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07:53.42 | perseiver | how to register asterisk as Client for SIP provider |
07:53.59 | perseiver | I am using format authuser:secret@domain/extension |
07:54.39 | perseiver | But I have three NIC, two for internal and 1 for public network |
07:55.24 | perseiver | I 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 |
07:55.52 | perseiver | how to differentiate the register request, as all sip register request from single bind address, |
07:56.25 | perseiver | Can I set another bind address ? may be there should be some setting in PEERS or users type? I don't know |
07:56.33 | perseiver | Do any one have idea? |
07:56.54 | perseiver | I want to assign bind address for two SIP provider. |
07:57.27 | perseiver | So that when asterisk send register request, appropriate Network interface selected. |
08:15.35 | Posterdati | hi |
08:17.18 | Posterdati | please help, I cannot receive incoming calls: https://paste.debian.net/1069616/ |
08:17.34 | Posterdati | I was able to register the ip phone to asterisk (same lan) |
08:18.05 | perseiver | Can 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 |
08:18.11 | perseiver | Is this possible? |
08:19.29 | perseiver | hey Posterdati: Its 403 forbidden request. Either asterisk is not register with sip provider or Endpoint is not reachable |
08:20.03 | perseiver | It also means you are making some mistakes in configuration |
08:20.26 | Posterdati | perseiver: registration with provider occurred |
08:20.54 | Posterdati | x.y.z.telecomitalia.it:5060 N +39xxx 3309 Registered Fri, 22 Feb 2019 09:06:41 |
08:21.03 | Posterdati | seems to be registered to me |
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08:31.44 | Posterdati | ? |
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09:14.39 | Posterdati | puzzola: :) |
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09:57.07 | Posterdati | please help, what is this error about? --> SIP/2.0 500 Cx Unable To Comply 03023012A5004 |
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10:59.58 | Posterdati | wow now was able to receive a call! But still cannot do one from the ip phone :) |
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11:31.14 | perseiver | Posterdati have you register your ip phone with freeswitch? If not then it will not able to make call |
11:31.56 | Posterdati | freeswitch? |
11:32.01 | Posterdati | what is it? |
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14:51.24 | Posterdati | persever: 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 |
15:09.13 | perseiver | Posterdati why you want to call from ip phone to ip phone itself... |
15:10.16 | perseiver | how one can call himself from same phone? |
15:12.26 | Posterdati | no... |
15:12.36 | Posterdati | persever: 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-Fender | Posterdati, Show us the actual failure |
15:23.28 | [TK]D-Fender | ~pb |
15:23.29 | infobot | somebody 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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15:47.59 | Posterdati | [TK]D-Fender: https://paste.debian.net/1069666/ |
15:48.08 | Posterdati | from reload on... |
15:48.50 | [TK]D-Fender | Reliably Transmitting (NAT) to 5.97.53.8:5060: |
15:48.56 | [TK]D-Fender | Contact: <sip:s@10.0.0.2:5060> |
15:49.14 | Posterdati | what about? |
15:49.16 | [TK]D-Fender | your registration to your provider is showing that you did not configure your PBS to work properly from behind NAT |
15:49.24 | [TK]D-Fender | You are missing required settings |
15:49.33 | Posterdati | on the firewall? |
15:50.09 | [TK]D-Fender | PBX <- |
15:50.12 | [TK]D-Fender | Reliably Transmitting (NAT) to 156.54.82.96:5060: |
15:50.12 | [TK]D-Fender | OPTIONS 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-Fender | SIP/2.0 500 Cx Unable To Comply 03023012A5004 |
15:50.41 | Posterdati | ? |
15:50.42 | [TK]D-Fender | They also don't like you sending OPTIONS packets to them a yuo have your peer configured to do. Stop. |
15:51.00 | [TK]D-Fender | And finally I see no call coming in from your phone at all |
15:51.06 | Posterdati | how can I stop to send options packet? |
15:51.13 | [TK]D-Fender | qualify=NO |
15:51.19 | Posterdati | ah I placed YES |
15:51.21 | Posterdati | ok |
15:52.08 | [TK]D-Fender | Go fix your general SIP settings relating to NAT & local addressing |
15:52.28 | Posterdati | in the firewall or sip.conf? |
15:52.30 | [TK]D-Fender | Then once you've fixed the things that are already wrong go try and show us an actual call |
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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 |
15:52.55 | Posterdati | ok |
15:52.57 | Posterdati | thanks |
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20:07.12 | viebig | Is it possible to implemetn an machine detection (like AMD) but listening in early media? |
20:11.10 | Samot | Not that I'm aware of. |
20:23.50 | seanbright | app_amd doesn't answer the channel and doesn't appear to check the channel state |
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20:30.38 | seanbright | put another way: try calling Progress(), then AMD(), and then Answer() |
20:40.16 | Slade | FreePBX on one of those $5 vultrs should be plenty to handle just a few lines right? |
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20:42.15 | seanbright | shrugs |
20:42.21 | seanbright | cheap to test |
20:43.42 | Slade | looks like it only integrates with sipstation for SMS :/ |
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21:29.20 | Samot | Yes, the SMS integration in the FreePBX UCP is for SIPStation only. |
21:30.28 | Samot | SIPStation also doesn't deliver SMS over HTTP API. |
21:30.35 | Samot | They do it over SIMPLE. |
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21:47.00 | as3ty | <PROTECTED> |
21:52.20 | igcewieling | I have access to Level3 T.38, but it is a larger commercial account. |
21:54.19 | as3ty | We probably won't be able to meet the monthly minimum for L3. |
21:54.39 | Kobaz | last time i worked with level3 the commit was like a million minutes a month |
21:54.51 | igcewieling | Kobaz: *nod* |
21:55.19 | Kobaz | i think my logrotate fixer is finally production ready |
21:55.58 | Kobaz | so 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 |
21:56.07 | Kobaz | smacks logrotate |
22:35.19 | Slade | Samot, is SIMPLE a better way to do it, or is the flowroute method better.. i guess http api integrates better |
22:36.29 | Samot | Those are just delivery methods. |
22:36.57 | Samot | SMS is done over SMPP at the carrier level. |
22:37.56 | Slade | right. just curious about the flexibility of each, if a standard is going to win |
22:39.05 | Samot | Pretty must any ITSP offering SMS is going to do it over HTTP API |
22:39.14 | Samot | SIPStation has Asterisk on both ends. |
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