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| 00:20.53 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); 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 |
| 01:31.37 | igcewieling | Sorry, I'm allergic to queues. |
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| 08:24.56 | dnit | Hi |
| 08:29.16 | dnit | Hi how can I bridge an agent with channel after the call has been initiated and the channels have been created, but before the channel starts ringing. |
| 08:29.46 | dnit | So that the agent can hear all the early media which the outgoing channel has |
| 09:01.54 | dnit | Hello |
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| 14:05.57 | *** topic/#asterisk by gtjoseph -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.17.0 (2017/07/12), 11.25.1 (2016/12/08), Standard: 14.6.0 (2017/07/12); 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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| 14:28.37 | darkunderlord | hey all |
| 14:28.57 | eric_hill | hey |
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| 14:51.26 | imcdona | Anyone know when we might see https://gerrit.asterisk.org/#/c/5857/ released in a certified Asterisk 13? I'm assuming it will be included in cert 5? Any idea on when that is scheduled to be released? |
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| 15:21.16 | dnit | Hi , is there a way to brige a channel which is has just been dialed and not yet answered with an agent ? |
| 15:22.13 | dnit | After Dial is executed |
| 15:23.33 | rrittgarn | you could use a call file to originate a call into a queue that is outbound/ringing (not sure if that answers what you're looking to accomplish) |
| 15:25.12 | igcewieling | Sorry, I'm allergic to queues. |
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| 15:31.18 | dnit | I am using AMI interface for originating the calls |
| 15:32.44 | dnit | What I am looking for is once the Dial is executed and asterisk sends the dialing event , then I want to assign an agent and make him hear all the early media which is there on the outbound channel |
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| 15:34.09 | dnit | I tried with Confbridge , agentrequest |
| 15:34.38 | dnit | but using these hangs up the outbound channel which I think is happening because it gets answere ? |
| 15:37.50 | bluez_ | can extensions be setup to do things when we receive progress and ringing? |
| 15:38.16 | bluez_ | eg dial(ext1), on progress dial(ext2), on ringing hangup |
| 15:38.20 | [TK]D-Fender | <dnit> I tried with Confbridge , agentrequest <- none of these hook into another call. |
| 15:38.42 | [TK]D-Fender | BRIDGE() <- |
| 15:38.44 | [TK]D-Fender | that hook in |
| 15:39.01 | [TK]D-Fender | bluez_, Nope. |
| 15:39.13 | bluez_ | ok |
| 15:42.19 | [TK]D-Fender | <imcdona> Anyone know when we might see https://gerrit.asterisk.org/#/c/5857/ released in a certified Asterisk 13? I'm assuming it will be included in cert 5? Any idea on when that is scheduled to be released? <- are you paying Digium for support? |
| 15:43.13 | igcewieling | "Zombie famine in Washington DC -- no brains found for miles around." |
| 15:48.35 | igcewieling | Does anyone know if func_hash does anything special other than setting channel variable with the ~HASH prefix? I need to emulte what func_hash is doing. |
| 15:49.35 | igcewieling | i.e. ~HASH~sm_call[1]~route_name~=VMAX (DIOD) |
| 16:06.35 | dnit | [TK]D-Fender: I tried with bridge too it hangs up both my agent channel and outbound channel. Will look into it though. |
| 16:11.26 | igcewieling | apparently the answer is "yes" |
| 16:15.40 | file | imcdona: as [TK]D-Fender sorta mentioned - certified only receives fixes as a result of commercial agreements, it would only appear in the next major certified release |
| 16:15.44 | file | whatever that may be |
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| 17:21.27 | td34 | hey all, has anyone had any experience with cisco phones when using asterisk SIP server? |
| 17:21.42 | salviadud | td34, I have cisco phones working right now |
| 17:22.08 | salviadud | I recommend you take a look at usecallmanager.nz |
| 17:22.44 | salviadud | You could make them work with sccp, but limited functionality |
| 17:23.39 | td34 | oh id rather do SIP |
| 17:23.53 | td34 | I've got a DX80 phone |
| 17:24.06 | td34 | which I want to use to communicate video with other linphone devices |
| 17:24.16 | td34 | but first thing is first, i'm trying to get it to register lol |
| 17:27.03 | td34 | have you got any experiencce with the DX80? I'm guessing it's a pretty similar experience to the phones... salviadud |
| 17:29.59 | salviadud | td34, I don't have a DX80. If you still have the xml file generated by CUCM, you can use that to edit a few settings and make it register with asterisk. |
| 17:30.14 | td34 | I do not have the xml file |
| 17:30.29 | td34 | however... there was one posted on another site |
| 17:30.32 | td34 | so i'll use that |
| 17:30.47 | salviadud | check out that site I posted. |
| 17:30.52 | salviadud | It should help |
| 17:31.05 | td34 | well.. it's just a patch for asterisk right |
| 17:31.10 | td34 | that doesn't help me register the device? |
| 17:31.15 | salviadud | You would need to patch asterisk so that chan_sip does stuff it usually doesn't do natively |
| 17:31.26 | salviadud | It's a patch for chan_sip to be precise. |
| 17:31.41 | td34 | and the patch is 100% nessesary? |
| 17:31.51 | salviadud | No |
| 17:32.14 | salviadud | The site has info on the xml files as well, so it's worth a look. |
| 17:32.54 | salviadud | Patch helps for native conferencing and some other things like call forwarding. |
| 17:33.00 | td34 | ahh nah thats ok |
| 17:33.00 | td34 | thanks |
| 17:33.05 | td34 | i'll just use the xml file from the other site |
| 17:33.07 | salviadud | But from what I see, that DX80 doesn't even look like a phone. |
| 17:33.17 | td34 | although good to read, thanks - looks like I have an interesting day ahead of me. |
| 17:33.24 | td34 | it's a teleconference thingy |
| 17:34.01 | td34 | https://support.cafex.com/hc/en-us/articles/201620801-How-to-Run-Cisco-DX80-DX650-against-Trial-Environment-running-on-Mac |
| 17:34.07 | salviadud | You'll probably need to enable tcp on sip |
| 17:34.22 | salviadud | and max out your bandwidth on that device. |
| 17:34.47 | salviadud | I have 9971's and I cannot get video to work, but it's not really that important. |
| 17:35.58 | salviadud | Good luck with that, come back and comment progress. |
| 17:38.09 | td34 | ok will do |
| 17:38.12 | td34 | why can't you get video to work? |
| 17:39.53 | salviadud | I'm not entirely sure. I know I enabled it on the global section of sip. |
| 17:40.09 | td34 | this is gonna take a while |
| 17:40.11 | td34 | i can feel it |
| 17:40.17 | salviadud | I read that if I do patch chan_sip that video will not work when I enable the use_callmanager flag |
| 17:40.29 | td34 | fair nuff |
| 17:40.42 | salviadud | So, I lose video capability if I want to do native conferencing. |
| 17:41.01 | td34 | lmao |
| 17:41.06 | td34 | i only need video capability |
| 17:41.08 | salviadud | So, I didn't look to much into it, because we don't have clients that are out of the office. |
| 17:41.21 | td34 | this is only for testing nayway |
| 17:41.21 | salviadud | Well, for starters, the patch is not necessary for you. |
| 17:41.32 | td34 | if i was going to use this in production |
| 17:41.36 | td34 | i would demand cucm lll |
| 17:41.37 | td34 | lol* |
| 17:42.06 | salviadud | I would never pay for such piracy. |
| 17:42.24 | salviadud | Asterisk > cucm |
| 17:42.56 | salviadud | I know for a fact that back in 2007, cisco would test sip firmware with asterisk. |
| 17:43.22 | salviadud | I don't know about now though. |
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| 17:52.40 | td34 | it's not really in their interest |
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| 18:18.58 | igcewieling | hugs his Polycom phones |
| 18:20.12 | td34 | lol |
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| 18:46.45 | salviadud | igcewieling, can you get native conferencing on polycom? |
| 18:52.32 | igcewieling | salviadud: Why wouldn't it? |
| 18:53.03 | salviadud | Oddly enough, I am sort of a polycom partner, but we don't own a single phone. |
| 18:53.15 | salviadud | It's probably because my boss is a cheap sob |
| 18:53.25 | salviadud | And he doesn't want to invest in new phones. |
| 18:53.39 | salviadud | So, I've never really used a polycom firsthand. |
| 18:53.57 | igcewieling | I've not looked at N-way conferencing in years, but people use 3-way conferencing all the time. |
| 18:54.44 | igcewieling | If you are referring to phones who don't support more than 1 g729 call, that is, as far as I know, only something Cisco does. |
| 18:55.53 | igcewieling | We must have about 1,200 Polycom phones across 200 or so clients. |
| 18:56.07 | salviadud | igcewieling, what I mean is this: Does polycom have a "button" of sort that allows you to call another extension while placing the first call on hold so you can later join those calls through confbridge without actually opening up a room? |
| 18:56.10 | igcewieling | 72+ FreePBX boxes too. |
| 18:56.46 | igcewieling | salviadud: Some have hard keys, some have softkeys, but they all have a conference button |
| 18:57.16 | igcewieling | I can't imagine a company so brain dead they don't support a conference button on the hardphone. |
| 18:57.47 | salviadud | haha |
| 18:58.02 | igcewieling | links to phone quickstart guides and user guides: http://help.nyigc.net/ |
| 18:58.06 | salviadud | I had a hard time getting the ciscos to do that. |
| 18:58.38 | salviadud | who is interglobe com? |
| 18:58.57 | igcewieling | salviadud: my employer. |
| 18:59.45 | igcewieling | salviadud: assume Cisco 79xx types of phones are unlike any other phones. |
| 19:00.39 | salviadud | igcewieling, it's a cisco thing... You patch it and it works, I just had a hard time finding that. |
| 19:00.53 | salviadud | Still, the patch is third-party, I doubt cisco is happy about it |
| 19:01.05 | salviadud | They get to sell less cucms because of that. |
| 19:02.11 | salviadud | I remember I had a conversation with someone that was a CCNP, and he said that they lost a lot of market, huawei making switches, the fact that they got into Voip, and that the phones are cool but the licensing for cucm is above average. |
| 19:02.23 | salviadud | Well, not only huawei. |
| 19:02.46 | salviadud | Anyways, cisco has found that a lot of ppl have gone to other vendors for ALL of their solutions. |
| 19:03.09 | salviadud | And all they got now, is bragging rights for protocols. |
| 19:03.22 | igcewieling | salviadud: the Cisco Linksys (or SPA series) are the "SIP" phones. |
| 19:03.27 | salviadud | but, IMO, cisco routers/switches are good quality. |
| 19:03.55 | salviadud | igcewieling, 79xx series and above have SIP firmware. |
| 19:04.05 | igcewieling | salviadud: that firmware is crap. |
| 19:04.18 | salviadud | knows that firsthand. |
| 19:04.32 | igcewieling | They added SIP as an afterthought. The SPA series were designed for open SIP, not SCCP. |
| 19:04.37 | salviadud | Like I said, for SIP to work properly, chan_sip needs to get patched. |
| 19:04.47 | salviadud | But some models only do SIP |
| 19:05.06 | salviadud | 9971 and 89xx series phones don't do sccp |
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| 20:15.55 | td34 | hey |
| 20:16.01 | td34 | is the chap still here who uses asterisk with their cisco phones? |
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| 21:33.08 | panatrator69 | so I'm having a client complain about excessive background noise in a confbridge to the point where they're saying that other callers that are physically in the same room and tlaking are heard over the client. I've been looking into implementing the denoise option from the confbridge user profile or using the DENOISE dialplan function, which all required speex. Installing speex wouldn't be a hassle, but |
| 21:33.11 | panatrator69 | I've heard that it's deprecated and Opus has kinda taken it's place. Do you guys have any other recommendations on background noise reduction? Or maybe even setting a volume silence threshold on SIP channels in a confbridge? Since we drop clients and callers together into a confbridge, we've tweaked the dsp_silence and dsp_talking threshold values to try to remedy their complaints, but that seems to only |
| 21:33.13 | panatrator69 | deal with time duration constraints for mixing a channel's audio into the confbridge rather than setting the gain or volume threshold for what's considered silence. Suggeestions? |
| 21:35.08 | lorsungcu | panatrator69: the issue is more translation |
| 21:35.15 | lorsungcu | i dont trhink there's paths for speex > anything |
| 21:35.33 | lorsungcu | but |
| 21:35.46 | lorsungcu | a better mic would help more than anything. what device are you using |
| 21:36.58 | panatrator69 | I'm actually not sure. It's for clients with agents in a call center so it could be a toss up. I'll have to get in contacts with their point of contact. |
| 21:37.10 | panatrator69 | I know addressing the issue at the hardware would probably be the proper solution |
| 21:37.20 | panatrator69 | but we're receiving it from multiple clients across multiple locations |
| 21:37.34 | panatrator69 | multiple hardware configurations, etc. |
| 21:37.46 | lorsungcu | you arent sure what the hardware config is |
| 21:38.18 | lorsungcu | getting that corrected would be the most effective use of time |
| 21:38.31 | lorsungcu | if it's still bad, go look for a magic bullet in some codec |
| 21:39.21 | panatrator69 | gotcha, thanks for the help |
| 21:39.43 | panatrator69 | also do you mind clarifying what you meant when you said that there's any paths for speex > anything? |
| 21:40.14 | lorsungcu | asterisk can't transcode speex to anything |
| 21:40.28 | panatrator69 | ohhh codec translations |
| 21:40.31 | panatrator69 | gotcha |
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| 21:56.49 | panatrator69 | exit |
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