | drmessano | Same PBX I would presume |
| 06:02.05 | drmessano | Chances are, they have a number of redundancies for their SIP proxies and like 1 IAX box |
| 06:02.12 | drmessano | So that's even more BENT |
| 06:03.05 | MRH2 | it was a real simple question... iax.conf setting for callerid on the channel, no, ok then |
| 06:03.47 | drmessano | Forget IAX.. Simple answer |
| 06:04.07 | drmessano | What you have is not a "backup" in any sense <= Expanded answer |
| 06:05.23 | MRH2 | i get cases where there has been network issues to certain destination IPs = backup |
| 06:05.50 | drmessano | 01:02:05 <drmessano> Chances are, they have a number of redundancies for their SIP proxies and like 1 IAX box |
| 06:06.00 | MRH2 | maybe .... |
| 06:06.04 | MRH2 | prob not |
| 06:06.08 | drmessano | Extremely likely |
| 06:06.08 | MRH2 | but possible |
| 06:06.36 | drmessano | How long have you been dealing with "suppliers", as you call them? |
| 06:06.41 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 06:06.43 | MRH2 | 20+years |
| 06:07.01 | drmessano | So you've been using Asterisk with SIP and IAX for 20+ years? |
| 06:07.09 | drmessano | That's longer than the developers |
| 06:07.18 | MRH2 | no suppliers |
| 06:07.22 | MRH2 | that's it |
| 06:07.30 | MRH2 | asterisk must have been |
| 06:07.47 | MRH2 | well before 1.2 anyway |
| 06:08.07 | MRH2 | it used to be alot friendlier on IRC |
| 06:08.25 | drmessano | So you should be well aware of the extremely limited provider support for IAX2, limited implementations, etc |
| 06:08.28 | mikekane | Well, I learned something very important tonight: No matter how important IRL issues are (excluding death) do not step away while someone is trying to brute force your Asterisk server |
| 06:09.00 | mikekane | (Asterisk is down now while I attend to my firewall |
| 06:09.02 | drmessano | mikekane: What do you mean? |
| 06:09.08 | drmessano | oh |
| 06:10.03 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 06:11.08 | drmessano | SIP scanning has been going on since the beginning of SIP.. one might argue that if you have strong SIP credentials and don't allow guest calls, you're fine. But the firewall helps too |
| 06:11.26 | drmessano | Actually the firewall is good to keep a really high bandwidth attack from shutting down a box |
| 06:12.04 | drmessano | Put a dinky Asterisk VM on a gigabit pipe and it won't last long |
| 06:12.17 | mikekane | That was part of the issue before. I could barely type due to the lag. |
| 06:12.56 | drmessano | I learned the hard way a few years ago. I would get scanned and Asterisk would lock up until I restarted |
| 06:12.56 | mikekane | I think I very much lucked out by deciding to work from home tonight |
| 06:13.18 | drmessano | So even a little bit of iptables magic goes a long way |
| 06:13.59 | drmessano | Honestly, that worries me more than getting brute forced |
| 06:14.17 | mikekane | What does? |
| 06:14.58 | *** join/#asterisk clopez_ (~tau@neutrino.es) |
| 06:14.58 | *** join/#asterisk ShuttleDuck (~heya@ec2-35-157-68-150.eu-central-1.compute.amazonaws.com) |
| 06:14.59 | drmessano | The attack locking up Asterisk and making in inoperable until a restart, vs the brute force actually matching an account and doing anything useful |
| 06:16.03 | mikekane | Yeah, I'm fairly certain in my case it couldn't have matched. The only credentials I have are for FlowRoute. |
| 06:16.08 | drmessano | I could install Asterisk on a VM at some provider, spin up another VM, run SIPp on the second VM attacking the first VM, and chan_sip would die |
| 06:16.28 | drmessano | iptables helps that 100% |
| 06:17.55 | mikekane | Although, this time they managed to get as far as trying to place calls with extensions |
| 06:18.15 | drmessano | In what way? |
| 06:18.17 | *** join/#asterisk Nugget (~nugget@mezger.macnugget.org) |
| 06:18.21 | *** join/#asterisk smkelly (~smkelly@mykonos.smkelly.org) |
| 06:18.29 | mikekane | The constantly streaming error message was: Call from '' (23.239.85.66:7304) to extension '66589' rejected because extension not found in context 'public'. |
| 06:18.39 | drmessano | Ok? |
| 06:18.43 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 06:18.52 | drmessano | Thats.. normal |
| 06:18.58 | mikekane | oh, no. That's making calls TO extensions |
| 06:19.05 | drmessano | Yes |
| 06:19.14 | mikekane | Neeeevermiiiiind |
| 06:19.19 | drmessano | What youre describing is a standard, everyday scan |
| 06:19.54 | mikekane | What would they gain from calling an extension, though? |
| 06:19.57 | drmessano | This is no different than any HTTP server getting hit thousands of times a day |
| 06:20.29 | drmessano | Youre probing at that point |
| 06:20.38 | drmessano | If extensions start matching |
| 06:20.50 | drmessano | You try 12025551212 |
| 06:21.10 | drmessano | See what it can throw at the box and what will pass |
| 06:21.43 | mikekane | Ah, makes sense |
| 06:22.17 | drmessano | I used to get calls to my extensions all the time |
| 06:22.35 | drmessano | They would ring, I would answer, and then nothing |
| 06:22.58 | drmessano | No more SIP guests |
| 06:23.19 | mikekane | I wouldn't have thought that was enabled by default |
| 06:23.36 | mikekane | googles disabling SIP guests |
| 06:23.54 | drmessano | Hippies.. open calling for everyone.. peer to peer SIP calling.. free the dialtone #allthethings |
| 06:24.07 | drmessano | That's half joking |
| 06:24.15 | mikekane | I figured |
| 06:24.35 | drmessano | Many people are lazy and leaving that at enabled permits inbound calls |
| 06:24.46 | drmessano | So at that point you're hopefully using a sane dialplan |
| 06:25.05 | drmessano | Where an inbound call doesn't get direct access to an outbound context |
| 06:25.34 | drmessano | Though there is something to be said for peer to peer SIP calling |
| 06:25.59 | drmessano | But I think technology ruined that |
| 06:26.55 | mikekane | I was planning to eventually have a means of conference calling the admin on our projects, but I'll have to work around sip guests now |
| 06:27.00 | *** join/#asterisk kunwon1 (~kunwon1@unaffiliated/kunwon1) |
| 06:27.35 | mikekane | So, how does one enable inbound calling while disabling SIP guests? |
| 06:27.49 | mikekane | Or maybe I'm barking up the wrong tree |
| 06:28.37 | MRH2 | just don;t mention iax :P |
| 06:29.21 | mikekane | Ha, you got it |
| 06:30.02 | drmessano | mikekane: It's part of your peer configuration. Are you using the details supplied by Flowroute? |
| 06:30.55 | mikekane | Slightly modified, yes. The examples did not account for IP Auth using your techprefix |
| 06:31.12 | mikekane | Otherwise, yes |
| 06:33.16 | drmessano | I think the supplied config should be sufficient. Haven't looked at their defaults in a while |
| 06:36.44 | mikekane | I was in here earlier in my work day because I couldn't connect with their config |
| 06:37.22 | mikekane | It turned out that when you restricted Flowroute to IP Auth you had to do this: |
| 06:37.35 | mikekane | [outgoing] |
| 06:37.35 | mikekane | exten => _1NXXNXXXXXX,1,Dial(SIP/${TECHPREFIX}*${EXTEN}@flowroute) |
| 06:38.18 | mikekane | as opposed to their default: [outgoing] |
| 06:38.18 | mikekane | exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@flowroute) |
| 06:38.19 | *** join/#asterisk jkroon (~jkroon@uls-154-73-35-201.wall.uls.co.za) |
| 06:38.59 | mikekane | That's the only difference |
| 06:40.28 | igcewieling | 2 months until April 1, I am ready for it this year. |
| 06:41.05 | MRH2 | you have half price chocolate day to look forward to first |
| 06:41.14 | mikekane | hah |
| 06:42.42 | igcewieling | On my test version of our internal asterisk GUI an spider runs across the screen occasionally. An April 1 I'll deploy it with 5ish spiders running around all the time |
| 06:43.44 | MRH2 | you could spice things up with an alligator |
| 06:45.18 | MRH2 | passing out zzzzzzzzzzzzzzz thanks for the input |
| 06:45.25 | MRH2 | byeee |
| 06:45.41 | mikekane | Play the occasional sound of a spider skittering across the floor into your reps' channels. |
| 06:53.26 | WIMPy | mikekane: Are you from LA? |
| 06:53.57 | mikekane | WIMPy, nope. MI |
| 06:54.09 | mikekane | I wish I was that warm |
| 06:54.21 | WIMPy | I thought only Hollywood has spiders that make sounds. |
| 06:55.01 | mikekane | Ah, because April Fools day pranks have to be completely logical |
| 06:55.38 | WIMPy | But they only work if people understand them. |
| 06:56.34 | mikekane | You don't think the fact that Hollywood has audible spiders combined with the GUI graphic will explain? |
| 06:56.57 | WIMPy | Maybe |
| 06:57.31 | mikekane | Ya know, we might be overthinking this |
| 06:57.35 | mikekane | a little |
| 06:57.39 | mikekane | maybe |
| 06:58.28 | WIMPy | would never ever do that. Never. No. |
| 06:58.33 | *** join/#asterisk Demon_VoIP (~demon@109.60.222.253) |
| 07:04.06 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 07:24.33 | mikekane | Flowroute is recommending this for "RTP Media" iptables -A INPUT -p UDP --dport 10000:20000 -j ACCEPT But that seems incredibly permissive |
| 07:24.38 | mikekane | Any thoughts? |
| 07:30.34 | WIMPy | relies on conntrak magic. |
| 07:31.39 | mikekane | WIMPy, Since I don't know what that is, I don't know if that has anything to do with my question or is separate |
| 07:32.15 | mikekane | is new to asterisk |
| 07:32.34 | WIMPy | It's about iptables RELATED rule. |
| 07:33.06 | noobkane | maybe I'm just exhausted. |
| 07:33.49 | noobkane | I'd be sleeping now, but I want to get my server locked down |
| 07:33.57 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:34.11 | noobkane | but I FEEL like I should remember what the RELATED rule is |
| 07:34.20 | noobkane | laughs quietly at self |
| 07:34.39 | WIMPy | If you have ever used iptables, you should. |
| 07:35.01 | mikekane | I have, for years... lol |
| 07:35.29 | mikekane | CentOS 7 uses a new firewall by default and I replaced it |
| 07:36.00 | mikekane | I think maybe I'll just call it a night and remember in the morning, lol |
| 07:36.22 | WIMPy | Sounds like a plan. |
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| 07:55.03 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 08:13.39 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 08:14.07 | madduck | am stuck behind a carrier-grade NAT setup, and while SIP/RTP worked for a couple of days, it suddently stopped. Here's the SIP/RTP debug output, as w |
| 08:14.46 | madduck | ell as the SIP config: http://slexy.org/raw/s2aRsIJAyb |
| 08:14.50 | madduck | any clues appreciated! |
| 08:15.29 | madduck | (ISPs that use CGNAT and refuse IPv6 should have their licences remocvdâ¦) |
| 08:15.33 | madduck | removed even |
| 08:33.04 | *** join/#asterisk UncleKiwi (~UncleKiwi@unaffiliated/unclekiwi) |
| 08:34.34 | WIMPy | Should be enough if they have their customers removed. |
| 08:48.31 | madduck | turns out it works with kernel 3.16.0 and not with 4.8.0 |
| 08:48.39 | madduck | simply reboot to kernel 3 and it works. |
| 08:58.12 | *** join/#asterisk cyberik (~ejsf@81.30.38.29) |
| 09:00.19 | madduck | why?? |
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| 09:53.43 | phrearch | hello |
| 09:54.41 | phrearch | i was wondering how asterisk webrtc support works. does it use sipml/websockets to negotiate a webrtc-like connection between asterisk and a browser, or is the websocket also used to transfer media on? |
| 09:59.27 | Samot | It's WebRTC. |
| 09:59.43 | Samot | How it works doesn't change. It's still WS/WSS. |
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| 10:37.42 | wasanzy | is there a split function in asterisk to split values? |
| 10:39.41 | Samot | CUT |
| 10:40.21 | Samot | Or better yet, clarify what you mean by "split". |
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| 10:45.39 | *** join/#asterisk Maliuta (~nobusines@unaffiliated/maliuta) |
| 10:48.56 | wasanzy | Samot: something like cut yes. What I want to do is I have say "One Thousand and seventeen" so I want to split this by space. meaning each field will be assign to a variable but in the dialplan, I may not know the number of fields when accessing the values one by one. |
| 10:49.31 | Samot | Then you should loop the CUT |
| 10:49.34 | Samot | So you can count. |
| 10:49.40 | Samot | And see when it ends. |
| 10:50.23 | wasanzy | ok, while() could do the magic right? |
| 10:51.52 | Samot | Should, you'll have to read up. |
| 10:52.08 | wasanzy | yea I will read |
| 10:53.49 | wasanzy | FIELDQTY() am looking at this. it counts number of fields |
| 11:36.10 | wasanzy | is it possible to write a function in asterisk dialplan and call it under different context we it is done in programming languages? |
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| 12:26.43 | dokma | I'm trying to run linphone on the same machine where asterisk is running and it looks to me as if though it's talking to itself? |
| 12:27.02 | dokma | You can see the session here: http://david.evorion.hr:9000 |
| 12:28.05 | dokma | I've set linphone peer to port 5067 |
| 12:28.35 | dokma | I can make calls from linphone but cannot receive any |
| 12:30.38 | dokma | It makes sense to me since apparently both asterisk and linphone are trying to talk on port 5067 on the same machine. |
| 12:30.39 | *** join/#asterisk nemmeviu (~nemmeviu@unaffiliated/nemmeviu) |
| 12:30.52 | nemmeviu | hy misters |
| 12:30.56 | dokma | What baffles me though is that linphone can even open the port if asterisk already got it? |
| 12:31.24 | nemmeviu | one simple question about extensions |
| 12:31.57 | *** join/#asterisk Tiffon (~name@unaffiliated/tiff0n) |
| 12:32.17 | dokma | hmmmm... now I see that asterisk does not open port 5067 when linphone is closed |
| 12:32.30 | nemmeviu | i can create one exten like this: exten => ${read_number},1,Playback(some_sound) |
| 12:32.34 | nemmeviu | ? |
| 12:33.14 | nemmeviu | then use one exten to many options? |
| 12:34.09 | nemmeviu | exten => ${read_number},1,Playback(${read_number}) |
| 12:39.06 | dokma | anyone knows how to paginate the asterisk cli? |
| 12:43.07 | *** join/#asterisk Tiffon (~name@unaffiliated/tiff0n) |
| 12:44.16 | *** join/#asterisk newtonr (~newtonr@173-17-133-211.client.mchsi.com) |
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| 12:50.49 | nemmeviu | sorry masters, puedo trabajar con exten => _XX y tener las variabes perfecto |
| 12:50.52 | nemmeviu | Asterisk Rulez!! |
| 12:50.54 | nemmeviu | gracias |
| 12:51.04 | *** part/#asterisk nemmeviu (~nemmeviu@unaffiliated/nemmeviu) |
| 12:53.40 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 12:57.42 | Samot | dokma: You cannot paginate the CLI |
| 13:00.55 | *** join/#asterisk nemmeviu (~nemmeviu@unaffiliated/nemmeviu) |
| 13:02.44 | dokma | Samot: thanks |
| 13:03.00 | dokma | I managed to return up with Shift+PgUp which didn't work under gotty |
| 13:04.21 | dokma | Samot: that dialing through HT503 FOX in the end got solved with the simple: exten => _XXXXX.,1,Dial(SIP/FXO/${EXTEN}) |
| 13:05.56 | Samot | Right, treating the FXO peer as a trunk. |
| 13:06.07 | Samot | Instead of just trying to dial it like a device. |
| 13:23.08 | dokma | Samot: so you knew that? |
| 13:24.55 | Samot | Yes. |
| 13:25.01 | Samot | It's actually something I told you. |
| 13:25.11 | Samot | I said you have to treat FXO as a trunk peer not a device peer. |
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| 14:11.07 | phrearch | hi |
| 14:13.08 | phrearch | im fiddling around with asterisk. i wonder how i can call a softphone thats registered to my local asterisk. https://paste.kde.org/pgeizrgly should i always use the address/port on line 19? |
| 14:14.35 | phrearch | i get this error https://paste.kde.org/pbw1pwcdv |
| 14:15.28 | phrearch | calling from softphone to mobile works |
| 14:17.43 | [TK]D-Fender | @localhost:5062) <- remove all those suffixes |
| 14:18.19 | [TK]D-Fender | SIP/6055551235 <- this is how you define a device |
| 14:18.51 | [TK]D-Fender | You only use @ where yyou are crafting a URI tto call an outside resource that you don't have a device definition for in sip.conf |
| 14:25.56 | phrearch | aha thanks. i'll give that a try |
| 14:28.22 | wasanzy | Set(fd[i]=${System(echo ${TOTAL} | awk '{print $i}' )}) |
| 14:28.33 | phrearch | any tip on a good softphone for linux? |
| 14:28.50 | phrearch | i tried blink and linphone so far |
| 14:29.18 | [TK]D-Fender | both are fine. Ekiga is pretty good |
| 14:29.18 | mub | phrearch: ekiga |
| 14:29.33 | mub | or maybe a webphone |
| 14:31.40 | phrearch | ah thanks. ekiga looks solid |
| 14:32.02 | mub | make sure to set the DTMF mode to RFC2833 |
| 14:32.12 | mub | by default it's INFO |
| 14:33.04 | [TK]D-Fender | nothing wrong witth using either |
| 14:33.58 | mub | This one time a user was having trouble getting through an IVR, setting it to RFC2833 solved it! |
| 14:38.06 | [TK]D-Fender | Both work fine as long as you are smart ... and make them match on both ends. |
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| 14:46.09 | wasanzy | is there a problem with how am calling System? Set(fd[${i}]=${System(echo ${TOTAL} | awk '{ print ${i} }' )}) |
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| 14:53.52 | [TK]D-Fender | wasanzy, Clearly. |
| 14:54.26 | [TK]D-Fender | wasanzy, there is no function with that name |
| 14:54.59 | wasanzy | maybe spelling mistake? |
| 14:55.02 | [TK]D-Fender | wasanzy, Perhaps you should read the LIST of *'s functions, pick one that does exist and use itt properly... |
| 14:55.28 | wasanzy | is an application? |
| 14:56.07 | [TK]D-Fender | Do you SEE one with that name? |
| 14:57.08 | wasanzy | maybe? SHELL() |
| 14:57.27 | wasanzy | am surprised because in one of my asterisk setup, I systed System and it worked |
| 14:58.36 | [TK]D-Fender | You clearly have no idea HOW you used any of these commands |
| 14:59.22 | [TK]D-Fender | because that line you pasted wasn't valid for a function even if it did exist |
| 15:00.36 | wasanzy | This works: |
| 15:00.42 | wasanzy | System( curl -o /tmp/marrytts.wav "http://localhost:59125/process?INPUT_TEXT=${KEYWORD}&INPUT_TYPE=TEXT&OUTPUT_TYPE=AUDIO&AUDIO=WAVE_FILE&LOCALE=en_US&VOICE=dfki-prudence") |
| 15:00.55 | wasanzy | so am surprised |
| 15:01.13 | [TK]D-Fender | Why are you surprised? |
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| 15:05.43 | wasanzy | [TK]D-Fender: am surprised it is not recognizing the function |
| 15:06.03 | wasanzy | I will appreciate if you can point out the problem with the line I posted earlier |
| 15:06.37 | [TK]D-Fender | IT ISN'T A FUNCTION |
| 15:06.44 | [TK]D-Fender | rEAD THE list of functions. |
| 15:06.48 | [TK]D-Fender | dO YOU see IT THERE? |
| 15:06.59 | [TK]D-Fender | ***** NO ***** |
| 15:07.01 | wasanzy | am using SHELL now, and same issue |
| 15:07.12 | wasanzy | Set(fd[${i}]=${ SHELL(echo ${TOTAL} | awk '{ print ${i} }' ) }) |
| 15:07.40 | [TK]D-Fender | Why do you have GARBAGE SPACES in there? |
| 15:08.03 | [TK]D-Fender | Since when do you get to putt a space between { and S? |
| 15:08.09 | [TK]D-Fender | Your coding is sloppy |
| 15:08.21 | [TK]D-Fender | <wasanzy> is there a problem with how am calling System? Set(fd[${i}]=${System(echo ${TOTAL} | awk '{ print ${i} }' )}) <- this was never a functtion in the firstt place |
| 15:08.27 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 15:08.29 | [TK]D-Fender | And all functions are UPPERCASE ONLY |
| 15:09.12 | [TK]D-Fender | ${System( <- so this is 2 mistakes at once right off the bat |
| 15:09.25 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
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| 15:09.51 | wasanzy | there was no spaces yet it wasn't working that is why I put the space just to see |
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| 15:13.24 | wasanzy | now the problem is with this: Set(ic=${FIELDQTY(TOTAL,"")}) how do I represent space as delimiter? |
| 15:15.43 | [TK]D-Fender | "" is clearly not good. |
| 15:16.00 | wasanzy | five thousand and fifty |
| 15:16.11 | wasanzy | that is the string |
| 15:16.24 | wasanzy | space is the delimiter |
| 15:20.48 | wasanzy | core show function FIELDQTY did not mention space |
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| 15:22.34 | [TK]D-Fender | it doesn't have to. |
| 15:23.14 | wasanzy | [TK]D-Fender: so how do I represent space? |
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| 15:28.29 | [TK]D-Fender | wasanzy, What does the functions SYNTAX and SAMPLES show you? |
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| 15:43.49 | wasanzy | FIELDQTY(<varname>,<delim>) |
| 15:44.55 | Northcode_ | I'm trying to build asterisk 14 with opus but in the menuselect screen codec_opus is XXX'd. Opus and xmlstarlet is installed but it doesn't change anything. Could someone help? |
| 15:47.18 | Northcode_ | Oh nvm... needed to rerun ./configure after installing xmlstarlet... |
| 15:52.32 | [TK]D-Fender | wasanzy, So put it |
| 15:54.19 | wasanzy | [TK]D-Fender: That is what I put but I need help with how to show it is space |
| 15:54.38 | [TK]D-Fender | PUT IT AS A SPACE |
| 15:54.43 | [TK]D-Fender | You seem to be skipping the obvious |
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| 15:59.33 | wasanzy | [TK]D-Fender: Got it now. thanks. I think instead of 3, outputs 4. eg. five thousand and fifty |
| 15:59.56 | wasanzy | that is 3 fields, but it is saying 4 |
| 16:00.12 | wasanzy | same => n,Set(ic=${FIELDQTY(TOTAL, )}) |
| 16:09.32 | wasanzy | same => n,Set(fd[${i}]=${SHELL(echo ${TOTAL} | awk '{ print ${i} }')}) the awk, part, am I doing something wrong? |
| 16:11.26 | [TK]D-Fender | <wasanzy> five thousand and fifty |
| 16:11.35 | [TK]D-Fender | That's funny, I see FOUR words. |
| 16:13.22 | wasanzy | [TK]D-Fender: Yea bad of me |
| 16:14.06 | wasanzy | my concern now is the awk part |
| 16:15.55 | *** join/#asterisk friedrich (~friedrich@2a03:4000:7:5e::2) |
| 16:18.20 | [TK]D-Fender | So far there are a lot of variables I don't see |
| 16:18.57 | [TK]D-Fender | And we're not looking at what's actually happening |
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| 16:21.30 | wasanzy | I will pastebin the code |
| 16:21.34 | [TK]D-Fender | the CALL |
| 16:21.39 | [TK]D-Fender | you aren't looking at what is HAPPENING |
| 16:21.42 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 16:24.31 | wdoekes | wasanzy: you cannot compare awk-splitting (trimmed and by '\s+') with FIELDQTY splitting (by ' ') |
| 16:25.28 | wdoekes | $ echo ' abc d' | awk '{print NF}' |
| 16:25.28 | wdoekes | 2 |
| 16:25.35 | wdoekes | fieldqty counts 4 |
| 16:27.16 | wasanzy | am ok now |
| 16:27.20 | wasanzy | thank you all |
| 16:27.26 | wasanzy | will be back if something else happen |
| 16:29.18 | igcewieling | an AGI would make your life a lot easier. |
| 16:29.47 | wasanzy | igcewieling: an example will help |
| 16:30.21 | wasanzy | this is the code I was working on. https://paste.linux.community/view/6a38d527 |
| 16:31.07 | wasanzy | Now what I want to do is, I want to make it as a user defined function so that I can call it in other context to do what I want. Is that possible? |
| 16:31.23 | igcewieling | sample agi code to run this? t(fd[${i}]=${SHELL(echo ${TOTAL} | awk '{ print ${i} }')}) the awk, part, am I doing somet |
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| 16:34.29 | igcewieling | Set(ic=${FIELDQTY(TOTAL, )}) might be $ic = substr_count($total, " "); |
| 16:34.42 | wasanzy | igcewieling: oh I get you now |
| 16:34.50 | igcewieling | the while statement would be trivial |
| 16:35.31 | wasanzy | igcewieling: trivial in the sense that? |
| 16:35.44 | igcewieling | wasanzy: any time you need to run an external program (like awk) consider and AGI instead. |
| 16:36.10 | igcewieling | while ($bob < 5) { // do stuff } |
| 16:36.12 | [TK]D-Fender | Why is awk being used at all here? |
| 16:36.23 | igcewieling | I'm not going to write your AGI for you. |
| 16:36.23 | wasanzy | igcewieling: ok that is cool |
| 16:36.36 | [TK]D-Fender | well? Why? |
| 16:36.51 | wasanzy | igcewieling: am not asking you to write for me, don't worry |
| 16:36.52 | igcewieling | [TK]D-Fender: because it is the tool he knows. |
| 16:37.21 | igcewieling | I'd paste one of my agis, but they are always hundreds and thousands of lines long. |
| 16:37.22 | wasanzy | [TK]D-Fender: I will appreciate if you can show me anther way |
| 16:37.59 | [TK]D-Fender | What's the goal here? |
| 16:37.59 | wasanzy | igcewieling: ok |
| 16:38.03 | [TK]D-Fender | to parse out each word? |
| 16:38.43 | wasanzy | [TK]D-Fender: yes to parse out each word to PLAYBACK() |
| 16:38.52 | [TK]D-Fender | "core show function CUT" <--- Way to reinvent the wheel and place a giant load on your server for NOTHING |
| 16:38.56 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
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| 16:40.05 | wasanzy | Ok, I will look at CUT, I looked at it earlier on though |
| 16:40.12 | stefan27 | I'd avoid using a combo of shell one-liners and dialplan functions for implementing your dialplan string manipulation logic... If ${TOTAL} had value "Hello world" and ${i} has value 1 wouldn't asterisk send echo Hello World | awk '{print 1}' to the shell? |
| 16:40.29 | stefan27 | Oh sorry I wasn't scrolled down so i responded to something from half an hour ago |
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| 16:50.01 | [TK]D-Fender | Some people have a really hard time reading a simple list of APPLICATIONS and FUNCTIONS. |
| 16:50.49 | [TK]D-Fender | So far at least a half a dozen straight up errors including not know whether a given name is actually even one or the other, inability to follow the samples, etc |
| 16:50.53 | [TK]D-Fender | So yeah.... fun |
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| 16:53.53 | wasanzy | <PROTECTED> |
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| 16:58.28 | [TK]D-Fender | <[TK]D-Fender> "core show function CUT" <--- |
| 16:58.35 | [TK]D-Fender | MORE reading problems.... |
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| 17:03.15 | [TK]D-Fender | Try again |
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| 17:05.16 | wasanzy | I read the wiki http://www.voip-info.org/wiki/view/Asterisk+func+cut |
| 17:05.29 | [TK]D-Fender | <[TK]D-Fender> <[TK]D-Fender> "core show function CUT" <--- |
| 17:05.40 | [TK]D-Fender | Where did I ever send you to than ancient WIKI? |
| 17:06.01 | wasanzy | [TK]D-Fender: CUT(varname,char-delim,range-spec) |
| 17:06.07 | *** join/#asterisk mrspence (~Asterisk@68-115-161-74.static.ahvl.nc.charter.com) |
| 17:06.14 | wasanzy | not much info when I did core show function CUT |
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| 17:07.25 | [TK]D-Fender | well you are clearly NOT passing it what it is asking for |
| 17:07.46 | wasanzy | https://paste.linux.community/view/a2f099c7 |
| 17:07.57 | wasanzy | that is all I get when core show ... |
| 17:08.17 | [TK]D-Fender | And you are not passing what it tells you you have to pass |
| 17:09.27 | wasanzy | space is space in this case too or? |
| 17:09.46 | [TK]D-Fender | What does it TELL you you should pass to it? |
| 17:09.47 | igcewieling | you are passing a string not a variabl3 |
| 17:09.55 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 17:10.08 | igcewieling | classic n00b mistake. |
| 17:10.10 | [TK]D-Fender | <wasanzy> same => n,Set(fd[${i}]=${CUT(${TOTAL}, ,${i})}) <-- that is a VALUE, not a VARIABLE NAME |
| 17:10.31 | wasanzy | ${TOTAL} is variable |
| 17:10.48 | wasanzy | that contains the data |
| 17:11.05 | [TK]D-Fender | no |
| 17:11.10 | [TK]D-Fender | TOTAL is the variable |
| 17:11.17 | [TK]D-Fender | ${TOTAL} is NOT |
| 17:11.28 | [TK]D-Fender | that returns the CONTENTS |
| 17:11.28 | wasanzy | oh I see |
| 17:11.44 | wasanzy | I get it now. maybe the wiki confused me |
| 17:11.53 | [TK]D-Fender | <wasanzy> same => n,Set(ic=${FIELDQTY(TOTAL, )}) <-- you already did it HERE... and don't seem too awake |
| 17:11.59 | [TK]D-Fender | NAME. |
| 17:12.01 | [TK]D-Fender | not the VALUE |
| 17:12.17 | [TK]D-Fender | When it says variable NAME it is not looking for the TEXT it holds |
| 17:12.36 | wasanzy | oh ok |
| 17:12.52 | wasanzy | same => n,Set(fd[${i}]=${CUT(TOTAL, ,${i})}) |
| 17:13.24 | wasanzy | hope am not in trouble here : ${i} |
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| 17:17.58 | wasanzy | works. thanks [TK]D-Fender |
| 17:18.10 | wasanzy | I can't have user defined function? |
| 17:18.22 | wasanzy | or AGI is the way? |
| 17:20.13 | igcewieling | wasanzy: An AGI will let you write the program in whatever language you are most comfortable with. That must be a lot better than trying to learn the dialplan "language" |
| 17:25.03 | [TK]D-Fender | ~book |
| 17:25.06 | 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 |
| 17:25.08 | [TK]D-Fender | ^^^^ |
| 17:25.13 | [TK]D-Fender | How long have you been here? |
| 17:25.19 | [TK]D-Fender | Should this be a question? |
| 17:30.59 | mrspence | Anyone have any experience sending an audio file over multicastRTP using either the page or dial application? |
| 17:36.59 | *** join/#asterisk EnrgySmth (d8eba101@gateway/web/freenode/ip.216.235.161.1) |
| 17:50.08 | wasanzy | [TK]D-Fender: I have not read the entire book. I ask because I don't know |
| 17:50.22 | wasanzy | anyways can I download the 4th Edition? |
| 17:51.08 | [TK]D-Fender | Three is also the WIKI |
| 17:51.20 | wasanzy | please which chapter is related to my question? |
| 17:51.20 | [TK]D-Fender | 4th edition is not free last I checked |
| 17:51.42 | [TK]D-Fender | user defined function might as well be a GOSUB |
| 17:51.51 | [TK]D-Fender | You should already knwo your dialplan basics |
| 17:52.02 | [TK]D-Fender | * dialplan is not a "real" langue. |
| 17:52.14 | [TK]D-Fender | If yyou want better parsing then everything is extternal. AGI or ARI. |
| 17:52.20 | [TK]D-Fender | both well documented. |
| 17:52.36 | wasanzy | <PROTECTED> |
| 17:52.44 | [TK]D-Fender | All on the WIKI |
| 17:53.08 | wasanzy | the wiki I sent earlier you said is outdated, which one is the current one? |
| 17:53.22 | [TK]D-Fender | The official Asterisk one you've been linked tto many many times |
| 17:53.46 | [TK]D-Fender | You'd think there would be a link on the ASTERISK web-site or something... |
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| 18:00.32 | wasanzy | https://wiki.asterisk.org |
| 18:00.59 | [TK]D-Fender | WOW! |
| 18:01.59 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 18:03.21 | WIMPy | BTW: Is there any theory why Asterisk downloads are only gzipped? |
| 18:04.12 | file | it's just what the person who did the release process made it do? |
| 18:04.12 | wasanzy | I think GOSUB will help me |
| 18:04.44 | WIMPy | Migt make downloads a lot faster if that was a little modernized. |
| 18:04.54 | *** join/#asterisk tompaw (~tompaw@tompaw.xxx) |
| 18:05.32 | file | the release scripts are publicly available and we accept changes |
| 18:06.03 | WIMPy | s/gzip/xz -9/ |
| 18:06.14 | file | you'd have to do both |
| 18:06.21 | file | we can't just abruptly change |
| 18:07.08 | WIMPy | Does anything automatically generate download filenames? |
| 18:07.12 | *** join/#asterisk Tiffon (~name@unaffiliated/tiff0n) |
| 18:07.49 | file | the release script generates the filenames and such, but outside consumers depend on the existing .gz |
| 18:07.58 | file | thus why it can't just go away |
| 18:08.25 | WIMPy | Just out of interest: Where does that happen? |
| 18:09.27 | file | https://gerrit.asterisk.org/#/admin/projects/repotools is the repository that contains the scripts and other assorted things |
| 18:09.35 | file | mkrelease.py being the major one |
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| 18:10.13 | WIMPy | No I mean things that depend on filenames ending in .gz. |
| 18:10.30 | file | automated scripts people use to build RPMs or deploy Asterisk |
| 18:10.40 | igcewieling | indeed! |
| 18:10.46 | file | some people use the -current gz to grab the latest version automatically |
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| 18:14.01 | WIMPy | Seems very minor compared to other things that break regularly. But OTOH that's only one file. |
| 18:14.16 | file | no, it's not "very minor" |
| 18:14.47 | file | breaking something like that (removing gz files) is not something we would ever do to our users unless we absolutely had to |
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| 18:15.34 | wasanzy | ~wiki |
| 18:15.50 | file | is rather passionate when it comes to breaking changes anywhere |
| 18:17.34 | WIMPy | In the past years I had the ompression that breaking things has become very fashionable. |
| 18:17.59 | WIMPy | impression coff orse |
| 18:18.19 | file | we try our best not to introduce breaking changes into Asterisk |
| 18:18.25 | WIMPy | And often for much less benefit. |
| 18:18.34 | [TK]D-Fender | ~asteriskwiki |
| 18:18.34 | infobot | asteriskwiki is probably http://wiki.asterisk.org |
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| 18:48.30 | pjensen00 | I have a raw PCM audio file and the playback is BRRGHEGHIEG. It's named "test_tts.pcm". I see that asterisk uses the .sln exention for signed linear PCM, so changing the file extension to .sln (default 8k sample rate) made it work perfectly |
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| 18:48.46 | pjensen00 | My question is - what was asterisk playing the audio file with the .pcm extension as? |
| 18:50.56 | WIMPy | 'core show file formats' |
| 18:51.51 | pjensen00 | ah, neat. |
| 18:52.29 | pjensen00 | So it was deciding it was a ulaw file |
| 18:52.36 | pjensen00 | thanks |
| 18:58.01 | igcewieling | pjensen00: also see "core show translations" |
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| 19:04.59 | Centinel | One of our endpoints is set up with nat=yes since it's located remotely, but evidently RTP traffic is still being sent to the wrong address - the endpoint's local IP address. |
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| 20:55.35 | pjensen00 | When playing a sound file back in asterisk, is the entire file loaded into memory or does it require the file to exists throuought the entire playback? |
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| 20:56.15 | pjensen00 | *exist throughout |
| 20:57.38 | theGoat_ | so here's an intersting one. I have a cisco 7921 connecting to asterisk via sccp. When asterisk dies, i have my phones fail over to my voice gateway which is my cisco router via SRST. so when asterisk comes back, all my phones will fail back to asterisk, but my 7921 stays in SRST, has anyone else have a similar issue? |
| 21:02.28 | igcewieling | pjensen00: in linux/unix if a file is open when it is deleted, the file is not actually deleted until nothing has the file open anymore. |
| 21:03.24 | pjensen00 | Huh... thanks for the datas |
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| 21:07.48 | igcewieling | pjensen00: http://stackoverflow.com/questions/2028874/what-happens-to-an-open-file-handler-on-linux-if-the-pointed-file-gets-moved-de |
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| 21:17.54 | danielyk | Hi! When will Asterisk 13.14.0 come? |
| 21:18.12 | [TK]D-Fender | wHEN IT'S READY |
| 21:18.39 | danielyk | I mean, when is it scheduled? |
| 21:18.47 | [TK]D-Fender | When it's ready |
| 21:19.20 | danielyk | You do not want me to understand... |
| 21:20.01 | [TK]D-Fender | I do want you to understand |
| 21:20.13 | [TK]D-Fender | It'd help if you understood what I said |
| 21:20.24 | [TK]D-Fender | This is OPEN SOURCE SOFTWARE |
| 21:20.31 | [TK]D-Fender | There is *NO* schedule |
| 21:20.34 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 21:20.43 | [TK]D-Fender | What you are asking for does not exist |
| 21:20.48 | [TK]D-Fender | there is no date on a piece of paper |
| 21:20.52 | [TK]D-Fender | there is no commitment to a date |
| 21:21.09 | [TK]D-Fender | It is released WHEN IT IS READY. |
| 21:21.16 | [TK]D-Fender | Clearer now? |
| 21:21.24 | danielyk | yes, thanks |
| 21:21.31 | [TK]D-Fender | :) |
| 21:21.41 | danielyk | ;) |
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| 21:21.47 | [TK]D-Fender | "When itt is ready" clearly means "no date" |
| 21:22.02 | [TK]D-Fender | If it takes a MONTH... then it takes a month. If it's in 2 hours, then it's in 2 hours |
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| 22:51.11 | mknooihuisen | I've been googling around but I can't find much info on this problem. I'm trying to originate calls and playback "Hello, World" as you do when testing, but when I answer the call the message has already started and I only hear part of it. Any thoughts? |
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| 22:59.23 | Snwspeckle | I'm seeing some latency in call disconnect callbacks and I was wondering if there are any suggestions to solve for this? |
| 23:03.00 | igcewieling | mknooihuisen: are you using analog lines? |
| 23:03.52 | mknooihuisen | Only as the recipient of the call |
| 23:04.09 | igcewieling | analog lines are considered "answered" as soon as dialing finishes. |
| 23:04.19 | igcewieling | you are using a SIP provider? |
| 23:04.25 | mknooihuisen | Yes, Flowroute |
| 23:04.56 | igcewieling | are you answering anywhere in your dialplan? |
| 23:05.44 | mknooihuisen | Well, I just switched the originate call to send it to an ext@context rather than just an application |
| 23:05.50 | mknooihuisen | So yes, now |
| 23:06.03 | [TK]D-Fender | <igcewieling> analog lines are considered "answered" as soon as dialing finishes. <- depends |
| 23:06.23 | igcewieling | [TK]D-Fender: close enough for this discussion |
| 23:06.49 | igcewieling | mknooihuisen: you'll have to pastebin the output of the problem calls and hope someone wants to help. |
| 23:09.29 | igcewieling | don't know how much help this will be. |
| 23:09.31 | igcewieling | http://pastebin.com/bzgSvzSx |
| 23:10.16 | igcewieling | [incoming] context (not shown) is where a Dial() to the destination number happens. |
| 23:13.20 | mknooihuisen | igcewieling, how are lines 3-10 used, may I ask? |
| 23:14.12 | mknooihuisen | I read something about .call files that looked similar to that, but I wasn't sure how they were implemented or how they helped. |
| 23:14.35 | igcewieling | that is a sample of a call file. |
| 23:14.50 | igcewieling | you must be using Originate, but call files work similarly |
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| 23:15.40 | igcewieling | watch your asterisk console, it should tell you where the call is answered. |
| 23:18.55 | mknooihuisen | Based on SIP debug, it appears to be around the time the call rings |
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