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| 02:16.12 | igcewieling | Wow, AT&T DSL service is $49.95/month for 10Mbps, 18Mbps, 25Mbps, 50Mbps, and 100Mbps. Yes, the same price for all those speeds. |
| 02:16.50 | igcewieling | heh, the 758K, 1.5Mbps, and 5Mbps speeds are the same price. |
| 02:16.58 | Samot | Because it's DSL. |
| 02:17.10 | Samot | DSL is now the new dialup. |
| 02:17.23 | igcewieling | It is still dickish |
| 02:17.32 | Samot | I'm not sure how. |
| 02:18.08 | igcewieling | I have a $10/month 15/5Mbps DSL service as a backup ISP for home. |
| 02:18.28 | Samot | From ATT? |
| 02:18.37 | igcewieling | yup |
| 02:18.51 | Samot | I'm sure it's something that's been there for some time. |
| 02:20.09 | Samot | This isn't a new thing. |
| 02:20.11 | igcewieling | I'm trying to find a better modem, the model I have doesn't have a bridge mode. It hasn't been a problem until I started doing more advances stuff. |
| 02:20.35 | Samot | Amazingly, companies will start to price out old crap to make it unappealing while they try to get rid of it. |
| 02:20.41 | igcewieling | I doubt it is a new thing, but it is still |
| 02:21.02 | igcewieling | they do the same on their fiber service, from the price list I/'m looking at. |
| 02:22.17 | igcewieling | the fiber versions are $10/month cheaper. |
| 02:22.28 | Samot | Yes but is that standalone? |
| 02:22.59 | igcewieling | only the fiber plans |
| 02:26.49 | Samot | So it's $39.95/month for Fiber? |
| 02:27.07 | igcewieling | correct. |
| 02:28.34 | Samot | Well I see that offer but for 12 months with a bundle not including taxes/fees and equipment costs. |
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| 05:47.59 | Ravenheart | morning |
| 05:48.06 | Ravenheart | or evening if you are in the americas |
| 05:50.48 | gry | good afternoon from Australia, Ravenheart |
| 05:51.52 | Ravenheart | juuuust got into the office |
| 05:52.01 | Ravenheart | time to continue my asterisk woes |
| 06:05.02 | wyoung | asterisk is the best! |
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| 06:35.55 | *** join/#asterisk ehsanv (~AdminUser@185.179.171.235) |
| 06:38.43 | ehsanv | Hi, I have a problem in Outgoing Calls in issabel (which is using asterisk), I look through logs and found `Got SIP response 502 "Bad Gateway" back from` |
| 06:39.40 | ehsanv | since i can directly connect to my trunk through my mobile phone and use Outgoing call, i know there is no problem with my service provider |
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| 06:55.48 | *** join/#asterisk Pinchiukas (~keps@unaffiliated/pinchiukas) |
| 06:56.25 | Pinchiukas | If I have a fleet of Asterisks running and want to shut one down, is it possible to transfer all the ongoing calls on it to another instance without disruption? |
| 07:09.13 | Ravenheart | ehsanv, are you using chan_sip or pjsip |
| 07:09.23 | Ravenheart | post a phsip.conf/sip.conf |
| 07:09.33 | Ravenheart | pjsip.conf* |
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| 08:21.17 | khobo | yo wyoung |
| 08:21.37 | khobo | are ports 10,000-20,000 only for video&voice? |
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| 10:04.10 | User738648723 | Hi, is there any way to specify a per caller periodic-announcement? |
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| 10:20.01 | ehsanv | Hi, I have a problem in Outgoing Calls in issabel (which is using asterisk), I look through logs and found `Got SIP response 502 "Bad Gateway" back from xxx.xxx.xxx.xxx:5060` and since i can directly connect to my trunk through my mobile phone and use Outgoing call, i know there is no problem with my service provider |
| 10:22.25 | ehsanv | I need to know is there any way to find out the problem more clearly? I googled a lot and even looked through detailed logs also but don't know what to do :( |
| 10:28.08 | Ravenheart | hi ehsanv |
| 10:28.22 | Ravenheart | are you using chan_sip or pjsip |
| 10:28.31 | Ravenheart | post a pjsip.conf/sip.conf |
| 10:28.53 | Ravenheart | and your dialplan :) |
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| 12:25.28 | Samot | 2:56:25 AM <Pinchiukas> If I have a fleet of Asterisks running and want to shut one down, is it possible to transfer all the ongoing calls on it to another instance without disruption? <<- Nope. Not at all. |
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| 13:27.10 | Pinchiukas | Samot: I take it I can only re-route media with reinvites? |
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| 13:30.00 | Samot | What about the rest of the call? |
| 13:30.28 | Samot | The question was about moving all calls from box A to box B |
| 13:30.34 | Samot | You cannot do that. |
| 13:45.03 | ehsanv | Ravenheart: thanks for your answer. I'm trying to figure out what is my dialplan and what sip i'm using ! :) i'm newbie! |
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| 13:49.18 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 13:52.20 | Ravenheart | ssh into issabel |
| 13:52.26 | Ravenheart | goto /etc/asterisk |
| 13:52.42 | Ravenheart | and checkout pjsip.conf and extensions.conf |
| 13:53.19 | Ravenheart | also from a second ssh you should open the asterisk cli |
| 13:53.25 | Ravenheart | asterisk -rvvvvv |
| 13:53.38 | Ravenheart | and try dialing an outgoing call |
| 13:53.46 | Ravenheart | you should get some more info on whats wrong |
| 13:54.09 | Ravenheart | but if i had to guess you either have a badly configured gateway or your dialplan doesn't know where to call |
| 14:00.32 | Samot | Wow, Issabel finaly got PJSIP support? |
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| 14:01.06 | Ravenheart | doesn't it ? |
| 14:01.23 | Samot | Well as of Oct 2019 the lead dev said no. |
| 14:01.26 | Ravenheart | i only installed it to mess around with the gui |
| 14:01.40 | Samot | Not without it being a paid support thing. |
| 14:01.52 | Samot | Well Issabel is based on FreePBX 2.11 |
| 14:01.57 | Samot | So that should give you some idea. |
| 14:02.18 | Ravenheart | me personally i built from source and did a bunch of experiments |
| 14:02.37 | Ravenheart | got asterisk running with all theb ells and whistles i want in under a day |
| 14:02.45 | Samot | OK. |
| 14:03.08 | Samot | So in the GUI, can you manage Chan_PJSIP? |
| 14:03.21 | Samot | Does it have options to setup extensions or trunks using Chan_PJSIP? |
| 14:07.23 | Samot | Ravenheart: Those were serious questions, I'm curious. |
| 14:07.39 | Samot | When I installed Issabel in early 2019, it didn't have it. |
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| 14:23.49 | ehsanv | Ravenheart: thanks a lot! i found bad configuration. the correct username is 285598 but it was 2855598 in configuration! |
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| 15:44.47 | igcewieling | *sigh* Sales strikes again. Apparently our damn sales rep never even went to the customer site to see if what he was selling them would even work for them. |
| 15:45.24 | khobo | igcewieling: what do you do? |
| 15:45.50 | igcewieling | khobo: I told our tech to do a detailed site survey and kick it back to the sales rep. |
| 15:46.50 | khobo | lol |
| 15:46.55 | igcewieling | Even if I was willing to run our phones on the customer switches, since the phones are PoE and the customer switches are not PoE, we still can't do the install today. |
| 15:47.12 | khobo | =| |
| 15:47.25 | khobo | are you in installation department? |
| 15:47.33 | igcewieling | We normally install Adtran PoE switches for the phones. |
| 15:49.09 | igcewieling | khobo: Depending on your point of view, I'm the neckbeard who is a bit of an asshole and works remote OR I'm the Chief of Technology. |
| 15:49.59 | igcewieling | I prefer the first point of view, it involves fewer phone calls. |
| 15:50.49 | khobo | LOL |
| 15:50.50 | igcewieling | Our techs install the router and switches enough for me to get remote access, then I configure the router and switches. |
| 15:51.08 | khobo | oh that's pretty cool |
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| 15:57.16 | Chainsaw | igcewieling: They'll never let you have that job title. They wouldn't give me Keeper of Legacy Systems, it had to be Senior Systems Engineer. |
| 16:00.36 | igcewieling | I'm officially some kind of engineer. |
| 16:01.00 | Chainsaw | igcewieling: More like VoIP magician. |
| 16:01.19 | Chainsaw | igcewieling: I bet you have a pointy hat. |
| 16:01.50 | igcewieling | Nah, though I had a rubber chicken for a while. |
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| 16:53.09 | khobo | https://imgur.com/SbpwxSo |
| 16:53.14 | khobo | how do you disable trunks? |
| 16:53.23 | khobo | i clicked the edit setting and didn't see anything within that |
| 16:54.26 | khobo | ignore this |
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| 16:57.31 | Samot | Wow, 32 pages of Chan_SIP trunks. |
| 16:57.42 | Samot | er 32 Chan_SIP trunks. |
| 16:58.29 | Samot | Probably could be trimmed down to like 3-4 Chan_PJSIP trunks. |
| 17:03.17 | khobo | https://pastebin.com/2u7qSVc1 |
| 17:03.20 | khobo | here's what my iptables looks like |
| 17:05.13 | khobo | WARNING: Server access locked down to server IP address and your current IP address. |
| 17:05.18 | khobo | i've read two different things about how this comes configured... ^ |
| 17:05.37 | khobo | the other thing i've read which i feel is conflicting is that anythign on my LAN gets auto accepted |
| 17:05.58 | khobo | looking at my iptables ACCEPT all -- 192.168.0.0/16 0.0.0.0/0 |
| 17:06.07 | khobo | i would guess that that's true? |
| 17:07.30 | Samot | If this is IncrediblePBX you need to go to their forums for support. |
| 17:08.06 | Samot | Someone want to relay that. |
| 17:09.16 | drmessano | lol |
| 17:09.49 | Samot | Well not you. |
| 17:10.03 | Samot | Because someone would have to relay you relaying what I said. |
| 17:10.26 | drmessano | I haven't a care in the world for this one |
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| 17:42.35 | igcewieling | Links to things like Polycom manuals, SIP paging units, diagrams, etc. Might be helpful to some people: https://help.nyigc.com/ |
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| 19:16.37 | Pinchiukas | Samot: you cannot think of a scenario where that would be possible? |
| 19:17.02 | Samot | Not with Asterisk |
| 19:18.07 | Samot | It is not possible |
| 19:26.38 | Pinchiukas | Not even with having some smartass LB in front? :) |
| 19:29.45 | khobo | https://dpaste.com/C4FNWY8U9 |
| 19:29.54 | khobo | any idea what's going on in here? |
| 19:38.58 | Samot | Pinchiukas: no |
| 19:39.23 | Samot | Pinchiukas: You cant move all the channel(s) data over |
| 19:40.59 | Samot | I have an idea of whats going on with that pastebin |
| 19:41.33 | [TK]D-Fender | khobo, they don't like your call. enable SIP debug and actually look at the packets going out |
| 19:42.01 | khobo | will do! thanks |
| 19:44.12 | Pinchiukas | Samot: I admit to being pretty ignorant but was hoping doing reinvites in some way might achieve what I wanted. |
| 19:44.34 | Samot | It will not |
| 19:46.08 | *** join/#asterisk sa02irc (~mbax@155-079-043-212.ip-addr.inexio.net) |
| 19:47.17 | Pinchiukas | So how do people usually drain calls from an Asterisk instance? Either wait for all of them to complete after removing from a LB or just break them? |
| 19:47.31 | Samot | They break |
| 19:47.47 | Samot | Asterisk is a Back2Back User Agent |
| 19:47.56 | Samot | One call = 2 channels |
| 19:48.02 | Pinchiukas | Well that's not nice is it? :) |
| 19:48.09 | khobo | annnnd that's a bunch of debug |
| 19:48.43 | Samot | So call details, variables, etc are associated to those channels |
| 19:48.57 | Samot | There is no clean way to move everything |
| 19:49.18 | Pinchiukas | How about an unclean one? :) |
| 19:49.40 | Samot | The calls will drop |
| 19:49.57 | Samot | To do what you want requires spending money |
| 19:50.09 | kjetilho | Pinchiukas: DNS failover + wait for people to stop talking. and if they are still talking after 15 minutes, they need a break! |
| 19:50.14 | Samot | Because it is costly |
| 19:50.16 | Pinchiukas | Pretty much everything requires that. |
| 19:51.08 | Pinchiukas | kjetilho: can you elaborate what you mean by failover and 'wait for people to stop talking'? |
| 19:51.30 | kjetilho | Pinchiukas: move the service name to point to a different IP address (and server) |
| 19:51.51 | kjetilho | look at active channels. start maintenance when there are none |
| 19:52.05 | kjetilho | obviously standard DNS TTL issues apply |
| 19:52.18 | Samot | That isnt what is being asked |
| 19:52.28 | Samot | Because it still drops calls |
| 19:53.14 | kjetilho | what do you mean? |
| 19:53.29 | Pinchiukas | Well it might kinda sorta be but ops people tell me there are people that have calls hanging in there for days. |
| 19:53.39 | Samot | Take active calls from box a and put them on box b |
| 19:53.53 | Pinchiukas | So this is me looking for an ideal solution to all of the problems. :) |
| 19:54.26 | kjetilho | yes yes, I just described my best effort to reduce inconvenience of maintenance |
| 19:54.48 | Samot | Being able to move active SIP calls between systems is a costly endeavor. |
| 19:55.23 | Pinchiukas | This is making me think it's possible. :) |
| 19:55.44 | kjetilho | Pinchiukas: how? |
| 19:55.50 | khobo | got it!!! |
| 19:55.55 | khobo | was about the callerID |
| 19:55.59 | khobo | thanks for your help |
| 19:56.02 | khobo | outbound works!! |
| 19:56.08 | khobo | everything works |
| 19:58.14 | Pinchiukas | kjetilho: as I mentioned, I'm pretty ignorant of how things work. But my hope is that there is a way to transfer all the relevant state and renegotiate the media streams on the fly. |
| 19:58.56 | kjetilho | the protocol is not well suited for that |
| 19:59.13 | kjetilho | it is designed by phone people, not Internet people |
| 19:59.41 | kjetilho | (ok, it could be much worse :) |
| 20:01.05 | Pinchiukas | That's fine, I was looking a way to hack things together. |
| 20:03.01 | Samot | Well what you are looking to do isn't something you really hack together. |
| 20:54.50 | khobo | Format: "caller name" <#######> |
| 20:57.30 | igcewieling | leave off the quotes |
| 21:00.29 | *** join/#asterisk drathir_tor (~drathir@gateway/tor-sasl/drathir) |
| 21:07.02 | khobo | thanks! |
| 21:07.07 | khobo | still didn't push... weird |
| 21:07.23 | khobo | i should be able to dial from poly to my cellphone and my cellphone should show "name & number" as the caller id, yes? |
| 21:07.37 | khobo | i set it correctly in voip.ms interface |
| 21:07.40 | khobo | must be freepbx side |
| 21:25.12 | khobo | hmmm weird!! |
| 21:25.18 | khobo | i see it being passed in my invite |
| 21:25.25 | khobo | when i turn debug on |
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| 21:46.32 | jsmith | khobo: At least in the US, the caller-id name isn't passed along to the phone... the cell phone company does a database 'dip' to get the name associated with a number. |
| 21:46.52 | jsmith | khobo: So you can change the name in the INVITE all you like, it's not going to make that name appear on your cell phone. |
| 21:46.58 | khobo | ahhhhhhh |
| 21:47.06 | khobo | any idea how i can test this to make sure it's passing along properly? |
| 21:48.29 | sibiria | make a call to a DID pointing back to your asterisk setup |
| 21:48.53 | sibiria | and take a look at the invitation |
| 21:48.59 | sibiria | or just call locally if you have that option |
| 21:49.29 | khobo | good idea! |
| 21:49.30 | khobo | thanks |
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