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| 07:43.42 | Ionic | is an USB ISDN TE adapter enough to create a bridge for my internal ISDN network to SIP? |
| 07:43.57 | Ionic | (I hope that question isn't too generic) |
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| 08:29.11 | WIMPy | Yes |
| 08:53.05 | Ionic | okay, thanks :) |
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| 10:05.46 | penguin_ice | good morning all!! |
| 10:06.05 | penguin_ice | wishes everyone a great and wonderful Sunday day! |
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| 11:34.16 | dan_j | Asterisk v13 seems to set the src cdr field to equal CALLERID(name). This did not occur in asterisk v11. Is there anyway to restore the v11 functionality? |
| 11:37.54 | dan_j | actually, that should read CALLERID(num) |
| 11:39.22 | dan_j | Alternatively, is there an way to add a new field to the CDR (using an ODBC backend) so that I store my own data? I'm already using userfield for something else. |
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| 13:38.30 | qakhan | strang thing happened yesterday. i had problem with outbound calls |
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| 13:39.02 | qakhan | i was getting message No audio format found to offer. Cancelling call |
| 13:39.58 | qakhan | [TK]D-Fender asked me to check sip show pree preename |
| 13:41.09 | qakhan | when i checked there was Codecs : (nothing) in sip show peer |
| 13:41.39 | qakhan | but in sip config there was disallow = all and allow = ulaw |
| 13:42.35 | qakhan | i reloaded the asterisk many time, reloaded the sip. but still Codecs : (nothing) |
| 13:46.14 | qakhan | then i removed the [vitelity-outbound] config (from line 1165 in sip.conf) and recreate after some lines (1178 in sip.conf) |
| 13:46.26 | qakhan | then it started working |
| 13:47.53 | qakhan | i am curious how these changes made differnece ? |
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| 19:50.14 | v0lZy | hello |
| 19:50.40 | v0lZy | Quick question ... I have 2 locations and a site-2-site tunnel between them |
| 19:51.01 | v0lZy | one side has asterisk in the same network as the phone |
| 19:51.04 | v0lZy | phones* |
| 19:51.14 | v0lZy | the other side registers to said asterisk |
| 19:51.25 | v0lZy | I can make incoming and outgoing calls |
| 19:51.34 | v0lZy | but I'm only getting incoming audio |
| 19:51.38 | v0lZy | (in both cases) |
| 19:51.57 | v0lZy | I suspect this might have to do with outgoing port randomization or something |
| 19:52.10 | v0lZy | but then again |
| 19:52.15 | v0lZy | theres no NAT between the two sites |
| 19:52.26 | v0lZy | its a VPN tunnel with pure routing |
| 19:55.40 | [TK]D-Fender | Then there shouldn't be a randomization issue if that's straight routing |
| 19:56.12 | v0lZy | Ok, that eliminates one possibility |
| 19:56.20 | v0lZy | the next possibility is the firewall then I guess |
| 19:56.53 | v0lZy | from asterisk box, i can ping the phone at the remote end |
| 19:57.13 | [TK]D-Fender | I'm not seeing any debug to validate that you haven't made a mistake directly in your config |
| 19:57.17 | [TK]D-Fender | and ping means nothing |
| 19:57.23 | [TK]D-Fender | this is one-way audio |
| 19:57.27 | [TK]D-Fender | the SIP end made it |
| 19:57.53 | v0lZy | well, im saying ping because im adding allow all rules on the firewall |
| 19:57.59 | v0lZy | just trying to eliminate it as being the issue |
| 19:58.45 | [TK]D-Fender | invalid test |
| 19:58.53 | [TK]D-Fender | ping is not sip/rtp |
| 19:58.56 | [TK]D-Fender | and you aren't loking at the call |
| 20:00.28 | v0lZy | [TK]D-Fender: Yes, conceded that ping != sip or RTP, but obviously SIP is going through if my remote phone can ring another phone, and can be ringed back in turn. RTP is arriving to the remote phone, but RTP from the remote phone isnt arriving back |
| 20:00.43 | v0lZy | that results in 1 way audio, right? |
| 20:01.07 | [TK]D-Fender | lack of RTP... is lack of audio |
| 20:02.27 | v0lZy | yeah, but its a lack only in one direction, regardless of whether its the remote phone calling out or being called |
| 20:02.48 | v0lZy | so somethings wrong with delivery of RTP from phone to asterisk, obviously |
| 20:02.52 | [TK]D-Fender | Why are we not looking at the call already? |
| 20:02.59 | v0lZy | or from asterisk to the other phone |
| 20:03.12 | v0lZy | welll |
| 20:03.16 | v0lZy | sip packets? |
| 20:03.37 | [TK]D-Fender | * CLI w/ sip debug |
| 20:03.47 | [TK]D-Fender | The thing that proves you aren't passing it garbage |
| 20:03.58 | [TK]D-Fender | and are sending the right things |
| 20:04.21 | v0lZy | hm, ok, sec |
| 20:07.22 | v0lZy | argh, putty and CLI buffer... |
| 20:07.26 | v0lZy | this is a bit ocmplicated to capture. |
| 20:07.28 | v0lZy | one sec |
| 20:10.01 | v0lZy | ok |
| 20:10.04 | v0lZy | got it captured... |
| 20:11.26 | v0lZy | --- SIP read from UDP:10.222.224.116:5060 ---> |
| 20:11.26 | v0lZy | INVITE sip:0030380002@192.168.1.3 SIP/2.0 |
| 20:11.47 | v0lZy | I made a call with the remote phone to my mobile phone |
| 20:11.52 | [TK]D-Fender | PASTEBIN is your friend |
| 20:11.55 | v0lZy | 10.222.224.116 is the remote phone |
| 20:12.08 | v0lZy | 192.168.1.3 is asterisk |
| 20:12.18 | [TK]D-Fender | those 2 lines don't say anything of value yet |
| 20:12.25 | [TK]D-Fender | stop focussing on2 dumb lines and show the WHOLE thing |
| 20:13.55 | [TK]D-Fender | ~pb |
| 20:14.00 | 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. |
| 20:14.01 | [TK]D-Fender | ^^^ |
| 20:14.09 | v0lZy | http://pastebin.com/CJjMkFQz |
| 20:14.33 | [TK]D-Fender | What part of WHOLE CALL are you having trouble with? |
| 20:17.28 | v0lZy | have to mask some IPs, company policy... |
| 20:17.33 | v0lZy | gimme a sec |
| 20:18.36 | [TK]D-Fender | Don't |
| 20:18.43 | [TK]D-Fender | that's known as 'FUCKING THE EVIDENCE" |
| 20:19.29 | [TK]D-Fender | do You just said this was all VPN on LAN subnets |
| 20:19.56 | [TK]D-Fender | there should be ZERO outside accessible info in there anyway |
| 20:20.06 | v0lZy | yeah it is, but when i do a sip debug, i capture a lot of other stuff too |
| 20:20.13 | v0lZy | its a production system. |
| 20:20.17 | [TK]D-Fender | The password and PM it |
| 20:20.28 | [TK]D-Fender | Because if you fuck with the evidence I'm nto going to waste my time |
| 20:22.40 | v0lZy | it exceeds the pastebin limit, let me find another way... |
| 20:23.06 | [TK]D-Fender | You're clearly grabbing too much for too ling |
| 20:23.18 | [TK]D-Fender | This should be all of 3-5 seconds of debug tops |
| 20:23.24 | [TK]D-Fender | it's shouldn't be that big |
| 20:24.14 | v0lZy | its just that theres a lot captured by sip debug thats just stuff with the phones on the local network |
| 20:24.18 | v0lZy | BLFs etc. |
| 20:24.27 | v0lZy | pasted in private |
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| 20:29.37 | dan_j | One way audio sounds like a firewall issue to me if there is no nat. Make sure you are allowing the RTP ports on the firewall. |
| 20:30.06 | dan_j | Look at rtp.conf on your asterisk box to see what ports you should allow. |
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| 21:10.51 | [TK]D-Fender | Was improper NAT setup as expected |
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| 21:28.16 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.12.1 (2016/10/27), 11.24.1 (2016/10/27), Standard: 14.1.1 (2016/10/27); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 21:42.04 | drmessano | lol |
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| 23:38.03 | Samot | So all the emails were dups it looks like and what's Trump's play on that. Insult the Jews. |
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