| 00:19.33 | *** join/#asterisk JeffC_NN (32cad19e@gateway/web/freenode/ip.50.202.209.158) |
| 00:21.06 | JeffC_NN | I'm trying to troubleshoot an asterisk server that becomes unresponsive. There are 2 threads with medium CPU usage, and the server doesn't respond to SIP requests. |
| 00:21.36 | *** join/#asterisk Akuma (~Akuma@apn-31-1-20-182.dynamic.gprs.plus.pl) |
| 00:21.36 | JeffC_NN | It's happening daily, very low call volume. It's a remote branch that trunks over IAX2, but uses SIP at the branch. |
| 00:23.14 | JeffC_NN | I'm running Asterisk 13.0.1 on Ubuntu 14.04.1 |
| 00:30.35 | *** join/#asterisk Valduare (~Valduare@108-198-116-80.lightspeed.mdsnwi.sbcglobal.net) |
| 00:31.05 | ChannelZ-Wk | Quickest place to start, do you monitor the console? Is there any external scripts you call that ar misbehaving, or are calls getting stuck, etc. |
| 00:31.09 | ChannelZ-Wk | sip show channels |
| 00:56.38 | JeffC_NN | There are 2 channels that are active, Sip to IAX, but the call itself is no longer in progress |
| 01:00.35 | *** join/#asterisk Dovid (~Dovid@ool-2f113961.dyn.optonline.net) |
| 01:00.50 | newtonr | JeffC_NN, if you have logs running with warning,error, notice, etc, you might see if you see errors similar to what this guy sees https://issues.asterisk.org/jira/browse/ASTERISK-24478 |
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| 01:09.05 | JeffC_NN | I'm trying to reproduce the issue. It seems to happen every day after normal use of the server, a handful of calls per day. |
| 01:28.48 | *** join/#asterisk pankid (~quassel@108-223-192-152.lightspeed.livnmi.sbcglobal.net) |
| 01:29.03 | JeffC_NN | I'm having this issue on Asterisk 13: |
| 01:29.04 | JeffC_NN | [Dec 4 17:27:30] NOTICE[1724]: acl.c:748 ast_apply_acl: IAX2 user ACL: Rejecting '<censored>' due to a failure to pass ACL '(BASELINE)' |
| 01:29.09 | JeffC_NN | (IP removed) |
| 01:30.20 | JeffC_NN | But afaik, I have no ACL restrictions. I've tried adding permit = 0.0.0.0/0.0.0.0 with the same result |
| 01:32.30 | MarkS- | JeffC_NN: do you have "deny" in your sip.conf? |
| 01:33.23 | JeffC_NN | (this is IAX, but I don't have it in either) |
| 01:33.35 | MarkS- | missed that part, sorry |
| 01:33.59 | MarkS- | no idea what it could be in that case |
| 01:34.26 | JeffC_NN | Yeah I haven't found any help online :( |
| 01:34.39 | newtonr | JeffC_NN, yeah double check that you don't have any permit, deny or acl keywords in the file |
| 01:34.56 | newtonr | maybe accidentally uncommented |
| 01:34.59 | newtonr | or in an included file |
| 01:36.23 | JeffC_NN | funky. I looked again, and found one in a different IAX user context. removing it seems to have fixed this part of the issue |
| 01:36.53 | JeffC_NN | I also put in some extra permit = 0.0.0.0/0.0.0.0, and that may have been it as well |
| 01:41.46 | newtonr | cool |
| 01:49.25 | JeffC_NN | ok. I can replicate my IAX issue where Asterisk stops responding and 2 threads are at 15% cpu without any calls |
| 01:50.38 | JeffC_NN | I just spammed IAX channels, and eventually I stop getting audio (~70 channels) and then with SIP debug and IAX debug, I see no IAX messages, but a TON of application/pidf+xml presence stuff even though there aren't any presence changes |
| 01:52.04 | newtonr | hrm |
| 01:52.33 | JeffC_NN | and now the presence stuff has stopped, but the 2 threads are still there, and I can't get calls over IAX still |
| 01:53.41 | newtonr | hmm |
| 01:53.47 | newtonr | there is this one recently https://issues.asterisk.org/jira/browse/ASTERISK-24540 |
| 01:53.51 | newtonr | but it appears SIP related |
| 01:54.01 | newtonr | chan_sip specifically |
| 01:54.54 | JeffC_NN | any way to see what these 2 threads are doing? |
| 01:55.06 | JeffC_NN | I haven't compiled for debug |
| 01:55.48 | newtonr | uhh |
| 01:55.52 | newtonr | "core show threads" I think |
| 01:56.09 | newtonr | I'm not a real developer though so I can't parse it very well |
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| 01:56.27 | JeffC_NN | aha, thanks |
| 01:57.26 | newtonr | if it looks like it is getting into a locked state where you can't start new calls or anything, then you might want to recompile with DEBUG_THREADS and reproduce, get "core show locks" output and a backtrace after it happens and then file a bug |
| 01:57.29 | JeffC_NN | once is 3 below the lowest PID on the list of core show threads, and the other is network_thread |
| 01:58.44 | newtonr | https://wiki.asterisk.org/wiki/display/AST/CLI+commands+useful+for+debugging |
| 01:58.56 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationForADeadlock |
| 01:59.41 | JeffC_NN | I assume I can just make menuselect, make, make install over my existing install? |
| 02:00.23 | newtonr | yah |
| 02:01.46 | JeffC_NN | hmm, looks like I need BFD(E), DLADDR(E) for the BETTER_BACKTRACES option. ideas? |
| 02:02.36 | newtonr | what distro do you use? they should be in most distro repos |
| 02:02.45 | JeffC_NN | Ubuntu 14 |
| 02:04.21 | JeffC_NN | having trouble finding package name for apt-get |
| 02:05.01 | newtonr | yeah i'm looking too |
| 02:05.49 | JeffC_NN | installed gdb, seeing if that helps me |
| 02:06.22 | newtonr | try the install_prereq script under contrib/scripts (in your asterisk source) |
| 02:06.39 | JeffC_NN | ah, helps if I read. it says Use libbfd (GPL) to generate better inline backtraces |
| 02:07.10 | JeffC_NN | binutils-dev |
| 02:07.34 | newtonr | yeah and i see that gets install with the install_prereq script |
| 02:08.01 | JeffC_NN | oh. nice. I had trouble with that script. It just said "abort" at the end after listing the packages that would be installed |
| 02:08.22 | newtonr | I haven't tried it in 14. it worked in 12.04 |
| 02:09.16 | newtonr | you can copy paste the libraries from the script :D |
| 02:10.18 | JeffC_NN | yeah, that's what I ended up doing (except I was on a headless server, so it was good ol' fashioned read and type, hahaha) |
| 02:12.02 | JeffC_NN | wow, with debug stuff on, it's WAY easier to reproduce :) |
| 02:12.19 | newtonr | lol i occasionally see problems like that |
| 02:13.12 | newtonr | the opposite is more fun, the problem stops occurring when various debug or compiler flags are enabled |
| 02:21.44 | newtonr | Okay I'm out. Good luck. If you file it, be sure to describe reproduction as well if that is possible and include logs of the calls if you can. |
| 02:21.50 | newtonr | Later! |
| 02:22.04 | JeffC_NN | thx for all your help. gotta run, but I have the files to submit a bug tomorrow :) |
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| 03:34.47 | Valduare | so I was scrolling up and the other day guys were talking about attacks on their asterisk servers? |
| 03:34.55 | Valduare | didnt know that was a âthingâ :P |
| 03:35.15 | scv | rather common |
| 03:35.20 | Mango45 | Quite common. |
| 03:35.35 | Valduare | what do they gain from doing it |
| 03:35.51 | scv | abuse of your telco services? |
| 03:36.32 | Mango45 | They'll route calls via your PBX - and either charge their customers for what they get for free, or they'll call high-cost numbers that they get compensation for. |
| 03:36.40 | ChannelZ | I get tons of attempted calls to Palestine. |
| 03:36.50 | scv | 37.? |
| 03:36.51 | scv | :) |
| 03:37.29 | ChannelZ | and right on queue, there went one in my console |
| 03:37.35 | scv | acl the heck out of that |
| 03:38.17 | ChannelZ | Ah, this one seems to be from the US for a change. 107.182.20.226 |
| 03:38.57 | Valduare | what all do you have to do to combat it |
| 03:39.05 | scv | firewalling |
| 03:39.07 | scv | acls |
| 03:39.12 | Valduare | ban on ip? |
| 03:39.17 | Valduare | what is acls |
| 03:39.30 | scv | asterisk allows you to list what ip addresses are allowed to connect to the pbx |
| 03:39.55 | scv | https://wiki.asterisk.org/wiki/display/AST/Named+ACLs |
| 03:40.03 | Valduare | btw im new to this - I dont have a server or anything yet im just learning |
| 03:40.19 | scv | well this is a very important step that a lot of people miss |
| 03:40.21 | Valduare | but isnt asterisk an inbound server suppose to take calls from anywhere |
| 03:40.48 | ChannelZ | 011 972.. Israel |
| 03:43.44 | ChannelZ | Valduare: only if you really want to |
| 03:44.19 | Valduare | I was looking at using this to handle my google voice number for the business and then be able to route calls to key members cell phones |
| 03:44.22 | Valduare | that possible? |
| 03:44.32 | ChannelZ | It's actually fairly rare get (legitimate) SIP calls from randoms (someone other than your ITSP) |
| 03:45.01 | ChannelZ | s/get/to get/ |
| 03:46.19 | ChannelZ | Yes, for as long as Google Voice continues to work outside of Google (I really don't know what the status of that is) |
| 03:46.43 | scv | i was under the impression they no longer offered a sip gateay |
| 03:46.44 | Valduare | at that point I can still port the number |
| 03:46.45 | scv | gateway |
| 03:46.50 | ChannelZ | they never had a SIP gateway |
| 03:46.54 | Valduare | so its fairly safe to establish a google voice number for your business. |
| 03:47.05 | scv | ChannelZ: they did quite some time ago |
| 03:47.10 | scv | shortly after the grand central transition |
| 03:47.14 | scv | it wasn't publicly documented. |
| 03:47.18 | ChannelZ | after or before? |
| 03:47.30 | scv | just after |
| 03:47.33 | scv | for a few months |
| 03:47.58 | ChannelZ | I didn't know that. But in any event it was more widely accessable via XMPP |
| 03:48.40 | ChannelZ | Supposedly they were going to close the protocol/servers off to prevent all the abuse but I've heard people say it's still working. |
| 03:49.11 | ChannelZ | so it could go at any moment, or never. |
| 03:50.31 | ChannelZ | not necessarily something I'd depend my own business on, a 'free' telecom service, but.. |
| 03:51.17 | Valduare | been working fine for few years for us now. |
| 03:51.27 | Valduare | and I am assured that i can port the number out of google |
| 03:51.42 | Valduare | we are not a high call volume place anyhow - |
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| 06:01.39 | Penguin | I guess that was when they bought Gizmo5. I used it until it was shut down. |
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| 09:55.12 | Fira | Hey :) I'm interested in using Asterisk with the Opus Codec. I saw that it previously was completely unsupported and tried to apply the Asterisk 11 patch to Asterisk 13, but now I see that Opus passthrough has been added.. Assuming all the VoIP clients talk Opus, do I still need transcoding support for anything ? |
| 09:56.09 | WIMPy | You need it of you want to play messages that aren't recorded in the same format. |
| 09:56.24 | ThoMe | hihio |
| 09:56.29 | Fira | Oh. I see. Recording aswell then. Makes sense. |
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| 09:56.30 | WIMPy | Where I'm not sure Asterisk would read them. |
| 09:56.37 | ThoMe | can anyone help me with macro and catch? |
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| 10:55.18 | pbxman | hello How can I play something in the background and go on with the dial plan? |
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| 10:56.07 | pbxman | is that even possible to do with Asterisk? |
| 10:56.45 | WIMPy | StartMusicOnHold ist the only thing in that direction. |
| 11:00.52 | pbxman | thank you WIMPy |
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| 11:28.16 | pecanha | Hello all, I have monitor-type = MixMonitor set at queues.conf for my queue group. How can I change its storage path? |
| 11:44.55 | kleszcz | people never read a documentation or file conf. On queues.conf TYPE MIXMONITOR OPTIONS |
| 11:47.16 | Lycnixd | ooh, when did LTS get released? |
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| 12:50.39 | ThoMe | hiho |
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| 13:23.32 | stefan27 | How are the sip.conf per peer "icesupport=yes" and rtp.conf (global) icesupport=true related? |
| 13:24.21 | pa | WIMPy, ping :) |
| 13:24.40 | stefan27 | does icesupport=true have to be enabled in rtp.conf in order for icesupport=yes to work in sip.conf? |
| 13:25.16 | pa | do you know what chips would support chan_capi? |
| 13:25.31 | pa | or any other asterisk native isdn channel? |
| 13:25.50 | pa | i concluded that perhaps changing card would be the least of the pains |
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| 13:52.38 | file | stefan27, yes |
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| 14:02.05 | ThoMe | file: hihio |
| 14:02.20 | file | ...hi |
| 14:02.26 | ThoMe | when i recieve a fax via "ReceiveFAX" then asterisk hangup after, sometimes |
| 14:02.43 | ThoMe | when i use catch h, then jump sometimes in h and sometimes not. |
| 14:02.54 | ThoMe | how i can debug this? |
| 14:03.05 | ThoMe | (I would like send after recieve fax an email) |
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| 14:34.22 | pederindi | a question about hold musi |
| 14:34.24 | pederindi | music* |
| 14:34.48 | pederindi | I'm trying to get creative commons music, but it says that needs attribution, this is needed in the same piece of audio (hold music)? |
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| 14:38.12 | [TK]D-Fender | That means don't distribute without their license file / reference material concerning it's origin |
| 14:40.10 | vassilux | I try to use oslec. I set echocancel=256 but when execute dahdi show channel 1 , I got Echo Cancellation : 1 taps. Is is possible to change trazps with oslec ? |
| 14:40.22 | vassilux | sorry taps |
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| 15:11.00 | Fira | Hey guys, i'm hacking around with codecs but i have no idea how to actually test them in Asterisk :p... I'm trying to get two calls bridged together, I made an extension which calls confbridge(conference) but it seems only one call can get to it at any time. Is that normal ? Can someone tell me what the hell i'm doing wrong ;-; ? |
| 15:12.03 | [TK]D-Fender | Show us the actual failure |
| 15:12.07 | [TK]D-Fender | ~pb |
| 15:12.07 | 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. |
| 15:12.09 | [TK]D-Fender | ^^^ |
| 15:12.52 | Fira | Nothing much to show, first call goes through, next call is answered and immediately hanged up |
| 15:13.00 | [TK]D-Fender | Show us the calls |
| 15:13.15 | Fira | I'll turn off automated join messages to be sure cause i don't have proper translators for them and it keeps spitting random errors... |
| 15:14.08 | [TK]D-Fender | For reference, "testing" means "doing". |
| 15:26.18 | Fira | actually it probably doesn't matter because i got the thing all wrong :) |
| 15:28.36 | Fira | i just have this in extensions.conf |
| 15:28.37 | Fira | https://gist.github.com/fira/95a21a51abe6daeb23e7 |
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| 15:31.29 | Fira | right now the first call to 500 is bridged, but a second call puts the first one on hold |
| 15:32.18 | [TK]D-Fender | [10:12][TK]D-FenderShow us the actual failure |
| 15:33.41 | Fira | What do you mean by failure >-> |
| 15:34.27 | [TK]D-Fender | Fira: Show us that call's CLI output where we see it actually FAIL. |
| 15:34.28 | TSM | anyone looking for a used Sangoma A101D PCI-E EchoCanceller card (UK) |
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| 15:35.25 | Fira | Doesn't spit anything on CLI, which is why i assume this is just me doing it wrong |
| 15:35.50 | [TK]D-Fender | Fira: You showed us only 1 little dialplan snippit. We don't acutally know if it's even getting used, and regardless is comparable to showing us a showrrom photo of your car before you drove it off the lot and asking us why you crashed sintead of showing us actual video footage. |
| 15:36.00 | TSM | ive also got a Sangoma A200 + 1xFXS-2 module PCI-E |
| 15:36.12 | [TK]D-Fender | [10:13]FiraI'll turn off automated join messages to be sure cause i don't have proper translators for them and it keeps spitting random errors... <--- where are these errors? |
| 15:36.24 | [TK]D-Fender | If you're placing a call then you should be seeing it in CLIU |
| 15:36.52 | *** join/#asterisk crised (~crised@186.67.181.203) |
| 15:37.14 | crised | Any tip on reflashing Cisco 7961 to SIP firmware? |
| 15:37.23 | Fira | that's just me doing crap, i'm trying to reimplement the opus transcoder for Ast 13 but because there's no slin16 translator it can't playback sounds (yet) |
| 15:37.43 | [TK]D-Fender | Fira: That'll do it |
| 15:37.53 | [TK]D-Fender | Fira: codec != format |
| 15:39.09 | Fira | .. |
| 15:39.34 | [TK]D-Fender | Sound files are played back using format_XXX modules. |
| 15:39.57 | [TK]D-Fender | (and read) |
| 15:40.10 | Fira | but I can playback slin and have it transcoded into opus ? i'm talking about the opus transcoder fork here, not passthrough |
| 15:40.29 | *** part/#asterisk mdhas (~mdhas@206.223.170.186) |
| 15:40.47 | [TK]D-Fender | that shoud work if the file is not in your native codec |
| 15:41.09 | [TK]D-Fender | has your modeul loaded and shows up in the translation matrix? |
| 15:41.13 | *** join/#asterisk jamesc (~kvirc@mail2.inetplc.com) |
| 15:42.28 | jamesc | Hi it does not seem possible to GoTo priorities in the h exten , is this a bug ? |
| 15:43.02 | Fira | module is loaded, i believe it tries to transcode, but i'm getting "no samples for.." errors. how do i show the translation matrix ? googling around says show translation but i must be missing a module here.. |
| 15:44.58 | [TK]D-Fender | Fira: You mean "missing mine"? |
| 15:46.07 | Fira | very funny. |
| 15:46.35 | [TK]D-Fender | I'm serious. If it just missing yours, or not functioning at all? |
| 15:46.38 | [TK]D-Fender | Is* |
| 15:47.49 | Fira | the module is up, i'm not getting any encoding path errors |
| 15:48.24 | Fira | i tried to hack around the samples counting with what i could find since it seems it changed from Ast 11 -> 13 and apparently i just did it bogus |
| 15:48.40 | [TK]D-Fender | Fira: Given the nature of what you're doing you should definitely take this to #asterisk-dev |
| 15:48.54 | [TK]D-Fender | Fira: They are far better equipped to assist you there... |
| 15:49.21 | Fira | k thanks |
| 15:50.05 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 15:51.40 | [TK]D-Fender | Fira: Want you to the best, most focused support for this |
| 15:51.45 | [TK]D-Fender | to get* |
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| 15:54.49 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 16:11.21 | Fira | [TK]D-Fender: i was actually firing calls from the same softphone so it auto put on hold them so only one was active at a time... took 30 minutes to realize *facepalms* |
| 16:11.28 | Fira | not like the transcoding works anyway :p |
| 16:11.51 | [TK]D-Fender | Talking to yourself = bad... |
| 16:12.06 | [TK]D-Fender | (arguing with yourself and LOSING = worse) |
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| 16:20.38 | stefan27 | i have a sip peer A with transport WSS talking to a normal snom sip phone B with transport udp. when A gets powerfailure I want the call to be killed by asterisk as soon as possible, but now it only happens after x=60 seconds where x=rtptimeout in sip.conf for A ... is there some other configuration I can do so that when asterisk detects that the WSS connection is no longer ESTABLISHED it |
| 16:20.38 | stefan27 | kills all calls to that device? |
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| 17:46.29 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 17:47.10 | ChannelZ-Wk | Huh.. I've just noticed I've been getting CNAMs from my telco with upper and lower case letters recently.. and this is POTS. I didn't think that was possible? |
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| 18:25.55 | Mango45 | It is technically possible, though many telcos just shout out the CNAM. |
| 18:26.44 | Mango45 | looks at his call records |
| 18:27.12 | Mango45 | Looks like mine is a mixture. |
| 18:35.47 | ChannelZ-Wk | Scrolling back a bit it just started happening to me on the 4th. |
| 18:40.24 | ChannelZ-Wk | It doesn't matter much I guess, I already run a little script that cleans up caller ID and does initial caps (and overrides some from a local DB) but I just happened to notice this morning what I initially receive has changed. |
| 18:50.43 | *** join/#asterisk picard276 (80547c88@gateway/web/cgi-irc/kiwiirc.com/ip.128.84.124.136) |
| 18:50.59 | picard276 | hey guys i am having a problem connecting my Cisco 7940 phones to my asterisk machine |
| 18:51.04 | picard276 | im doing an upgrade from asterisk 1.4 to asterisk 13 |
| 18:51.20 | picard276 | currently i have all outbound calls working correctly... but any inbound calls just goes directly to voicemail... |
| 18:52.23 | picard276 | i noticed if i do sip show peers (on the old system) i see that the port says 5061... Does this mean that i shoulud setup my asterisk sip settings to be listening on 5061 or still listening on 5060 but the phones are listening on 5061? |
| 18:52.28 | [TK]D-Fender | And what is the call doing? |
| 18:52.36 | ChannelZ-Wk | What does the console say? |
| 18:52.45 | picard276 | the phoens are not registering |
| 18:52.54 | [TK]D-Fender | the PEER port has nothing to do with *'s bindport |
| 18:53.01 | ChannelZ-Wk | It means your peer "came from" 5061.. the port asterisk should talk to _it_ on |
| 18:53.02 | [TK]D-Fender | then that's your problem. |
| 18:53.28 | picard276 | so does that mean the phone is trying to register on port 5061 to the asterisk system? |
| 18:53.44 | picard276 | or does that mean that the asterisk system is trying to send to 5061 on the phone? |
| 18:53.46 | ChannelZ | no; asterisk wouldn't know that |
| 18:53.58 | WIMPy | Don't say "on". There's a from and a to. |
| 18:54.12 | picard276 | ok...the problem is the phones are not registering with asterisk correctly |
| 18:54.34 | ChannelZ | It means the source port of the phone was 5061 and asterisk will try to send it traffic there. |
| 18:55.06 | picard276 | ok... so on my old system i see 5061 and then status is ok |
| 18:55.50 | ChannelZ | When you upgraded *, did you do other system changes as well (linux upgrade, change firewalls, routing...) |
| 18:56.04 | ChannelZ | wait.. your old system is still running? |
| 18:56.32 | [TK]D-Fender | Nothing means anything yet |
| 18:56.38 | [TK]D-Fender | We don't SEE it trying to register |
| 18:56.55 | [TK]D-Fender | And it isn't registered now. |
| 18:57.04 | picard276 | yes ChannelZ.. i have only moved one phoen over |
| 18:57.14 | picard276 | so taht new phone is pointing to the new IP |
| 18:57.26 | picard276 | the new system is behind hte same firewall, just a different server IP |
| 18:58.16 | ChannelZ | well if you see no register attempts on the console for that phone, I'd check your work.. that you have the right IP all around |
| 18:58.35 | ChannelZ-Wk | oops |
| 18:58.58 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 19:00.10 | [TK]D-Fender | .... AND HE'S GONE |
| 19:01.40 | ChannelZ-Wk | Just like his phone calls |
| 19:03.01 | *** join/#asterisk picard276 (80547c88@gateway/web/cgi-irc/kiwiirc.com/ip.128.84.124.136) |
| 19:03.06 | picard276 | sorry my internet died |
| 19:03.23 | picard276 | i guess what i am asking is that 5061... this is something i do not setup on asterisk... |
| 19:03.34 | ChannelZ-Wk | it's the phone |
| 19:03.43 | *** join/#asterisk riess82 (~riessma@88-117-63-248.adsl.highway.telekom.at) |
| 19:03.44 | ChannelZ-Wk | the port of the phone |
| 19:03.46 | ChannelZ-Wk | not asterisk |
| 19:04.11 | picard276 | ok. |
| 19:04.22 | picard276 | so it is something then now where the phone can't register with asterisk 13 |
| 19:04.24 | ChannelZ-Wk | Did you totally reboot the phone after updating the proxy IP? |
| 19:04.26 | picard276 | but can register with asterisk 1.4 |
| 19:04.28 | picard276 | yes |
| 19:04.40 | ChannelZ-Wk | And did you see it re-register on the OLD system instead perhaps? |
| 19:04.56 | picard276 | no.. |
| 19:04.58 | picard276 | it does not |
| 19:05.26 | picard276 | i guess could it be something with chanSIP vs. pjSIP |
| 19:05.33 | picard276 | i tried going about disabling pjSIP |
| 19:05.38 | picard276 | and using chanSIP |
| 19:05.41 | ChannelZ-Wk | then you've either botched something in the phone's config such that it doesn't want to register at all, or your network is crazy |
| 19:05.57 | picard276 | but it should be nothing with asterisk, correct? |
| 19:06.03 | ChannelZ-Wk | You should only have one or the other loaded. Presumably chan_sip unless you've rebuilt your entire config for pjsip |
| 19:06.11 | picard276 | right... |
| 19:06.51 | ChannelZ-Wk | But with some console verbose on, if you reboot the phone and see no attempts of it registering, then you have some network problem. |
| 19:07.11 | picard276 | ok |
| 19:07.18 | picard276 | is there something i have to do to disable pjSIP? |
| 19:07.23 | picard276 | i made chanSIP the default 5060 port |
| 19:07.38 | ChannelZ-Wk | Either the traffic is not making it to that server, or it's being blocked from entering, or something else has clogged up port 5060 such that asterisk isn't even listening to it. |
| 19:08.19 | ChannelZ-Wk | make sure you don't have pjsip.conf in /etc/asterisk |
| 19:08.30 | picard276 | kk |
| 19:08.50 | picard276 | ill debug and see if i can find any other issues.. it might be pjsip being a bitch |
| 19:08.54 | WIMPy | Uness you're into interesting setups, it's bet to noload either of the sip modules. |
| 19:08.54 | picard276 | thanks! |
| 19:09.08 | WIMPy | s/bet/best/ |
| 19:09.11 | ChannelZ-Wk | and yeah probably noload chan_pjsip at minimum |
| 19:09.20 | picard276 | kk |
| 19:11.02 | ChannelZ-Wk | curious though you said earlier 'outgoing calls work' (presuming phone -> asterisk -> world) but that incoming didn't (asterisk -> phone) |
| 19:11.21 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 19:12.31 | ChannelZ-Wk | which would mean asterisk is seemingly getting SIP from the phone OK, but unable to send to it. |
| 19:13.03 | picard276 | correct |
| 19:13.05 | ChannelZ-Wk | Do you have the peer in sip.conf set to an IP/hostname or is it indeed dynamic? |
| 19:13.16 | picard276 | it is dynamic |
| 19:15.13 | ChannelZ-Wk | Is it type=user or type=friend then? Otherwise I don't know why Asterisk would accept a call from it in the first place, if the phone never registered and asterisk didn't know its IP |
| 19:15.48 | picard276 | no its a peer on the old system |
| 19:15.53 | picard276 | on the new system it should be the same |
| 19:15.55 | picard276 | let me check |
| 19:17.04 | picard276 | they are peers |
| 19:17.29 | picard276 | i just disabled the pjsip driver.. ill check to see if that makes a difference |
| 19:17.56 | ChannelZ-Wk | do what does 'sip show peers' look like on the new system that's half working? |
| 19:18.27 | picard276 | status UNKNOWN (unspecified) |
| 19:18.39 | picard276 | dyanmic D |
| 19:19.43 | picard276 | also... is there anyway to enable better debugging in asterisk 13 in asterisk 1.4 all i have to do is "asterisk -r" to get all the messages for a call etc... but in in 13.. i never see the call info with asterisk -r |
| 19:19.54 | ChannelZ-Wk | Then for you to be able to make outgoing calls with it, you must either have guest access turned on (which is bad because apparently guests can make outgoing calls in your setup!) or it's matching some other user |
| 19:20.00 | ChannelZ-Wk | asterisk -rvvv |
| 19:20.04 | picard276 | kk thanks |
| 19:20.06 | ChannelZ-Wk | or 'core set verbose 3' |
| 19:21.05 | picard276 | alright ill take a look. Thanks for all the help.. when i figure it out ill report back |
| 19:21.11 | ChannelZ-Wk | You need to do some legwork to make sure the phone is talkng |
| 19:21.14 | ChannelZ-Wk | ok nevermind then |
| 19:21.37 | ChannelZ-Wk | (...talking to the right server which I doubt it is...) |
| 19:24.06 | crised | [TK]D-Fender: hi |
| 19:24.28 | crised | I remember you suggested me time ago, a good sip phone, can't recall the brand... |
| 19:27.52 | lvlinux | crised: there are many good SIP phones---what features do you need? |
| 19:28.18 | crised | lvlinux: not much, good quality |
| 19:28.26 | crised | voip.ms compatible |
| 19:28.46 | lvlinux | You mean direct connection to voip.ms or through Asterisk? |
| 19:28.52 | crised | direct |
| 19:29.13 | lvlinux | price range? |
| 19:29.17 | crised | lWhat is that HD voice |
| 19:29.25 | crised | lvlinux: show me options |
| 19:29.58 | lvlinux | HD Voice is the marketing term for high samplerate voice codec---like g.722, AMR-WB, etc. |
| 19:30.13 | crised | lvlinux: is it good? |
| 19:30.18 | crised | lvlinux: tell me model options |
| 19:30.28 | crised | Is yealink a good brand? |
| 19:30.38 | lvlinux | You don't get any advantage from it making PSTN calls, but for HDVoice to HDVoice calls it's great. |
| 19:30.41 | lvlinux | I like Yealink |
| 19:30.50 | lvlinux | Polycom is also highly regarded |
| 19:31.04 | lvlinux | And some people like Snom (I do). |
| 19:31.14 | crised | lvlinux: tell me good values options |
| 19:31.28 | lvlinux | well do you have a price range? Used or new? |
| 19:31.31 | crised | lvlinux: I'll only use one or two or 3 phones |
| 19:31.35 | crised | don't mind, can be used |
| 19:31.54 | crised | not sure the price range, the lower the better, but looking for good qualiuty |
| 19:32.15 | lvlinux | I personally really like the Yealink SIP-T46G, about $160 new. |
| 19:32.30 | lvlinux | But you can get good Snoms on eBay for around $50. |
| 19:32.50 | lvlinux | Something like a Snom 320 or 360 |
| 19:33.05 | lvlinux | Polycoms are all over the map depending on which models. |
| 19:33.09 | crised | lvlinux: SIP T46G looks like a computer |
| 19:33.23 | WIMPy | They ARE computers. |
| 19:33.25 | lvlinux | The 450/550/650 are nice---can get them around $100 or so. |
| 19:33.29 | lvlinux | yep they are :-) |
| 19:33.33 | crised | lol yes |
| 19:33.46 | crised | I just trashed a cisco 7961 |
| 19:33.52 | lvlinux | The Snoms actually run a Linux kernel :-) |
| 19:33.54 | crised | could not configure it for SIP |
| 19:34.03 | lvlinux | did you have the SIP image on it? |
| 19:34.03 | crised | maybe I should not do that... |
| 19:34.06 | WIMPy | So if you already hae a computer, you can just connect a handset or headset to it. |
| 19:34.07 | crised | lvlinux: yes |
| 19:34.13 | WIMPy | that's the ceapest option. |
| 19:34.17 | crised | WIMPy: yes |
| 19:34.30 | lvlinux | Ciscos are good phones but they sometimes don't play well with NAT |
| 19:34.38 | crised | lvlinux: wouldn't want to struggle with the long xml config |
| 19:34.43 | lvlinux | they work well in a local PBX situation though. |
| 19:34.54 | crised | ok |
| 19:35.04 | WIMPy | If price is of interest, I'd look for the Snom 320. It's a really good phone for little money. |
| 19:35.16 | lvlinux | I have a 7960 and 7940 in use here on my home * box. Have configured a couple 8961s remotely. |
| 19:35.21 | lvlinux | Yes I agree with Wimpy. |
| 19:35.52 | WIMPy | Well, "good" in VOIP ranges. |
| 19:35.58 | lvlinux | If you don't really have a preference, the 320 is a good phone, does lots of stuff, can be had for good prices, and are plenty available. |
| 19:36.00 | crised | lvlinux: I can untrush my cisco 7961g, but how to configure the XML? |
| 19:36.14 | crised | snom... is it germany? |
| 19:36.22 | lvlinux | yes |
| 19:36.27 | lvlinux | i believe so |
| 19:36.35 | WIMPy | It is |
| 19:36.50 | crised | made in germany? |
| 19:37.01 | WIMPy | Liek Yealink. |
| 19:37.13 | lvlinux | I don't think the 7961 will work well with voip.ms through your NAT though---probably a lot of trouble for unreliability. |
| 19:37.14 | crised | not made in CHina? |
| 19:37.25 | lvlinux | Yealink are Taiwan aren't they? |
| 19:37.26 | crised | lvlinux: so it will stay in the trash |
| 19:37.35 | WIMPy | The lable says "engeneered in Germany" :-) |
| 19:38.01 | WIMPy | Noone would produce anything in Germany. |
| 19:38.04 | lvlinux | like I said the Ciscos work fine with Asterisk, but they're kindof picky on the config. |
| 19:38.25 | lvlinux | Yes I'm sure the assembly is farmed out somewhere else. |
| 19:39.13 | crised | lvlinux: is the snom poe? |
| 19:39.19 | lvlinux | So yeah crised it sounds like the Snom 320 is going to be a good bet for you. |
| 19:39.21 | WIMPy | yes |
| 19:39.23 | lvlinux | Yes it does poe |
| 19:39.25 | crised | cool |
| 19:40.17 | crised | Is the snom 320 Made IN germany? not designed or not engineered... |
| 19:40.32 | crised | http://www.amazon.com/SNOM-Technology-snom320-Snom-Phone/dp/B000FIHMSK/ref=sr_1_2?ie=UTF8&qid=1417808181&sr=8-2&keywords=snom+320 |
| 19:40.46 | lvlinux | like we said---I doubt if much of anything is actually made in Germany---why do you ask? |
| 19:40.47 | WIMPy | Off course not. |
| 19:41.07 | crised | because it would really worth the money |
| 19:41.18 | WIMPy | If it was really made in Germany noone could afford it. |
| 19:41.23 | crised | es |
| 19:41.25 | crised | yes |
| 19:42.08 | crised | lvlinux: is $130 cheap for a sip phone/? |
| 19:42.09 | WIMPy | Or maybe you could because they probably wouldn't exist any more then. |
| 19:42.10 | lvlinux | DON'T pay $131 for a 320 |
| 19:42.18 | lvlinux | NO it's not cheap |
| 19:42.30 | lvlinux | not for what you can get them used for anyway. |
| 19:42.44 | lvlinux | Yes for a high end phone that is cheap but not for a mid-grade or low end |
| 19:43.03 | lvlinux | Like I said you can get the 320s on eBay for under $50 |
| 19:43.49 | crised | http://www.ebay.com/itm/Snom-320-/171539139287?pt=Business_Phone_Sets_Handsets&hash=item27f0878ad7 |
| 19:44.18 | lvlinux | yeah there you go |
| 19:44.32 | crised | cool, thanks man |
| 19:44.35 | crised | thanks WIMPy |
| 19:44.54 | lvlinux | http://www.ebay.com/itm/Snom-320-VoIP-Telephone-/171329846412?pt=US_VoIP_Business_Phones_IP_PBX&hash=item27e40dfc8c |
| 19:45.02 | lvlinux | there's one for $30 |
| 19:45.38 | lvlinux | without power supply---which you don't need if your gonna do poe |
| 19:46.24 | crised | offered $15 |
| 19:46.28 | crised | :P |
| 19:46.35 | *** join/#asterisk riess82 (~riessma@88-117-63-248.adsl.highway.telekom.at) |
| 19:46.42 | lvlinux | hehe i don't think they will go that low, but you never know :-) |
| 19:46.55 | crised | oops The handset makes a minor rattling noise when moved. |
| 19:48.49 | lvlinux | ehh there's just a piece of plastic loose in it. no big deal lol |
| 19:49.08 | crised | lvlinux: wouldn't the speaper be banged up |
| 19:49.31 | lvlinux | probably not |
| 19:49.48 | crised | ok |
| 19:50.04 | lvlinux | but anyway there's tons of them on ebay for reasonable prices, and they come up all the time. |
| 19:50.13 | crised | :-D |
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| 20:11.01 | lorsungcu | anyone know if it's possible to get a screenshot of a D40 screen via URL? |
| 20:14.31 | newtonr | malcolmd, ping |
| 20:17.27 | malcolmd | pong |
| 20:17.29 | malcolmd | yes, it's possible |
| 20:18.22 | malcolmd | http://[phone-ip]/cgi-bin/screen_shot?screen=main |
| 20:19.11 | WIMPy | What name is the 2nd screen? |
| 20:19.57 | malcolmd | if you've a D70 then it's http://10.24.74.201/cgi-bin/screen_shot?screen=blf for the side-screen |
| 20:20.09 | malcolmd | er..hah, insert your ip there, o'course |
| 20:20.38 | file | DoSes malcolmd |
| 20:20.50 | malcolmd | file: you're only hurting yourself when you do that :( |
| 20:21.19 | file | malcolmd, bah |
| 20:22.01 | malcolmd | you keep doing that and pretty soon we'll run out of electrons and then where will we be, hunh? |
| 20:23.23 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 20:23.44 | malcolmd | lorsungcu: works? |
| 20:28.16 | malcolmd | also..needs to be firmware >=1.3.0 for that to be possible; won't work prior |
| 20:29.36 | newtonr | neat, didn't know you could do that |
| 20:30.44 | malcolmd | it was tucked into one of the release notes for firmware - https://wiki.asterisk.org/wiki/display/DIGIUM/Firmware |
| 20:32.32 | *** join/#asterisk wolrah_ (~wolrah@24.239.210.140) |
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| 21:27.45 | *** join/#asterisk WooAf (18d4f2c3@gateway/web/freenode/ip.24.212.242.195) |
| 21:27.52 | WooAf | Hello all |
| 21:29.09 | WooAf | How I can reload my sip config on the web gui ? |
| 21:29.25 | WooAf | I edited the file directly but I want to load them on the gui |
| 21:33.06 | WIMPy | There are different web GUIs, but we don't support any of them here. |
| 21:33.31 | WooAf | I use Elastix |
| 21:33.32 | WIMPy | And wen you're using such a thing, editing files yourself is usually a bad idea. |
| 21:33.52 | WooAf | I just want to restore a backup sip.conf |
| 21:33.54 | WIMPy | Try #elastix |
| 21:33.58 | WooAf | Alright, thanks |
| 21:34.25 | WIMPy | That's almost certain to fail. |
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| 21:54.13 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-sltysgsqxopgynld) |
| 21:55.08 | KNERD | On one of the voice mail boxes I am getting an response -- "i'm sorry I did not understand your response" when pressing "3" for Advanced Options. NO error is appearing in the console and DTMF shows " DTMF[15083]: channel.c:3998 __ast_read: DTMF end '3' received on " any ideas? |
| 22:14.35 | WIMPy | Hmm. Is it possible to associate contacts to accounts ("lines") on the Digium phones? |
| 22:20.08 | *** join/#asterisk calum_ (~calum_@cpc70817-harg4-2-0-cust754.7-1.cable.virginm.net) |
| 22:30.06 | WIMPy | Looks like I'm not the only one who doesn't find that option. |
| 22:44.31 | ChannelZ-Wk | Call from '' (85.25.74.32:5354) to extension '!#485)(9e370?476587' rejected because extension not found in context 'incoming'. |
| 22:44.44 | ChannelZ-Wk | Really? That's going to be some magic extension someplace? |
| 22:45.07 | ChannelZ-Wk | or '!afow+342FASF^&' ? Stupid-ass script kiddies can't control their shit |
| 22:45.08 | *** join/#asterisk Akuma (~Akuma@apn-31-1-20-182.dynamic.gprs.plus.pl) |
| 22:46.06 | WIMPy | nice |
| 22:54.58 | rrittgarn | yeah channelZ... i block 85.0.0.0/8 but i feel your pain... they are definitely persistent |
| 22:56.01 | WIMPy | Evil swiss hackers? |
| 22:58.58 | Valduare | the ones you gotta watch out for are the russian hackers named Yuri - they are often found in stone wall leaky basements with single hanging lightbulb in the room with a super computer built out of random parts with coords connecting them stringing everywhere, and somehow the guy is built like a tank with arm tatoos (not sure where he finds the time to workout cause heâs spends all his time hacking YOUR* server) |
| 22:59.56 | WIMPy | wonders how Valduare know so much abot Yuri... |
| 23:00.28 | Valduare | he does freelancing work building Etsy shops for knitters :P |
| 23:00.34 | Valduare | my mom hired him |
| 23:00.48 | Valduare | pft :P lol |
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| 23:46.34 | Penguin | channelz-wk: Sure it is! Don't you dial things like !#)( regularly from your phone? I thought that was the new trend for vanity phone numbers. |
| 23:48.10 | Penguin | <PROTECTED> |
| 23:48.18 | Penguin | Seems like I'm blocking it, too. |
| 23:53.24 | ChannelZ-Wk | Germans. |
| 23:55.43 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 23:56.28 | snadge | my employer just received 3cx certification.. just throwing that out there :p |
| 23:59.08 | ChannelZ-Wk | which is what? |