| 00:28.25 | Samot | ? |
| 00:28.43 | Samot | I installed 13.11 in distro |
| 01:00.03 | *** join/#asterisk mmlj4 (1000@ip98-163-252-185.no.no.cox.net) |
| 01:04.33 | *** join/#asterisk masked (~masked@hpavc/masked) |
| 01:04.49 | masked | Samot: got it figured out |
| 01:05.19 | masked | with asterisk, the only way to get the media to transport from loopback to the real world interface is to use a media proxy |
| 01:06.08 | masked | or so it would seem |
| 01:07.36 | Samot | Why is the media on the loopback to begin with? |
| 01:10.22 | masked | because defining media_address etc doesn't do what it's meant to do lol |
| 01:10.50 | masked | i've got asterisk on loopback due to cost factors and security |
| 01:14.57 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 01:17.20 | Samot | What do you mean defining the media_address doesn't do what it's meant to do? |
| 01:18.11 | masked | well, that long with bind_rtp_to_media_address, i would have thought, would set the media to the ip specified.. |
| 01:20.48 | Samot | And what about the external_media_address? |
| 01:30.09 | masked | Samot: yeah it changes the i= but not the c= |
| 01:30.28 | Samot | Is Asterisk behind NAT? |
| 01:30.31 | masked | no |
| 01:30.36 | Samot | So it's public? |
| 01:30.36 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 01:30.49 | masked | well, its on loopback |
| 01:30.56 | drmessano | Is this the Fritzbox install? |
| 01:31.16 | Samot | So it doesn't have a LAN or WAN IP? |
| 01:31.29 | masked | asterisk on a fritzbox? neat, nah, this is freebsd |
| 01:31.36 | drmessano | No, behind it |
| 01:31.38 | masked | Samot: correct |
| 01:31.45 | Samot | Why? |
| 01:31.49 | Samot | Seriously. |
| 01:31.49 | drmessano | wrong guy |
| 01:32.13 | Samot | Why is Asterisk only listening on 127.0.0.1? |
| 01:32.18 | masked | Samot: for fun and fancy to please old nancy |
| 01:32.24 | drmessano | .... |
| 01:32.28 | Samot | OK, I don't care. |
| 01:32.31 | Samot | Have your issues. |
| 01:32.36 | masked | doesn't matter, it works now |
| 01:33.04 | drmessano | So you have Asterisk on localhost and you're proxying ? |
| 01:33.28 | masked | however, having said that. i know you'll say fs and asterisk arent the same.. but if you set rtpip on fs to the realworld ip, it works. however this bind_rtp_to_media address doesn't appear to work |
| 01:33.35 | masked | drmessano: correct. |
| 01:33.58 | Samot | Are you telling it to set the external IP in any way? |
| 01:34.09 | drmessano | Thats incredibly odd and impractical |
| 01:34.16 | masked | yeah external_media_address is set to the external ip |
| 01:34.53 | Samot | And does Asterisk know it's basically behind NAT? |
| 01:35.10 | Samot | If you are translated a public IP to a private IP that's NAT. |
| 01:35.12 | masked | i tried turning on nat and it behaves the same.. |
| 01:35.29 | Samot | What did you have as the settings? |
| 01:35.56 | masked | yeah but rdr-to isn't implemented in freebsd's pf, so i can't really redirect it afaik |
| 01:37.02 | Samot | So far it sounds like this deployment has more fixes for the issues it causes than being a practical solution. |
| 01:37.10 | drmessano | Indeed |
| 01:37.27 | drmessano | I dont see the secruity of putting another daemon in front of Asterisk for RTP.. |
| 01:37.42 | Samot | He's using OpenSIPS. |
| 01:37.47 | Samot | So the device register to OpenSIPs |
| 01:37.51 | drmessano | okay |
| 01:37.55 | Samot | OpenSIPS sends the calls to Asterisk. |
| 01:38.02 | Akari | Is there any security advantage to that |
| 01:38.09 | Samot | Somehow between OpenSIPS and Asterisk it has to be "extra secure" |
| 01:38.15 | drmessano | ROFL |
| 01:38.18 | Samot | By having loopbacks. |
| 01:38.30 | Samot | Really, OpenSIPS should be handling all the public facing stuff... |
| 01:38.37 | drmessano | So you move your vulnerablities from one daemon to another |
| 01:38.41 | Samot | And routing the calls to Asterisk on a private LAN. |
| 01:38.41 | Akari | masked: Remember to enable TLS/SRTP between OpenSIPS and Asterisk |
| 01:38.42 | masked | yeh.. drmessano, i've tried to map asterisk's rtp socket to the real world ip, but it doesn't work. |
| 01:38.47 | masked | Akari: i have. |
| 01:38.53 | Akari | OK good |
| 01:38.54 | drmessano | masked: Because youre binding it to localhost |
| 01:39.00 | drmessano | For some insane reason |
| 01:39.22 | Samot | It needs to be on a private or public IP |
| 01:39.36 | drmessano | If anyone has tested this, they're dead now and we never heard from them |
| 01:40.05 | Samot | Even lidl, the master of FreeBSD Asterisk would scratch his head at this. |
| 01:40.20 | masked | Akari: yeah, if it's udp, it passess udp to asterisk, tcp == tcp, and tls == tls |
| 01:40.36 | masked | this is hardly a freebsd specific thing |
| 01:40.59 | Samot | No. |
| 01:41.03 | drmessano | Youre right |
| 01:41.11 | drmessano | Ive never heard of anyone else doing it |
| 01:41.19 | Samot | Nope. Never. |
| 01:41.52 | masked | i did it with kamailio + freeswitch quite successfully, having a snapshot of a working config here i used for months. |
| 01:42.08 | Akari | masked: What is the functional difference you're achieving though |
| 01:42.26 | drmessano | I didnt see Asterisk anywhere in that sentence, and I still don't see the actual need |
| 01:43.12 | masked | opensips afa security goes has the pike module which i find better than fail2ban, and it handles nat better |
| 01:43.23 | Samot | drmessano: I built my Dragula from a Mustang. Now I built it with a Chevy but it's not doing the same things the same way. |
| 01:43.34 | Samot | ...... |
| 01:43.52 | masked | i've yet to try directmedia with endpoints behind nat with just asterisk alone |
| 01:43.56 | masked | but with opensips |
| 01:44.05 | masked | i get peer-to-peer media traffic |
| 01:44.11 | drmessano | You keep justifying your use of OpenSIPS... and no one is disputing that |
| 01:44.12 | Samot | You misunderstand how Pike works. |
| 01:44.21 | drmessano | Your implementation is the issue |
| 01:44.32 | masked | and only when they fail to initiate a peer-to-peer hand-off, does the proxy come in |
| 01:44.37 | masked | the rtp proxy that is. |
| 01:44.44 | Samot | So wait.. |
| 01:44.59 | Samot | You're doing direct media with an endpoint that listens on loopback?! |
| 01:45.35 | masked | no i dont even use asterisk for subscribers/users |
| 01:45.41 | masked | only for media services |
| 01:45.47 | Samot | Right. |
| 01:45.48 | masked | like conference |
| 01:45.53 | Samot | Were the media would be direct. |
| 01:46.10 | masked | behind nat? |
| 01:46.23 | masked | oh |
| 01:46.25 | masked | nvm |
| 01:46.28 | Samot | PHONE --> ASTERISK |
| 01:46.31 | Samot | That's the RTP path.. |
| 01:46.33 | masked | i misunderstood. |
| 01:46.34 | masked | yeah |
| 01:46.34 | Samot | Directmedia.. |
| 01:46.42 | masked | yuppers |
| 01:46.44 | Samot | And Asterisk is listening on the loopback. |
| 01:46.48 | Samot | This is why you need a proxy. |
| 01:46.54 | Samot | Because you're deployment is wrong. |
| 01:47.04 | Samot | You put yourself in a position to need it. |
| 01:47.10 | masked | haha |
| 01:47.30 | masked | or because asterisk isn't binding the rtp to the ip i specify correctly |
| 01:47.42 | Samot | Because you're listening on the LOOPBACK |
| 01:47.46 | Samot | Which is for LOCAL ONLY |
| 01:47.54 | Samot | Not for anything else. |
| 01:47.56 | masked | im doing signaling on loopback |
| 01:48.01 | drmessano | .....but it works on $another_telephony_app |
| 01:48.06 | WIMPy | o.O |
| 01:48.14 | Samot | You have Asterisk configured wrong. |
| 01:48.18 | Samot | Accept it. |
| 01:48.19 | Samot | Fix it. |
| 01:48.25 | drmessano | .....but it works on $another_telephony_app |
| 01:48.38 | WIMPy | Since whencan you bind RTP ports to a specific IP? |
| 01:48.38 | Samot | Yup. |
| 01:48.49 | Samot | Not RTP ports. |
| 01:49.00 | Samot | The IP RTP uses. |
| 01:49.02 | masked | and who is the main author of freeswitch and is/was a main author of asterisk? |
| 01:49.18 | drmessano | He was the main author of Asterisk? No |
| 01:49.28 | Samot | He was a developer. |
| 01:49.33 | masked | a |
| 01:49.35 | Samot | Mr. Spencer is the author. |
| 01:49.48 | masked | nevermind the semantics |
| 01:49.53 | drmessano | Of course not |
| 01:50.19 | drmessano | Your context and pretenses are all wrong.. so lets forget them and go back to arguing over a scenario that's known to not work |
| 01:50.49 | drmessano | Sorry, thats more semantics |
| 01:51.08 | masked | well the only caveat is through the proxy it's a bit slow to initiate a call |
| 01:51.20 | drmessano | Of course it is |
| 01:51.26 | Samot | I've never had these problems with OpenSER/Asterisk. |
| 01:51.31 | Samot | In almost 12 years. |
| 01:52.22 | Samot | I've also never heard of this type of deployment, ever. |
| 01:55.09 | Samot | Oh and what'cha doing so far north? |
| 01:55.49 | masked | yeah.. well i haven't used asterisk in almost 12 years.... |
| 01:55.55 | Samot | I noticed your check-in... |
| 01:58.10 | drmessano | A lot has changed in 12 years |
| 01:58.16 | masked | Samot: me? |
| 01:58.36 | masked | drmessano: sure has, from what i've seen the last few days im quite impressed |
| 01:58.37 | WIMPy | You missed the good times. |
| 02:00.49 | Samot | No, drmessano. |
| 02:01.12 | drmessano | Samot: Theres this whole solidarity thing for the Pipeline protesters |
| 02:01.18 | drmessano | Everyone is checking in |
| 02:01.30 | drmessano | Because the feds were using FB checkins to target protesters |
| 02:01.45 | drmessano | So we're all checking in |
| 02:03.52 | drmessano | Makes you want to run Asterisk on loopback WITHOUT a proxy |
| 02:07.43 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 02:19.46 | Akari | Makes you want to run Asterisk over the TOR network and then through 7 proxies and then encrypted over the loopback interface. |
| 02:20.46 | Samot | While running it on a WD MyCloud |
| 02:21.20 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 02:22.20 | drmessano | Akari: I like to use PGP-Net on my loopbacks |
| 02:34.08 | Penguin | [Nov 2 21:25:12] ERROR[19865]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("m", "(null)", ...): Name or service not known |
| 02:34.11 | Penguin | [Nov 2 21:25:12] WARNING[19865]: chan_sip.c:18035 check_via: Could not resolve socket address for 'm' |
| 02:34.29 | Penguin | WIMPy: I'm inclined to agree with you and the other guy who said this is simply corruption during the lookup. |
| 02:51.11 | *** join/#asterisk tuxd00d (~tuxd00d@24.121.219.186) |
| 03:05.38 | *** join/#asterisk nathani (~nathani@2607:f2f8:ac88::) |
| 03:06.55 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 03:08.19 | *** join/#asterisk dadrc (~quassel@unaffiliated/dadrc) |
| 03:10.06 | *** join/#asterisk monsterco (~monsterco@207.35.222.227) |
| 03:10.51 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 03:13.21 | *** join/#asterisk danjenkins (danjenkins@gateway/shell/firrre/x-qwxwlzrvaoxbnadn) |
| 03:13.21 | *** mode/#asterisk [+o danjenkins] by ChanServ |
| 03:18.41 | *** join/#asterisk monsterco (~monsterco@207.35.222.227) |
| 03:21.07 | *** join/#asterisk [NC]_ (~nc@rv1.sabius.net) |
| 04:21.02 | masked | drmessano: arguable.. with a proxy, at least you can have the call drop out if it's intercepted |
| 04:21.12 | masked | but regardless |
| 04:21.18 | masked | i having it working without the proxy now |
| 04:22.10 | masked | media_address and external_media_address,e tc etc only confused the situation |
| 04:22.24 | masked | works without them |
| 04:46.02 | *** join/#asterisk Prelude2004c (~Prelude20@anonymousvpn.zazeen.com) |
| 04:46.54 | Prelude2004c | hey guys.. good day to everyone. I have a question, is it possible to receive text messages on asterisk ? meaning if i send a text message from my phone carrier to my sip phone number, is it possible to identify and allow the message to be delivered to my sip application ? |
| 04:47.08 | Prelude2004c | on the reverse side i guess i am able to send the message from my sip address outside the network |
| 04:48.18 | Samot | How would the call get intercepted? And by what? |
| 04:48.54 | Samot | Prelude2004c: Yes, you can do text messages on Asterisk with SIP SIMPLE |
| 04:49.12 | Prelude2004c | right but that is sip to sip right? |
| 04:49.20 | Prelude2004c | how do i do that to get messages from an outside carrier |
| 04:49.26 | Samot | Yeah, you'll need a SMS system. |
| 04:49.57 | Samot | Or a provider that supports SMS and use their APIs to send/receive... |
| 04:50.44 | Prelude2004c | the send i can figure out but to receive i am concerned... because if for example someone has a Bell or Rogers ( in canada ) and they send to the phone number, how does asterisk now to send it to SMS server |
| 04:51.01 | Prelude2004c | i sent a message to my sip now and my cellular phone said.. thank you for using text to landline |
| 04:51.16 | Prelude2004c | somehow the answer didn't contain anything saying SMS support so it left a message instead |
| 04:52.18 | Samot | The number has to be able to receive SMS |
| 04:53.57 | Prelude2004c | i control the asterisk server so that is what i am asking.. how to configure asterisk to pickup when its an sms call and route to the right sms server or simply write script to send message or whatever |
| 04:54.48 | Samot | As far as I know you would need an SMS gateway that either connects to another SMS gateway or a SMSC system (usually at the carrier level). |
| 04:55.18 | Samot | Or combination of both. |
| 04:56.50 | masked | Samot: https://www.vocal.com/lawful-interception/voip-decoding/ |
| 04:57.53 | Samot | So you're saying someone is going to hack your calls with that? |
| 04:58.42 | masked | yeah or a rootkit an zrtp mtim |
| 04:58.56 | Samot | Do you think your calls are that important? |
| 04:59.00 | masked | no |
| 04:59.04 | Samot | Because you are going way overboard. |
| 04:59.31 | masked | well mate |
| 05:00.04 | masked | you haven't had to type up hours upon hours of telephone intercepts to keep your family and friends out of jail beofre now have you? |
| 05:00.30 | Samot | Oh so this is for illegal activity. |
| 05:00.40 | masked | no |
| 05:02.13 | Samot | I don't think many have had to spend hours doing that. |
| 05:02.50 | masked | well, yes.. |
| 05:02.53 | masked | but. |
| 05:02.59 | masked | more police went to jail than my friends |
| 05:03.10 | Samot | Yay? |
| 05:03.32 | masked | laughs |
| 05:05.35 | Samot | drmessano: |
| 05:09.09 | *** join/#asterisk O47m341 (~Suzeanne@2602:306:3676:c60:6daa:3775:2463:c785) |
| 05:12.15 | drmessano | Yes? |
| 05:13.03 | drmessano | Samot: |
| 05:13.17 | Samot | Hold me gently while I weep. |
| 05:13.22 | Samot | I have to buy a Yealink. |
| 05:13.41 | drmessano | You mean a Sang.. NM |
| 05:14.07 | Samot | lol. |
| 05:14.09 | Samot | Yeah. |
| 05:14.11 | Samot | Er no |
| 05:14.23 | drmessano | NO REZEMBELENZES |
| 05:14.30 | drmessano | PROMISS |
| 05:14.46 | Samot | New client.. |
| 05:14.55 | Samot | Already has 275 users |
| 05:15.09 | Samot | Expanding to another 200 over the next 3 months.. |
| 05:15.27 | Samot | They all have yealinks, so now I need to have one to interop and test things. |
| 05:15.38 | drmessano | Which one are you getting? |
| 05:15.48 | Samot | Me? Low end. |
| 05:16.00 | Samot | 3 lines..probably the T40 |
| 05:16.34 | drmessano | 3 lines just fscks with my OCD so bad |
| 05:16.40 | Samot | It's NAT handling is..... |
| 05:16.51 | drmessano | 2 or 4 |
| 05:16.59 | Samot | They don't have 2 or 4 |
| 05:16.59 | drmessano | 3? ZOMGWTFBBQ |
| 05:17.02 | Samot | 3 or 6 |
| 05:17.04 | Samot | or more |
| 05:17.06 | drmessano | I know |
| 05:17.12 | drmessano | WTF is 3 |
| 05:17.15 | Samot | haha. |
| 05:17.24 | drmessano | Apparently Yealink CANT EVEN |
| 05:19.15 | Samot | It's so awful. |
| 05:19.39 | Samot | All the locations have Cisco RV325's. |
| 05:19.51 | drmessano | Great god |
| 05:20.11 | drmessano | My recommendation |
| 05:20.15 | Samot | Yeah, the NAT is so bad between that and the Yealinks.... |
| 05:20.24 | Samot | They have them set to 30 registers. |
| 05:20.26 | drmessano | Is to get them all on a 127.0.x.x /24 |
| 05:20.31 | drmessano | So they cant be hacked |
| 05:20.35 | Samot | Yeah. |
| 05:21.03 | Samot | I did install a Kamailio server...lol. |
| 05:22.07 | masked | lol |
| 05:22.25 | Samot | So far the phones that are doing pass-thru no longer need to have 30 second registers..NAT issues are pretty much gone.. |
| 05:22.34 | drmessano | have you tried a proxy? |
| 05:22.44 | Samot | Meh, I don't need to proxy the audio. |
| 05:23.00 | drmessano | Thats very narrowminded |
| 05:23.04 | Samot | I could. |
| 05:23.29 | drmessano | I proxy everything |
| 05:23.40 | drmessano | I have an ICMP Proxy to proxy my pings |
| 05:23.45 | drmessano | Because, cant trust |
| 05:24.02 | Samot | Gotta keep those family members outta jail. |
| 05:24.13 | drmessano | Especially Fat Tony |
| 05:24.17 | drmessano | Who did NOTHING wrong |
| 05:24.22 | Samot | Oh yeah. |
| 05:24.26 | drmessano | Despite his mafioso name |
| 05:24.32 | Samot | He was with Little Gino.. |
| 05:24.35 | Samot | At the club. |
| 05:24.47 | Samot | No WHERE near where that thing happened. |
| 05:24.53 | Samot | You know, the thing. |
| 05:24.57 | Samot | That happened. |
| 05:25.07 | Samot | Not where Fat Tony was. |
| 05:25.07 | drmessano | Making calls on our PBX named definitely-not-a-mafia-pbx.mafiapbx.com |
| 05:25.30 | Samot | OMG |
| 05:25.42 | Samot | Wait, it might exist.. |
| 05:25.49 | Samot | Let me check.. |
| 05:26.06 | drmessano | You asshole |
| 05:26.18 | drmessano | Youre gonna steal my domain you squatter SOB |
| 05:26.51 | Samot | Oh not that. |
| 05:27.00 | drmessano | You're gonna wake up with a bed full of Microsoft Lynx ISOs |
| 05:27.01 | drmessano | Oh |
| 05:27.03 | Samot | Allison... |
| 05:27.08 | Samot | IVR greetings.. |
| 05:27.08 | masked | https://au.linkedin.com/in/joelemond |
| 05:27.14 | Samot | All mafia move line classics. |
| 05:27.14 | masked | who's this |
| 05:27.25 | drmessano | I dont follow links from IRC |
| 05:28.45 | Samot | Call up and get Allison with "I know it was you Fredo. You broke my heart...(pause for a few seconds)...broke my heart" |
| 05:29.13 | drmessano | HAH |
| 05:29.23 | Samot | I wonder if Allison would do the Mafia PBX recording pack? |
| 05:29.40 | drmessano | Oh im sure |
| 05:29.55 | igcewieling | Allison will say almost anything. |
| 05:29.59 | drmessano | I really hate working for Trump |
| 05:30.00 | Samot | "Guido is not available, please don't leave a message.." |
| 05:30.08 | drmessano | Im patching these Windows 2003 server |
| 05:30.10 | drmessano | Im patching these Windows 2003 servers |
| 05:31.06 | Samot | "We're sorry but Benny is busy.....sleeping with the fishes. Please 0 to reach the on call Capo" |
| 05:31.12 | Samot | +press |
| 05:31.47 | Samot | Yes, windows servers? |
| 05:32.04 | drmessano | Actually I am installing SCCM on Windows 2003 servers |
| 05:32.17 | Samot | 2003? |
| 05:32.22 | drmessano | Yup |
| 05:32.59 | Samot | Exchange 5 on there? |
| 05:33.04 | drmessano | Ha no |
| 05:33.08 | *** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net) |
| 05:33.21 | Samot | OH FrontPage 2000 server? |
| 05:33.30 | drmessano | I really want to set up an NT server again |
| 05:34.05 | Samot | Good ole NT |
| 05:34.27 | drmessano | So my 2003 DHCP/File Server is like 8 years old now |
| 05:34.41 | drmessano | Reason it's only 8 years old |
| 05:34.48 | drmessano | I had an NT Server that wouldn't die |
| 05:35.07 | drmessano | NT4 on a Poweredge 2500.. Lasted 7 years |
| 05:35.11 | Samot | Yeah, I had one that lasted forever... |
| 05:35.19 | Samot | Yup, about the same run.. |
| 05:35.34 | Samot | I used it to run Platypus. |
| 05:36.03 | Samot | ICVerify iirc... |
| 05:36.09 | Samot | For CC processing. |
| 05:36.54 | drmessano | I miss those days |
| 05:38.44 | Samot | Yeah.. |
| 05:38.47 | Samot | Because now you have |
| 05:38.48 | Samot | https://www.rt.com/viral/364809-pilot-cockpit-sex-pictures/ |
| 05:39.28 | Samot | What I want to know is, where was the co-pilot? From what I understand there always has to be two in the pit.. |
| 05:41.50 | Samot | "Aaaaa, this is your captain..aahhhhh..oooohhh...yeah...speaking...if you...ooo..aaa..oohh..look out the left side of the cabin...ooo..deep...naughty...you'll see the huuuuge landmark...oooooo BIG BEN!...thank you" |
| 05:43.52 | drmessano | Pinging away at his loopback interface |
| 05:43.57 | drmessano | Hope he wore a proxy |
| 05:45.55 | Samot | Well it must not have been enough, looks like he got intercepted. |
| 06:00.15 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 06:09.37 | drmessano | Im trying to save up a little money |
| 07:02.24 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 07:15.42 | masked | lol |
| 07:16.25 | masked | i dont have the money to run 2 vps's and i dont need asterisk listening for sip requests when i have opensips, it defeats the purpose. |
| 07:16.38 | masked | i could run it on the real ip on a different port then block the port |
| 07:16.43 | masked | isn't tht what you would do? |
| 07:27.57 | *** join/#asterisk mirela666 (~mirkob@89.184.168.160) |
| 07:32.50 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:47.02 | *** join/#asterisk defsdoor (~andy@cpc35-sutt4-2-0-cust184.19-1.cable.virginm.net) |
| 07:53.29 | *** join/#asterisk jkroon (~jkroon@dustpuppy.is.co.za) |
| 08:03.45 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 08:10.47 | *** join/#asterisk Rasputin3711 (~Rasputin3@87.255.254.66) |
| 09:06.37 | *** join/#asterisk kako (~Thunderbi@b2b-94-79-159-22.unitymedia.biz) |
| 09:06.55 | kako | Hi, I have two sip clients. (A) can use alaw,g722 and (B) can use g722,g726,speex. In other words, both can do "g722", but when checking it by "sip show channels" I see that (A) uses alaw and (B) g722. Why is asterisk translating instead of both use g722? Is there anywhere a config option to use best common codec if there is a common codec and only do translation if no common was found? |
| 10:03.37 | *** join/#asterisk tirej (~tirej@unaffiliated/tirej) |
| 10:04.03 | *** join/#asterisk jozza (~chatzilla@BSN-77-186-224.static.siol.net) |
| 10:04.34 | jozza | hi all, i have a problem with phone provisioning module on 13.11.2. It often decides that everything is 404 not found. But when i un/load the res_phoneprov module, it starts working again |
| 10:05.07 | jozza | does anyone else experience this? |
| 10:09.02 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 10:12.02 | WIMPy | kako: I think that's still being worked on. So you have to sort codecs manually to make them fit. |
| 10:13.14 | *** join/#asterisk Tiffon (~name@unaffiliated/tiff0n) |
| 10:26.43 | pchero_work | Hi, I have a question regarding queue. Can I make the queue DO NOT ANSWER to the entered call? So, the entered call would be still the ringing. |
| 10:27.30 | WIMPy | That IS how it works. |
| 10:28.51 | pchero_work | Ooops.. I thought the queue answered the entered call immediately. |
| 10:28.59 | pchero_work | ACDÂ (Automatic Call Distributor) distributes incoming calls in the order of arrival to the first available agent. The system answers each call immediately and, if necessary, holds it in a queue until it can be directed to the next available call center agent. Balancing the workload among agents ensures that each caller receives prompt and professional service. - http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf |
| 10:30.06 | WIMPy | Well, I always play a greeting before using Queue(). But I am pretty sure that it doesn't answer without doing so. |
| 10:31.54 | pchero_work | Hm.. Thank you. At least I could try. ;) |
| 10:42.01 | pchero_work | WIMPy: Thanks. You're right. The queue doesn't answer by itself. ;) |
| 10:57.10 | *** join/#asterisk rwsq1 (~rw4c@host-92-25-254-200.as13285.net) |
| 11:10.52 | kako | WIMPy: thanks |
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| 11:20.45 | rwsq1 | hi. I'm trying to build branch 13 on debian jessie. The linker doesn't seem to be find asound symbols in pjproject http://pastebin.com/AKrTdE8D |
| 11:22.48 | rwsq1 | a forget it I found the configure option :) |
| 11:31.31 | rwsq1 | ok so that didn't help. Any ideas? |
| 11:35.45 | rwsq1 | I'm using the bundled pjproject |
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| 12:55.56 | tig | hi all, I don't suppose anyone is using voiptalk with asterisk in the UK |
| 13:02.44 | *** join/#asterisk monsterco (~monsterco@toroon474aw-lp130-01-70-49-228-179.dsl.bell.ca) |
| 13:09.19 | [TK]D-Fender | tig, Does it matter if they are in the UK? |
| 13:11.20 | tig | [TK]D-Fender: not really, I am having some fun with one way audio and I think it might be todo with how their proxy needs to be setup |
| 13:11.45 | tig | just to make things more complex I am using xivo, it is very close to working |
| 13:11.47 | [TK]D-Fender | I'm pretty sure they set up their own proxy, and not you |
| 13:12.10 | tig | It must be a NAT issue (I know it is evil) |
| 13:12.56 | tig | [TK]D-Fender: yes, but trying to work out how to connect to it to help with outgoing calls |
| 13:13.08 | [TK]D-Fender | it is common and no big deal unless your router is doing particularly screwy stuff |
| 13:13.33 | [TK]D-Fender | Show us a call attempt with SIP debug enabled |
| 13:13.34 | [TK]D-Fender | ~pb |
| 13:13.34 | infobot | pastebin is, like, a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 13:13.35 | [TK]D-Fender | ^^^ |
| 13:13.55 | tig | ok will try capturing one now |
| 13:17.41 | tig | [TK]D-Fender: hmmm, not a call yet but this leaps out http://pastebin.com/Q6RETLMR |
| 13:18.20 | [TK]D-Fender | not indicative of anything |
| 13:18.32 | [TK]D-Fender | From: "nest" <sip:nest@192.168.10.36>;tag=as4d8bd3e7 |
| 13:18.50 | [TK]D-Fender | Except a hint that you've misconfigured your system |
| 13:19.09 | [TK]D-Fender | Don't just show the response.. we need to see the the request you send doesn't look like garbage... |
| 13:19.24 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-pvekawstwonxnmne) |
| 13:19.24 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:19.38 | [TK]D-Fender | We don't want to see only the other side saying "bad", we want to see the request that caused that response. |
| 13:20.54 | tig | just trying to wrestle the log into pastebin :) |
| 13:27.28 | aiksa[LV] | Hey guys; Anyone has seen asterisk setting different SSRC value for each RTP package in a call? |
| 13:29.24 | tig | [TK]D-Fender: http://pastebin.com/zyT08qCe <-- line 263 seems to be the start of the call |
| 13:30.30 | [TK]D-Fender | Even outside the call problems are evident |
| 13:30.38 | [TK]D-Fender | [Nov 3 13:14:21] VERBOSE[2147] chan_sip.c: Reliably Transmitting (NAT) to 77.240.48.94:5060: |
| 13:30.38 | [TK]D-Fender | OPTIONS sip:voiptalk.org;user=phone SIP/2.0 |
| 13:30.58 | [TK]D-Fender | #1 No sane provider is expected to be considered to be behind NAT |
| 13:31.03 | [TK]D-Fender | Contact: <sip:nest@192.168.10.36:5060> |
| 13:31.14 | tig | hmmm seeing an anauthorised error in there but it seems to register according to the console |
| 13:31.19 | [TK]D-Fender | #2 you're providing your PRIVATE IP as the return contact and not your WAN IP. |
| 13:31.56 | [TK]D-Fender | <PROTECTED> |
| 13:33.09 | [TK]D-Fender | You're giving them the wrong addresses and you're not trust where they say to send audio. |
| 13:34.16 | [TK]D-Fender | externaddr, localnet, nat <- all need to be set right. directmedia=no <- should be set everywhere so endpoints also don't try to reinvite from behind NAT either |
| 13:34.22 | *** join/#asterisk matrix1233 (~matrix123@197.0.119.228) |
| 13:36.41 | tig | [TK]D-Fender: thanks, some of this is due to me pulling at straws, I took the external IP address out when I read a page suggesting that somewhere. Now I know what I am looking for I will get all that sorted out |
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| 13:37.31 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 13:38.34 | aiksa[LV] | so - for some calls (like 2-3%) I experience a weirdo SSRC values in the RTP packages |
| 13:38.45 | aiksa[LV] | Asterisk is always in the middle of media flow |
| 13:39.09 | aiksa[LV] | so - according to tcpdump: decent SSRC values received from one endpoint |
| 13:39.37 | aiksa[LV] | but when RTP get forwarded further each an every RTP package has different SSRC value |
| 13:39.40 | aiksa[LV] | any ideas? |
| 13:39.52 | aiksa[LV] | "each an every RTP package" - for the same call |
| 13:40.04 | aiksa[LV] | I have never ever seen this before |
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| 14:19.20 | *** part/#asterisk monsterco (~monsterco@toroon474aw-lp130-01-70-49-228-179.dsl.bell.ca) |
| 14:53.14 | tig | hmmm my local phones seem to be trying to register directly with my sip provider, is that normal? |
| 14:55.12 | [TK]D-Fender | They are your phones... why are you pointing them there? |
| 14:56.03 | Samot | Only if you tell them to. |
| 14:58.33 | tig | ah I think they are going via the server but it is still sending the internal IP address to the provider :( |
| 14:59.01 | *** join/#asterisk Penguin (~xwQ5kwYl6@our.systems.are.full.of.penguins.at.penguinsystems.net) |
| 14:59.15 | igcewieling | sounds like a NAT setting isn't set. |
| 15:01.03 | [TK]D-Fender | tig, Where do we see thatt? |
| 15:01.44 | tig | http://pastebin.com/LsaPdVJF |
| 15:02.11 | [TK]D-Fender | Retransmitting #4 (NAT) to 77.240.48.201:5060: |
| 15:02.15 | tig | I think there must be a NAT option somewhere in the xivo gui I am missing |
| 15:02.16 | [TK]D-Fender | Contact: <sip:xivo@192.168.10.36:5060> |
| 15:02.29 | [TK]D-Fender | GUI's are not supported in here |
| 15:03.09 | tig | [TK]D-Fender: understood, was hoping it would steer me in the right direction |
| 15:03.34 | [TK]D-Fender | tig, I already told you the settings (based on anything vaguely recent) that should be configured |
| 15:03.44 | [TK]D-Fender | tig, How you do that in your environment is another matter |
| 15:08.05 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 15:08.05 | *** mode/#asterisk [+o DivideBy0] by ChanServ |
| 15:09.02 | M2K | Hi D-Fender, Thanks fro yesterdays advice by the way, how are you? |
| 15:10.16 | [TK]D-Fender | Still breathing |
| 15:13.57 | *** join/#asterisk RolledGold (47c09026@gateway/web/freenode/ip.71.192.144.38) |
| 15:16.52 | RolledGold | Hello everyone, I have an issue with confbridge not playing 'conf-onlyperson.ulaw' if announce_only_user=yes if I set a user profile in the confbridge () |
| 15:17.14 | RolledGold | IF a don't set a user profile it plays the file. |
| 15:18.16 | RolledGold | on version 11.14.1 |
| 15:19.07 | igcewieling | RolledGold: have you checked the changelogs to see if the issue is already fixed in a later release? |
| 15:20.07 | RolledGold | no but I have crawled issues.asterisk.org |
| 15:21.44 | igcewieling | fixed 4/2015 Confbridge dynamic profiles did not have a default profile unless you explicitly used Set(CONFBRIDGE(bridge,template)=default_bridge). If a template was not set prior to the bridge being created then some options were left with no default values set. |
| 15:21.53 | igcewieling | I'm sure there are others which might apply. |
| 15:22.12 | igcewieling | ^^^ ASTERISK-24749 |
| 15:23.50 | igcewieling | I'm known for refusing to upgrade almost anything unless there is no other choice, but even I am running 11.21.2 |
| 15:25.42 | RolledGold | yes these are production servers as well so upgrades are ugly. |
| 15:25.50 | [TK]D-Fender | half a branch behind, and 11 is NOT getting any more bug-fixes |
| 15:25.58 | [TK]D-Fender | 11 = DISCO BABY |
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| 15:28.00 | *** join/#asterisk tuxian (~tuxian@igilmour.plus.com) |
| 15:31.45 | Samot | Oh yeah, 11 when to SFO last week didn't it? |
| 15:31.57 | [TK]D-Fender | 2 IIRC |
| 15:32.23 | Samot | Oh..Oct 25th. |
| 15:33.18 | RolledGold | I am trying to set a conf user option should that be a bridge option? |
| 15:39.41 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
| 15:42.29 | Samot | RolledGold, show your confbridge.conf |
| 15:42.52 | Samot | And the user profile, as it's set when it doesn't work. |
| 15:43.27 | RolledGold | [agent] type=user music_on_hold_when_empty=no announce_only_user=yes announce_join_leave=no end_marked=yes dtmf_passthrough=yes quiet=yes |
| 15:43.33 | Samot | ~pb |
| 15:43.33 | infobot | pastebin is, like, a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:43.48 | Samot | Clean and readable. |
| 15:45.36 | RolledGold | http://pastebin.ca/3735866 |
| 15:48.57 | RolledGold | the confbridge.conf is the boilerplate with just users added |
| 15:51.58 | Samot | Show the confbridge.conf file, please. |
| 15:55.53 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 15:57.17 | RolledGold | http://pastebin.ca/3735868 |
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| 16:01.17 | tig | [TK]D-Fender: many thanks, I think I have got it sorted now :) |
| 16:03.25 | RolledGold | announce_only_user=yes is a user option and not a bridge (you can set the file to play but not the behavior as to when it plays) |
| 16:03.52 | RolledGold | if i need to set both the user and the bridge that's fine too. |
| 16:04.05 | Samot | Right. It's also on by default. |
| 16:04.22 | Samot | It's the default behaviour. |
| 16:04.29 | *** join/#asterisk tuxd00d (~tuxd00d@24.121.219.231) |
| 16:04.38 | Samot | If you don't want it to announce the user is the only user, you need to set it to no. |
| 16:04.44 | Samot | Otherwise, it's doing what you want by default. |
| 16:05.48 | [TK]D-Fender | tig, Let's see the difference |
| 16:07.17 | RolledGold | I do want it to announce and it is not if I pass the user with exten => _.,n,ConfBridge(${agent_id},,agent,) |
| 16:07.49 | RolledGold | if I just use exten => _.,n,ConfBridge(${agent_id}) it plays |
| 16:18.09 | RolledGold | confbridge.conf user has announce_only_user=yes yet it does not announce if passed as an arg to ConfBridge. |
| 16:20.37 | Samot | Show a call doing this. |
| 16:20.48 | Samot | ~pb |
| 16:20.49 | infobot | pastebin is probably a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:21.39 | *** join/#asterisk tuxd00d (~tuxd00d@24.121.219.231) |
| 16:28.27 | RolledGold | http://pastebin.ca/3735875 |
| 16:28.29 | RolledGold | http://pastebin.ca/3735876 |
| 16:29.53 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
| 16:30.07 | RolledGold | one has audio files the other does not |
| 16:30.09 | RolledGold | <PROTECTED> |
| 16:33.17 | Samot | -- Executing [2@sc_agent_context:4] ConfBridge("SIP/test1-00000003", "2,,sc_agent,") in new stack |
| 16:33.26 | Samot | You do not have a sc_agent profile. |
| 16:35.45 | RolledGold | it's just agent, sorry. |
| 16:36.11 | RolledGold | -- Executing [2@sc_agent_context:4] ConfBridge("SIP/test1-00000003", "2,,agent,") in new stack |
| 16:37.56 | Samot | So a new call that shows that. |
| 16:38.03 | Samot | Because your first one didn't. |
| 16:39.52 | aiksa[LV] | btw what is tghe typical reason for "...requested media update control 26, passing it to ..." flood messages |
| 16:39.52 | aiksa[LV] | chnage in timing? |
| 16:40.38 | M2K | Guys has Anyone come accross Retell call recording? |
| 16:41.24 | RolledGold | http://pastebin.ca/3735884 file plays. |
| 16:41.33 | [TK]D-Fender | M2K, as in? |
| 16:42.04 | M2K | their product range for call recording, analog, isdn, sip |
| 16:42.26 | Samot | RolledGold: I want to see a call that doesn't play it. |
| 16:42.39 | Samot | Like the first one you posted, with the wrong agent profile. |
| 16:42.53 | Samot | So me a call that is trying to load the proper user profile. |
| 16:43.00 | RolledGold | http://pastebin.ca/3735885 |
| 16:43.04 | Samot | s/so/show |
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| 16:43.45 | RolledGold | if I load not profile it plays. Add profile, no play |
| 16:46.39 | RolledGold | user profile [agent] has announce_only_user=yes |
| 16:48.38 | Samot | Stop. |
| 16:48.46 | Samot | Again, that is default to YES |
| 16:48.54 | Samot | You don't need to have it if that's what you want. |
| 16:49.07 | Samot | quiet=yes <-- Fix that |
| 16:49.12 | Samot | quiet=no |
| 16:49.28 | Samot | Otherwise you stop all join/leave announcements outside of PINs, etc. |
| 16:49.40 | Samot | Including not playing that you're the only user. |
| 16:49.40 | [TK]D-Fender | M2K, if that's a productt then I've nott heard of it |
| 16:50.04 | RolledGold | quiet is not just leave/enter? |
| 16:50.16 | Samot | It's the leave/enter announcements. |
| 16:50.22 | Samot | Playing that you are the only user... |
| 16:50.27 | Samot | Is a enter announcement. |
| 16:50.35 | Samot | Is an enter announcement. |
| 16:51.49 | M2K | :D |
| 16:52.42 | Samot | ;quiet=yes ; When enabled enter/leave prompts and user intros are not played. ; There are some prompts, such as the prompt to enter a PIN number, ; that must be played regardless of what this option is set to. ; Off by default |
| 16:52.53 | Samot | It's right there in your conf file. |
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| 17:15.50 | RolledGold | Thanks Samot. |
| 17:16.01 | RolledGold | dev team is happy with that. |
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| 17:24.08 | Samot | I've actually never configured a conference bridge on Asterisk directly... |
| 17:26.59 | Samot | I've always done it from a *gasp* GUI. |
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| 17:39.32 | igcewieling | I've never setup a confbridge via a GUI. Seemed terribly complicated. |
| 17:40.00 | igcewieling | I went the easy route and configured them by hand |
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| 18:57.51 | mmlj4 | igcewieling: ping |
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| 20:23.02 | rmudgett | hi |
| 20:25.00 | rmudgett | file: ping |
| 20:25.03 | rmudgett | test |
| 20:25.05 | rmudgett | ing |
| 20:25.10 | file | pong |
| 20:25.18 | rmudgett | new machine |
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| 21:27.10 | gtjoseph | rmudgett: ping |
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