| 00:28.58 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:28.58 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.14.0 (2017/02/13), 11.25.1 (2016/12/08), Standard: 14.3.0 (2017/02/13); 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 |
| 00:31.21 | vader- | Anyone have any suggestions for running asterisk on vmware esx? I seem to be having some timing issues in an extremely small environment. The ESX box current has only the one VM running. It doesn't seem to be struggling in any department, CPU, Memory, I/O, etc. But voicemail, MOH and all seem to be very choppy. Calls seem to be semi ok |
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| 05:16.41 | tuxd00d | vader-_: Have you tried it on OpenVZ? |
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| 05:34.59 | vader- | I haven't... I had an ESX server setup already. My end goal is probably AWS |
| 05:38.46 | drmessano | AWS is going to suck even more |
| 05:39.08 | drmessano | Try Linode or Vultr |
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| 06:26.25 | tuxd00d | vader-: AWS used to have a latency issue with RTP, but Iâve been told it works well now, but I havenât tried it myself recently. |
| 06:31.38 | drmessano | AWS has a problem with overselling |
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| 06:46.30 | igcewieling | heh, we decomissioned our last ESXi server today. |
| 06:49.17 | drmessano | He probably has an old copy of ESX and a keygen from a torrent |
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| 08:59.46 | guest1010 | Does anyone know a softphone for linux that can handle the god damn ringinuse bug that has been following me all since 1.8 up to latest 11? Meaning that agents being a member of more than 1 queue can recieve calls when busy. |
| 09:00.08 | guest1010 | Any devs for ring.cx speak up! |
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| 09:11.38 | guest1010 | Also, I am wondering about sip show channelstats. What does it display? When I look at it I can see no disturbance towards the trunk, but towards the clients on 192.168.0/24 it shows heavy jitter. http://codepad.org/9ixGcDTK |
| 09:11.40 | guest1010 | Does this say that the connection between the asterisk and the agents are faulty? And the connection towards the trunk OK? |
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| 09:39.52 | guest1010 | or maybe it displays legs? |
| 09:39.59 | guest1010 | One outbound and one towards the agent |
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| 10:39.00 | catphish | is parsing SIP headers as complicaed as it looks? |
| 10:40.27 | adnauseum | looking at call in cli, how does one determine what trunk was used to make the call? |
| 10:40.41 | adnauseum | err not trunk, route |
| 10:55.32 | detha | catphish: no, it's more complicated than that |
| 10:55.45 | catphish | detha: i feared as much :) |
| 10:56.16 | catphish | just trying to get my head around the structured fields, they have so many options |
| 10:57.16 | detha | parse what you can, ignore what you don't understand/don't want to handle |
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| 10:58.27 | catphish | that's mostly what i want to do, but the issue is getting the information i want regardless of the format |
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| 11:03.03 | todwetsprock | Hello, quesition: I get the following mistake Ignoring call to '55513161' on span 2. Its not in the MSN list:_1316N,_108185Z _108985Z,_108987N,1552 . Here my pastebin: https://paste.pound-python.org/show/Oa1S9oenOBAhdlG8fNY0/ . Ideas someone ? |
| 11:05.12 | todwetsprock | as of now, i notice in anomymizing my post theres of course "N" doesnt fit "1" . but that is not the point |
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| 12:53.02 | kannan | hello all. i would like to know of recommendations for a good router that can handle upto 200 concurrent VoIP SIP calls between asterisk servers. |
| 12:53.40 | [TK]D-Fender | todwetsprock, it doesn't match any of those |
| 12:54.04 | [TK]D-Fender | 200 calls between just 2 servers? |
| 12:55.05 | kannan | yes between 2 servers. |
| 12:55.32 | [TK]D-Fender | The is kinda big for IAX2. |
| 12:55.38 | kannan | i want to setup a IPSec VPN site to site connectivity |
| 12:55.41 | [TK]D-Fender | though the savings would add up quite a bit |
| 12:55.54 | [TK]D-Fender | I'm partial to MikroTik personally |
| 12:56.11 | kannan | MicroTek ? |
| 12:56.36 | [TK]D-Fender | MikroTik |
| 12:56.38 | kannan | ok ok i see MicroTik |
| 12:57.08 | [TK]D-Fender | I own 2 x RB2011 series routers (1 home, 1 office) |
| 12:57.51 | kannan | i have to check if it support IPSec VPN |
| 12:58.32 | kannan | I want to connect to an asterisk VM in google compute engine, where this is the only VPN that is supported (outside of doing an OpenVPN on an instance of course) |
| 13:01.35 | [TK]D-Fender | it does |
| 13:01.41 | [TK]D-Fender | and 5 other kinds |
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| 13:25.55 | adnauseum | outbound route seems to be ignored even when placed at top of outbound routes list. Checked matched pattern many times, no mistakes. |
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| 13:27.20 | adnauseum | tried dialing tf numbers which apply to other outbound routes and they all work as expected. |
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| 13:55.26 | Samot | Yay. |
| 13:58.33 | [TK]D-Fender | That sounds kinda generic |
| 13:58.42 | [TK]D-Fender | *I'M* a community |
| 13:58.54 | [TK]D-Fender | has you all outnumbered |
| 13:58.54 | Samot | Of one? |
| 14:10.48 | adnauseum | 1866NXXXXXX is in matched pattern of toll-free outbound route, extension is blocked. Dial the toll-free number from blocked extension and call is answered. |
| 14:10.54 | adnauseum | wtf |
| 14:11.01 | adnauseum | why is this happening |
| 14:11.02 | adnauseum | hmm |
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| 14:14.41 | Samot | Not sure how the extension is blocked but really, there aren't "Outbound Routes" in Asterisk. |
| 14:17.08 | adnauseum | every outbound route works properly except toll-free. I'm stumped. Done lots of searches and haven't found any concrete answers |
| 14:19.08 | Samot | Actually sounds like it's working, dial 1866NXXXXXX and call goes out trunk. Sounds working to me. |
| 14:19.29 | adnauseum | but the extension is blocked. |
| 14:19.39 | Samot | How is it blocked? |
| 14:19.41 | adnauseum | put it this way |
| 14:20.08 | adnauseum | if i block the ext in north-america outbound, call does not go to destination |
| 14:20.13 | adnauseum | same with pinset |
| 14:20.22 | adnauseum | its just the toll-free outbound route |
| 14:20.39 | Samot | How is it blocked? |
| 14:20.46 | adnauseum | deleted toll-free, recreated, asterisk -rx 'dialplan reload' no help |
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| 14:21.22 | adnauseum | its blocked from the module |
| 14:21.32 | adnauseum | forgot the exact name |
| 14:22.03 | adnauseum | extension routing? |
| 14:22.39 | Samot | Sounds like a FreePBX issue. |
| 14:22.49 | Samot | looks at the room name. Yup, #asterisk |
| 14:23.21 | adnauseum | your funny |
| 14:23.28 | adnauseum | but i am not laughing |
| 14:24.46 | Samot | I am. Just a little. |
| 14:24.52 | Samot | More like a chuckle |
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| 14:43.45 | *** topic/#asterisk by file -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.14.1 (2017/04/04), 11.25.1 (2016/12/08), Standard: 14.3.1 (2017/04/04); 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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| 14:57.24 | adnauseum | Samot: blaze studios is you |
| 14:57.27 | adnauseum | ? |
| 14:58.25 | Samot | It's one of my oldest domains.. |
| 14:59.13 | adnauseum | you still work it? |
| 14:59.27 | Samot | The domain? Meh, it's just a domain. |
| 15:00.42 | adnauseum | https://www.upwork.com/o/profiles/users/_~012ed8f50eb8da6aa4/ |
| 15:01.06 | Samot | OK. |
| 15:02.14 | Samot | So you're stalking me? |
| 15:02.49 | adnauseum | no |
| 15:02.52 | adnauseum | not at all |
| 15:04.54 | adnauseum | the person I work with sometimes is in this channel but currently /away |
| 15:06.21 | adnauseum | this damm iptv i wish they would get their act together |
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| 18:02.15 | Foxi352 | ^^ maybe important: Asterisk 14.3.1 on Debian 8 |
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| 18:05.35 | orn | Are there any other flags needed than R for Queue to generate a ringback instead of MOH when an agent's line is actually ringing? |
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| 18:07.19 | orn | I'm on 1.8.15, and it doesn't seem to do anything |
| 18:07.47 | orn | (the agent is a Local channel, maybe that is why?) |
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| 18:10.46 | [TK]D-Fender | Most likely you put the option in the wrong place |
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| 18:12.03 | orn | Hmmm, OK... it seems that when a local channel rings, the R option does not do anything. |
| 18:12.15 | orn | I registered a SIP client and once that was ringing, i got ringback instead. |
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| 18:46.24 | Kobaz | 35892017-04-04 14:14:18.33203910.0.90.610.0.90.1RTP223216130PT=DynamicRTP-Type-127, SSRC=0x5FE90D5A, Seq=58115, Time=3631420123 |
| 18:46.35 | Kobaz | so, RTP DTMF, payload type 127 |
| 18:46.54 | Kobaz | aaand asterisk does not generate a DTMF manager event |
| 18:47.21 | Kobaz | asterisk -rx "sip show peer 4277" | grep -i dtmf DTMFmode : rfc2833 |
| 18:47.34 | Kobaz | 4277 = 10.0.90.6 |
| 18:52.33 | *** join/#asterisk scgm11_ (~scgm11@r186-52-85-139.dialup.adsl.anteldata.net.uy) |
| 18:52.49 | Kobaz | 35892017-04-04 14:14:18.33203910.0.90.610.0.90.1RTP EVENT223216130Payload type=RTP Event, DTMF Five 5 (end) |
| 18:53.04 | Kobaz | stuff like that tells me that DTMF is making through to the server just fine |
| 18:57.46 | Kobaz | the SDP in the INVITE as well as the OK. contains Media Format: DynamicRTP-Type-127 |
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| 19:34.40 | Kobaz | where's TK-Fender when you need em |
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| 20:03.37 | Kobaz | [2017-04-04 16:03:32.526] [C-00001581] Sent RTP P2P packet to 10.0.90.6:2250 (type 00, len 000160) |
| 20:03.41 | Kobaz | i'm realizing this is the problem |
| 20:03.53 | Kobaz | asterisk is doing P2P rtp despite having directmedia off |
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| 20:42.29 | saint_ | is NAT supposed to worked between 2 private network, without using a SBC (not even talking aboutasterisk, but about sip itself) |
| 20:43.11 | Samot | I'm not sure what that means. |
| 20:48.23 | saint_ | i have one way audio in my lab. my coworker is telling me i need a SBC because I use 2 NAT between 2 private networks. |
| 20:48.42 | saint_ | I thought SBC were only required when one deal with public/private networks. |
| 20:48.46 | Samot | That's not entirely true at all. |
| 20:48.50 | Samot | No. |
| 20:49.00 | Samot | There are various case uses for SBC/Proxies. |
| 20:49.21 | Samot | You have to make sure that both sides have their NAT rules correct. |
| 20:49.34 | Kobaz | throwing in an SBC does not magically fix nat issues |
| 20:49.41 | Samot | No, it doesn't. |
| 20:49.54 | Samot | Specially if the NAT issue is in the router/firewall |
| 20:50.13 | saint_ | his point was you cant use NAT for SIP on 2 private networks. |
| 20:50.20 | saint_ | I do not agree with that. |
| 20:50.33 | Samot | He has no idea about how NAT works. |
| 20:50.36 | Samot | So ignore him. |
| 20:51.04 | Samot | Or her.. |
| 20:51.05 | Samot | It |
| 20:51.16 | Samot | I don't want to upset drmessano by assuming genders. |
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| 20:59.46 | drmessano | #Triggered |
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