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| 05:20.30 | ruben23 | hi guys |
| 05:20.50 | ruben23 | anyone familiar with this error ---> installed asterisk 1.8 ---> http://pastebin.com/Xq2xkuxV |
| 05:26.44 | juned | ruben23: g++: Command not found |
| 05:28.23 | ChannelZ | IE your c++ compiler is missing |
| 05:28.51 | WIMPy | IE you need to install an OS before trying to build Asterisk. |
| 05:30.29 | juned | ruben23: seems like you haven't installed Asterisk dependencies |
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| 06:39.30 | MonkehParade | Hi c: |
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| 09:21.02 | telenieko | Hi. I have a few pjsip phones on my asterisk system, and when I establish a call Asterisk always stays "in-the-middle" of the media. How can I know what is preventing asterisk to leave the media and connect the phones directly? any checklist :) |
| 09:34.51 | utrack | Hey! Is there any guide setting up * for absolute noobs? |
| 09:36.53 | utrack | I'm a dev naturally, and out admin does not know a thing about the Aster. It would be used as dumb switch controlled through AMI by our app, pool of internal numbers on one side, around 20-25 external channels to usual phone service. |
| 09:36.57 | utrack | our* |
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| 09:37.44 | utrack | I did set up a test server, however I'm not sure how to work with routing/external channels. |
| 09:47.24 | ChannelZ | ~primer |
| 09:47.24 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
| 09:47.54 | ChannelZ | though it might not go as far as you want |
| 09:48.28 | ChannelZ | I need to add on to it sometime. Time and all that... |
| 09:51.41 | ChannelZ | telenieko, Asterisk isn't a SIP proxy so it generally stays in the media path... but you can look at direct_media for the endpoint(s) config |
| 09:54.20 | utrack | ChannelZ: thank you! |
| 09:55.11 | ChannelZ | sure, hope it helps |
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| 10:50.59 | utrack | ChannelZ: actually, it helps a lot :) thanks again! That's a great starting guide |
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| 11:39.41 | telenieko | ChannelZ: thanks for the tip |
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| 12:14.25 | juned | How to originate calls to two users |
| 12:14.34 | juned | channel originate SIP/1001 extension 1002@my_extension |
| 12:15.06 | juned | this works well with two sip users but what if want to call to mobile numbers using gateway ? |
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| 12:23.28 | stefan27 | those of you that deploy asterisk with sip, do you allow access to <your asterisk> : 5060 from the outside world? |
| 12:24.10 | WIMPy | Where's the point of using VoIP if you don't at least try/allow to use it P2P? |
| 12:25.10 | WIMPy | (not that I see anything the like happening...) |
| 12:25.17 | stefan27 | well you could restrict access to only certain sip proxys or special domains |
| 12:26.10 | WIMPy | I suspect, most people do so. |
| 12:26.54 | stefan27 | I never really had to before, but the amount of unsolicited invite spam has increased the last month |
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| 12:42.47 | juned | How to set caller ID in asterisk CLI originate command ?\ |
| 12:45.21 | acidfoo_ | from what I heard the last time I asked, you can't |
| 12:46.03 | acidfoo_ | but you can use a callfile |
| 12:46.32 | acidfoo_ | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files |
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| 13:20.05 | juned | acidfoo_: Thanks :) |
| 13:21.27 | juned | I want to do something like this |
| 13:21.27 | juned | channel originate SIP/gw/1001 application DIAL SIP/gw/1002 |
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| 14:40.58 | Spengler | anyone else in here use vitelity? |
| 14:42.30 | [TK]D-Fender | probably several |
| 14:43.44 | Spengler | i was just wondering if anyone else was experiencing calls not connecting on their network |
| 14:43.56 | Spengler | intermittently get a fast busy |
| 14:48.51 | smkelly | Exiting! |
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| 15:54.39 | TandyUK | fuck me how stupid are suppliers |
| 15:54.47 | TandyUK | I ordered 5 SIP phones for delivery direct to site, |
| 15:54.59 | TandyUK | asked supplier to clearly label the boxes as 201, 202, 203, etc |
| 15:55.06 | TandyUK | and let me know the MAC address of each phone |
| 15:55.13 | TandyUK | can they do that??? No , fucking retards |
| 15:55.49 | TandyUK | any UK voip hardware suppliers that arent complete morons? |
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| 15:56.31 | robmal | provu? |
| 16:00.11 | TandyUK | omfg 6 emails later thay have managed it! |
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| 16:37.29 | *** join/#asterisk RZer0 (55769ffa@gateway/web/freenode/ip.85.118.159.250) |
| 16:39.14 | RZer0 | Hi all, I am unsing asterisk 11.11, and I am trying to figure out how to set the sip uri 'telephone-context=+foo'. Is it possible in Asterisk? |
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| 16:47.11 | *** join/#asterisk gaps (7aa4e714@gateway/web/freenode/ip.122.164.231.20) |
| 16:47.21 | [TK]D-Fender | as in? |
| 16:47.50 | gaps | am using Asterisk 11.2.1, connected to Verizon ISP via IPSEC tunnel for testing, incoming works fine, but outgoing results to 408 request timeout |
| 16:48.57 | [TK]D-Fender | Tha version is ancient, insecure and unsupported. We highly recommend you upgrade immediately. |
| 16:49.10 | [TK]D-Fender | 11.21.2 (2016/02/11) <-------- |
| 16:49.57 | gaps | [TK]D-Fender: Ok |
| 16:50.18 | [TK]D-Fender | I'd also recommend looking at the SIP debug for that call to see what is actually happening |
| 16:50.49 | gaps | in sip debug i see 408 request time out from Verizon server after 100 Trying |
| 16:50.51 | *** join/#asterisk juned (~juned@43.250.164.138) |
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| 16:51.50 | [TK]D-Fender | Show us. |
| 16:52.20 | gaps | [TK]D-Fender: In Asterisk console i get Busy/Congested message whereas in the packet capture I see 408... one sec... |
| 16:52.36 | [TK]D-Fender | "sip set debug on" <- |
| 16:52.43 | [TK]D-Fender | Nothing external |
| 16:52.49 | *** join/#asterisk saint_ (~saint_@unaffiliated/saint-/x-0540772) |
| 16:53.46 | saint_ | hi all - i am resuming my Asterisk 13 installation. Since I am using pjproject , do i still need chan_sip driver ? |
| 16:54.00 | [TK]D-Fender | no. |
| 16:54.08 | saint_ | ok, thanks |
| 16:54.15 | [TK]D-Fender | You could simply noload it |
| 16:54.20 | [TK]D-Fender | jsut like always |
| 16:54.52 | keithf | gaps i saw something similar from one of our remote phones going over the internet to our pbx. his ISP was not forwarding the udp traffic back to us. and he got the 408 each time could be different but thats what happened. he had to pay extra because this was his home. DAMN time warner |
| 16:55.00 | saint_ | instead of having a huge noload file , i'd rather simply uncheck it from menuselect. it's just two differents way of doing clean things, no ? |
| 16:55.55 | saint_ | keithf are you serious ? They charge you more depending on what kind of usage he does from his internet connection ? |
| 16:56.04 | saint_ | THAT'S BS !!!! |
| 16:56.10 | newtonr | saint_, Yeah you can just avoid building the module |
| 16:56.14 | keithf | because it is SIP and he could be hosting his own phone service |
| 16:56.21 | [TK]D-Fender | Why would you have a huge nolad list? |
| 16:56.30 | keithf | and they have voip so to discourage it they blocked / will unblock for a nominal fee |
| 16:56.35 | [TK]D-Fender | How many things do you actually feel the need to prevent any chance of loading? |
| 16:56.39 | saint_ | [TK]D-Fender because there is a ton i am not using |
| 16:56.43 | gaps | http://pastebin.com/W5ULZFBv |
| 16:56.46 | [TK]D-Fender | like? |
| 16:57.34 | [TK]D-Fender | chan_sip has at least valid REASONS for not loading (potential security risk), but besides API & channel drivers, what qualifies to be part of this "tons" you're talking about? |
| 16:58.13 | *** join/#asterisk d00gster (~d00gster@unaffiliated/d00gster) |
| 16:58.52 | gaps | keithf: whats your PBX IP? |
| 16:59.03 | saint_ | [TK]D-Fender like 99% of the cdr, anything related to agent / call center , anything related to not sip (iax, gsm, ulaw), the festival, dictate, image, etc etc ... i start small, then go the way around and add whatever module i need so i understand how each work. right now, i'm doing a strict basic minimal install. |
| 16:59.27 | [TK]D-Fender | <PROTECTED> |
| 16:59.48 | [TK]D-Fender | Ah yes, slowly breaking this as you realize you need them. |
| 16:59.50 | keithf | externally it is a 173 ip |
| 17:00.05 | saint_ | yes |
| 17:00.05 | [TK]D-Fender | This gets pretty sad ... Gentoo-style... |
| 17:00.10 | saint_ | no |
| 17:00.16 | keithf | in that debug i saw a 401 unauthorized but i didnt see a 408 |
| 17:00.18 | saint_ | it's a different way of doing it |
| 17:01.04 | keithf | there it goes my eyes are getting old :) |
| 17:01.46 | saint_ | newtonr thank you |
| 17:01.57 | juned | Guys I am getting this error while making call using WebRTC, I'm using SIpml5 |
| 17:01.59 | juned | DTLS failure occurred on RTP instance '0x7f6474070468' due to reason 'no shared cipher', terminating error |
| 17:02.19 | juned | RTCP Read error: Unspecified. Hanging up. |
| 17:03.21 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 17:06.41 | gaps | 173 am not using |
| 17:06.41 | gaps | I mean am not sending any request |
| 17:08.19 | gaps | keithf: after 100 trying you can see the 408 request timeout |
| 17:09.13 | keithf | yeah i sees it gaps |
| 17:09.39 | gaps | keithf: yeah.. since everything looks normal.... not sure why we are getting 408.... |
| 17:10.50 | keithf | there is no way to get a capture from both ends would love to see if its making it round trip |
| 17:11.48 | gaps | keithf: round trip between my PBX and verizon SBC? |
| 17:12.44 | keithf | whats the ip of your sbc and whos the manufacturer of the sbc |
| 17:12.48 | keithf | we have sangoma sbc's |
| 17:13.05 | gaps | SBC is from Verizon.... |
| 17:13.25 | keithf | i know with us if the caller ID outbound isnt correct verizon blocks the call from being made |
| 17:13.27 | gaps | myside Asterisk is 121.242.139.57 |
| 17:13.33 | keithf | its annoying |
| 17:13.42 | gaps | am using Verizon's caller id given by them |
| 17:13.45 | keithf | do you guys have the same thing |
| 17:13.52 | gaps | and this is Verizons SBC - 63.79.179.178 |
| 17:14.07 | keithf | ok and you dont have your asterisk side to over write the cid ? just trying to think outloud |
| 17:14.19 | keithf | whats 10.60.10.15 ? |
| 17:14.34 | gaps | thats my laptop ip address... where my softphone installed |
| 17:15.41 | keithf | ok and its using the extension 3081 |
| 17:16.01 | gaps | yes correct |
| 17:17.33 | keithf | stupid question but the trunk between the asterisk and the sbc is up right |
| 17:17.43 | gaps | hahaha yeah its up |
| 17:17.50 | gaps | because my inbound works fine |
| 17:20.41 | RZer0 | Is your route config set correctly for the VPN, route -n |
| 17:21.40 | gaps | yes my route goes to my switch and then to firewall |
| 17:21.40 | gaps | from my firewall the VPN tunnel is established |
| 17:21.43 | gaps | both way we are able to use its established |
| 17:22.16 | keithf | what firewall are you using ? |
| 17:22.20 | keithf | sonic wall ? |
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| 17:24.35 | gaps | yes |
| 17:24.37 | gaps | sonicWALL |
| 17:25.14 | keithf | there is a setting on the sonicwall let me see if i can remember we had to set |
| 17:25.27 | gaps | you mean the SIP Transformation? |
| 17:25.43 | keithf | I think thats what it was i know we had a TON of issues getting it set just right |
| 17:26.25 | gaps | yeah we tried that setting.... if we enable that... inbound also won't work |
| 17:27.01 | gaps | really its strange that ... i have been struggling for couple of days on this... |
| 17:27.09 | gaps | even verizon also stuck here |
| 17:27.33 | keithf | we had the same thing i think your sonic wall is the culprit |
| 17:27.40 | keithf | have you run a trace on the sonic wall |
| 17:27.45 | keithf | to see exactly whats flowing thru it |
| 17:28.07 | gaps | hmm we tried to see any drop packets in sonicWALL but there are no packet drops... |
| 17:28.38 | keithf | we had to get sonic wall on the phone and into their device to fix it and it took about a month before they got it working. i dont remember what they did our networking team had all of the conversations with them |
| 17:28.52 | gaps | oh ok |
| 17:29.10 | Chubby | is this the right room to ask a question about fail2ban? |
| 17:29.50 | keithf | Chubby: i would just ask |
| 17:29.57 | keithf | if not someone will point you in the right direction |
| 17:30.34 | Chubby | having an issue with fail2ban not starting, it was running until 5AM or so this morning then just stopped |
| 17:30.37 | Chubby | if I try to start it via fail2ban-client it throws an error ERROR /etc/fail2ban/filter.d/freepbx.conf and /etc/fail2ban/filter.d/freepbx.local do not exist ERROR Unable to read the filter ERROR Errors in jail 'pbx-gui'. Skipping... |
| 17:30.47 | Chubby | log files don't contain anything useful that I can see |
| 17:31.32 | keithf | and the files it say do not exist are in the directory ? |
| 17:31.59 | RZer0 | <PROTECTED> |
| 17:32.04 | Chubby | they are not, but I don't know very much about fail2ban so I'm not sure if it was a link that obliterated itself or there was something else going on |
| 17:33.21 | Chubby | or if it should be pointed to asterisk.conf which does exist |
| 17:33.28 | keithf | as RZer0 said try run a yum update |
| 17:33.35 | keithf | looking at fail2ban on one of my servers |
| 17:33.40 | keithf | the freepbx.conf is in the directory |
| 17:33.49 | keithf | something may have blown your file out |
| 17:33.56 | Chubby | I wonder what happened |
| 17:34.01 | file | raises eyebrow |
| 17:34.11 | Chubby | I should in theory just be able to copy freepbx.conf from my running instance yeah? |
| 17:34.24 | keithf | are they mirrors ? |
| 17:34.29 | keithf | if so then i dont see why not |
| 17:34.57 | Chubby | lol yum update <3 |
| 17:35.26 | Chubby | That was out of left field for my approach and it worked, thank you very very much |
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| 17:39.41 | ruben23 | hi guys any idea when im on CLI> iax2 show peers -------------> No such command 'iax2 show peers' (type 'core show help iax2 show' for other possible commands) |
| 17:40.14 | ruben23 | CLI> sip show peers ----------------------------> No such command 'sip show peers' (type 'core show help sip show' for other possible commands) |
| 17:40.21 | ruben23 | any idea guys why they are not working |
| 17:40.46 | keithf | is the module running ? |
| 17:41.20 | keithf | module load chan_iax2.so |
| 17:41.27 | keithf | i would run on the cli |
| 17:42.03 | keithf | we run dundi and i get the same error when i try to do a dundi lookup and dundi module is not running |
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| 18:15.11 | gaps | Hi all after removing the E164 format as callerid it started working for me |
| 18:18.25 | keithf | good to know |
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| 18:53.30 | RadicalDev | Anyone know the latest version of Dahdi that wanpipe 7.0.16 supports? |
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| 20:29.16 | Tachoh | is there any kind of limit to the number of active AMI connections I can have open at any given time? |
| 20:30.50 | Tachoh | oh i see i can use a proxy |
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| 21:03.01 | RadicalDev | Anyone know why my kernel panics when I disconnect a crossover cable from the CPE T1 interface using Asterisk 11, Wanpipe 7.0.16, dahdi current, libpri current and Centos 6.7? |
| 21:03.54 | RadicalDev | it only happens while asterisk is running, maybe HWEC? |
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