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| 00:19.52 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.21.0 (2018/05/01), Standard: 15.4.0 (2018/05/01); 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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| 04:45.40 | FarhaadN | Hi everyone, is there any command for all phones to be re-registered? |
| 04:52.52 | [TK]D-Fender | no |
| 04:52.58 | [TK]D-Fender | You can't make a phone register |
| 04:54.37 | FarhaadN | D-Fender: are you sure? |
| 04:55.41 | FarhaadN | becouse cisco phones had this options |
| 04:56.57 | FarhaadN | i want when phone is register |
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| 05:14.42 | drmessano | You can't make an endpoint register |
| 05:14.46 | drmessano | There is no option for that |
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| 05:29.57 | FarhaadN | drmessano: thank you |
| 05:30.05 | FarhaadN | [TK]D-Fender: thank you |
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| 08:11.25 | pawiecki | I have both: hint and BLF configured, but 'core show hints' shows zero watchers. Is that an error? |
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| 09:15.47 | pawiecki | with sip debug I've found that my phone is sending incorrect SUBSCRIBE requests, different to what is setup in the BLF field. |
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| 16:54.57 | beight0 | When I set up a user in voicemail.conf and do "voicemail show users" it shows NewMsg count. If I set up users in SQL it and run "voicemail show users for default" there is no NewMsg count. Is this correct behavior? |
| 17:07.18 | *** join/#asterisk Toerkeium (be0237fa@gateway/web/freenode/ip.190.2.55.250) |
| 17:07.21 | Toerkeium | hello guys |
| 17:07.23 | Toerkeium | does anyone know how can I find who provides phone numbers starting with 5984 in my country? |
| 17:16.45 | igcewieling | what is your country? |
| 17:20.19 | Toerkeium | argentina |
| 17:21.03 | igcewieling | See http://www.itu.int/oth/T0202000009/en it might help. |
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| 17:26.15 | beight0 | To answer my own question, yes that is the correct behavior. |
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| 18:32.35 | Nivex | I realize this is not asterisk specific, but hoping someone here can point me in the right general direction: I have a Polycom VVX 600 with latest firmware. If you send it a SIP text message, it hangs up 30 seconds later. |
| 18:40.27 | Samot | It shouldn't do that. |
| 18:41.39 | Samot | A SIP MESSAGE is not handled the same was as a SIP INVITE. If the phone is ringing and hanging up. It thinks it has a call. |
| 18:41.40 | igcewieling | Nivex: what EXACTLY is sent as the "text message" |
| 18:41.42 | Samot | Not a text. |
| 18:42.04 | Nivex | Samot: well I know it _shouldn't_ do that, but it does. |
| 18:42.12 | Nivex | igcewieling: hang on lemme pull a trace |
| 18:44.51 | Nivex | https://p.6core.net/p/F7JOBS2rlVtedjemgJ2i9BIH |
| 18:45.20 | Nivex | sadly the debug doesn't have timestamps, but I watch the timer on the phone display and the screen |
| 18:45.34 | Nivex | you can see the BYE originating from the Polycom though |
| 18:46.25 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 18:46.57 | Samot | Does the message appear on the polycom? |
| 18:47.02 | Nivex | so yes, it's Polycom's fault. An acceptable workaround would be to inhibit MESSAGE frames getting sent to the peer |
| 18:47.04 | *** join/#asterisk Hazel-rah (~Hazel-rah@190-77-134-42.dyn.dsl.cantv.net) |
| 18:47.04 | Nivex | Samot: it does not |
| 18:47.15 | Samot | Do you have it enabled? |
| 18:47.29 | Samot | It's not something that is enabled via the GUI. |
| 18:47.34 | Samot | It's a config setting. |
| 18:47.43 | Samot | You have to tell the device to accept SMS/Text |
| 18:48.26 | Samot | The polycom is sending back 200 OK's to the MESSAGE requests. |
| 18:48.38 | Nivex | Probably not then since I basically used the Simple Setup to load creds and that's about it |
| 18:48.55 | igcewieling | you might consider using XMPP and the Polycom's IM features. |
| 18:49.00 | Samot | Yeah, it's an option that has to be enabled. |
| 18:49.09 | Samot | to use IM/SMS |
| 18:49.19 | Nivex | I don't even care if it doesn't display the message. I just want it to not hang up. |
| 18:49.47 | igcewieling | read up on PUSH messages in the Admin guide. |
| 18:51.17 | Nivex | ok, will do |
| 18:51.27 | Samot | feature.messaging.enabled="1" |
| 18:51.37 | Samot | ^^ That's needs to be enabled on the polycom |
| 18:52.08 | Samot | And MessageSend() to send it from Asterisk works just fine. |
| 18:52.28 | igcewieling | I thought that controlled XMPP messaging |
| 18:52.31 | Hazel-rah | Good day everyone! i'm looking for a way to listen an active call in a browser; i kwon about Chanspy and Extenspy but i need an alternative. Thanks and cheers! |
| 18:52.38 | Samot | It does both |
| 18:52.47 | Samot | SIP: or XMPP: |
| 18:53.16 | Samot | Well PJSIP for that driver as well |
| 18:54.33 | Samot | Hazel-rah: There isn't an alternative. |
| 18:54.42 | Samot | That's why they exist. |
| 18:55.03 | igcewieling | I found users didn't like texting on their Polycoms so we never deployed it |
| 18:55.14 | Samot | I don't use it for sending |
| 18:55.18 | Samot | I use it for alerts. |
| 18:55.34 | Samot | So they are mainly just receiving. |
| 18:55.36 | igcewieling | Hazel-rah: there are a number of applications which capture SIP/RTP calls |
| 18:55.42 | igcewieling | What do you tell your users? |
| 18:55.54 | Samot | me? |
| 18:55.58 | igcewieling | yeah. |
| 18:56.17 | Samot | Well I do it on both Polycom and Yealinks. |
| 18:56.29 | Samot | For hotels and offices it's used for 911 alerts. |
| 18:56.53 | Samot | Some use it for different notices. |
| 18:57.14 | Nivex | alrighty! Got the feature enabled. It displayed the message. ... and then it still hung up 30 sec later :( |
| 18:57.31 | igcewieling | I'd use text messaging to a cellphone for that sort of thing, but I see your point. |
| 18:57.35 | Samot | Nivex: Show the dialplan you are using. |
| 18:57.59 | Samot | In most cases the alert stays until acknowledged. |
| 18:58.11 | Samot | So for hotels and the front desk phones, it means someone can see it. |
| 18:58.17 | Nivex | It's being originated by an Allstar node at the far end of a SIP line. Lemme see if I can reproduce with a SendMessage. |
| 18:58.24 | Samot | But as for having a "conversation" that way...no. |
| 18:58.37 | Samot | I have a GUI for that. |
| 18:59.24 | igcewieling | My users are unusually.....dense. They complain when they can't use the mailbox number for their voicemail password. |
| 19:00.10 | Samot | I have some users that don't use most features. |
| 19:00.24 | Samot | Some don't even use the visual voicemail in the GUI I setup.. |
| 19:00.38 | Samot | But then others, they are using that, the SMS and XMPP. |
| 19:01.01 | igcewieling | Sometimes I wish I had your users. |
| 19:01.13 | Samot | I don't do resi |
| 19:01.33 | Samot | And i really don't do SOHO |
| 19:01.39 | igcewieling | Neither do I. |
| 19:01.42 | Samot | I have both but not directly. |
| 19:01.49 | Samot | I mainly do hosted voice. |
| 19:01.55 | Samot | Not PBXes, voice. |
| 19:02.34 | igcewieling | *nod* at least half of my users are on their own PBX with analog or PRI handoff from our on-site adtran. |
| 19:02.59 | Samot | For the hotels it's "virtual PRIs" because that's what they understand.. |
| 19:03.07 | Samot | And a hybrid of hosted/on site. |
| 19:03.23 | Samot | When I install the guest hotel PBX |
| 19:03.35 | Samot | So all the "office/hotel" services are on the hosted platform |
| 19:03.44 | Samot | While the IP phones for the guests are on an internal PBX |
| 19:04.06 | igcewieling | *nod* I really hate hosted. Hosted users are too cheap to get a real PBX and more support too. |
| 19:04.13 | Samot | Uhm. |
| 19:04.27 | Samot | I got people paying for 20 seats at $22+ per seat per month |
| 19:04.39 | Samot | They are not "cheap" |
| 19:04.40 | Nivex | grumble. Using MessageSend() sends the message and it doesn't hang up :/ |
| 19:04.56 | Samot | Nivex: Then the other system is sending it wrong. |
| 19:05.44 | Samot | I'm not sure an office that pays $600 a month for their voice services are going to say they are too cheap to get a PBX |
| 19:05.48 | igcewieling | Ah. Most of our hosted users could have had an on-site PBX, but decided to go with hosted, since it is a lot cheaper unless there are a lot of users. |
| 19:05.52 | Nivex | ok. Is there some way to inhibit MESSAGE packets from a given place then? |
| 19:06.04 | Nivex | I don't _need_ the messages, but I do want it to not hang up. |
| 19:06.21 | Samot | Then why are you sending the messages? |
| 19:06.59 | Samot | The issue is how whoever is originating this message to you is doing it. |
| 19:07.13 | Nivex | I understand that. I want to drop those messages on the floor before they get to the Polycom |
| 19:07.14 | Samot | It's making the phone think there is a dialog or something happening. |
| 19:07.51 | Nivex | Allstar (old asterisk 1.4 custom job) -> My main Asterisk (13) -> Polycom phone |
| 19:08.09 | Nivex | Allstar is originating those messages |
| 19:08.33 | Samot | Then drop them at your Asterisk server. |
| 19:08.35 | Nivex | Interestingly Linphone on my Android device doesn't have any problem with those messages |
| 19:08.45 | Samot | 1) Your Asterisk trunk has to be setup to accept those messages |
| 19:08.52 | Samot | 2) You have to process those messages. |
| 19:09.02 | Samot | Or at least you should be doing it that way |
| 19:09.05 | Nivex | How do I drop them at my Asterisk server. That is what I do not know how to do. Help me Obi-Wan Samot :) |
| 19:09.09 | Samot | Don't send them to the phone. |
| 19:09.21 | Samot | Well you have a trunk to this allstar node right? |
| 19:09.45 | Samot | In that trunk set the accept_messages context (can't remember the exact name) and send those there to just drop them. |
| 19:09.49 | Samot | Never pass them through |
| 19:10.50 | Nivex | Dial(SIP/47201@allstar.nivex.lan) |
| 19:10.56 | Nivex | That's what I'm doing to call it right now |
| 19:11.16 | Samot | How do you accept calls from them? |
| 19:11.22 | Nivex | I don't. |
| 19:11.25 | Nivex | I place that call. |
| 19:11.31 | Nivex | It sends the messages back along that channel. |
| 19:11.34 | Samot | Then how are you getting a SIP MESSAGe from them? |
| 19:11.42 | Samot | Ahh. |
| 19:11.49 | Samot | So they send that as a reply in a call? |
| 19:11.55 | Samot | So a hangup should be here. |
| 19:12.02 | Samot | you still have to hangup the call |
| 19:12.19 | Nivex | The hangup is being originated by the Polycom, which it is apparently doing because it saw something it didn't like. |
| 19:12.31 | Samot | Right which is no audio |
| 19:12.40 | Nivex | There's audio flowing during that time. |
| 19:12.47 | igcewieling | Sam: $14/month (our hosted price I think) .vs. a server, a telephony card for analog backup, polycom phones, and PoE ethernet switches, analog lines for backup. |
| 19:13.05 | Samot | $14 is way to cheap |
| 19:13.12 | Samot | I wouldn't even be considered. |
| 19:13.23 | Samot | Not in my market area. |
| 19:13.56 | igcewieling | I'm not 100% sure. I don't talk to sales much after one of them admitted never reading my e-maiols. |
| 19:14.16 | Samot | $14 is like a "basic" seat. |
| 19:14.23 | Samot | Almost no features. |
| 19:15.21 | Samot | I complete against the LECs (like you) and Telco's/VSPs. |
| 19:15.55 | Samot | And if you are too off from them, no one takes you seriously. |
| 19:16.29 | igcewieling | I'd LOVE to double our hosted seat price. Might push more of them to a full PBX. |
| 19:16.38 | *** join/#asterisk sebastienthiry (~Thunderbi@109.134.212.7) |
| 19:16.51 | Samot | Dude, I can charge my resellers there $25.00 for unlimited basic phone service |
| 19:17.12 | Samot | Because they can still mark it up pretty well and still be competing against the LECs. |
| 19:17.46 | Samot | Granted I still have to do thinks like support 7 digit dialing. |
| 19:17.55 | Samot | Because it's still a thing there. |
| 19:18.10 | igcewieling | The problem with that is you need a decent sales force. Unfortunately, that's not what we have, though that could change after some upcoming management changes. |
| 19:18.48 | igcewieling | just confirmed, $16/month |
| 19:19.40 | Samot | Yeah, you can't provide LEC level services and try to compete with none LEC/Telco cloud providers. |
| 19:20.06 | Samot | $16/month as the standard is a wash. |
| 19:21.11 | Samot | But I don't have an internal sales force. |
| 19:21.15 | Samot | Well there's a guy. |
| 19:21.31 | Samot | But all of the core sales are outside sales guys |
| 19:21.38 | Samot | Who like that commission money |
| 19:27.53 | *** join/#asterisk lilDangerz (~lildanger@fixed-187-189-68-170.totalplay.net) |
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| 19:43.49 | Sepultura | Hallo |
| 19:44.00 | Sepultura | is Asterisk IPv6 on by default? |
| 19:46.56 | Samot | That is a config setting |
| 19:47.23 | Samot | You can use IPv6 on Chan_SIP but not Chan_PJSIP if you'd like. |
| 19:47.30 | Samot | Same with IPv4. |
| 19:47.42 | Samot | You have to tell Asterisk this stuff. |
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| 20:13.48 | igcewieling | hah! From a vendor regarding an older monitoring server we are having trouble with: " I am honestly surprised your server's hardware is still functional, " |
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| 21:03.54 | Bhakimi | hi guys im a bit desperate |
| 21:04.11 | Bhakimi | out of nowhere asterisk started to throw these errors: unable to send SIGHUP to AGI process 28250: No such process |
| 21:04.25 | Bhakimi | and when it does cpu goes throuhg the roof and kills the box |
| 21:04.33 | Bhakimi | only way to solve it is to restart asterisk |
| 21:04.59 | Bhakimi | i need some type of lead to look into why this is happening, can anyone possibly help me find out why it uses so much cpu ? |
| 21:05.11 | Bhakimi | this error may be a side effect of a different issue |
| 21:11.16 | scgm11_ | <PROTECTED> |
| 21:11.28 | scgm11_ | will dump the memory at that moment and give some feedback |
| 21:12.56 | scgm11_ | you probably will need to need to compile with dont optimze |
| 21:16.36 | Bhakimi | but not compile with debug correct |
| 21:16.44 | Bhakimi | cause debug causes deadlocks in productions |
| 21:18.23 | scgm11_ | there are many options |
| 21:18.33 | scgm11_ | dont optimize will at least let see something |
| 21:18.42 | scgm11_ | but maybe you will have to add more |
| 21:18.49 | scgm11_ | compiler flags |
| 21:19.25 | scgm11_ | check https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 21:22.09 | Bhakimi | thanks |
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| 23:45.59 | beight0 | I'm trying to set up a voicemail only server with LDAP storage and SQL voicemail users. I found that for pollmailboxes to work, I need a sip peer with that mailbox. If the peer is stored in SQL, pollmailboxes will not work because the peer is not loaded into cache. Setting rtcachefriends does not help as the peer never registers. Is there a way to automatically cache SIP peers from an SQL database? |