00:01.22 | josefig | hi, i know this is not elastix application, i need to setup a lower time between call and call on the call center application, but i don't know where to find it, can you give me a hand please? I'm using elastix |
00:01.37 | josefig | <PROTECTED> |
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01:18.46 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
01:19.43 | n\ack | OK, so restarting asterisk cleared the phantom pheature [sic] |
01:21.40 | n\ack | WTF? Shouldn't dialplan reload and module reload features be the entirety?? |
01:22.18 | n\ack | Na Ja. |
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06:04.36 | BeeBuu | hi,all. is webrtc make asterisk's performance low? |
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06:20.34 | drmessano | no |
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08:32.15 | Aur313 | hi everyone |
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08:35.41 | Aur313 | I'm using port 5070 for SIP, and I noticed that the "From" uri does not contain the port. this disturb some of my equipements... does anyone has any idea? (I tried from_domain but it adds some "[" "]" in around the address... and it doesn't seems good to me...)) |
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13:46.18 | jkroon | sigh .. ok, so tcptlsserver.c at some point stops accept()ing new connections. |
13:48.32 | jkroon | well, the main/tcptls.c code. |
14:18.53 | mneth1 | Anyone around that could help with a stereo recording sync/alignment issue? |
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14:20.57 | jkroon | mneth1, g729? |
14:21.12 | mneth1 | jkroon: g711u. |
14:21.22 | mneth1 | it's sync in the stereo recording |
14:21.41 | jkroon | hmm, that doesn't have VAD afaik, which is what normally causes it in that case. |
14:21.51 | jkroon | more specifically the CNG ... |
14:22.23 | mneth1 | Are you familiar at all with this bug from the tracker? https://issues.asterisk.org/jira/browse/ASTERISK-26875?jql=text%20~%20%22mixmonitor%20sync%22 |
14:22.53 | mneth1 | well more specifically this: https://gerrit.asterisk.org/#/c/5209/ |
14:23.15 | jkroon | interesting. never had that specific problem. Used Monitor() mostly. |
14:25.43 | Samot | menth1: What is the actual issue? |
14:25.53 | Samot | mneth1: ^^ |
14:27.11 | mneth1 | Samot: We're still having MixMonitor stereo recordings end up out of sync even after upgrading to 13.18.3, and it looks like that fix has been in since 13.16. |
14:27.48 | Samot | OK |
14:27.56 | Samot | So the exact same thing? |
14:28.06 | Samot | 183 with SDP but no RTP? |
14:29.21 | mneth1 | I need to dig into more pcaps to confirm that, it does still seem linked to packet loss like in this: https://gerrit.asterisk.org/#/c/5209/ |
14:30.31 | Samot | WEll.. |
14:31.14 | Samot | I'm not saying you don't have problems. But I've seen plenty of problems and latch on to an old bug ticket because "it's like their issue". |
14:31.37 | Samot | So if you are pointing to a specific bug issue, then that means you are having the exact same issues reported. |
14:32.09 | Samot | If you are out of sync in other ways then focusing on the "I thought this was fixed already" line of thought is bad. |
14:32.32 | mneth1 | Yeah for sure. I can't say for sure it's the exact same issue. Just researching where to look or what to test and I haven't been able to come up with much yet. |
14:32.53 | Samot | You're also using PJSIP? |
14:34.35 | mneth1 | Still using chan_sip right now. I do plan to try to migrate to pj_sip at some point though. |
14:34.49 | Samot | Ok |
14:34.50 | Samot | So |
14:35.02 | Samot | It is in no why the same issue |
14:36.02 | Samot | Its hars to get help when you point at a known issue for people to look at then turns out youre not even in that type of setup |
14:36.32 | Samot | Please provide actual details of your issue |
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14:39.22 | mneth1 | Hm. We're using 13.18.3, chan_sip, and MixMonitor's t and r options to create separate recording files for the two stereo channels but in some instances the two recordings are ending up as different lengths. This makes it so when the two recording are merged into one, the conversations get out of sync/alignment. I've tried with the MixMonitor b option as well and the issue was still present. |
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16:20.35 | wim_ | Hello, is it normal that when using pjsip with path support that when using contact sip:s@x.x.x.x it doesn't seem to work? |
16:22.01 | wim_ | The Asterisk was receiving a REGISTER with Supported: path and a Path header, but when asterisk tried to send a qualify, it would ignore the path and not add a route header |
16:23.09 | wim_ | when i changed the registration on the other end to have a contact with sip:sip_user@x.x.x.x it now work. It does't seem to like the contact s@x.x.x.x |
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17:32.56 | imcdona | What determines if Asterisk is going to install default conf files in /etc/asterisk/ ? Is it merely the presence of any files /etc/asterisk/ ? There's no make target that says "don't put any config files in /etc/asterisk " |
18:07.00 | rmudgett | imcdona: Doing "make samples" puts the sample config files in /etc/asterisk |
18:07.22 | rmudgett | Otherwise you have to put files there. |
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22:51.52 | Kobaz | is there a way to run a single line of dialplan |
22:52.15 | Kobaz | say you have a macro/context/etc, and you want to jump right into a specific label, run that code, and then jump back |
22:52.27 | Kobaz | without modifying the original macro/context |
22:52.49 | Kobaz | GoSubSingleLine(foobar,baz,labelname) |
22:53.02 | Kobaz | i guess i could write one |
22:53.50 | Kobaz | maybe just a general introspection would be useful |
22:54.25 | Kobaz | like, Set(dialplan=${READ_DIALPLAN(exten.context,prio)}) |
22:54.41 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_Gosub |
22:54.48 | [TK]D-Fender | Or ... you could just USE Gosub |
22:55.09 | Kobaz | right, i know how to use gosub, i'm talking about a specific-case |
22:55.30 | [TK]D-Fender | Kobaz> say you have a macro/context/etc, and you want to jump right into a specific label, run that code, and then jump back <- it already DOES that |
22:55.37 | Kobaz | no it does not |
22:55.44 | Kobaz | you need to explicitly Return(); |
22:55.48 | [TK]D-Fender | Single line? No. Jump to specific point, sure |
22:55.55 | Kobaz | or implicitly run out of dialplan in that context |
22:56.18 | Kobaz | run a single line, and jump back, without any existing modifications |
22:56.29 | [TK]D-Fender | There is no "single line". |
22:56.32 | Kobaz | sure there is |
22:56.34 | [TK]D-Fender | THere is "go", and you go. |
22:56.47 | Kobaz | <PROTECTED> |
22:56.50 | Kobaz | there's your single line |
22:56.53 | [TK]D-Fender | no... |
22:56.54 | Kobaz | or whatever |
22:57.05 | [TK]D-Fender | there is no way to jump BACK if thtere ar other priorities after it |
22:57.10 | Kobaz | right, didn't think so |
22:57.21 | Kobaz | so i like my READ_DIALPLAN idea the best |
22:57.25 | Kobaz | it's the most flexible |
22:57.34 | Kobaz | no jumping needed... and you can read as many lines as you like |
22:58.03 | Kobaz | basically just a quick query to confirm if that exists or not, if it was something i may have overlooked |
22:58.13 | [TK]D-Fender | No. |
22:58.31 | [TK]D-Fender | What actual usage scenarion do you see as making this worth creating? |
22:58.41 | Kobaz | working with vendor-specific asterisk platforms |
22:58.50 | Kobaz | writing custom code for said platforms |
22:59.06 | [TK]D-Fender | So only to hack other peoples "moving target" code.... |
22:59.22 | Kobaz | if need be |
22:59.23 | [TK]D-Fender | yeah, * isn't built with trying to get competing code to cooperate |
22:59.41 | Kobaz | right, that's not the intention |
22:59.46 | Kobaz | but there's no harm in giving the tools |
23:00.00 | Kobaz | a READ_DIALPLAN type thing would work well for my own platform |
23:00.16 | Kobaz | where sometimes i generate dialplan to be used by other bits of dialplan, and i only need some snippits |
23:00.29 | [TK]D-Fender | You COULD hack this up in external scripting |
23:00.45 | Kobaz | like hunt groups, i have lists of hunt group members, and if i want member 1, i can gosub to member 1, which does a Set() Return(); |
23:00.47 | Kobaz | and then i have my value |
23:01.16 | Kobaz | grabbing dialplan would be a lot more direct |
23:01.30 | Kobaz | you could hack anything any which way |
23:01.32 | Kobaz | doesn't make it a good idea |
23:02.06 | [TK]D-Fender | You're hacking someone else's GUI ... so yeah, swimming upstream. |
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23:23.57 | kambei | I'm having a little difficulty with an Asterisk machine using a Sangoma A200 with a daughterboard, which should provide access to 8 POTS lines. When I run the config, I'm seeing only two channels. What am I doing wrong? Does channell == POTS line? |
23:24.20 | kambei | I'm far from adept when it comes to Asterisk, needless to say. |
23:26.07 | kambei | I have my system.conf in /etc/dahdi which is showing fxsks 1 and 2, and those are reflected in the configs in /etc/asterisk ... I seem to have access to only two lines, or am I misunderstanding? |
23:26.20 | [TK]D-Fender | a defined channel is a single port in the case of a card like that |
23:26.27 | [TK]D-Fender | Show us your configs and the status dump |
23:26.31 | [TK]D-Fender | ~pb |
23:26.32 | 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. |
23:26.33 | [TK]D-Fender | ^^^ |
23:27.45 | kambei | https://pastebin.ca/3964211 <- /etc/dahdi/system.conf |
23:28.03 | kambei | 02:01.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card |
23:28.17 | [TK]D-Fender | Yup, that's 2 channels |
23:28.30 | [TK]D-Fender | So if you have more you expect to use... they're not configured |
23:29.05 | kambei | [TK]D-Fender: What am I doing wrong? I was using the wanrouter config tools to generate this, and that's what I got |
23:29.24 | kambei | [TK]D-Fender: Why in particular is it detecting only two channels? |
23:29.27 | [TK]D-Fender | That file only specifies 2 channels |
23:29.33 | [TK]D-Fender | Go change it |
23:29.45 | kambei | But that's what was generated. Why did it generate two channels? Why not one or three? |
23:30.16 | [TK]D-Fender | I don['t know what it think's it's detecting, but we shouldn't have to care. You know what you bought right? You know you have the right modules to support more channels on it? |
23:30.32 | kambei | Yes |
23:30.45 | [TK]D-Fender | so just fix your config and specify the rest |
23:31.15 | kambei | When I try to add more channels, I get this... |
23:31.41 | kambei | http://pastebin.ca/3964212 |
23:35.02 | [TK]D-Fender | wanpipe configs too..... |