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| 00:22.31 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.20.0 (2018/03/15), Standard: 15.3.0 (2018/03/15); 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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| 09:15.18 | bllackjack | taskprocessor.c: The 'subp:ast_channel_topic_all-000029e6' task processor queue reached 500 scheduled tasks. |
| 09:15.23 | bllackjack | Can anyone help me with this? |
| 09:15.39 | bllackjack | The calls get stuck in waiting and this is what I get in the log |
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| 10:15.32 | [J]oules | bllackjack: you have pjsip extensions loosing registration too? |
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| 10:36.29 | bllackjack | [J]oules: Just SIP, using the legacy. |
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| 12:55.17 | sigmaorion | hi! I've registered two Cisco 7960 phones with Asterisk (TCP transport) from different networks, different ISPs. If only one phone is connected, everything works fine, but once I connect the second one I get "forbidden" when I try to dial. Asterisk logs show "username mismatch, has <xxx> digest has <yyy>". |
| 12:55.54 | sigmaorion | also, when I connect the second phone, the first one changes as if it has registered from the same external IP the second one has |
| 12:56.25 | sigmaorion | Both extensions are configured as type = peer. Any ideas would be really appreciated! |
| 12:57.42 | Samot | Don't use peer |
| 12:57.46 | Samot | Because it's matching on IP |
| 12:58.11 | sigmaorion | Samot, should I use "friend"? |
| 12:58.17 | Samot | Yes. |
| 12:58.25 | Samot | For phones, yes. |
| 13:00.44 | sigmaorion | thank you very much for you answer! I have read the difference between friend, user and peer, but haven't completely understood... would you please briefly try to clarify why, in this case, I should use type = friend? |
| 13:03.18 | [TK]D-Fender | sigmaorion, Are you trying to register 2 phones to the same peer entry in *? |
| 13:04.05 | sigmaorion | no, each phone is registering to its own extension definition in sip.conf |
| 13:04.25 | sigmaorion | phone A -> [100], phone B -> [200] |
| 13:04.33 | [TK]D-Fender | ok |
| 13:04.42 | sigmaorion | I have all my non-Cisco phones working perfectly like "peer" and UDP |
| 13:04.52 | sigmaorion | this issue is showing only when the transport is TCP |
| 13:05.52 | sigmaorion | it looks like Asterisk thinks the SIP messages are all coming from the same phone when on TCP, no matter what |
| 13:08.14 | file | chan_sip didn't use the port when looking up from TCP or TLS sources |
| 13:08.30 | file | it was changed, https://gerrit.asterisk.org/#/c/7430/, and will be in the next release |
| 13:09.22 | sigmaorion | hi @file, and will "friend" type workaround that in the meantime? |
| 13:09.30 | file | I don't remember chan_sip. |
| 13:10.49 | file | can only keep one SIP stack in his head at a time |
| 13:13.57 | Samot | https://www.irccloud.com/pastebin/8Zkg1riq/ |
| 13:14.13 | Samot | ^ That's how Chan_SIP matches incoming calls |
| 13:15.18 | Samot | https://www.irccloud.com/pastebin/noSDPMKL/ |
| 13:15.32 | Samot | ^^ That's the differences between friend/user/peer |
| 13:15.38 | Samot | https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample |
| 13:15.42 | Samot | ^^ Full details |
| 13:16.49 | sigmaorion | Samot, that's clear like water, and that makes sense with all my UDP phones defined as peer and working... but why is TCP not working? |
| 13:17.12 | file | because for TCP and TLS it did not check using the port number. |
| 13:17.40 | sigmaorion | @file, yes, got it... that's what I didn't know and didn't find anywhere |
| 13:17.47 | sigmaorion | thank you very much for that tip! |
| 13:17.52 | Samot | If the device needs to do stuff based on username then peer is wrong. |
| 13:18.14 | Samot | Peer should be mainly for your "trunks" |
| 13:18.48 | sigmaorion | but I've read some posts discouraging the use of friend for security reasons... |
| 13:18.58 | Samot | "some" |
| 13:19.01 | sigmaorion | something about an "enumeration attack" |
| 13:19.13 | sigmaorion | isn't that true? |
| 13:19.25 | Samot | Those are posts made by people who have no idea about SIP security. |
| 13:19.48 | Samot | If you need your devices to do things based on the user.. |
| 13:20.06 | Samot | the device username, then you need to use friend. |
| 13:20.15 | Samot | So it makes both a user and a peer entry |
| 13:21.34 | sigmaorion | Samot, thank you for your help! |
| 13:21.49 | sigmaorion | I will try type friend right now and see if it works! :) |
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| 13:40.54 | sigmaorion | gents, so far it looks good! I cannot confirm until another guy reaches the other phone, a mile away from here, and we can fully test it. But I think it's working... |
| 13:41.00 | sigmaorion | thank you for your help! |
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| 13:44.10 | sigmaorion | @file, where can I read about UDP and TCP differences? I haven't ever found anything about Asterisk not checking port when on TCP or TLS... |
| 13:44.22 | sigmaorion | any specific documentation where I can read more? |
| 13:44.34 | file | I don't know if there was ever any documentation or mentions |
| 13:46.25 | sigmaorion | so, may I ask, how do you know it? I mean, is it because you are involved with the code or something? |
| 13:46.43 | sigmaorion | just being curious so I can look for more information somewhere, I'm interested in digging |
| 13:46.54 | file | I reviewed the code review that I linked |
| 13:47.10 | file | https://gerrit.asterisk.org/#/c/7430/ |
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| 13:55.16 | sigmaorion | well, unfortunately it has failed again... 'sip show peers' show the same IP address for both phones (on different ISPs) and the phone cannot place calls |
| 13:55.26 | sigmaorion | so, it didn't actually work |
| 13:57.52 | sigmaorion | there must be something wrong with the registration process... how could both extensions end up registered as if they were under the same IP? |
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| 14:21.49 | [J]oules | anyone happen to know a2billing ? |
| 14:23.01 | [J]oules | having an issue where a2billing is sending a terminated call to another pbx. found what its doing but unsure where to remove it.. It's not in any of the settings in gui |
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| 14:31.32 | sigmaorion | hi J]oules, I use a2billing... could you please give some more detail? is it resending the call through another trunk? |
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| 14:56.55 | [J]oules | a2billing sends the call to another pbx. Even though our freepbx terminated the call |
| 14:57.13 | [J]oules | i have a short debug which clearly shows whats happening |
| 14:57.34 | [J]oules | but I am guessing the issue is in the mya2billing db |
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| 15:11.47 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.20.0 (2018/03/15), Standard: 15.3.0 (2018/03/15); 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 |
| 15:17.54 | [J]oules | Samot: originally i asked if anyone knows a2b. |
| 15:18.19 | [J]oules | there are a lot of people here so perhaps someone knows and wont kill me on cost to fix it |
| 15:18.45 | [TK]D-Fender | And even now you're showing half a call worth of processing |
| 15:18.51 | [TK]D-Fender | never the full call |
| 15:19.06 | [TK]D-Fender | not showing matching config setting to back it up |
| 15:19.10 | igcewieling | Thankfully I don't have to deal with CDRs and billing, we use the CDRs from the carrier for billing. |
| 15:20.07 | [TK]D-Fender | http://newsroom.plantronics.com/press-release/plantronics-acquire-polycom-2-billion |
| 15:20.24 | [TK]D-Fender | there goes the neighbourhood |
| 15:21.28 | igcewieling | Plantronics is the only headset I recommend. It could be worse, Microsoft could be buying Polyvom. |
| 15:21.37 | Samot | ^^^ |
| 15:21.58 | Samot | Plantronics/Polycom is the combo I use too. |
| 15:22.06 | Samot | So yeah, I'm good with this. |
| 15:22.31 | Samot | Could've been the other way as well. Still be cool. |
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| 15:24.00 | igcewieling | Digium needs to buy Polycom so Broadsoft doesn't 8-) |
| 15:25.21 | Samot | I like Broadsoft. |
| 15:25.37 | Samot | Way better than the M6 platform. |
| 15:29.11 | [TK]D-Fender | <igcewieling> Digium needs to buy Polycom so Broadsoft doesn't 8-) <- Digium already has their own phones and other hardware, they should probably concentrate on making those products better |
| 15:30.34 | igcewieling | If they bought polycom they would not need to make their phones better. |
| 15:31.14 | [TK]D-Fender | igcewieling, but I'm actually against the hardware & software side merging. What made * great was that it was intented to work with all sorts of random devices hence its name. When you start developing specific hardware to match special functions in software then you end up being just another proprietary solution and standards can go out the window ... along with options. |
| 15:32.01 | [TK]D-Fender | If one day my phones piss me off I can ditch them in a heartbeat. If * somehow becomes some kind of serious pain for me it can go too and my hardware isn't just written off. |
| 15:32.16 | [TK]D-Fender | McDLT the bitch. That's what I'm saying.... |
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| 15:43.21 | igcewieling | Polycom already does special stuff for Broadsoft and MS Link -- which is really annoying. |
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| 17:00.40 | [J]oules | igcewieling: you want annoying? How about grandstream phone that boots then locks up? |
| 17:03.59 | igcewieling | [J]oules: No thanks. I tried Grandstream in 2005, never again. |
| 17:06.37 | [TK]D-Fender | BT-1XX & GXP-2000 era. Yeah, those were garbage |
| 17:07.50 | file | has more Digium phones than he needs |
| 17:08.15 | [J]oules | never tried digium phones |
| 17:08.27 | [J]oules | perhaps i should |
| 17:08.33 | [TK]D-Fender | I just made an actual purchase of one of thier newest executive models because of the sidecar & management and it orks decently. The build quality, sound, etc were all what one should expect in general (though of course Polycom does that little bit extra. Build I'd compare to Aastra & Digium's 1st get (that I own) |
| 17:08.37 | file | they help pay my mortgage |
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| 17:22.02 | igcewieling | If Digium had phones in 2003 when I was deciding what to standardize on there is a good chance I'd have chosen them. |
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| 17:26.20 | [J]oules | u use digium phones igcewieling |
| 17:26.23 | [J]oules | ? |
| 17:30.43 | igcewieling | No, I use Polycom |
| 17:30.57 | igcewieling | since Digium didn't make phones at the time. |
| 17:32.53 | meyou | you guys played with Yealinks? |
| 17:38.08 | [TK]D-Fender | From what I've heard they look nice (kinda) in as much as they knock-off Polycom & Cisco, but the sound, fit & finish, and stability is really second-rate |
| 17:40.17 | [J]oules | my disti just said grandstream/yealink have the most returns/warranty claims. |
| 17:40.27 | [J]oules | polycom/digium hardly ever |
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| 17:41.05 | [J]oules | but he told me digium phones cant be automated unless using digium pbx |
| 17:41.15 | [J]oules | seems strange |
| 17:44.33 | meyou | rip we just replaced polycoms with yealinks |
| 17:44.52 | meyou | i haven't heard any SQ complaints yet, and featurewise they seem pretty slick |
| 17:45.16 | meyou | although i did find a funny bug, as soon as 12:00AM UTC rolls over, all of your calls from before that will show "Yesterday" in history |
| 17:45.45 | meyou | like 5PM here is 12 UTC, and at 5:01 a call from 4:59 says yesterday, even though the times show correct PST times |
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| 21:31.47 | [J]oules | why would this be in cli when openssl connect shows certs are OK |
| 21:31.47 | [J]oules | 5:31 PM pjproject:0 <?>: SSL 6 [SSL_ERROR_ZERO_RETURN] (Read) ret: 0 len: 32000 |
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| 23:05.52 | sawgood | Question: In general when using PJSIP (and you have) (3 phones registered as x1000) ... is there anyway to "share" HOLD (so that) if one phone as x1000 puts a call on HOLD ... the other (2) phones would be able to press a button an take that call as x1000 as well? |
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| 23:08.54 | rpifan | ihi |
| 23:18.58 | sawgood | Thatâs correct. That type.of feature does not exist in the SIP world with asterisk. |
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| 23:31.56 | Samot | Actually, it does. Just has to be supported. |
| 23:32.15 | Samot | In Asterisk, that's what Parking Lots/BLF is kinda for. |
| 23:32.29 | Samot | To hold so others can pick it up. |
| 23:34.47 | sawgood | right parking lots ... awesome ... but not natively |
| 23:35.41 | sawgood | some like to call that SLA or SCA ... but that is so broken or has conditions that make you not use it |
| 23:36.15 | Samot | SCA/BLA has to be supported. |
| 23:36.24 | Samot | By both the server and the device. |
| 23:36.32 | sawgood | Samot: What about getting a phone to have a DSS/BLF key which monitors a VM-box for a non-registered phone? |
| 23:36.47 | Samot | I just answered in #freepbx |
| 23:36.52 | Samot | Read the wiki. |
| 23:37.03 | sawgood | ah! thanks! |