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| 07:35.41 | andycol_500 | Hi All |
| 07:35.57 | andycol_500 | i am struggling to find info as to why i am getting this error on pjsip when i dial out |
| 07:36.01 | andycol_500 | Unable to create request with auth.No auth credent als for any realms in challenge. |
| 07:36.10 | andycol_500 | has anyone seen it before |
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| 09:55.54 | stefan27 | If asterisk13.5 on some channel sip_Q executes Dial(SIP/77@peer_A) and peer_A in sip.conf has outboundproxy=X and host=Y it seems asterisk creates a request URI like sip:77@Y, which is sent to ip X, but if I want to override the value of Y by Z in the dial if the channel sip_Q satisfies some condition, how do I best do this? |
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| 10:01.46 | Zogot | stefan27: Dial(SIP/user:secret@newY/77) |
| 10:01.47 | Zogot | i think |
| 10:02.01 | stefan27 | the only solution we've come up with is patching chan_sip's 'sip_call' to consider a new special dialplan variable OUT_NEW_REQ_URI which is sometimes set to Z and make sip_call use that instead of the peer's default: Y. |
| 10:03.16 | stefan27 | but I want all the other properties of peer_A, I'd have to rewrite the entire dialplan if Dial(SIP/<boo>@peer_A) is to be replaced with Dial(SIP/user:secret...) |
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| 10:06.43 | Zogot | stefan27: maybe you can mess with SIPAddHeader? |
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| 11:00.10 | tiuman | how run parralel thread extention?? |
| 11:00.21 | tiuman | howt to run parralel thread extention?? |
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| 11:26.38 | phpboy | tiuman: what are you trying to achieve? |
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| 12:45.26 | zaf | can anyone recommend a phone to use in a hospital environment? I guess it needs to be anti-microbial, no screen, and simple, large-ish buttons? |
| 12:46.37 | [TK]D-Fender | Forget anything "VoIP" at that point and get an analog one. |
| 12:46.53 | [TK]D-Fender | "Antimicrobial" is not going to be on the list of normal stuff |
| 12:47.10 | [TK]D-Fender | Same with "big button" |
| 13:00.32 | lvlinux | zaf: yes, find an analog phone that fits your criteria, and then run FXS interfaces to talk to them. |
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| 13:15.26 | zaf | found these - http://www.vtechhotelphones.com/s2220.html |
| 13:18.32 | stefan27 | When asterisk crashes and a sip device is registered via UDP, the sip device will usually not notice the crash since the registration is stored in asterisk database and udp is stateless, but if the sip device is registered via TCP, what is supposed to happen? The server machine seems to send some F flag on the tcp connection, is a good sip device supposed to detect the termination of the tcp |
| 13:18.32 | stefan27 | connection and attempt some kind of special re-register sooner than the usual re-registration time given by last REGISTER's dialog's expire attribute? |
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| 13:30.18 | Tagor | can someone tell me if this sip client config is ok http://pastebin.com/4hLKi7wg ? although it has a really strong password some bot was able to log in and make calls. * version is 11.13.1 |
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| 13:45.04 | [TK]D-Fender | Any password could get brute forced |
| 13:45.17 | [TK]D-Fender | Do you have prrof that this was the peer that those calls were authed as? |
| 13:45.30 | [TK]D-Fender | A secret alone is not "secure" |
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| 13:53.09 | Tagor | [TK]D-Fender: I killed * and checked the logs. They logged in on multiple user accounts and successfully made calls. I changed the passwords but it happened again. According to the logs there were about 20 failed log in attempts from other IPs. That shouldn't be enough to crack one of the used passwords |
| 13:53.43 | Tagor | [TK]D-Fender: by the way I'm also using BFD |
| 13:53.48 | [TK]D-Fender | Depends if you were getting flooded with attacks so fast they wouldn't log. |
| 13:53.59 | [TK]D-Fender | Which is a thing... |
| 13:54.10 | [TK]D-Fender | What is this BFD you're referring to? |
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| 13:55.09 | Tagor | [TK]D-Fender: similar to fail2ban. A cron that checks logs and adds ips to iptables |
| 13:55.31 | [TK]D-Fender | You should be rate limiting as well |
| 13:59.05 | Tagor | [TK]D-Fender: thanks I will look into that. do these flood attacks show up when sip debug is on? |
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| 14:00.46 | file | falls over |
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| 14:01.21 | [TK]D-Fender | Tagor: IIRC |
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| 14:05.52 | Tagor | [TK]D-Fender: anything else I could look into? I used a password like this one: V$S=gtJ3;m-P3x<8A:5FZ$2u>R2^L+}uZpmg?zRtk and they got in within a few hours. I wonder if that's enough time to brute force such a password |
| 14:06.16 | glNito | No, not unless there are thousands of computers attempting. |
| 14:06.34 | glNito | And that would still take a very long time. |
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| 15:52.28 | stefan27 | how does ${CHANNEL(rtpqos,audio,all)} which I report just before hangup relate to the output of sip show channelstats? |
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| 15:56.04 | WIMPy | There's also RTPAUDIOQOS, RTPAUDIOQOSJITTER, RTPAUDIOQOSLOSS, RTPAUDIOQOSRTT, RTPAUDIOQOS |
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| 19:38.02 | rrittgarn | having a senior moment, if i specify operator=yes and exitcontext for a voicemail entry, that goes to the 'o' extension in the exit context right (if the caller presses zero)? |
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| 19:45.10 | newtonr | Does the docs say? I'll look |
| 19:45.58 | rrittgarn | i ended up just testing it - worked |
| 19:46.00 | newtonr | yeah voicemail.conf sample says "Context to go to on user exit such as * or 0" for exitcontext |
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| 20:11.35 | CurtN | hello everybody, can i ask a question plz ? :) |
| 20:11.47 | WIMPy | ~ask |
| 20:11.48 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 20:15.58 | CurtN | Well i have just installed Asterisk, WebRTC & SIPML5, created two accounts but the problem is when i try to call i have this problem : |
| 20:16.22 | CurtN | == Using SIP RTP CoS mark 5 [Oct 16 15:15:38] WARNING[17969][C-00000000]: chan_sip.c:10519 process_sdp: Can't provide secure audio requested in SDP offer |
| 20:17.16 | CurtN | res_srtp.so is already loaded |
| 20:18.34 | CurtN | anyone can help ? :) |
| 20:18.58 | [TK]D-Fender | It may be loaded... and available.. but did you actually setup your certificate? |
| 20:19.22 | CurtN | i have no wss just ud ws |
| 20:19.31 | CurtN | just ws* |
| 20:20.26 | [TK]D-Fender | CurtN== Using SIP RTP CoS mark 5 [Oct 16 15:15:38] WARNING[17969][C-00000000]: chan_sip.c:10519 process_sdp: Can't provide secure audio requested in SDP offer |
| 20:20.34 | [TK]D-Fender | offer was for secure |
| 20:20.37 | [TK]D-Fender | no certificate = fail |
| 20:20.49 | [TK]D-Fender | Either fix the request, or satisfy it |
| 20:21.22 | CurtN | ok so i have to get ssl certifacte and enable wss transport ? (i am new to Asterisk :) ) |
| 20:21.31 | CurtN | thanks a lot for helping me ! :D |
| 20:23.24 | CurtN | or i have to set transport = tls |
| 20:23.25 | CurtN | ? |
| 20:23.41 | [TK]D-Fender | tls=secure certificate |
| 20:23.43 | CurtN | protocol = tls sorry >< |
| 20:23.56 | [TK]D-Fender | Transport Layer SECURITY |
| 20:24.05 | [TK]D-Fender | You expect TLS ... then set up the cert |
| 20:24.32 | CurtN | so i follow this https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial should it work |
| 20:24.48 | CurtN | sorry for being a newbie ^^ |
| 20:24.56 | [TK]D-Fender | I'm pretty sure the official WIKI's instructions do pretty much what they say they do... |
| 20:25.05 | [TK]D-Fender | if that's what you want then I suggest trying to follow it |
| 20:25.20 | CurtN | ok thanks a lot i will follow it |
| 20:26.38 | [TK]D-Fender | packs up to head home... |
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| 23:35.29 | Tagor | What function can I use to tell the calling party that a number is not in use? |
| 23:36.41 | robmal | Hangup |
| 23:37.06 | Tagor | robmal: Do I need to use Answer first? |
| 23:39.27 | WIMPy | no |
| 23:40.48 | WIMPy | And if you narrow "not in use" down a little, we might be ablw to suggest the right parameter. |
| 23:41.43 | Tagor | WIMPy: I mean 'number does not exist' |
| 23:42.05 | Tagor | 'no such number' |
| 23:42.12 | WIMPy | Hangup(1) |
| 23:42.33 | Tagor | Thanks |
| 23:44.56 | robmal | I'd love to make everything Hangup() |
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| 23:55.29 | WIMPy | Saves a lot of traffic! |
| 23:57.34 | robmal | ('d love to make a hangup() day but tue users won't understand :-? |
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