IRC log for #asterisk on 20190222

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07:53.42perseiverhow to register asterisk as Client for SIP provider
07:53.59perseiverI am using format authuser:secret@domain/extension
07:54.39perseiverBut I have three NIC, two for internal and 1 for public network
07:55.24perseiverI am using another asterisk installation and want to register 1st Asterisk with another Asterisk.  At the same time I want 1st asterisk to my SIP provider
07:55.52perseiverhow to differentiate the register request, as all sip register request from single bind address,
07:56.25perseiverCan I set another bind address ?  may be there should be some setting in PEERS or users type? I don't know
07:56.33perseiverDo any one have idea?
07:56.54perseiverI want to assign bind address for two SIP provider.
07:57.27perseiverSo that when asterisk send register request, appropriate Network interface selected.
08:15.35Posterdatihi
08:17.18Posterdatiplease help, I cannot receive incoming calls: https://paste.debian.net/1069616/
08:17.34PosterdatiI was able to register the ip phone to asterisk (same lan)
08:18.05perseiverCan we bind two interface on asterisk ?  My Server has three interface, Out of which I need to bind two NIC and one should not bind.  Means Asterisk should not send any register or sip request to 3rd one
08:18.11perseiverIs this possible?
08:19.29perseiverhey Posterdati: Its 403 forbidden request.  Either asterisk is not register with sip provider or Endpoint is not reachable
08:20.03perseiverIt also means you are making some mistakes in configuration
08:20.26Posterdatiperseiver: registration with provider occurred
08:20.54Posterdatix.y.z.telecomitalia.it:5060     N      +39xxx      3309 Registered           Fri, 22 Feb 2019 09:06:41
08:21.03Posterdatiseems to be registered to me
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08:31.44Posterdati?
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09:14.39Posterdatipuzzola: :)
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09:57.07Posterdatiplease help, what is this error about? -->  SIP/2.0 500 Cx Unable To Comply 03023012A5004
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10:59.58Posterdatiwow now was able to receive a call! But still cannot do one from the ip phone :)
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11:31.14perseiverPosterdati have you register your ip phone with freeswitch?  If not then it will not able to make call
11:31.56Posterdatifreeswitch?
11:32.01Posterdatiwhat is it?
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14:51.24Posterdatipersever: the phone correctly register to asterisk, I can receive a phone call on the ip phone, but I cannot place a call from the ip phone itself
15:09.13perseiverPosterdati  why you want to call from ip phone to ip phone itself...
15:10.16perseiverhow one can call himself from same phone?
15:12.26Posterdatino...
15:12.36Posterdatipersever: the phone correctly register to asterisk, I can receive a phone call on the ip phone, but I cannot place a call from the ip phone to the outside
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15:23.26[TK]D-FenderPosterdati, Show us the actual failure
15:23.28[TK]D-Fender~pb
15:23.29infobotsomebody said pastebin was 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.
15:23.30[TK]D-Fender^^^^
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15:47.59Posterdati[TK]D-Fender: https://paste.debian.net/1069666/
15:48.08Posterdatifrom reload on...
15:48.50[TK]D-FenderReliably Transmitting (NAT) to 5.97.53.8:5060:
15:48.56[TK]D-FenderContact: <sip:s@10.0.0.2:5060>
15:49.14Posterdatiwhat about?
15:49.16[TK]D-Fenderyour registration to your provider is showing that you did not configure your PBS to work properly from behind NAT
15:49.24[TK]D-FenderYou are missing  required settings
15:49.33Posterdation the firewall?
15:50.09[TK]D-FenderPBX <-
15:50.12[TK]D-FenderReliably Transmitting (NAT) to 156.54.82.96:5060:
15:50.12[TK]D-FenderOPTIONS sip:telecomitalia.it SIP/2.0
15:50.18[TK]D-Fender<--- SIP read from UDP:5.97.53.8:5060 --->
15:50.18[TK]D-FenderSIP/2.0 500 Cx Unable To Comply 03023012A5004
15:50.41Posterdati?
15:50.42[TK]D-FenderThey also don't like you sending OPTIONS packets to them a yuo have your peer configured to do.  Stop.
15:51.00[TK]D-FenderAnd finally I see no call coming in from your phone at all
15:51.06Posterdatihow can I stop to send options packet?
15:51.13[TK]D-Fenderqualify=NO
15:51.19Posterdatiah I placed YES
15:51.21Posterdatiok
15:52.08[TK]D-FenderGo fix your general SIP settings relating to NAT & local addressing
15:52.28Posterdatiin the firewall or sip.conf?
15:52.30[TK]D-FenderThen once you've fixed the things that are already wrong go try and show us an actual call
15:52.40*** join/#asterisk Ai9zO5AP (~BQcdf9eiZ@91.240.67.45)
15:52.46[TK]D-Fender<[TK]D-Fender> your registration to your provider is showing that you did not configure your PBX to work properly from behind NAT
15:52.55Posterdatiok
15:52.57Posterdatithanks
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20:07.12viebigIs it possible to implemetn an machine detection (like AMD) but listening in early media?
20:11.10SamotNot that I'm aware of.
20:23.50seanbrightapp_amd doesn't answer the channel and doesn't appear to check the channel state
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20:30.38seanbrightput another way: try calling Progress(), then AMD(), and then Answer()
20:40.16SladeFreePBX on one of those $5 vultrs should be plenty to handle just a few lines right?
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20:42.15seanbrightshrugs
20:42.21seanbrightcheap to test
20:43.42Sladelooks like it only integrates with sipstation for SMS :/
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21:29.20SamotYes, the SMS integration in the FreePBX UCP is for SIPStation only.
21:30.28SamotSIPStation also doesn't deliver SMS over HTTP API.
21:30.35SamotThey do it over SIMPLE.
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21:47.00as3ty<PROTECTED>
21:52.20igcewielingI have access to Level3 T.38, but it is a larger commercial account.
21:54.19as3tyWe probably won't be able to meet the monthly minimum for L3.
21:54.39Kobazlast time i worked with level3 the commit was like a million minutes a month
21:54.51igcewielingKobaz: *nod*
21:55.19Kobazi think my logrotate fixer is finally production ready
21:55.58Kobazso that this crap can get fixed... -rw-rw-r--+ 1 pbx pbx  93K Dec 22 03:36 AGI.Dialer.Postback.log.63.gz  -rw-rw-r--+ 1 pbx pbx  54M Feb 22 03:36 AGI.Dialer.Postback.log.1
21:56.07Kobazsmacks logrotate
22:35.19SladeSamot, is SIMPLE a better way to do it, or is the flowroute method better..   i guess http api integrates better
22:36.29SamotThose are just delivery methods.
22:36.57SamotSMS is done over SMPP at the carrier level.
22:37.56Sladeright. just curious about the flexibility of each, if a standard is going to win
22:39.05SamotPretty must any ITSP offering SMS is going to do it over HTTP API
22:39.14SamotSIPStation has Asterisk on both ends.
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