| 16:51.15 | *** join/#asterisk infobot (ibot@208.53.50.136) |
| 16:51.15 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0-rc1 (2018/07/03), Standard: 15.5.0-rc1 (2018/07/03); 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 |
| 16:53.36 | *** join/#asterisk mutin-sa (~s-mutin@85.234.114.134) |
| 17:14.10 | *** join/#asterisk miralin (~Thunderbi@194.8.128.63) |
| 17:14.31 | *** join/#asterisk waldo323 (~waldo323@75-151-31-89-Michigan.hfc.comcastbusiness.net) |
| 17:22.11 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 17:39.46 | *** join/#asterisk miralin1 (~Thunderbi@195.209.246.194) |
| 17:46.54 | *** join/#asterisk shanth (~shanth@140.55.197.35.bc.googleusercontent.com) |
| 17:56.21 | TandyUK2 | anyone know how to do "sip set debug ip ip1,ip2,ip3" or equivalent? |
| 17:57.02 | file | chan_sip has no CLI command to do multiple IPs - it's all or 1 specific one |
| 17:57.37 | TandyUK2 | 5 ssh sessions it is then lol |
| 17:57.45 | TandyUK2 | that would be a _really_ useful feature |
| 17:58.00 | file | the command isn't per console, it's to the entire process |
| 17:58.18 | TandyUK2 | ok wierd then |
| 17:58.47 | Samot | Are you just wanting to see what is in the actual SIP message? |
| 17:59.03 | TandyUK2 | i need to track why we are rejecting some calls from simwood with 401's |
| 17:59.13 | TandyUK2 | but their request can reasonably come from any of 5 ips |
| 17:59.31 | Samot | Are they an inbound provider? |
| 17:59.42 | TandyUK2 | usual story atm is, i enable debug on one of the ips, and ofc the actual call comes from anothr |
| 17:59.50 | TandyUK2 | yeah |
| 17:59.59 | Samot | Are you authing their INVITEs? |
| 18:00.05 | TandyUK2 | they have the first anycasted sip network ive heard of |
| 18:00.13 | Samot | 401 is a Auth rejecting meaning "Send me Auth Digest" |
| 18:00.22 | TandyUK2 | thats the problem, NO, but apparantly we are rejecting _some_ requests for lack of auth |
| 18:00.38 | Samot | Are you using Chan_SIP or Chan_PJSIP? |
| 18:00.43 | TandyUK2 | all inbound trunks have insecure invites set, and we have 1 per remote ip |
| 18:00.50 | TandyUK2 | *11, so chan_sip |
| 18:01.02 | Samot | Is it insecure=invite,port? |
| 18:01.06 | TandyUK2 | correct |
| 18:01.16 | Samot | So a rejected call |
| 18:01.19 | Samot | Show |
| 18:01.39 | TandyUK2 | this is the problem lmao |
| 18:01.48 | TandyUK2 | actualy _finding_ logs from a failed call |
| 18:02.04 | TandyUK2 | I have ~1000 handsets registeredo n this system, so the noise is insane with global debug on |
| 18:02.52 | file | do a packet capture externally and filter afterwards |
| 18:04.52 | Samot | tcpdump |
| 18:05.50 | Samot | Should be able to do a host 1.1.1.1. or host 1.2.2.2 or host 1.2.3.4 |
| 18:06.21 | file | you probably don't want to do that though... because you're operating under the assumption that you really do know all the IP addresses |
| 18:08.11 | Samot | OK, ngrep it |
| 18:08.33 | TandyUK2 | i think this is because the call im getting from 178.22.139.77 (what we have trunk setup with), is sendign a call FROM: "mymobile" <sip:mymobile@178.22.139.4> |
| 18:08.44 | TandyUK2 | we dont know who the fk 178.22.139.4 is, rejected |
| 18:09.00 | Samot | ngrep -W byline -d eth0 'NXXXXXXX' port 5060 -O capture_file |
| 18:09.06 | file | chan_sip doesn't care about that. |
| 18:09.37 | Samot | ngrep -W byline -d eth0 'NXXXXXXX' port 5060 <-- Live output instead of file. |
| 18:09.49 | Samot | NXXXXX being the DID being called |
| 18:10.06 | Samot | It will get _any_ SIP packet with that DID in it. |
| 18:10.36 | Samot | Change the eth interface and sip port to match your setup if needed. |
| 18:12.02 | TandyUK2 | https://gist.github.com/tandyuk/8eaae57bd960ff734e98e52206d3fd22 |
| 18:12.23 | TandyUK2 | and you all have my mobile number lol |
| 18:12.35 | TandyUK2 | other one was public to begin with :) |
| 18:13.21 | igcewieling | nobody cares about your mobile number. |
| 18:13.23 | Samot | OK, now show the peer config... |
| 18:14.33 | TandyUK2 | https://gist.github.com/tandyuk/9c270bcefe5f0fd4c05d48731ff8f7b5 |
| 18:15.15 | Samot | Why are you using "friend"? |
| 18:15.18 | Samot | This is a peer. |
| 18:18.46 | TandyUK2 | i think im just getting confused by shit documentation |
| 18:18.55 | TandyUK2 | "peer: an entitry to which we send calls" |
| 18:19.16 | Samot | A peer doesn't have a user. |
| 18:19.25 | Samot | "friend" makes both a peer and a user |
| 18:19.31 | Samot | so it wants to auth by the user. |
| 18:19.36 | Samot | Which you don't have. |
| 18:19.45 | Samot | Your provider trunks are peers. |
| 18:19.56 | Samot | Unless you are using SIP Registration. |
| 18:19.59 | TandyUK2 | yeah i mean this is an onbound only trunk, so scopserv's shit labelling was throwing me off |
| 18:20.01 | Samot | Then it's a "friend" |
| 18:20.45 | Samot | peer's will auth on the IP and the port |
| 18:20.53 | Samot | Friend will auth on the user and IP |
| 18:23.52 | Samot | Though, this still shouldn't send a 401 challenge. |
| 18:23.53 | TandyUK2 | ok, so now theyre all peers. |
| 18:24.00 | TandyUK2 | any idea why this still happened? |
| 18:24.00 | TandyUK2 | https://gist.github.com/tandyuk/fc21014d4f0c907861fe728294396120 |
| 18:24.24 | Samot | I'm not sure. Even with friend the insecure setting shouldn't send a 401 challenge. |
| 18:28.36 | [TK]D-Fender | I'd want to see actual * CLI debug for this call... |
| 18:28.44 | [TK]D-Fender | to prove what it's hitting |
| 18:30.23 | Samot | We covered this. |
| 18:30.32 | Samot | The provider can send the call from multiple IPs. |
| 18:31.10 | Samot | sip debug can do 1 IP. |
| 18:31.34 | TandyUK2 | this is wierd, identical settings bar the ip on the 2 trunks, call from 185.63.140.77 works, from 178.22.139.77 does not (401 response) |
| 18:31.44 | TandyUK2 | and ive just repeated that about 4 times per ip |
| 18:31.56 | Samot | Need to see the debugs |
| 18:32.36 | Samot | You might have to enable "sip set debug ip 178.22.139.77" and make a test call until it hits. |
| 18:36.09 | TandyUK2 | ok so lets rewind having juts figured out why the 401's are happening |
| 18:36.25 | TandyUK2 | prior to the weekend, we just used a plain IP trunk with simwood |
| 18:37.16 | TandyUK2 | due to taking on a new customer, who has their own account with simwood, we have replaced the single (set of) trunk with seperate Client-Outgoing trunks, that each have their own auth on simwoods end |
| 18:37.16 | Samot | So you had one trunk/peer? |
| 18:37.38 | TandyUK2 | no there was one per simwood ip, but just one set of them |
| 18:37.48 | Samot | Oh.. |
| 18:37.50 | *** join/#asterisk miralin (~Thunderbi@194.8.128.63) |
| 18:37.58 | TandyUK2 | all outgoing calls just went over a signle trunk, regardless of who it was from |
| 18:38.00 | Samot | You have TWO peers with the same IP? |
| 18:38.00 | *** join/#asterisk wtf911 (42184ca9@gateway/web/freenode/ip.66.24.76.169) |
| 18:38.10 | TandyUK2 | yeah this seems to be the issue |
| 18:38.13 | Samot | OK |
| 18:38.17 | Samot | so which one is first? |
| 18:38.24 | TandyUK2 | i need to send calls to simwood, and recieve them from simwood, on different trunks |
| 18:38.25 | Samot | Your "global" or the new customers? |
| 18:39.10 | TandyUK2 | atm, the customer specific ones are all listed first |
| 18:39.21 | Samot | Which should receive inbound calls? |
| 18:39.36 | TandyUK2 | the global incoming trunks should recieved all calls |
| 18:39.42 | Samot | OK |
| 18:39.51 | TandyUK2 | the customr one should only be used for that specific customers outbound |
| 18:40.06 | Samot | So which peer where you showing? |
| 18:40.08 | Samot | Earlier... |
| 18:40.13 | Samot | Yours or the customers? |
| 18:40.15 | [TK]D-Fender | If you have 2 you're asking to get screwed |
| 18:40.19 | TandyUK2 | that was one of the global incoming ones |
| 18:40.23 | [TK]D-Fender | Which is why I was asking about that... |
| 18:40.24 | Samot | OK |
| 18:40.30 | Samot | So here's what is happening |
| 18:40.35 | Samot | your client's peer is first |
| 18:40.39 | Samot | So it matches first |
| 18:40.53 | Samot | Put your peer above it. |
| 18:41.23 | *** join/#asterisk jamesaxl (~James_Axl@41.140.35.122) |
| 18:41.28 | TandyUK2 | this is where gui fucks me over, as it has no 'order', its just the order in which they are defined |
| 18:41.43 | Samot | It doesn't do it by name? |
| 18:41.49 | Samot | Or number order? |
| 18:42.01 | Samot | It just sorts by the database "id"? |
| 18:42.07 | TandyUK2 | yeah, 'order in which it was added to db' |
| 18:42.18 | Samot | So you're screwed. |
| 18:42.21 | TandyUK2 | which isnt necesarily the id |
| 18:42.31 | Samot | If you can't adjust the order |
| 18:42.38 | Samot | You can't adjust what matches first. |
| 18:42.40 | TandyUK2 | cd .. |
| 18:43.01 | TandyUK2 | whe nthe gui builds the configs, its using includes |
| 18:43.37 | TandyUK2 | customer/sip, customer/sip-extras, customer2/sip, customer2/sip-extras, ...... all/sip, all/sip-extras |
| 18:43.42 | TandyUK2 | the global stuff is in all |
| 18:43.57 | Samot | If "all" is last... |
| 18:44.01 | Samot | Then it matches last. |
| 18:44.03 | TandyUK2 | yeah theres the bug |
| 18:44.32 | Samot | Should "all" be first? |
| 18:44.41 | TandyUK2 | certainly sounds like it |
| 18:45.00 | Samot | Well it depends on how their "multi-tenant" logic is handled. |
| 18:45.35 | TandyUK2 | well i just manaully adjusted the configs and reloaded |
| 18:45.38 | igcewieling | I didn't know FreePBX has a multi-tennant option. |
| 18:45.41 | TandyUK2 | its still handled by the same wrong trunk |
| 18:45.43 | Samot | It doesn't. |
| 18:45.47 | Samot | This is ScopServ. |
| 18:45.48 | TandyUK2 | its not freepbx |
| 18:46.41 | TandyUK2 | ah, this is why i was setting them to 'friend' |
| 18:46.53 | Samot | Yes, that does make more sense. |
| 18:47.07 | Samot | Didn't know you had multiple peers to the same IP. |
| 18:47.18 | Samot | Until after, so yeah friend does make more sense. |
| 18:47.25 | TandyUK2 | when i do that, gui gives me an incoming calls tab, which gives me an option 'use global incoming context' |
| 18:47.53 | TandyUK2 | that seemed to work, but again may have just been luck that calls were hitting a different peer |
| 18:50.34 | TandyUK2 | kinda wishing i'd gone with my other option for sorting this on saturday.... giving th fking box another ip |
| 18:52.51 | TandyUK2 | if i did that going forward, would having the same set of peers using 2 ips |
| 18:52.56 | TandyUK2 | 1 ip purely for inbound |
| 18:53.00 | TandyUK2 | and 1 purely for outbound |
| 18:53.06 | TandyUK2 | would that play nicely? |
| 18:53.15 | TandyUK2 | regardless of the peer order in sip.conf |
| 18:54.18 | Samot | No. |
| 18:54.21 | TandyUK2 | that said i can see any way to set which ip is used |
| 18:54.25 | Samot | A peer is a peer |
| 18:54.28 | TandyUK2 | ^^ kinda figured :( |
| 18:54.38 | Samot | The host defines the IP the PBX will accept/send calls to/from. |
| 18:54.44 | Samot | Doesn't matter what IP. |
| 18:55.31 | Samot | That being said.... |
| 18:55.36 | Samot | You could use the domain setting. |
| 18:56.03 | Samot | Technically each of your IPs would be a domain. |
| 18:56.03 | TandyUK2 | all my calls are delivered to number@voip.tandyuk.com if that helps |
| 18:56.10 | Samot | OK |
| 18:56.14 | Samot | You would need a second domain. |
| 18:56.26 | TandyUK2 | if i can tie that domain only to the 5 incoming trunks wold that work? |
| 18:56.32 | Samot | Because you can tell a peer to accept calls to X domain and what context to use |
| 18:56.58 | Samot | If you add a second IP to the PBX you can make voip2.tandyuk.com |
| 18:57.25 | Samot | And tell the peer that domain=voip2.tandyuk.com,global-incoming-context |
| 18:57.57 | Samot | domain=voip.tandyuk.com,customer-incoming-context |
| 18:58.01 | Samot | Or however... |
| 18:59.16 | Samot | Hell, you don't even need a second IP if Simwood allows FQDNs for their SIP URIs. |
| 18:59.41 | Samot | You can point all the domains to the PBXes IP and use the domain= setting to route calls based on the domain. |
| 19:03.27 | TandyUK2 | i think i have a better temporary solution... use simwoods beta anycast ips for all outgoing calls |
| 19:03.40 | TandyUK2 | so the 5 ips for existing trunks wouldnt be in use |
| 19:10.10 | TandyUK2 | yeah that seems to be working |
| 19:10.25 | TandyUK2 | now all my client specific peers are on a single different ip to what i will ever recieve calls from |
| 19:10.42 | TandyUK2 | even though that single ip is anycasted to all of simwoods sites |
| 19:11.34 | *** join/#asterisk billxx (49958984@gateway/web/freenode/ip.73.149.137.132) |
| 20:18.19 | *** join/#asterisk wtf911 (42184ca9@gateway/web/freenode/ip.66.24.76.169) |
| 20:19.10 | wtf911 | WARNING[9536]: pjproject:0 <?>: SSL STATUS_FROM_SSL_ERR (connecting): Level: 0 err: <336151598> <SSL routines-SSL3_READ_BYTES-tlsv1 alert protocol version> len: 0 |
| 20:19.44 | wtf911 | SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336130315> <SSL routines-SSL3_GET_RECORD-wrong version number> len: 0 |
| 20:20.27 | wtf911 | getting those errors when trying to register google voice with the new naf google voice pjsip. i'm using a virtual machine of debian 8.9 and openssl 1.0.1 2016 |
| 20:54.57 | rfr__ | Still stuck with this no dial tone on new dahdi / * machine if anyone can help. https://pastebin.com/UyNnHwT6 |
| 20:56.57 | *** join/#asterisk scgm11_ (~scgm11@r186-52-166-87.dialup.adsl.anteldata.net.uy) |
| 20:57.02 | drmessano | HA |
| 20:57.12 | drmessano | I downloaded the gvsip package |
| 20:57.19 | drmessano | Holy patches and patches |
| 20:57.39 | drmessano | I've never seen anything like it |
| 21:05.56 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:13.18 | *** join/#asterisk rwb (~Thunderbi@65.183.151.121) |
| 21:38.05 | Samot | That bad? |
| 21:41.53 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 21:45.15 | file | while Google is using SIP it's not the implementation that everyone else is using, they are using RFCs and standards that are... newer |
| 21:45.34 | rfr__ | In extesnions.conf is the global 'Dahdi/14' properly expressed or must it be DAHDI/14? |
| 21:49.38 | *** join/#asterisk jetlag (~jetlag@c-71-226-222-56.hsd1.nj.comcast.net) |
| 21:54.59 | jpsharp | gvsip looks like a clusterfuck of epic proportions. |
| 21:59.40 | drmessano | The patches are just a freaking nightmare, from what I saw |
| 22:00.00 | drmessano | Sure, it works TODAY.. but someone is going to have to maintain it |
| 22:00.09 | drmessano | It's unmanageable, IMO |
| 22:00.38 | drmessano | They're basically going to have to maintain their own Asterisk branch |
| 22:00.46 | drmessano | GFLWT |
| 22:04.17 | file | the person plans on putting it up for review |
| 22:04.37 | file | it's going to take a lot to clean it up and get stuff sorted, will involve PJSIP upstream changes most likely |
| 22:07.55 | jpsharp | Just suck it up and pay for a good provider. Not one that's milking your traffic for ad targeting info. |
| 22:09.21 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
| 22:09.28 | drmessano | SMH |
| 22:09.39 | drmessano | I guess some people think it's worth it |
| 22:10.01 | file | jpsharp: I brought up similar sentiment on DSLReports but was shot down a bit |
| 22:10.32 | drmessano | Well, how dare you suggest people spend money.. |
| 22:10.39 | drmessano | Asterisk is free, the calls should be too |
| 22:11.30 | file | so should my mortgage! |
| 22:12.16 | jpsharp | Only if you use asterisk to handle your mortgage. |
| 22:12.34 | jpsharp | app_paymortgage.so |
| 22:14.08 | drmessano | If you install Asterisk with Increbigasnitron PBX on a VM4Cheap instance, at $10 a year, your PBX is dirt cheap too* |
| 22:14.43 | drmessano | * Note that this provider may go under at any time, for example Aug 17th |
| 22:16.42 | *** join/#asterisk boson (~boson@204.48.27.25) |
| 22:18.53 | drmessano | I love how there is always some new VPS provider that they "Partnered" with to offer one-button installs of Incredigasmic PBX, for like $10 a decade, like it's some big deal.. when in reality it's some basement startup.. |
| 22:19.00 | drmessano | They ask the admins to partner with them, and the guy replies back "Yeah, I think we can do that, but I gotta ask my Mom" |
| 22:20.22 | drmessano | Imagine if people put this kind of effort into useful contributions to Asterisk, and not just trying to fuck Google out of a few bucks |
| 22:21.06 | drmessano | NO TIME TO WORK ON ADDING ____ TO ASTERISK, GOOGLE VOICE STOPPED WORKING!!! CLEAR MY CALENDAR |
| 22:21.10 | drmessano | </rant> |
| 22:22.33 | Kobaz | [2018-07-09 18:21:51] WARNING[24747] db.c: Couldn't bind key to stmt: out of memory |
| 22:22.37 | Kobaz | hmmmmm first time i've seen that |
| 22:23.33 | Kobaz | definitely not OOM |
| 22:23.45 | Kobaz | plenty of sheeps and rams to go around |
| 22:32.00 | jpsharp | ulimits are fine? |
| 22:35.05 | Samot | drmessano: LOL |
| 22:48.36 | Kobaz | no ulimit defined |
| 22:51.58 | *** join/#asterisk sibyakin (~sibyakin@188.162.238.228) |
| 23:09.17 | Kobaz | [2018-07-09 19:08:41] ERROR[22779]: manager.c:6376 process_message: Unable to process manager action 'DBPut'. Asterisk is shutting down. |
| 23:09.18 | Kobaz | ugh |
| 23:09.34 | Kobaz | wow this is really fscked somehow |
| 23:09.45 | Kobaz | certified asterisk 13.21 |
| 23:12.08 | *** part/#asterisk kharwell (kharwell@nat/digium/x-qpgomsrmqgydsmxo) |
| 23:17.46 | Kobaz | oh okay that's nothing to do with DBPut |
| 23:17.55 | Kobaz | i thought it was quiting because of DBPut |
| 23:21.23 | *** join/#asterisk zigggggy (ffssd@unaffiliated/zigggggy) |
| 23:22.32 | *** join/#asterisk saint__ (~saint_@unaffiliated/saint-/x-0540772) |
| 23:23.55 | Samot | https://community.asterisk.org/t/asterisk-exit-automatically/74109/3 |
| 23:24.03 | Samot | Sounds like a memory leak |
| 23:24.51 | Samot | How much memory is in the system? |
| 23:24.59 | Samot | Kobaz: ^^^ |
| 23:25.13 | Kobaz | 4 gigs :( |
| 23:25.21 | Samot | How much CPU? |
| 23:25.29 | Samot | And how many calls/users does this thing handle? |
| 23:25.37 | Kobaz | <PROTECTED> |
| 23:26.06 | Kobaz | voicemail for 600 users, and like a 5 person call center |
| 23:26.10 | Kobaz | so... not much |
| 23:26.26 | Samot | Except for 600 MWI subscriptions. |
| 23:26.40 | Samot | And how many calls? |
| 23:26.52 | Samot | 5 people does not equal 5 calls. |
| 23:27.00 | Samot | You could have 20 sitting in a queue. |
| 23:27.15 | Kobaz | right |
| 23:27.20 | Kobaz | it's a single PRI, so... max 20 |
| 23:27.26 | Kobaz | but generally there's like 2-3 people calling in |
| 23:27.27 | Samot | What's the average? |
| 23:27.37 | Samot | No one sits in the queue? |
| 23:27.39 | Kobaz | it's capped at 20 on the avaya side |
| 23:27.41 | Kobaz | not really |
| 23:28.21 | Samot | What else is this box doing? Recording calls, etc or any other things like AMI/ARI commands from outside the box |
| 23:28.23 | Samot | remote sql? |
| 23:28.33 | Kobaz | local sql |
| 23:28.39 | Kobaz | no call recording |
| 23:28.58 | Kobaz | gonna install valgrind on this thing |
| 23:29.08 | Samot | And how long after restarting asterisk does this happen? |
| 23:29.18 | Kobaz | so i do a 'core reload' |
| 23:29.33 | Kobaz | and within about 10-15 seconds asterisk quits |
| 23:29.55 | Kobaz | Samot: really not much |
| 23:30.19 | Samot | And if you restart the server? |
| 23:30.41 | Kobaz | that's not generally part of my troubleshooting tool kit |
| 23:30.44 | Kobaz | last resort... |
| 23:30.47 | Samot | OK |
| 23:30.51 | Kobaz | but in this case the box has been restarted a number of times |
| 23:30.52 | Samot | So this is a constant issue |
| 23:30.56 | Kobaz | yeah |
| 23:30.56 | Samot | It might be time. |
| 23:31.07 | Samot | OK so it sounds like this box is hosed. |
| 23:31.11 | Kobaz | well no |
| 23:31.24 | Kobaz | we just switched out asterisk, so it's something with this astersk version |
| 23:31.39 | Samot | So you downgraded? |
| 23:31.43 | Kobaz | upgraded |
| 23:32.01 | Kobaz | was 13.17.2 |
| 23:32.23 | Samot | Oh.. |
| 23:32.32 | Samot | So do you have a cert support contract? |
| 23:32.40 | Kobaz | no |
| 23:32.51 | Kobaz | just keeping in line with what was already here |
| 23:35.17 | Kobaz | inherited box |
| 23:35.19 | Kobaz | xorcom system |
| 23:35.22 | Kobaz | fixing bugs in it |
| 23:35.50 | Samot | 2018-01-31 15:40 +0000 [665444b772] Richard Mudgett <rmudgett@digium.com> |
| 23:35.50 | Samot | * manager.c: Fix potential memory leak and corruption. |
| 23:35.50 | Samot | <PROTECTED> |
| 23:35.50 | Samot | <PROTECTED> |
| 23:36.02 | Kobaz | what rev? |
| 23:36.51 | Samot | For you? The one that was just released. |
| 23:36.57 | Samot | That's the downside of cert... |
| 23:37.04 | Samot | We had this issue fixed awhile ago. |
| 23:37.20 | Kobaz | what's the git commit revision id? |
| 23:37.51 | Samot | More details in the changelog for Cert 13 |
| 23:38.17 | Samot | <PROTECTED> |
| 23:38.18 | Samot | I think |
| 23:38.36 | Kobaz | got it |
| 23:38.58 | Kobaz | commit 665444b77281bc7e744534721747429c4671df0b |
| 23:39.05 | Samot | It looks like it was fixed in 13.20 |
| 23:39.17 | Kobaz | this is 13.21 |
| 23:39.25 | Samot | Cert |
| 23:39.33 | Samot | Cert has a limited release cycle. |
| 23:39.38 | Kobaz | it has that change |
| 23:39.45 | Kobaz | do a git log in 13.21, and you'll see commit 665444b77281bc7e744534721747429c4671df0b |
| 23:39.50 | Samot | 2018-06-11 21:09 +0000 Asterisk Development Team <asteriskteam@digium.com> |
| 23:39.50 | Samot | * asterisk certified/13.21-cert2 Released. |
| 23:39.59 | Samot | Cert gets all the fixes in one release. |
| 23:40.24 | Samot | So when a new Cert version comes out, it contains fixes on releases we already had. |
| 23:40.31 | Samot | There was no cert-20 |
| 23:40.39 | Kobaz | right |
| 23:40.49 | Samot | That's why I said, that's the downside. |
| 23:40.53 | Samot | We got the fix in Feb. |
| 23:40.57 | Samot | You got it in June. |
| 23:41.28 | Kobaz | well yeah, in general |
| 23:41.29 | *** join/#asterisk Iamnacho (~Iamnacho@ip68-102-129-52.ks.ok.cox.net) |
| 23:41.38 | Kobaz | i'm more concerned with this partiaular problem |
| 23:44.45 | Samot | Well you upgraded, right? |
| 23:44.46 | Samot | Went away? |
| 23:44.57 | Kobaz | what? no |
| 23:45.03 | Kobaz | the problem appeared |
| 23:45.19 | Samot | What version of 21 did you install? |
| 23:45.25 | Samot | 21-1 or 21-2? |
| 23:45.48 | Kobaz | whatever's in git certified/13.21 |
| 23:46.02 | Kobaz | i'm going to try out mainline 13.21, there's a bit of a difference |
| 23:47.44 | TrickkyTyper | Hey guys waht is the best gateway and sip phone i will use for Linux |
| 23:48.31 | TrickkyTyper | to make calls |
| 23:48.33 | Kobaz | gateway for doing what |
| 23:48.37 | TrickkyTyper | making phone calls |
| 23:48.42 | Kobaz | from what, to what |
| 23:48.52 | TrickkyTyper | from PC to landline |
| 23:48.53 | TrickkyTyper | phone calls |
| 23:48.56 | TrickkyTyper | using Headset to make calls |
| 23:48.59 | TrickkyTyper | im using Debian |
| 23:49.12 | Kobaz | you're doing this all wrong |
| 23:49.22 | Kobaz | it's 2018, use a ITSP |
| 23:49.27 | Kobaz | ~itsp |
| 23:49.27 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 23:49.40 | TrickkyTyper | perfect thats what i wanted to know |
| 23:49.41 | Kobaz | if you REALLY want to use your home copper line |
| 23:49.41 | TrickkyTyper | ITSP |
| 23:49.54 | Kobaz | get like a linksys SPA-FXO ATA |
| 23:49.56 | TrickkyTyper | well i wanna set something up like a telemarketing center |
| 23:50.04 | TrickkyTyper | so i want to be able to set up phone numbers on computers, and allow people to call |
| 23:50.10 | Kobaz | right |
| 23:50.15 | Kobaz | take a number :) |
| 23:50.40 | Kobaz | everyone else and their goldfish is doing contact centers these days |
| 23:50.42 | Kobaz | but anyway |
| 23:50.54 | Kobaz | if you want to start and roll your own, get the book |
| 23:50.55 | Kobaz | ~book |
| 23:50.55 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 23:51.15 | TrickkyTyper | okay but ideally if i want to keep it simple id find a ITSP like you said |
| 23:51.20 | TrickkyTyper | and just set up Sip software on compouter? |
| 23:51.25 | Kobaz | be prepared to invest 6 months to a year, depending on your experience level, to get something worthwhile in place |
| 23:51.30 | TrickkyTyper | or what would that provider call that would allow me to physically make the calls |
| 23:51.33 | TrickkyTyper | from SIP |
| 23:51.40 | TrickkyTyper | i used to use Magic Jack from Windows |
| 23:51.47 | TrickkyTyper | but i went to Linux so i want to replace that company with another gateway |
| 23:51.55 | Kobaz | linux... check out twinkle |
| 23:51.57 | Kobaz | or linphone |
| 23:52.05 | Kobaz | i like twinkle a bit better |
| 23:52.09 | Kobaz | and you can start making calls |
| 23:52.21 | Kobaz | but if you want a whole call center, reports, tracking, metrics... |
| 23:52.27 | Kobaz | you're better off buying something, from what it sounds like |
| 23:53.22 | TrickkyTyper | thank you |
| 23:53.29 | TrickkyTyper | well im just starting a small start up right now |
| 23:53.35 | TrickkyTyper | so i wont be doing something big scale yet. |
| 23:53.43 | TrickkyTyper | but i am a linux geek so i wanted to do it as manual as possible |
| 23:53.58 | Kobaz | allllrightey then |
| 23:54.00 | Kobaz | yeah, start with the book |
| 23:54.02 | Kobaz | go from there |
| 23:54.15 | Kobaz | it's free online, or you can buy the book |
| 23:54.57 | Kobaz | dinner time |
| 23:55.31 | TrickkyTyper | ok ty |