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| 00:21.17 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:21.17 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.15.0 (2017/04/07), 11.25.1 (2016/12/08), Standard: 14.4.0 (2017/04/07); 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 |
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| 03:39.31 | nafg_ | Hi, I have an asterisk FastAGI app and I want to add some functionality that uses Originate and I need to test it on my laptop |
| 03:39.40 | nafg_ | I have a soft phone (linphone) |
| 03:40.06 | nafg_ | I'm not sure what settings or values to use to make a call to my local softphone |
| 03:52.09 | nafg_ | this channel didn't used to be this quiet did it? |
| 03:52.52 | Samot | You really haven't asked a question about a problem. |
| 03:52.55 | Samot | ~wiki |
| 03:53.43 | Samot | Originate is covered in various forms on there. |
| 03:54.59 | nafg_ | Samot: I guess I don't have a deep enough understanding to articulate it better. What strings would I pass originate agi command to get to the local sip phone? |
| 03:56.03 | Samot | No, I understood what you were asking. |
| 03:56.22 | nafg_ | ok, so what did I omit? |
| 03:56.27 | Samot | I'm saying that Originate is covered in the Asterisk wiki. |
| 03:56.51 | Samot | Under ARI, AMI, Callfiles, Originate itself.. |
| 03:58.13 | nafg_ | Samot: this page? https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Originate |
| 03:58.53 | nafg_ | Doesn't answer |
| 03:58.57 | Samot | That's one of them. |
| 03:59.09 | Samot | If you want to issue the Originate via dialplan. |
| 03:59.42 | Samot | You can do it via AMI, ARI, the use of Callfiles, AGI classes.. |
| 04:02.14 | nafg_ | right, I know how to "do" originate. My question is what to put in the string for my scenario that I described |
| 04:02.43 | Samot | Uhm. |
| 04:03.32 | Samot | You originate a call to the endpoint peer |
| 04:03.52 | Samot | Like you would any other call you wanted to send to the peer. |
| 04:04.13 | Samot | If you're registered to the peer on linphone, then it will get the call. |
| 04:04.30 | nafg_ | can you show the exact string I would use? |
| 04:04.39 | nafg_ | (with placeholders) |
| 04:04.48 | Samot | If you know how to originate then you know what strings to use. |
| 04:05.31 | nafg_ | Ipso facto I don't know how to originate |
| 04:07.29 | nafg_ | In any case I don't know what strings to use. Can you tell me? |
| 04:07.36 | Samot | Dude. |
| 04:07.41 | nafg_ | I.e. what is the syntax/format of the string |
| 04:07.43 | Samot | It's documented in the wiki |
| 04:07.49 | nafg_ | Where |
| 04:07.54 | Samot | I told you |
| 04:08.05 | nafg_ | All I see is "tech_data - Channel technology and data for creating the outbound channel. For example, SIP/1234." |
| 04:08.23 | Samot | That's from the DIALPLAN |
| 04:08.34 | nafg_ | ok. So can you point me to the RIGHT page? |
| 04:08.42 | Samot | Originate can be called MULTIPLE ways. |
| 04:08.52 | Samot | I don't know HOW you are going to issue the Originate... |
| 04:08.56 | nafg_ | I'm calling it from asterisk-java |
| 04:09.01 | nafg_ | via fastagi |
| 04:09.06 | Samot | I have no experience with that. |
| 04:09.34 | lorsungcu | https://asterisk-java.org/tutorial/#apidocsorgasteriskjavamanageractionOriginateAction.html |
| 04:10.36 | Samot | If you're using Asterisk-Java for this, Asterisk-Java has its own method for calling Originate. |
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| 04:10.46 | Samot | Oh...apparently there it is. |
| 04:17.52 | nafg_ | There *what* is, that doesn't answer anything |
| 04:18.02 | nafg_ | What is the syntax/format of the string |
| 04:18.49 | Samot | It's in the documentation. |
| 04:19.02 | Samot | You're using Asterisk-Java's FastAGI script. |
| 04:19.08 | Samot | You have to do thing ITS way. |
| 04:19.24 | Samot | That page gave a fine example to do what you need. |
| 04:19.27 | Samot | Go do it. |
| 04:19.39 | Samot | It has all the syntax you're looking for. |
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| 07:40.56 | Haris | hello all |
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| 07:43.49 | Haris | I have 4 interfaces on this box. 1x to SIP provider. 1x to LAN. other 2x not being used. my local hostname is not resolving to any IPs. asterisk is filling log with it. Is it important where the hostname points ? |
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| 08:09.26 | dokma | On incoming I ring SIP/101&SIP/105 however 105 is giving me low TX volume. |
| 08:09.43 | dokma | Is there a way to increase TX only if 105 answers the incoming? |
| 08:10.31 | dokma | 105 is an Android SIP softphone. I tried both Linphone and Zoiper and have the same low TX on both apps. |
| 08:13.24 | dokma | Another thing is that I get low volume only for calls that route through Grandstream HT503. |
| 08:13.42 | dokma | Calls through LAN are of normal volume. |
| 08:14.10 | sekil | sounds like issue on the endpoint |
| 08:14.56 | dokma | sekil: which endpoint? Android or the incoming caller. |
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| 09:48.49 | MacroMan | When I start an attended transfer, the caller ID of the initiator is shown as 'asterisk' and not their number. |
| 09:49.04 | MacroMan | How can I set it? I've Googled but failed to find an answer |
| 09:51.28 | MacroMan | I'm using the atxfer feature in features.conf |
| 10:02.24 | MacroMan | Upon some more investigation, this only occurs when the original call is via a queue |
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| 13:40.42 | DexDeadly | I have a question, does anyone integrate cucm with asterisk? Is this done so that you can use cucm to properly manage the phones and then askterisk is used to maintain the sip trunks to providers? |
| 13:40.59 | Samot | How many Cisco phones do you have? |
| 13:56.18 | voipmonk | skinny box + Asterisk box ? |
| 13:57.09 | voipmonk | skinny on asterisk box? |
| 13:57.46 | voipmonk | skinny at customer prem out sip to asterisk box in the cloud ? |
| 13:58.08 | Samot | This is a FreePBX system |
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| 14:01.54 | Samot | Bringing the CUCM in to the mix just makes it worse. |
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| 14:10.10 | Maliuta_ | needs to take sweet, sweet, vengance on a phone scammer ... how many people would be prepared to make 1 call from a blocked number playing a short pre-generated message to a phone number in .au? |
| 14:10.34 | Maliuta_ | either that or I need to set up a VPN to a site outside .au and then just have it call them _all_ the time |
| 14:12.32 | Samot | Sounds like too much work. |
| 14:14.03 | Maliuta_ | Samot: a simple script set to call them every 5 minutes would do the job. I just need something outside the jurisdiction of the AFP |
| 14:14.23 | Maliuta_ | because it might be considered illegal use of a carriage service |
| 14:14.36 | Samot | Yeah, when jurisdiction is a factor of things...too much work. |
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| 16:47.54 | Kobaz | sooooo |
| 16:47.57 | Kobaz | ugh, sip |
| 16:48.04 | Kobaz | having registration problems |
| 16:48.25 | Kobaz | polycom phone -> vpn box -> routed subnet -> asterisk |
| 16:48.53 | Kobaz | polycom is not sending the second REGISTER that contains the authentication |
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| 17:06.27 | ghoti | Still having my "stuck channel" problem. I have a conference that continues to record (its .wav file keeps growing) When I "channel request hangup" the channel, nothing appears to happen. How do I debug this? |
| 17:06.41 | Kobaz | ghoti: sounds like an internal locking problem |
| 17:06.44 | Kobaz | what asterisk version? |
| 17:08.29 | ghoti | Kobaz: This is a 3-year-old FreePBX install .. Asterisk 11.8.1. |
| 17:11.42 | ghoti | I'm getting the channel from `core show channels concise`. I actually have TWO of these growing recordings at the moment... |
| 17:12.04 | Kobaz | ghoti: upgrade first, before doing any hair pulling |
| 17:13.58 | ghoti | Alas, not so easy... We're running 24/7 support on this thing and I don't have a backup (due to recent hardware fault)... |
| 17:14.43 | ghoti | With some prep, I could reboot, but that would only address symptoms, and I'd rather know what actually *causes* this, since it happens a few times a year. |
| 17:14.45 | [TK]D-Fender | wonders how a backup can't be made... |
| 17:15.26 | ghoti | D-, I could back things up, but I need a RUNNING system, 24/7. The back-up hardware is out for repair at the moment. (We only keep one spare phone system, not two.) |
| 17:15.52 | Kobaz | ghoti: this sounds probably 99% like an internal bug... and it's almost pointless to debug an old version unless you specifically need that version |
| 17:16.32 | ghoti | Kobaz: fair enough. I had another long-running recording that I *was* able to end by running "channel request hangup" on the connected SIP/* channel. Just have these other ones that appear to be ignored. |
| 17:16.32 | [TK]D-Fender | Kill all the channels in the conference |
| 17:16.50 | [TK]D-Fender | if one refuses to consider disconnecting then use an AMI Redirect to toss it off a cliff |
| 17:16.57 | Kobaz | mm |
| 17:16.58 | Kobaz | interesting |
| 17:17.20 | Kobaz | [TK]D-Fender: but if it's blocking on something hardcore in conference, it may not respond to a redirect |
| 17:17.36 | ghoti | [TK]D-Fender: In each case, there's one SIP/ connection that won't die, and four or five "ConfBridgeRecorder/conf-829-uid-#####" channels that may or may not be related... |
| 17:17.48 | [TK]D-Fender | Try it. Good odds it'll do so violently |
| 17:18.04 | ghoti | Try ... what? Killing the ConfBridgeRecorder channels too? |
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| 17:18.47 | [TK]D-Fender | Everything you have to in your path |
| 17:18.57 | [TK]D-Fender | #scorchedearth |
| 17:19.59 | ghoti | My goodness. |
| 17:20.04 | ghoti | The files have stopped growing. |
| 17:20.39 | ghoti | I was so certain that it was the SIP/* channels that needed to be demolished, I hadn't tried the ConfBridgeRecorder ones. One of them must have been the thing at fault. |
| 17:20.40 | Kobaz | ghoti: you can always echo "" > recordingfile |
| 17:20.49 | Kobaz | if you can't kill the recorders and you're running out of disk space |
| 17:21.05 | Kobaz | clear the file on disk, and it will still grow again, but it'll reclaim space so far |
| 17:21.24 | ghoti | Kobaz: if a file is kept open with some process's fopen(), will that still truncate the file? I know that *deleting* the file wouldn't work because fopen() attaches to the inode... |
| 17:22.05 | Kobaz | yeah |
| 17:22.08 | ghoti | # find 2* -type f -size +10G -print0 | xargs -0 ls -lh |
| 17:22.08 | ghoti | -rw-rw---- 1 asterisk asterisk 126G Apr 25 13:19 2017/01/17/conf-832-832-20170117-155916-1484686751.63669-1484686767.wav |
| 17:22.08 | Kobaz | it'll clear blocks on disk |
| 17:22.11 | ghoti | -rw-rw---- 1 asterisk asterisk 123G Apr 24 08:37 2017/01/19/conf-800-800-20170119-095613-1484837767.63972-1484837786.wav |
| 17:22.14 | ghoti | -rw-rw---- 1 asterisk asterisk 110G Apr 25 13:19 2017/01/30/conf-829-829-20170130-143032-1485804625.65964-1485804641.wav |
| 17:22.14 | Kobaz | haha |
| 17:22.17 | ghoti | :-) |
| 17:22.17 | Kobaz | that's large |
| 17:22.24 | Kobaz | echo > |
| 17:22.28 | Kobaz | if you dont care about the data |
| 17:23.07 | ghoti | Thanks for the tip, I'll keep that in mind for next time. |
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| 17:24.37 | orn | Quick question; all the documentation for Queues says that ENTERQUEUE has two parameters, url and callerid |
| 17:25.04 | orn | I find however that there are three parameters... does anyone know what the third one is? My best guess is the original position within the queue |
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| 17:38.38 | orn | OK. Just looked at the source code. It is with all likelihood the original position (qe.opos within the code) |
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| 23:17.28 | TandyUK | [2017-04-26 00:17:01] WARNING[23490]: tcptls.c:683 handle_tcptls_connection: FILE * open failed! |
| 23:17.45 | TandyUK | seeing lots of these suddenly, despite hundreds of phones being registered ok |
| 23:17.57 | TandyUK | what can i do to get more info about what is causing that? |
| 23:18.22 | TandyUK | It seems to be cert related, but i dont understand why it only appears to affect certain connections, and not all of them |
| 23:23.14 | Samot | Are all the connections TLS? |
| 23:23.37 | TandyUK | yes |
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| 23:27.08 | TandyUK | its usually preceeded by |
| 23:27.08 | TandyUK | <PROTECTED> |
| 23:27.16 | TandyUK | which is highly descriptive :P |
| 23:28.05 | TandyUK | 11.25.1 btw |
| 23:28.18 | TandyUK | im just reading the source trying to make sense of it, and failing lol |
| 23:32.00 | TandyUK | any way to verify the certificate etc is ok externally |
| 23:32.14 | TandyUK | i know of tools for checking https etc, but none for sip/tls |
| 23:39.22 | Samot | Well.. |
| 23:39.40 | Samot | The certificate is valid or not regardless of https or sip/tls |
| 23:39.55 | Samot | Do you have more than one cert? |
| 23:40.41 | TandyUK | no theres only the one cert on the server |
| 23:41.21 | TandyUK | its always "WARNING[<<NUMBER>>} ...." |
| 23:41.45 | TandyUK | any idea what that number relates to, might help me track down whats causing it |
| 23:41.53 | TandyUK | the number is always different |
| 23:54.46 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |