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| 01:09.20 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0 (2018/10/09), Security Only: 15.6.1 (2018/09/20); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 03:23.47 | simbalion | Hi, what controls the length of time a queue extension is tried? Is it 'timeout'? Mine is set to 10 presently and it only rings a line twice before trying others, I want to increase that |
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| 03:28.08 | simbalion | nvm found out that is right, through testing :) |
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| 12:39.20 | jamesaxl | Hello |
| 12:39.31 | jamesaxl | Samot: hi, I hope you are good. |
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| 13:15.14 | pchero_work | Hi guys, I would like to know the sequence number of ari events of the channel. Is there any good idea..? |
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| 13:19.57 | file | I don't think anyone has really documented the sequence |
| 13:23.09 | pchero_work | ah, not that one. |
| 13:23.41 | pchero_work | If the ari send a event via websocket, I would like to know the sequence of the event of the channel. |
| 13:24.29 | pchero_work | If the channel sent a 5 events, every each event has a sequence 1 to 5. |
| 13:24.44 | file | they don't have sequence numbers |
| 13:24.51 | file | but they are guaranteed to be in order over the websocket |
| 13:25.37 | pchero_work | file: thanks. You're right. I need to look at it more in different way.. |
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| 16:02.10 | Kobaz | i don't know what's going on... i can't log into this one polycom anymore |
| 16:02.36 | Kobaz | the web portal says invalid password... but i have a forced password set in <device> in the polycom cfg |
| 16:16.10 | shanth | might be why the sales guys email is slow |
| 16:16.19 | shanth | he's gonna talk to the CEO about it |
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| 17:17.17 | d1gital | \o hello |
| 17:17.42 | d1gital | I have libopus installed, but it doesn't seem to be loaded when I try to load the opus codec: [2018-10-23 16:58:34] WARNING[22106]: loader.c:573 load_dynamic_module: Error loading module 'codec_opus_open_source.so': /usr/lib/asterisk/modules/codec_opus_open_source.so: undefined symbol: opus_encode |
| 17:17.57 | d1gital | any idea what I've missed? |
| 17:18.05 | FrankyCyborg | that's a linker problem |
| 17:18.48 | FrankyCyborg | your module is missing the named symbol - most likely, because your .so file is not linking against the opus library |
| 17:19.21 | FrankyCyborg | are you on linux or mac os x? |
| 17:19.34 | d1gital | ubuntu, installed from apt |
| 17:19.46 | d1gital | you're right; libopus isn't in the ldd output for that module |
| 17:19.48 | FrankyCyborg | have you installed the opus library on your system? |
| 17:19.57 | FrankyCyborg | yep, that's it |
| 17:21.11 | d1gital | strange. I installed asterisk-opus from the ubuntu repos. I wonder why it wouldn't be linked against libopus |
| 17:21.37 | FrankyCyborg | maybe because that wouldn't install the opus library itself? |
| 17:22.16 | FrankyCyborg | do you have header/library files of opus in /usr/include or /usr/lib (or whatever paths you are using)? |
| 17:22.35 | d1gital | I do have libopus installed, and I can see the opus_encode symbol defined in "nm -D /usr/lib/x86_64-linux-gnu/libopus.so.0.5.2" |
| 17:24.00 | FrankyCyborg | hmm, if it's not a platform mismatch (32/64-bit), it may be possible to link the .so file afterwards against that opus library? (I would make a backup of it first) I know how to do that on mac os x, but not on linux |
| 17:26.40 | FrankyCyborg | but it's strange, that this issue happens, when you used the repos |
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| 18:15.14 | igcewieling | It appears the sip to pjsip config conversion script puts all the permits in a single ACL. |
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| 18:49.28 | shanth | when i register/connect with my sip provider to trunk, is that unencrypted when it sends the password? |
| 18:55.33 | [TK]D-Fender | No, it is MD5'd for auth, but the audio is unencrypted most of the time. |
| 18:55.57 | [TK]D-Fender | Then again an attacker would have to have a way to get themselves in the packet path |
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| 19:04.45 | trmg | I'm having an odd issue that I'm hoping I can get some pointers on. |
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| 19:09.40 | kfife | Can I limit the number of concurrent calls for a specific peer in sip.conf, or must that be done in the dialplan? |
| 19:12.36 | [TK]D-Fender | either |
| 19:34.47 | qxork | Mr Fife! |
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| 19:35.40 | qxork | kfife: thereâs a call-limit= availabile in sip.conf |
| 19:35.57 | qxork | you can also use group in dialplan which is my prefered method |
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| 19:40.40 | shanth | thinking of using Flowroute but the number of 404 pages they have is making me nervous |
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| 19:52.51 | trmg | shanth: FWIW I used Flowroute for a while and have no complaints with their service. |
| 20:08.37 | *** join/#asterisk DanFromUK (sid21651@gateway/web/irccloud.com/x-nmqvyhdvvlkccsgo) |
| 20:09.49 | DanFromUK | Hi. I'm trying to configure a new Polycom phone. It's something I've done hundreds of times. I'm using the same provisioning files, just changing the sip login details. But Asterisk seems to be simply ignoring the REGISTER packets. |
| 20:09.55 | DanFromUK | It's not replying at all. |
| 20:10.20 | DanFromUK | I have captured the packets on the firewall, and also on the server itself, but pjsip logger shows nothing from that ip address. |
| 20:10.48 | DanFromUK | I've got another phone, on the same external network, and it's working fine. Any ideas what could cause asterisk to simply ignore packets? |
| 20:11.42 | [TK]D-Fender | Because maybe it's not passing the firewall to get to * |
| 20:11.56 | [TK]D-Fender | go prove there are no rules that could block it, then prove that * is listening on the right port, etc |
| 20:12.03 | DanFromUK | tcpdump on the asterisk box says it is passing the firewall |
| 20:12.34 | DanFromUK | and i've got another phone sat next to me which is able to communicate with no problems at all |
| 20:13.50 | [TK]D-Fender | Less anecdotes, more backup.... |
| 20:16.32 | DanFromUK | Might be some form of SIP ALG issue actually. Running tcpdump on the Asterisk machine, it appears that the "broken" phone and one of the working phones are using the same source port. I wonder if asterisk is ignoring it because the credentials don't match another phone using that source port. |
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| 20:22.12 | igcewieling | DanFromUK: if the source port shown in sip show peers (or whatever is right for pjsip) is 5060 for more than one device behind the same router, that strongly indicates ALG |
| 20:23.17 | DanFromUK | yes. this is at home. just trying to catch up work and set up a device for a client. it's just odd that i've got 6 other devices working fine here. going to check out the router. |
| 20:24.16 | DanFromUK | nope. sip alg turned off. time for a reboot i think. |
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| 20:37.42 | [TK]D-Fender | Or... we could try NOT fucking with the evidence... |
| 20:37.52 | [TK]D-Fender | And actually look |
| 20:58.47 | trmg | Any pointers on what to troubleshoot when PJSIP falls apart and you're seeing a ton of various "task processor queue reached 500 scheduled tasks again." in the log? |
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| 22:34.29 | Slade | was at a little think tank for small businesses today. one of the ideas made me wonder the feasibility of little guys/providers doing their cellphone time in the same manner that they can do their voip time |
| 22:35.17 | efm | Slade: like a MVNO? |
| 22:35.36 | Slade | yes |
| 22:47.37 | DanFromUK | I realise now that the other phones at home are connected to another asterisk server at the datacentre. Either way, the packets from this phone are not being accepted by Asterisk, even though the packets are being received by the server os. |
| 22:49.21 | DanFromUK | <PROTECTED> |
| 22:49.24 | efm | Slade: did you wonder in a positive way? |
| 22:49.50 | DanFromUK | [TK]D-Fender: Not really sure why you are responding like that. tcpdump clearly shows the packets are being received by the server os, although you are implying that it's either not or i'm lying? sending you a tcpdump from the os will only allow you to validate what i'm saying. or you can simply trust me when I say that asterisk is not responding to a specific ip address or not receiving the packets. |
| 22:50.12 | [TK]D-Fender | or that we're seeing prefiltered results |
| 22:50.25 | [TK]D-Fender | So far we are not seeing a firewall having been emptied out or anything at all |
| 22:50.32 | Slade | efm, hum.. a cynical but hopeful way? |
| 22:50.33 | DanFromUK | tcpdump is post filter? |
| 22:52.01 | [TK]D-Fender | https://superuser.com/questions/925286/does-tcpdump-bypass-iptables |
| 22:52.20 | file | tcpdump is before iptables |
| 22:52.32 | [TK]D-Fender | ^ |
| 22:52.38 | DanFromUK | noted. |
| 22:52.48 | [TK]D-Fender | You see things * may never get to |
| 22:53.08 | DanFromUK | ok. i understand. |
| 22:57.40 | efm | Slade: you could call it 'decentralized' |
| 23:09.12 | Slade | efm, i have to think the carriers would overcharge.. but it would be awesome if it was possible |
| 23:18.00 | efm | Slade: I use project fi, and think it's awesome. But that's the sort of thing you are describing |
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| 23:18.28 | Slade | i know very little about project fi |
| 23:19.12 | efm | phone + voip + cellular + internet for a modest price and pay as you go bw |
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