| 00:00.54 | [TK]D-Fender | Remember randon no-name cheap shit bought on ebay (or China-Direct-Inc), .... tends to be SHIT |
| 00:01.03 | UncleKiwi | ahaha |
| 00:01.14 | UncleKiwi | i want to try it |
| 00:01.39 | UncleKiwi | do know of name brand stuff that will do the job |
| 00:01.48 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 00:02.33 | [TK]D-Fender | If you want to go on a budge you can use a cheap mobile phone directly with chan_mobile |
| 00:02.48 | UncleKiwi | oh wow |
| 00:02.55 | UncleKiwi | how does it interface ? |
| 00:03.06 | UncleKiwi | with asterisk |
| 00:03.38 | UncleKiwi | thanks for the tip |
| 00:06.44 | UncleKiwi | Asterisk Channel Driver to allow Bluetooth Cell/Mobile Phones to be used as FXO devices and Bluetooth Headsets as FXS devices |
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| 00:19.59 | *** part/#asterisk kharwell (kharwell@nat/digium/x-mrsuttumimfhhzxo) |
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| 02:40.50 | SaintPeteJackBoy | Okay |
| 02:41.08 | SaintPeteJackBoy | I think I installed and configured Asterisk correctly... I have a Ubuntu box. |
| 02:41.14 | SaintPeteJackBoy | I even have some DIDX numbers |
| 02:41.32 | SaintPeteJackBoy | how hard is it to get to hve those DID numbers actually call another number? |
| 02:42.04 | SaintPeteJackBoy | like, I can 11234567890, but instead, 11233214321 rings |
| 02:42.28 | SaintPeteJackBoy | I'm trying to set up a local number service for people and have the $$ for it, but my Asterisk skills are n00b level |
| 02:42.46 | SaintPeteJackBoy | at best :( |
| 02:43.00 | SaintPeteJackBoy | I've run tons of other servers and stuff before, though, just none related to phones. |
| 03:08.33 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 03:09.01 | *** join/#asterisk dkwiebe (add178ea@gateway/web/freenode/ip.173.209.120.234) |
| 03:11.20 | dkwiebe | Good day everyone, I'm looking for a product recommendation. I have a user who has color vision deficiency. The BLF keys on his phone work perfectly but he can barely differentiate between the green and red. Any suggestions? |
| 03:33.59 | *** join/#asterisk jjrh (~jjrh@2607:f0b0:8:8035:a565:bf22:6427:2da7) |
| 03:37.30 | SaintPeteJackBoy | Hmm how long does the asterisk community forums usually take to "approve" my topic? It was automatically hidden... |
| 03:43.07 | [TK]D-Fender | SaintPeteJackBoy, it's your dialplan, you can do whatever you want |
| 03:43.22 | [TK]D-Fender | Want to call out, it's just a Dial() like any other... |
| 03:52.39 | SaintPeteJackBoy | hmm, yeah but it can be configured automatic, yes? |
| 03:52.53 | SaintPeteJackBoy | I'm trying to figure out what exactly my extensions.conf should look like |
| 03:53.24 | SaintPeteJackBoy | I'm on Asterisk 13.11.2, by the way, not sure if that makes much of a difference |
| 03:53.49 | SaintPeteJackBoy | Or perhaps I am too lame and extensions.conf is not even the proper place for such an assignment? |
| 03:55.59 | SaintPeteJackBoy | part of my problem is, I do not know the correct terminology, " call forwarding " is inapplicable here... |
| 03:56.32 | SaintPeteJackBoy | My DID # is say, 17274204201, I want them to dial that, but my personal number, 17272660666 rings instead, what is that called? |
| 03:57.05 | SaintPeteJackBoy | Usually I can find documentation, but not when I am too ignorant to know the phrases ;_; |
| 04:03.10 | *** join/#asterisk pcAngel (~dade@S0106d050995743c0.vc.shawcable.net) |
| 04:04.12 | SaintPeteJackBoy | starts reading Asterisk Definitive Guide - 4th Edition... |
| 04:04.17 | SaintPeteJackBoy | Maybe my answers are in here :D |
| 04:11.40 | pcAngel | Hi guys, I have a client who, whenever I put them on any of our servers, winds up with Asterisk causing Linux to indicate load averages of anywhere between 40 and 150, with call quality symptoms present at the time. The problem builds up the longer they're on a server without asterisk being restarted -- it goes away for 1-2 hours after each asterisk restart. Asterisk 11.18 (haven't updated |
| 04:11.40 | pcAngel | recently due to dtls patches we developed & lack of issues until lately - submitted most of them back to gerrit under username Dade =\). These same servers do NOT have the problem for other customers with equal call volume/activity. They've been migrated across five servers in the past two weeks and the problem follows them to each server. We had them replace their modem, and sent them |
| 04:11.40 | pcAngel | a router with our own known-good QoS configuration. "top -H" (thread mode) combined with the "WCHAN" field (to show which syscall is causing threads to go in to uninterruptible sleep for long durations) shows suspect activity where many threads simultaneously are in "call_rwsem_down_read_failed", or less (much) commonly in the series of functions listed in this pastebin: https://imagebin. |
| 04:11.40 | pcAngel | ca/v/321XeSkCXbFK We've troubleshot drive access (iftop, interrupts) network access (irqs are spread across servers), nothing is being swapped. I suspect that the call_rwsem_down_read_failed calls and futex_wait_queue_me calls are related to threads waiting for locks. We've intermittently suspended asterisk with gdb and dumped thread apply all bt and info threads, and I can't see a |
| 04:11.40 | pcAngel | good reason. |
| 04:12.11 | pcAngel | I can submit patches upstream for any fix we have to put in, but I need some help in finding where in the source to look, or how to find where to look |
| 04:12.42 | [TK]D-Fender | SaintPeteJackBoy> hmm, yeah but it can be configured automatic, yes? <- what does automatic mean? |
| 04:13.00 | [TK]D-Fender | SaintPeteJackBoy, All call processing = extensions.conf. It does whatever you tell it to |
| 04:13.03 | *** join/#asterisk rpifan (~rpi@73.106.73.90) |
| 04:13.11 | [TK]D-Fender | SaintPeteJackBoy, It does the steps you tell it to. |
| 04:13.25 | [TK]D-Fender | "Automatic" isn't a thing |
| 04:13.58 | [TK]D-Fender | <SaintPeteJackBoy> My DID # is say, 17274204201, I want them to dial that, but my personal number, 17272660666 rings instead, what is that called? <- you need to be more specific here |
| 04:14.34 | [TK]D-Fender | They may call a DID that gets routed to your server ... but what is this "17272660666" you say you want to "ring"? |
| 04:14.50 | [TK]D-Fender | is taht to say you want to call OUT using your provider? |
| 04:15.28 | [TK]D-Fender | bkeeping in mind that this isn't "instead" either. Call comes in... then goes through step you tell it to. There is no assumed "normal" for there to be an "instead" |
| 04:17.41 | SaintPeteJackBoy | Well no |
| 04:17.48 | SaintPeteJackBoy | Say 7272660666 is my personal cell phone |
| 04:17.50 | SaintPeteJackBoy | an external number |
| 04:19.26 | SaintPeteJackBoy | I'm getting into this book and trying to understand it better. Sorry for my lack of understanding currently. :x but yeah, 7274204201 is an example of a DID that routes to my server, or at least I hope it does, I have some test DID from a company |
| 04:19.39 | SaintPeteJackBoy | I want to be able to dial that DID, but then my cell phone rings (the other number). |
| 04:19.47 | SaintPeteJackBoy | This is possible correct? |
| 04:20.24 | *** join/#asterisk jkroon (~jkroon@2c0f:f720:1:0:fcb3:e235:8b0a:1f6b) |
| 04:20.42 | [TK]D-Fender | of course |
| 04:20.50 | [TK]D-Fender | call comes in to your server ... you then tell it to Dial() out |
| 04:20.55 | [TK]D-Fender | it's a step like any other |
| 04:21.04 | [TK]D-Fender | it does whatever you tell it to |
| 04:22.04 | [TK]D-Fender | Check the time. Check the weather. Check to Laker last game score. Flip a coin. but then do somehting else if you dialed out for it last time |
| 04:23.07 | SaintPeteJackBoy | I see, and all of this happens in extensions.conf, correct? |
| 04:23.13 | [TK]D-Fender | yes |
| 04:23.24 | [TK]D-Fender | [TK]D-Fender> SaintPeteJackBoy, All call processing = extensions.conf. It does whatever you tell it to |
| 04:23.49 | SaintPeteJackBoy | Sounds fairly simple haha, is there a place I can find maybe an example or a sample of somebody doing something similar with extensions.conf so I can get a better idea of what the syntax there should look like? |
| 04:24.04 | SaintPeteJackBoy | Sorry if that is a bit much :x I'm only on about page 100 or so of the book at the moment |
| 04:24.54 | [TK]D-Fender | Dial(myproviderpeerinsipconf/1234567890,30) |
| 04:24.59 | [TK]D-Fender | Dial() <-------------------- |
| 04:25.07 | [TK]D-Fender | Dial(SIP/myproviderpeerinsipconf/1234567890,30) |
| 04:25.09 | [TK]D-Fender | ^actually |
| 04:25.22 | [TK]D-Fender | (assuming using chan_sip AKA sip.conf) |
| 04:25.56 | SaintPeteJackBoy | Yeah, seems fairly simple :X |
| 04:26.15 | SaintPeteJackBoy | haha, every time I think Asterisk is going to be daunting it turns out to be easier than I imagined it in my mind |
| 04:26.46 | [TK]D-Fender | calling out a provider is no differnt from calling another sip device registered to your server |
| 04:27.09 | [TK]D-Fender | Every normally bridged call is a Dial() like any other |
| 04:32.31 | SaintPeteJackBoy | Hmm I think part of my issue is the company I'm getting these DID from, the documentation they provide is a bit obfuscated, imo. |
| 04:32.45 | SaintPeteJackBoy | So it made the issue seem a bit more difficult than it really was |
| 04:32.49 | [TK]D-Fender | Almost nothing you should need from them |
| 04:33.20 | SaintPeteJackBoy | yeah, just the numbers :D hahaha |
| 04:33.20 | [TK]D-Fender | All any provider gives you is a SIP account to register against. The difference from one to any otheris 2-3 lines of sip.conf at most normally. |
| 04:33.47 | [TK]D-Fender | user, pass, host to send calls to, and an understanding of the #'s they'll throw at you (your DID's) |
| 04:33.56 | SaintPeteJackBoy | yeah, they had a big of stuff in the other asterisk.conf they wanted configured a certain way, but yeah the sip conf stuff was I dunno... 7 lines? |
| 04:35.04 | SaintPeteJackBoy | 9 actually, but I'm not sure if the alaw, ulaw and jitterbuffer stuff, what they wanted is likely the default configuration for that anyway |
| 04:36.03 | pcAngel | Re ^^ I just finished checking over 'thread apply bt all' on an instance of asterisk that was still running after the client signed out for the day, and there's no noticeable issues with threads - ie deadlocks left over from their last round of working, no threads in a state that isn't reasonable. |
| 04:36.25 | SaintPeteJackBoy | I'm glad you helped me see Dial() as more of a function that gets passed parameters... I've been programming in PHP and stuff for many years, so it makes a lot more sense to me that way, [TK]D-Fender. |
| 04:40.48 | [TK]D-Fender | "core show application dial" |
| 04:40.49 | [TK]D-Fender | ^ |
| 04:42.54 | SaintPeteJackBoy | Dial(Technology/Resource[&Technology2/Resource2[&...]][,timeout[,options[,URL]]]) |
| 04:43.00 | SaintPeteJackBoy | :D |
| 04:47.33 | [TK]D-Fender | For parameters refer to CLI help, for general theories, the book, for the stuff in between, the WIKI |
| 04:49.13 | *** join/#asterisk matrix1233 (~matrix123@41.230.143.234) |
| 04:50.36 | SaintPeteJackBoy | yeah :D man I just figured out that context=(USER) is the same [user] as is in the extenstions.conf file |
| 04:50.40 | SaintPeteJackBoy | a revelation, for me lol |
| 04:52.23 | [TK]D-Fender | Sounds like progress |
| 04:54.14 | SaintPeteJackBoy | I should have stuck to the wiki :( part of my problem came from trying to follow the terrible information provided by the DID company, I mean, their tutorial contains such gems as if u send it to guest:guest@domain.com/12126555763" |
| 04:54.40 | SaintPeteJackBoy | (forgot a " somewhere after gems as) |
| 04:55.07 | SaintPeteJackBoy | Typing 'you' instead of 'u' might not seem like much, but speaks volumes about their ability to convey a point. |
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| 07:41.28 | SaintPeteJackBoy | Hmm, okay, I tried to set up Google Voice with asterisk, and when I asterisk -r I get a flood on xmpp stuff, both receive and send. |
| 07:41.32 | SaintPeteJackBoy | I'm guessing that is not normal. |
| 07:45.17 | SaintPeteJackBoy | looks into less secure apps... |
| 07:46.21 | SaintPeteJackBoy | omg that was it lol |
| 07:46.39 | SaintPeteJackBoy | holy crap tonight has been eventful |
| 07:47.18 | SaintPeteJackBoy | now the GV works, and I understand the asterisk configs on a pr0-n00b level at this point and even got some custom hold music :D |
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| 09:45.16 | SaintPeteJackBoy | Ugh, :( |
| 09:45.36 | SaintPeteJackBoy | asterisk only works from my local ip and I can't for the life of me figure out how to get it to work either from my DNS or from my external ip |
| 09:45.39 | SaintPeteJackBoy | ANy advice? |
| 09:48.34 | SaintPeteJackBoy | Hmm, I was using bindaddr, but may need to ue externip |
| 10:00.59 | *** join/#asterisk KaliLinuxGR (~alexandro@unaffiliated/kalilinuxgr) |
| 10:10.49 | TandyUK | externip, along with relevant firewall rules, and dns entries |
| 10:11.15 | TandyUK | oh hes quit |
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| 11:17.08 | Guest9030 | Something makes me furious, randomly some outgoing calls are missing sound. The SDP block sends an internal address however comedia and force_rport is activated and hey it works on 50% of the calls. I fail to see why some calls with the exact same setup would fail. |
| 11:17.11 | Guest9030 | *pulls hair* |
| 11:18.59 | Guest9030 | The only difference I can find is that in the working example I receive a RTCP Source Description. |
| 11:19.07 | Guest9030 | I do not even know what that is |
| 11:22.17 | Guest9030 | RTCP carries a persistent transport-level identifier for an RTP source called the canonical name or CNAME |
| 11:22.32 | Guest9030 | Ah? So does that mean the Asterisk is never told where to send the RTP when it lacks the RTCP? |
| 11:34.19 | file | when you say "SDP block sends an internal address" do you mean their SDP? |
| 11:54.13 | Guest9030 | The agents INVITE to Asterisk has a internal sdp |
| 11:54.39 | Guest9030 | This is true for both examples, both working and not working. |
| 11:54.53 | file | are they on the same network? |
| 11:55.07 | Guest9030 | No, the Asterisk is on the internet while agents on the other side behind FW |
| 11:55.40 | file | if you don't get media from them then Asterisk can't switch to their mapped IP address and port, and you'd get no audio |
| 11:56.03 | Guest9030 | Something like that yes. It bugs me that the next time I dial it works. |
| 11:56.06 | Guest9030 | 50/50 as said above |
| 11:56.13 | Guest9030 | and the signaling looks the same |
| 11:56.30 | Guest9030 | Asterisk tries to send rtp to 172.17.17.17 |
| 11:56.45 | Guest9030 | Next call it sends it to 196.69.69.69 |
| 11:56.56 | file | you'd need to determine why Asterisk is not getting media from it |
| 11:57.19 | Guest9030 | Yeah I was hoping the SIP would tell me that but no, guess I'll have to jack up the core debug for this and hold my thumbs. |
| 11:58.06 | file | if the IP address and port in the SDP from Asterisk to the device is reachable by the device, then it's something outside of Asterisk |
| 11:58.17 | file | and a core debug wouldn't tell you |
| 11:59.48 | Guest9030 | The SDP is always an internal adress, and it is not reachable from Asterisk |
| 11:59.56 | Guest9030 | Even tho it works sometimes |
| 12:00.03 | file | that's not what I said though |
| 12:00.09 | file | the SDP from Asterisk to the device |
| 12:00.29 | file | the SDP from the device to Asterisk can contain an internal address given the config you are using |
| 12:03.29 | Guest9030 | That SDP has the asterisk IP |
| 12:03.31 | Guest9030 | which is correct |
| 12:03.52 | file | then you'd need to look outside of Asterisk to determine why it is not getting media |
| 12:04.09 | file | ie: do a packet capture and see if the RTP arrives at Asterisk at all |
| 12:04.16 | file | if not, then it's something else |
| 12:06.30 | Guest9030 | Yeah, I do not receive rtp from the agent |
| 12:06.35 | Guest9030 | I am just trying to send to it. |
| 12:07.37 | file | then it's not something in Asterisk |
| 12:10.33 | Guest9030 | I can see from a Wireshark on the Agent machine that it is sending RTP to the Asterisk |
| 12:10.38 | Guest9030 | But yeah, it never reaches it. |
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| 14:44.50 | rah | [Nov 15 14:44:19] ERROR[2752]: tcptls.c:609 handle_tcptls_connection: Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) |
| 14:44.53 | rah | [Nov 15 14:44:19] WARNING[2752]: tcptls.c:684 handle_tcptls_connection: FILE * open failed! |
| 14:44.56 | rah | umm.. help? |
| 14:48.12 | Guest9030 | https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial |
| 14:48.27 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 14:49.58 | rah | Guest9030: that doesn't help me I'm afraid |
| 14:50.03 | rah | I've read that |
| 14:50.19 | rah | and https://wiki.asterisk.org/wiki/display/AST/SIP+TLS+Transport |
| 14:50.22 | Guest9030 | Well I do not know what will help you since you did not really elaborate your question. |
| 14:50.45 | rah | Guest9030: that's OK, thanks :-) |
| 14:50.51 | rah | is there anyone else here who can help? |
| 14:51.20 | rah | obviously the error message is completely unhelpful so I'm wondering if anyone here has had experience with this kind of asterisk error before? |
| 14:51.20 | [TK]D-Fender | You should already have started by chowing more of the call around that, being specific about whatt version you're running, the scop of when it's happening, the operating environment, etc |
| 14:51.50 | Guest9030 | Also, try increasing the core debug. That info would be helpfull |
| 14:51.55 | rah | this isn't from a call, this seems to be during registration |
| 14:52.38 | rah | [TK]D-Fender: in my experience, when it comes to openssl errors like this, none of that kind of information helps |
| 14:53.35 | [TK]D-Fender | In my experience only spitting out 1 error line and expecting it to lead to a proper resolution helps even less |
| 14:53.43 | rah | but this is asterisk 13.8.0 on openwrt |
| 14:54.01 | rah | [TK]D-Fender: indeed |
| 14:54.03 | Guest9030 | rah: what does an elevated core debug say? |
| 14:54.07 | rah | [TK]D-Fender: it's a bit of a shot in the dark |
| 14:54.22 | [TK]D-Fender | Running on the worst platform imaginable... |
| 14:55.00 | rah | Guest9030: running "asterisk -rvvvvv" instead of "asterisk -rvvv" gives no more information |
| 14:55.07 | Guest9030 | core set debug |
| 14:55.30 | Guest9030 | also make sure the logger.conf allows console output |
| 14:56.09 | rah | Guest9030: how do I determine the current debug level? |
| 14:56.48 | Guest9030 | Unsure. Just make sure debug is at the console => row in logger.conf |
| 15:00.02 | igcewieling | rah, just set it. |
| 15:00.23 | igcewieling | by default the debug level is 0 unless otherwise set in a config file. |
| 15:01.49 | *** join/#asterisk rpifan (~rpi@73.106.72.200) |
| 15:03.07 | rah | increasing the debug level to 10 and retrying registration produces, again, the two lines above and nothing else |
| 15:04.12 | rah | http://codepad.org/mFMn3r03 |
| 15:04.58 | rah | alas, I fear I will have to RTFS |
| 15:06.42 | *** join/#asterisk kharwell (kharwell@nat/digium/x-dsukevpwidjtaeoc) |
| 15:06.42 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:08.29 | [TK]D-Fender | I'd look at SIP debug as well... |
| 15:10.33 | rah | [TK]D-Fender: no effect |
| 15:11.03 | rah | http://codepad.org/GZA3y042 |
| 15:11.24 | igcewieling | looks to me like Asterisk is not set up to trust the sender's certificate issuer |
| 15:11.32 | igcewieling | But I've not done much with TLS |
| 15:12.01 | rah | igcewieling: what makes you say that? |
| 15:12.04 | igcewieling | maybe you didn't load a key. e_tcptls_connection: FILE * open failed! |
| 15:13.09 | rah | igcewieling: what makes it look like Asterisk is not set up to trust the sender's certificate issuer? |
| 15:14.04 | igcewieling | because that is the same error *I* got when trying to do TLS. I decided it was not worth all the extra works. also try this: https://www.google.com/search?q=handle_tcptls_connection%3A+FILE+*+open+failed+solved |
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| 15:14.58 | rah | igcewieling: I've already googled the error message |
| 15:15.26 | igcewieling | so this was not helpfuil: http://forums.asterisk.org/viewtopic.php?f=14&t=91525 |
| 15:15.27 | rah | or I should say, I've searched the web for the error message |
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| 15:16.05 | igcewieling | In any case, I've given my advice, you don't have to take. I need to get back to work anyway. |
| 15:16.15 | rah | igcewieling: you mean the thread where the person says "no problem, I solved it!" without saying anything at all about how they solved it? |
| 15:16.20 | rah | igcewieling: yeah, saw that, didn't help :-) |
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| 16:31.55 | pcAngel | rah: I don't know if this will help, and I didn't read a lot of the earlier messages, but one thing that gave me a lot of problems when configuring srtp & https for webrtc was that a lot of the SSL config options in the config files don't get reset if you delete them from the config file and issue a reload (vs a restart) -- for example, if you load a config with dtlscertfile=x, dtlsprivatekey=y |
| 16:31.55 | pcAngel | and dtlscafile=z, but then try reloading with a new config for the same peer in sip.conf with only dtlscertfile=a and dtlsprivatekey=b (no cert file, ie self signed) then that cert file config line is going to stick in there until a restart. Same things happen with dtlsenable, dtlsverify etc, at least in Asterisk 11. Seems like the kind of thing they would backport if they fixed it. |
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| 19:09.12 | kubrrick | hello guys i am trying to encrypt my iax2 trunk between 2 asterisk but it doesn't works, i am using auth=rsa both side. can someone help me ? thanks |
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| 19:37.19 | jeffspeff | when doing a netstat -a should i see sip listening as "localhost.localdoma:sip" ? it seems odd to me that it says localhost.localdomain instead of 0.0.0.0 or the actual interfaces ip address. |
| 19:38.27 | Kuunsi | localhost.localdomain means "listen on localhost only" |
| 19:39.15 | jeffspeff | thx |
| 19:39.52 | Kuunsi | but, you want to use `netstat -l`, to see what's listening. netstat -a will return current connections |
| 19:41.16 | Kuunsi | or maybe even -tulpe, which will also tell you process name and stuff |
| 19:42.14 | jeffspeff | Kuunsi do you use pjsip? |
| 19:42.25 | Kuunsi | no |
| 19:44.02 | jeffspeff | on my production box, which uses sip "netstat -l | grep sip" shows that it's binding to the ip of the interface. when i run the same command on my test box which has pjsip i see that it's binding localhost.localdomain, yet it can still send/receive traffic. |
| 19:44.56 | kubrrick | !clear |
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| 20:07.40 | drmessano | jeffspeff: Does your box have a public IP or private IP? |
| 20:07.55 | jeffspeff | private |
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| 20:41.19 | bravvve | i have asterisk installed on wheezy openvz template and icant connect,host machine is debian wheezy |
| 20:45.06 | spearo_ | assuming it's a fresh install - to start with are you in root? |
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| 21:00.51 | bravvve | well,i have fulle controll,to every think |
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| 21:49.20 | rrittgarn | using mixmon via AMI - sending recordings to voicemails - the caller ID on the messages for outbound calls is always unknown - is there an easy way to just update that to the dialed number? |
| 22:16.42 | rah | error:00000005:lib(0):func(0):reason(5) |
| 22:17.02 | rah | it would appear that asterisk is calling the wrong function; ERR_get_error() instead of SSL_get_error() |
| 22:18.00 | rah | and in master, too |
| 22:20.14 | war9407 | question, I used to use asterisk to make calls with acrobits softphone or bria but since Ive last tried it, there have been a few updates, whenever I make calls now it seems to call itself? |
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| 22:58.30 | [TK]D-Fender | war9407, does it? |
| 22:58.37 | [TK]D-Fender | * does what you tell it to.... |
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| 23:19.31 | saul | hello, i have some ring groups and when i dial them all the member phones ring. however, i dont hear any ringing on the phone that is placing the call |
| 23:19.47 | saul | i just get silence until someone answers or doesnt answer, but no indication that the other end is ringing |
| 23:20.09 | saul | is this normal ? |
| 23:20.40 | dym | saul: do you hear the other party, after pickup? |
| 23:23.16 | dym | Also, what do you mean by ringgroups? |
| 23:23.29 | dym | Please provide logs/excerpt of config. |
| 23:23.50 | dym | Or are you using queues? |
| 23:24.56 | [TK]D-Fender | Don't need config, just the call debug |
| 23:25.15 | dym | ohwell, just beeing thorough :) |
| 23:25.20 | dym | but you're right ofc. |
| 23:25.50 | dym | afaik theres no such thing as "ring groups" that ring multiple lines. either queues, or dial to multiple lines. |
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| 23:39.01 | saul | dym: once they answer we can hear each other |
| 23:39.22 | dym | saul: as [TK]D-Fender mentioned: Please provide Debug CLI output |
| 23:39.32 | saul | i'm using FreePBX by the way, it has a Ring Groups feature where you can gather a few extensions so that when you call one, all those extensions ring |
| 23:39.43 | saul | dym: ok |
| 23:42.01 | [TK]D-Fender | #freepbx |
| 23:42.08 | [TK]D-Fender | ~freepbx |
| 23:42.08 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 23:42.18 | [TK]D-Fender | pick it up there |
| 23:42.30 | saul | ok |
| 23:43.53 | dym | "FreePBX is unable to be supported here." odd syntax, imo. |
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