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| 00:33.23 | maruen | Hi all, I'getting the folowing error when trying to call with asterisk: AT^DDSETEX failed |
| 00:33.34 | maruen | Could anyone help to solve this? I'm newbie with asterisk |
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| 00:41.22 | maruen | anyone? |
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| 00:46.40 | WIMPy | Looks like you're trying to use a dongle that fails. |
| 00:46.50 | maruen | Yes, but it worked sometime ago |
| 00:46.58 | maruen | and it stopped to work |
| 00:47.26 | maruen | and the state is shown after that failing in: Dialing |
| 00:47.38 | maruen | and it stays that foverer |
| 00:47.54 | maruen | Any clue on what may be happening? |
| 00:49.09 | WIMPy | What has changed since it worked? |
| 00:49.48 | maruen | I changed the file extensions.conf , but I made a backup, and I put the same that was working before |
| 00:49.56 | maruen | and now, it didnt work anymore |
| 00:50.15 | maruen | I made all configs through freepbx |
| 00:50.33 | maruen | create an extension, trunk and a outgoing route through there |
| 00:51.11 | WIMPy | Dialplan is surely not related. |
| 00:51.38 | WIMPy | And noone knows what freepbx might have done. |
| 00:51.53 | maruen | Dialplan not related? How do I fix then? |
| 00:53.04 | WIMPy | Find out what has changed since it worked. |
| 00:53.46 | maruen | How? I'm a complete newbie in asterisk |
| 00:53.54 | WIMPy | Maybe the audio device has gone or got renamed or something? |
| 00:55.22 | maruen | I dont know. I can send sms |
| 00:55.27 | maruen | but not call |
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| 03:36.04 | rue_house | can asterisk do the sip and leave the two clients to directly transfer the rtp data? |
| 03:37.17 | WIMPy | yes |
| 03:38.25 | rue_house | is it a flag or ? there must be a distinction between connections that can be routed directly, and those that needs to be forwarded |
| 03:39.01 | WIMPy | directmedia/directrtpsetup/canreinvite |
| 03:41.12 | rue_house | flags in sip.conf? |
| 03:41.32 | WIMPy | yes |
| 03:41.42 | rue_house | thankyou |
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| 03:53.27 | hackershack | Hi. Just deployed my first Asterisk server with freePBX. I've confiugred a few extensions and have it all working pretty well! I do have a concern though, I established a call with an outside caller and the caller was able to push *1 to stop\start recording and *2 to transfer themselves. Are they supposed to be able to do that? If not, how can I disable callers from having that access? |
| 03:54.51 | WIMPy | #freepbx |
| 03:58.31 | hackershack | thanks. i thought that was just a GUI on top of Asterisk. I didnt see any options there so I thought iâd start here. Iâll give them a shout. |
| 03:59.32 | WIMPy | It is, but that where you do your configuration/ |
| 03:59.37 | WIMPy | . |
| 04:00.05 | hackershack | Ah, understood. thanks. |
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| 04:06.55 | Surye | I have been trying to find any documentation or examples, and I am hoping maybe someone here has this set up. I have MeetMe going, but now that we have virtualized the server the quality has really degraded, I think due to DAHDI timing. I am trying to switch to ConfBridge, but the way we need it is to have a single extension everyone dials to, it then prompts for the conference room number, then they are joined to it after entering it. We have a we |
| 04:08.13 | WIMPy | Your message was cut off. |
| 04:10.00 | Surye | Ah boo, was trying not to spam with a bunch of lines. |
| 04:11.10 | Surye | WIMPy: Where did it cut off? |
| 04:11.32 | WIMPy | <Surye> then they are joined to it after entering it. We have a we |
| 04:12.04 | Surye | We have a webui to create meetme rooms, but I am okay if we don't have any kind of self service for creating the conf rooms, I just can't get ConfBridge to work in that model |
| 04:13.15 | Surye | From what I can tell, ConfBridge works out of the box by creating meeting rooms that are dialed like extensions themselves (using FreePBX mostly but am trying to aproach it by hand if needed). |
| 04:13.56 | WIMPy | The only difference is that you don't have to pre-define the rooms. |
| 04:15.01 | Surye | So the issue I see with that personally is that if someone miss dials the conf room, they wouldn't know it was invalid till they waited long enough in silence waiting for others to join |
| 04:16.39 | WIMPy | Then validate it on your dialplan. |
| 04:18.41 | Surye | Okay, that makes sense, and is a little more obvious now that I have expressed my full problem and requirements, I didn't think of that exactly when I was in the thick of it |
| 04:20.54 | hackershack | WIMPy: nobody answering at #freepbx , can you recommend a search term? what feature am I looking for? âStar codesâ just gives me a list of *69, *67 and what they do. |
| 04:25.01 | WIMPy | Not sure what they call the "features" there. |
| 04:25.48 | hackershack | Ok, so how would I accomplish this in plain Asterisk config files? |
| 04:26.10 | hackershack | Is it even possible? |
| 04:26.34 | WIMPy | Sure. You have to specify the allowed features in Dial(). |
| 04:28.24 | hackershack | Ok. Let me see what I can turn up. |
| 04:28.25 | hackershack | thanks. |
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| 06:36.40 | Surye | Hmm, Playback can find a sound file no problem, conf-getconfno, but using the same in Read says it can't find the sound file. Do you need to specify them differently for both? |
| 06:38.07 | Surye | Nevermind, space before the string... |
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| 13:10.09 | Get_The_Fish | I keep having Asterisk 13.7.2 crash on me, and when it comes back up I get "No SIP authenticator registered. Assuming authentication is not required" for quite a while before asterisk magically starts |
| 13:10.22 | Get_The_Fish | And when I say quite a while, I mean like 20 mins |
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| 13:28.28 | Get_The_Fish | Are we still not recommending that we run asterisk at the current version in production? If not, which version should I be running? |
| 13:30.38 | file | there are 2 currently supported "versions" both of which are used in production environments |
| 13:30.42 | file | Asterisk 11 and Asterisk 13 |
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| 13:52.33 | Get_The_Fish | file: ok, so are we recommending the 13.7.2 tag for production then? |
| 13:53.16 | file | I can say it is in use in production environments |
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| 14:08.14 | Get_The_Fish | OK thanks |
| 14:08.47 | Get_The_Fish | I'm not doing anything special, just a file based little PBX, nothing fancy whatsoever. |
| 14:12.25 | Get_The_Fish | So I'm guessing you cant use a relative path with app Background? |
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| 19:38.59 | LemensTS | If I have a Public IP on my asterisk server, in sip.conf for the [general] context do I just put nat=no ? I've had nat=force_rport/comedia for a few years and it has worked, but today I am getting nat like issues. And I am wondering why I did not have it at nat=0 |
| 19:39.05 | LemensTS | nat=no I mean |
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| 19:40.04 | LemensTS | Its an Azure VM with an Instance IP (aka public ip) |
| 19:42.04 | [TK]D-Fender | "nat like issues" is something you should be showing us the debug to match |
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| 19:48.46 | LemensTS | [TK]D-Fender: http://pastebin.com/QTmQh3UJ |
| 19:49.04 | LemensTS | Dialing to 831-920-4866 |
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| 19:54.44 | [TK]D-Fender | all I see are retransmit attempts |
| 19:54.54 | [TK]D-Fender | nothing is failing auth witth a finalitty that would cause filtering |
| 19:55.01 | [TK]D-Fender | So you're going to have to filter them yourself |
| 19:56.15 | LemensTS | Do you have a utility that you read these sip debug dumps in? I just used notepad++ surely their is something faster |
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| 19:56.45 | [TK]D-Fender | I just read htem in my browser |
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| 20:26.24 | Janos | hey there, question, is there any way to tell dahdi_genconf to not turn crc4 on my E1 lines by default ? My provider is reluctant to enable it on their end and every time I run dahdi_genconf I have to manually disable it |
| 20:27.35 | Janos | already checked https://github.com/asterisk/dahdi-tools/blob/master/xpp/genconf_parameters but did not find anything relevant there |
| 20:27.57 | [TK]D-Fender | Stop running genconf |
| 20:28.04 | [TK]D-Fender | Why would your situation be CHANGING? |
| 20:28.17 | [TK]D-Fender | Make your configs and leave them along like any sane person would do |
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| 20:30.18 | [TK]D-Fender | alone* |
| 20:30.23 | [TK]D-Fender | checks out for a while |
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