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| 02:06.10 | Katty | i recommend chocolate. |
| 02:06.21 | Katty | maybe some chocolate syrup. |
| 02:06.30 | Katty | on top of a chocolate cupcake. |
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| 02:26.57 | [Consultant]C-As | hi, i am having issues actually disconnecting a call.. the Hangup() emediattly brings me to the h extenstions but the caller on the other ends does not get a disconnect. i have tested it on alot of differnt phones, what on asterisk can cause it.. seems to be only on external trunk calls , through any provider |
| 02:27.29 | [Consultant]C-As | its getting in the way of my origination() |
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| 09:57.14 | tparcina | Sometimes my phones (if not all, than most of them) lose connection to the Asterisk server. |
| 09:58.43 | tparcina | In notice I see that they are Lagged, TOO LAGGED or UNREACHABLE, and after short period of time (form few seconds up till one minute), they connect again. |
| 09:59.53 | tparcina | They usually louse connection (or become lagged) in period of several seconds, and they usually reconnect around the same time (most of them in the same second). |
| 10:00.07 | tparcina | Server is virtual on XenServer. |
| 10:00.36 | tparcina | How to check is the problem in GNU/Linux server, XenServer or network itself? |
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| 11:02.55 | sekil | is there any way to play with auth parameters when challenged on INVITE, in the dialplan? |
| 11:10.46 | wdoekes | play? no, but you can pass user/pass in the Dial string |
| 11:11.40 | wdoekes | and can use the auth= parameter in sip.conf to set different responses based on the realm |
| 11:12.55 | sekil | wdoekes: I have single trunk...but I'd like to send different user/pass when challenged ...like "catching" them from db |
| 11:13.18 | sekil | wdoekes: the alternative is to create many sip trunks in sip.conf which bores me ..and it's not nice |
| 11:14.20 | *** join/#asterisk Quo-fan (quofan@unaffiliated/quo-fan) |
| 11:14.45 | Quo-fan | I have asterisk set to listen to my servers public ip on udp port 5060 |
| 11:15.15 | Quo-fan | but nmap -sU -p 5060 ip says the port is open|filtered |
| 11:15.27 | Quo-fan | what the hell is wrong with it |
| 11:16.09 | WIMPy | Nothing? |
| 11:17.11 | Quo-fan | ok this is not working |
| 11:17.17 | Quo-fan | thx for nuttin |
| 11:17.18 | Quo-fan | bye |
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| 11:18.09 | WIMPy | Connection status information shouldn't be trusted on UDP ports for obvious reasons. |
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| 11:41.17 | wdoekes | haha |
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| 11:42.25 | RadJackson | Hi ,is there any cool desktop software or web app to write dialplan code? other than Virtual DialPlan. |
| 11:45.00 | sekil | wdoekes: to what were you referring about auth in Dial? |
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| 12:00.52 | YellowS4 | Hi everyone! |
| 12:01.54 | YellowS4 | I wasn't able to find if it was possible to have a trunk to multiple IPs in asterisk |
| 12:01.58 | YellowS4 | can we? |
| 12:04.19 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:04.45 | YellowS4 | s/trunk/SIP trunk/ |
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| 12:24.50 | [TK]D-Fender | YellowS4: Multiple IP's for what? |
| 12:25.28 | YellowS4 | [TK]D-Fender, Multiple IPs for the client of the trunk |
| 12:26.16 | [TK]D-Fender | for INBOUND matching sure. not sure about outbound. |
| 12:26.24 | [TK]D-Fender | And that's only using chan_pjsip |
| 12:28.28 | file | Fun fact: chan_pjsip also has the ability to associate an inbound call with the registration and direct it accordingly, without doing IP based matching - although it's dependent on the remote server not modifying your Contact header, but not many do |
| 12:29.25 | YellowS4 | The asterisk is a gateway between trunk clients and SIP Transit providers, one of the clients might be set up as an active HA cluster with a different IP on each node of the cluster, but all nodes MUST have access to the trunk at the same time |
| 12:29.55 | YellowS4 | but if it doesn't work we'll just rely on datacenter HA and Virtualization to achieve the HA |
| 12:30.38 | sekil | file: peer line in register string can't help with associating inbound call? |
| 12:31.07 | file | in chan_sip? no, it doesn't have the functionality I'm referring to |
| 12:32.14 | sekil | file: so what is peer for then? to use the user/password/auth parameters from peer config? |
| 12:32.23 | sekil | file: in the registration |
| 12:32.38 | YellowS4 | file, you are saying that it would be able to send an invite to multiple IPs when it receives an Inbound call on a trunk that has to be redirected to an other trunk which is used by multiple nodes ? |
| 12:32.42 | [TK]D-Fender | sekil: Registration has nothing to do with matching an inbound call |
| 12:33.04 | [TK]D-Fender | YellowS4: There is no such thing as "redirect". |
| 12:33.37 | sekil | [TK]D-Fender: ok.. |
| 12:33.44 | file | YellowS4, oh you mean outgoing? anything can do that if you configure stuff... in the case of chan_sip that's multiple peer entries |
| 12:34.06 | YellowS4 | [TK]D-Fender, I used the wrong term, send an invite to multiple nodes and bridge to the one that answers the call |
| 12:35.11 | sekil | [TK]D-Fender: "If you want to control where the call enters your dialplan, which context, you want to define a peer with the hostname of the provider's server" |
| 12:35.20 | sekil | [TK]D-Fender: is this what is peer for? |
| 12:35.23 | [TK]D-Fender | file: PJSIP can dial multiple contacts associated with a "peer" automatically, right? |
| 12:35.30 | YellowS4 | file, It has to handle incoming and outgoing calls, since it is used only as gateway for trunks and there's no end devices directly registering to it |
| 12:35.35 | file | [TK]D-Fender, if you use the dialplan function - yes |
| 12:36.02 | [TK]D-Fender | YellowS4: So PJSIP can do what you're looking for |
| 12:36.05 | file | or chan_sip can if you create multiple peers |
| 12:36.12 | YellowS4 | [TK]D-Fender, multiple contact is not the solution, it's one contact with multipleIPs |
| 12:36.24 | [TK]D-Fender | file: YellowS4 Each IP *is* a contact |
| 12:36.26 | file | that's not what a contact is |
| 12:36.38 | file | yes, each target is a contact |
| 12:39.11 | YellowS4 | <PROTECTED> |
| 12:39.34 | YellowS4 | but only one trunk account for all the nodes, defined by domain name |
| 12:40.12 | [TK]D-Fender | YellowS4: Yes. Now go read up on PJSIP and get working on it |
| 12:40.34 | YellowS4 | [TK]D-Fender, ok thanks :) |
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| 12:43.02 | sekil | wonders about peer in register |
| 12:43.28 | [TK]D-Fender | there is no 'peer' is 'register'. |
| 12:43.33 | [TK]D-Fender | They are separate |
| 12:43.48 | [TK]D-Fender | I can register to a server and have NOTHING defined to match and ID them when they send me a call |
| 12:44.07 | [TK]D-Fender | I can have a peer defined so that I can dial them, and match them when they send me a call and NEVER register |
| 12:44.19 | [TK]D-Fender | these are 1000% separate things |
| 12:45.09 | sekil | yes I agree...I'm asking about the peer in register line string in the example in sip.conf |
| 12:45.32 | sekil | is there some use of it and if is what is it? |
| 12:45.53 | sekil | ; Format for the register statement is: Â Â Â Â Â Â register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] |
| 12:45.53 | sekil | <PROTECTED> |
| 12:45.56 | sekil | sorry |
| 12:49.09 | [TK]D-Fender | sekil: Unsure. On a guess I might think that it would use the settings of a defined peer for the registration. |
| 12:49.46 | sekil | [TK]D-Fender: which helps a lot...I don't have to put the params twice then..thank you..will test it |
| 12:50.23 | YellowS4 | [TK]D-Fender, only in the asterisk world, on other PBXs you can define who can access a trunk by registration AND hostname/IP |
| 12:50.57 | [TK]D-Fender | YellowS4: You are mixing terms & context up |
| 12:51.04 | [TK]D-Fender | YellowS4: Those are all CONTACTS |
| 12:51.21 | [TK]D-Fender | YellowS4: And do not say "access a trunk". This is already a bad term to start with |
| 12:51.50 | [TK]D-Fender | YellowS4: You are MATCHING an inbound contact. How many you can have, and how they are defined is an entirely different matter |
| 12:53.11 | YellowS4 | [TK]D-Fender, what defines a contact then? |
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| 12:53.34 | [TK]D-Fender | [08:40][TK]D-FenderYellowS4: Yes. Now go read up on PJSIP and get working on it <--- |
| 12:53.48 | [TK]D-Fender | AOR <- |
| 12:55.50 | *** join/#asterisk doobeh (~doobeh@ergo.io) |
| 12:55.59 | YellowS4 | [TK]D-Fender, it begins to make sense, step by step |
| 12:58.03 | doobeh | my system is having a bit of an issue detecting when the remote caller hangs up, I'm using an astribank via FXO ports, any suggestions where to look? |
| 13:00.02 | tzafrir | Does the line have any type of disconnect supervision? |
| 13:00.56 | doobeh | I'm not sure to be honest-- but that I guess is a country/provider specific thing, so I'll have a dig into the settings relating to that-- thanks :) |
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| 13:04.17 | sekil | [TK]D-Fender: works fine |
| 13:04.31 | sekil | [TK]D-Fender: takes the credentials from the peer configured |
| 13:05.54 | sekil | [TK]D-Fender: so one don't need to specify username and secret in the string |
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| 13:10.16 | wdoekes | sekil: SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port] |
| 13:11.11 | sekil | wdoekes: thanks |
| 13:14.59 | [TK]D-Fender | wdoekes: that's Dial syntax.... |
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| 13:28.22 | madduck | with the automixmon feature beautifully working, I wonder how people handle this in a multi-user-scenario? Ideally, if an extension starts it, I'd like the result to be available in their voicemail or mabe sent as an email to an address associated with the extension? |
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| 13:31.38 | [TK]D-Fender | E-mail is your job as is moving it to some other folder like voicemail as well as creating the .txt file it requires as well... |
| 13:31.58 | [TK]D-Fender | madduck: Recording itself has nothing to do with anything else without you coding all of it |
| 13:33.00 | madduck | i know recording is not sending, hence I wondered how other people handle it |
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| 13:33.21 | madduck | for a start, I |
| 13:33.25 | [TK]D-Fender | madduck: Call a script to send an e-mail if that's what you want to do |
| 13:33.35 | madduck | I'd wonder how to reliably identify the SIP entity that initiated the recording. |
| 13:33.44 | [TK]D-Fender | Many people jsut leave the files in whatever folder they told it to save in in the first palce |
| 13:37.59 | *** part/#asterisk mjordan (~mjordan@75.76.55.191) |
| 13:38.22 | wdoekes | [TK]D-Fender: am I allowed to answer questions? like 13:45 < sekil> wdoekes: to what were you referring about auth in Dial? |
| 13:39.11 | [TK]D-Fender | wdoekes: Of course. Everyone can ask whatever they want. |
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| 14:09.15 | madduck | [TK]D-Fender: but it seems not possible to set the save location for the automixmon feature from the dialplan, eh? |
| 14:09.49 | WIMPy | Just set a full path. |
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| 14:13.36 | madduck | WIMPy: except automixmon doesn't seem to be configurable at all. I am talking about features.conf, not the application |
| 14:13.59 | WIMPy | Ah, well. |
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| 14:18.23 | madduck | maybe I can make a dynamic feature that spawns the MixMonitor app |
| 14:18.25 | madduck | since it has: |
| 14:18.26 | madduck | m - Create a copy of the recording as a voicemail in the indicated mailbox(es) separated by commas eg. m(1111default,â¦). Folders can be optionally specified using the syntax: mailbox@context/folder |
| 14:18.30 | madduck | mailbox |
| 14:18.51 | madduck | and also support for a command |
| 14:19.17 | madduck | but i somewhat doubt that the dynamic feature has access to the variables in the context of the dialplan. |
| 14:19.27 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_features#Asterisk12Configuration_features-featuremap_automixmon |
| 14:20.00 | [TK]D-Fender | There are channel variables you can set for the name... |
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| 14:23.59 | madduck | [TK]D-Fender: seems to be a v12 addition :/ |
| 14:24.06 | madduck | :/ since Debian stable has v11 |
| 14:24.21 | [TK]D-Fender | Glaciers are VERY stable ;) |
| 14:24.35 | *** join/#asterisk wonderworld (~ww@p50858F12.dip0.t-ipconnect.de) |
| 14:24.43 | cbeyerlein | I was just wondering why asterisk complains about not found include file... looks like #include<file> in asterisk.conf only works with absolute path, whereas in all other config files it works relative to astetcdir? |
| 14:25.06 | cbeyerlein | and more strange, if you #include<relativefile> in asterisk.conf, the complete asterisk.conf will be ignored |
| 14:25.17 | cbeyerlein | e.g. stuff you set in [directories] |
| 14:25.33 | cbeyerlein | thats not very straight-forward... |
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| 14:28.23 | [TK]D-Fender | cbeyerlein: Because * doesn't assume asterisk.conf is IN astetcdir. |
| 14:29.00 | [TK]D-Fender | cbeyerlein: Otherwise it wouldn't even have to be a setting... if * was already reading asterisl.conf from somewhere it's just assume that for the others. But since it's a setting it clearly isn't an assumption |
| 14:29.08 | [TK]D-Fender | #logic |
| 14:29.25 | cbeyerlein | good point! |
| 14:29.44 | cbeyerlein | but what about this one: |
| 14:29.46 | cbeyerlein | [root@fravocmg05t asterisk]# asterisk -C /opt/etc/asterisk/asterisk.conf -rx "core restart now" |
| 14:29.46 | cbeyerlein | The file 'include/config/asterisk.inc' was listed as a #include but it does not exist. |
| 14:29.46 | cbeyerlein | Unable to open specified master config file '/opt/etc/asterisk/asterisk.conf', using built-in defaults |
| 14:30.02 | cbeyerlein | it still ignores the complete asterisk.conf when it has an relative include :D |
| 14:30.13 | cbeyerlein | and even says "unable to open"... |
| 14:30.14 | Greenlight | I see calls from witheld numbers come in with a callerid of "anonymous" -- I should be able to pattern match on my extensions with this to treat these calls differently, right? eg exten => 0123456/anonymous,1,Hangup() |
| 14:30.36 | [TK]D-Fender | PASTEBIN all the actual backup for this... |
| 14:31.26 | sgriepentrog | cbeyerlein: don't use -C with -r - put it on the initial non-r (non-remote) start of asterisk. |
| 14:36.38 | [TK]D-Fender | Greenlight: Depends. |
| 14:37.01 | Greenlight | [TK]D-Fender: On ? |
| 14:37.12 | [TK]D-Fender | Greenlight: Where it is |
| 14:37.31 | Greenlight | It doesn't appear to be working, and I was wondering if "anonymous" was a special CallerID or something |
| 14:37.43 | [TK]D-Fender | Greenlight: Lets look at the actual call.... |
| 14:37.51 | [TK]D-Fender | And actual config while we're at it. |
| 14:40.26 | cbeyerlein | sgriepentrog: yes, my real error was usage of relative include in asterisk.conf.. but ignoring the complete asterisk.conf when the include cannot be found is cumbersome |
| 14:41.39 | *** join/#asterisk davlefouAMD (~davlefouA@197.15.62.182) |
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| 14:47.33 | Greenlight | [TK]D-Fender: Server's a little busy to capture any useful output at the moment.. pushing 20cps or so. Will see if perhaps a GoToIf using the ${CALLERID} variable works any better |
| 14:47.46 | *** join/#asterisk Kunsi (kunsmannf@unaffiliated/kunsi) |
| 14:48.07 | [TK]D-Fender | Greenlight: That isn't a variable that has been relevant since before * 1.2 |
| 14:48.24 | [TK]D-Fender | Greenlight: And you aren't looking at what's actually coming in.... |
| 14:48.54 | WIMPy | Greenlight: Usually it is "". I don't know if exten => exten/,1,bla() works again. |
| 14:49.14 | [TK]D-Fender | WIMPy: Highly doubtful |
| 14:49.49 | WIMPy | It used to work. But some time, a long time ago, it was broken. Haven't tried since. |
| 14:50.22 | Greenlight | Ahh okay, cool, I'll give that a shot,thanks! |
| 14:50.58 | Greenlight | Hmm... no joy |
| 14:51.32 | [TK]D-Fender | Greenlight: Go look at the actual call.... |
| 14:52.41 | Greenlight | A SIP trace? |
| 14:53.30 | WIMPy | Or just a Verbose(${CALLERID(num)})? or (pres) |
| 14:53.59 | *** part/#asterisk Kunsi (kunsmannf@unaffiliated/kunsi) |
| 14:56.03 | Greenlight | I'll see what I can do.... getting like 1000 lines a second in verbose output |
| 14:58.25 | WIMPy | Turn down the verbose level? |
| 14:58.58 | Greenlight | Ooo, forgot Verbose takes a level arg |
| 14:59.23 | WIMPy | Yes. My line was incomplete. |
| 14:59.35 | WIMPy | Sorry, if that was confusing. |
| 15:01.46 | *** join/#asterisk markusl (~markus@2001:8d8:924:5b10:b8a9:8bff:fe08:fd08) |
| 15:01.47 | markusl | hi |
| 15:02.56 | markusl | I'm currently porting extension monitoring stuff from pull based to a push based approach. |
| 15:03.29 | markusl | so what i've done before is poll a script that queries ExtensionState on the AMI. |
| 15:03.48 | Greenlight | Ahh okay, I see why it wasn't working now. The callerid is indeed "anonymous", but I was doing a GoTo first, and trying to match with /anaoymous after that, apparently that doesn't play ball with the callerid match. Doing it at the context it first comes in to works a treat though |
| 15:03.52 | Greenlight | Thanks for your help! |
| 15:04.04 | markusl | Now I want to have a persistent AMI connection and consume ExtensionStatus events right when they happen. |
| 15:05.21 | markusl | It looks like those events are only emitted when there are 'watchers' registered on the extension state (see core show hints) |
| 15:06.16 | markusl | I'd love to connect to AMI and say 'yo asterisk, watch this extension for me and give me ExtensionState events for it' |
| 15:06.19 | markusl | how to do that? |
| 15:06.37 | markusl | (we're using patterns to generate hints on request) |
| 15:06.51 | WIMPy | Just sit there and listen. |
| 15:07.01 | WIMPy | Ah. That would be an issue, I guess. |
| 15:07.10 | Greenlight | Hmm.. maybe not quite working then, seems it rejects non-anonymous calls now too. How odd. |
| 15:07.14 | markusl | I mean ExtensionStatus events. ExtensionState is what you request. |
| 15:08.10 | WIMPy | Greenlight: Are you using / or GotoIf now? |
| 15:08.46 | WIMPy | And do you have the priorities right? |
| 15:10.50 | Greenlight | Just swapped over to using GoToIf instead -- all working good now |
| 15:11.02 | Greenlight | Guess the pattern matching as some oddities |
| 15:11.17 | Greenlight | Rejecting anonymous calls, and allow the rest. All good |
| 15:11.50 | Greenlight | As much fun as it was having some annoyed customer calling in and conferening two of the call centre agents together.... |
| 15:12.27 | WIMPy | Interesting idea :-) |
| 15:13.47 | Greenlight | Some of the call recordings are still funny on the 10th play... |
| 15:13.54 | *** join/#asterisk vader- (~Adium@204.183.88.4) |
| 15:18.49 | markusl | ok. I solved my problem. Looks like I receive ExtensionStatus events once the hint is added by querying ExtensionState. |
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| 15:48.55 | qakhan | i am using page app to dial 5 extension. is there any way i can check which extension are registered when using page app. |
| 15:53.16 | [TK]D-Fender | qakhan: No. Page does not check. It simply dials. YOU can check BEFORE calling Page |
| 15:53.28 | [TK]D-Fender | qakhan: "core show function DEVICE_STATE" |
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| 16:08.42 | qakhan | [TK]D-Fender here is my pastbin http://pastebin.com/NTiH9k7W |
| 16:08.48 | qakhan | i am getting invalid |
| 16:09.16 | [TK]D-Fender | qakhan: Who said you pass MULTIPLE items ot that function? |
| 16:09.59 | [TK]D-Fender | qakhan: Read the instructions |
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| 16:14.20 | qakhan | [TK]D-Fender i need to check exten which are not registered |
| 16:14.55 | [TK]D-Fender | qakhan: And I just gave you the function and the means of getting the instructions on how to use it. |
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| 16:27.40 | qakhan | [TK]D-Fender i can check single ext state |
| 16:28.26 | qakhan | these ext can be 100 and i have to write 100 lines to check their state |
| 16:29.13 | *** part/#asterisk josecapurro (~jcapurro@214.2.23.190.res.adsl.dyn.click.com.py) |
| 16:29.54 | qakhan | can we get multiple ext state and save "UNAVAILABLE" state exten in db |
| 16:30.07 | [TK]D-Fender | qakhan: You aren't going to have 100. Especially for that method because a dialplan line can't even be long enough to support it |
| 16:30.34 | [TK]D-Fender | qakhan: You see what that command does. That is ALL. |
| 16:30.48 | [TK]D-Fender | qakhan: You'd better be capable of understanding what you have to do with this.... |
| 16:34.57 | qakhan | [TK]D-Fender it is true i will have 100 even more. |
| 16:35.25 | qakhan | so you are telling me that dialplan will not support 100 ext? |
| 16:39.05 | [TK]D-Fender | qakhan: You won't be able to pass that much as DATA to an Application |
| 16:49.10 | qakhan | ok how many data we can pass? |
| 16:51.11 | [TK]D-Fender | a LOT less. Something under 255 byes IIRC |
| 16:54.26 | qakhan | so how many exts we can dial? |
| 16:54.33 | qakhan | same time |
| 16:54.38 | [TK]D-Fender | same answer |
| 16:54.46 | [TK]D-Fender | this is a DATA limit |
| 16:54.51 | [TK]D-Fender | not an "extension limit" |
| 16:55.06 | file | The asteriskdocs.org site is now back online. |
| 16:55.07 | [TK]D-Fender | ie: how much crap can I put on 1 line of text |
| 16:55.26 | qakhan | ok |
| 16:55.29 | file | after part of the rack lost power and then a surge seemingly killed a disk and BIOS |
| 16:58.17 | qakhan | [TK]D-Fender so how can we page around 100 exts as time, |
| 16:58.38 | [TK]D-Fender | qakhan: Break it into local channels |
| 17:01.44 | qakhan | hint please :) |
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| 17:20.56 | [TK]D-Fender | That WAS a hint. |
| 17:21.17 | [TK]D-Fender | qakhan: In fact a very implicit usage. |
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| 17:53.32 | qakhan | [TK]D-Fender i can dial 110 exts in dialplan |
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| 17:53.57 | [TK]D-Fender | Show us |
| 17:54.14 | WIMPy | Wow. |
| 17:55.55 | qakhan | http://pastebin.com/S0JEsFbj |
| 17:57.05 | [TK]D-Fender | Guess the limit is bumped. What version are you on now? |
| 17:58.00 | qakhan | 11.10 |
| 17:58.14 | [TK]D-Fender | That might do it. Sure seems to. So you are paging that many.... |
| 17:58.24 | qakhan | yes |
| 17:58.34 | [TK]D-Fender | So you just have to build up the list of which ones to actaully call. |
| 17:59.59 | qakhan | i have to dial all of them either they are registered or not, but i want to log those exts which were not registered |
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| 18:16.26 | qakhan | [TK]D-Fender any suggestion on unregistered exts log |
| 18:21.11 | [TK]D-Fender | Do it yourself |
| 18:21.23 | [TK]D-Fender | You don't seem to understand that ALL of this is your job in the dialplan |
| 18:21.33 | [TK]D-Fender | All these concepts are things you invent in your head |
| 18:21.40 | [TK]D-Fender | And you have to code. |
| 18:22.02 | WIMPy | qakhan: Just listen for state changes on AMI and log the state when you execute that Dial or Page. |
| 18:26.50 | qakhan | ok |
| 18:27.02 | [TK]D-Fender | WIMPy: Wondow how identifiable that would be... or if it is really visible since you won't see a "new channel" event since * won't even try |
| 18:27.06 | [TK]D-Fender | wonder* |
| 18:28.28 | WIMPy | You would see the dialpan priority being executed. |
| 18:29.18 | [TK]D-Fender | WIMPy: but he's dialing them as devices within a page. |
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| 18:29.22 | [TK]D-Fender | that 1 call to 100 devices |
| 18:29.23 | WIMPy | My initial idea was to save the sates continousely. |
| 18:29.30 | [TK]D-Fender | no way to catalog individual failures to dial |
| 18:29.46 | [TK]D-Fender | If you're playing the "gap" method... EW! |
| 18:29.48 | [TK]D-Fender | :p |
| 18:30.09 | WIMPy | But you could also just take the argument of that Daial or Page and remove eveything you see a newchannel event for. Then you have the unreachable ones left at the end. |
| 18:30.12 | [TK]D-Fender | "See a PageAll, subtract tracked non-regs = PAIN |
| 18:30.30 | [TK]D-Fender | Just poll device_state for all it's worth! ;) |
| 18:30.39 | [TK]D-Fender | Better that a persistent tracker. |
| 18:30.53 | [TK]D-Fender | Less crap to blow up and you can do it in pure dialplan |
| 18:31.23 | WIMPy | Eats lots of resources. |
| 18:32.00 | [TK]D-Fender | 100 lines of non-lookup dialplan. Should execute somewhere on the scale of "instant". |
| 18:32.19 | [TK]D-Fender | everything stays within the core |
| 18:32.35 | WIMPy | If you do it via AMI it would execute in parallel. |
| 18:33.06 | [TK]D-Fender | but susceptable to race conditions, tracked states, extra code. |
| 18:33.23 | WIMPy | is a self confessed friend of the nanoseconds :-) |
| 18:33.25 | [TK]D-Fender | Far more complicated. I doubt the load could amtter at all in the grand scheme |
| 18:34.02 | WIMPy | I find it much easier than doin a humongous dialplan. |
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| 18:34.18 | WIMPy | But that's obviousely a personal thing. |
| 18:34.56 | [TK]D-Fender | WIMPy: He can't even handle the basics.... I'd hate to imagine him having to do something "serious" like that.... |
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| 19:03.28 | qakhan | [TK]D-Fender thank you |
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| 19:15.04 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 19:18.03 | Scunizi | making a google voice call through asterisk on Raspberry PI2b results in garbled voice on the asterisk end. The called number voice quality is excellent. Any solutions? |
| 19:23.33 | [Consultant]C-As | if i enter a X.,n,hangup() on a channel does it send a bye emidiatlly or does it wait to the h channel is completed? |
| 19:27.57 | [TK]D-Fender | immediate |
| 19:28.33 | gusto | he |
| 19:28.33 | gusto | what is going on? someone already has the raspberry pi2? |
| 19:28.33 | gusto | well, I have the same problem with that "garbeled sound" but it is not through raspberry pi causing this |
| 19:28.33 | gusto | it is when the codecs change |
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| 19:33.42 | [TK]D-Fender | Trnacoding = CPU killer... and you don't ahve much to spare on a Pi |
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| 19:57.53 | tm1000 | newtonr: ping |
| 19:59.40 | newtonr | tm1000, hay |
| 20:01.06 | tm1000 | newtonr: see your pm |
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| 20:45.21 | Scunizi | FYI - Yate client-Google Voice-Asterisk I've had audio issues with. Switching to SFLphone fixed all audio issues for me. |
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| 23:14.05 | Get_The_Fish | So I am trying to use the python pyst2 framework to issue originate commands to the AMI. I am getting the super helpful RequestNotAllowed response when issuing the originate. Is there any way to debug this AMI session? |
| 23:14.58 | Get_The_Fish | When I do it by hand with a telnet session to port 5038 it works. |
| 23:16.47 | kplong | In a fairly "default" scenario, day you have 2 sip phones and an asterisk server. When they call eachother, does both the audio data and the SIP signalling traverse through the asterisk server? Or is RTP/SRTP data going directly phone-to-phone |
| 23:17.15 | [TK]D-Fender | kplong, no real such thing as "default". |
| 23:17.26 | [TK]D-Fender | kplong, There's something in your configs, and you are responsible for that |
| 23:17.35 | Get_The_Fish | kplong, it depends on the value of the setting "directmedia" in sip.conf when using the chan_sip stack |
| 23:17.40 | [TK]D-Fender | kplong, did YOU tell it it could, or could not reinvite? |
| 23:18.01 | [TK]D-Fender | Any DTMF feautres allowed in a Dial will also keep * in the way |
| 23:18.07 | [TK]D-Fender | As will transcoding, recording, etc |
| 23:18.22 | Get_The_Fish | kplong, for this to work properly you will also need "canreinvite=yes" |
| 23:21.06 | Get_The_Fish | so using the command "manager set debug on" I dont see anything new coming across the console. Am I doing something wrong here? |
| 23:21.37 | [TK]D-Fender | Not looking at the raw AMI comms |
| 23:22.02 | [TK]D-Fender | TCPDUMP the whole thing and see what the actual call/response is looking like |
| 23:22.59 | kplong | What I am really trying to ask is for SRTP specifically - are keys exchanged between phones or between each phone and the asterisk server. For example if I have encryption enabled on the extension but then I call a phone which does not, or a phone in the outside world, is SRTP still being used for the first leg between the origin phone and asterisk |
| 23:23.10 | Get_The_Fish | I am getting garbled packets when I do that. I see some of the text but not all. Suggestions on the optimal tcpdump cap would be appreciated. |
| 23:23.37 | Get_The_Fish | I am using "sudo tcpdump -i lo -s 0 -w /vagrant/c2c.cap tcp port 5038" as the command |
| 23:24.45 | Get_The_Fish | kplong in the the case of srtp you would need to use direct media. Keys are only exchanged between the endpoint and the asterisk server. |
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| 23:26.09 | kplong | Get_The_Fish - are you saying that I would need to use directmedia if I wanted keys to exchange directly between phones, or that i would need directmedia to use SRTP at all |
| 23:26.36 | kplong | I dont want phones connecting directly to eachother at all. I do however want to make certain my SRTP is working. I think it is finally though |
| 23:27.20 | kplong | So SRTP is a secure connection for RTP between the endpoint and the server, not some kind of encryption negotation between endpoints |
| 23:27.36 | kplong | I think ZRTP does that which is why I figured it worked that way. |
| 23:27.41 | Get_The_Fish | kplong, then disable direct media and enable srtp. |
| 23:27.51 | Get_The_Fish | I cant speak to zrtp. |
| 23:28.05 | kplong | I have to directmedia setting and canrevinite=no on all extensions |
| 23:28.19 | kplong | so there should not be any phones connecting with eachother other than through asterisk, I hope. |
| 23:28.29 | kplong | no directmedia setting in my config I mean. |
| 23:29.35 | [TK]D-Fender | <kplong> I dont want phones connecting directly to eachother at all. I do however want to make certain my SRTP is working. I think it is finally though <- Then just LOOK at your call |
| 23:34.36 | kplong | [TK]D-Fender that would be great to do. I don't see anything in my "full" log that actually says anything about SRTP now that I think it's working |
| 23:34.43 | kplong | if a call is active can I check it from the asterisk CLI or somethin? |
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| 23:45.24 | [TK]D-Fender | You should have been watching the SIP DEBUG |
| 23:45.51 | [TK]D-Fender | just saying "full log" doesn't actually mean anything as to what gets in there |
| 23:45.54 | [TK]D-Fender | logs= trash |
| 23:46.04 | [TK]D-Fender | full CLI with proper things being debugged = real |
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