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| 00:12.59 | X-Rob | File... So would *this* convince you to come to Toronto? https://www.facebook.com/xrobau/posts/10153812531567752 |
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| 05:01.10 | pools0n | dudes ! |
| 05:01.22 | pools0n | hows life |
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| 05:01.58 | pools0n | to whichever jokers helped out yesterday you'll be relieved to know i got the stuff working that i was trying to do ! |
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| 12:44.15 | jkroon | Chainsaw, i think we should bring that discussion here... https://bugs.gentoo.org/show_bug.cgi?id=107482 for everybody else's sake. |
| 12:44.53 | Chainsaw | Probably #gentoo-voip instead of #asterisk though. |
| 12:45.10 | Chainsaw | It's competitor after all ;) |
| 12:45.21 | jkroon | not really :) |
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| 14:21.28 | ruben23 | hi guys any idea have this issue displaying on my asterisk -------> my CLI is flooded with this ------> WARNING[3792]: chan_iax2.c:4929 handle_call_token: Address xxx.xxx.xxx.xx failed CallToken hash inspection |
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| 16:26.04 | enoch | -bash: /var/run/asterisk/asterisk.ctl: No such device or address |
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| 17:10.45 | Dpunkt | An incoming, accepted, working call got terminated unexpected by the caller with "Reason: SIP ;cause=404;text="Not Found (1:27)"" after 2:26 minutes. Any ideas? |
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| 17:15.29 | [TK]D-Fender | idea : show us |
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| 17:15.46 | [TK]D-Fender | * CLI. Verbose 210. SIP debug enableed |
| 17:15.51 | [TK]D-Fender | Actual call with actual proof |
| 17:16.17 | [TK]D-Fender | and while we're at it... always good to know what you're actually running. |
| 17:16.55 | Dpunkt | Asterisk 13.7.0 with pjsip |
| 17:18.32 | Dpunkt | i got a network trace but no cli output |
| 17:18.52 | [TK]D-Fender | Go get a proper debug |
| 17:19.54 | Dpunkt | problem is hard to reproduce but often enougth to be disturbing |
| 17:20.03 | Dpunkt | How do i set up debug right? |
| 17:22.01 | [TK]D-Fender | [12:15][TK]D-Fender* CLI. Verbose 210. SIP debug enableed |
| 17:22.05 | [TK]D-Fender | verbose 10 |
| 17:22.14 | [TK]D-Fender | 210 would still do the samee thing mind you |
| 17:22.32 | catphish | surely a non-ok SIP status should not change the state of the session |
| 17:23.23 | catphish | a failed reinvite should leave the previous parameters of the session in place shouldn't it? |
| 17:29.16 | K0HAX | Is there a way to get the IP address of someone sending a SIP INVITE ignoring what's in the From: section of the INVITE? |
| 17:29.55 | Dpunkt | ok, now trying to reproduce this issue |
| 17:29.57 | K0HAX | Asterisk is saying one thing, and Wireshark is saying another, the packets are coming from someone who's attacking my PBX |
| 17:30.22 | catphish | K0HAX: well wireshark is obviously right, and SIP/PJSIP certainly knows the real address |
| 17:30.30 | catphish | since that's how it matches peers |
| 17:31.25 | K0HAX | Yeah, the source IP in wireshark is a public IP (as it should be) but the output of ${CHANNEL(uri)} shows a spoofed IP |
| 17:31.33 | catphish | i'd also guess that it logs the IP in the security log, for the benefit of tooks like fail2ban |
| 17:31.52 | catphish | CHANNEL(uri) will show the From: URI, not the source IP |
| 17:32.05 | catphish | the URI is not related to the proxy sending you the message |
| 17:32.35 | K0HAX | [Feb 18 11-13-47] SECURITY[] Unknown Call from to 99992 IPdetails sip:99992@10.5.50.2 |
| 17:32.39 | catphish | i dont know if the real source IP can be found in a variable or not, but definitely not in the URI |
| 17:32.40 | K0HAX | that's in my sipsec |
| 17:33.10 | catphish | that's in the security log? |
| 17:33.12 | K0HAX | yeah |
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| 17:33.19 | catphish | disappointing :( |
| 17:33.22 | K0HAX | I know. :( |
| 17:33.25 | catphish | that just looks like a URI |
| 17:33.57 | K0HAX | and that's the 'to' |
| 17:34.04 | K0HAX | that IP is my Asterisk IP |
| 17:34.30 | catphish | i get this: |
| 17:34.32 | catphish | [Feb 18 17:04:49] NOTICE[11274] res_pjsip/pjsip_distributor.c: Request from '"sipvicious" <sip:100@1.1.1.1>' failed for '85.93.91.138:5156' (callid: 962162157763124368629214) - No matching endpoint found |
| 17:34.41 | K0HAX | that's way better.. |
| 17:34.52 | K0HAX | I'm running: Asterisk GIT-master-3b426a8 built by michael @ asterisk on a x86_64 running Linux on 2015-10-16 20:22:59 UTC |
| 17:35.02 | catphish | are you using pjsip? |
| 17:35.04 | K0HAX | no |
| 17:35.07 | K0HAX | normal sip |
| 17:35.13 | catphish | ah, i dont know then |
| 17:35.16 | K0HAX | I can't get pjsip to work behind NAT correctly |
| 17:35.26 | catphish | that's a shame |
| 17:35.46 | catphish | i never run a PBX behind a NAT, so not had to look at that |
| 17:35.59 | catphish | anyway, home time now, i hope you find something |
| 17:36.30 | K0HAX | oh hold on.. let me try changing something in my logger.conf |
| 17:36.32 | K0HAX | :) |
| 17:36.48 | K0HAX | sipsec was just output from my dialplan to block unauthorized INVITEs |
| 17:37.37 | catphish | must go now :) |
| 17:44.53 | Dpunkt | i got my issue reproduced by forcing a reregistration using "pjsip send register trunk" http://pastebin.com/qxts6GtU |
| 17:45.18 | Dpunkt | i see that the contact in the registration is the internal ip, not the external |
| 17:51.42 | K0HAX | Woo, sent an 8mb log of the same IP attempting to break into my PBX to their ISP's abuse contact. :) Unlikely that they'll do anything, but it is at least a US company... |
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| 18:04.38 | Dpunkt | The Problem can be fixed by setting external_signaling_address and restart |
| 18:05.41 | Dpunkt | But Asterisk is sitting behind a Nat with dynamic external ip. Is there any other solution to tell asterisk which ip it should use as external ip? |
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| 18:40.15 | elcontrastador | Greetings, I've got a voice down emergency. Power outage and digium phones are sip registering. ITSP and polycom speaker phones are fully operational. Logs show "[Feb 18 10:28:34] WARNING[2739] pbx.c: PBX requires Asterisk to be fully booted" many times after each reboot. Unsucessful googling so far |
| 18:40.32 | elcontrastador | digium phones are NOT sip registering |
| 18:41.06 | elcontrastador | I am running DPMA and the corresponding module is loading |
| 18:44.41 | newtonr | If it is a true emergency then I'd calling up a consultant or someone who can help you get it working immediately. If you post a debug log we can take a look |
| 18:44.47 | newtonr | *recommend |
| 18:44.49 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 18:45.00 | newtonr | pastebin a debug log from startup |
| 18:45.24 | newtonr | preferably two logs, one log without the debug channel and one log with it |
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| 19:37.17 | vader- | hello |
| 19:38.00 | vader- | Quick question, is there any recommend extension prefix or numbers to use to setup the ability for people to forward a call to voicemail instead of an extension? |
| 19:38.33 | vader- | I am being asked if I can create a dial path for our secretaries to send people to a person's voicemail instead of their extension |
| 19:39.18 | robmal | 8 |
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| 20:41.25 | vader- | i need to figure out a way to create a block list so we can enter numbers in we want blocked |
| 20:41.27 | [TK]D-Fender | vader-: Pick whatever you want. some use * + ext. Others #. Sometimes after the number |
| 20:41.28 | [TK]D-Fender | all depends |
| 20:41.32 | [TK]D-Fender | Doesn't amtter really |
| 20:41.47 | vader- | tkd ya i did *6(3 digit extension) so *6123 |
| 20:41.55 | [TK]D-Fender | For the block list... go shove the # in a list you can look up |
| 20:42.08 | [TK]D-Fender | And look it up |
| 20:44.50 | vader- | happen to have any example code for that? |
| 21:05.09 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 21:10.13 | *** join/#asterisk saint_ (~saint_@unaffiliated/saint-/x-0540772) |
| 21:16.57 | [TK]D-Fender | vader-: "core show function DB" |
| 21:17.16 | [TK]D-Fender | vader-: "core show function SHELL" |
| 21:17.26 | [TK]D-Fender | vader-: Or read the book for DB use, etc |
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| 21:34.19 | nickgaw | Hi, What brand of sip phone do you suggest that connects using a network cable but has a wireless receiver that has two line buttons so I can answer two extentions using one phone and the wireless handset can chage on the base? |
| 21:35.41 | saint_ | check grandStream maybe , or Yealink |
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| 21:35.52 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 21:36.48 | nickgaw | Do some office supply stores sell the phones for asterisk? I always like to go and look at the device in a store before I buy it? |
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| 21:39.38 | nickgaw | How long is Asterisk 11 supported? |
| 21:45.56 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 21:46.13 | [TK]D-Fender | nickgawDo some office supply stores sell the phones for asterisk? I always like to go and look at the device in a store before I buy it? <- there is nt such thing as "for Asterisk" |
| 21:46.17 | [TK]D-Fender | Some may sell SIP phone. |
| 21:46.19 | [TK]D-Fender | Which * can speak. |
| 21:46.33 | [TK]D-Fender | And yes you obviously have to look UP the device to know what tit does. |
| 21:47.08 | [TK]D-Fender | 'Whether that means physically at before you by ... not is you look up the proper product number and things are listed properly. |
| 21:47.49 | nickgaw | I have read that some sip phones are locked to carriers so I would want to make sure they were unlocked phones would most sip phones sold in office supply stores be unlocked? |
| 21:48.05 | [TK]D-Fender | No such thing as "most" |
| 21:48.09 | [TK]D-Fender | Go see what yours offers |
| 21:48.17 | [TK]D-Fender | Except remember ... this isn't CONSUMER stuff. |
| 21:48.38 | [TK]D-Fender | Joe Blow doesn't typically go buying SIP phones from Office Depot. |
| 21:48.59 | [TK]D-Fender | Consumer stores don't like selling things they can't help ignorant users with |
| 21:49.10 | [TK]D-Fender | Because ignorantt consumers throw tantrums |
| 21:49.16 | nickgaw | true and I do hope the people know what they are talking about but I would think they would at an office supply store. |
| 21:49.39 | [TK]D-Fender | Maybe you should NOT be trying to by quality phone equipment from Walmart... |
| 21:50.02 | nickgaw | no I would never buy from there as they sell junk. |
| 21:50.40 | nickgaw | that is why I am switching to sip and asterisk to get away from the trash that loads of people sell. |
| 21:50.47 | [TK]D-Fender | Yes, well your office supply store isn't a telcom or high-tech dealer. |
| 21:50.58 | [TK]D-Fender | Electric staplers would be pusing it... |
| 21:51.02 | [TK]D-Fender | pushing* |
| 21:52.32 | nickgaw | are there any such stores that specialise in telecom? |
| 21:52.47 | [TK]D-Fender | Expect nothing in brick & mortar |
| 21:53.09 | [TK]D-Fender | you'll find Telecom Integrators... |
| 21:53.14 | [TK]D-Fender | but not a place to walk in for them |
| 21:53.20 | [TK]D-Fender | walk in = consumer |
| 21:53.35 | nickgaw | so most places are on line then? |
| 21:53.45 | [TK]D-Fender | Or through integrators |
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| 21:54.37 | nickgaw | I had to port my two numbers to a sip company and am currently using soft phones would contacting my sip company be the best thing as I know they themselves use Asterisk? |
| 21:55.37 | [TK]D-Fender | Perhaps you should just ask US what we recommend |
| 21:56.25 | [TK]D-Fender | haven't you already head tons of different makes & models thrown around? |
| 21:56.31 | [TK]D-Fender | heard* |
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| 21:58.13 | [TK]D-Fender | packs up to head home |
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| 22:01.38 | nickgaw | What store is a good store as voip-supply has so much to offer that I would rather talk to someone on the phone to explain what I am looking for? |
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