IRC log for #asterisk on 20190402

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01:50.03life_of_eThere's a bit of silence on the ATA when it dials out (a good four or five seconds).  Is there a way to send a ringing signal back to the calling handset?  I tried using Ringing() but I still get silence.
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01:51.07[TK]D-Fenderthere is ATA dial timeout before it even passes a call to the PBX
01:51.16[TK]D-Fenderwhich is quite likely what you're hitting
01:51.34[TK]D-Fendergo look at CLI and compare the time between your last digit and the actual start of processing
01:51.40life_of_eIt's about ten seconds
01:52.35life_of_eBut I have every setting I can think of in the ATA set as short as I can for that timeout.  But while the handset is waiting for that connection, those ten seconds are silent
01:54.55[TK]D-FenderI recall some of those GS ATA's not even HAVING a timeout setting and you're stuck with one basic value
01:55.00[TK]D-Fenderand no local dialplan
01:55.13[TK]D-FenderNot sure on your specific model if they've changed....
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02:04.57life_of_eI may need to check via the SSH interface
02:05.59life_of_eBUt it does have local dialplans
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02:15.07life_of_eThere's 10 seconds between Asterisk in the console saying Called the endpoint and the endpoint "is ringing"
02:28.02[TK]D-Fenderfrom where to where?
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02:47.55life_of_eFrom any internal phone dialing out to the FXO
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09:06.08lostinsipHi I dont know if anyone can offer any ideas as to why I cant make a call between extensions, although the phones are registered to the PBX?
09:06.26lostinsipthis happened since a new switch installation
09:07.14lostinsipif I dial an extension that is actually not connected I go to the unavailable/voicemail
09:08.20lostinsipbut with a pair of "Test" extensions sat next to me I get "The person at extension xx is unavailable" if i dial either extension from the other
09:08.47lostinsipI have made no changes to the PBX setup
09:08.51lostinsipany ideas?
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11:14.29Shorty_has anyone had any issues with T.38 re-invitations? I'm seeing a T.38 reinvite from my provider, which we reject with a 488 (I don't want to do it), I then see another re-invite for audio, which we also 488. The other end then hangs that channel up on us
11:28.13Shorty_best I can tell, asterisk should *not* be 488'ing that second re-invite
11:28.40filewhat SIP implementation?
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11:40.24Shorty_file: PJSIP
11:41.04fileWhat version of Asterisk and PJSIP?
11:41.14Shorty_for comparision, I've done a call to the same number and provider on my asterisk 11 instance running normal SIP.. it handles the call correctly
11:41.25Shorty_Asterisk 16.2.1~dfsg-0~ppa1 built by nobody @ buildd.debian.org on a unknown running Linux on 2019-02-28 21:36:30 UTC
11:41.35Shorty_PJPROJECT version currently running against: 2.8
11:42.27Shorty_I did capture some debug out of it at some point in the past I can probably dig up, I saw that the media request was saying it was an image, even though the re-invite definately wasn't.. almost like it was caching the media type of 'image'
11:42.50Shorty_I tried digging through the code, but I'm not very strong with C so I got lost
11:43.04Shorty_as it was I have to lodge 3 crashing bugs with T.38 handling (one of the reasons it's now off)
11:45.05fileI don't know if anyone has tested the scenario, although it should be fine
11:45.23Shorty_I have a PCAP showing the rejection
11:45.28fileIt's not necessary to reinvite again after the 488 so few impls do
11:45.46Shorty_the other end is asterisk fwiw
11:45.52Shorty_which version I don't know
11:46.18fileI'd file an issue then
11:48.46fileThis does assume the second reinvite isn't wonky
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11:55.23Shorty_as mentioned, asterisk 11 handles the same thing fine, so
11:55.33Shorty_where's the best place to whack issues these days?
11:56.55fileJIRA. https://issues.asterisk.org/jira
11:57.03fileNo time frame on when it will get looked at
11:57.33Shorty_*nod* fair call
11:59.44Shorty_am I missing something, there's no way to make an issue without an account and I have to contact an admin for one?
12:00.55Shorty_ah found the guidelines
12:02.41fileThere's also the signup notice in big letters on the left hand side of the main page
12:02.54fileWe can't alter things further without forking JIRA and modifying it
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12:13.18Shorty_fair call, I must've missed it, sorry about that
12:15.42fileMany do
12:16.00fileI've tried to find other ways to make it stand out without luck
12:30.30Shorty_submitted, thanks for your help
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12:42.05Shorty_I tell you what, I'm so far over faxing xD
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12:46.35fileShorty_: you should always attach files, not link elsewhere as external can go away - even if it's your own external source
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12:52.36Shorty_will fix shortly
13:04.38sibiriadid asterisk in the past use "end + 1" of the rtp port range in rtp.conf?
13:04.56sibiriaie. setting the range to -20000 could lead to 20000 for RTP and 20001 for the connected RTCP
13:05.01sibiriaif yes, is it still like this?
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13:05.49sibiriai suppose it should be so, since RTCP should always land on the following uneven port
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13:15.11ThE-mASkBGHello guys!
13:15.11ThE-mASkBGI have a problem.
13:15.11ThE-mASkBGCan anybody help me with configuration of confbridge.conf
13:16.39fileif you ask a question, someone may answer
13:21.24ThE-mASkBGI want to make a conferencing with a user and an administrator
13:21.24ThE-mASkBGWhen dial number 123456, ask for identification with a pin if it is a user or an administrator
13:34.29Guggeplayback a soundfile, read some dtmf, check the variable, start the conference
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13:36.11SamotWell I'm still waiting to hear what the actual problem is.
13:36.32SamotSo far we've gotten a "would like list"
13:37.16ThE-mASkBGexten => 1,1,Answer()
13:37.16ThE-mASkBGsame => n,Set(CONFBRIDGE(user,template)=default_user)
13:37.16ThE-mASkBGsame => n,Set(CONFBRIDGE(user,admin)=yes)
13:37.16ThE-mASkBGsame => n,Set(CONFBRIDGE(user,marked)=yes)
13:37.16ThE-mASkBGsame => n,ConfBridge(1)
13:37.24ThE-mASkBGThis works
13:38.39ThE-mASkBGI want to make sure that anyone who chooses 1 asks for a pin if they will be a user or an admin in the conference
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13:39.34ThE-mASkBGI dont know how to do it
13:40.02Samothttps://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_app_confbridge
13:40.43Samothttps://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_ConfBridge
13:41.08Samothttps://github.com/asterisk/asterisk/blob/master/configs/samples/confbridge.conf.sample
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14:07.16ThE-mASkBGthank you
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17:33.24life_of_eI guess I might not consider a GDS3710 door camera after all.  Someone just found it making SIP connections to an IP address in China and that IP is not found anywhere in the configuration files.
17:34.10seanbrightsource?
17:35.00life_of_eGrandstream's helpdesk
17:35.14seanbrightno source then?
17:35.24life_of_ehttps://forums.grandstream.com/t/gds3710-backdoor/35491
17:35.33life_of_ePCAP data in that post
17:37.11seanbrightno payload? shame
17:37.42seanbrightit's using SIP backoff timing
17:37.54seanbrightso probably just a misconfiguration
17:39.20seanbrightand isn't 1.1.0.0 a network? can you even route packets to it?
17:39.30seanbrightnot really a networking guy
17:39.34zafthe chances of "1.1.0.0" being a misconfiguration is pretty high compared to the chances of it being something nefarious
17:42.53seanbrightyeah, seems like a whole lotta nothing
17:44.06fileI don't even think that range is announced on the public internet
17:44.09life_of_eYou can address something ending in .0 if it's part of a larger block.  I do agree that it's a misconfiguration but why is there a hardcoded IP in the firmware?  If that IP isn't showing up in any configuration file there's no way to put it in.
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17:44.31seanbrightkinda shitty of that user to use the subject "GDS3710 backdoor" as if it was a foregone conclusion
17:44.39seanbrightFUD
17:44.42life_of_eI agree with that
17:45.00life_of_eI'm just not a general fan of things making hidden contacts no matter the reason
17:45.22life_of_e1.1.0.0 is indeed routed and advertised.  It belongs to China Telecom
17:45.27seanbrighthave you never had a device require a factory reset?
17:45.58life_of_eSure, but I always can see what it attempts to contact.
17:45.59seanbrightit's possible that the config is just corrupted
17:46.25seanbrighti'd like to see the payload of those packets
17:46.48life_of_eThat would be fascinating.
17:46.49seanbrightwell, i take that back
17:46.55seanbrightit's not asterisk
17:47.00seanbrightand i don't own a grandstream
17:47.11seanbrightso i don't really care
17:47.15life_of_e:)
17:47.34life_of_eI'm just fascinated from a network security perspective.  It's one of my hobbies
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18:02.42[TK]D-Fender"some guy in a forum" = wrong by default 99% of the time :)
18:04.38Samot"some guy on IRC" = wrong by default 99% of the time :)
18:04.43SamotSame applies.
18:06.59life_of_eIt got me curious enough to put a monitor on the firewall just to see if my ATA is doing anything
18:08.15life_of_eI know I have enough chatty devices that I have to block already.
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19:05.27life_of_eGrandstream's support is a bit slow on the reading comprehension
19:06.40life_of_eThey asked for PCAPs and debug syslogs.  So I made them, zipped them up and sent them in.  Then they return a message saying "We need the debug logs."  Uh, open the zip file.
19:09.28seanbrightoh
19:09.31seanbrightyou're the OP?
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19:16.52life_of_efor the camera? no, this is captures trying to figure out why there's a 10 second delay dialing out the FXO
19:17.12seanbrighthmmm...
19:18.50life_of_eI can see it in the Asterisk console, between the time Asterisk says "Called..." to the time it says "...is ringing" is about ten seconds.  Nothing else shows up in the console between those.
19:21.18life_of_eIt takes 5 seconds from the time the endpoint sends off the dial to the time that the ATA starts sending DTMF tones to the PSTN.  Then the tones play very slowly (about 0.5 seconds per digit).
19:21.25[TK]D-FenderThen maybe the HT is really slow before, while dialing, and slow to pick up progress in audio (which is probably what it is doing).
19:22.11life_of_eperhaps so, I had to add the ring option to the dial command so someone would know the dialing was happening.  Otherwise the channel is dead silent.
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19:32.59life_of_eWe'll see what they say.  The person responding on the helpdesk says they'll take the captures "So we can check in laboratory"
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20:15.52life_of_eThey seem lost now.  Every thing they've asked me to set, reset, change, whatever, are all related to inbound calls from PSTN when I've stated multiple times the delay is calls to the PSTN not from it.
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21:04.33life_of_eI guess the concept with any Grandstream product is that it better work exactly right for what you need because anything else and you won't really get any help from them.
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21:16.33life_of_eThey're trying to blame it all on the POTS line taking 10 seconds to process the DTMF tones
21:21.05life_of_eI can hear everyone rolling their eyes. :)
21:28.31avblife_of_e: :) i think the concept of all GS AT always been 'or its working or not' :)
21:29.00avbif something is wrong, its less likely that you will make it work
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21:43.00life_of_eI know, but boy is the support system infuriating
21:43.59life_of_eSomehow the phone company is causing the delay even though the ATA hasn't even picked up the phone line to begin dialing.  This is Grandstream's claim anyway.
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