| 00:00.05 | Ionic | nope |
| 00:00.33 | Ionic | ah, indeed... it crashed the kernel module |
| 00:01.00 | Ionic | I can't even SIGKILL asterisk anymore |
| 00:02.25 | WIMPy | nice |
| 00:03.11 | Ionic | hehe |
| 00:03.28 | Ionic | given chan_capi wouldn't even compile with asterisk 11, that's way better |
| 00:03.41 | Ionic | admittedly not perfect, but there's always a catch |
| 00:04.05 | WIMPy | Noone said it would be easy. |
| 00:05.28 | WIMPy | is digging in to much mor luxurious issues. Trying to find the LEDs of a multiport card. |
| 00:08.20 | Ionic | I created this patch one year ago, but failed to properly debug asterisk crashing... and was fed up at some point |
| 00:08.44 | Ionic | Really looking forward to buy and fiddle with that draytek card next month |
| 00:09.08 | Ionic | probably also incooporating it via chan_capi or just the misdn2 interface |
| 00:09.43 | WIMPy | I haven't tried the mISDN CAPI. |
| 00:10.03 | WIMPy | But I believe you can actually use mISDN as hardware driver for I4L via CAPI. |
| 00:10.29 | WIMPy | Might be a good way to be able to use divertctl witout having to switch drivers. |
| 00:11.53 | Ionic | oh, 1 EUR starting price (+ 6 EUR shipping) on ebay, 3 days remaining... I guess I could spare that right away, iff I'm the only buyer |
| 00:23.08 | WIMPy | Ionic: Seen my message? |
| 00:31.33 | Ionic | yeah, sorry, been afk :) |
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| 02:03.33 | cstk421 | ot sure what to make of this but im getting a call from extensions that dont exist every 33-35 minutes to ext 100 which is a physical phone. durations are consistent 0 sec 1 sec and 31 sec and then it starts over again. any ideas |
| 02:03.59 | cstk421 | sorry those are the voicemails. as for calls they are every 3 seconds |
| 02:09.28 | ChannelZ | the console should say who/what IP these are coming from. |
| 02:11.56 | ChannelZ | do you have allowguest turned on in sip.conf such that you're accepting unauthenticated calls? |
| 02:12.31 | cstk421 | is that the same as allow anonymous sip calls ? |
| 02:12.34 | cstk421 | if so yes |
| 02:13.26 | ChannelZ | ok.. so then you can't be too surprised that you're receiving these things.. |
| 02:14.09 | cstk421 | the source ip is the same as the asterisk box itself |
| 02:14.16 | cstk421 | its not coming from the outside |
| 02:14.43 | cstk421 | here is a report example |
| 02:14.44 | cstk421 | http://pastebin.com/BhsBNKda |
| 02:16.09 | ChannelZ | do you have something crazy going on in a call file? |
| 02:16.30 | ChannelZ | an external AMI script? |
| 02:17.22 | ChannelZ | it's got to be being trigger from somewhere but the CDR isn't going to help a lot. Look at the console with some verbose on and see what is actually being initiated. |
| 02:17.34 | cstk421 | not that familiar with asterisk to answer that. this setup is very basic. 2 did's 4 extensions. a call group and voicemail thats it. nothing extra or custom was added. |
| 02:17.42 | cstk421 | ok |
| 02:17.44 | cstk421 | ill try that |
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| 04:58.13 | floren | evening :) |
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| 05:03.10 | floren | i have a quick question related to MWI from a voip provider. do I have to setup voicemail locally, in order to get the messages from voip voicemail? |
| 05:07.20 | ChannelZ | no |
| 05:07.55 | ChannelZ | you presumably have to dial some extension/number at your ITSP to get into your voicemail there |
| 05:07.59 | floren | ok, well i get the headers properly: Messages-Waiting: yes |
| 05:07.59 | floren | Message-Account: sip:asterisk@67.205.74.184 |
| 05:08.00 | floren | Voice-Message: 17/0 (0/0) |
| 05:08.27 | floren | i was thinking all I have to do is add mwi_from = voipprovider |
| 05:11.31 | ChannelZ | oh you said "get messages" but you just meant indications |
| 05:11.39 | floren | ya |
| 05:11.52 | floren | the flashing bulb :) |
| 05:14.04 | ChannelZ | You'd have to do a remote subscription with an mwi => line in your sip.conf |
| 05:14.30 | floren | ya is done. i guess that is how i get the headers with 17 messages |
| 05:14.34 | ChannelZ | and then make your phone subscribe to a special mailbox on your own system |
| 05:14.36 | floren | ? |
| 05:14.45 | floren | i see |
| 05:14.54 | floren | thanks ChannelZ |
| 05:15.17 | floren | i think the cleanest way is to use my own mailbox |
| 05:15.24 | floren | much easier to configure and is local |
| 05:15.27 | ChannelZ | you set your peer to mailbox=xxxx@SIP_Remote where xxxx is whatever 'virtual' mailbox number you have in your mwi line |
| 05:16.30 | floren | i ddi not have time to look into voicemail setup but is pretty straight forward. into asterisk definite guide 4th edition they tell you everything nice and concise |
| 05:17.01 | floren | right now i'm working on a phone personal directory synced directly from google contacts, scripted in php with oauth 2 |
| 05:17.17 | ChannelZ | it takes like 30 seconds to setup voicemail in asterisk and another minute or so to write some dialplan to hook it up |
| 05:17.29 | floren | ya, you are right |
| 05:18.14 | floren | i get after a while a SIP/2.0 489 Bad event |
| 05:18.26 | floren | let say i restart asterisk service, no bad event |
| 05:18.49 | floren | after about 30min or so, it pops.. from voip.ms ip to my nat address |
| 05:19.20 | floren | is a 102 notify |
| 05:20.40 | floren | probably related to crisco phone :) |
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| 05:22.02 | ChannelZ | they might expire subscriptions or registrations, I don't know |
| 05:26.43 | ChannelZ | although if it's you sending a NOTIFY it's probably your qualify, or keepalive packets they are barfing on |
| 05:27.26 | floren | is from provider to me |
| 05:28.25 | ChannelZ | The 'bad event' |
| 05:28.47 | floren | i'm waiting for one to pop and paste it in pastebin |
| 05:33.30 | ChannelZ | I'll take a guess you're using keepalive |
| 05:33.43 | floren | ya |
| 05:33.49 | floren | 180secs |
| 05:35.08 | floren | http://codepad.org/wTmbvFHm |
| 05:35.53 | ChannelZ | oh it's the MWI notify from them to you that you're rejecting |
| 05:37.06 | floren | hmm rejecting like firewall rejecting? because i have the firewall disabled just for testing purposes |
| 05:37.46 | floren | that explains why MWI does not work |
| 05:38.29 | floren | anyways, is good that i know what is it from that 489. once i have my voicemail local, i will not have this issue anymore |
| 05:38.41 | ChannelZ | no asterisk is rejecting it |
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| 05:38.52 | floren | i'm interested on that 401, why is it sending from 1.8 to 1.8 ip |
| 05:38.57 | floren | i see |
| 05:39.59 | ChannelZ | I think you'd need to use unsolicited_mailbox or something. Not sure I've never really done remote MWI subscription. |
| 05:40.11 | floren | ya, i dont care :) |
| 05:40.16 | floren | i'll have it set proper |
| 05:40.17 | ChannelZ | and it's not going from 1.8 to 1.8, it's going to 1.9 |
| 05:40.33 | floren | Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK7cacb83a;received=192.168.1.9 |
| 05:40.33 | floren | From: <sip:101@192.168.1.8>;tag=58bc2774e85300b33556d509-120ff62c |
| 05:40.34 | floren | To: <sip:101@192.168.1.8>;tag=as25b0f2de |
| 05:40.35 | floren | ya |
| 05:41.12 | ChannelZ | <--- Transmitting (no NAT) to 192.168.1.9:5060 ---> |
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| 05:42.07 | floren | after that 401 i get the retransmission |
| 05:42.22 | floren | 10 times |
| 05:42.32 | ChannelZ | the IPs in the headers are.. well, complicated. It's just how SIP is |
| 05:42.47 | floren | i see :) |
| 05:42.58 | floren | hours of fun with asterisk |
| 05:43.02 | ChannelZ | That's your phone not responding, for whatever reason (because it's a POS) |
| 05:48.43 | floren | :D |
| 05:48.56 | floren | i still love it |
| 05:48.58 | floren | heh |
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| 09:06.54 | AliRezaTaleghani | I had install asterisk via yum (repo) |
| 09:07.11 | AliRezaTaleghani | how can I install mp3 support for moh :-? |
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| 09:49.56 | adnc | hello, I've an asterisk pbx on my lan and my peers out from time to time since I use pfSense as my router. When I reset the "States" than the peers can register again, has someone got any ideas how to track this down? I know it has nothing todo with asterisk, but maybe someone had similar problems? |
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| 11:13.08 | Ice_Strike | What port do I need to allow tftp in the firewall (iptable)? |
| 11:13.18 | Ice_Strike | Its for polycom phones |
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| 11:58.46 | WIMPy | grep tftp /etc/services |
| 12:06.35 | Ice_Strike | WIMPy Do you know how to prevent zap loading during reboot? |
| 12:11.15 | WIMPy | Don't load it. |
| 12:11.25 | WIMPy | I don;t know how/where YOU load it. |
| 12:11.42 | WIMPy | But You seem to need an upgrade. |
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| 12:29.14 | Ice_Strike | During the reboot - I get "waiting for zap to come online" |
| 12:29.20 | Ice_Strike | I need to take it off. |
| 12:29.22 | Ice_Strike | I dont have it. |
| 12:30.55 | WIMPy | You don't have what? |
| 12:31.55 | Ice_Strike | zap |
| 12:32.19 | WIMPy | Why do you have it if you don't have it? |
| 12:32.37 | Ice_Strike | I didnt install it :P |
| 12:32.52 | Ice_Strike | also no longer needed |
| 12:34.27 | WIMPy | Get rid of the ancient stuff and get yourself a fresh new Asterisk. |
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| 13:40.42 | rfmonk | ~. |
| 13:40.42 | infobot | methinks ~. is not the escape sequence you're looking for. |
| 13:41.25 | WIMPy | A thinking infobot? |
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| 17:44.51 | ChannelZ | yawns |
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| 18:32.29 | maxhbp204 | Hi, Can we put background music in ivr calls when there is no sound in case of they are entering dtmfs? i have tried with local channel, meetme as well to generate new channel and also generated with music on hold? please help me |
| 18:39.20 | ChannelZ | not sure I understand - you want music to just fade in during periods of silence while in the middle of a call? |
| 18:39.51 | ChannelZ | or you want to play something while people are deciding which digits to hit in an IVR? |
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| 18:43.10 | maxhbp204 | thanks for reply, i want background music during whole call |
| 18:43.29 | maxhbp204 | if there is silence then it will automatically play and if it is ivr then it is in background |
| 18:44.03 | maxhbp204 | if we can do in silence then also it is great too, because i can not put silence in ivr |
| 18:45.24 | [TK]D-Fender | The only way to have continuous backgrounding like that is bridging another channel. Forget the idea of having it auto-muted when people are speaking though. |
| 18:46.08 | maxhbp204 | ok thanks for reply, so if i can use G or M parameter of dial application and that another local channel will goes to meetme |
| 18:46.32 | maxhbp204 | and in meetme i can play moh to make it working, but in that case dtmf is not working |
| 18:47.57 | [TK]D-Fender | maxhbp204: that will not let it sit in the BACKGROUND |
| 18:48.26 | maxhbp204 | oh ok, then do i have to playback or something else? |
| 18:48.32 | [TK]D-Fender | no. |
| 18:48.55 | [TK]D-Fender | They channels have to be actually bridged |
| 18:49.02 | [TK]D-Fender | You cannot just throw them in a meetme |
| 18:49.06 | [TK]D-Fender | DTMF will not pass |
| 18:49.37 | maxhbp204 | ok, so what will be the best method because the incoming call will reach to IVR and then i have to also start background music as well |
| 18:50.16 | [TK]D-Fender | If you are making the IVR in Asterisk ... just use the musac AS a Background() |
| 18:50.25 | [TK]D-Fender | And your description is very bad. |
| 18:50.40 | [TK]D-Fender | You just mentioned doing something in a DIAL and then started mention of sitting in a IVR... which is NOT a call |
| 18:51.21 | maxhbp204 | i am sorry for bad description, i will try with background application |
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| 20:28.19 | boichev_ | can someone help me configure an OpenVox B800P card on asterisk 11.4.0 ? I installed libpri and dahdi it detected the card and dahdi_genconf generated the file /etc/asterisk/dahdi-channels.conf as in the manuals (everything is ok inside)... but in asterisk console "dahdi show channels" shows only the psedo channel...... |
| 20:36.41 | boichev_ | got it :) |
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| 22:31.45 | juhas | Good evening :-) I'm trying to debug a asterisk 1.8.10.1~dfsg-1ubuntu1 where Queue members (SIP/blabla) get stuck in a "ringing" state and can't accept more calls |
| 22:33.46 | juhas | Any suggestions on what could be causing this? This seems to be happening reliably on every call (ie, the queue member gets the first call and no more calls) |
| 22:34.12 | WIMPy | the only thing I can give you is~upgrade asterisk |
| 22:34.21 | WIMPy | oops |
| 22:34.26 | WIMPy | ~upgrade asterisk |
| 22:34.27 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 22:37.21 | juhas | Well, I'd like to avoid upgrading (I'm running ubuntu 12.04's default package version) unless this is a known bug. Tried to google around for it, but so far no luck. |
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