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| 03:59.22 | Bonte | I'm trying to create a dynamic conference bridge that rings a line internally and when that line is picked up, adds a third party to the call. |
| 03:59.50 | Bonte | So part A calls in, party B's phone rings, when party B answers, party C is called and then they're all put into a confbridge. |
| 04:01.04 | Bonte | Using asterisk manager or .call files, I'm able to place the call to party B and bring them into the bridge, but I can't find a way to determine if party B answered to bring party A into the conference. |
| 04:01.41 | Bonte | Any thoughts? |
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| 10:54.30 | Joonake | any ideas if this is a feature or a bug and if it is fixed in latest versions: https://haste.joonake.fi/uwukoqafow ? |
| 10:55.28 | Joonake | asterisk does not increment the session version id on SDP o= line even if SDP body has changed |
| 10:56.47 | Joonake | should it when only number of offered codecs is changed? |
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| 12:10.07 | HenrikJott | Hi! I trying to get "Attended call transfer" to work and when i use the Dial-command i can pass the options "tTkK" to be able to transfer and park the call, but when i use call files i cannot get it to work. Does anyone know how i can pass those options from a call file? |
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| 12:12.18 | HenrikJott | Maybe i should mention that i use Asterisk 12 and the call im trying to transfer is in a queue, connected to an agent. |
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| 12:23.09 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 12:25.01 | [TK]D-Fender | HenrikJott, * 12 is no longer supported |
| 12:25.50 | [TK]D-Fender | Next, what phone is the transferer using? |
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| 12:37.32 | HenrikJott | [TK]D-Fender: Sorry, typo, i am using asterisk 13! The phone is sipml5 (javascript loaded in browser) |
| 12:41.16 | HenrikJott | [TK]D-Fender: I can see that the dtmf gets correctly signalled to asterisk but nothing happens. When i tried the same thing with the Dial command instead i got it working when i passed "tTkK" as options. That was on a X-lite softphone though... |
| 12:41.43 | [TK]D-Fender | You should make sure your client has REAL SIP transfer capabilities. |
| 12:42.07 | [TK]D-Fender | DTMF should only be used when calling over the PSTN or another interface where it isn't possible to have native transfers, etc. |
| 12:42.25 | [TK]D-Fender | You alternative for this case might be to use a Local channel to dial the Channel: |
| 12:42.32 | [TK]D-Fender | and dial that with the options |
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| 12:48.04 | HenrikJott | [TK]D-Fender: Ok, you mean that I shouldn´t involve asterisk in the transfer? (i realize im on thin ice here).. What we tried is to use the asterisk built in functionality for attended call transfer but as agents wait in the queue and we auto dial out i think we need to add those options in the call file. I´ve tried to add "Data: tTkK" to the call file but it does not seem to make any |
| 12:48.05 | HenrikJott | difference. |
| 12:52.45 | Dovid | morning |
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| 13:16.12 | [TK]D-Fender | DTMF transfers = ass |
| 13:16.30 | [TK]D-Fender | I told you what to do. Dial a LOCAL channel |
| 13:16.35 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 13:16.45 | [TK]D-Fender | and pass Dial your options |
| 13:19.55 | Samot | What is a "native transfer" and why can't DTMF be used for it? |
| 13:23.58 | [TK]D-Fender | SIP transfer inside a SIP client. Like it should be |
| 13:24.49 | Samot | And how do things on "one-touch" get done? |
| 13:25.07 | [TK]D-Fender | as in? |
| 13:25.08 | Samot | One-touch parking, one-touch recording, one-touch anything... |
| 13:25.18 | [TK]D-Fender | that's clearly different |
| 13:25.23 | Samot | How? |
| 13:25.23 | [TK]D-Fender | that isn't a native SIP thing |
| 13:25.28 | Samot | BS. |
| 13:25.53 | [TK]D-Fender | Where is "record" in the SIP protocol? |
| 13:25.58 | [TK]D-Fender | I don't think you're following the context |
| 13:26.17 | Samot | Sure. |
| 13:26.21 | file | there is a Snomism for one-touch recording |
| 13:26.21 | Samot | I have a button.. |
| 13:26.31 | Samot | It does *271 |
| 13:26.37 | Samot | That drops it in a parking log |
| 13:26.38 | [TK]D-Fender | the WAY he's calling that channel doesn't tell * to accept DTMF call control features. THAT is what I'm saying he shouldn't need. |
| 13:26.42 | Samot | That drops it in a parking lot |
| 13:26.48 | [TK]D-Fender | the fact your phone is sparter doens't make the action singular |
| 13:26.51 | Samot | Sure it does |
| 13:27.00 | Samot | When you add the Dial() options. |
| 13:27.09 | [TK]D-Fender | sure and you can tell your wife "make me a cake". |
| 13:27.13 | [TK]D-Fender | I guess that's a 1-step cake |
| 13:27.22 | [TK]D-Fender | Whatever |
| 13:27.41 | [TK]D-Fender | Still not what we're actually trying to resolve here |
| 13:27.54 | Samot | Well... |
| 13:28.05 | Samot | If you're going to suggest stuff.. |
| 13:28.15 | Samot | Don't sit there and say DTMF can't be used "natively" |
| 13:28.18 | Samot | Because that is wrong. |
| 13:28.23 | [TK]D-Fender | I didn't say "can't" |
| 13:28.30 | [TK]D-Fender | So actually read what I said |
| 13:28.31 | Samot | No you're right |
| 13:28.36 | Samot | Just that you shouldn't. |
| 13:28.54 | [TK]D-Fender | If you have a SIP device let it it ITS thing |
| 13:29.08 | [TK]D-Fender | You don't use a highly functional thing and then drive it like a chump |
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| 13:38.22 | Samot | Well as far as I can tell.. |
| 13:38.28 | Samot | *2 puts the call on hold.. |
| 13:38.32 | Samot | Asterisk "holds" it |
| 13:38.39 | Samot | You dial where you want it to go.. |
| 13:39.08 | Samot | And that is connected and you hangup, it bridges the held channel into the newly answered channel and dumps your channel. |
| 13:39.47 | Samot | The "SIP" way just puts the call on hold, dials the new calls as a new call... |
| 13:40.01 | Samot | And when that transfer button is hit a second time..it sends a REFER |
| 13:40.16 | Samot | To tell the system, "Hey, bridge these other two channels together" |
| 13:40.32 | Samot | So one way has Asterisk do it all internally |
| 13:40.42 | Samot | The other requires the device to send additional SIP messages. |
| 13:41.39 | *** join/#asterisk u0m3_ (~u0m3@86.120.99.58) |
| 13:41.41 | Samot | Neither of those two processes are dependent or work better/worse over the PSTN vs "native/local" devices. |
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| 16:35.20 | jamesaxl | hi |
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| 16:36.20 | jamesaxl | sometimes when I run queue show my_queue, I see that number of calls become to 0 , is it a configuration issue ? |
| 16:36.56 | [TK]D-Fender | No, those stats reset every now & again |
| 16:37.08 | [TK]D-Fender | Don't remember the trigger or terms |
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| 16:46.33 | jamesaxl | [TK]D-Fender: because i create a real-time application for monitoring a small call center and i got this issue |
| 16:46.59 | jamesaxl | maybe I will took info from database queue_log |
| 16:47.08 | jamesaxl | s/took/take* |
| 16:47.37 | Samot | well logs do rotate |
| 16:48.45 | jamesaxl | Samot: is it a safe solution or not ? |
| 16:49.16 | Samot | Well most queue reporting software I've seen for Asterisk, reads from the log and stores the information in a backend database. |
| 16:49.35 | Samot | So it's not trying to parse an active log file for data |
| 16:50.05 | Samot | Or having to keep that log file from rotating or becoming enormously huge |
| 16:51.34 | jamesaxl | Samot: in my case, I think that i am smart and I parse the output of asterisk CLI :D |
| 16:52.01 | Samot | Why would you do that? |
| 16:52.19 | Samot | I mean the CLI is just showing you what is logged. |
| 16:52.41 | jamesaxl | Samot: because I don't have experience, but I am going to take the info directly from database |
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| 17:10.20 | [TK]D-Fender | i disabled queue log rotation on my server |
| 17:10.25 | [TK]D-Fender | and you can also log direct to SQL |
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| 18:05.31 | twitchnln | morning, trying to setup remote queue agent under asterisk 11.25 am able to get logged in and send queue calls to remote agent, but queue does not seem to respect 'skip busy agent' and I see the following in CLI when agent is on call: 1000 has 0 calls (max unlimited) in 'ringall' strategy (7s holdtime, 25s talktime), W:0, C:6, A:4, SL:100.0% within 60s |
| 18:05.31 | twitchnln | <PROTECTED> |
| 18:05.31 | twitchnln | <PROTECTED> |
| 18:05.32 | twitchnln | <PROTECTED> |
| 18:07.22 | roswell | twitchnln, do you have hints enabled for queue members? |
| 18:08.16 | twitchnln | roswell, how do you enable hints on local channel? |
| 18:08.52 | roswell | twitchnln, interesting. how do you have queue members defined? |
| 18:10.54 | *** join/#asterisk Dovid (~dovid@67.231.202.3) |
| 18:11.11 | twitchnln | roswell, https://pastebin.com/HstUpmBc |
| 18:11.57 | [TK]D-Fender | (Not in use) <- doesn't look BUSY to me.... |
| 18:12.32 | twitchnln | I know, that's the problem, the status was run while agent was on call |
| 18:13.14 | [TK]D-Fender | Outside shit doesn't HAVE a status |
| 18:13.23 | [TK]D-Fender | There is no presence for a "PSTN phone #" |
| 18:14.10 | [TK]D-Fender | And tehre was no queue call going on at the time of your paste there |
| 18:14.27 | Samot | Wait..wait wait |
| 18:14.32 | Samot | "Remote" agent |
| 18:14.45 | Samot | You're assuming Asterisk can tell the state of a mobile phone? |
| 18:14.59 | Samot | It knows if the agent is on a call |
| 18:15.10 | Samot | But doesn't know about CF, CW |
| 18:15.29 | Samot | How many calls marks that device as "BUSY" |
| 18:15.30 | twitchnln | No, I'm expecting asterisk to remember it called the mobile and started the queue call, and not to send another until first is finished. |
| 18:15.44 | [TK]D-Fender | I don't see a queue call in progress there at all |
| 18:16.04 | [TK]D-Fender | Show me otherwise |
| 18:18.57 | Samot | Plus.. |
| 18:19.03 | Samot | Local is a 2-part channel. |
| 18:19.19 | Samot | You have to make sure Asterisk is watching for the 2 leg of the channel to be in use |
| 18:19.23 | Samot | Not the first. |
| 18:19.34 | Samot | Because as soon as that call bridges, the first channel is gone. |
| 18:19.50 | Samot | So it doesn't exist anymore ergo it is "not in use" |
| 18:21.00 | twitchnln | D-Fender/Samot, Here is trace with call up and queue showing nothing. |
| 18:21.02 | twitchnln | https://pastebin.com/cyfd5X6q |
| 18:21.35 | Samot | And what was the state of the agent during the call? |
| 18:21.39 | Samot | When it was connected? |
| 18:22.03 | twitchnln | in call, but second call still came in |
| 18:22.27 | Samot | <PROTECTED> |
| 18:22.39 | Samot | So which is being monitored for follow up calls? |
| 18:23.00 | Samot | The status that shows "in call" or the device/extension status that shows it not in use? |
| 18:23.19 | [TK]D-Fender | <PROTECTED> |
| 18:23.22 | Samot | You need to make sure Asterisk is remember the right thing. |
| 18:23.26 | [TK]D-Fender | -- Executing [h@macro-dialout-trunk:1] Macro("Local/4443335552@from-internal-00000010;2", "hangupcall,") in new stack |
| 18:23.30 | [TK]D-Fender | I see a HANGUP right after |
| 18:23.38 | Samot | Right |
| 18:23.47 | Samot | One of the Local channels hangsup |
| 18:23.50 | Samot | Like it should. |
| 18:23.50 | [TK]D-Fender | <PROTECTED> |
| 18:23.53 | [TK]D-Fender | EXTRA dea |
| 18:23.55 | [TK]D-Fender | dead |
| 18:24.04 | [TK]D-Fender | <PROTECTED> |
| 18:24.13 | Samot | What about the OTHER LEG? |
| 18:24.35 | [TK]D-Fender | Think we're going to see a channel listing? |
| 18:24.40 | *** join/#asterisk u0m3_ (~u0m3@86.120.99.58) |
| 18:24.49 | Samot | My point. |
| 18:25.12 | Samot | Is the exception based on watching the wrong status/channel-leg? |
| 18:25.24 | twitchnln | so I should be monitoring IAX2/AgudathTestElastix-4839 |
| 18:25.27 | Samot | One status shows in call the other shows the device not in use |
| 18:25.40 | twitchnln | which is the 2nd leg of call |
| 18:26.00 | Samot | So |
| 18:26.01 | Samot | Sure |
| 18:26.06 | Samot | Or the "in call" status |
| 18:26.13 | Samot | Just not anything related to the device |
| 18:26.25 | Samot | Since it will never have a status other than "not in use" |
| 18:26.40 | Samot | Unless *you* set it yourself based on actions. |
| 18:26.41 | twitchnln | How would I monitor the "in call" status, that sounds easier than a constantly changing 2nd leg |
| 18:28.55 | Samot | state_interface |
| 18:29.20 | Samot | Instead of it being something like SIP/<device> or IAX/<device> |
| 18:29.47 | Samot | You use Agent:1000 |
| 18:30.00 | *** part/#asterisk roswell (roswell@roswell.systems) |
| 18:30.46 | Samot | I don't generally deal with remote agents... |
| 18:31.12 | Samot | So the rare times I do, I just have to tweak things until they play how I want |
| 18:31.30 | Samot | Or compromise to how Asterisk wants it |
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| 19:39.07 | *** join/#asterisk skies (67ff044c@gateway/web/freenode/ip.103.255.4.76) |
| 19:40.01 | skies | hello |
| 19:40.14 | skies | please i need some help .. |
| 19:40.33 | skies | can i register multiple soft phone on 1 SIP extension? |
| 19:41.04 | skies | and what i want when there incoming call on that SIP extension it should ring on all soft phone |
| 19:41.07 | skies | and what i want when there incoming call on that SIP extension it should ring on all soft phones |
| 19:42.41 | roswell | skies, https://wiki.asterisk.org/wiki/display/AST/SLA+Trunk+Configuration |
| 19:45.00 | skies | Thanks let me check. |
| 19:45.41 | thiagoc | skies, why not use different extensions and a queue with ringall strategy? |
| 19:46.38 | skies | @thiagoc, reason is becuase same extension.. i want to configure on mobile and softphone and hardphone. |
| 19:54.17 | [TK]D-Fender | SLA is a lie |
| 19:54.45 | [TK]D-Fender | and has nothing to do with this |
| 19:54.56 | [TK]D-Fender | (as * calls it) |
| 19:55.01 | [TK]D-Fender | chan_sip cannot do this |
| 19:55.05 | [TK]D-Fender | PJSIP can |
| 19:57.33 | skies | https://wiki.asterisk.org/wiki/display/AST/SLA+Trunk+Configuration -> i have read this but this is using trunk. what if there is 100 make a to 200. and 200 is register on mulitple phone then how would we handle it |
| 19:59.42 | roswell | skies, https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=4816923 |
| 20:00.00 | skies | thanks let me check this |
| 20:00.19 | [TK]D-Fender | SLA has NOTHING to do with this |
| 20:00.47 | [TK]D-Fender | You are wasting your time |
| 20:01.12 | [TK]D-Fender | <skies> can i register multiple soft phone on 1 SIP extension? -----> PJSIP ONLY <----- |
| 20:02.58 | skies | @TK]D-Fender, please can you give any exmple links which i can follow? |
| 20:03.10 | skies | using PJSIP on asterisk? |
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| 21:00.45 | wyoung | Service Level Agreement Trunk Configuration? |
| 21:08.44 | rmudgett | Shared Line Appearance |
| 21:12.56 | wyoung | Ah, old fashion PBX tech |
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| 22:20.33 | _Maik_ | why the hack is yealink dailing to extension 1 when it recives a BLF NOTIFY |
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