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| 02:05.53 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.1 (2018/11/14), Security Only: 15.6.2 (2018/11/14); 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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| 06:13.57 | Milos | Does anyone know how I can inspect a certificate from the command-line? Like, using nc/openssl? So I can verify why Asterisk says that it's expired. |
| 06:14.08 | Milos | I'm talking about a remote certificate on port 5061. |
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| 06:18.27 | Milos | ^ |
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| 06:31.11 | wyoung | Milos: I know |
| 06:31.52 | wyoung | openssl x590 -in ya_cert -noout -text |
| 06:32.00 | wyoung | x509 even |
| 06:32.10 | wyoung | (sorry typoe) |
| 06:32.16 | Milos | I don't have the cert, it's at the remote server on port 5061 |
| 06:32.43 | wyoung | use openssl sclient then, should I find the command or are you able to read man files? |
| 06:33.09 | wyoung | s_client even |
| 06:33.31 | wyoung | https://stackoverflow.com/questions/7885785/using-openssl-to-get-the-certificate-from-a-server |
| 06:33.38 | wyoung | First result of google search |
| 06:33.52 | Milos | yes, wasn't sure if that would work here but it does, thanks |
| 06:34.02 | wyoung | <3 |
| 06:34.08 | wyoung | Anytime |
| 06:34.55 | Milos | well I have another question...... |
| 06:35.21 | Milos | I spent the last 2 hours trying to debug something that started happening after I restarted the asterisk server, and randomly went away after 2 hours of debugging and another restart of the asterisk server |
| 06:35.26 | Milos | I was getting this in the logs: |
| 06:35.43 | Milos | WARNING[25819] chan_sip.c: Retransmission timeout reached on transmission 698654759-5060-8@BJC.BGI.II.CBB for seqno 61 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 06:35.49 | Milos | what the hell is BJC.BGI.II.CBB? |
| 06:35.52 | Milos | I googld it and had no hits |
| 06:35.58 | Milos | I downloaded the source code and got no hits |
| 06:36.07 | Milos | 2 hours of debugging later and 1 restart later, it doesn't happen anymore? |
| 06:36.12 | Milos | I changed nothing |
| 06:36.32 | Milos | it's meant to show an IP address there |
| 06:36.34 | Milos | not random letters |
| 06:36.43 | Milos | the letters are always the same |
| 06:37.20 | wyoung | Milos: This is a question for [TK]D-Fender. He is a lesser god of asterisk, but greater than a demi-god. |
| 06:37.44 | Milos | I'll ask when he's around. |
| 06:37.51 | Milos | s/he/they/ |
| 06:37.53 | Milos | Oh, they are around. |
| 06:37.55 | wyoung | :+1: |
| 06:37.56 | [TK]D-Fender | Some would sooner say "persistent rash", but we don't need that kind of negativity :p |
| 06:39.45 | Milos | I am running 13.23.1 |
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| 12:29.26 | Samot | Milos: That is the SIP Call-ID |
| 12:30.17 | Samot | So it wouldnt reallt hold something that is searchable as those are unique. |
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| 12:46.14 | *** join/#asterisk csguth (c8af3d51@gateway/web/freenode/ip.200.175.61.81) |
| 12:46.33 | csguth | hi all |
| 12:46.55 | csguth | can someone help me with setting up SRTP on my asterisk server? |
| 12:47.16 | csguth | I'm able to use TLS transport, but I can't enable media encryption :( |
| 12:47.53 | csguth | I'm on Asterisk13 cert3 |
| 12:48.11 | csguth | pjsip for sip stack |
| 12:49.49 | Samot | What are your settings now? |
| 12:50.18 | Samot | And you set media_encryption? |
| 13:02.18 | Samot | Alrighty. |
| 13:16.59 | *** join/#asterisk csguth (c8af3d51@gateway/web/freenode/ip.200.175.61.81) |
| 13:17.07 | csguth | damn |
| 13:17.12 | csguth | I've tried that |
| 13:17.22 | csguth | I still get 488 on my Zoiper phone |
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| 13:27.25 | Samot | That has multiple meaninga and causes |
| 13:33.39 | csguth | If I disable srtp on zoiper and remove that media_encryption line from asterisk config, it works |
| 13:33.57 | Samot | Show a call that fails. |
| 13:34.00 | Samot | asterisk -rvvvvvvv |
| 13:34.05 | Samot | pjsip set logger on |
| 13:34.17 | csguth | just a second |
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| 13:39.08 | csguth | I don't think thats something useful |
| 13:39.14 | csguth | I'm getting the sdp with the ciphers |
| 13:39.17 | csguth | and then |
| 13:39.18 | csguth | <--- Transmitting SIP response (328 bytes) to TLS:200.175.61.81:56832 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS 200.175.61.81:56832;rport=56832;received=200.175.61.81;branch=z9hG4bK-524287-1---73de19223473b898 Call-ID: MH_e6dn04A2eY8Ued_Ue1Q.. From: <sip:alice@54.186.222.123>;tag=47ec990a To: <sip:1@54.186.222.123> CSeq: 2 INVITE Server: Asterisk PBX certified/13.21-cert3 Content-Length: 0 <--- Transmitting SIP response (382 bytes |
| 13:39.23 | file | pastebin it. |
| 13:39.29 | file | the entire thing. |
| 13:39.54 | csguth | sorry |
| 13:41.28 | csguth | https://pastebin.com/zMct3zXs |
| 13:42.08 | file | and provide the current configuration. |
| 13:42.31 | csguth | https://pastebin.com/BApwCihJ |
| 13:43.24 | Samot | You shouldnt be using sslv23 |
| 13:43.36 | csguth | If I remove both media_encryption=sdes and disable srtp on zoiper, it works |
| 13:43.44 | Samot | Shoudnt that be tlsv1 or another tlsv |
| 13:43.49 | file | Is the res_srtp module loaded? |
| 13:43.49 | csguth | hmm |
| 13:44.19 | Samot | Uhm |
| 13:44.24 | Samot | Fix your codecs. |
| 13:44.28 | Samot | Order mattets |
| 13:44.31 | Samot | Matters |
| 13:44.46 | Samot | Your pbx and zoiper do not have codecs ordered the same way |
| 13:45.03 | Samot | Possible problem as well |
| 13:45.17 | csguth | hm |
| 13:45.28 | csguth | regarding the method |
| 13:45.43 | csguth | zoiper is using sslv2/3 |
| 13:45.45 | Samot | But what file asked is important |
| 13:46.08 | csguth | should I change to tlsv1? |
| 13:46.29 | Samot | Answer files question |
| 13:46.44 | csguth | ahh |
| 13:46.51 | csguth | res_srtp |
| 13:46.52 | csguth | hm |
| 13:48.12 | csguth | I think I don't have that module installed ^^' |
| 13:48.20 | csguth | should I have a /usr/lib/asterisk/modules/res_srtp.so ? |
| 13:48.40 | file | if your Asterisk was built with SRTP support, yes |
| 13:48.46 | csguth | ok |
| 13:48.49 | csguth | what a shame |
| 13:48.59 | csguth | I probably didn't do that on build time |
| 13:49.23 | Samot | OK |
| 13:49.26 | Samot | asterisk -r |
| 13:49.36 | Samot | module show like srtp |
| 13:49.40 | Samot | What is the output |
| 13:49.58 | csguth | 0 modules loaded |
| 13:50.03 | csguth | :) |
| 13:50.09 | csguth | I'll have to re-build asterisk |
| 13:50.33 | csguth | one more question, for asterisk to use TURN: |
| 13:50.49 | csguth | adding turnaddr, turnusername and turnpassword on rtp.conf |
| 13:50.59 | csguth | and |
| 13:51.03 | csguth | ice_support=yes direct_media=no on pjsip.conf |
| 13:51.07 | csguth | is it enough? |
| 13:51.30 | Samot | I honestly don't know. I've never used Asterisk and TURN together. |
| 13:53.20 | file | ICE has to actually be negotiated and used. |
| 13:53.40 | file | It also has to be built with pjproject to have ICE/STUN/TURN support. |
| 13:54.16 | csguth | @file and Samot, thank you all very much :) |
| 13:54.30 | csguth | I'm rebuilding asterisk with srtp supprot |
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| 15:54.34 | cal-tec | does anyone on here have any idea what is needed to implement a custom event package in asterisk? I believe it can be done using a pjsip module but I am lost as to where to start. I want to respond to a SIP SUBSCRIBE with a custom Event type. Specifically avaya-cm-feature-status |
| 15:54.59 | file | res_pjsip_mwi would be the smallest module which implements such functionality, otherwise there isn't really a guide |
| 15:57.00 | cal-tec | ok.... is the source for res_pjsip_wmi in the asterisk sources? |
| 15:57.57 | cal-tec | never mind... I found it |
| 16:00.10 | cal-tec | file: thank you for your help |
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| 16:28.03 | *** mode/#asterisk [+o bford] by ChanServ |
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| 16:45.19 | trmg | Is it possible to configure two different email templates for the native voicemail app? I'd like to send an abbreviated version to a cell phone (10digitnumber@txt.carrier.com or whatever). Would it be possible for a given mailbox to send, say, the full message plus attachment to email A, and then an abbreviated message without attachment to email B? |
| 16:59.39 | [TK]D-Fender | trmg, that's what the pager e-mail address is for |
| 17:00.58 | trmg | orly? |
| 17:02.31 | trmg | Ok so the second email address on a mailbox is by default the pager email? |
| 17:05.29 | trmg | Answered my own question... |
| 17:05.31 | trmg | "Each mailbox is listed in the form <mailbox>=<password>,<name>,<email>,<pager_email>,<options> |
| 17:05.34 | trmg | " |
| 17:05.51 | trmg | I will have to play with this. |
| 17:11.40 | life_of_e | trmg: voicemail.conf has configurations for the pager versions of the body |
| 17:11.56 | trmg | Yeah, that's what I was looking at. |
| 17:12.02 | trmg | I never thought to search for the word "pager". |
| 17:12.03 | life_of_e | as pagerfromstring, pagersubject and pager body |
| 17:12.05 | trmg | But now it all makes sense. |
| 17:12.27 | life_of_e | the alternative would be to use mail filtering on the server and let it reparse the email |
| 17:13.07 | trmg | Makes sense |
| 17:13.14 | life_of_e | I do this on my own email server to take certain messages (like from banks), reformat and send it out to my phone as an SMS. |
| 17:14.10 | trmg | Nice |
| 17:14.22 | life_of_e | I get to avoid the spam messages that the bank will send if I give them my cell number (I pay per SMS message, no unlimited text on my plan) |
| 17:28.53 | *** join/#asterisk IsUp (5f0897b6@gateway/web/freenode/ip.95.8.151.182) |
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| 17:31.48 | IsUp | hello, i am trying to make a queue with single agent which is myself (over SIP), however AgentLogin keeps me waiting on MOH. is there any way to login and then hangup then receive calls from queue? |
| 17:34.25 | [TK]D-Fender | nobody (literally0 uses Agentlogin |
| 17:34.46 | [TK]D-Fender | "core show applications like queue" <- |
| 17:35.02 | [TK]D-Fender | there are add, remove, pause commands to add devices as members |
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| 17:40.01 | IsUp | [TK]D-Fender: and should i use Agent channels at all? |
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| 17:44.24 | [TK]D-Fender | It's 1/100000 for people who want to work that way |
| 17:44.47 | [TK]D-Fender | Only the worst call centers force their agents to literally sit chained to their phones like that and also force-answer calls. |
| 17:44.53 | [TK]D-Fender | Total dick move |
| 17:47.25 | IsUp | got it |
| 17:50.17 | trmg | We have our folks log in/out of our queues. |
| 17:50.20 | trmg | shrugs |
| 18:00.25 | IsUp | adding this much easier :p member => SIP/20 |
| 18:12.40 | [TK]D-Fender | only downside: they are always logged in |
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| 19:54.30 | drmessano | AgentLogin should come with a disclaimer |
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