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| 00:13.16 | Apteryx | anything like the CLI "set sip debug on" for pjsip? |
| 00:15.37 | tm1000 | Apteryx: pjsip set logger on |
| 00:15.45 | tm1000 | Apteryx: pjsip set logger <host> |
| 00:15.53 | tm1000 | Apteryx: pjsip set logger off |
| 00:16.28 | tm1000 | Apteryx: also Asterisk supports âtabbingâ so. pjsip <tab> === list of commands :-) |
| 00:17.20 | *** part/#asterisk bkruse (~Adium@64.89.97.113) |
| 00:17.29 | Apteryx | tm1000: that's just what I was looking for! Thanks :) I thought setting debug=yes under [globals] in pjsip.conf was enabling debug mode. |
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| 01:49.03 | Apteryx | Anyone knowledgeable in SDP codecs handling would mind to help me understand why Jitsi and Bria won't talk together using the Opus codec? I've put a paste of the pjsip log here: http://pastebin.com/uNBzzuXB |
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| 01:54.34 | Apteryx | Jitsi uses: opus/48000/2, whilst Bria uses OPUS/48000, if I'm following the sequence correctly. |
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| 01:56.47 | Apteryx | Would the agents/Asterisk be dumb enough to do a case sensitive compare between those fields? |
| 02:03.56 | Apteryx | It seems PJSIP is case insensitive regarding codec ID. See http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__CODEC.htm for more info. |
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| 02:22.42 | Apteryx | I'm pretty sure the problem is Bria not using the latest Opus RTP Format defined in the latest IETF draft. |
| 02:31.44 | *** part/#asterisk vinhdizzo (~vinh@cpe-76-173-168-125.socal.res.rr.com) |
| 02:55.10 | Apteryx | ok, so this was the problem: Bria on iOS (it works on Mac) refuses the opus/48000/2 offer. They are working on a fix for the next releases. |
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| 03:11.21 | Apteryx | How can I force some codec on incoming calls? |
| 03:12.50 | Apteryx | I enabled G722, and now Asterisk bridges G722 to G711 instead of using G711 on the receiver side's as well. I'd like to tell it to use G711 on all incoming call (but still allowing G722 for my internal calls). |
| 03:14.39 | Apteryx | I tried set(SIP_CODEC=ulaw) |
| 03:14.57 | Apteryx | didn't work, the receiving UA is still on G722. |
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| 04:08.35 | Apteryx | Using pjsip and tcp, I get: TCP connect() error: Connection refused. |
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| 09:10.02 | WHiZZi | g'day all |
| 09:11.59 | WHiZZi | I was wondering if anyone can confirm or deny if there's a setting for an Asterisk call file to be retried after a number of minutes. |
| 09:12.18 | WHiZZi | I create a working file, move it to the outgoing spool directory |
| 09:12.27 | WHiZZi | call is being made, so far so good |
| 09:12.45 | WHiZZi | I answer the call and everything is setup as it should |
| 09:12.58 | WHiZZi | but the .call-file remains in the outgoing spool directory |
| 09:13.01 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 09:13.27 | WHiZZi | and after approx 5 minutes asterisk does a new dial based on that .call-file |
| 09:13.54 | WHiZZi | while I'm actually having a fully functional call already |
| 09:14.34 | WHiZZi | as soon as I hangup, the .call-file is removed |
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| 09:29.05 | hindi | hi |
| 09:29.11 | robscow | Is it possible to cancel a transfer? We can transfer a call, but if the other party doesn't pick up, and it goes to voicemail, we'd like to cancel the transfer? |
| 09:29.43 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 09:29.48 | hindi | Ive configured a session timer for my trunk. But It seems not working.... I want asterisk to re-register the trunk every 800 seconds |
| 09:30.00 | robscow | currently we have [featuremap] atxfer => *2 in features.conf |
| 09:32.18 | mirela666 | robscow: use attended transfer, so that agent can call 3rd party and see if they will pick up |
| 09:33.17 | mirela666 | hindi: yu must be having session-timer in general section that is re-writing it ? |
| 09:33.49 | hindi | nope, I don't have any other session-timers |
| 09:34.30 | hindi | in sip trace i can see the register packets from asterisk to the provider... but asterisk doesnt send the register every 800 seconds... |
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| 09:37.19 | hindi | mirela666, peer settings overwirte the global settings... |
| 09:37.58 | mirela666 | hindi: and in what time does it send ? |
| 09:38.47 | hindi | between 8 and 10 minutes.. |
| 09:39.12 | hindi | but I think it takes the providers expiretime... |
| 09:39.36 | hindi | because sip show registry shows: Refresh = 585 |
| 09:40.40 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 09:44.05 | hindi | mirela666, yep it takes the expire in the contact field |
| 09:44.14 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 09:45.17 | mirela666 | hindi: and what does sip show registry says for expires in field |
| 09:45.54 | mirela666 | Refresh* |
| 09:46.37 | hindi | 585 |
| 09:46.59 | mirela666 | so it is set to 600 sec |
| 09:47.26 | mirela666 | 10 min |
| 09:48.01 | hindi | yep |
| 09:48.22 | mirela666 | maybe you can't overwrite the general section per peer |
| 09:51.15 | mirela666 | so you have defaultexpiry=600 in general section |
| 09:51.26 | mirela666 | and what do you use in peer settings |
| 09:54.09 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 09:54.26 | hindi | nope... defaultexpiry is set to 120 |
| 09:55.12 | hindi | tried to force asterisk to use the configured session-timer with session-refresher=uac.... but still uses the providers expire... |
| 09:55.34 | hindi | just called the provider.. they check the expire in their response |
| 09:58.18 | mirela666 | http://paste.ubuntu.com/8478204/ |
| 09:58.48 | mirela666 | Asterisk will never override the ; preferences of the other endpoint, to avoid fiting |
| 09:58.54 | mirela666 | fighting* |
| 10:00.59 | hindi | yep... so it takes the providers expire :) |
| 10:01.37 | hindi | mirela666, so its not my fault :D |
| 10:02.34 | mirela666 | well, looks like you can do nothing about it |
| 10:02.54 | MintberryCruNCH | anyone worked with app mp4? |
| 10:14.37 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 10:39.51 | hindi | mirela666, just tried to refuse the session timer... but its not possible... |
| 10:40.18 | mirela666 | :/ |
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| 10:55.21 | hindi | mirela666, ok.. on step closer :) set it to originate and refresher = uas |
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| 10:55.39 | hindi | now the session timer is mine :) |
| 10:56.07 | hindi | stupid computers... |
| 10:56.13 | mirela666 | hehe :) good |
| 10:56.48 | hindi | why i didnt take another job... something with stones... or with cars .. |
| 10:56.54 | hindi | ;) |
| 10:57.04 | mirela666 | yeah, driving big truks |
| 10:57.36 | mirela666 | or Executive cutter of watermellons |
| 10:57.42 | hindi | but I though... do something with computers... computers are the future.. |
| 10:57.48 | mirela666 | :D |
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| 10:58.14 | hindi | now i working on UC-Applications for a few years and eyery week is another stupid case... |
| 10:59.33 | mirela666 | so originate + uas was the winning combo ? |
| 10:59.38 | mirela666 | or uac? |
| 10:59.38 | hindi | mirela666, this is the real executive cutter of watermelones |
| 10:59.39 | hindi | http://www.youtube.com/watch?v=Lq14KIgdJOA |
| 11:00.55 | mirela666 | :D |
| 11:01.00 | hindi | mirela666, pffffff..... session timer changed to providers one... |
| 11:01.21 | hindi | I think, asterisk never win the challenge... |
| 11:01.45 | hindi | because its the last stupid asterisk box in the whole data center... |
| 11:01.58 | mirela666 | :) the boss said "seddle the horse" |
| 11:02.01 | hindi | and so it cares a shit about our feelings... |
| 11:02.34 | hindi | because all his friends are down and replaced by freeswitch servers |
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| 11:33.00 | hindi | why there is way to decrease the session timer... I only want to set a session timer lower than the timer which is sent by provider :( |
| 11:33.15 | hindi | *no way... |
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| 12:18.29 | ovoshlook | hello. How to configure menuselect from command line (something like menuselect --enable) |
| 12:18.32 | ovoshlook | ? |
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| 12:30.07 | Aamit | ovoshlook, make menuselect |
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| 12:34.30 | ovoshlook | I mean - make menuselect from scripr oranothes app. I want to automate asterisk installer |
| 12:34.54 | ovoshlook | like apt-get |
| 12:35.09 | ovoshlook | but install it with my own options and packets |
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| 12:41.49 | wdoekes | ovoshlook: $ menuselect/menuselect --disable app_dial; grep app_dial menuselect.makeopts |
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| 12:43.51 | Apteryx | Good morning! |
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| 12:55.19 | Apteryx | Would anyone have an idea why I keep getting Connection refused [code=120111] when using Asterisk 12, pjsip and tcp transport? My config works fine using udp... My log: http://pastebin.com/keCQbTB3 |
| 12:58.49 | ovoshlook | wdoekes: thanks! |
| 13:00.46 | nkkromhof | Apteryx: at the risk of asking the obvious, you have enabled 'transport=tcp' in sip.conf, right? |
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| 13:01.20 | wdoekes | nkkromhof: PJsip |
| 13:01.36 | nkkromhof | oh I see, sorry, my bad! |
| 13:02.07 | Apteryx | nkkromhof: in pjsip.conf, yes. |
| 13:04.08 | Apteryx | When I do, from the CLI, pjsip show endpoints, I get: http://pastebin.com/tqTHRQrV |
| 13:04.44 | Apteryx | One thing that it could be, is a module I forgot to load, as I'm manually managing modules. But usually Asterisk has good messages about a missing module. |
| 13:04.54 | wdoekes | Apteryx: ? |
| 13:04.58 | wdoekes | your tcp works just fine |
| 13:05.05 | wdoekes | it's the endpoint that has issues on that port |
| 13:05.27 | wdoekes | go so a shell and type: nc -zvw2 192.168.0.100 60057 |
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| 13:05.57 | wdoekes | *go to |
| 13:06.54 | Apteryx | On the first TCP connection, I got: nc: connect to 192.168.0.100 port 60057 (tcp) timed out: Operation now in progress |
| 13:07.08 | Apteryx | (for the yuki endpoint.) |
| 13:07.29 | Apteryx | On the second TCP address registered, I got: nc: connect to 192.168.0.100 port 60263 (tcp) failed: Connection refused |
| 13:08.03 | Apteryx | hmmm. Seems you've nailed the issue. Strange that it happens only when using TCP and not UDP? |
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| 13:09.51 | wdoekes | are you trying to register tcp endpoints from behind nat? |
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| 13:10.09 | wdoekes | (two different endpoints behind 192.168.0.100 ?) |
| 13:11.01 | Apteryx | wdoekes: the only thing which is the otherside of the nat in this scenario is Asterisk registration address (sip.apteryx.ca), that resolves back to my router and then port forward to the asterisk server at 192.168.0.10. |
| 13:11.37 | Apteryx | I'm trying to use such a scheme because I want to connect to my server from remote locations as well and I don't want to change my UA settings every time. |
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| 13:13.38 | wdoekes | you didn't answer my question: is the 192.168.0.100 the IP of a device, or is it an IP of a nat-gateway? |
| 13:14.04 | Apteryx | wdoekes: sorry. Yes, 192.168.0.100 is the IP of a device. |
| 13:15.51 | wdoekes | well.. apparently the phone/device registers with a port it doesn't open. but we need to see a sip capture of the REGISTER to be certain |
| 13:17.16 | WIMPy | Isn't the idea of using tcp that you have a connection? |
| 13:21.04 | Apteryx | wdoekes: here's my registration process, for the endpoint at 192.168.0.101: http://pastebin.com/xxE9j4sa |
| 13:22.15 | wdoekes | and if you then 'nc' to 192.168.0.101 50871, does it open the port? |
| 13:23.05 | Apteryx | nope, still getting: nc: connect to 192.168.0.101 port 50871 (tcp) failed: Connection refused |
| 13:23.13 | wdoekes | hm.. "In a Contact or Route header field value, it indicates that the UA would like other requests in the same dialog to be routed over the same flow. |
| 13:23.16 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 13:23.40 | wdoekes | as long as the connection is open, asterisk should attempt to use that open tcp connection to contact said device |
| 13:23.52 | wdoekes | don't know if that requires extra modules |
| 13:24.03 | wdoekes | check: netstat -apnAinet | grep 50871 |
| 13:24.08 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:24.20 | wdoekes | it should be ESTABLISHED if it's still open |
| 13:24.37 | Apteryx | got: tcp 0 0 192.168.0.10:5060 192.168.0.1:50871 ESTABLISHED 12121/asterisk |
| 13:24.58 | Apteryx | But notice the 192.168.0.1 |
| 13:25.02 | Apteryx | this is due to sip.apteyrx.ca |
| 13:25.17 | wdoekes | ok, so that would never work |
| 13:25.37 | Apteryx | :( |
| 13:25.37 | wdoekes | since the .1 would have to forward all ports 1-1 to .101 |
| 13:26.04 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 13:26.05 | wdoekes | unless you can convince asterisk to actually use the "ob" flag and talk over the existing tcp connection |
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| 13:28.28 | Apteryx | What is this flag about? |
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| 13:39.23 | wdoekes | Apteryx: this? https://issues.asterisk.org/jira/browse/ASTERISK-22658 |
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| 14:17.30 | *** join/#asterisk upp (~ialami@146.140.16.62) |
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| 14:18.49 | upp | hello, can i tell asterisk to ignore the first 3 digit if a call is comming from a known peer? |
| 14:20.19 | [TK]D-Fender | You can't. You CAN however make different extens in the context it lands in to strip them and then pass onwards. |
| 14:21.15 | upp | ok thanks [TK]D-Fender |
| 14:30.43 | WIMPy | The first digits of what? |
| 14:34.15 | *** join/#asterisk e4voip (uid13742@gateway/web/irccloud.com/x-qtbqtdxeyazutnvv) |
| 14:46.17 | upp | WIMPy: peer1 call 555222 and asterisk sip user is 222 for example |
| 14:46.52 | upp | WIMPy: trunk work well, i see at the log 555222 rejected because extension not found in context |
| 14:48.23 | WIMPy | Yes, you need to make a special context for them then. |
| 14:48.42 | [TK]D-Fender | upp: "asterisk sip user" means nthing |
| 14:49.07 | [TK]D-Fender | upp: phone's don't dial "sip users". They dial EXTENSIONS> What your extensions fo is another matter |
| 14:51.04 | upp | yes i understand what you mean, i tought that i can do that at the trunk declaration |
| 14:51.13 | upp | at sip.conf |
| 14:51.22 | [TK]D-Fender | no |
| 14:51.29 | [TK]D-Fender | what you dial = dialplan |
| 14:51.36 | [TK]D-Fender | this is the very basics of * |
| 14:52.02 | [TK]D-Fender | nothign you put in sip.conf tells it what do do when user dials X. |
| 14:52.35 | WIMPy | It would make sense, however. |
| 14:53.17 | upp | there are some other software like VCX when you can tell it when calls come from this peer, do that.... |
| 14:54.49 | [TK]D-Fender | upp: you can.. it's called DIALPLAN |
| 14:54.55 | *** join/#asterisk calum_ (~calum_@host86-176-63-147.range86-176.btcentralplus.com) |
| 14:55.02 | [TK]D-Fender | upp: Everything the dial = dialplan. |
| 14:55.07 | [TK]D-Fender | upp: Go make some for this peer |
| 14:55.27 | upp | ok thanks |
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| 17:07.35 | *** join/#asterisk ikevin (~kevin@lisa.illux.org) |
| 17:07.41 | ikevin | hello |
| 17:09.05 | ikevin | i'm trying to setup tls on asterisk 1.8, i'm following some howto about that, so i've a problem, all howto refer to tcp/tls port 5061 which is not open on my server |
| 17:09.44 | ikevin | i've set a "port=5061" in my extension, do i need to add other options? |
| 17:09.56 | *** join/#asterisk Dovid (~Dovid@ool-2f113961.dyn.optonline.net) |
| 17:10.30 | WIMPy | 1. You may want to use a more recent version of Asterisk. |
| 17:10.52 | WIMPy | 2. If the port isn't open, you either failed to activate it or used another port. |
| 17:11.07 | [TK]D-Fender | ikevin: that port will not be open on your server |
| 17:11.14 | WIMPy | 3. "port" and "extension" don't make sense. |
| 17:11.22 | [TK]D-Fender | ikevin: that is the port on the OTHER END you defining |
| 17:11.33 | [TK]D-Fender | ikevin: that is not *'s LISTENING port |
| 17:12.11 | WIMPy | What? |
| 17:13.14 | WIMPy | Sure that's that's the port it would listen on if it's enabled. |
| 17:13.34 | [TK]D-Fender | [13:09]ikevini've set a "port=5061" in my extension, do i need to add other options? <- he's talking about PEER settings |
| 17:13.44 | [TK]D-Fender | not *'s actual [general] settings |
| 17:13.51 | ikevin | to enable tls i've added "tlsenable=yes" on [general] |
| 17:14.07 | ikevin | [TK]D-Fender, oh sorry, yes, in peer not extension :x |
| 17:14.18 | WIMPy | Can you use TLS outbound without having it enabled inbound? |
| 17:14.43 | [TK]D-Fender | WIMPy: that peer port says nthing about setting the bound port it will use |
| 17:15.06 | [TK]D-Fender | WIMPy: Also... we've seen no statement of anything at all working.. so.... |
| 17:15.14 | WIMPy | And it doesn;t say anything about using TLS, either. |
| 17:15.37 | ikevin | i'm following: https://wiki.asterisk.org/wiki/display/AST/SIP+TLS+Transport |
| 17:16.14 | ikevin | so while i try to use this (just changing IP), this don't work, i get a connexion refused error |
| 17:16.32 | ikevin | (i use debian's package) |
| 17:16.45 | [TK]D-Fender | ikevin: So far i'm not seeing you defining the actual bound port * will use |
| 17:17.29 | WIMPy | What says connection refused? |
| 17:18.16 | ikevin | [TK]D-Fender, i'm not sure to understand what you said (sorry for my bad english :x) |
| 17:19.08 | [TK]D-Fender | ikevin: You haven't shown us where you actually told Asterisk to listen to that port. |
| 17:19.26 | [TK]D-Fender | ikevin: And we have no idea how many other mistakes you may have made in your configs. |
| 17:19.33 | [TK]D-Fender | ikevin: You should re-read the sample config |
| 17:21.56 | ikevin | i don't specify any specific port, just use the default debian config and apply https://wiki.asterisk.org/wiki/display/AST/SIP+TLS+Transport |
| 17:22.19 | [TK]D-Fender | ikevin: Default means nothing. |
| 17:22.26 | [TK]D-Fender | ikeGo make actual settings and confirm them |
| 17:23.34 | ikevin | default mean i've installed package from debian and just changing sip.conf with what i found on the howto |
| 17:24.27 | [TK]D-Fender | ikevin: You're not showing what you actually did, status dumps, etc. |
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| 17:27.40 | ikevin | what i did, installing 2 fresh debian 7 on VMs, installing asterisk from apt, and the only thing i've change in asterisk config is to apply the examples config for tls from asterisk wiki |
| 17:27.41 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 17:31.53 | WIMPy | If you made those changes, restarted Asterisk and it doesn't listen on 5061, then you either made a mistake, or you don't have TLS support. |
| 17:33.17 | [TK]D-Fender | ikevin: Nothing you have done has set the listening port so far as we'vce seen |
| 17:33.36 | *** join/#asterisk riess82 (~riessma@188-22-235-251.adsl.highway.telekom.at) |
| 17:33.36 | [TK]D-Fender | ikevin: And what we have seen ... was 1 line you posted here. |
| 17:34.17 | WIMPy | What's on the wiki, would set it to listen. |
| 17:35.29 | *** part/#asterisk riess82 (~riessma@188-22-235-251.adsl.highway.telekom.at) |
| 17:35.52 | ikevin | http://pastebin.com/upknZwXg this are my 2 sip.conf |
| 17:37.39 | [TK]D-Fender | ikevin: http://svnview.digium.com/svn/asterisk/branches/1.8/configs/sip.conf.sample?revision=423066&view=markup <---- read the sample config |
| 17:39.19 | [TK]D-Fender | ikevin: You have not set the bind for it |
| 17:42.39 | ikevin | i think it's that, so i don't found info about "tls bind port" |
| 17:43.10 | WIMPy | Yes you did. You even told us where you found it. |
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| 17:55.33 | *** join/#asterisk eduzimrs (~adada@mail.aytycrm.com.br) |
| 17:56.44 | eduzimrs | guys, im using * 1.8 and im trying to schedule a call in the future doing (touch -t YYYYMMMDDhhmm.ss somefile.txt) and moving it to outgoing, so its dialing right after i move it, its not respecting the date a changed |
| 17:56.47 | eduzimrs | ideas? |
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| 18:09.34 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 18:18.04 | [TK]D-Fender | eduzimrs: Same as always : Show us |
| 18:18.38 | ChannelZ | how are you movingh it? |
| 18:20.03 | ChannelZ | ('stat' the file and see if the timestamp is what you think it is) |
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| 19:00.42 | KNERD | What happened to the Voicemail option - > 0 Mailbox options -> 4 Record your temporary message -> 2 Erase your temporary message (going back to the standard message) ? |
| 19:01.25 | *** join/#asterisk mirela666 (~mirko.bra@95.180.126.160) |
| 19:01.28 | WIMPy | Does anyone have an idea why Asterisk 11.13.0 sometimes decides to try to play a non existing file instead of an existing one? |
| 19:02.21 | WIMPy | It seems to only happen for the queue-minutes files and only sometimes. |
| 19:03.09 | WIMPy | Here's a verbose/debug of such an occurance: http://wimpy.yeti.dk/pastebin |
| 19:03.55 | WIMPy | The files all exist as gsm and only as gsm. The failed one is then replaced by the english version. |
| 19:05.10 | KNERD | you need o fix that "pastebin" of yours. Nobody wants to try to read all of that if it is all on one line |
| 19:05.41 | WIMPy | Well, it's a unix text, not a DOS one. |
| 19:07.12 | KNERD | should be in displayed in compliance with w3c |
| 19:07.33 | WIMPy | It's a text file, not html. |
| 19:08.06 | KNERD | exaclty :-) |
| 19:08.31 | WIMPy | Do you want it as html? |
| 19:09.52 | WIMPy | Now you can get it as html. |
| 19:17.21 | KNERD | sure :-) |
| 19:19.11 | KNERD | well...it wants the version according to what CODEC is being used |
| 19:19.58 | WIMPy | Which is alaw all the way. |
| 19:20.59 | KNERD | so it is trying to place a alaw version of that message which does not exists so it plays what it has avaiable |
| 19:21.27 | WIMPy | Or something that isn't. |
| 19:22.02 | KNERD | so if you have the message in alaw it woul dbe playing that one |
| 19:22.13 | WIMPy | Maybe. |
| 19:22.21 | KNERD | or you make the system only use GSM CODEC then it will play that GMS one |
| 19:22.46 | WIMPy | The PSTN is alway only. |
| 19:23.23 | KNERD | not inside the PBX |
| 19:24.29 | WIMPy | Well, as I said, I only have gsm file. |
| 19:24.36 | WIMPy | Which Asterisk uses most of the time. |
| 19:24.51 | KNERD | try converting it |
| 19:25.27 | WIMPy | And how do I know if that would work? |
| 19:25.39 | WIMPy | That seems to be a very strange and random bug. |
| 19:29.55 | *** join/#asterisk jokke (~jokke@2a03:4000:2:4f5::1) |
| 19:29.59 | jokke | hi |
| 19:30.10 | jokke | i'm having some trouble with asterisk |
| 19:30.30 | jokke | when someone is calling me i see in the console: failed to extend from 256 to 373 |
| 19:30.43 | jokke | my number isn't 256 though |
| 19:30.50 | jokke | oer 373 |
| 19:30.53 | jokke | *or |
| 19:31.07 | jokke | where should i start debugging? |
| 19:35.00 | WIMPy | First of all the real message would help. |
| 19:35.17 | WIMPy | And maybe it's just someone trying to hack you at the same time. |
| 19:36.04 | jokke | WIMPy: at the exact same time? |
| 19:36.12 | jokke | that's the real message |
| 19:38.09 | WIMPy | he part at the beginning might tell you what's happening. |
| 19:38.14 | WIMPy | The... |
| 19:38.26 | WIMPy | The part you didn't paste. |
| 19:38.40 | jokke | just a sec |
| 19:40.11 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 19:43.11 | jokke | WIMPy: ok.. https://p.jreinert.com/1qX68/ |
| 19:44.07 | jokke | the push thing is from my extensions |
| 19:44.11 | jokke | i'll paste them |
| 19:46.15 | WIMPy | That does not look like Asterisk log at all. |
| 19:46.43 | jokke | here's the relevant extends https://p.jreinert.com/2CP/ |
| 19:46.51 | WIMPy | But something seems to have run out of some resource, probably memory. |
| 19:47.06 | jokke | naaah i don't think so |
| 19:47.25 | jokke | Mem: 3.9G 1.9G 2.0G 17M 53M 438M |
| 19:47.34 | WIMPy | Ok, so where is the verbose of at least 3? |
| 19:47.45 | jokke | 3? |
| 19:47.51 | WIMPy | And maybe you should set debug to at least 1 as well. |
| 19:48.01 | jokke | ah ok |
| 19:48.04 | jokke | how do i do that? |
| 19:48.29 | WIMPy | core set ... verbose|debug. |
| 19:48.52 | jokke | ah |
| 19:48.54 | jokke | ok |
| 19:49.19 | jokke | ok done |
| 19:49.43 | jokke | i can't call myself though because my softphone on linux can't connect to the server |
| 19:49.48 | jokke | no idea why |
| 19:49.55 | jokke | the mobile client works fine |
| 19:50.14 | jokke | and i've tried several softphones |
| 19:50.46 | jokke | can you try to call me from another sip server? Don't know if thats possible... |
| 19:53.03 | Katty | file: 5 DAYS |
| 19:55.41 | *** join/#asterisk mcrownover (~mcrownove@remote.gawest.com) |
| 19:58.55 | WIMPy | SIP is P2P. |
| 19:59.16 | WIMPy | So if only depends if you allow it. |
| 19:59.28 | jokke | ah |
| 19:59.32 | jokke | ok |
| 19:59.34 | *** join/#asterisk Cynagen (~cynagen@ip72-208-60-104.ph.ph.cox.net) |
| 19:59.41 | jokke | i don't know.. what are the default settings? |
| 19:59.55 | WIMPy | Depends on the version. |
| 20:00.42 | jokke | 12.4.0 |
| 20:01.58 | WIMPy | no |
| 20:02.24 | jokke | no? |
| 20:02.27 | jokke | what no? |
| 20:02.31 | WIMPy | Default=no |
| 20:02.36 | jokke | ah |
| 20:02.39 | jokke | ok |
| 20:03.20 | jokke | what setting is this? |
| 20:03.34 | WIMPy | allowguests |
| 20:04.31 | jokke | um.. does this mean just incoming calls or also allow guests to make outgoing calls? |
| 20:05.59 | WIMPy | Both |
| 20:06.13 | jokke | ah that's bad |
| 20:06.18 | WIMPy | Your extensions define what you're allowed to call. |
| 20:06.26 | jokke | hm sure |
| 20:07.01 | jokke | i'll activate it temporarily if you want to help me out and try to call |
| 20:07.40 | WIMPy | I don't have a softphone here. |
| 20:07.46 | jokke | ah :D |
| 20:07.50 | jokke | ok then |
| 20:10.40 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 20:17.55 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 20:29.02 | jokke | ok here: |
| 20:29.12 | jokke | WIMPy: i got a softphone working |
| 20:29.21 | jokke | here's the verbose output |
| 20:29.25 | jokke | https://p.jreinert.com/WA7/ |
| 20:29.51 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 20:36.08 | WIMPy | So it wasn't able to communicate with the peer jokke. |
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| 21:18.07 | *** join/#asterisk SirLouen (~pickupdog@188.84.87.20) |
| 21:18.16 | SirLouen | hi all |
| 21:19.04 | SirLouen | anyone know if its possible to eliminate ringing tone in early media during a noanswer-playback for example? |
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| 21:43.44 | *** join/#asterisk Neo` (~neo@unaffiliated/neo/x-8965769) |
| 21:43.46 | Neo` | hi all |
| 21:44.08 | Neo` | guys, mb anybody can help me with linksys pap2t |
| 21:44.20 | Neo` | i have 2 ata, cisco spa 122, and linksys pap2t |
| 21:44.42 | Neo` | when anybody call me to spa122 - i get long rings |
| 21:44.59 | Neo` | when anybody call me to pap2t - i get many many short rings :( |
| 21:45.12 | Penguin | Change the ringer on the device. |
| 21:45.17 | Neo` | how i can configure pap2t to long rings too :( |
| 21:45.35 | Neo` | i can't find ringer settings on the device :(((( |
| 21:45.58 | Neo` | (sorry for my english, i`m from Russia) |
| 21:46.05 | Penguin | I always do everything on admin/advanced, so click on admin and then click on advanced. |
| 21:46.43 | Neo` | i try change mode to admin/advanced, but i can't find this options |
| 21:46.57 | Neo` | mb you can say me name of this options? |
| 21:47.23 | Penguin | Did you log on from your browser? |
| 21:48.19 | Neo` | at now moment - yes |
| 21:51.36 | Neo` | can you help me with it? please :( |
| 21:52.14 | Penguin | One second. |
| 21:52.18 | Neo` | ok |
| 21:53.33 | Penguin | I think it might be on the line 1 tab. |
| 21:54.19 | Neo` | but... where? what name is option? :( |
| 21:55.53 | Penguin | ringer |
| 21:57.48 | Neo` | This tab does not have this option |
| 22:01.24 | Penguin | Give me a minute to log in on a pap2t and look for you. |
| 22:04.09 | Neo` | ofc |
| 22:04.38 | Penguin | Go to the User 1 tab for port 1. |
| 22:05.27 | Penguin | There is a Ring Settings section. |
| 22:05.59 | Penguin | Default ring option |
| 22:06.12 | Penguin | 1 should be long ring |
| 22:09.15 | Neo` | wow! |
| 22:09.20 | Neo` | one sec, i try it! |
| 22:10.50 | Neo` | hmmmm |
| 22:10.59 | *** join/#asterisk calum_ (~calum_@cpc67428-harg5-2-0-cust142.7-1.cable.virginm.net) |
| 22:11.14 | Neo` | on default selected 1, but i get: long, short, short, short |
| 22:12.38 | Penguin | Try a different one on the list. |
| 22:13.21 | Neo` | ofc, i try all from this list ) |
| 22:13.37 | Penguin | You can also change what the ringers sound like in the regional tab. |
| 22:15.33 | Penguin | My Ring1 Cadence (in north america) is: 60(2/4) |
| 22:15.42 | Penguin | one 2-second ring |
| 22:18.42 | Penguin | Oh, that might be the one you hear on the earpiece when dialing out. :( |
| 22:18.59 | Penguin | I don't have a pap2t here to play with. |
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| 22:23.29 | Neo` | - |
| 22:23.29 | Neo` | <Penguin> My Ring1 Cadence (in north america) is: 60(2/4) |
| 22:23.29 | Neo` | <Penguin> one 2-second ring |
| 22:23.30 | Neo` | - |
| 22:23.31 | Neo` | !!! |
| 22:23.37 | Neo` | it is very important! |
| 22:24.01 | Neo` | cadence settings is idenidentical from cisco spa122 and linksys |
| 22:24.13 | Neo` | it settings is: 60(2/4) |
| 22:24.27 | Neo` | but on cisco it is "one 2-second ring" |
| 22:24.52 | Neo` | on linksys it is: one 1-second ring, and 2 short rings |
| 22:28.58 | *** join/#asterisk vinhdizzo (~vinh@cpe-76-173-168-125.socal.res.rr.com) |
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| 22:39.10 | Penguin | (1718.40) <Penguin> Oh, that might be the one you hear on the earpiece when dialing out. :( |
| 22:39.14 | Penguin | ^ |
| 22:47.10 | Neo` | in pap2t has a feature call longer 0.75señ he divides into parts |
| 22:47.44 | Neo` | if i set ring 2 sec, he ring 0.75 sec, and last short rings |
| 22:47.48 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 22:47.56 | Neo` | i dont know how i can fix it |
| 22:48.59 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 22:59.11 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 23:01.53 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 23:13.42 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 23:26.20 | Neo` | Penguin, very very very thx you for help! |
| 23:26.25 | Neo` | i fix this bug ) |
| 23:29.17 | *** join/#asterisk blee (~blee@72.189.166.247) |
| 23:36.03 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:44.06 | *** join/#asterisk fling (~fling@fsf/member/fling) |