IRC log for #asterisk on 20161211

00:33.00Samotdrmessano:
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01:58.07budman_mtlhello ppl
01:58.39budman_mtli need some advise on a project creation
01:59.31budman_mtlneed to use asterisk to call ppl from db list and have it run a scripr according to user dtmf entry
01:59.39budman_mtl*script
02:00.24budman_mtlany starting point would be apreciated, i did some project involving asterisk, but i'm a n00b. thxs in advance
02:02.02budman_mtlps this would call internal extention only, but i dont think it make a diff to asterisk .
02:05.13budman_mtlplz pm if you have any info, ill be around. Thxs
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02:36.34[TK]D-Fender~book
02:36.34infobotAsterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
02:36.35[TK]D-Fender^^^
02:36.55[TK]D-FenderFor the callouts : AMI Originate, or Call Files.
02:37.00[TK]D-Fender~asteriskwiki
02:37.00infobotasteriskwiki is probably http://wiki.asterisk.org
02:37.02[TK]D-Fender^^^^
02:37.19[TK]D-Fenderbudman_mtl, ^
02:50.09budman_mtlthxs
02:50.19budman_mtlwill do my homework
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08:18.41MilosWhat is this bridge_simple.so which was added/forced in Asterisk 13?
08:20.19Miloshttp://infobot.rikers.org/%23asterisk/20140829.html.gz
08:20.31MilosApparently "bridging has been split out into modules"
08:20.41Milosmentioned by file back at the end of 2014
08:21.20MilosCan I get some more context on that? If it was split out into modules, I would have expected to still see bridges being joined in the Asterisk 11 logs, but it seems to be new in Asterisk 12+ (xxx joined 'simple_bridge' basic-bridge <537840ed-ef7e-446c-91db-17632e159acc>)
08:21.30MilosWhat is simple_bridge and/or basic-bridge exactly?
08:25.14Miloshttps://wiki.asterisk.org/wiki/display/AST/Bridges - is this just an internal thing that you've decided to make a public interface for?
08:25.30Milos"Dial - a bridge is created for the two channels when the outbound channel answers. Both the inbound channel and the outbound channel are placed into the bridge."
08:25.41MilosObviously this was possible before Asterisk 12, hence my question.
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09:18.51SamotAsterisk 12 is EOL
09:19.38MilosI never used Asterisk 12.
09:19.48MilosMade the switch from 11 to 13.
09:22.26SamotWhat is the issue you are having?
09:23.12MilosNo issue - just wondering what the new bridge stuff is and how it relates to Asterisk 11. It seems in 12+ a bridge is created for each call. Calling works in 11 without these bridges, so what gives?
09:23.48SamotWell a bridge happens when there are two or more channels that share media..
09:24.06SamotSo yes, when you make a call from your device to another device, those two channels are bridged together.
09:25.12MilosI had to add load => bridge_simple.so to make this work, since I only load the modules that I choose. Now, logs mention (xxx joined 'simple_bridge' basic-bridge <537840ed-ef7e-446c-91db-17632e159acc>) which again wasn't seen in 11 and below.
09:25.27MilosMy question is, is this a core feature that was hidden from the end user, now made into a module, or what?
09:26.43SamotWell bridging happened in Asterisk 11
09:26.46SamotAnd before.
09:26.51MilosTo rephrase my question: In Asterisk 11, calling worked fine without any mention of bridges neither as a module nor in the regular logs. In 12+ this module has to be loaded and the logs talk about creating a bridge for new calls.
09:27.00MilosSo does that mean it was an internal feature that was made public in 12+ ?
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09:35.53SamotI'm not sure what you mean, but I just found an Asterisk 11 box and bridge_simple.so is a module that needs to be loaded.
09:38.20SamotIn fact, going back to a box running 1.8, bridge_simple.so is a module that needs to be loaded.
09:38.26SamotSo this isn't a new change.
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20:00.19bravvvehi am having some truble to conect to my server from wan,need some help pleeze
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20:04.48WIMPyWe have a strict policy here. You won't get any answers without a question.
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20:14.51bravvveWIMPy:some help in testing !!!!
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20:22.25Samotbravve: Then actually state what you are trying to achieve, what you have actually done and what type of errors are you getting.
20:23.00SamotUntil you actually tell us the issue and what you are doing, we will not...er..cannot help.
20:29.53bravvveam having asterisk server installed behined DSL modem under a openvz host machine,i can connect from machine in LAN network,but not WAN,i can't figure why
20:31.30bravvveip is 41.100.88.172 tcp 5062,account name 2201 password p2201
20:32.11RS232Cbravvve, have you forwarded the udp port to your internal firewall?
20:32.20RS232Cerr server
20:33.08bravvvetcp,yes
20:33.09RS232Cor are you intentionaly using tcp?
20:34.48igcewielingudp is not tcp.
20:34.55bravvveusing wireshark,i get resquet from distant machin,and the host is sending reponce,but nothing in the server
20:35.42RS232Cnmap says your port is closed
20:35.45bravvveyes i know udp,and tcp,am using tcp to folow ACKs
20:36.24bravvveops,5060 not 5062
20:36.48RS232Cya that shows open
20:37.46RS232Chmmm I have never used tcp for sip
20:38.22RS232Cwireshark should be able to follow a udp sip conversation...
20:38.46bravvveam using tcp cause it should give ACKs
20:38.53RS232CI assume it can follow a tcp one as well
20:39.05bravvveudp do no
20:39.40RS232Chave you turned on "sip set debug on" in your asterisk console?
20:41.16RS232Cyou might also need something like "nat=force_rport,comedia" in your device config in sip.conf
20:41.35RS232Cbut I don't know if that applys to tcp
20:42.32bravvvewhat nat=force_rport.comedia  do?
20:43.31RS232Cnat = force_rport       ; Pretend there was an rport parameter even if there wasn't
20:43.40RS232Cnat = comedia           ; Send media to the port Asterisk received it from regardless
20:44.45bravvverport?
20:44.46RS232Cbut I think thats for media once sip is established
20:44.55RS232Cremote port
20:45.16bravvveyes ,the problem with me that i can't connect
20:45.44bravvveRS232C:can connect to my server?
20:46.07bravvveRS232C:from your location????
20:46.22RS232Cabout to try
20:47.53bravvveok
20:48.25RS232Cmy ciscoSPA525G says registration failed
20:49.36bravvvehow can do virbos mode ?
20:50.29RS232Cin asterisk -r "core set verbose 999" and "core set debug 999"
20:51.02SamotA core debug isn't needed at this point.
20:51.19SamotAnd it will produce a bunch of unneeded output.
20:51.47RS232CI usually turn it on. but your right
20:52.07Samot1) Are all the ports forwarded properly from the router to Asterisk?
20:52.23Samot2) Are things like SIP ALG or SPI Firewall disabled in the DSL router?
20:52.49Samot3) Are the external ip, local_networks, etc configured properly?
20:54.41SamotUntil all the NAT/network stuff is configured properly in Asterisk, doing a debug for remote connections is kinda pointless.
20:54.57SamotAnd in the router.
20:55.49bravvvehhh,am using GNS to test network stuffs
20:56.00RS232Cbravvve, is your setup like this ... phone <--> firewall <--> internet <--> firewall <--> asterisk ?
20:56.13SamotI don't know what GNS is.
20:56.33RS232Ccisco router emu
20:59.32bravvveyes is Cisco router emulator,but i get to configure a router too hhhhhhh,and am using openvz,too many things to configure hhhhhhh
20:59.57bravvveand at last see what happend with wireshark
21:05.54RS232Cbravvve, I am assuming you are running asterisk on some for of linux?
21:06.26RS232Cform even
21:06.57RS232Cyou might check your firewall setting on that server as well
21:09.37bravvveyes a debian vz template,some cained of virtualization
21:10.13RS232Cso I am working with 2 asterisk servers one is voice mail... but I can't seem to get "mwi => 1234:password@mysipprovider.com/1234" working... I can get voice mail and remove voice mails but MWI is not working. any gotchas I should be aware of?
21:10.57RS232Calso the main server was compiled for imap support but not file storage support
21:11.28RS232Cbut not sure if that impacts MWI
21:13.09RS232Cbravvve, you might check is it is blocking non local access to tcp 5060
21:13.43RS232Cs/is/if/g
21:14.12RS232Clol
21:15.19RS232Cinfobot, you almost got it right!
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22:29.05bravvveRS232C:hi
22:29.16RS232Cyo
22:31.12bravvvecan help to test with 41.100.40.122:5060 username 2201 password p2201,plz
22:31.18bravvvecan help to test with 41.100.40.122:5062 username 2201 password p2201,plz
22:31.27RS232Ctcp still?
22:31.27bravvve5062 not 5060
22:31.32bravvvetcp yes
22:31.37RS232Cya
22:33.05RS232Cloons like its not registring
22:35.18bravvveand now?
22:35.33RS232Chold on turning up syslog and rebooting phone
22:37.42RS232Clooks like its not connecting
22:38.40bravvveok
22:40.03RS232CDec 11 16:42:53 10.42.0.199 [1:0]SIP/TCP:ConnErr:110
22:40.03RS232CDec 11 16:42:53 10.42.0.199 [1:0]SIP/TCP:Connect=-1
22:40.03RS232CDec 11 16:42:53 10.42.0.199 [1]SIP/TCP Backoff 500 ms
22:40.03RS232CDec 11 16:42:53 10.42.0.199 [1]SIP:RegFailed;Retry in 30s
22:40.20RS232Cthats the syslog from my phone
22:41.39RS232Coh I just noticed a diff ip... my bad
22:42.13bravvveok,give it again
22:42.22bravvveor hit it again
22:42.49RS232C5060 or 5062?
22:43.40RS232C5062 is failing a lot faster now
22:43.41bravvve5062
22:45.55bravvvecan tray again plz
22:47.24RS232Cstill fail
22:47.34RS232CSIP/TCP:ConnErr:111
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22:52.21bravvvenot getting you at all
22:53.05RS232Cnothing from 216.106.72.202 ?
22:54.17bravvveno ,and its working very well between 2 routed networks,i'v tested it with gns3
22:55.48bravvvenow the problem is in the router or ISP is blocking it
22:55.56bravvveam getting 211.103.199.98
22:56.07bravvveam getting 211.103.199.98,but http not 5062
22:56.56RS232Cthis feels like a firewall issue to me
22:57.22bravvveport is open ,have tested it
22:59.02RS232C... what country are you in?
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23:02.22RS232Cthat ip says Algeria... mabe a government level block?
23:03.56RS232CI know some countries block voip
23:05.26RS232Cif thats the case you may have to tunnel though a VPN and hope that is not blocked
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