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| 08:07.26 | ashutoshkrchaube | Hi i am trying to do a playback on a specific channel however playback only takes current channel and plays on it how i can do that without changing the current channel? |
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| 11:31.44 | duyleekun | Hi |
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| 12:11.21 | Zevv | Hi; I'm running asterisk 13.1 from the stock ubuntu package. At startup if fails trying to load modules |
| 12:11.50 | Zevv | lots of unresolved symbols. It seems that this is case of https://gerrit.asterisk.org/#/c/6649/ |
| 12:12.01 | Zevv | Is there a workaround that does not require (re)building from source? |
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| 12:17.44 | _ppp | hi |
| 12:18.03 | _ppp | any recommended pci cards to install on a desktop machine? need to have 1fxo 1fxs |
| 12:18.28 | wdoekes | Zevv: yes, there is a workaround, which involves you ordering the modules explictly in modules.conf |
| 12:18.36 | wdoekes | not sure if that's easier |
| 12:19.24 | *** join/#asterisk zbro (b0dd7a8f@gateway/web/freenode/ip.176.221.122.143) |
| 12:19.37 | zbro | hello everyone |
| 12:19.56 | Zevv | wdoekes: hm probably not, that's too bad |
| 12:20.07 | Zevv | Funny that a bug like this creeps into the debian/ubuntu packages |
| 12:20.21 | Zevv | completely defunct from default install |
| 12:20.46 | zbro | Is there anyone willing to help with trivial (hopefully) configuration matter ? |
| 12:22.03 | _ppp | ? |
| 12:22.30 | zbro | I'm having two PBX gateways and I'm trying to make a call between the two |
| 12:23.00 | zbro | and I'm getting Call from 'Company' (192.168.2.100:5060) to extension '192.168.1.100' rejected because extension not found in context 'from-sip'. |
| 12:23.27 | zbro | dialplan show @from-sip [ Context 'from-sip' created by 'pbx_config' ] '192.168.1.100' => 1. Answer() [pbx_config] 2. Playback(hello-world) [pbx_config] 3. Hangup() [pbx_config] -= 1 extension (3 priorities) in 1 context. =- |
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| 12:24.20 | zbro | sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Company 192.168.2.100 Auto (No) No 5060 OK (55 ms) |
| 12:25.23 | zbro | 192.168.1.100 has an entry in extensions.conf , and as you can see it is visible in asterisk but for some reason the call is rejected |
| 12:25.49 | zbro | I'm pretty sure there's something trivial that I'm missing here |
| 12:26.08 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 12:27.17 | zbro | any clues ? |
| 12:29.31 | [TK]D-Fender | PASTEBIN it |
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| 12:34.35 | zbro | https://pastebin.com/1PiDsFex |
| 12:39.34 | [TK]D-Fender | and the call..... |
| 12:39.46 | [TK]D-Fender | why are you attempting to us an IP as an extension? |
| 12:40.01 | zbro | actually it's because |
| 12:41.09 | zbro | receiving side complains about extension called '192.168.1.100' which is btw receiving side's IP |
| 12:41.26 | [TK]D-Fender | Why are they sending that? |
| 12:41.27 | zbro | so I figured that I can simply add this extension, and it'll work. But it doesn;t |
| 12:41.29 | [TK]D-Fender | show us the actual call |
| 12:41.37 | [TK]D-Fender | and the raw config as well |
| 12:42.24 | zbro | ok, one moment |
| 12:43.28 | _ppp | any recommended pci cards to install on a desktop machine? need to have 1fxo 1fxs |
| 12:43.36 | _ppp | something with a good reputationete |
| 12:43.41 | [TK]D-Fender | TDM410P |
| 12:43.46 | [TK]D-Fender | Or whatever the replacement was |
| 12:43.59 | [TK]D-Fender | Digium or Sangoma card with HWEC |
| 12:49.00 | zbro | ok, so here it is: https://pastebin.com/iEaZ1MPK |
| 12:49.19 | zbro | as you can see it's a mess right now, as I've been experimenting with it |
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| 13:46.59 | Stokkeland | i have multiple sites running Asterisk 13.8-cert2 - all are stable except 1, which was fine and recently started segfaulting every so many days (from 4 to 10). I enabled -g and have a core file and trace but dont know what to look for |
| 13:47.15 | Stokkeland | guessing the version is a bit old to be submitting issues |
| 13:48.08 | Stokkeland | have been looking at https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=5243139 but it doesnt really guide me to look for what events may trigger the issues |
| 13:49.36 | [TK]D-Fender | are you paying Digium for support? |
| 13:50.27 | Stokkeland | no |
| 13:51.41 | [TK]D-Fender | then you should be using the far more updated ergular branch |
| 13:51.53 | [TK]D-Fender | LTS: 13.18.0 (2017/10/30), |
| 13:52.04 | [TK]D-Fender | Which got an update even just yesterday |
| 13:52.42 | Samot | Plus opening issues on Cert and not having an agreement is kinda pointless. |
| 13:54.14 | Stokkeland | yeah i know.. now I am just kind of looking for pointers to what may be causing it, if there are any info out there on how to figure the point of where things go wrong in a trace |
| 13:54.43 | Stokkeland | I am working on getting a budget for digium agreement - working in a startup makes such tricky sometimes :) |
| 13:55.07 | file | not really, the trace gives you the state of things and allows you to examine it - but you have to be familiar with Asterisk internals in order to understand and comprehend what is going on |
| 13:55.33 | Stokkeland | okay - maybe this event will help me get a budget |
| 13:55.40 | [TK]D-Fender | AKA "If you have to ask, it's not for you..." |
| 13:55.55 | [TK]D-Fender | Or you could start by using the regular branch more people can support |
| 13:56.23 | Samot | I'm not sure how Cert makes a difference for a startup |
| 13:56.49 | Samot | Stokkeland: Did you actually read the Cert stuff? |
| 13:57.08 | Samot | Or are you just basing this on the fact it has "Certified" in the name? |
| 13:57.09 | Stokkeland | I think the reason cert was chosen what it was believed to be more stable - we dont need features, just plain sip calling |
| 13:57.30 | Samot | By stable, you mean less updates. |
| 13:57.35 | Samot | Not bugs. |
| 13:57.41 | Samot | You have bugs we don't have |
| 13:57.46 | Samot | Because they were fixed. |
| 13:57.57 | Stokkeland | heh it may have been thought of as less bugs, guessing that may not be the case |
| 13:58.00 | Samot | You're still open to the RTP bleed bug. |
| 13:58.09 | Stokkeland | oh okay |
| 13:58.18 | [TK]D-Fender | Upgrade time.... |
| 13:58.26 | Stokkeland | sounds like it |
| 13:58.35 | Samot | You're still opened to bugs that prompted v11 to be updated TWICE within 3 weeks of being fully dead. |
| 13:58.38 | voipmonk | happy migrations. file your change management down the hall , when done make a left and grab some Timbits and Coffee while yet another team decides if your request is relevant enough to be granted. Wait about a day and go upstairs and let the communications team know when this is happening and dont forget any steakholders or you'll be toast like those in that bin of powder on your right next to the recycled parts. Have a nice day! |
| 13:59.18 | Stokkeland | no tim hortons in this area - have to stick to Dunkin Donuts :p |
| 13:59.24 | voipmonk | oh, and i need that plan in case you need to roll back. like yesterday |
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| 14:01.13 | voipmonk | and your dev box, the one you test the migrations on is down again... that wont be available until after the holiday. |
| 14:02.11 | zaf | mmm, steakholder |
| 14:03.42 | Stokkeland | https://pbs.twimg.com/media/ClD6aWmVAAE0mEu.jpg |
| 14:03.52 | [TK]D-Fender | https://lh4.googleusercontent.com/8P9MNMOR4oqBkRY85sfOqbArgC_pgnERNCbwG6X8yKAO9I3bD_FmKKEP13d2is5ryTrZI810TD7ekaf6LloO8ThfLCZ56aGdZ5Er7mct04tfHDZbJfcJnMQ2WQ |
| 14:04.17 | voipmonk | haha |
| 14:04.47 | voipmonk | i love that purple background, reminds me of a certain place... |
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| 16:11.57 | drab | hi, I'm running vanilla asterisk 13 on ubuntu 16.04. I'm trying to set up google voice. I've configured motif.conf and xmpp.conf as described in the tutorials, here's a paste: http://dpaste.com/0QF66FA |
| 16:12.07 | drab | Iv'e also logged into the gmail account, enabled less secure apps |
| 16:12.41 | drab | chan_motif and res_xmpp are loaded in modules.conf |
| 16:13.04 | drab | however when I issue xmpp show connections it shows the account as Disconnected |
| 16:13.21 | drab | and after a while in a cli started with asterisk -cvvvv I see this: |
| 16:13.49 | drab | ERROR[17926]: res_xmpp.c:3630 xmpp_client_reconnect: No XMPP connection available when trying to connect client 'google' |
| 16:14.55 | drab | I tried to put debug=yes in the [general] section, but I don't see much going on other than that same message I get in console |
| 16:15.04 | drab | otheriwse it all checks out, configs are found and loaded etc |
| 16:16.55 | drab | password and account match, as I logged in through the web interface and then pasted into config, so I'm certain those work |
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| 16:35.58 | Stokkeland | so... how do I become an expert in everything coding asterisk in no-time? Read "C for dummies" today? expert tomorrow? j/k |
| 16:38.10 | [TK]D-Fender | Or just get the latest fixes in the standard stream |
| 16:39.16 | Samot | drab: Google Voice is dead in that manner. |
| 16:39.45 | Samot | drab: They officially shut down at the end of June and I'm sure are slowly shutting down XMPP connectors. |
| 16:40.09 | Samot | drab: They no longer allow XMPP Fed projects to connect to them like they used to. |
| 16:42.29 | *** part/#asterisk c|oneman (cloneman@vps.digitaloverhaul.com) |
| 16:47.00 | drab | Samot: oh ok, well for now it's working, I found my issue, it was a dns problem |
| 16:47.39 | drab | but understood that I should not be relying on it, I'm trying to find/convince ppl to pay for some service, just not sure which one yet |
| 16:48.00 | Samot | Jesus. |
| 16:48.12 | Samot | Why do people need "convincing" for their phone service? |
| 16:48.53 | Samot | At least with the point of paying for it. |
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| 16:50.12 | Samot | You know, for the average user the price of a Starbuck's Grande PSL would cover service with Flowroute for like a f'ing month. |
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| 17:09.43 | *** join/#asterisk Nextlink (~tyui@luminet-2-130-190.cust.b2b2c.ca) |
| 17:11.30 | Nextlink | hey guys i am having this message saying username mismatch on trunks " have <nec2> digest have <501> |
| 17:11.35 | *** join/#asterisk polysics (~polysics@host52-216-dynamic.57-82-r.retail.telecomitalia.it) |
| 17:11.35 | Nextlink | but i know they are right |
| 17:11.50 | Nextlink | i remember i had to put a setting in the trunk to get rid of this but i forgot |
| 17:12.08 | Nextlink | my type is peer for outgoing and friend for incomming |
| 17:12.24 | polysics | hey there! has there ever been any progress on support for HTTP ASR/TTS services like Watson? |
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| 17:16.02 | Samot | What do you mean? |
| 17:16.13 | [TK]D-Fender | Nextlink, that wil happen when you have 2 devices at the same IP using type=peer |
| 17:16.17 | Samot | Progress on support in what manner? |
| 17:18.39 | Samot | Watson is probably a bad example considering most services like that have you POST the text you want converted and they write the audio file straight back to you. |
| 17:19.04 | Samot | Which you then have to make sure is in a spot that Asterisk can call on and in the right format for Asterisk to use it. |
| 17:20.30 | Samot | On the fly, dynamic TTS should be done on the system. |
| 17:21.41 | Samot | And Watson conversion is fast but you're still look at 3-5 seconds at least of dead air. |
| 17:21.54 | polysics | Samot: yeah, I was specifically referring to doing HTTP chunked streaming instead of fetch-then-play |
| 17:21.56 | Samot | While the POST happens, the conversion happens and the write out happens. |
| 17:22.16 | Samot | You shouldn't be POSTing to something like Watson *during* the call. |
| 17:22.19 | polysics | doing the non-streaming is a few lines of script rally |
| 17:22.27 | polysics | *really |
| 17:22.48 | Samot | Depending on the size of the text, the load and other variables.. |
| 17:22.58 | Samot | Your call will sit there doing nothing while your TTS happens. |
| 17:24.53 | Samot | I use Watson for TTS/STT stuff quite a bit now but only to generate recordings for scripts submitted by users. |
| 17:25.05 | polysics | yes, that is not what I was referring to. There had been a discussion at AstriDevCon 2016 regarding supporting some type of streaming |
| 17:25.30 | Samot | I use Cepstral and app_swift for prompts that are generated dynamically based on user input. |
| 17:25.49 | Samot | Well Asterisk 14 supports audio streaming from remote locations. |
| 17:25.51 | polysics | how's the quality? has it improved? |
| 17:25.59 | Samot | I don't have an issue with it |
| 17:26.12 | Samot | And I paid the extra $40 for the Allison recordings. |
| 17:26.17 | polysics | having streaming ASR/TTS would be killer |
| 17:26.25 | Samot | You're missing the point. |
| 17:26.37 | Samot | That would require a streaming server. |
| 17:26.46 | polysics | I am not. I am thinking "replace UniMRCP with Watson/Amazon" |
| 17:26.51 | Samot | Watson would have to generate the audio file and then stream it back to you |
| 17:27.13 | polysics | they do HTTP streaming - download starts before it is finished creating hte file, and ASR is ran in chunks over audio |
| 17:27.33 | Samot | Yeah, I guess you can do that. |
| 17:27.39 | Samot | And hope for the best. |
| 17:27.42 | polysics | you get ASR results back immediately for various definitions of "immediately" |
| 17:27.57 | Nextlink | samot can i ask you a question |
| 17:27.59 | polysics | but it's faster than record-POST-recog-act |
| 17:28.15 | Samot | Nextlink: Sure. |
| 17:28.27 | Samot | polysics: I prefer that type of stuff done on system. Faster. |
| 17:28.48 | Samot | Since you are already doing other things to make that happen. |
| 17:28.48 | polysics | I do too, but TTS/ASR options are expensive for some applciations |
| 17:28.58 | polysics | what do you use for ASR? |
| 17:29.59 | Samot | I don't do a lot of ASR. |
| 17:30.25 | Samot | Or any at this time. |
| 17:30.41 | Nextlink | Samot: I connected a sip trunk to another system ( not asterisk ) i am receiving a message saying "Username mismatch, have <nec2>, digest has <501>... but nec2 is just the trunk name in asterisk. the user context is 501. i had this issue in the past i fixed it with a setting but i forgot. type=friend for incomming |
| 17:30.54 | Samot | OK. |
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| 17:31.18 | Nextlink | have any idea? |
| 17:31.26 | Samot | Don't use the user context. |
| 17:31.31 | Samot | There's really no need for it. |
| 17:31.38 | Samot | Just put it in the peer settings. |
| 17:31.46 | Nextlink | user=501 ? |
| 17:31.52 | Samot | I can't see your configs. |
| 17:31.55 | Samot | So I don't know |
| 17:31.57 | Samot | Show them |
| 17:31.58 | Samot | ~pb |
| 17:31.58 | infobot | it has been said that pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:32.53 | [TK]D-Fender | type=friend <---- |
| 17:33.00 | [TK]D-Fender | What you should be using |
| 17:33.24 | Nextlink | its already friend |
| 17:34.47 | Nextlink | i am sorry i tought i was in freepbx channel |
| 17:34.53 | Nextlink | this change things |
| 17:35.04 | Nextlink | so i am forced to use user context |
| 17:35.26 | [TK]D-Fender | no |
| 17:35.44 | [TK]D-Fender | there is almost no need for that other section ever |
| 17:36.00 | Nextlink | for the incomming section? |
| 17:36.17 | Nextlink | ok |
| 17:39.48 | Nextlink | in the peer section should i put type=friend? |
| 17:40.45 | *** join/#asterisk IPalmer (~IceChat9@88.211.120.74) |
| 17:45.14 | IPalmer | Afternoon: I'm using asterisk 14.4.0, I'm using ari push to configure my endpoints. I'm having an issue when adding a record to identify (ps_endpoint_id_ips) The record I add is visible when I do a "PJSIP show identify 1234" but it doesn;t write it to the database, I get an error from MySQL saying WARNING[60963]: res_config_odbc.c:120 custom_prepare: SQL Prepare failed! [INSERT INTO ps_endpoint_id_ips (id, match, endpoint, match_header, srv_lookups) VA |
| 17:45.27 | Nextlink | now its saying wrong password |
| 17:45.43 | [TK]D-Fender | Nextlink> for the incomming section? NONE |
| 17:46.05 | [TK]D-Fender | it should never be used PERIOD 99.9999% of the time |
| 17:46.32 | Nextlink | https://pastebin.com/eG2e42gt |
| 17:46.35 | Nextlink | this is my setting |
| 17:46.38 | IPalmer | Does anyone know of a way I can troubleshoot this, I've looked at column counts and have replaced the enums with varchars (which was an issue previously) |
| 17:46.47 | Nextlink | i removed everything in incomming |
| 17:46.58 | [TK]D-Fender | username=501 <- not even valid anymore |
| 17:47.02 | [TK]D-Fender | not for a LONG time now |
| 17:47.05 | [TK]D-Fender | defaultuser <- |
| 17:47.11 | Nextlink | thanks |
| 17:47.50 | [TK]D-Fender | Also no codecs are properly defined for it, no implication on nat or directmedia, etc |
| 17:54.07 | Nextlink | https://pastebin.com/hNrLj54s |
| 17:54.09 | Nextlink | new settings |
| 17:56.57 | Nextlink | wrong password again |
| 18:03.35 | Nextlink | is secret ok? |
| 18:04.37 | Nextlink | secret= |
| 18:10.10 | *** join/#asterisk jamesaxl (~James_Axl@109.172.62.242) |
| 18:10.33 | [TK]D-Fender | You're still not showing us the FAILURE |
| 18:11.18 | *** part/#asterisk IPalmer (~IceChat9@88.211.120.74) |
| 18:11.38 | Samot | Is this the peer that is being connected to? |
| 18:12.02 | jamesaxl | hi |
| 18:12.16 | Nextlink | this is the setting for the asterisk |
| 18:12.22 | Nextlink | the other peer is a nec pbx |
| 18:12.22 | Samot | I get that |
| 18:12.35 | Samot | Is the NEC connecting to Asterisk? |
| 18:12.35 | [TK]D-Fender | Show us the attempt |
| 18:12.45 | Samot | secret=debug123 <-- That is for the NEC to use. |
| 18:12.51 | Nextlink | it's not because now it saying wrong password |
| 18:12.55 | Samot | I.e. debug123 is the password. |
| 18:13.03 | Nextlink | yeah the nec have 501 and debug123 as credentials |
| 18:13.09 | [TK]D-Fender | Show us the attempt |
| 18:13.13 | Samot | ^^^^ |
| 18:13.18 | Nextlink | hold on |
| 18:13.45 | Samot | oh wait.. |
| 18:14.06 | Samot | Answer this, since I saw you mention FreePBX. Is this a v13 or v14 distro install? |
| 18:14.22 | Nextlink | v13 |
| 18:14.25 | Samot | OK |
| 18:14.27 | [TK]D-Fender | JUST SHOW THE ATTEMPT |
| 18:14.31 | Samot | STOP |
| 18:14.35 | jamesaxl | if I have an issue with outgoing and incoming calls(provider), the client that calls us or that we call, (s)he does not hear us. Should I check the firewall? because intern calls work very good |
| 18:14.37 | Samot | Did you modify the SIP ports? |
| 18:14.40 | [TK]D-Fender | I'm giving rope |
| 18:14.57 | Samot | Nextlink: Did you modify the SIP ports? |
| 18:15.23 | *** join/#asterisk jkroon (~jkroon@165.16.204.168) |
| 18:15.53 | Nextlink | i am |
| 18:15.55 | Samot | Nextlink: Did you tell Chan_SIP to use 5060? |
| 18:16.19 | Samot | These are yes/no questions. |
| 18:16.43 | Nextlink | yes port 5060 |
| 18:16.47 | Nextlink | its working with other trunks |
| 18:16.52 | Samot | OK. |
| 18:16.57 | Nextlink | i am getting logs now |
| 18:17.02 | Samot | OK. |
| 18:17.02 | jamesaxl | [TK]D-Fender: any idea? |
| 18:17.32 | [TK]D-Fender | jamesaxl, Could be a lot of things |
| 18:17.33 | Samot | jamesaxl: Is it to just this one client? |
| 18:17.44 | Samot | jamesaxl: Or from the one client? |
| 18:17.48 | [TK]D-Fender | You've told us almost nothing and shown even less |
| 18:18.25 | jamesaxl | Samot: from all clients (outgoing and incoming calls) external |
| 18:18.46 | Samot | jamesaxl: Then yes, you should look at your firewall. You clearly have a NAT issue. |
| 18:19.01 | Samot | Your internal phones don't use NAT |
| 18:19.48 | Nextlink | https://pastebin.com/ChvSALrr |
| 18:19.49 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 18:19.58 | Samot | No. |
| 18:20.02 | Samot | The ENTIRE attempt |
| 18:20.05 | Samot | With a SIP debug |
| 18:20.14 | Samot | sip set debug on <-- That command |
| 18:20.15 | jamesaxl | Samot: no they do not, I should check the server that is connected to internet, because it is the gate of my asterisk server. |
| 18:20.24 | Samot | Then make the other side try to connect. |
| 18:20.47 | Nextlink | ok |
| 18:21.05 | Samot | jamesaxl: Is the PBX registered to the provider? |
| 18:21.21 | Samot | jamesaxl: Do calls connect but just have no audio? |
| 18:24.56 | jamesaxl | Samot: yes it is , and I can make and receive calls, i can hear clients but clients can not hear us |
| 18:26.28 | Samot | OK |
| 18:26.35 | Samot | So why would you need to confirm Internet access? |
| 18:26.46 | Samot | This has nothing to do with the Internet access or connectivity |
| 18:27.05 | Samot | This has to do with how your firewall is handling the packets and NAT |
| 18:27.06 | jamesaxl | Samot: I have an other asterisk server connect directly to internet, and it works well, I use the same configuration in the second one, but i have this issue with it |
| 18:27.16 | Samot | This has to do with how your firewall is handling the packets and NAT |
| 18:28.37 | jamesaxl | Samot: I configure firewall very on the srever where exist asterisk, but I did not check rules of iptables in the server that guide me to outside, |
| 18:29.16 | Nextlink | its weird i cant find the sip attempt but i receive the info saying wrong password like i sent |
| 18:30.10 | jamesaxl | Samot: anyway, i will check hoe firewall handling the packets and NAT, and thank you very much |
| 18:30.50 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 18:33.54 | Nextlink | got it |
| 18:34.44 | Nextlink | https://pastebin.com/jR86RWnt |
| 18:34.54 | Nextlink | Samot: got the request |
| 18:36.49 | [TK]D-Fender | Your NEC is trying to register to * and that is not allowed |
| 18:36.55 | [TK]D-Fender | because you set a fixed host |
| 18:37.13 | [TK]D-Fender | host=dynamic <---------- |
| 18:37.36 | Nextlink | ok trying |
| 18:38.23 | Nextlink | same result |
| 18:38.44 | [TK]D-Fender | I'm not seeing updated configs, prrof they are applied, and the debug to match |
| 18:39.05 | Nextlink | ok hold on |
| 18:39.39 | *** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com) |
| 18:41.15 | Nextlink | https://pastebin.com/n4nPgmWv |
| 18:42.34 | *** join/#asterisk jamesaxl (~James_Axl@109.172.62.242) |
| 18:43.00 | Samot | There is only one peer? |
| 18:43.15 | Samot | You don't have any other peers setup to match on that NEC |
| 18:43.45 | Nextlink | there is 3 different peer in the pbx |
| 18:44.01 | Nextlink | 3 trunks |
| 18:44.24 | [TK]D-Fender | <[TK]D-Fender> I'm not seeing updated configs, prrof they are applied, and the debug to match |
| 18:44.28 | Samot | Where's the new trunk config? |
| 18:44.55 | Nextlink | coming |
| 18:46.26 | Nextlink | https://pastebin.com/8bUUx8tm |
| 18:46.39 | [TK]D-Fender | host=66.158.130.xxx&dynamic |
| 18:46.40 | [TK]D-Fender | NO |
| 18:46.45 | Samot | No kidding. |
| 18:46.47 | Nextlink | its doing that automaticly |
| 18:46.48 | [TK]D-Fender | Did I say to GLUE SHIT ON THE END? |
| 18:46.50 | Samot | We said change it. |
| 18:46.54 | Nextlink | when i set host=dynamic |
| 18:46.55 | Samot | Not append it |
| 18:47.03 | Nextlink | ho ok |
| 18:47.04 | Nextlink | sorry |
| 18:47.24 | [TK]D-Fender | That thing doesn't come up with an IP address out of NOWHERE |
| 18:49.07 | Nextlink | https://pastebin.com/XNyqeZsc |
| 18:49.30 | Nextlink | i am collecting the debug now |
| 18:51.17 | Nextlink | https://pastebin.com/snsTu5ii |
| 18:51.19 | Nextlink | wrong password |
| 18:51.56 | [TK]D-Fender | "sip show peer 501" and screenshot the entire trunk |
| 18:52.57 | Nextlink | peer 501 not found |
| 18:53.55 | [TK]D-Fender | Well I guess you should have named the TRUNK 501 |
| 18:54.11 | [TK]D-Fender | in that box right above |
| 18:54.34 | Nextlink | ok |
| 18:55.59 | Nextlink | https://pastebin.com/thcpgm5b |
| 18:56.30 | Nextlink | how should i send you the screenshot |
| 18:56.34 | Samot | <PROTECTED> |
| 18:56.44 | [TK]D-Fender | looks like it has registered now |
| 18:56.44 | Samot | Status : OK (86 ms) |
| 18:56.56 | [TK]D-Fender | And is qualifying |
| 18:57.07 | Samot | Having correct configs and settings in them is a wonderful thing. |
| 18:59.19 | Nextlink | ok not its registered. |
| 18:59.21 | Nextlink | thanks |
| 18:59.45 | Nextlink | call is failing from nec saying rejected because extension not found in context internal, should i change the context? |
| 19:00.07 | [TK]D-Fender | You should read your trunk basics |
| 19:00.15 | [TK]D-Fender | Because you clearly haven't |
| 19:00.18 | Nextlink | from-internal |
| 19:00.20 | Nextlink | oops |
| 19:00.21 | [TK]D-Fender | And you shouldn't be doing any of this in here |
| 19:00.28 | [TK]D-Fender | GUI's aren't supported in the channel |
| 19:00.36 | [TK]D-Fender | ~freepbx |
| 19:00.36 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:00.59 | Nextlink | yeha anyway thank you very much |
| 19:01.02 | Nextlink | i wish you a good day |
| 19:02.46 | *** join/#asterisk Typhon (~Typhon@dslb-092-078-207-053.092.078.pools.vodafone-ip.de) |
| 19:04.48 | *** join/#asterisk tzafrir (~tzafrir@212.29.194.37) |
| 19:06.54 | *** join/#asterisk jamesaxl (~James_Axl@109.172.62.242) |
| 19:14.51 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 19:41.39 | Samot | Oh please. |
| 20:30.17 | *** join/#asterisk drab (~administr@23-24-198-190-static.hfc.comcastbusiness.net) |
| 20:31.35 | drab | hi, I'm migrating from asterisk 11 on a server to asterisk 13 on another machine. I've ported and adjusted all the configs in /etc/asterisk, however I'm wondering what else needs to be moved over. |
| 20:31.48 | drab | I have a bunch fo custom sounds in /var/lib/asterisk and voicemails in /var/spool/asterisk/voicemail/ |
| 20:32.04 | drab | but for example I'm not even sure if porting over the entirety of those dirs make sense |
| 20:32.08 | drab | as things in 13 may have changed |
| 20:32.18 | drab | these are vanilla install from packages on ubuntu 14.04 and 16.04 |
| 20:32.49 | drab | no mysql or other external dependency, plain and simple installs |
| 20:35.18 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:a0fe:42bd:3e6d:2de3) |
| 20:47.27 | *** join/#asterisk aness (~aness@cm-84.209.56.255.getinternet.no) |
| 21:03.34 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 21:16.30 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:18.36 | Samot | No. |
| 21:18.54 | Samot | Things like that would have been noted in the wiki and in the sample configs |
| 21:19.21 | Samot | 13 and 15 still use /var/spool/asterisk/voicemail as the local directory for storing voicemails. |
| 21:20.58 | *** join/#asterisk tuxd00d (~tuxd00d@wsip-24-248-81-186.ph.ph.cox.net) |
| 21:22.07 | *** join/#asterisk Penguin (~xwQ5kwYl6@our.systems.are.full.of.penguins.at.penguinsystems.net) |
| 21:31.05 | drab | thank you Samot |
| 21:37.03 | [TK]D-Fender | DAHDI if applicable, AstDB |
| 21:39.51 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 21:46.39 | Samot | Are these VMs? |
| 21:47.06 | Samot | Because I would just snapshot the 11 machine and load it up on another machine and just recompile Asterisk on it. |
| 21:47.09 | Samot | Go from there. |
| 21:56.43 | drab | Samot: current is baremetal, new one is a lxc container |
| 22:06.32 | *** join/#asterisk MrTAP (~MrT@unaffiliated/mrtap) |
| 22:06.47 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 22:37.59 | *** part/#asterisk kharwell (kharwell@nat/digium/x-cdyhrnhtqijwjdhy) |
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| 23:10.46 | *** join/#asterisk [[thufir]] (~thufir@192.157.119.2) |
| 23:21.22 | *** join/#asterisk rangotail (~rangotail@unaffiliated/rangotail) |
| 23:58.27 | *** join/#asterisk saint_ (~saint_@unaffiliated/saint-/x-0540772) |
