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| 14:25.43 | Someone_Else | is it possible to keep a aor in the database (astdb) when a extension disconnects? |
| 14:26.29 | Someone_Else | in chan_sip, after a extension disconnects, it is still possible to retrieve the sip_uri in SIP/Registry/{ext} |
| 14:28.33 | file | contacts are stored in the astdb by default and persisted across restarts and such. |
| 14:28.53 | file | they exist for the lifetime that the contact exists |
| 14:30.24 | Someone_Else | file: is it possible to persist until the extension registers again? |
| 14:30.39 | file | no |
| 14:30.42 | Samot | It's persistent until the Expire is hit. |
| 14:30.51 | file | indeed |
| 14:30.56 | Someone_Else | mmm |
| 14:31.00 | Samot | If you tell the contact to expire in 60 seconds, that's how long Asterisk will hold it |
| 14:31.04 | Samot | That's standard. |
| 14:31.09 | Samot | Not just Asterisk. |
| 14:31.11 | Someone_Else | is it possible to log register events? |
| 14:31.27 | Samot | It logs them already |
| 14:31.38 | Someone_Else | no, it doesn't |
| 14:31.53 | Someone_Else | in the full log I don't see any endpoint reachable blah |
| 14:31.56 | Someone_Else | only in the console |
| 14:32.16 | Samot | Then you're logging isn't at a high enough verbosity |
| 14:32.54 | Someone_Else | are those registered events notice? |
| 14:33.00 | Someone_Else | or debug? |
| 14:33.07 | file | verbose. |
| 14:33.25 | Samot | When you log into the console and do "core set verbose 10" |
| 14:33.39 | Samot | Then your full log will have level 10 logging during that time |
| 14:33.54 | Samot | When you exit it goes back to the default setting in the logger config |
| 14:34.28 | Someone_Else | the point is, sip_uri contains the push token to wake up the device prior to calling |
| 14:34.55 | Someone_Else | so the only way I see to make this work is dump the register lines into a log, parse it and save it to the database |
| 14:35.03 | Someone_Else | *a database |
| 14:35.18 | Samot | So you're trying to get headers? |
| 14:35.27 | Someone_Else | yes |
| 14:35.49 | Samot | file: You can't parse headers on a REGISTER can you? Just the INVITE, right? |
| 14:36.02 | file | nothing happens on a REGISTER to be able to act on |
| 14:36.16 | Samot | Someone_Else: So that's going to be an issue. |
| 14:36.22 | file | may go out over AMI, I forget |
| 14:37.28 | Someone_Else | Samot: many VoIP apps work that way: https://www.zoiper.com/en/support/answer/for/windows-phone/105/Push_notifications |
| 14:37.52 | Samot | I understand how push notices work. |
| 14:38.07 | Samot | That doesn't change the fact you can't parse the headers on a REGISTER in Asterisk. |
| 14:38.25 | Samot | So if you're trying to parse the token from the URI in a header, you can't do it on a REGISTER in Asterisk. |
| 14:38.39 | Someone_Else | Samot: logging it is, then.... |
| 14:39.13 | Samot | Or you use something like Kamailio in between your users and the Asterisk box. |
| 14:39.18 | file | you can get the contacts and contact URI in dialplan... |
| 14:39.23 | file | PJSIP_AOR and PJSIP_CONTACT |
| 14:39.26 | Samot | Because you can tear apart a SIP message at any level there. |
| 14:40.01 | Someone_Else | and what I tried to say is that they benefit from the fact that chan_sip doesn't remove the sip_uri after a disconnect |
| 14:40.17 | Samot | It will. |
| 14:40.34 | file | PJSIP doesn't remove the SIP URI unless it is asked to remove it, or the expire happen |
| 14:40.50 | file | so "after a disconnect" depends on what exactly that inherently means |
| 14:41.02 | Someone_Else | file: the client immediately disconnects after being used to save battery life |
| 14:41.26 | file | if it explicitly does not remove the Contact, then the Contact remains until the expiration time |
| 14:41.32 | file | it would go unreachable, but it would still exist |
| 14:41.54 | file | if it does not re-register, then it would indeed go away |
| 14:42.04 | Someone_Else | file: the last one is the case |
| 14:42.17 | Samot | Because the app is asleep |
| 14:42.30 | Samot | Or off to conserve battery. |
| 14:42.56 | Someone_Else | but I'm not fond of a Kamilio before Asterisk setup, too much for such a simple task (get the sip_uri) |
| 14:42.57 | Samot | If the app being backgrounded doesn't send REGISTERS or respond to keep alives |
| 14:43.09 | Samot | OK. |
| 14:43.10 | file | mobile is not a simple task. |
| 14:43.16 | Samot | There's that. |
| 14:43.46 | Someone_Else | file: depends on how you look at it |
| 14:43.53 | file | no, it's really not |
| 14:44.04 | file | to make a truly good mobile experience is not a simple task |
| 14:44.08 | Samot | Mobile devices background apps. |
| 14:44.20 | Samot | Most the apps don't respond to SIP message when in that mode |
| 14:44.32 | Someone_Else | file: can you explain that a bit further? |
| 14:44.33 | Samot | This is why you have a push service. |
| 14:45.00 | Samot | Someone_Else: Most softphone clients have issues with backgrounding. |
| 14:45.04 | file | nothing is truly guaranteed in mobile, so you can have lots of areas where the experience falls short |
| 14:45.10 | Samot | Someone_Else: They don't respond to keep alives |
| 14:45.13 | file | this push aspect, communicating with the softphone, audio, networking |
| 14:45.15 | Samot | It causes missed calls |
| 14:45.22 | Samot | Or features not working |
| 14:45.40 | Someone_Else | Samot: my client works fine as far as I tested it |
| 14:45.49 | Samot | What client is that? |
| 14:45.55 | Someone_Else | Linphone |
| 14:46.05 | Samot | Is this the same client you just said will not re-register? |
| 14:46.13 | Samot | And goes unreachable? |
| 14:46.18 | Someone_Else | yes |
| 14:46.25 | Samot | So it works "fine"? |
| 14:46.32 | Samot | This is why I use Bria. |
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| 14:47.04 | Samot | My Bria app has been running in the background on my phone for over a week without being opened. |
| 14:47.11 | Samot | I still get calls, it's registered without issue. |
| 14:47.15 | Samot | It responds to keepalives. |
| 14:47.31 | Someone_Else | Samot: Linphone can do the same, if you don't mind draining your battery |
| 14:47.36 | Someone_Else | that's where push is for |
| 14:47.44 | Someone_Else | same goes for Bria |
| 14:47.49 | Samot | It doesn't drain my battery |
| 14:48.07 | Samot | I haven't had to change my charging or battery powering at all. |
| 14:48.11 | file | they take care of that, and of making a good experience |
| 14:48.18 | Samot | Correct. |
| 14:48.38 | Someone_Else | wait a sec, is it like groundwire where you provide them access to your sip account? |
| 14:48.49 | Samot | Nope. |
| 14:49.05 | Samot | I do use the Stretto platform. |
| 14:49.13 | Samot | I have push servers/services I could use |
| 14:49.17 | Samot | But I don't use them |
| 14:49.23 | Samot | Because I was testing that |
| 14:49.23 | Someone_Else | does Bria use push? |
| 14:49.29 | Samot | It can if I tell it to |
| 14:49.38 | Samot | Again, I have the Stretto platform. |
| 14:49.52 | Samot | It gives me everything I need. |
| 14:50.00 | Samot | Including provisioning of the clients. |
| 14:50.17 | Samot | It auto provisions the client based on their account login. |
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| 14:52.32 | Someone_Else | one thing I don't get... |
| 14:52.43 | Samot | Just one thing? |
| 14:53.06 | Samot | Sorry, not often I can quote Clue. |
| 14:53.25 | Someone_Else | why all this talk of push (because, the conventional wisdom is that keeping the app registering drains the battery) |
| 14:53.31 | Someone_Else | as this is no issue at all? |
| 14:53.47 | Someone_Else | I mean, the point is not that it didn't work... it works, but drains the battery |
| 14:53.52 | Someone_Else | hence the push solution |
| 14:53.58 | Samot | Right. |
| 14:54.12 | Samot | And sometimes its used for more than just that. |
| 14:54.30 | Samot | See if you move from network to network your SIP server might keeps the old location and just register a new one |
| 14:54.41 | Samot | By using the push server, you always are at the same IP. |
| 14:54.56 | Samot | Because the client is going to the push system and that's dealing it with it. |
| 14:55.10 | Someone_Else | by the way, how many data is consumed by those registers? |
| 14:55.18 | Samot | Minimal. |
| 14:55.22 | Someone_Else | hmmm |
| 14:55.58 | Someone_Else | that IP thing isn't a issue when you use max_contacts and remove_existing |
| 14:56.00 | Samot | See Asterisk was being in the game. |
| 14:56.13 | Samot | Until PJSIP it could support 1 contact per peer. |
| 14:56.22 | Samot | Unlike almost all the other SIP servers out there. |
| 14:56.34 | Samot | That could support multiple contacts per AOR. |
| 14:57.17 | Samot | Someone_Else: I have my desk phone and my mobile app. |
| 14:57.22 | Samot | I can't have just 1 contact. |
| 14:57.28 | Samot | Would break my setup. |
| 14:57.58 | Someone_Else | I would just use 2 extensions |
| 14:58.04 | Samot | Why? |
| 14:58.09 | Samot | Now you're complicating things. |
| 14:58.27 | Samot | So you don't want to use Kamailio for such a simple thing.. |
| 14:58.29 | Someone_Else | how about calling home using a internal address, why not? |
| 14:58.38 | Samot | OK. |
| 14:58.48 | Samot | So you're talking home user level stuff. |
| 14:58.53 | Someone_Else | yep |
| 14:59.32 | Someone_Else | I will disable the push notification thing and see how that plays out with my battery |
| 14:59.55 | Someone_Else | if not, I can always re-enable it and find a workaround for the sip_uri |
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| 15:23.56 | Someone_Else | is it possible to pipe a asterisk log? |
| 15:24.16 | Someone_Else | so, instead of writing it to a file, redirecting it to a program? |
| 15:24.26 | igcewieling | yes. |
| 15:24.30 | igcewieling | sort of |
| 15:24.38 | Someone_Else | how? |
| 15:24.43 | igcewieling | have asterisk log to syslog, then handle it with syslog |
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| 16:58.25 | igcewieling | "It was originally ordered as a 3MB fiber but due to build out costs, $6,520, it was changed to a T1" I have Verizon |
| 16:58.32 | igcewieling | I HATE Verizon |
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| 16:59.46 | MLC | I had some carriers quoting us $50K build out costs for fiber. They would have had to tunnel under a freeway. |
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| 17:32.52 | bkervaski | Man, we have it so good here. 400mb business with 50mb up is only $100/month. We all have business connections at our homes now. |
| 17:33.15 | bkervaski | No love for any of these carriers. |
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| 17:38.30 | igcewieling | bkervaski: you must be outside the USA |
| 17:38.39 | bkervaski | Alabama |
| 17:38.57 | igcewieling | Huntsville? |
| 17:39.04 | bkervaski | Birmingham |
| 17:39.13 | bkervaski | Huntsville is getting huge. |
| 17:39.27 | igcewieling | wow, impressive. I used to live in Huntsville and near Gadsden. |
| 17:39.28 | bkervaski | I'm actually in Trussville, east of Birmingham. |
| 17:39.57 | bagira | hail from ohio |
| 17:40.00 | bkervaski | I fly up to the Gadsden airport all the time! Got a buddy who paints airplanes up there. How long ago? |
| 17:40.01 | bkervaski | hail! |
| 17:40.51 | bagira | is that for the ANG? (painting planes) |
| 17:40.52 | igcewieling | Maybe 5 years ago or so. I moved to Pensacola when I got tired of tornadoes every day. |
| 17:41.14 | bkervaski | lol |
| 17:41.40 | bkervaski | bagira .. ANG? |
| 17:42.23 | bagira | air national gaurd |
| 17:42.39 | bkervaski | oh .. no .. private |
| 17:42.45 | bagira | 'Merica Reserve, Brown Label |
| 17:43.12 | bagira | nice |
| 17:43.21 | bagira | i love driving through bammer |
| 17:43.31 | bagira | never get a chance to get out and do stuff there though |
| 17:43.49 | bkervaski | When I was getting my private I got in the pattern up there with two guard fighter jets .. scared the heck out of me .. thought I accidentally flew to a military airport |
| 17:44.07 | bagira | lol |
| 17:44.47 | bagira | so what are you guys doing with asterisk? im brand new to it so still getting a feel for the use cases. looks like it does alot. |
| 17:52.30 | igcewieling | I have a lot of customers with FreePBX, but others have an Adtran box to convert analog or PRI to SIP, then send it to some non-freepbx servers I use for call routing. |
| 17:53.10 | igcewieling | last count was around 75 FreePBX servers and 6 non-FreePBX boxes, 4 for call routing, 2 for our hosted service. |
| 17:53.48 | igcewieling | maybe close to 200 of the Adtran customers. |
| 17:54.10 | MLC | We use asterisk on-premise for our call center phones. Toll free number from voip.ms to that server. Lot's of integration between asterisk and our application software. |
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| 18:19.30 | MLC | Is there a way in a pjsip type = identify section to match on the server's port number? I'm experimenting with using an alternate port at voip.ms to make use of both my WAN connections. . |
| 18:22.26 | MLC | the "match" statement does not allow a port number |
| 18:22.42 | Samot | No. There isn't. |
| 18:24.14 | MLC | Maybe it doesn't matter? I have two endpoints pointing to the same voip.ms server, one on port 5060 and the other on 5080. An inbound call from the 5080 account matches the identify for the 5060 account, but seems to work just fine. |
| 18:31.52 | Samot | Do they go to the same context? |
| 18:31.58 | MLC | yes |
| 18:32.06 | Samot | Then you don't need two trunks. |
| 18:32.26 | Samot | They just tell Asterisk where calls can and cannot come from/go to. |
| 18:33.14 | MLC | I'm hoping to be able to route one trunk out WAN1 and the other out WAN2, so that if/when WAN1 goes down new calls will come in on the WAN2 link. Is there a better way to accomplish that? |
| 18:35.09 | MLC | with the firewall I can force the traffic out different WANs based on the port number. voip.ms has a "failover" feature that will try a second "account" (= trunk) if the primary is unreachable. |
| 18:37.56 | bagira | so what my intent is |
| 18:38.10 | bagira | i've got an asterisk server in a vm compiled with sip support |
| 18:38.19 | bagira | and i have signed up for a trunk service at gotrunk |
| 18:38.33 | bagira | and on my host machine i will run a softphone of some kind (linux) |
| 18:38.40 | bagira | and use that for inbound and outbound |
| 18:39.13 | bagira | having difficulty getting all the pieces working together |
| 18:44.08 | MLC | bagira: what kind of trouble are you having? |
| 19:13.06 | igcewieling | Whoo! Whoo! I just tested some network failover and we didn't even drop the test call. |
| 19:13.32 | MLC | Nice. What kind of equipment do you have that makes that black magic possible? |
| 19:14.14 | igcewieling | MLC: Me? OSPF and GRE tunnels. 8-) |
| 19:14.39 | Samot | MLC: I don't think he was referring to it at the Asterisk level. |
| 19:14.54 | igcewieling | MLC: I control the router on the carrier side of the circuit so that sort of stuff is possible. |
| 19:15.09 | MLC | got it. Black magic indeed! |
| 19:15.23 | igcewieling | SSH session didn't even drop. |
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| 21:59.51 | hecline | I'm not sure if it's okay to ask here, but I have a Digium switchvox API and I'm trying to create a call rule+set for an extension using their API, but it appears the API doesn't have a method for that. Anyone know if I'm missing something? The documentation is here http://developers.digium.com/switchvox/wiki/index.php/WebService_methods |
| 22:05.22 | bkervaski | Asterisk 16.latest ... pjsip ... some sip registrations get an ";rinstance=" identifier, some do not. Looks like Polycom phones do not but others do. Curious if I'm missing something obvious here. Any ideas? |
| 22:06.38 | file | it's up to them to do it, and Asterisk itself does nothing with that information |
| 22:07.32 | bkervaski | Thanks. The issue I seem to be having is that the contacts with an rinstance will continue to stay registered whereas coicidentally I guess the ones without will not renew their registration. Digging into the debug now. |
| 22:09.40 | file | that is not the cause |
| 22:11.01 | bkervaski | Yea, I understand. Seems that only the last contact that registered is the only one allowed to update. |
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| 22:17.04 | kharwell | hecline: not sure, but my guess is rules are added through methods specifically related to a "subsystem". So for instance if you add a distinctive ring rule for a user using the api method for that it might add it to a set for the user? |
| 22:17.45 | kharwell | can always try it as a test and then call the rulesset get list to see what shows up? |
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| 22:20.58 | hecline | switchvox.users.distinctiveRing.rules.add you mean? |
| 22:22.50 | kharwell | yes, but that is just one example, and again not sure if that it how it works |
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| 22:23.16 | hecline | mhm. i just ran that method on an extension and got nothing of value unfortunately |
| 22:24.35 | kharwell | hrm I see too now it has it's own getList method. |
| 22:24.57 | kharwell | have you tried just issuing an update? Maybe it's an update/add action, so if not found it adds it vs updating the record? |
| 22:25.07 | hecline | it's so strange that there's no switchvox.users.callRuleSets.add |
| 22:25.29 | hecline | one second |
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| 22:26.34 | hecline | no, i can't. it requires a call_rule_set_id |
| 22:27.24 | *** join/#asterisk MLC (~MLC@63.249.40.11) |
| 22:34.11 | hecline | 'A call rule set with that id does not exist or can not be accessed by your account.' |
| 22:45.02 | jsmith | hecline: You can always ask questions... but just to set expectations appropriately, the extended API isn't really supported for external consumption |
| 22:46.40 | hecline | yeah i know. i asked digium directly for support and they said it's not supported |
| 22:47.03 | hecline | really unfortunate for my case, i have quite a few extensions i need to create and didn't want to manually create unique call rules for each |
| 23:13.19 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |