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| 10:17.53 | wasanzy | using zoiper to dial into asterisk, I can't hear the sound even though I selected the right codecs |
| 10:18.03 | wasanzy | a-alaw and u-law |
| 10:26.45 | evilman_work | wasanzy: don't you hear sound on both sides or only in one direaction? |
| 10:26.51 | evilman_work | **direction |
| 10:27.51 | wasanzy | when I run the test sound in zioper, I hear sound, but when I dial in, I hear nothing at all |
| 10:29.30 | wasanzy | my call: http://pastebin.com/Z8CkH3jh |
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| 10:47.59 | r00f | no sound is usually related to network and nat, not codec |
| 10:55.29 | wasanzy | r00f: so what could be the problem? |
| 10:55.49 | wasanzy | it was working fine but after reinstalling my pc and zoiper, it stops working |
| 10:56.12 | wasanzy | am installing twinkle too to see |
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| 12:33.32 | dan_j | Hi, I'm starting to use AMI for a few things and when i do, Asterisk repeatedly crashes. The logs don't appear to show anything obvious. Its just like the process has been terminated without a chance for it to write anything. |
| 12:33.40 | dan_j | Has anyone experienced this? |
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| 12:59.16 | anonymouz666 | hello |
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| 13:07.21 | WIMPy | dan_j: Not something I managed so far. |
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| 13:14.47 | anonymouz666 | so... |
| 13:15.01 | anonymouz666 | whatsapp API is making some sucess integrating systems out there |
| 13:18.03 | Johnny5_ | is alive. |
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| 13:30.51 | Kunsi | hm, i'm trying to get VoiceMailMain() working, however it hangs up just after announcing it's going to play new messages ... |
| 13:30.54 | Kunsi | http://pastebin.com/b5njuqSr |
| 13:32.23 | Kunsi | any ideas what's causing this? emailed wav plays fine, msg0000.wav plays fine directly on server,too |
| 13:33.14 | *** join/#asterisk Quastor (d44700fe@gateway/web/freenode/ip.212.71.0.254) |
| 13:35.39 | Quastor | Hi Guy's, does anyone know software to detect phone numbers with OCR or similar wich makes them clickable. Not only for browsers but also in CRM or in-mail phone numbers |
| 13:37.01 | [TK]D-Fender | Quastor: That'd be like something you'd find in a "screen reader" for the visually impaired. This is highly specialized and not something you'd find here. |
| 13:37.22 | [TK]D-Fender | Quastor: Once you find a solution we can help you try to integrate it with Asterisk. |
| 13:37.57 | [TK]D-Fender | Kunsi: "sip set debug on" <- You need to actually look at the SIP comms to confirm which side aborted and why |
| 13:46.10 | Quastor | Ok but [TK]D-Fender but you are not aware of any API doing that? |
| 13:46.12 | Kunsi | [TK]D-Fender: http://pastebin.com/131Zu5sQ <- i don't really see any errors? |
| 13:47.10 | Quastor | [Jan 16 14:41:57] Retransmitting #8 (no NAT) to 173.208.168.34:5074: [Jan 16 14:41:57] SIP/2.0 401 Unauthorized |
| 13:47.26 | Kunsi | that's a noth american IP |
| 13:47.26 | Quastor | no nat? |
| 13:47.44 | Kunsi | someone trying to contact my asterisk |
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| 13:47.56 | Kunsi | all here is 192.168.0.* |
| 13:49.20 | Quastor | Did you try to change NAT mode from rport to NAT? |
| 13:49.54 | Kunsi | uh |
| 13:49.55 | [TK]D-Fender | QuastorOk but [TK]D-Fender but you are not aware of any API doing that? <- once you HAVE a number there are several ways you can make Asterisk call out and bridge 2 ends |
| 13:50.07 | Kunsi | there is no NAT between asterisk and my phone |
| 13:50.23 | [TK]D-Fender | Quastor: Getting that number however has nothing to do with telephony at all... that is something for a programming / UI channel |
| 13:51.54 | Quastor | True |
| 13:53.34 | [TK]D-Fender | Kunsi: We don't see the end of the call in there |
| 13:53.46 | [TK]D-Fender | Kunsi: grab the ENTIRE call from beginning to end |
| 13:55.55 | Kunsi | http://pastebin.com/4TcgXgXv |
| 13:56.11 | Quastor | No NAT, are u sure? :p |
| 13:56.12 | Kunsi | phone is 192.168.0.52, asterisk is 192.168.0.20 |
| 13:56.31 | Kunsi | Quastor: why would i use NAT if it's all in same local net? |
| 13:57.02 | Quastor | Ok ;) |
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| 13:58.25 | [TK]D-Fender | Kunsi: [Jan 16 14:41:58] == Spawn extension (internal, 98, 1) exited non-zero on 'SIP/91-00000063' |
| 13:58.47 | Kunsi | ? |
| 13:59.00 | [TK]D-Fender | Kunsi: Seems like dialplan simply exited and ran out of steps. Funny thing is it sees there should be a message : [Jan 16 14:41:57] == Parsing '/var/spool/asterisk/voicemail/default/800/INBOX/msg0000.txt': Found |
| 13:59.40 | [TK]D-Fender | Kunsi: But in the core debug we don't see it trying to actually play that audio recording. I'm thinking you may be MISSING the actual audio file and VMM is quitting out because of that failure |
| 14:00.14 | Kunsi | <PROTECTED> |
| 14:00.17 | Kunsi | msg0000.alaw msg0000.gsm msg0000.txtmsg0000.ulaw msg0000.wav |
| 14:00.29 | [TK]D-Fender | check permissions as well. |
| 14:00.33 | Quastor | idd |
| 14:00.38 | Kunsi | (there's a space between txt and msg0000.ulaw) |
| 14:01.12 | Kunsi | everything's owned by asterisk user (asterisk itself recorded this message with no errors |
| 14:02.21 | [TK]D-Fender | Kunsi: PASTEBIN msg0000.txt |
| 14:02.22 | [TK]D-Fender | ~pb |
| 14:02.23 | infobot | A "pastebin" is 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://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:03.53 | Kunsi | http://pastebin.com/PMWXnrbk |
| 14:04.20 | Kunsi | (did you notice i've nopasted everything unless `ls` of INBOX/?) |
| 14:04.53 | [TK]D-Fender | Reflex... |
| 14:05.02 | Kunsi | ⺠|
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| 14:07.36 | [TK]D-Fender | Kunsi: Something is wrong here. If the files look right all we see is VMM aborting without any statement and your dialplan terminating normally from there. |
| 14:08.13 | [TK]D-Fender | Kunsi: Might be a bug, but I can't see any special circumstance that it would happen. |
| 14:08.50 | [TK]D-Fender | Kunsi: issues.asterisk.org <- I might post that debug up on the tracker to have a dev look at it. |
| 14:11.01 | Kattyroo | gooooood morning cupcakes! |
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| 14:15.09 | Johnny5_ | Moooorning |
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| 14:18.22 | Kattyroo | [TK]D-Fender: good morning, fenderbender |
| 14:18.32 | [TK]D-Fender | Kattyroo: Mew. |
| 14:18.40 | Kattyroo | [TK]D-Fender: how're you this morning dear? |
| 14:18.55 | [TK]D-Fender | Kattyroo: Hours from freedom... |
| 14:19.09 | Kattyroo | [TK]D-Fender: woohoo! big plans for tonight? |
| 14:19.31 | [TK]D-Fender | Kattyroo: I'm currently working on a "Bobby Tables" issue for the first time... |
| 14:19.57 | Kattyroo | is that the sql injection thing? |
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| 14:21.10 | [TK]D-Fender | Kattyroo: yup |
| 14:21.25 | Kattyroo | [TK]D-Fender: sounds like a project for sure (= |
| 14:21.39 | Kattyroo | [TK]D-Fender: you're not going to spend your entire friday night working on it, are you? |
| 14:21.46 | [TK]D-Fender | Kattyroo: Shouldn't be. just need to sanitize a few fields and have never done much of this before |
| 14:22.04 | [TK]D-Fender | Kattyroo: Not for tonight.. doing that right NOW. Tonight is quiet.... |
| 14:22.05 | Kattyroo | nods |
| 14:22.32 | Kattyroo | [TK]D-Fender: well hey, any friday night i don't have to work is awesome in my book (= |
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| 14:37.30 | necronian | I'm completely new to asterisk/telephony in general and I'm having some issues. I'm trying to connect an grandstream HT-503 into a key phone system using the fxs port to use as a voicemail system. |
| 14:37.45 | necronian | I have the hello world example working from the wiki and I can see the HT-503 register with asterisk. When I call the extension I can see in the HT-503 admin page that it detects ringing.... but then I see no activity in asterisk. |
| 14:38.10 | necronian | I'm not really sure where to go from here? Could someone point me in the right direction? |
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| 14:40.50 | Kunsi | necronian: did you enable verbose mode? |
| 14:40.57 | Kunsi | (core set verbose 10) |
| 14:41.35 | necronian | I launched it with a lot of vvvvvv's |
| 14:41.39 | necronian | like 9 or 10 |
| 14:41.51 | Kunsi | then you should see calls, yeah |
| 14:48.36 | necronian | is there an easy way for me to quickly dial out from the console? to see if that is working at least? I'm looking at the originate command and I don't understand it. |
| 14:51.35 | [TK]D-Fender | necronian: just set up a softphone to do a test call. |
| 14:51.49 | [TK]D-Fender | necronian: Also, that would only be for dialing OUT... inbound handling is another matter |
| 14:51.57 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 14:52.18 | [TK]D-Fender | necronian: Did you read their maule for pstn-to-voip calls? |
| 14:53.02 | [TK]D-Fender | manual* |
| 14:53.29 | necronian | I have several manuals open, I only sort of understand what I'm reading and I have a non standard setup to start which is confusing things |
| 14:54.18 | necronian | I'm running it into an old digital phone system using an analog adaptor that allows me to map an extension number to a regular phone |
| 14:54.20 | Kunsi | necronian: best would be, get asterisk (voip-only) working, then try to integrate POTS |
| 14:55.16 | necronian | Well I have a softphone on my computer and dial in to hear hello world |
| 14:55.23 | necronian | so I know asterisk works? |
| 14:55.40 | [TK]D-Fender | necronian: what you have plugged in doesn't matter. If its signalling is normal then the only thing you care about is the HT to * |
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| 14:56.08 | [TK]D-Fender | Show us what you set up on the HT for PSTN |
| 14:56.48 | [TK]D-Fender | [09:55]necronianso I know asterisk works? <- this really doesn't mean much. Just because your softphone works doesn't mean it's accepting a possible call from the HT trying to forward on sue to the FXO port ringing... |
| 14:57.12 | [TK]D-Fender | necronian: You need to prove if it is even trying to talk to your server |
| 14:57.20 | [TK]D-Fender | necronian: "sip set debug on" <- from * CLI |
| 14:57.34 | [TK]D-Fender | necronian: You should see an INVITE being sent to pass a call over. |
| 14:58.32 | necronian | This is all I've configured so far http://pastebin.com/3cFxfUzi |
| 14:59.10 | [TK]D-Fender | necronian: screenshot your HT side |
| 14:59.19 | [TK]D-Fender | picpaste.com , etc |
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| 15:07.28 | necronian | [TK]D-Fender: for some reason that was really hard http://necronian.uberpurple.com/stuff/settings.png |
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| 15:09.13 | Kunsi | necronian: "deskserv" looks wrong. does "deskserv" resolve to you asterisks IP? |
| 15:09.21 | necronian | Kunsi: yes |
| 15:09.54 | Kunsi | password is set, too? |
| 15:10.23 | Kunsi | "outgoing call without registration" should be off, i think |
| 15:10.25 | stefan27 | quick JIRA-question: Assignee says Assignee: Stefan Engström (me) on a jira ticket (ASTERISK-24673), does that mean the ball is in "my court" i.e. no one else will look at it until i perform some action? I tried to 'send it back to developers' but im not sure I succeeeded, do I have to click "Request Feedback" again? |
| 15:10.45 | necronian | the password was set yes |
| 15:12.34 | necronian | Well now I really borked it, for some reason now it thinks the phone is off the hook :S |
| 15:12.36 | stefan27 | aha nvm I think it's okay because status went back to "Triage" |
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| 15:18.51 | necronian | Kunsi: turning off outgoing call without registration does nothing |
| 15:19.02 | Kunsi | hm |
| 15:19.28 | [TK]D-Fender | necronian: that pic you gave what the ***FXS PORT*** |
| 15:19.33 | [TK]D-Fender | was for the* |
| 15:20.05 | necronian | yes the fxs port |
| 15:20.48 | [TK]D-Fender | necronian: Which if you are using as an "extension" on the other system is wrong |
| 15:20.50 | Kunsi | [TK]D-Fender: that's correct. FXO connects to telephone network, FXS connects to phone |
| 15:21.11 | [TK]D-Fender | necronian: Are you treating it as a "line" on the other system, or as a "phone"? |
| 15:22.14 | necronian | it should appear to the phone system as an analog phone |
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| 15:22.36 | necronian | I know that it is hooked up correctly to the phone line |
| 15:22.46 | [TK]D-Fender | necronian: then you're doing it wrong |
| 15:22.50 | necronian | I can call it and I can see on the admin page that it is ringing |
| 15:23.00 | [TK]D-Fender | necronian: the FXS port on the HT is for YOU to plug a phone into. |
| 15:23.39 | [TK]D-Fender | necronian: it needs to ACT line a phone... that is what the FXO port is for |
| 15:23.53 | [TK]D-Fender | necronian: it needs to ACT like a phone... that is what the FXO port is for |
| 15:26.13 | necronian | [TK]D-Fender: I just tried plugging it into the fxo port and it shows up as busy and the phone system sees it as not installed |
| 15:26.40 | [TK]D-Fender | Shows up WHERE as "busy"? |
| 15:26.43 | necronian | when it is plugged into the fxs port the phone system sees it and I can ring it and see ringin/on hook status |
| 15:26.57 | [TK]D-Fender | You haven't shown having configured anything on it either so my expectations are low... |
| 15:27.16 | necronian | The status page of the HT-503 |
| 15:27.58 | necronian | I haven't configured anything other than the sip info and outgoing call without registration that Kunsi recommended |
| 15:27.59 | Kunsi | necronian: had same issue with my spa3102. have to plug it into fxs port, but configure fxo port, then it worked |
| 15:28.38 | necronian | Kunsi: well that is just confusing! |
| 15:28.55 | [TK]D-Fender | Kunsinecronian: had same issue with my spa3102. have to plug it into fxs port, but configure fxo port, then it worked <_ No, that is horribly wrong |
| 15:29.23 | Kunsi | [TK]D-Fender: it's not. port description was sawpped |
| 15:30.14 | Kunsi | necronian: http://www.grandstream.com/files/7412/9468/8057/ht503_single_2.png <- did you connect your phone system to "line" or to "phone"? |
| 15:31.17 | necronian | Kunsi: Okay I'm just going to wire both of them right now and try everything twice |
| 15:31.24 | [TK]D-Fender | necronian: DON'T |
| 15:32.02 | Kunsi | necronian: connect your phone system to "line", and configure FXO port |
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| 15:33.06 | [TK]D-Fender | FXO ports are for connecting to lines. That means they ACCEPT ringing voltage from the telco. FXS ports are for connecting PHONES. FXS port SUPPLY voltage and aren't designed to ACCEPT it. |
| 15:33.31 | [TK]D-Fender | Get this wrong and you'll send ring voltage to something that doesn't take that load and you can fry something |
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| 15:36.14 | Kunsi | hm, [TK]D-Fender, regarding my voicemail issue, Matt Jordan says "not a bug" :/ |
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| 15:38.12 | [TK]D-Fender | Kunsi: All good and fine... so what DOES he think it is? :) |
| 15:38.30 | Kunsi | Thanks for your comments. This does not appear to be a bug report and we are closing it. We appreciate the difficulties you are facing, but it would make more sense to raise your question in the support tracker |
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| 15:41.27 | cyford33 | as far as fail2ban goes, does pjsip write to the same logs as asterisk in same format, when registering? |
| 15:42.32 | *** join/#asterisk darkdrgn2k (~darkdrgn2@209.90.253.66) |
| 15:42.42 | darkdrgn2k | Hey any old school PBX techs around that know stuff about the old nortels? |
| 15:43.21 | newtonr | darkdrgn2k, i used to go to http://www.tek-tips.com/ when I worked with the older stuff |
| 15:43.35 | newtonr | darkdrgn2k, lots of old guys on there willing to share |
| 15:43.58 | darkdrgn2k | problem is asside from changin line names i havent touched this stuff to much |
| 15:44.13 | darkdrgn2k | got a client that just ordered a telephone line form bell didnt realize they need to feed it into the pbx |
| 15:44.41 | darkdrgn2k | ..its nto even a BCM its a NMTM040 |
| 15:45.57 | newtonr | I've wiped it all from my mind. Been ~9 or 10 years since I've touched Nortel stuff at this point. I'd go to the nortel forums on tek-tips |
| 15:46.08 | [TK]D-Fender | darkdrgn2k: that doesn't look like a proper model # |
| 15:46.44 | darkdrgn2k | ...dam photo is a little blur NNTM040CCTBD maybe |
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| 15:47.45 | [TK]D-Fender | Looks like an ICS |
| 15:48.43 | [TK]D-Fender | darkdrgn2k: Basically all the Norstar stuff is about 95% identical really... go grab the admin guide https://www.google.ca/#q=norstar+ICS+admin+guide |
| 15:48.58 | [TK]D-Fender | darkdrgn2k: Hardly Raw-Cat Sigh Hence |
| 15:49.18 | darkdrgn2k | thank you! |
| 15:50.02 | [TK]D-Fender | darkdrgn2k: My company had a Norstar 8x24 when I started 20 years ago (and had it for a many years prior). one of our brnaches used to have an ICS. They're basically the same really. Dumb key-systems... |
| 15:51.11 | darkdrgn2k | some one once told me you can get a one-time keycode for VNC access into the system... since its running windows (drum roll) NT |
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| 15:52.57 | [TK]D-Fender | darkdrgn2k: I only know of the BCM using NT. Never a CICS, etc |
| 15:53.00 | mjordan | Kunsi: two points. |
| 15:53.06 | mjordan | 1. Put your full logs on the issue. |
| 15:53.12 | mjordan | Don't link to a pastebin, since those go away. |
| 15:53.24 | Kunsi | ok |
| 15:53.31 | darkdrgn2k | maybe i donno i did my best to stay away from key systems....... and now this |
| 15:53.45 | mjordan | 2. Don't use allow=all. Configure your SIP peer to match to what it is actually offering. |
| 15:53.50 | mjordan | That should fix the issue you're having. |
| 15:53.59 | mjordan | If it does, then that issue is known about and already reported. |
| 15:54.31 | mjordan | But, keep in mind the issue tracker is *not* a support forum. It's up to the issue reporter to provide all of the information as per the bug reporting guidelines, which are linked to when you filed the issue |
| 15:54.44 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines |
| 15:54.54 | darkdrgn2k | hey random question.. 16 mins into a call between asterisk and a polycom.... all of a sudden the people cant hear each other even though packet capture from the server shows both channels are active/... both ppl are saying "hellow... hello" what coudl it be? |
| 15:55.00 | Kunsi | i don't have allow=all, if you're referring to allowed codecs in sip.conf |
| 15:55.19 | mjordan | you have a lot of codecs set up for that peer, regardless of what it happens to be |
| 15:55.42 | mjordan | your peer is offering g722,ulaw,alaw,gsm,aal2,g729 |
| 15:55.50 | mjordan | configure your peer's codecs to match that. |
| 15:56.03 | mjordan | At a minimum, just do allow=!all,g722,ulaw,alaw,gsm |
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| 15:57.47 | Kunsi | arr, sh*t. moved allowed codecs from peer definition into general section, and "dis" got missing â¹ |
| 15:57.57 | Kunsi | so, yeah, i had allow_all |
| 15:58.01 | Kunsi | allow=all* |
| 15:58.58 | Kunsi | let me test that |
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| 16:03.57 | [TK]D-Fender | mjordan: He didn't get a codec warning and every other sound file played. through VMM. No statement of * even trying.... |
| 16:06.12 | [TK]D-Fender | mjordan: is just ... stopped. |
| 16:17.25 | Kunsi | mjordan: no, setting codecs does not resolve my issue |
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| 16:21.23 | Kunsi | same as before: http://pastebin.com/Tt4kHKE2 - tried with an old and with an new recorded message |
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| 16:29.55 | mjordan | Kunsi: k, I'll reopen. Provide the full DEBUG log generated using the instructions on the wiki, along with 'sip set debug on'. Make sure you attach your sip.conf, extensions.conf, and voicemail.conf. |
| 16:30.17 | Kunsi | ok |
| 16:30.29 | mjordan | Do you have the issue on both 11 and 13? |
| 16:30.34 | mjordan | what specific version are you testing with? |
| 16:31.42 | Kunsi | i've had this issue with 11.9.0, but stopped looking at it since i didn't have asterisk connected to an actual phone line |
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| 16:32.06 | Kunsi | updated to 13.1.0 this week, connected an ATA, and tried to get voicemail working, but no success |
| 16:32.50 | Kunsi | 13.1. was downloaded and compiled yesterday, so it should be pretty up-to-date |
| 16:37.49 | mjordan | k, so you did have it with 11 as well |
| 16:38.01 | mjordan | can you attach the .ulaw file that is supposed to be played back as well? |
| 16:39.31 | Kunsi | only msg0000.ulaw, or all (ulaw, alaw, wav, txt)? |
| 16:41.09 | mjordan | go ahead and attach them all |
| 16:41.16 | mjordan | I'd like to make sure the files are actually generated correctly |
| 16:44.31 | Kunsi | everything should be uploaded now |
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| 16:54.40 | Kunsi | hm, issues.asterisk.org won't work with latest safari/mac |
| 16:54.57 | Kunsi | server requests a certificate to identify myself |
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| 20:04.44 | WangDang | Asterisk the Definitive Guide mentions SIPVicious as a good vulnerability tester. When I go to the vicious site, the release date on the downloads page says Dec 2012. That's not very active development. Is it still considered a good tool? Any others people recommend? |
| 20:05.32 | WangDang | I'm reading the 3rd Ed (asterisk 1,8) btw, not the most recent edition |
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| 20:09.57 | [TK]D-Fender | yes |
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