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| 06:52.40 | saxa | Hello, I want to use a Grandstream HT813 ATA as my sip trunk, to receive and send calls. Right now I set it up but when I try to place a call from my voip phone I get a "Forbidden from sip....." answer on my asterisk console. But I can answer calls. |
| 06:53.23 | saxa | Any idea what is makeing asterisk to think that my voip phone cant call out ? |
| 06:54.10 | saxa | I searched the net but could not find an answer as nearly all answers were ask your provider |
| 06:54.21 | saxa | but I do not have a sip provider |
| 06:58.25 | saxa | https://pastebin.com/fuUkCcd1 |
| 06:58.53 | saxa | this is the sip show peers and sip.conf |
| 07:00.01 | saxa | in some way asterisk does not route the call to the correct direction I think, of course due to a misconfiguration. |
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| 07:26.41 | saxa | c u l8r |
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| 10:21.59 | Martin` | Hello world! |
| 10:22.53 | Martin` | I've configured asterisk 16 with phoneprov, the users/macs are in users.conf but when I restart asterisk the routes of phoneprov are not correctly loaded, I have to reload phoneprov.conf before it works, is this a known problem? |
| 10:23.06 | Martin` | looks like he does not get the macs during startup |
| 10:23.30 | Martin` | create routes like cfg.xml instead of cfgHEREMAC.xml |
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| 14:22.07 | Samot | So is Asterisk 17 on the calendar for 2019? Or will a year be skipped for stable like in 2015? |
| 14:25.30 | life_of_e | saxa: the HT813 has two different ports for the SIP connection. By default it uses port 5062 for the FXO connection. You need to add that to relevant section in sip.conf |
| 14:25.48 | life_of_e | The FXS port uses 5060. |
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| 14:29.48 | Samot | Or pjsip.conf depending on your tech choice. |
| 14:30.12 | life_of_e | yeah, he quoted his sip.conf file |
| 14:30.25 | Samot | I wonder what is going to happen with all these FXS/FXO in the near future. |
| 14:30.36 | life_of_e | Meaning? |
| 14:30.53 | Samot | All the documentation relies heavily and 99% of the time on chan_sip. |
| 14:31.14 | Samot | Which is on its death bed. |
| 14:32.37 | life_of_e | As long as the port can be specified it should be fine, right? |
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| 14:35.21 | Samot | That was never an issue before. |
| 14:35.31 | Samot | You can specific whatever port you'd like for chan_sip. |
| 14:35.41 | saxa | life_of_e: can i just change that port in the HT813 admin panel in the FXO tab ? |
| 14:36.04 | life_of_e | saxa: no, you can't have the FXS and FXO port sharing the same UDP port number |
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| 14:36.16 | saxa | life_of_e: let me see how I add that port to sip.conf |
| 14:36.24 | life_of_e | It's one device with one IP, it has to use two different port numbers |
| 14:36.42 | life_of_e | You just add it as port=NNNN to the configuration |
| 14:36.45 | saxa | i c |
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| 14:36.57 | life_of_e | Just log into the HT813 and verify the SIP port for the FXS and FXO |
| 14:37.37 | life_of_e | Samot: can you not do the same for pjsip? |
| 14:37.48 | Samot | Of course you can. |
| 14:38.04 | Samot | 10:30:54 AM <Samot> All the documentation relies heavily and 99% of the time on chan_sip. |
| 14:38.14 | life_of_e | Oh I see what you meant |
| 14:38.23 | Samot | No one has had the guts to do it with PJSIP |
| 14:38.47 | Samot | The fact that 10 year old documentation on the Internet can still apply to most things in Asterisk today is a double edge sword. |
| 14:38.59 | saxa | life_of_e: https://pastebin.com/aHrFMune |
| 14:39.09 | saxa | life_of_e: i am still getting that forbidden |
| 14:39.11 | Samot | On the one edge you have the fact that certain core things just don't change. |
| 14:39.31 | Samot | On the other edge, you have people not using current things because of that same fact. |
| 14:40.08 | life_of_e | The auto conversion script managed to get my FXO configuration *almost* right. It lost the port= value but otherwise it worked with the FXS configuration. |
| 14:41.10 | saxa | life_of_e: yeah it is 5062 as port in the FXO section. |
| 14:41.54 | life_of_e | Is FXO enabled (mine defaulted to off) and have you verified the authentications? |
| 14:43.21 | saxa | life_of_e: Ok, let me check that well again, step by step. |
| 14:43.27 | saxa | life_of_e: thanks for now ! |
| 14:46.09 | life_of_e | Yep, the grandstream docs aren't great but be sure to check all the configuration options especially things like autoprovision and various registration and authentication settings (like turning off authenticate incoming invite, etc) |
| 14:46.43 | saxa | i c, i left all as default for now, not touched anything. |
| 14:47.18 | life_of_e | Yeah, the defaults didn't work for me either, I did have to make an adjustment here and there but I can't remember what those were compared to the defaults |
| 14:47.55 | saxa | perfect |
| 14:48.30 | saxa | I will play a bit with it during the weekend |
| 14:48.35 | saxa | i hope to make it work |
| 14:48.51 | life_of_e | Sorry, I tinkered with mine until it worked but I didn't note what I changed. I only backed up the configuration afterwards so I could always restore it to the known working state. |
| 14:49.03 | saxa | the thing is that is probably something with authentication for sure, since it says forbidden from >( |
| 14:49.06 | wonderworld | i want to implement busygroups, ie. only one peer in a group can be called. if one peer in the group is in a call, the others should signal busy when other calls come in. i thought about configuring the busygroups in astdb and check all channels in the group before a dial. |
| 14:49.18 | wonderworld | is there a more simple solution? |
| 14:50.06 | life_of_e | It does work, I've been using it for the same purpose. Keep in mind there's a delay when it dials out. It takes about 10 seconds between the time you finish dialing on your phone to the time the FXO finishes dialing and reports back a ringing signal. |
| 14:50.11 | saxa | life_of_e: i want to make it work with a BananaPi Pro board and astersisk on it, so it wont need to leave my whole computer up just for the phone when I am away. |
| 14:50.28 | life_of_e | I run Asterisk on a RaspberryPi, works fine |
| 14:50.45 | saxa | cool |
| 14:53.00 | life_of_e | My new toy is being delivered today, a SIP intercom gateway |
| 14:53.50 | saxa | what is this _ |
| 14:53.52 | saxa | ? |
| 14:53.52 | wonderworld | life_of_e: ha, i am building one just now with a raspberry pi |
| 14:54.18 | wonderworld | will be used in a car park |
| 14:54.23 | saxa | life_of_e: do you know if I have to register with both ports FXS and FXO ? |
| 14:54.48 | life_of_e | Yes, you do have to have both ports authenticate |
| 14:55.16 | bipolar | Can anyone point me to a comprehensive tutorial/documentation on using the AMI command UpdateConfig? The documentation is incredibly confusing. |
| 14:55.22 | saxa | so I have to create one section for the FXS and one for the FXO port in sip.conf ? |
| 14:55.50 | life_of_e | Yes, that's correct. The device acts as two endpoints, they just share the same IP |
| 14:56.46 | saxa | i think i read somewhere on the net its not needed |
| 14:57.06 | life_of_e | I have two sections in my configuration |
| 14:57.22 | life_of_e | I gave the two ports different user IDs |
| 14:58.28 | life_of_e | and of course the section names match those user IDs |
| 14:58.47 | life_of_e | wonderworld: a car park? |
| 15:00.51 | life_of_e | saxa: You should have both ports set up to allow outgoing calls without registration. That would be the registration you don't want. |
| 15:01.46 | wonderworld | life_of_e: yes as intercom on the gates when you drive out. to contact the guard on duty . |
| 15:02.34 | life_of_e | wonderworld: oh, that's what a car park is, ok. :) |
| 15:02.58 | wonderworld | maybe it's not the correct english word, i am unsure :) |
| 15:04.08 | life_of_e | Sounds like a British term so it's English :) |
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| 17:23.36 | kevinnn | how can I get the name of a channel so I can hang up on it? |
| 17:24.44 | file | you can list channels on the console... |
| 17:25.25 | kevinnn | https://pastebin.com/mQgAwCpy |
| 17:25.27 | kevinnn | @file |
| 17:25.35 | file | right |
| 17:25.50 | file | those are the channel names, and you can tab complete in the console too for the hangup command |
| 17:26.00 | kevinnn | Local/4000@from-internal should be a channel name right? |
| 17:26.16 | file | it's an incomplete one, but tab completion will fill it in |
| 17:26.22 | kevinnn | jeez |
| 17:26.30 | kevinnn | there's a ton of zero's after it |
| 17:26.32 | kevinnn | why |
| 17:26.49 | file | because that is how Local channel naming works. |
| 17:26.59 | kevinnn | okay cool, thanks for the help |
| 18:00.57 | Reinhilde | fef |
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| 19:24.10 | Deeewayne | anyone have an opinion on what the best solution for phone calls through a browser are these days? WebRTC vs SIP over WebSocket? jssip vs sip.js ? I've tried to get a sample working Asterisk master and can get the registrations working, I have even gotton calls to work, but no audio in the browser |
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| 19:24.36 | Deeewayne | RTP tracing seems to indicate packets are being sent and my firewall is open, but not sure where the issue is |
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