| 00:00.07 | [TK]D-Fender | atan: "core show function CONNECTEDLINE" |
| 00:00.23 | [TK]D-Fender | heads off for the evening |
| 00:00.30 | newtonr | atan, https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information |
| 00:03.58 | atan | newtonr, looks like we're onto something here |
| 00:04.06 | atan | bsdice, ugh, they didn't did they?! :( |
| 00:04.12 | bsdice | no |
| 00:04.18 | bsdice | well, not sure |
| 00:04.41 | atan | The place I saw it was not using Apple products, unless Apple makes desk phones now |
| 00:04.46 | atan | It was just a simple 2 line display. |
| 00:04.57 | atan | Was really cool to see it work like it did though. |
| 00:05.09 | bsdice | and you are SURE this was not the phonebook in that phone? |
| 00:05.33 | atan | Unless for some reason this museum had me in their phone book... but I doubt it |
| 00:05.42 | atan | I think the connected party id is what I'm after |
| 00:06.47 | bsdice | is this the secret Digium phone collection? https://twitter.com/stgnet/status/387708344913625088 |
| 00:07.41 | file | nope |
| 00:07.43 | atan | Speaking of museums... that photo |
| 00:07.53 | file | that is Schmoozecom's |
| 00:10.08 | bsdice | does Digium even have a phone :) |
| 00:10.29 | newtonr | http://www.digium.com/en/products/asterisk/phones |
| 00:11.18 | atan | Looks to me like there's a D70, D40, D50 in that pic you posted |
| 00:11.25 | WIMPy | Usually companies selling phones or phone services don;t know how to use these things themselves. |
| 00:12.06 | atan | What do those cameras in the twitter photo have to do with voip? |
| 00:12.11 | file | has a D70 right there ----> |
| 00:13.01 | atan | I was looking at the D70 something fierce. Ended up getting a Cisco 99xx. |
| 00:13.03 | newtonr | atan, they probably use VOIP for two way audio, i've seen several security cams with two way audio |
| 00:13.07 | bsdice | any PCB photographs of Digium phone innards? |
| 00:13.11 | atan | Now I'm full of regret. |
| 00:13.28 | bsdice | Cisco... |
| 00:13.45 | bsdice | developed in China huh |
| 00:13.51 | atan | I don't care what they say. They can call it a "SIP" phone all they want. I can't for the life of me get it to work right. |
| 00:14.06 | ChannelZ-Wk | Maybe they meant to call it PIS |
| 00:14.07 | atan | Connects fine, makes and takes calls, but it's all wonky. |
| 00:14.19 | bsdice | define wonky |
| 00:14.43 | atan | When you take a call, or miss a call, it leaves a line open on the screen which disables all functions that would normally only be disabled when on an active call. |
| 00:14.55 | atan | Also, there's no "missed calls" type feature, as that is handled by CUCM |
| 00:15.23 | atan | + no way to define your own as far as I can see. Techs are only willing to help if connected to CUCM or CME. Of course. |
| 00:15.35 | bsdice | bleh |
| 00:15.45 | ChannelZ-Wk | which phone? |
| 00:16.05 | atan | 9951+9971 |
| 00:16.37 | ChannelZ-Wk | hmm |
| 00:16.37 | bsdice | just do useragent=RTCC/5.0.0.0 MediationServer sdpsession=RTCC/5.0.0.0 MediationServer sdpowner=Administrator callerid=Administrator and claim "hey it works with Microsoft!" |
| 00:17.13 | atan | After something like 40 hours on the phone with New Delhi I wasn't able to find any workarounds. The fella I talked to last was even cool with the whole Asterisk thing, he just didn't have anything in his bag of tricks that could help |
| 00:17.48 | atan | So now it's sitting in a box. Bloody expensive paper weight! |
| 00:17.57 | bsdice | I gambled, got a Snom 821 for home office and works very well |
| 00:18.31 | bsdice | did not try anything with provisioning though |
| 00:18.54 | WIMPy | Reminds me that I should update te Digium phones. |
| 00:20.11 | WIMPy | Lets see how long I take to find it this time. |
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| 01:20.53 | WIMPy | Hmm. Is there a way to get a GROUP_COUNT in to a hint? |
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| 02:09.05 | bengoodman | Are there any wide problems with the lastest asterisk 11 update? |
| 02:09.57 | WIMPy | wide? |
| 02:11.16 | bengoodman | I upgraded from 11.6 to 11.7...latest and I can't even dial other numbers in the same context. |
| 02:11.54 | WIMPy | Looks like you broke something. But that's almost definitely not because of the version. |
| 02:12.01 | bengoodman | haha |
| 02:12.08 | bengoodman | yum update |
| 02:12.13 | carrar | Sounds like a wide issue |
| 02:12.14 | bengoodman | hard to break something with tht |
| 02:12.17 | carrar | uninstall wide |
| 02:12.29 | bengoodman | seriously i made no changes |
| 02:12.39 | bengoodman | just a yum update |
| 02:12.53 | bengoodman | it's nuts...i did an asterisk -rvvvvvvvv |
| 02:13.24 | bengoodman | and nothing even shows up....did a sip debug on but nothing popped out at me |
| 02:13.33 | carrar | downgrade |
| 02:13.41 | bengoodman | it's really odd..i've reverted for the time being but wanted to see if others had any issues |
| 02:13.44 | bengoodman | yeah i did |
| 02:13.47 | WIMPy | What else did you upgrade? |
| 02:13.54 | bengoodman | nothing |
| 02:13.58 | bengoodman | seriously |
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| 02:14.13 | bengoodman | ok...so nothing major i see |
| 02:14.15 | bengoodman | dang |
| 02:18.17 | bengoodman | I wonder if something was changed at the sip level than snoms don't like |
| 02:19.21 | WIMPy | I'm only using 11.7-rc1, but that does work. |
| 02:19.47 | WIMPy | Obviousely not using yom, however. |
| 02:20.16 | bengoodman | It's AsteriksNow...maybe it just has something to do with that distribution. |
| 02:21.09 | bengoodman | I've had this problem twice now |
| 02:21.34 | bengoodman | Once I did a yum update and I had to copy all the sound files to a different location...I forgot exactly |
| 02:21.57 | bengoodman | and now this...I'm really starting to not trust the update process. :( |
| 02:25.08 | bsdice | yumyum :) |
| 02:25.59 | WIMPy | sings "No yum here, no cry!" |
| 02:26.25 | bsdice | hey WIMPy! |
| 02:26.37 | bsdice | no unauthorized singing in thias channel! :D |
| 02:26.41 | bsdice | this |
| 02:27.06 | WIMPy | What? No singing? |
| 02:27.23 | bsdice | unless you got a Digium license from the president :) |
| 02:27.35 | bsdice | (or Windows Vista running) |
| 02:27.44 | WIMPy | Is that installed on the phone? |
| 02:27.50 | bsdice | yes |
| 02:27.51 | WIMPy | Err |
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| 02:28.13 | WIMPy | Then I might have one. I just upgraded them. |
| 02:28.22 | bsdice | with yum? |
| 02:28.25 | WIMPy | No |
| 02:28.54 | WIMPy | Although the D70 needed a factory reset after upgrading to work propperly again. |
| 02:29.12 | bsdice | how many D70 do you have? |
| 02:29.19 | bsdice | if you don't mind me asking |
| 02:29.20 | WIMPy | 1 |
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| 02:41.22 | bengoodman | [2014-01-03 21:38:14] WARNING[1892][C-00000003]: chan_sip.c:10518 process_sdp: We are requesting SRTP for audio, but they responded without it! |
| 02:41.31 | bengoodman | I missed this before |
| 02:41.34 | bengoodman | for some reason |
| 02:42.07 | bsdice | check for transport=tls |
| 02:42.08 | bsdice | encryption=yes |
| 02:42.13 | bsdice | turn to no |
| 02:42.22 | bsdice | does your upspream SIP even support SRTP? |
| 02:42.30 | bsdice | upstream |
| 02:42.37 | bsdice | sorry, Tippgicht |
| 02:43.00 | WIMPy | Too much Gyros? |
| 02:43.33 | bsdice | no, pretty tolerant to meat without joint problems even in large doses |
| 02:43.53 | bsdice | too much sports, shower, a beer, feet on table, trouble typing |
| 02:44.56 | bengoodman | bsdice you mean on the phone itself? |
| 02:45.05 | bsdice | bengoodman no on your SIP server |
| 02:45.11 | bsdice | encryption=no |
| 02:45.34 | bsdice | uncomment any transport=tls |
| 02:46.00 | bsdice | why is your Asterisk suddently trying SRTP? that is encrypted audio |
| 02:46.06 | bsdice | needs both sides to support it |
| 02:46.49 | bsdice | for security to hold up you'd also need TLS set up with an X.509 certificate |
| 02:46.53 | WIMPy | TLS has nothing to do with SRTP. |
| 02:47.24 | bengoodman | i put encryption=no in the sip definition but still get the same message |
| 02:47.45 | bsdice | SRTP keys are sent in SDP field over TLS encrypted SIP |
| 02:47.51 | WIMPy | Did you restart? |
| 02:47.52 | bsdice | (usually) |
| 02:48.39 | bengoodman | yeah |
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| 02:51.13 | bengoodman | damn |
| 02:52.17 | bengoodman | I'm also getting provisioning server failed on the phone |
| 02:52.20 | bengoodman | man this is so odd |
| 02:57.00 | bengoodman | I have a feeling that Digium is going to get a lot of calls from folks with Snom 720s |
| 02:57.06 | bengoodman | :) |
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| 03:04.47 | bengoodman | Hmm...I turned off rtp encryption on the phone and it worked |
| 03:05.21 | bengoodman | Yeah...I really don't want to do that on 100s of phones deployed. I wonder what changed |
| 03:06.34 | WIMPy | What about SRTP Auth-tag and RTP/SAVP? |
| 03:11.04 | WIMPy | And if you have 11s of phones, don't you provision them? |
| 03:13.07 | bengoodman | No we haven't looked too much into it. We have a bunch of different types of phoens |
| 03:13.14 | bengoodman | bad on us |
| 03:13.19 | WIMPy | yes |
| 03:13.35 | bengoodman | yeah that srtp auth tag is on the client too |
| 03:13.38 | bengoodman | rtp/savp |
| 03:13.43 | bengoodman | i wonder what changed |
| 03:14.08 | WIMPy | hasn't tried encryption for quite some time. |
| 03:14.24 | bengoodman | it feels like i could add a setting to sip.conf to make it work |
| 03:14.36 | bengoodman | well that's the odd part, the server wasn't doing srtp before |
| 03:15.22 | WIMPy | Well, the message said your server wanted it. |
| 03:15.43 | bengoodman | isn't that odd though |
| 03:15.49 | bengoodman | say if the phone has it turned off |
| 03:15.53 | bengoodman | and it works |
| 03:16.13 | WIMPy | Might be the way it rejects it. |
| 03:16.26 | bengoodman | yes |
| 03:16.38 | bengoodman | So if I set rtp encryption off on the phone it works |
| 03:16.49 | bengoodman | If I set rtp encryption on it throws that error |
| 03:17.05 | bengoodman | is there an encryption=negotiage |
| 03:17.08 | bengoodman | te |
| 03:17.32 | WIMPy | The snom can do it as option. |
| 03:58.39 | bengoodman | if (!secure_audio && p->srtp) { |
| 03:58.39 | bengoodman | 10518 ast_log(LOG_WARNING, "We are requesting SRTP for audio, but they responded without it!\n"); |
| 03:58.39 | bengoodman | 10519 res = -1; |
| 03:58.39 | bengoodman | 10520 goto process_sdp_cleanup; |
| 03:58.39 | bengoodman | 10521 } |
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| 04:40.51 | atan | Anyone aware of similar providers to cidname? |
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| 05:02.38 | overyander | are AMI connection/authentication attempts logged in the security log? |
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| 10:40.38 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.7.0 (2013/12/17), 1.8.25.0 (2013/12/17); Standard: Asterisk 12.0.0 (2013/12/20); DAHDI: DAHDI-linux 2.8.0.1 (2013/12/19), DAHDI-tools 2.8.0 (2013/12/06); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 13:07.02 | jks | I have a problem with Asterisk 11.7.0 crashing in pj_timer_heap_schedule() during ICE negotiation (it seems) when using webrtc... is this a known bug? |
| 13:08.59 | file | https://issues.asterisk.org/jira/browse/ASTERISK-20762 |
| 13:10.00 | jks | file, thanks, seems like the same hting |
| 13:11.01 | jks | although I can make it crash every time, and I do not have to make two calls... only have to accept a call in the webrtc client |
| 13:12.34 | jks | but it only happens with multi-homed clients |
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| 13:34.22 | powerunits | hi all .. i need some help on tls configration |
| 13:34.42 | powerunits | i have configure tls on asterisk but when i try to register my softphone |
| 13:34.50 | powerunits | using tls |
| 13:34.58 | powerunits | i get error message on asterisk CLI |
| 13:35.07 | powerunits | Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) |
| 13:35.13 | powerunits | tcptls.c:235 handle_tcptls_connection: FILE * open failed! |
| 13:35.29 | powerunits | please any help on this? |
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| 13:42.47 | powerunits | ? |
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| 15:45.08 | admin0 | hey .. any a2billing uses/experts around |
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| 17:52.21 | jks | any ideas on speeding up ICE in Chrome when using WebRTC with Asterisk? (currently takes ~5 seconds before a call can be made... using local STUN/TURN server) |
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