| 00:02.33 | *** join/#asterisk brettnem (~bnemeroff@207.183.229.86) |
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| 00:29.14 | *** join/#asterisk rapha (~rapha@unaffiliated/rapha) |
| 00:29.22 | rapha | hi! |
| 00:30.22 | rapha | is it possible to use asterisk with chan_lantiq on a small router with TAE ports as an ATA connecting to another asterisk instance connected to the internet? |
| 00:34.53 | *** part/#asterisk kharwell (kharwell@nat/digium/x-ezazibwuuszwvopq) |
| 01:01.22 | [TK]D-Fender | TAE? |
| 01:01.33 | *** join/#asterisk Dovid (~dovid@ool-45738ae3.dyn.optonline.net) |
| 01:03.15 | Samot | German thing. |
| 01:03.47 | Samot | It's akin to a RJ11 |
| 01:20.44 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:20.44 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); 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 |
| 01:22.40 | *** join/#asterisk ddickenson (sid179041@gateway/web/irccloud.com/x-huckpxwvtqusveht) |
| 01:26.31 | rapha | [TK]D-Fender: looks much different though: https://upload.wikimedia.org/wikipedia/commons/thumb/d/d7/TAE-Stecker-F.jpg/250px-TAE-Stecker-F.jpg |
| 02:27.09 | KNERD | must be something recent |
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| 03:45.08 | drmessano | KNERD: Pretty recent |
| 03:45.12 | drmessano | 1980's |
| 03:47.32 | KNERD | If you got it, Flaunt it! |
| 03:49.41 | KNERD | Last time I was in Germany , such connectiosn did not exist and was called Deutsche Bundespost |
| 03:54.07 | Samot | So pre-1986? |
| 03:54.44 | Samot | And there were no connectors pre TAE. |
| 03:54.55 | Samot | Literally. |
| 03:57.06 | Samot | I had a German exchange student at our high school during 1989. She was amazed we could put a phone in any room. |
| 03:57.29 | Samot | Because it still was a new thing for her. |
| 04:06.02 | KNERD | thereabouts |
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| 10:28.18 | Alblasco1702 | Hello is it possible for asterisk to send answered elsewhere only if it is answhered else where? |
| 10:28.18 | Alblasco1702 | And where is it set? on the dial plan? |
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| 12:20.39 | Sprinterfreak | Samot. Me again. To my issue with pjsip. Now figured out that my sip provider does sometimes not respond with an ACK on an Status: 200 response after an INVITE |
| 12:22.52 | Sprinterfreak | Then asterisk repeats the response a few times, re-registers , repeats the 200 response a few times more and then finally asterisk gave up with BYE. |
| 12:23.58 | Sprinterfreak | Guess what? After this my sip provider simply replies with an 200 OK |
| 12:24.40 | sekil | ACK on 200 OK is mandatory |
| 12:25.01 | sekil | to establish transaction |
| 12:25.24 | Samot | No. |
| 12:25.29 | Samot | That is completely wrong. |
| 12:25.37 | Samot | You send them a 200 OK and they should ACK it. |
| 12:25.45 | Samot | If they are not doing that get a real provider. |
| 12:25.58 | Sprinterfreak | They do that sometimes |
| 12:26.09 | Samot | It's not a "sometimes" |
| 12:26.12 | Sprinterfreak | With chan_sip everytime |
| 12:26.14 | Samot | There are standards. |
| 12:26.24 | Samot | Chan_SIP and Chan_PJSIP are SIP |
| 12:26.25 | Samot | Period. |
| 12:26.36 | Samot | Just different stacks/drivers. |
| 12:26.42 | Samot | How they do SIP is still the same. |
| 12:27.51 | Sprinterfreak | But with chan_sip it works 100% stable, with pjsip 10% the calls get established because my 200 after an INVITE don't get ACKed |
| 12:28.11 | sekil | Alblasco1702: should be sent on Cancel Reason.. |
| 12:28.30 | sekil | Alblasco1702: and as per standards...endpoint should not mark a missed call then.. |
| 12:29.44 | Samot | Sprinterfreak: So every INVITE they send you over a chan_SIP trunk works without issue? |
| 12:29.50 | file | the ACK is sent to the Contact address in the 200 OK, have you confirmed it contains the correct information? |
| 12:29.54 | Samot | They always return an ACK to your 200 OK? |
| 12:30.04 | Sprinterfreak | Samot: Yes |
| 12:30.43 | Samot | But somehow with PJSIP they decide not to? |
| 12:30.52 | Samot | Thats not how that works |
| 12:31.21 | Samot | They have no clue what driver you are using. |
| 12:32.00 | Samot | The otherside doesnt care if its chan_sip or chan_pjsip |
| 12:32.22 | Sprinterfreak | Right. Thats my problem acually. It didn't work properly. |
| 12:33.00 | Samot | OK, who is this provider? |
| 12:33.06 | Sprinterfreak | sipgate |
| 12:33.08 | file | Sprinterfreak: my comment was to you |
| 12:33.20 | sekil | Sprinterfreak: what file said.. |
| 12:33.41 | Sprinterfreak | moment pls |
| 12:36.20 | Sprinterfreak | file: sekil: identical contact in both of my 200 replies. One time I get a proper ACK and the call get established, the other time I get no reply |
| 12:36.49 | file | is it correct to reach the machine? is it the same as chan_sip? |
| 12:37.28 | Sprinterfreak | Its absolutely correct. Sometimes it works... |
| 12:37.44 | file | I understand sometimes it works. |
| 12:37.59 | file | The other option is to do a call from chan_sip and one from chan_pjsip and then compare the signaling. |
| 12:42.54 | Sprinterfreak | wait. no. There was a typo the external_signaling_address |
| 12:43.17 | Sprinterfreak | wich is in my case a dns name |
| 12:44.52 | Sprinterfreak | But why does it then sometimes work and sometimes not? |
| 12:45.35 | file | I don't know what sipgate would be doing. |
| 12:46.14 | file | It's up to them to send the ACK, they should do it according to the Contact but if NAT traversal is enabled then it could go to the source IP address/port instead |
| 12:47.44 | Sprinterfreak | Wich could be the case if my correct address was still in their cache anywhere |
| 12:49.10 | Sprinterfreak | Fixed that and will see if that solves the "somtimes working" state |
| 12:50.43 | Sprinterfreak | but in both cases, working and not working, my correct ip was in the contact-header of my 200 replies |
| 12:57.30 | Sprinterfreak | No. Still some calls don't get an ack |
| 12:59.56 | Sprinterfreak | Absoluty odd. Their 200 OK after the BYE due to the timeout gets through everytime |
| 13:01.57 | file | then you'd need to compare the signaling between chan_sip and chan_pjsip to see if there is a difference |
| 13:02.49 | Sprinterfreak | with chan_sip i use their proxy for signaling. But this should work anyway |
| 13:03.13 | file | ...so they are configured differently? |
| 13:03.53 | Sprinterfreak | Yes they are. But pjsip sadly does not accept the sip.conf format |
| 13:04.08 | file | outbound_proxy=sip:IP address\;lr |
| 13:04.14 | file | or host. |
| 13:04.56 | Sprinterfreak | in wich contexts? |
| 13:05.12 | file | the endpoint in the case of calls. |
| 13:06.29 | *** join/#asterisk k-man (~jason@unaffiliated/k-man) |
| 13:10.16 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 13:10.45 | Sprinterfreak | file: even with proxy the same |
| 13:10.58 | file | then you'd need to do as I said, compare |
| 13:11.11 | Samot | OK, show it. |
| 13:11.23 | Samot | Show full attempt with SIP debug/pjsip logger/whatever. |
| 13:11.32 | Samot | Let's see this from the INVITE to the the end. |
| 13:14.44 | *** join/#asterisk somepoortech (~somepoort@72-0-128-179.static.firstlight.net) |
| 13:22.53 | Sprinterfreak | Samot: There is a working one https://pastebin.com/GXCV41Dp |
| 13:23.54 | *** join/#asterisk miralin (~Thunderbi@194.8.128.76) |
| 13:24.41 | Samot | Do you have one of the actual issue? |
| 13:24.46 | Samot | That's really the one I want to see. |
| 13:33.15 | Sprinterfreak | Samot: Here is a not working one https://pastebin.com/dZnwN5RH |
| 13:33.58 | file | those are the same. |
| 13:33.59 | Samot | How is that not working? |
| 13:34.23 | Samot | Yeah. |
| 13:34.28 | Samot | Those are the same exact call. |
| 13:39.17 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:40.23 | Sprinterfreak | Samot: Updated the not working one. C+P-Fault |
| 13:40.59 | Samot | Dude. |
| 13:41.01 | Samot | Seriously. |
| 13:41.19 | Samot | We're trying to figure out why crap isn't being responded/routed properly |
| 13:41.39 | Samot | And you have sanitized these debugs. |
| 13:41.43 | Samot | They are useless. |
| 13:43.06 | Samot | And are these PJSIP debugs? |
| 13:43.12 | Samot | Both Chan_SIP? |
| 13:43.24 | Samot | The working one Chan_SIP and the broken one PJSIP? |
| 13:44.11 | Sprinterfreak | This would take me more time to do because of the whole reconfiguration |
| 13:45.07 | Samot | I'm asking wht these are. |
| 13:45.19 | Samot | What are we looking at? |
| 13:45.41 | Samot | PJSIP or Chan_SIP? |
| 13:49.17 | Samot | See in the working debug you have this: c=IN IP4 b.core.saenet.de |
| 13:49.34 | Samot | In the non working debug you have this: c=IN IP4 b.my.address.net |
| 13:50.03 | Samot | Now was that because they were actually different or because you just made it that? |
| 13:54.19 | *** join/#asterisk mahafyi (~chatzilla@182.65.10.214) |
| 13:54.59 | Samot | I mean, I see a difference but I would like some answers before giving a conclusion. |
| 13:55.24 | Sprinterfreak | Samot the working was pjsip too |
| 13:55.46 | Samot | So the SDP information was the same? |
| 13:55.51 | Samot | Between the two? |
| 13:56.00 | Samot | And you just obscured them differently? |
| 13:56.02 | Sprinterfreak | I've updated the not working one again. This time the pjsip debug output from asterisk cli |
| 13:57.02 | Samot | And this one doesn't work either? |
| 13:57.15 | Samot | Because this is completely different. |
| 13:58.13 | Sprinterfreak | Samot: This time the debug output from the asterisk cli. The prevous one where made with wireshark |
| 13:58.27 | Samot | OK |
| 13:58.34 | Samot | So all three of these debugs... |
| 13:58.43 | Samot | They are sourcing the calls from three different sources. |
| 13:58.52 | Samot | The second one, the one you just updated.. |
| 13:58.59 | Samot | Had a Diversion tag in it from them.. |
| 13:59.02 | Samot | For some reason. |
| 13:59.16 | Samot | But the two failed calls are from two different systems. |
| 13:59.36 | Samot | And those two systems are not where the good call came from it came from a third system all together. |
| 13:59.45 | Samot | Look at your 4th VIA: header in their invite. |
| 14:00.04 | Samot | That's where it actually came from, the other three (the ones with Record Routes) are the proxies. |
| 14:01.00 | Sprinterfreak | Oh god damed. Yes, in the original ones these addresses where all the same and correspond to my system |
| 14:02.01 | Samot | Via: SIP/2.0/UDP 217.116.117.8:5060;branch=z9hG4bK72a75155 <- Non working |
| 14:02.18 | Samot | Contact: <sip:01788675631@217.116.117.8:5060> <-- Non working |
| 14:02.19 | Samot | Now.. |
| 14:02.40 | Samot | Via: SIP/2.0/UDP 217.10.77.45:5060;branch=z9hG4bK1dffade2 <-- Working |
| 14:02.53 | Samot | Contact: <sip:80.111.222.333:5061> <-- Working |
| 14:03.17 | Samot | So in the working one, it sources from another IP but the Contact it claims to be from is a completely different IP. |
| 14:03.51 | Samot | I've come to the conclusion that Germany cannot SIP properly. |
| 14:03.51 | Sprinterfreak | Oh, yes. That could be the proxy... |
| 14:04.23 | Sprinterfreak | Wich we tried earlier without any change in behavior |
| 14:04.38 | *** join/#asterisk saint_ (~saint_@unaffiliated/saint-/x-0540772) |
| 14:06.37 | *** join/#asterisk retentiveboy (~retentive@c-73-82-30-193.hsd1.ga.comcast.net) |
| 14:09.12 | sekil | private IPs in RR...grrr |
| 14:10.00 | *** join/#asterisk chandoo (~chandoo@pool-74-105-13-92.nwrknj.fios.verizon.net) |
| 14:11.06 | Sprinterfreak | sekil: how to avoid this? |
| 14:11.51 | sekil | Sprinterfreak: it's not an error per-se...it's the providers internal network.. |
| 14:16.05 | *** join/#asterisk rwb (~Thunderbi@74.85.159.242) |
| 14:16.38 | sekil | Sprinterfreak: they should hide their internal topology... |
| 14:16.46 | sekil | Sprinterfreak: nothing to do with you.. |
| 14:18.32 | Sprinterfreak | Well in my sdp is a private ip4 10.88.0.1 . Wich also was in there if I got a working one |
| 14:19.12 | Sprinterfreak | o=- 2000795089 2000795092 IN IP4 10.88.0.1 |
| 14:19.37 | sekil | Sprinterfreak: that's wrong from your side |
| 14:20.06 | Sprinterfreak | How to change this? |
| 14:20.46 | sekil | Sprinterfreak: you should advertise your public IP address |
| 14:21.48 | Sprinterfreak | Thats also done in the same packet |
| 14:21.50 | Sprinterfreak | v=0 |
| 14:21.50 | Sprinterfreak | o=- 2000795089 2000795092 IN IP4 10.88.0.1 |
| 14:21.50 | Sprinterfreak | s=Asterisk |
| 14:21.50 | Sprinterfreak | c=IN IP4 b.core.saenet.de |
| 14:26.20 | Sprinterfreak | This I think is not the problem. Because its exactly the same in both cases. Working and not working |
| 14:26.49 | sekil | it's a not a problem...it's just wrong practice imo |
| 14:27.26 | Sprinterfreak | what does this o= stand for? |
| 14:28.03 | file | origin like, not really used, in later versions of Asterisk it is substituted for external |
| 14:28.06 | file | er line |
| 14:29.15 | Sprinterfreak | So the important part is |
| 14:29.22 | Sprinterfreak | c=IN IP4 b.core.saenet.de |
| 14:29.22 | Sprinterfreak | m=audio 30060 RTP/AVP 8 101 |
| 14:29.39 | Sprinterfreak | Wich is totally correct |
| 14:30.06 | Sprinterfreak | but does only sometimes work |
| 14:30.55 | sekil | FQDN in c is legal I think...although uncommon.. |
| 14:31.13 | file | latest version of Asterisk puts an IP address there |
| 14:31.28 | Samot | Asterisk is sending the same details. |
| 14:31.38 | Samot | For a working call and a non working call. |
| 14:31.55 | sekil | yeah...it's legal definitely.. |
| 14:31.55 | Samot | What needs to happen now is more testing |
| 14:32.07 | Samot | Do all working calls come from the same server? |
| 14:32.15 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-157-202.satx.res.rr.com) |
| 14:32.16 | Samot | Or do other servees work? |
| 14:32.30 | Samot | Do all non working calls come from the same servers? |
| 14:32.41 | Samot | Three debugs.. |
| 14:32.59 | Samot | All three incoming calls sourced from three different places |
| 14:33.07 | Samot | 1 worked |
| 14:34.22 | Sprinterfreak | That could be since we tried teh outbound_proxy |
| 14:34.25 | sekil | why is contact IP different? |
| 14:34.30 | rapha | so, after last nights discussion about the age of TAE connectors and how great of an invention connectors as such were... |
| 14:34.36 | sekil | you are registering from different IPs? |
| 14:34.45 | rapha | is it possible to use asterisk as an ATA connected to another asterisk? |
| 14:35.05 | Samot | Asterisk can support FXS/FXO cards. |
| 14:35.09 | Samot | But it is not an ATA. |
| 14:35.26 | Sprinterfreak | sekil: No. Mangled this |
| 14:35.29 | Samot | It would be a PBX with an analog card. |
| 14:35.31 | rapha | Samot: so it can't connect to a SIP server upstream? |
| 14:35.39 | Samot | Sure.. |
| 14:35.43 | rapha | oh okay |
| 14:35.45 | sekil | rapha: yes |
| 14:35.47 | Samot | I just said it supports FXS/FXO |
| 14:35.56 | Samot | But it is not an ATA. |
| 14:35.59 | Samot | It's a PBX |
| 14:36.04 | sekil | Sprinterfreak: mangled how? |
| 14:36.05 | rapha | i'm talking about a small router with analog phone ports, an ethernet port, and lantiq chipset |
| 14:36.09 | Samot | Those are two highly and completely different things. |
| 14:36.19 | Samot | No. |
| 14:36.24 | rapha | no to what? |
| 14:36.37 | Samot | rapha: If you don't know this stuff you're not going to turn some old router into a PBX |
| 14:36.46 | rapha | *sigh* |
| 14:36.49 | Samot | Just not happening. |
| 14:36.50 | Sprinterfreak | replaced my ip with 111.222.333 and one time my fqdn with my.address.net |
| 14:36.58 | Samot | 8:49:17 AM S<Samot> See in the working debug you have this: c=IN IP4 b.core.saenet.de |
| 14:36.58 | Samot | 8:49:34 AM S<Samot> In the non working debug you have this: c=IN IP4 b.my.address.net |
| 14:36.58 | Samot | 8:50:03 AM S<Samot> Now was that because they were actually different or because you just made it that? |
| 14:37.04 | Samot | ^^ hence me asking that |
| 14:37.14 | Samot | Again.. |
| 14:37.27 | sekil | Sprinterfreak: you should really give us clean logs/traces... |
| 14:37.33 | Samot | Giving us sanitized information leads to misinformed conclusions. |
| 14:37.36 | sekil | Sprinterfreak: you're misleading us..at least me.. |
| 14:37.40 | Samot | I already bitched. |
| 14:38.33 | rapha | Samot: in spite of my inadequateness as a human and all that, asterisk, if handled properly by an experienced operator such as yourself, possesses the ability to run on a lantiq chipset and connect an analog phone to an upstream sip provider? |
| 14:39.00 | Sprinterfreak | Ive verified this a few times in my wireshark captures. The addresses between working and not working are exactly identical |
| 14:39.16 | sekil | Sprinterfreak: so I'm losing interest... |
| 14:40.36 | Sprinterfreak | sekil: Well. Me too. Sadly the only thing I could do now is a roolback to chan_sip |
| 14:41.01 | Samot | rapha: I said "IF you don't know this stuff" |
| 14:41.04 | Samot | IF |
| 14:41.09 | Samot | I never said you didn't. |
| 14:41.30 | mneth1 | I upgraded to 13.18.3 to correct MixMonitor stereo recordings having channels out of sync but the issue is still present. This was the bug fix I thought would take care of it: https://issues.asterisk.org/jira/browse/ASTERISK-26875?jql=text%20~%20%22mixmonitor%20sync%22 |
| 14:41.31 | Samot | I never questioned your adequacy in any of this. |
| 14:41.53 | mneth1 | are there any other known fixes for getting MixMonitor stereo to have aligned recordings even during packet loss? I'm using the b option as well. |
| 14:42.00 | Samot | So don't take that attitude when you're asking for help on a project that makes no sense or has any real viability. Especially if you don't have clue one. |
| 14:44.42 | rapha | And you might want to start thinking about your own attitude as to why it is that you want to help people at all. Because one might suspect, hammering out phrases like these, that the motivation may be to fulfill the cravings of your inadequate ego for attention and worthfulness. One of the operators may kickban me now if they so please, I'll be in the basement. |
| 14:45.23 | sekil | rapha: helping is above the way how does one help.. |
| 14:45.27 | Samot | rapha: How much do you know about Asterisk? |
| 14:45.34 | sekil | rapha: you should be grateful in the first place |
| 14:46.04 | Samot | rapha: Or FXS/FXO modules? TAE connectors? |
| 14:46.38 | Samot | rapha: Or having to wire and/or solder chips/circuits to make this happen? |
| 14:46.44 | Samot | How much of that do you know? |
| 14:46.55 | [TK]D-Fender | <rapha> is it possible to use asterisk as an ATA connected to another asterisk? <- if * can handle the interface on that hardware then * takes calls from/to it, and processes them however you tell it to |
| 14:47.09 | [TK]D-Fender | rapha, Do you have it compiling and seeing your hardware? |
| 14:47.36 | Samot | Really? It's a router. |
| 14:48.24 | Samot | Let's make sure there is a proper knowledge/skillset in place before we start talking about how to turn an old router into a PBX |
| 14:48.27 | sekil | Sprinterfreak: I wanted to help...but these traces are really useless |
| 14:48.31 | Samot | With FXS/FXO hand offs |
| 14:49.19 | [TK]D-Fender | I don't care if it's a router. |
| 14:49.26 | Samot | Actually, the request was to make it an ATA |
| 14:49.26 | [TK]D-Fender | Has he gotten * compiled on it? |
| 14:49.31 | Samot | OK |
| 14:49.34 | Samot | Have fun with it. |
| 14:49.37 | [TK]D-Fender | If not... sit back, grab some popcorn and wait. |
| 14:49.48 | Samot | Encouraging bad ideas is never good. |
| 14:50.01 | Samot | That's how you get PBX In a Flash |
| 14:50.03 | [TK]D-Fender | Don't need fun, or deep involvement just to ask if he's gotten the first few steps done |
| 14:50.22 | Samot | You didn't even know what a TAE connector was. |
| 14:50.34 | [TK]D-Fender | Yeah this is probably more effort than it's worth... just as long as I don't invest disproportionately :) |
| 14:51.08 | [TK]D-Fender | <Samot> That's how you get PBX In a Flash <- PBXPILAF :) |
| 14:54.31 | Samot | And now nothing. |
| 14:54.51 | Samot | So the answers to my questions are 99% a "no" |
| 14:56.34 | [TK]D-Fender | "I'll be in the basement" <- we're in extended overtime.... |
| 15:02.19 | *** join/#asterisk miralin (~Thunderbi@194.8.128.76) |
| 15:06.56 | *** join/#asterisk kharwell (kharwell@nat/digium/x-ezvxpewmfucfjmgt) |
| 15:06.56 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:09.50 | Sprinterfreak | sekil: Samot: Checked this again. The real difference is really that Asterisk with chan_sip locally looks up the ip and sends the IP in SDP und PJSIP sends the dns |
| 15:10.31 | Sprinterfreak | It seems sipgate doesn't like to lookup fqdn's |
| 15:11.30 | Sprinterfreak | If I hardcode my dynamic ip into the pjsip.conf it seems to work consistently so far |
| 15:11.32 | Samot | The SDP has nothing to do with the ACK |
| 15:11.42 | Samot | That's just for the media |
| 15:11.50 | Samot | Does not impact signalling |
| 15:12.05 | sekil | Sprinterfreak: ACK should be back though...but as I said FQDN in c is very uncommon... |
| 15:12.48 | Samot | Not really |
| 15:12.49 | Sprinterfreak | They seem to silently drop the sdp answer |
| 15:12.51 | sekil | Sprinterfreak: maybe they condition on SDP body somehow...who knows.. |
| 15:13.03 | Samot | PBX's behind NAT on DHCP WANs. |
| 15:13.17 | Samot | Why? |
| 15:13.35 | Samot | The SDP can be different than the source/signaling IP |
| 15:13.59 | Samot | And if they don't support FQDN's wouldn't that be in their docs? |
| 15:14.10 | Samot | Like in the setup docs "Please don't use FQDNs" |
| 15:14.50 | sekil | Samot: one uses stun client for these setups |
| 15:15.15 | Sprinterfreak | pjsip doesn't implement stun |
| 15:15.22 | Samot | Not all ISP/ITSPs run STUN clients. |
| 15:15.39 | sekil | Samot: pbx should use stun if under nat imo.. |
| 15:15.45 | sekil | Samot: not itsp.. |
| 15:15.53 | Samot | That doesn't happen. |
| 15:16.22 | sekil | Sprinterfreak: so check with them about FQDN |
| 15:18.23 | sekil | Sprinterfreak: also see if you can use an STUN client on pjsip if available.. |
| 15:27.01 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 15:27.01 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 15:30.01 | *** join/#asterisk Janos (~Janos@181.194.24.51) |
| 15:31.17 | Sprinterfreak | sekil: Cannot realy find any detailed informations from them |
| 15:33.00 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 15:33.43 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 15:33.48 | Janos | hey there, got a problem with an asterisk, trying to register with my sip provider, I have the following in my sip.conf "register => user:secret@sip.provider.com" but I see no traffic to/from the provider ip and "sip show registry" does not show anything, any ideas what I could be missing ? |
| 15:35.23 | Sprinterfreak | Janos: Don't use chan_sip (and sip.conf) anymore. Its legacy and unsecure in some cases |
| 15:36.00 | Janos | Sprinterfreak, yeah I'm starting to switch to pjsip but it will take some time :) |
| 15:36.16 | Janos | I'm guessing pjsip was your recommendation yes ? |
| 15:36.47 | Sprinterfreak | Yes. Me too. But the configuration of pjsip is really awful |
| 15:38.05 | Sprinterfreak | It states its more secure in NAT scenarios. Shure. It doesn't implement neccessary functionality for that |
| 15:38.59 | sibiria | i found it rather messy and annoying as well, compared to chan_sip. thankfully the "pjsip wizard" helped a bit |
| 15:43.42 | Janos | sibiria, what is this pjsip wizard ? |
| 15:43.44 | *** join/#asterisk bford (d8cff501@gateway/web/freenode/ip.216.207.245.1) |
| 15:43.45 | *** mode/#asterisk [+o bford] by ChanServ |
| 15:44.01 | *** join/#asterisk aways (~aways___@unaffiliated/aways) |
| 15:44.15 | file | https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Wizard |
| 15:44.22 | Janos | thanks |
| 15:45.37 | *** join/#asterisk GeneralSpongebob (~Spongebob@cpc127156-mapp14-2-0-cust83.12-4.cable.virginm.net) |
| 15:49.05 | Sprinterfreak | Janos: And pjsip doesn't resolv domainnames in external_media_address wich seems to be incompatible with some sip providers |
| 15:49.43 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 15:50.03 | Janos | I'm starting to think I should stay with chan_sip :P |
| 15:50.18 | file | Sprinterfreak: it does in the latest version. |
| 15:51.07 | Sprinterfreak | Hoping that debian maintainers shortly release this |
| 15:51.15 | Sprinterfreak | Think not :( |
| 15:55.41 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-gmxonfkmgugxxpxu) |
| 15:55.41 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 16:00.24 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 16:00.32 | Sprinterfreak | But until that I'm stuck to a bodge where I hook a script to the ip change and sed the address parameters in pjsip.conf manually |
| 16:21.01 | *** join/#asterisk Worldexe (~Worldexe@95-107-33-134.dsl.orel.ru) |
| 16:24.46 | *** join/#asterisk qakhan (~qakhan@50-204-254-11-static.hfc.comcastbusiness.net) |
| 16:28.30 | qakhan | Hi all, |
| 16:30.22 | qakhan | there are 10 agents 1001-1010 in a queue, if queue receive a call from 3048931223 then send that call only to 1st 5 agents 1001-1005. |
| 16:30.26 | qakhan | how can we do this |
| 16:34.15 | [TK]D-Fender | Go read the sample configs. |
| 16:34.22 | [TK]D-Fender | They describe how to use priorities |
| 16:35.01 | qakhan | in dialplan? |
| 16:36.38 | [TK]D-Fender | no |
| 16:39.15 | qakhan | in queue.conf? |
| 16:40.26 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 16:50.03 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 16:50.35 | *** join/#asterisk SoBlindWolf (~SoBlindWo@go.pcshost.co) |
| 16:52.52 | *** join/#asterisk dnit (673399be@gateway/web/freenode/ip.103.51.153.190) |
| 16:53.32 | dnit | Hi |
| 16:53.39 | dnit | my astreisk logs are flooded with Local/15841080-4829@tempinbound-1-ringing-0000731e;1 requested media update control 26, passing it to SIP/Xo_main-00003bb5 |
| 16:54.01 | dnit | and users are not able to hear the ivr and other audio. |
| 16:55.25 | [TK]D-Fender | Go look at the SIP debug for that call |
| 17:05.01 | dnit | [TK]D-Fender : I have a lot of calls like that. |
| 17:05.23 | dnit | and how can I see the sip debug in production. |
| 17:06.14 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 17:08.02 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 17:09.42 | [TK]D-Fender | place a call. look at it |
| 17:09.45 | [TK]D-Fender | you can't go back in time |
| 17:22.19 | *** join/#asterisk Jesterboxboy (~Thunderbi@84-115-150-8.cable.dynamic.surfer.at) |
| 17:26.31 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 18:35.30 | Worldexe | you can enable sip debug only for some ip/peer |
| 18:35.39 | Worldexe | so you will get limited amount of traffic |
| 18:35.54 | Worldexe | or just tcpdump it and investigate that dump |
| 18:43.03 | *** join/#asterisk jamesaxl (~James_Axl@109.172.62.242) |
| 18:43.49 | jamesaxl | HEllo |
| 18:44.31 | jamesaxl | is firewall enough to protect asterisk? |
| 18:45.41 | *** join/#asterisk Dovid (~dovid@ool-45738ae3.dyn.optonline.net) |
| 18:52.07 | Samot | As far as? |
| 18:52.42 | Samot | I mean if you have all ips blocked except for the ones you need/want, then yeah. |
| 18:53.42 | Samot | If you have to let unknowns in, then you need to make some more complex rules and perhaps some extra layers since you are making compromises. |
| 19:05.24 | [TK]D-Fender | jamesaxl> is firewall enough to protect asterisk? <- how? From what? |
| 19:08.14 | jamesaxl | [TK]D-Fender: from hacker how wants to test default context and also from hackers who send a loop authentication |
| 19:09.52 | jamesaxl | Samot: I understand you, thank you very much |
| 19:19.18 | *** join/#asterisk nny (ad5db800@gateway/web/cgi-irc/kiwiirc.com/ip.173.93.184.0) |
| 19:21.08 | nny | Figured i'd pop in here cause this is a bit of a head scratcher. Over the weekend a client replaced their firewall with an Untangled device. Ever since I am getting tons of registration spam in the logs for all extension, with no UNREACHABLE first. Also any type of SIP phone dial out takes 10-15 seconds to establish (and hit asterisk console). Late |
| 19:21.08 | nny | ncy is fine, nothing has changed in the switches (which are basic/no vlan). I am gonna go do some testing tonight but any conjecture opinion or discussion is welcomed |
| 19:21.53 | nny | dial out includes peer to peer etc.they hit dial and it just sits there says it's dialing but nothing in console. I am gonna do some sip debugging but it feels like something is hijacking the sip traffic |
| 19:24.32 | *** join/#asterisk jkroon (~jkroon@165.16.204.171) |
| 19:25.48 | nny | think i found it Correct auth, but based on stale nonce received from |
| 19:26.06 | *** join/#asterisk Iamnach0 (~Iamnacho@ip72-213-56-35.om.om.cox.net) |
| 19:35.27 | *** join/#asterisk clarjon1 (~clarjon1@unaffiliated/clarjon1) |
| 19:35.31 | Janos | is there a way to log the asterisk AMI communications, preferably to it's own log file ? |
| 19:39.33 | Samot | nny: This is 100% Untangled. |
| 19:41.00 | nny | Hmm wonder if touch on all config files in the tftp directory will trick the phones into reloading them/rebooting all quickly |
| 19:41.54 | Samot | No. |
| 19:42.16 | Samot | You have to send a NOTIFY event to tell them to resync/reboot |
| 19:43.03 | Samot | The phones don't look at file timestamps. |
| 19:43.19 | Samot | They pull the config and basically do a diff on that and the running config. |
| 19:43.35 | Samot | If there is a difference, they take it and then reboot/resync if needed. |
| 19:44.09 | Hsilamot | Janos: what are AMI communications? |
| 19:44.30 | Samot | The devices going UNREACHABLE means that Asterisk isn't getting a response to the Keep-Alive OPTIONS sent to the devices. |
| 19:44.37 | Janos | all commands send/rec by the asterisk manager interface |
| 19:44.57 | Samot | Inbound calls not working, having delays with responses for outbound calls...this is all the Untangle firewall. |
| 19:45.13 | Hsilamot | Janos: if i wanted to log them, i would make a simple bridge, pass-trough the comms and log them myself |
| 19:46.40 | Janos | well AMI allows many actions, not all of them are calls, for example remove agent from a queue, not sure your suggestion would work on those cases |
| 19:47.47 | Janos | one suggestion on the forums is to rise the verbose level to 10, that´s suppose to work, but I guess the log would get crazy big |
| 19:48.00 | Hsilamot | Why not? it's a simple bridge, basically you make a small program to listen on port x, set the AMI server (asterisk, whatever) to port x+1 on loopback interface, you forward everything that comes from port x to port x+1 and biceversa, and log whatever you want from them |
| 19:48.55 | Hsilamot | i have done it many times with all kind of applications, whenever i need to see the communications or log them to see whats happening |
| 19:49.43 | nny | @Samot close but not quite. This is all local communications, as in extension to extension for testing. Granted outbound does it too, but that's not what I am seeing. Also not going UNREACHABLE. It seems this is a nonce issue after firewall swamp out as described here http://lists.digium.com/pipermail/asterisk-users/2009-April/230435.html |
| 19:49.49 | nny | swmap/swap |
| 19:51.08 | Samot | That's a 9 year old issue. |
| 19:52.15 | Samot | And it's also something that happens on various SIP servers. |
| 19:52.18 | Samot | Not just Asterisk |
| 19:52.28 | Samot | It triggers a 401 challenge. |
| 19:52.41 | nny | Samot: gotcha but the nonce spam is real and constant. Just gonna reboot phones first to see if it quells the spam. Going after smoking guns atm |
| 19:53.00 | Samot | It's just a warning. |
| 19:53.25 | Samot | To let you know, they are registering, the auth is good but the register/location is expired. |
| 19:53.31 | *** join/#asterisk mbecroft (~user@ak2.becroft.co.nz) |
| 19:53.36 | Samot | So it triggers a new auth |
| 19:53.39 | Samot | for a new session |
| 19:53.44 | Samot | new expiry time |
| 19:54.23 | nny | yeah. |
| 19:54.26 | Samot | They put in Untangle |
| 19:54.32 | Samot | Ever since then X has happened. |
| 19:54.37 | Samot | that's how you've stated it |
| 19:54.43 | nny | Yeah I disabled any SIP nonsense too. |
| 19:54.45 | Samot | That means, Untangle is the cause. |
| 19:54.50 | nny | I 100% agree |
| 19:55.51 | Samot | So there is no need to do stuff on the Asterisk side. |
| 19:55.53 | Samot | It works. |
| 20:05.51 | *** join/#asterisk elcontrastador (~textual@70-90-215-98-BusName-ca.sacra.hfc.comcastbusiness.net) |
| 20:06.55 | *** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust177.19-1.cable.virginm.net) |
| 20:11.55 | drmessano | Untangle seems very millenial |
| 20:12.09 | Samot | Rofl. |
| 20:13.29 | *** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust177.19-1.cable.virginm.net) |
| 20:16.42 | nny | hahaha |
| 20:16.48 | nny | not my choice :D |
| 20:17.05 | nny | i guess Lit Firewall 420 isn't out yet |
| 20:21.12 | nny | fun fact millenial is born after like 82 and on which means half the people mocking it are technically part of it' |
| 20:21.32 | drmessano | You can add apps to NG Firewall at any time. Simply complete the purchase of the app you want, then assign it to your Untangle instance from your My Account page. |
| 20:21.48 | drmessano | Sounds like a Sangoma product |
| 20:23.02 | Samot | http://hw-static.worldstarhiphop.com/pics/images/tp/1pyyia.jpg |
| 20:24.02 | drmessano | Samot: Yeah yeah, wait until they try to sell you a cloud hosted firewall |
| 20:36.01 | nny | I am not used to seeing so many channels in SIP SHOW CHANNELS, is this registration for hints or something else? https://pastebin.com/FKTS4pUn |
| 20:38.58 | *** join/#asterisk areski (~areski@37.223.2.207) |
| 20:40.05 | [TK]D-Fender | hints/vm subscription |
| 20:43.32 | nny | thanks |
| 20:46.21 | *** join/#asterisk tzafrir (~tzafrir@62-90-199-247.barak.net.il) |
| 21:08.30 | *** join/#asterisk Chotaire (chotaire@unaffiliated/chotaire) |
| 21:24.03 | *** join/#asterisk josefig (~josefig@unaffiliated/josefig) |
| 21:25.02 | josefig | one question, i need to setup asterisk to place 2 calls by our predictive, but i dont know where to set the option that when they are placed 2 calls, if 1 is answered the other hangup, i remember i needed to change but i dont rememeber where |
| 21:42.41 | kfife | Gosub has a ${GOSUB_RETVAL}. Very functional. Nice. Nice. |
| 21:42.55 | kfife | tips his hat |
| 21:45.48 | kfife | Looks like I was missing a parentesis: same=>n,ExecIf(${RECORDING}?Gosub(send-email,s,1(${DB(DN-EMAIL/${CALLERID(num)})},"Rec: ${STRFTIME(%c %Z)}","Attached.\n\nFr:${CALLERID(num)}\nTo:${OCN}\n${STRFTIME(%c %Z)}\n\",${TEMP-REC}):System(rm -I ${TEMP-REC}))) |
| 21:49.19 | *** join/#asterisk infobot (ibot@rikers.org) |
| 21:49.19 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); 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 |
| 21:54.56 | *** join/#asterisk defsdoor (~andy@cpc120600-sutt6-2-0-cust177.19-1.cable.virginm.net) |
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| 23:20.10 | *** join/#asterisk Cory (~Cory@unaffiliated/cory) |
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| 23:39.55 | *** join/#asterisk salviadud (~ralfalfa@200.188.129.226) |
| 23:40.29 | salviadud | I got a question related to P.A. systems. Is there an easy way to incorporate one with asterisk? |
| 23:41.04 | salviadud | I'm at a factory right now, and they got this old Panasonic KX-POS90 |
| 23:41.48 | salviadud | It barely works, I dunno who programmed it, so I was wondering about replacing it with some digium cards, or an E1, depends on how many lines they use, can't say for sure. |
| 23:41.58 | salviadud | But they dial an extension to make public announcements on floor. |
| 23:42.30 | salviadud | So, an extension heads out to a mixer, and the mixer does the rest. |
| 23:43.10 | salviadud | Now, there is also a laptop they used to fire up for music. But the supervisor got angry because of the bad taste in music. |
| 23:43.34 | salviadud | I could just install a softphone and auto-answer and that should go into the mixer. |
| 23:43.45 | salviadud | Comments are welcome. |
| 23:45.53 | Iamnacho | salviadud: I think there are devices you can buy that are SIP to Analog-sound |
| 23:46.30 | Iamnacho | so you can go Astersik -> SIP -> Analog Sound -> PA Amplifier |
| 23:51.58 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 23:56.58 | salviadud | Iamnacho, any examples? |
| 23:58.57 | salviadud | just googled something |
| 23:59.03 | salviadud | I'll keep looking, gotta go |
