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| 01:57.34 | ZeroGee | Hello! I'm looking for some help figuring out why my TrySystem() call in my dialplan is not working, even though running the exact same command (minus a channel variable) is capable of being ran successfully from the asterisk cli with ! in front. I get APPERROR returned for SYSTEMSTATUS. Checking the 'full' logs do not reveal much, from what I can tell. |
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| 05:54.05 | ZeroGee | For those that were curious, asterisk user wasn't capable of executing what was being called in TrySystem, and !command was being run as root... |
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| 15:20.15 | dan_j | Hi. I've got an asterisk server with two sip peers. When I call from one to the other, I get a foreign ring tone, even though i've set the variable in indications.conf to uk. What could be causing this? Is it changeable? |
| 15:22.21 | [TK]D-Fender | that has no impact on OOB |
| 15:26.28 | dan_j | Where does the OOB come from? The called endpoint itself? |
| 15:27.13 | dan_j | Actually, how is that possible if Dial() multiple endpoints in one go? |
| 15:27.32 | dan_j | It's surely generated by asterisk in that case? |
| 15:27.57 | [TK]D-Fender | SIP phones generate tone when they are TOLD they are "ringing" |
| 15:28.19 | [TK]D-Fender | If the state of the call is not "answered" then tones are OOB |
| 15:28.52 | dan_j | That doesn't make any sense. When I dial an internal extension, I get a different OOB tone to when I dial an external extension. Both unanswered. |
| 15:29.23 | dan_j | If the sip phone was told ringing and generates it's own tone, the tone should be the same regardless of where the call is going to. |
| 15:29.40 | [TK]D-Fender | Show the call |
| 15:30.15 | dan_j | Ok. I'll have to paste it later. My wife wants me to stop working for now. |
| 15:30.36 | [TK]D-Fender | if you dial with FORCED ringing, then it will answer and shove it inband |
| 15:30.55 | [TK]D-Fender | So you could be screwing up your interpretation of "answered" while not paying attention |
| 15:31.11 | dan_j | I'm not forcing ringing in this dial. |
| 15:31.36 | [TK]D-Fender | Come back with eveidence... ALWAYS have the ability to prove evidence otherwise all we can do is guess |
| 15:31.48 | [TK]D-Fender | And guessing wastes time |
| 15:32.16 | dan_j | It's just an experiement. I've got a SIP wholesale provider with some decent rates. And another SIP provider with OK rates. The cheaper one keeps playing the wrong ring-tone and my clients complain that they can't tell if the callee is roaming. |
| 15:33.07 | dan_j | Then I realised that my internal calls also play an international ring-tone rather than the normal UK ring tone which the more expensive SIP provider plays. So I'm trying to work out what is generating the tone. |
| 15:33.42 | dan_j | I'll run some tests shortly and see what I get. Probably with a simple Ring() cmd. |
| 15:33.49 | [TK]D-Fender | :work out" means showing the actual call. |
| 15:34.03 | dan_j | Will do. Later. Thanks. |
| 15:34.06 | [TK]D-Fender | Go do what you have to do and come back with the evidence so we can actually make a judgement on something |
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| 20:01.02 | lvlinux | Can Read() be made to put what it receives into a global variable? |
| 20:03.28 | lvlinux | Or better yet, write what it receives to a file that can be read later? |
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| 20:16.55 | [TK]D-Fender | You can put whatever you want into one |
| 20:16.58 | [TK]D-Fender | it's your dialplan.... |
| 20:17.35 | lvlinux | I know I can do it with Set, just curious if Read would do it directly. Core show application read doesn't say you can, so I'm assuming not. |
| 20:17.45 | lvlinux | But how do i write to a file? |
| 20:23.25 | [TK]D-Fender | By something to shells out |
| 20:23.29 | [TK]D-Fender | take your pick. |
| 20:23.36 | [TK]D-Fender | there's a half dozen different ways |
| 20:23.54 | [TK]D-Fender | Do you need it in a phsycial file? |
| 20:24.01 | lvlinux | Ah, I just found the FILE function. |
| 20:24.25 | lvlinux | Well I was going to do a variable first, but then thought it might be good to be persistent across an * restart. |
| 20:24.55 | lvlinux | What I'm doing is letting someone dial an extension and enter an emergency contatc number that can be used later on when someone presses the emergency button. |
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| 20:38.21 | snmp | nihao |
| 20:38.47 | snmp | well |
| 20:38.59 | snmp | call file + h323 |
| 20:39.04 | snmp | anyone |
| 20:39.06 | file | don't call me! |
| 20:39.13 | snmp | :D |
| 20:39.24 | snmp | ill call you later |
| 20:46.56 | snmp | http://pastebin.com/NxkDqyn3 |
| 20:48.40 | snmp | asterisk 1.8.13 |
| 20:49.17 | lvlinux | snmp: where have you defined ${play} ? |
| 20:49.30 | snmp | it doesnt matter |
| 20:49.50 | snmp | the problem is timing |
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| 20:50.28 | snmp | lines are slightly altered |
| 20:51.17 | lvlinux | says 0 attempts. I would think the waittime would only apply if it attempted. |
| 20:52.32 | snmp | it says 0 retries |
| 20:52.40 | snmp | _re_ |
| 20:53.53 | snmp | ..i guess.. |
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| 20:55.11 | lvlinux | huh? I don't see anywhere in that pastebin where the word "retries" is shown. ?? |
| 20:55.21 | snmp | well |
| 20:55.31 | WIMPy | snmp: Did you also read the 1st line of the log? |
| 20:55.33 | snmp | the phone exactly rings |
| 20:55.47 | snmp | about codec? |
| 20:56.00 | WIMPy | yes |
| 20:56.18 | snmp | er |
| 20:56.42 | snmp | well, everything perfect below 60 seconds |
| 20:57.17 | WIMPy | o.O |
| 20:57.24 | snmp | yep |
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| 21:11.43 | snmp | lvlinux: WaitTime 30, MaxRetries: 0 |
| 21:11.47 | snmp | Log: |
| 21:11.48 | snmp | [May 23 00:07:29] NOTICE[2580]: chan_ooh323.c:3909 ooh323_convertAsteriskCap |
| 21:11.48 | snmp | [May 23 00:08:00] NOTICE[26485]: pbx_spool.c:353 attempt_thread: Call failed |
| 21:11.49 | snmp | [May 23 00:08:00] NOTICE[26485]: pbx_spool.c:356 attempt_thread: Queued call |
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| 21:13.43 | snmp | 0 attempts after 30s timeout |
| 21:14.46 | lvlinux | Perhaps someone could better help you troubleshoot if you pastebinned a bit more info (and more accurate info). Otherwise, I'd look at your codec settings as WIMPy pointed out. |
| 21:15.09 | snmp | its just a notice |
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| 22:38.00 | linuxmint | Test |
| 22:38.58 | file | Test failed |
| 22:49.26 | lvlinux | lol |
| 22:49.35 | linuxmint | Hmm, inbound calls aren't ringing. |
| 22:49.53 | linuxmint | Also, I registered a landline and a softphone, but only the landline rings on inbound calls. |
| 22:50.02 | linuxmint | Would like a hand if possible. |
| 22:52.56 | lvlinux | k shoot |
| 22:53.47 | linuxmint | lvlinux: k, well, the phone and the mobile's softphone are both registered and call out ok, but incoming if faulty. |
| 22:54.04 | linuxmint | Phone incoming rings, but won't answer. Mobile won't ring with the incoming call. |
| 22:54.40 | lvlinux | I think you'll need to pastebin some config, and also define "won't answer" |
| 22:55.02 | linuxmint | k, let me access the logs. |
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| 23:06.08 | Prelude2004c | hey everyone good day |
| 23:06.33 | Prelude2004c | anyone know what this means ? [May 22 19:04:45] WARNING[28982][C-00000005]: codec.c:363 ast_codec_samples_count: Unable to calculate samples for codec opus |
| 23:06.33 | Prelude2004c | [May 22 19:04:45] WARNING[28982][C-00000005]: translate.c:407 framein: no samples for opustolin48 |
| 23:06.39 | Prelude2004c | i am trying to get opus workign with asterisk |
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| 23:18.08 | linuxmint | lvlinux: so, won'at answer means the phone rings, pick up and no tone. I've access the /var/log/asterisk/cdr-csv, which shows the call answered, but still looking for the Asterisk Elastix config. |
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| 23:21.24 | lvlinux | Ewww, Elastix. |
| 23:21.40 | lvlinux | linuxmint: your Asterisk config is /etc/extensions.conf and /etc/sip.conf |
| 23:21.50 | Prelude2004c | anyone know how to get opus going correctly on asteirsk ? |
| 23:23.28 | lvlinux | Prelude2004c: it's a tricky thing. I haven't tried it but you might check here: https://github.com/meetecho/asterisk-opus and |
| 23:26.06 | lvlinux | http://highsecurity.blogspot.com/2014/05/opus-codec-with-transcoding-on-asterisk_13.html |
| 23:27.16 | linuxmint | lvlinux: thanks, looks like Asterisk Elastix has a different path, which I'm looking up. |
| 23:28.08 | lvlinux | Yes Elastix uses FreePBX IIRC. So you might check in #freepbx as they might be a bit more able to help. Asterisk with a GUI is quite a different beast configuration wise than vanilla, like we do here. |
| 23:28.43 | lvlinux | linuxmint: you should sstill be able to run the asterisk console though and see what's going on better though. |
| 23:28.56 | lvlinux | "asterisk -rvvv" |
| 23:29.18 | lvlinux | Then try to cause your problem and it may give you a clue. |
| 23:29.44 | linuxmint | http://dpaste.com/2VA7YVH |
| 23:32.06 | lvlinux | linuxmint: yikes version 1.8? old stuff. But anyway, you have to do something for that console to show anything. When you make or receive a call you'll see stuff come up there in the console. |
| 23:38.02 | linuxmint | k, will setup a 2nd mobile to make test calls and try later. Thanks. |
| 23:42.25 | lvlinux | great :) |
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