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| 01:01.50 | lvlinux | how do i setup BLF for parked calls? |
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| 01:17.29 | mjordan | lvlinux: features.conf, set parkinghints=yes |
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| 01:21.08 | lvlinux | ah ok thanks |
| 01:23.00 | ipengineer | I noticed that the voicemail table has a field for email. Is there a mechanism for sending voicemails as attachments? |
| 01:23.22 | lvlinux | yes you put attach=yes in the options |
| 01:24.24 | lvlinux | 100 => 1234,Bob Smith,mail@domain.com,attach=yes|tz=pacific|maxmsg=100 |
| 01:24.34 | lvlinux | so your voicemail line would look something like this ^^^^^^^ |
| 01:25.41 | ipengineer | Ahh.. Nice thanks |
| 01:25.45 | ipengineer | Can I specify a template? |
| 01:26.12 | ipengineer | Do I have to put the email in the dialplan or can it pick it up from voicemail table? |
| 01:27.30 | lvlinux | everything it needs comes from the voicemail.conf file |
| 01:27.44 | lvlinux | and yes u can specify a template |
| 01:27.48 | ipengineer | Ok I will take a look. Thanks |
| 01:28.01 | ipengineer | I saw some scripts and was hoping that wouldnt have to be the case |
| 01:28.05 | lvlinux | look at the voicemail.conf sample config file---it has the template in there already that you can modify |
| 01:28.12 | ipengineer | kk |
| 01:28.59 | lvlinux | no you shouldn't need any kind of scripts at all---just set your voicemail.conf file with the template, add attach=yes in it, and make your voicemail box lines like what I showed you above and it should work great. |
| 01:32.04 | lvlinux | mjordan: what do I have the phone subscribe to then? |
| 01:33.41 | ipengineer | lvlinux: If I am sending calls to voicemail like this what would be the best way to do that? The voicemail.conf makes sense to me. |
| 01:33.41 | mjordan | lvlinux: off the top of my head, can't remember. 'core show hints' should tell you |
| 01:33.42 | ipengineer | exten=>s,1,Voicemail(201@context,u) |
| 01:33.59 | lvlinux | mjordan: oh ok thanks |
| 01:34.35 | lvlinux | ipengineer: yes? |
| 01:34.37 | mjordan | lvlinux: let me know if you don't see it in the list and I'll spin up an instance and take a peek |
| 01:35.39 | ipengineer | lvlinux: nevermind. Too late for me. If the voicemail is configured in the voicemail table it will send it. |
| 01:36.02 | lvlinux | ipengineer: yep |
| 01:37.12 | lvlinux | mjordan: ok i see all of them as 701@parkedcalls, 702@parkedcalls etc.... |
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| 01:48.04 | lvlinux | when i write a dialplan macro, does it need to be put above wherever it's used in the dialplan, like a variable? |
| 01:48.15 | lvlinux | or does it matter? |
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| 01:52.50 | KavanS | I'm trying to get a tdm400p card to answer w/out any delay when it detects an incoming call. right now, the best I've been able to do is adjust chan_dahdi.conf to immediate=yes, and that makes it so it rings SIP phone on 2nd ring of regular POTS telephone plugged into POTS line splitter (other side is TDM400p) |
| 01:53.00 | KavanS | how do I reduce the answer delay? |
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| 01:54.44 | lvlinux | do you have callerid enabled? |
| 01:54.52 | lvlinux | or rather do you need it enabled? |
| 01:55.47 | lvlinux | callerid is sent between the first and second rings here in the US, so if it's enabled, then the TDM400 is waiting to get the callerid info before answering the line. |
| 01:56.46 | KavanS | yep, in USA. |
| 01:56.47 | lvlinux | if you set usecallerid=no it will answer immediately |
| 01:57.03 | KavanS | anyway to get callerID later on in the process? (stupid question) |
| 01:57.08 | lvlinux | but of course you lose callerid |
| 01:57.09 | lvlinux | nope |
| 01:57.20 | lvlinux | telephone company won't send it if the line is off hook |
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| 01:58.03 | KavanS | damn... |
| 01:58.41 | lvlinux | why do you need it to answer asap? |
| 01:59.46 | KavanS | just trying to test * limitations |
| 02:00.04 | KavanS | guess it's really a limitation of the whole standard |
| 02:00.09 | lvlinux | hehe well it's not an * limitation---more of a PSTN limitation |
| 02:00.10 | lvlinux | yes |
| 02:00.19 | KavanS | yeah I'm def. a noob in that regard |
| 02:00.23 | KavanS | only know enough to get in trouble |
| 02:01.22 | lvlinux | I think w a T1 line it sends callerid immediately w the call, but on an analog line, you have to send it after one ring so a phone/device will be ready and listening for it. |
| 02:02.34 | KavanS | what about SIP? |
| 02:03.15 | mjordan | lvlinux: in general, I'd recommend not using macros. In sufficiently complex dialplans, they cause problems. The preferred mechanism are subroutines (GoSub) |
| 02:05.20 | lvlinux | really? i didn't know that. It's a simple dialplan. I'll have to look into doing stuff w gosub instead. |
| 02:05.41 | lvlinux | but in any case does it matter where it's actually located in the dialplan? |
| 02:09.36 | lvlinux | a macro i mean |
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| 02:18.11 | lvlinux | hey slav3_kitten! did u ever get ur audio problem straightened out? |
| 02:18.41 | lvlinux | was it ur ISP's fault? |
| 02:18.59 | slav3_kitten | i uh |
| 02:19.13 | slav3_kitten | it works when the ISP isn't sticking their dick in the rack and waving it about |
| 02:19.39 | slav3_kitten | if that helps answer your question |
| 02:20.05 | lvlinux | yep if i remember correctly that's what u suspected before |
| 02:20.09 | slav3_kitten | yep |
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| 02:20.25 | slav3_kitten | they regularly stick their dick in things though |
| 02:21.09 | slav3_kitten | and despite my contract saying they need to inform me of maint windows.... totally the rand() maint window game all last week |
| 02:21.33 | lvlinux | lol |
| 02:22.47 | slav3_kitten | yea... i called up after the 8th short outage like "what the fuck is up with this" and got "oh we are upgrading equipment here in the home office" |
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| 02:28.57 | ruben23 | hi guys any one have idea how to setup voicemail on asterisk will be automatically be emailed somehow..to recepient emails..please help |
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| 02:36.05 | [TK]D-Fender | ruben23: it's all in the sample voicemail.conf |
| 02:36.48 | [TK]D-Fender | ruben23: Fill in the address. have an MTA setup for the standard "sendmail" shell script. The end. |
| 02:37.50 | [TK]D-Fender | slav3_kitten: https://www.youtube.com/watch?v=diYS8jyOcFc |
| 02:38.38 | slav3_kitten | i'll remember this [TK]D-Fender |
| 02:38.59 | [TK]D-Fender | slav3_kitten: You couldn't afford the therapy it's cost to forget ;) |
| 02:39.04 | [TK]D-Fender | it'd* |
| 02:39.21 | slav3_kitten | lol |
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| 04:09.49 | neeby_goosey | Hello. |
| 04:10.42 | Penguin | katty: Oh yeah. We only had gobs of rain yesterday and only strong wind today. |
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| 04:37.32 | neeby_goosey | Could anyone help me with clarifying the difference between host, externhost, permit? |
| 04:56.54 | hebber | in sip.conf? |
| 04:58.17 | neeby_goosey | Yes. |
| 04:59.34 | hebber | externhost is used if you need to tell asterisk what external IP if its placed behind NAT and need to communicate on public IP |
| 05:00.23 | hebber | host and permit is used to define peers |
| 05:02.50 | hebber | For host and permit check in sip.conf under DEVICE CONFIGURATION |
| 05:03.38 | hebber | for externhost check in sip.conf under NAT SUPPORT |
| 05:03.45 | hebber | its all there :) |
| 05:07.08 | r00f | hi there |
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| 05:20.52 | neeby_goosey | Checked those, but couldn't find what RTP, SIP and SDP fields are affected by 'externhost' option. :( And I don't quite get, if I want to tell Asterisk that a local ATA will always stay at 192.168.1.2 and only there, which one should I add - 'host=192.168.1.2' or 'host=192.168.1.2 deny=0.0.0.0/0.0.0.0 permit=192.168.1.2'. |
| 05:21.42 | r00f | host alone should be enough. deny/permit are used for host=dynamic mostly |
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| 05:27.44 | hebber | externhost is for asterisk - host and permit is for devices |
| 05:29.52 | neeby_goosey | So, in the end, 'host' is where the connected device is, 'deny/permit' controls the 'host=dynamic' range, and 'externhost' is something that will be injected into SIP and RTP packets so that the connected device could see Asterisk behind NAT, right? |
| 05:32.24 | hebber | yes, great summary |
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| 05:37.55 | hebber | Application RetryDial: the voice file exists, but there is no sound from the file or ringtone - just dead sound. |
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| 05:43.41 | hebber | I think I will revert to Dial instead, can't get RetryDial to work as intended |
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| 05:59.31 | neeby_goosey | Argh. Configuring Asterisk behind NAT is tough as hell. |
| 06:00.41 | hebber | Yes, but not to many parameters needed hang on |
| 06:04.07 | hebber | on EC2 I used: localnet, externaddr, nat and media_address |
| 06:04.47 | hebber | also read the NAT SUPPORT section in sip.conf |
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| 06:20.11 | neeby_goosey | Oh, thanks, seems like I missed the "media_address" option - somehow I was sure it was included in 'externaddr'. |
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| 07:07.36 | hebber | neeby_goosey: I figured externaddr is for SIP and media_address is for the SDP |
| 07:08.26 | neeby_goosey | Woah 0_0 and what about RTP? |
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| 07:17.01 | hebber | Isn't SDP to establish RTP stream? |
| 07:18.33 | hebber | if not replace SDP with RTP |
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| 07:28.28 | neeby_goosey | Oh. |
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| 07:57.51 | neeby_goosey | By the way, anyone here ever worked with HyberTone GoIP series? I have a fully configured and working GoIP4, but still one thing bothers me - in Line Mode each channel must have a Routing Prefix. I have contacted Hypertone support, trying to explain this is an overkill feature, but they are chinese... and... erm... we didn't have a productive conversation. So I want to hear from other users,... |
| 07:57.53 | neeby_goosey | ...what they think about this feature. |
| 07:59.41 | r00f | i have experience talking to chinese support, at Draytek. it's not easy to explain them why their features are not features ) |
| 08:03.13 | neeby_goosey | I even tried using Google translate, trying to make my statements more convincing by converting them to chinese. It worked a little - responses to my emails were still in english, but at least contained more than "yes, sir" or "why? no why"... |
| 08:04.25 | neeby_goosey | And still the support continued feeding me with "reed the manual and do like that without any questioning". |
| 08:05.12 | neeby_goosey | So, any GoIP users here? |
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| 08:36.09 | flashdel | hi @all! I got a fritzbox 6360 and i am trying to register my asterisk (11.9) as sip client to it. It works for two sip clients for 1-2 hours, but after that time i get a timeout :-( If i register five numbers or more, i am getting this timeout faster... Can somebody help me please? Here is a trace and my sip.conf: http://pastebin.de/124255 |
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| 09:29.26 | flashdel | is there anybody who has got a hint for me? :S |
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| 11:48.09 | mjt | hello. What you guys recommend as a softphone for windows? I tried 3cx but that one appears to require their own PBX, does not work with asterisk anymore. Tried linphone (several versions), but it just crashes on first startup so does not work at all. Something else? |
| 11:49.50 | r00f | i like bria or eyebeam as a more lightweight version |
| 11:50.06 | r00f | or even xlite if you need it free |
| 11:50.25 | mjt | i also tried zoiper, but can't even download it (the d/load link doesn't work) :) |
| 11:50.35 | mjt | xlite |
| 11:51.00 | r00f | xlite doesnt support g729 if i am correct |
| 11:53.44 | mjt | is it important? |
| 11:53.53 | mjt | i dunno really what g729 is |
| 11:54.09 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 11:54.16 | r00f | low-bandwidth codek |
| 11:55.21 | mjt | heh. xlite doesn't start |
| 11:55.32 | mjt | that's just _lovely_ :) |
| 11:57.53 | mjt | so far, there's just one softphone which _starts_ (but is useless): it is 3cx. All the rest fails to _start_. |
| 11:58.03 | mjt | it is a fresh install of windows7 |
| 11:58.16 | r00f | maybe all get blocked by firewall |
| 11:58.27 | r00f | log in with admin account and it will offer to unblock |
| 11:58.32 | mjt | it's an interesting firewall which prevents applications from _starting_ |
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| 11:58.48 | mjt | (local firewall is turned off) |
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| 11:58.57 | r00f | lolz |
| 11:58.58 | r00f | isk |
| 11:59.07 | r00f | maybe av, or.... |
| 11:59.14 | mjt | no av, just fresh win |
| 11:59.16 | r00f | windows ) |
| 11:59.27 | mjt | the same is on another pc |
| 11:59.41 | mjt | with diffferent windows (32 vs 64bits) |
| 11:59.43 | r00f | i've used xlite and eyebeam for more than 5 years, never had problem |
| 12:00.00 | r00f | right now i'm writing from win 8 pro |
| 12:00.07 | r00f | and have bria up and running |
| 12:00.12 | mjt | ok. it looks like i know what's going on |
| 12:00.36 | mjt | lemme try to run one of them using local console, not using rdp |
| 12:01.20 | mjt | ..and all of them works |
| 12:01.28 | r00f | yeah if it has no audio devices to detect it will not work |
| 12:01.31 | r00f | (i guess) |
| 12:01.34 | mjt | it has audio |
| 12:01.48 | r00f | then why it hates rdp? strange |
| 12:01.56 | mjt | lack of audio is not a reason to crash at startup anyway |
| 12:02.19 | r00f | has no ideas |
| 12:02.24 | As001 | Hello is there an option in Queue application which will tell Asterisk 1.8 to go on next extension in dialplan after client from Queue hangs up call to queue agent not by * but by hangup of his phone ? I found out c option but but with that turned on I couldn't see that call goes to next extension after client hangup. |
| 12:02.31 | mjt | well, that's programming errors, aka bugs. in the mentioned software. |
| 12:04.46 | r00f | i've found only one support ticket for bria in rdp, but it's about citrix and it is not answered |
| 12:04.48 | r00f | 8) |
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| 12:08.45 | jbassett_ | Hello folks, having problems with DTMF being passed. Softphone client PhonerLite is working but other softphones such as Bria are not. Using rfc2833 across the board. Any similar experiences? Never had this issue before. |
| 12:12.41 | r00f | jbassett_ never had problems with bria |
| 12:13.23 | r00f | try to change to INFO and test |
| 12:14.54 | jbassett_ | I have not had problem with Bria in past, works fine on my other box, |
| 12:15.38 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 12:15.52 | r00f | google finds one similar problem, changing to INFO worked for them. you could just try |
| 12:18.41 | jbassett_ | tried that and inband but no luck |
| 12:19.23 | jbassett_ | using ISDN line for calls |
| 12:19.57 | r00f | does asterisk see the dtmf from phone? |
| 12:20.16 | [TK]D-Fender | [08:02]As001Hello is there an option in Queue application which will tell Asterisk 1.8 to go on next extension in dialplan after client from Queue hangs up call to queue agent not by * but by hangup of his phone ? I found out c option but but with that turned on I couldn't see that call goes to next extension after client hangup. <- the call is on the client's side. There is no continuing... |
| 12:20.18 | [TK]D-Fender | ...for the agent normally. |
| 12:21.35 | jbassett_ | r00f: /var/log/asterisk/full shows dtmf from the working client but not any other |
| 12:21.59 | jbassett_ | all clients set to use the same moderfc2833 |
| 12:23.06 | jbassett_ | codecs in use are alaw/ulaw |
| 12:23.23 | As001 | Ok thanks [TK]D-Fender |
| 12:23.44 | r00f | maybe ask overminds. i don't know what else could you try |
| 12:23.48 | [TK]D-Fender | As001: if you DIAL them yourself you could try that dial's option for it |
| 12:25.00 | As001 | At the matter of fact I want to achieve that when client hangup in Queue I emit some sound at agent who talked to that client like beep so he knows call has been hangup. |
| 12:25.51 | r00f | maybe beep it at h extension? but i dislike doing stuff with h |
| 12:26.27 | As001 | I want beep on channel like SIP/238-ancdege if agent is 238 and he is LoggedIn he has channel SIP/238-something. |
| 12:27.33 | jbassett_ | r00f: overminds? I assuming that is an irc user? |
| 12:27.56 | As001 | I think when call comes channel Agent/sip has been created but when any side hangup that channel is gone, only SIP/sip_number exist and I would like to beep on that channel. |
| 12:28.32 | r00f | jbassett_: i mean more experienced residents of this cavern |
| 12:28.42 | jbassett_ | aha, cheers |
| 12:30.23 | [TK]D-Fender | [08:24]As001At the matter of fact I want to achieve that when client hangup in Queue I emit some sound at agent who talked to that client like beep so he knows call has been hangup. <- the call ENDS. is that not enough of a sign? |
| 12:33.31 | mjordan | jbassett_: if the mode is RFC 2833, the DTMF should be arriving via RTP. You should be able to capture the DTMF sent from Bria either via 'rtp set debug on', or by getting a pcap |
| 12:33.50 | As001 | not for my agents I need to playback something to them so they know client side hangup. |
| 12:34.09 | r00f | [TK]D-Fender: agent could fall asleep with boring client, so we need to wake him up with a sound |
| 12:34.42 | jbassett_ | mjordan: Yes has set rtp debug on on and can see DTMF from the PhonerLite client but nothing from Bria or other phones |
| 12:34.42 | As001 | because when they start to talk like on horserace for 4-5 minutes and realise noone is listening they think it is some system error but I think client hang up. :) |
| 12:35.01 | As001 | I am sure client hangup |
| 12:35.03 | eirirs | silent hangup |
| 12:35.16 | [TK]D-Fender | As001: They can't look at their phone and see the call ended? |
| 12:35.26 | [TK]D-Fender | As001: You need better employees |
| 12:35.54 | [TK]D-Fender | As001: Or crappier phones like Linksys that give a reorder tone when a call terminates |
| 12:35.59 | mjordan | jbassett_: if you don't see it, it's because Bria didn't send it. |
| 12:36.31 | As001 | They are logged in and can see xlite and minutes and seconds they are logged in. |
| 12:36.39 | mjordan | jabassett_: The RTP Debug shows the RTP packets as they were received from the socket, with extremely minimal decoding. If you don't see DTMF at that level, we didn't get it. |
| 12:37.11 | [TK]D-Fender | As001: Have them use their eyes, or us a local channel to dial the agent and use the Dial() option to do your dirty-work |
| 12:37.46 | As001 | i can either do in java script rotation to signalise hangup state in browser or playback something to them. |
| 12:38.18 | As001 | So it's not possible to playback something on SIP/sipnumber channel who is logged in ? |
| 12:38.35 | r00f | installed 14 version today, it's great |
| 12:38.43 | r00f | sry wrong window |
| 12:39.06 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 12:40.26 | [TK]D-Fender | As001: What doe "logged in" actually mean in your case? |
| 12:40.28 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 12:42.58 | As001 | They log in and wait for calls and calls come to queue one by one with beep before every call. It means they did AgentLogin on their sip they logged in in Queue. |
| 12:43.27 | [TK]D-Fender | As001: Are they literally sitting in that queue app ON CALL wiaintg? |
| 12:43.30 | [TK]D-Fender | waiting* |
| 12:43.52 | As001 | they wait for calls in queue yes. |
| 12:46.41 | [TK]D-Fender | As001: Ok, then I don't know of any options for you at all if they don't get standard calls like almosts everyone uses app_queue like |
| 12:48.24 | As001 | I tried to catch up agentlogin channel like SIP/sip_number and to playdtmf via manager interface but without success. |
| 12:49.31 | As001 | ok if it's impossible to playback or playdtmf at least after client hangup/call ends then I will try to signalise them hangup on web browser via java script or something. |
| 12:49.52 | As001 | thanks for help |
| 12:49.58 | *** part/#asterisk As001 (~uros@82.117.198.142) |
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| 13:11.43 | *** join/#asterisk neeby_goosey (~chatzilla@62.141.95.18) |
| 13:11.49 | neeby_goosey | Hello. |
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| 13:14.52 | r00f | neeby_goosey hi |
| 13:31.04 | neeby_goosey | Could anyone please help me with troubleshooting the connection of a behind-the-NAT-Asterisk to a VoIP provider? |
| 13:31.40 | neeby_goosey | `sip set debug peer mcn-out` logs here: http://pastebin.com/nShP1uy3 |
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| 13:41.02 | neeby_goosey | With some details: http://pastebin.com/jEAySwHV |
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| 14:37.11 | M4dH4TT3r | hello |
| 14:37.22 | r00f | hi |
| 14:37.42 | M4dH4TT3r | does anyone know how to setup asterisk to use google voice |
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| 14:38.41 | M4dH4TT3r | including as the sip gate |
| 14:38.55 | M4dH4TT3r | dont I know you yano? |
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| 14:40.45 | M4dH4TT3r | howdy r00f |
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| 14:50.20 | malcolmd | setting up asterisk to use google voice, at this time, wouldn't be a very fruitful endeavor, what with google shutting down things on may 15, 2014 |
| 14:50.23 | [TK]D-Fender | http://www.androidpolice.com/2013/11/04/merging-of-google-voice-and-hangouts-will-result-shutting-down-all-3rd-party-voice-apps-in-may-2014/ |
| 14:50.24 | [TK]D-Fender | ^^^^^^^^^ |
| 14:50.28 | [TK]D-Fender | GV = DEAD |
| 14:51.55 | r00f | guys, any experience on befriending TDM410P 4FXO and Panasonic TSE824 ? |
| 14:52.16 | r00f | for some reason panasonic does not give me the callerid |
| 14:52.20 | M4dH4TT3r | :(:( |
| 14:53.02 | M4dH4TT3r | any other alternative thats not shutting down? (sip gate has been shutdown a while now) |
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| 14:53.37 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:54.46 | [TK]D-Fender | M4dH4TT3r: Any of the thousands of other providers out there |
| 14:56.34 | M4dH4TT3r | free |
| 14:57.38 | [TK]D-Fender | Free = going away everywhere |
| 14:58.02 | [TK]D-Fender | Somebody is paying for it... and the market isn't there for it not to be you. |
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| 15:01.36 | *** join/#asterisk iTrojan (~Adium@41.233.94.224) |
| 15:03.13 | litn | I'm trying to make it so that when the user dials *76 it also changes the device state to busy. In the examples I've seen of changing device states, it has the device name hardcoded into the config. How do I make it set the state to busy for the user calling *76? |
| 15:03.22 | litn | or make the device state unavailable or anything else |
| 15:03.39 | *** join/#asterisk navaismo (~navaismo@200-52-45-221.dynamic.axtel.net) |
| 15:04.16 | [TK]D-Fender | Use a custom device state |
| 15:04.32 | [TK]D-Fender | You do not ever set the state of an actual device. |
| 15:04.37 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:04.38 | [TK]D-Fender | that is read-only |
| 15:06.30 | litn | [TK]D-Fender: Set(DEVICE_STATE(Custom:lamp1)=BUSY |
| 15:06.37 | litn | in this example, isn't lamp1 the "device" ? |
| 15:06.47 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 15:07.18 | [TK]D-Fender | litn: that is a custom device, and not a literal ie: "sip/fred" |
| 15:07.28 | [TK]D-Fender | litn: So use those for your state |
| 15:09.36 | litn | I'm still not understanding, maybe if I explain what I am trying to do- I have a light on my phones that I can set to BLF. I am attempting to make it so that when they dial *76, the button for DND, it will reflect in the BLF light that they are busy |
| 15:10.03 | [TK]D-Fender | litn: So go set a device specific for that user |
| 15:10.33 | *** join/#asterisk theron (~theron@66.220.144.81) |
| 15:11.33 | litn | right but if I want *76 to work for everyone, how would I dynamically set it for their device? |
| 15:12.16 | [TK]D-Fender | Pick a name specifc to each |
| 15:14.53 | litn | in the extensions though when they dial *76, how would I define their device name there, so that *76 works for everyone? Would I check what their extension is or is there a way to get the device name for that given extension (hint right?), or something like that? |
| 15:17.04 | [TK]D-Fender | Yes |
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| 15:30.17 | mirela666 | Hi, has anyone encountered with VoiceMail unavailable message braking or loosing (like you hear 3s and then silence, sometimes whole msg, sometimes, half...) |
| 15:30.35 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 15:30.54 | loggiew | guys, my users are complaining that when they transfer calls to another phone there is about a 3 - 5 second delay before the new phone rings. Is this something I can adjust or a result of normal delays associated with SIP? |
| 15:31.36 | [TK]D-Fender | loggiew: Probably not SIP... but possibly networking |
| 15:31.44 | [TK]D-Fender | Because mine are instant on LAN |
| 15:32.09 | loggiew | Ok. So possibly something more along the lines of replacing the 10/100 switch with a 100/1000? |
| 15:32.34 | [TK]D-Fender | No |
| 15:32.54 | loggiew | Hm. |
| 15:33.00 | [TK]D-Fender | You should probably be looking at your calls a lot closer |
| 15:33.16 | [TK]D-Fender | How are you transferring the call? |
| 15:34.32 | loggiew | Erm. Im about to sound like a retard I know it. I just started working with this system for them. There is just a transfer button on the SIP phones. |
| 15:34.39 | loggiew | Hit the button, type the extension |
| 15:36.26 | [TK]D-Fender | The it's probably the phones LOCAL DIALPLAN that is creating the pause as it's waiting for more possible digits |
| 15:37.23 | loggiew | That's logical. Thanks for the suggestion |
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| 15:40.06 | loggiew | [TK]D-Fender, perfect. I think you are right. |
| 15:40.14 | loggiew | Thanks dude. |
| 15:41.40 | [TK]D-Fender | You're welcome |
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| 15:55.44 | litn | [TK]D-Fender: ok I think I am closer, still no go though- do you see what I am doing wrong? Here is the show core hints for the device, |
| 15:55.54 | litn | 645@ext-local : SIP/645 State:Idle Watchers 0 |
| 15:56.04 | litn | in the extentions for *78, I have this, |
| 15:56.14 | litn | exten => *78,n,Set(DEVICE_STATE(SIP/${AMPUSER})=NOT_INUSE) |
| 15:56.20 | brendan` | anyone able to recommend a good resource for learning asterisk? |
| 15:56.33 | litn | still no change in the status when I do core show hints though |
| 15:56.35 | brendan` | i saw on amazon there's the asterisk 'the definitive guide' but not sure about ppl's opinion on it |
| 15:57.53 | paulc | brendan`: It's a great book.. lots of pointers in the right direction to help you figure stuff out.. well written, easily read.. |
| 15:58.29 | navaismo | ~book |
| 15:58.29 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:59.30 | brendan` | paulc: awesome |
| 15:59.31 | brendan` | ty |
| 15:59.44 | paulc | no worries :) |
| 15:59.47 | brendan` | ~buybook |
| 15:59.47 | infobot | You can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/0636920025894 so go buy it SERIOUSLY |
| 16:07.42 | litn | [TK]D-Fender: also, in the logs, it shows the variable is correct, |
| 16:07.42 | litn | <PROTECTED> |
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| 16:13.18 | litn | [TK]D-Fender: so I tried doing devstate change from cli, and it complained that I can only change states on Custom: devices. I think I am closer? |
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| 16:18.16 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 16:18.20 | file | you can't arbitrarily change channel driver device states because the channel driver determines that |
| 16:23.38 | *** join/#asterisk onixx (1000@bas1-stetherese38-2925306377.dsl.bell.ca) |
| 16:24.30 | onixx | hello all. I have tried restricting udpbindaddr in sip.conf to only be on my local lan while tlsbindaddr to be on 0.0.0.0 |
| 16:25.29 | onixx | it is causing issues and I end up getting one way audio. as soon as I re-enable udpbindaddr to 0.0.0.0 it works again |
| 16:26.40 | onixx | transport is set to tls and encryption =yes for the peer |
| 16:27.08 | onixx | looks like it still needs udp somewhat ? |
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| 16:44.27 | navaismo | litn, did you checked this-->http://www.freepbx.org/forum/freepbx/users/blf-with-dnd ? |
| 16:45.25 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:49.08 | lodac | $3k to setup a ANI/ALI database for 911 for two numbers on a PRI. We should be able to send our own information.. of course, I suppose that could be abused and they are trying to prevent that... $3k though. |
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| 18:09.47 | pabelanger | anybody have dialplan to extract remote-party-id from sip header and setup callerid? |
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| 18:13.11 | pabelanger | sorry, P-Asserted-Identity |
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| 18:31.25 | navaismo | nope not me |
| 18:52.00 | puzzled | pabelanger: http://marc.info/?l=asterisk-users&m=123566332905584 |
| 18:59.09 | *** join/#asterisk Scott0_ (~Scott0_@unaffiliated/scotto/x-4000254) |
| 18:59.25 | Scott0_ | I wish switchvox didn't require subscription |
| 18:59.35 | Scott0_ | is asterisk my best option? |
| 18:59.47 | Scott0_ | getting tired of ringcentral |
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| 19:04.54 | Scott0_ | paying a subscriptiont per extension is crazy talk |
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| 19:11.24 | [TK]D-Fender | Scott0_: Depends what you want. |
| 19:11.52 | [TK]D-Fender | Scott0_: Remember that Asterisk is just a telephony toolikt.. it is what you make of it. That's what Switchvox is.... a front end over asterisk |
| 19:12.03 | Scott0_ | I need a phone system where I don't have to pay monthly fees that are outrageous |
| 19:12.20 | Scott0_ | I don't like paying for hardware as a monthly cost |
| 19:12.23 | Scott0_ | that's silly |
| 19:12.36 | Scott0_ | that's like some cisco devices where you pay a subscription |
| 19:12.48 | Scott0_ | if I own the hardware, I shouldn't be having to pay a monthly |
| 19:13.05 | Scott0_ | especially if I buy the $6000 switchvox |
| 19:13.11 | Scott0_ | that's a lot of money |
| 19:13.30 | Scott0_ | im stuck on ringcentral if there's no better option |
| 19:13.39 | Scott0_ | its too cheap |
| 19:13.52 | Scott0_ | im just exploring alternatives |
| 19:13.59 | [TK]D-Fender | FreePBX <--- |
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| 19:14.15 | Scott0_ | because ringcentral does not support multiple auto receptionists on a single account |
| 19:14.19 | [TK]D-Fender | they make a web admin frontend for Asterisk as well as supplying ready to use ISO's |
| 19:14.27 | Scott0_ | I have 2 companies im trying to serve under one roof |
| 19:14.39 | [TK]D-Fender | Multi-tennent is it's own problem |
| 19:14.39 | Penguin | [TK]D-Fender: its |
| 19:14.48 | Scott0_ | not multi tennant |
| 19:14.48 | [TK]D-Fender | And there are almost only commercial GUI's for it |
| 19:14.57 | Scott0_ | this is a conglomerate company |
| 19:15.02 | [TK]D-Fender | Could be depending on how you run things |
| 19:15.06 | Scott0_ | so im trying to share resources |
| 19:15.16 | [TK]D-Fender | You want them having the same #'s on both sides? |
| 19:15.25 | [TK]D-Fender | A has a 100 and so does B? |
| 19:15.30 | Scott0_ | different numbers but still beaing able to trasnfer in between |
| 19:15.32 | [TK]D-Fender | then that is definiately multi-tennent |
| 19:15.52 | Scott0_ | we are all in the same office |
| 19:15.53 | [TK]D-Fender | If you just want separate menu's, the that could be done with FreePBX |
| 19:16.11 | [TK]D-Fender | it's a question of logistical separation. |
| 19:16.21 | Scott0_ | different incoming numbers |
| 19:16.46 | Scott0_ | but it would be nice to have a users extension work from both companies |
| 19:16.55 | Scott0_ | and just server different menus and greetings |
| 19:19.26 | *** join/#asterisk roentgen (~irc@openvpn/community/support/roentgen) |
| 19:20.18 | [TK]D-Fender | Doable |
| 19:20.20 | [TK]D-Fender | Go try it |
| 19:20.27 | Scott0_ | with freepbx? |
| 19:20.29 | [TK]D-Fender | www.freepbx.org |
| 19:20.33 | Scott0_ | cool |
| 19:20.46 | Scott0_ | and I can use my existing IP phones |
| 19:20.47 | Scott0_ | ? |
| 19:20.50 | [TK]D-Fender | yes. With Asterisk itself you can do whatever you want.... but you also have to configure it all.... |
| 19:21.08 | [TK]D-Fender | And I'm suspecting you might not have much of a clue as to quite what would be involved in that... |
| 19:21.14 | Scott0_ | I do] |
| 19:21.30 | Scott0_ | I've dealt with some phone system linux stuff before |
| 19:21.40 | Scott0_ | I thnik I had a fax server that was linux |
| 19:21.43 | Scott0_ | its been some years |
| 19:22.21 | Scott0_ | ah avantfax |
| 19:22.49 | Scott0_ | will freepbx handle faxes and forwarding to mobile devices? |
| 19:25.41 | *** join/#asterisk bluOxigen (~a@unaffiliated/bluOxigen) |
| 19:28.29 | [TK]D-Fender | It can receive fax and then e-mail |
| 19:29.45 | Scott0_ | yep fax email |
| 19:30.08 | Scott0_ | can it send faxes too or will I have to setup a analog converter to use a local fax machine for outgoing? |
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| 19:39.45 | onixx | Hello, I have recently upgraded to asterisk 12 and noticed the following error message when a call is established: [2014-04-29 15:38:51] WARNING[8476][C-0000003e]: features_config.c:1301 ast_get_chan_applicationmap: Unknown DYNAMIC_FEATURES item 'automon' on channel SIP/2216-00000069. |
| 19:40.58 | lvlinux | How do i have a phone subscribe to a mailbox with BLF? Or rather how do I get * to set a hint for a voicemail box? |
| 19:41.18 | [TK]D-Fender | Scott0_: there are addon's for sending, some paid, but cheap |
| 19:41.29 | [TK]D-Fender | Scott0_: Or run hardware interfaces |
| 19:42.08 | lvlinux | onixx: sounds like you have something configured in your dialplan for automon but not in features.conf |
| 19:42.10 | onixx | the error message appears 3 times for some reasons. *1 to reocrd works as expected |
| 19:42.15 | [TK]D-Fender | onixx: Like it says..... there is no feature with that name. Go look at your features.conf |
| 19:43.10 | onixx | in extensions.conf I have DYNAMIC_FEATURES=>automon . is it still required ? |
| 19:43.56 | [TK]D-Fender | do you have an entry NAMED "automon"? |
| 19:44.48 | onixx | in features.conf, all I have under [featuremap] |
| 19:44.49 | onixx | [featuremap] |
| 19:44.59 | onixx | is automon => *1 |
| 19:45.28 | [TK]D-Fender | I'd restart * |
| 19:46.52 | *** part/#asterisk lodac (~lodac@unaffiliated/lodac) |
| 19:47.22 | onixx | [TK]D-Fender: did that a few times, still the same |
| 19:47.53 | [TK]D-Fender | pastebin actual configs and your actual call |
| 19:48.20 | [TK]D-Fender | ~pb |
| 19:48.20 | 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. |
| 19:48.24 | [TK]D-Fender | ^^^^^^^^^^^ |
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| 19:55.50 | Katty | hi kids. |
| 19:55.52 | lvlinux | How do i set a subscribe hint for a mailbox? I have a phone that needs to monitor two boxes, one with it's builtin MWI, the other with BLF. |
| 19:55.58 | lvlinux | hi Katty |
| 19:56.29 | onixx | [TK]D-Fender:http://pastebin.com/zbhUeUrX |
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| 19:58.22 | Katty | lvlinux: howdy. |
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| 20:11.11 | Alex_Bkash | hi |
| 20:13.33 | lvlinux | hi |
| 20:14.03 | navaismo | hi |
| 20:15.08 | c|oneman | hi |
| 20:15.11 | Alex_Bkash | :D |
| 20:16.08 | Katty | hi |
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| 21:06.27 | t4nk793 | quick question for anyone with a free moment. I am looking to create a hunt ring group without using the gui, any pointers or terms to look up as i havent had any luck as of yet |
| 21:07.06 | *** part/#asterisk Scott0_ (~Scott0_@unaffiliated/scotto/x-4000254) |
| 21:09.33 | lvlinux | t4nk793: define exactly what you mean by a "hunt ring group" |
| 21:09.57 | lvlinux | t4nk793: each phone rings 4 times, then next one, or only one phone rings per call? |
| 21:10.38 | lvlinux | t4nk793: then the next call another phone rings? |
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| 21:13.23 | t4nk793 | sorry user came to bothe rme |
| 21:13.29 | t4nk793 | the first one is what i am looking for |
| 21:13.48 | t4nk793 | each phone rings a few times before moving on to the next number |
| 21:14.30 | lvlinux | ok in ur extensions.conf where you have the incoming call, have it dial one phone for xx seconds, then next priority dial another phone, then next priority another phone. |
| 21:14.56 | lvlinux | then if you want to start over, do a Goto to go back to the priority where the first phone gets called. |
| 21:15.08 | t4nk793 | i see |
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| 21:15.40 | t4nk793 | can you point me somewhere with an example or the syntax for such a thing, im new to the command line side of asterisk |
| 21:20.52 | lvlinux | t4nk793: i made you up a little example here: http://pastebin.com/k8cN9ATX |
| 21:24.08 | t4nk793 | awesome, thank you. |
| 21:24.32 | lvlinux | or here w voicemail after: http://pastebin.com/BRcEabP4 |
| 21:25.49 | t4nk793 | ah one wrinkle i forgot to mention, this will be calling external devices (cell phones) |
| 21:26.24 | lvlinux | over a PSTN line or SIP trunk? |
| 21:26.40 | t4nk793 | SIP trunk |
| 21:27.43 | lvlinux | k then you just change what is called. instead of "SIP/phone1" you would put "SIP/myprovider/12223334444" |
| 21:28.04 | t4nk793 | ah, excellent, thank you |
| 21:28.09 | t4nk793 | really appreciate the help |
| 21:28.14 | lvlinux | np |
| 21:28.45 | lvlinux | check out the * book too---it will help you with your dialplan skills. You'll never go back to the gui lol |
| 21:28.50 | lvlinux | ~book |
| 21:28.50 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:30.45 | t4nk793 | awesome |
| 21:30.52 | lvlinux | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#asterisk-DP-Basics |
| 21:31.09 | lvlinux | ^^^^ starts the dialplan basics part---step by step |
| 21:34.58 | t4nk793 | one more question, i know in asterisknow and the liek there is an option to have the user confirm the call (ie tts says "press 1 to accept this call") it would be cool if it would stop calls once someone answers the phone and presses 1 to accept |
| 21:35.07 | t4nk793 | how can I go about doing that? |
| 21:35.15 | t4nk793 | or where would i look for info on it |
| 21:36.06 | lvlinux | what exactly do you mean by "stop calls"? |
| 21:37.18 | t4nk793 | so when it calls an extension if someone presses 1 in respons to the prompt it stops hunting, rather than stopping the hunt when the phone picks up |
| 21:37.48 | lvlinux | oh ok hmmm... |
| 21:38.29 | t4nk793 | so basically it is confirming that a human answered and understood the call |
| 21:39.33 | t4nk793 | the overall idea for this is for nagios to be able to run a script to generate a call to the IT department and make sure someone answers and responds to the issue |
| 21:39.45 | lvlinux | ah i see. |
| 21:40.00 | t4nk793 | and to stop calling once someone acknowledges the issue |
| 21:40.38 | t4nk793 | right now our system calls a script and just spams the hell out of the IT departments cell phones until nagios is happy again |
| 21:40.40 | lvlinux | you can do it, i think you use followme for that. |
| 21:42.49 | lvlinux | well actually you could use all dialplan but followme would probably be simpler |
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| 21:43.17 | t4nk793 | ok ill look into that |
| 21:44.44 | lvlinux | i think followme will do your ring hunts too. |
| 21:46.13 | lvlinux | http://answers.oreilly.com/topic/2705-asterisk-how-to-use-followme-to-call-a-series-of-phone-numbers/ |
| 21:46.28 | lvlinux | ^^^^^^^^^^^^^^^this might be helpful to you |
| 21:46.52 | vslap | I'm having some MWI issues. I'm not sure if I have done it all correctly. Whats going on is if I get a vm left for me, I don't see the MWI update. If I reboot my phone, MWI is flashing. If I then delete the VM, it doesn't go off, unless I reboot the device again. GXP-2160 w/ Asterisk 12.1.0. |
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| 21:56.46 | malcolmd | navaismo: stickied! |
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| 21:57.23 | navaismo | thaaanks |
| 21:58.03 | malcolmd | thank you :) |
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| 22:02.13 | ledoktre | greetings y'all. A quick question regarding webdav. Is there a reasonable way to run webdav accounts under the virtual host user, rather than the web server user? |
| 22:02.26 | vslap | Yea my phone are subscribed for MWI & their mailbox=### entry is set. I see asterisk removing/touching /var/spool/asterisk/voicemail/default/200/INBOX/msg0000.txt |
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| 22:05.42 | theqkash | hello guys, I'm new to asterisk and I have some problem with configuring it in simple way - eg. to answer incoming call and play something (like music or anything) - at the moment I'm stuck at this thing: asterisk is registering to sip (I see registered in console and on sip provider panel) but when I'm trying to call my number added to this account there is information about "service unavailable" |
| 22:06.02 | theqkash | Could you please help me? |
| 22:06.06 | *** part/#asterisk ledoktre (~ledoktre@216.51.224.229) |
| 22:07.09 | paulc | theqkash: so you have a line in sip.conf registering with your provider? and "sip show registry" shows it successfully registered? |
| 22:08.15 | theqkash | Exactly - it is registered fine, but I cannot call number connected with this login here. When I have normally logged this number into any sip client like X-Lite I can call it without problems |
| 22:08.39 | theqkash | I have set verbose to 5 when connecting to cli and even when I call there is nothing in console. |
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| 22:10.27 | paulc | I'd do "core set debug 5" and "core set verbose 5".. then I'd check the default "context=..." line in sip.conf - out of the box I think it's "default". I usually set it to "in-sip" or something similar.. then in that new context (in extensions.conf), make sure you have an extension that matches.. either the DID specifically, or do _xx. as a catch all (in case your provider is sending it full E.164 vs 10 digits only etc) |
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| 22:14.24 | lvlinux | theqkash: is your * box behind NAT? that could be the problem too. |
| 22:14.40 | theqkash | Nope it isn't |
| 22:14.51 | lvlinux | oh good :-) |
| 22:15.01 | theqkash | even when I set nat in config there is still the same problem |
| 22:15.19 | lvlinux | if it's not behind nat, then you don't want it set in the config. |
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| 22:26.34 | theqkash | so at the moment these are my config files: http://pastebin.com/vcQQEPci |
| 22:27.00 | theqkash | if somebody have some idea, please just tell me |
| 22:30.55 | lvlinux | theqkash: try changing your 12345 extension in the incoming context to "s" and see if the call goes through. |
| 22:31.28 | theqkash | I have recently changed from s to 12345 - not working |
| 22:32.02 | lvlinux | k then i run a sip debug during incoming call and PB it |
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| 22:33.28 | theqkash | sadly there is nothing in console when calling |
| 22:33.48 | theqkash | only > doing dnsmgr_lookup for 'sip.tlenofon.pl' |
| 22:33.54 | theqkash | but this happens more times |
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| 22:41.46 | navaismo | sip set debug on then call then pastebin the output |
| 22:41.51 | lvlinux | theqkash: do "sip set debug on" |
| 22:42.07 | lvlinux | yes what navaismo said. |
| 22:43.31 | theqkash | http://pastebin.com/crkC2Qic |
| 22:43.33 | theqkash | only this |
| 22:43.38 | theqkash | and when I'm calling there is nothing |
| 22:44.29 | navaismo | hmmm |
| 22:45.28 | navaismo | there is no invite in that paste, did you redirect your service to your PBX or something? |
| 22:46.03 | theqkash | I'm not sure what you're talking about... I have set no redirections at all. |
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| 22:47.19 | navaismo | for example in my voip.ms account i have a setting to send all callss to my PBX or my phone |
| 22:48.26 | theqkash | so then nope, it isn't. As I have said before, I have set it in the same settings using X-lite and I was able to call my number |
| 22:51.21 | lvlinux | theqkash: pb ur sip.conf |
| 22:53.01 | theqkash | okay guys, done this thing |
| 22:53.09 | theqkash | there was some weird thing |
| 22:53.16 | theqkash | I have changed password on my sip provider's panel and in asterisk |
| 22:53.22 | theqkash | and it have worked like a charm |
| 22:54.27 | vslap | How does MWI work? Does it get sent over normal SIP messages? Does asterisk send a packet to the phone when it changes, or does the phone request the status at certain intervals? I'm trying to verify functionality because my mwi isnt working so I want to see whats going on under the hood or not. |
| 22:57.17 | lvlinux | theqkash: heh sometimes that's all it takes---maybe provider password was buggered up and changing it reset things. |
| 23:03.03 | lvlinux | vslap: yes it is standard SIP notify I believe |
| 23:03.48 | lvlinux | vslap: * sends to the phone when it detects cahnge in vm, but i think phones can sometime be set to subscribe too |
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| 23:04.00 | lvlinux | vslap: have u run sip debug to see what comes accross |
| 23:04.02 | lvlinux | ? |
| 23:04.04 | vslap | mine is set to subscribe to MWI |
| 23:04.13 | vslap | My sip debug is really verbose. |
| 23:04.27 | vslap | Its constantly hitting my sip provider back and forth. |
| 23:04.52 | vslap | lots of 401 unauthorized sip stuff and 404 not found. |
| 23:07.13 | lvlinux | vslap: is this a production system or personal? |
| 23:07.22 | vslap | Production but its afterhours. |
| 23:08.10 | vslap | I was responding to an * ticket I put in a few weeks ago, figured id clean up the config a bit. |
| 23:08.16 | *** part/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 23:08.29 | lvlinux | k |
| 23:09.22 | lvlinux | well i would run a sip debug and leave a voicemail, wait for the pollinterval (set in voicemail.conf) and see what messages come across. |
| 23:10.04 | vslap | ok I just added pollinterval recently, but it was my understanding that could be uneccesary |
| 23:10.37 | lvlinux | yeah it may be, mine updates immediately |
| 23:24.41 | vslap | This log is a mess. |
| 23:41.06 | vslap | How would I see the server sending a notify MWI sip message? |
| 23:42.06 | vslap | I see the MWI subscription |
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