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| 00:21.33 | ChannelZ | Confbridge isn't that hard, just look through the sample config. |
| 00:25.02 | ChannelZ | I think you can basically just use the sample config unchanged, and then do a Confbridge(1) from your dialplan |
| 00:25.48 | Apteryx | Hello! I'm trying to establish a sip communication between a VPNed device (connected to my job network) and my home network as a test. The VPNed device received address 172.20.2.73 from my company's network, but when I visit 'whatsmyip.org' it says 64.235.209.34 |
| 00:26.19 | Apteryx | Asterisk sends back RTP to address 172.20.2.73, and without surprise I cannot hear audio on the VPNed device side. |
| 00:27.11 | Apteryx | Is there a way to tell Asterisk to use the last 'hop' address in the chain? It probably came from 64.235.209.34 even if it originated from 172.20.2.73. |
| 00:27.39 | ChannelZ | dejavu |
| 00:27.57 | ChannelZ | In sip.conf for the peer, set nat=auto_force_rport,comedia |
| 00:29.35 | ChannelZ | which aught make asterisk send RTP back to the ip/port it receives media from |
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| 00:30.37 | Apteryx | ChannelZ: I'm using Asterisk 12 with the new chan_pjsip, so I don't think there is a nat option in there. I've set local_net, external_media_address, external_signaling_address in my transport definition and direct_media=no in my endpoints definitions. |
| 00:31.44 | Apteryx | I think I saw something about force_rport, but I also think I saw it was on by default. Not sure about comedia, will look the configs help. |
| 00:38.11 | Apteryx | I confirm that force_rport is active by default. |
| 00:38.49 | Apteryx | The only other options I see could make a differences are "ice_support" and "rtp_symmetric". |
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| 00:50.32 | ChannelZ | sorry wandered off to eat. |
| 00:50.41 | ChannelZ | There is force_rport and rtp_symmetric in pjsip |
| 00:51.46 | ChannelZ | which should be roughly equiv from chan_sip, but I've not actually done any NAT with pjsip yet personally |
| 00:52.12 | ChannelZ | I'm also confused about your actual VPN setup |
| 00:54.36 | Apteryx | Sorry I was eating myself ;). Why is that? |
| 00:55.42 | ChannelZ | why is what |
| 00:56.36 | Apteryx | What is confusing you about my VPN setup? |
| 01:03.50 | ChannelZ | That it's a VPN at all. Where does the tunnel go from and to, and where does asterisk live? |
| 01:10.09 | Apteryx | The tunnel goes from my local home network to remote network (my company's network). Asterisk lives on my local home network. |
| 01:10.54 | Apteryx | I'm trying to establish communication between a device connected from my company's network and another device connected on my home local network. |
| 01:11.51 | Apteryx | This might not seem constructive but I'm using this setup to test if my Asterisk box is configured well for dealing with double NATs. |
| 01:21.08 | Apteryx | My VPNed contact is registering with Contact: maxim/sip:maxim@172.20.2.73:5060;transport=udp... hmm. |
| 01:21.21 | Apteryx | *registering as |
| 01:33.40 | ChannelZ | Isn't the VPN tunnelling that network to yours? IE some subnet or another of 172.20 |
| 01:37.06 | Apteryx | hmm. I guess so! I thougth being on the VPN it would act as if I was physically connected at my job's location, but I neglected the fact that VPN technology is bridging both networks... Hmm. |
| 01:41.57 | Apteryx | Here's the exact contact details (for the VPNed sip agent): http://pastebin.com/yPppjGe9 |
| 01:43.37 | Apteryx | 172.20.2.73 is the IP of the VPN interface. The WLAN interface is connected to my LAN with IP 192.168.0.101. |
| 01:56.57 | Apteryx | Ok, so it's not using my company's network as I thought. It's just passing from one interface (VPN) to another (WLAN), keeping the source address of the VPN interface. |
| 01:57.17 | Apteryx | So Asterisk was probably at least as confused as I was ;) |
| 01:58.51 | Apteryx | I'll install some 3G SIM in one of my device and retest. Hopefully this scenario will be more straigtforward (I'm thinking single NAT scenario). |
| 01:59.36 | Apteryx | 3G <--> NATed Asterisk |
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| 07:10.39 | KevinRoberts | Hello, I have a voicemail question I looked at the book, but it doesn't how to add a mailbox to a user or create mailbox users would it be in sip.conf? |
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| 07:14.21 | ChannelZ | no it's in voicemail.conf |
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| 07:18.13 | cjk | hi, is there a way in asterisk to know which channel leg send the BYE message. Did the caller or the callee haung up? |
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| 09:01.58 | ovoshlook | hello all! I have digium hardware TE133. By default as I see it at T1 mode. How to switch this card to E1? |
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| 09:07.29 | kaldemar | use t1e1override (older drivers) or default_linemode (newer drivers) module parameter for modprobe. |
| 09:08.48 | kaldemar | ovoshlook: t1e1override=0xFF or default_linemode=e1 |
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| 09:29.22 | ovoshlook | kaldemar thanks. already fixed |
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| 11:01.22 | ovoshlook | hello. I build little system for testing rtp traffic (2 ways or not). So I have 2 ways audio of one bridge for example Got RTP packet from 10.34.5.22:11798 (type 08, seq 000721, ts 1011208480, len 000160) |
| 11:01.29 | ovoshlook | Sent RTP packet to 10.34.5.35:11794 (type 08, seq 000360, ts 1011208480, len 000160) |
| 11:01.34 | ovoshlook | Got RTP packet from 10.34.5.35:11794 (type 08, seq 001008, ts 199879360, len 000160) |
| 11:01.41 | ovoshlook | Sent RTP packet to 10.34.5.22:11798 (type 08, seq 049291, ts 199879360, len 000160) |
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| 11:02.22 | ovoshlook | I drr same ts with from and to packets for different IPs- and I know that is first leg |
| 11:02.42 | ovoshlook | and with same results i see second leg |
| 11:03.05 | ovoshlook | How I can ccheck that this legs have one bridge |
| 11:03.16 | ovoshlook | ? |
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| 11:49.39 | KevinRoberts | <PROTECTED> |
| 11:51.03 | WIMPy | Well, you need to send the calls to voicemail in your extensions.conf. |
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| 11:51.44 | KevinRoberts | this is login to the mailbox not leaveing messages yet |
| 11:54.01 | salz212 | Hi would g722 enabled in general asterisk install to play IVRs? |
| 11:54.02 | ovoshlook | <PROTECTED> |
| 11:54.14 | ovoshlook | <PROTECTED> |
| 11:54.19 | ovoshlook | Sent RTP packet to 10.34.5.35:11794 (type 08, seq 000360, ts 1011208480, len 000160) |
| 11:54.27 | ovoshlook | Got RTP packet from 10.34.5.35:11794 (type 08, seq 001008, ts 199879360, len 000160) |
| 11:54.35 | ovoshlook | Sent RTP packet to 10.34.5.22:11798 (type 08, seq 049291, ts 199879360, len 000160) |
| 11:54.48 | ovoshlook | I see same ts with from and to packets for different IPs- and I know that is first leg |
| 11:54.54 | ovoshlook | and with same results i see second leg |
| 11:55.02 | ovoshlook | How I can ccheck that this legs have one bridge? |
| 11:56.51 | salz212 | Hi all, can we play sound file in astersk thorugh call having only AMR codec? |
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| 12:14.23 | pzn | I got surprised :-) my asterisk server has about 400.000 files in /var/log/asterisk/event_log.*, how can I diagnose why so many files? |
| 12:14.56 | [TK]D-Fender | I'd look at the files. |
| 12:15.40 | pzn | [TK]D-Fender, sorry, forgot to mention: all of them have size = 0 |
| 12:17.18 | WIMPy | And if you do an ls you can go and bake a cake to have with the coffee you have to drink befor anything happens? |
| 12:17.41 | pzn | WIMPy, for sure :-) |
| 12:18.25 | WIMPy | Before deleting them, rename the directory and copy over the files you want to keep. |
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| 12:31.48 | pzn | WIMPy, all ok now, asterisk running with a new /var/log/asterisk; old directory has been renamed to /var/log/asterisk-old/ for analysis |
| 12:32.57 | wdoekes | I'd look at logger.conf |
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| 12:41.25 | WIMPy | Logrotate every minute? |
| 12:41.59 | WIMPy | Would still be >9 months. |
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| 12:57.02 | wdoekes | or.. strftime-based logging |
| 12:57.06 | wdoekes | with %s |
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| 13:01.24 | wdoekes | oh, I thought that was a joke/misconfiguration, but that's the default (and only) config of the timestamp rotate strategy |
| 13:01.27 | wdoekes | <PROTECTED> |
| 13:01.30 | wdoekes | <PROTECTED> |
| 13:01.33 | wdoekes | <PROTECTED> |
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| 15:15.07 | fornax | Hi, I'm trying with the Asterisk REST interface ARI and want to manipulate a Custom Devstate. Now I'm not able to change the type from custom to Stasis and so I cannot set the state of the device. Is there any workaround? It absolutely makes no sense that I cannot change my custom devicestates. How can I make them Sasis controlled? |
| 15:18.15 | Kattyroo | has anyone heard of retron 5? |
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| 15:21.53 | [TK]D-Fender | fornax: https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Asterisk+REST+API |
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| 15:26.50 | fornax | [TK]D-Fender Yes, this is what I use |
| 15:27.41 | fornax | The problem is that ARI can only control Statis Device States and I have to control a Custom device state. When I try to control a custom state I get: stasis_app_device_state_update: Update can only be used to set 'Stasis:' device state! |
| 15:28.11 | fornax | But I cannot change the device state to Statis, because it is changed via sccp when I press a button |
| 15:28.12 | [TK]D-Fender | Show us |
| 15:29.07 | fornax | [TK]D-Fender What do you want to see? |
| 15:29.20 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-pyannswifhrexico) |
| 15:29.20 | [TK]D-Fender | What you're doing, and what you're getting. |
| 15:30.09 | fornax | I execute curl -v -u oser:password -X PUT "http://192.168.1.1:8088/ari/deviceStates/Custom:livingroom?devicestate=NOT_INUSE" |
| 15:30.18 | fornax | And I get ERROR[1205]: res_stasis_device_state.c:211 stasis_app_device_state_update: Update can only be used to set 'Stasis:' device state! |
| 15:30.48 | fornax | Here it is written that only devicestates with Stasis: can be controlled : http://www.siplab.cn/ast/Asterisk-Admin-Guide/Asterisk-12-Devicestates-REST-API_27197783.html |
| 15:31.23 | fornax | But I cannot change it to Stasis because it is set by my sccp conf: button = feature, Living Room, devstate, livingroom |
| 15:32.17 | [TK]D-Fender | <PROTECTED> |
| 15:32.19 | fornax | And I cannot couple a Custom: with a Stasis: state such as in http://lists.digium.com/pipermail/asterisk-app-dev/2014-March/000417.html because I do not dial an extension. The devstate changes asynchronous when I press a button on a phone. |
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| 15:36.47 | fornax | Okay, I do it with: http://192.168.1.1:8088/ari/asterisk/variable?variable=Custom:livingroom&value=NOT_INUSE |
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| 15:37.03 | fornax | I get "== Setting global variable 'Custom:livingroom' to 'NOT_INUSE'" in the cli |
| 15:37.17 | fornax | but when I run devstate list I get : --- Name: 'Custom:livingroom' State: 'INUSE' |
| 15:37.22 | fornax | So it does not change the devstate |
| 15:37.47 | [TK]D-Fender | that is not a variable.... |
| 15:38.04 | [TK]D-Fender | use the DIALPLAN FUNCTION |
| 15:38.49 | fornax | [TK]D-Fender But I can only execute it in the dialplan when I dial an extension |
| 15:40.34 | fornax | [TK]D-Fender But the devstate changes in the background without dialing an extension. When I hit the button, the light becomes yellow and the devstate is set. Now I want to get an event via ARI about the devstate event to set a light in my livingroom. |
| 15:41.17 | [TK]D-Fender | "core show help devstate change" <- |
| 15:41.43 | [TK]D-Fender | CLI & AMI hove options for this. |
| 15:41.45 | [TK]D-Fender | have* |
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| 15:43.42 | fornax | Yes, when I use devstate change (devstate change Custom:livingroom INUSE) the devstate changes wihtout problems. But how can I then get this event to ARI, is this still not possible? |
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| 15:44.53 | fornax | I need some webservice which gets notofied when the devstate changes. For example when i execute devstate change Custom:livingroom INUSE becuase i think it is the same which is executed internally when I press my phone button |
| 15:46.03 | fornax | If I could somehoe tell Stasis that it should monitor the Custom event it would work for me |
| 15:47.05 | [TK]D-Fender | "when I press my phone button" <- what does this actually mean & do? |
| 15:48.52 | fornax | I have a cisco 7914 extension module on my cisco phone with speed dial buttons. I press on such a button and I see the state change in cli when I enter devstate list |
| 15:49.59 | fornax | In my sccp.conf this is handled by: button = feature, Living Room, devstate, livingroom |
| 15:50.35 | [TK]D-Fender | Ok, that is foreign functionality to me. Does anything regiter in AMI? |
| 15:50.51 | [TK]D-Fender | that may be the only way to catch it as it happens |
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| 15:52.14 | fornax | How can I find this out? I did not work with ami right now |
| 15:52.50 | fornax | I think I need to use some AMI api, isn't it? |
| 15:53.43 | [TK]D-Fender | It's all in the book... |
| 15:54.35 | fornax | thank you very much |
| 15:54.48 | fornax | I will have a look and try to find a solution and post it here |
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| 16:27.42 | msaraiva | Hi all |
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| 16:33.40 | msaraiva | Is 10 the maximum available debug level? |
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| 16:38.13 | gesnaud | Hello there |
| 16:38.35 | gesnaud | I'm looking for support about WebRTC on asterisk v11.12 |
| 16:39.56 | gesnaud | I've this error: a=ice-ufrag:cAVEaxa8k8x/5jFL... UNSUPPORTED OR FAILED. |
| 16:40.23 | gesnaud | I think everything is well configured (avpf, encryption, ldd with uuid is ok etc.) |
| 16:40.49 | gesnaud | The sig make client ringing, but could not establish RTP |
| 16:41.40 | gesnaud | I've got this in my firebug: "Failed to set remote answer sdp: Called with SDP without ice-ufrag and ice-pwd." |
| 16:42.03 | gesnaud | I'm out of ideas here... please help :) |
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| 16:59.41 | msaraiva | What are the possible reasons for a 488 on an inbound call (i'm using Asterisk 12 with PJSIP)? |
| 16:59.48 | msaraiva | Codec? Malformed invite? |
| 17:01.05 | msaraiva | Actually, scratch codec, as i don't have the same issue with chan_sip |
| 17:03.56 | msaraiva | There's nothing else on the sip trace or debug logs that could point to the 488 reason :( |
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| 17:50.12 | klootzak | Hi guys! I have many "xmpp_client_receive: Parsing failure: Invalid XML." lines in log. Asterisk sending messages, but not receiving it. I tried with ejabberd 14.07 and Prosody from trunk and 0.9.4 version. No one works. Asterisk 12.6.0. |
| 18:01.10 | newtonr | klootzak, collect Asterisk logs and packet captures, post it on the users list. Perhaps someone will be able to look through it. |
| 18:01.27 | newtonr | They'll likely have to look at what is being sent and received to see why Asterisk doesn't like it. |
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| 18:42.54 | *** join/#asterisk polycomtesting (4028f472@gateway/web/freenode/ip.64.40.244.114) |
| 18:43.05 | polycomtesting | Hi everyone - how can I upgrade firmware on a Polycom 450 ? |
| 18:43.41 | polycomtesting | I am just browsing the menu on the phone and I don't see anywhere to allow me to enter HTTP server info for file download - also which file and what name should be put on server for the phone to recognize it? |
| 18:44.27 | klootzak | newtonr, seems like it works with usetls=yes with any server. |
| 18:45.28 | rrittgarn | @polycomtesting: use dhcp option 66 to tell it where to look |
| 18:45.38 | rrittgarn | and the filename must be the same as what you extract from the polycom website |
| 18:46.01 | rrittgarn | polycom also has a site that allows you to provision your phone with working fimware/configs but i dont remember it offhand |
| 18:46.53 | klootzak | newtonr, oh no, parsing failure again. |
| 18:47.40 | seanbright | polycomtesting: http://voipt2.polycom.com/ |
| 18:47.50 | polycomtesting | rrittgarn - oh my - this is so complicated. Thanks a lot for the response. Where does DHCP Option 66 get setup? on a router? |
| 18:51.15 | polycomtesting | seanbright - thanks a lot - so I have format the file system right now and there is no firmware on it I think. what steps should I take? the site you mentioned is for when the phone has a firmware on it already |
| 18:52.45 | seanbright | then you need to configure DHCP appropriately |
| 18:53.19 | seanbright | polycom has a lot of documentation on the topic. |
| 18:54.31 | seanbright | polycomtesting: http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip450.html |
| 18:55.34 | seanbright | check out the "How to Provision a Polycom Phone" document |
| 18:55.58 | polycomtesting | seanbright - can I go about this without "provisioning" as I can do the setup via webportal later on |
| 18:56.30 | polycomtesting | I find this much more complicated than Cisco and Aastra phone for a simple firmware upgrade |
| 18:56.30 | seanbright | read the documentation |
| 18:56.59 | seanbright | polycomtesting: well then you should have purchased a cisco or aastra |
| 18:57.28 | polycomtesting | can you please guide me to specific document - there are many documents there |
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| 18:57.37 | seanbright | go to that URL |
| 18:57.45 | seanbright | hit Ctrl-F on your keyboard |
| 18:58.04 | seanbright | paste "How to Provision a Polycom Phone" into your browser's find tool |
| 18:58.50 | seanbright | if you are unable to find it, perhaps you should hire a consultant |
| 18:59.31 | [TK]D-Fender | [14:43]polycomtestingI am just browsing the menu on the phone and I don't see anywhere to allow me to enter HTTP server info for file download - also which file and what name should be put on server for the phone to recognize it? <- it's in the BOOT ROM |
| 18:59.33 | klootzak | newtonr, I found it. This happening after receiving presence with status message "ð¤". After that asterisk stops receiving messages until restart. |
| 18:59.40 | [TK]D-Fender | polycomtesting: And you don't need DHCP to set it |
| 19:00.38 | polycomtesting | [TK]D-Fender - thanks - how do I get to BOOT ROM? |
| 19:01.28 | polycomtesting | what do you mean by don't need DHCP 66 to set it up? |
| 19:01.34 | seanbright | heh |
| 19:01.45 | seanbright | read. the. documentation. |
| 19:02.07 | polycomtesting | They use UC and SoundPoint firmware - this is all over. What is the basic firmware for these phones? |
| 19:02.11 | newtonr | klootzak, probably want to file a bug on the tracker and include your collected debug and packet captures |
| 19:02.50 | polycomtesting | for example, this "SoundPoint IP and SoundStation IP BootROM 4.3.0 Release Notes" OR this, "Polycom UC software 4.0.7 for SoundPoint IP, SoundStation IP and VVX 500 and VVX 1500 Business media Phones[combined] " |
| 19:03.18 | seanbright | well, one of those is release notes. the other is a firmware download. |
| 19:04.28 | polycomtesting | seems like there are many different types of firmware - anyone with experience here can tell me what their basic firmware name is? I am just using their SIP phones and not their whole backend system |
| 19:05.22 | seanbright | polycomtesting: check the "SIP Downloads Matrix" link on that page i sent you |
| 19:05.51 | seanbright | it shows you the correct version for the 450 |
| 19:08.09 | [TK]D-Fender | [15:01]polycomtestingwhat do you mean by don't need DHCP 66 to set it up? <- YOU DON'T |
| 19:08.47 | [TK]D-Fender | polycomtesting: Reboot the phone and BEFORE the SIP app loads you can log into the BOOT ROM and MANUALLY set an IP address, and protocol for provisioning |
| 19:11.53 | klootzak | newtonr, without packet captures not accepted? |
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| 19:12.27 | wdoekes | klootzak: that's not a nice nickname, now is it? |
| 19:12.59 | newtonr | klootzak, Probably not - you need to be able to demonstrate the issue. A description won't typically allow us to reproduce the issue for testing and what not unless it is a very simple issue. |
| 19:13.20 | newtonr | klootzak, In any case, report whatever you can get and if it isn't enough we'll decide from there. |
| 19:15.24 | Nugget | klootzak should change his nick to something less offensive, like "eikel" |
| 19:17.16 | test321 | newtonr, ok |
| 19:18.00 | newtonr | test321, here are the guidelines for the issue tracker BTW https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines |
| 19:18.09 | newtonr | test321, good to go through those if you haven't already. |
| 19:18.49 | test321 | newtonr, thanks |
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| 19:38.34 | SuperNull | hey guys, anyone ever experience asterisk not responsive to anything? core restart doesnt work yet asterisk -r lets you connect. No output running 1.8.27.0 |
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| 19:47.41 | rrittgarn | SuperNull: You running realtime? |
| 19:47.58 | rrittgarn | i've seen that when it can't connect to a DB, or when DNS servers aren't responding / timing out |
| 19:48.09 | rrittgarn | but then it was REALLY slow, not 'unresponsive' |
| 19:48.24 | rrittgarn | would also suggest a reboot... if same behavior after that.. has it ever worked? |
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| 22:33.10 | ovoshlook | Hello. How can I set RTP debug only to log files. Not to console? |
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