| 00:00.15 | *** join/#asterisk eschmidbauer (~chatzilla@unaffiliated/eschmidbauer) |
| 00:14.12 | *** join/#asterisk eschmidbauer (~chatzilla@unaffiliated/eschmidbauer) |
| 00:14.22 | mafairnet | how to disable the transfer option to same queue i'm logged in? |
| 00:14.40 | *** part/#asterisk tom_w_______ (~telcomind@wsip-24-120-113-148.lv.lv.cox.net) |
| 00:19.51 | *** part/#asterisk kharwell (kharwell@nat/digium/x-fxrefuxlatfgdkdp) |
| 00:20.04 | MaliutaLap | mafairnet: can you be a little more precise? How are things ending up in this queue? |
| 00:20.22 | MaliutaLap | some kind of config pastebin might help |
| 00:20.40 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 00:20.41 | MaliutaLap | and a precise description of the problem/desired outcome |
| 00:22.41 | mafairnet | Im logged to queue 222 but i don't want to answer this call. I transfer the incomming call to the same queue (222) and it's answered by another agent logged in queue (222). This bad practice is what i want to disable. |
| 00:23.22 | mafairnet | I have tried the dialplan but the logic I'm programming just don't work. |
| 00:23.24 | mafairnet | :/ |
| 00:33.34 | [TK]D-Fender | Fix your logic |
| 00:33.41 | [TK]D-Fender | Your dialplan does what you tell it |
| 00:45.19 | *** join/#asterisk juanmapalad (~junmapala@112.210.69.250) |
| 00:45.36 | juanmapalad | !ping |
| 00:48.28 | mafairnet | but in the dialplan how to know which queues I'm logged in? |
| 00:51.13 | [TK]D-Fender | mafairnet, What were you doing before? |
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| 05:12.52 | _omer | What is the use of Agents (agents.conf) if I dont want to use agentlogin() and if I have to authenticate agent using vmauthenticate and use AddQueueMember to add SIP/Local device into the queue? Is there a way to login Agent into the queue same like agentlogin() but without using agentlogin? |
| 05:20.11 | [TK]D-Fender | no |
| 05:20.20 | [TK]D-Fender | There isn't a second way to do the exact same thing |
| 05:20.25 | [TK]D-Fender | That would be REDUNDANT |
| 05:26.47 | *** join/#asterisk _omer (omer@184.175.79.212) |
| 05:27.56 | _omer | [TK]D-Fender : so it means, agents.conf is useful with agentslogin() only.....right? |
| 05:28.57 | [TK]D-Fender | pretty |
| 05:28.58 | [TK]D-Fender | much |
| 05:29.05 | [TK]D-Fender | chan_agent = toast |
| 05:32.58 | _omer | here comes a confusion then. if I want to use hotdesk feature, I track and add SIP/Local device to Queue but actual extension (SIP Peer) of an agent is 100 then I won't be able to fetch agent's CDR at the end of the day if I don't know which device he was logged in from ..... you understand what I mean? |
| 05:34.01 | *** join/#asterisk gryphon (~gryphon@82.140.120.164) |
| 05:34.23 | [TK]D-Fender | "core show application addqueuemember" <--- |
| 05:34.34 | [TK]D-Fender | YOU tell it what to call the device you add. |
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| 05:39.01 | _omer | I understand, for example if a person is added to the queue using AQM and his device is SIP/777 then his call history will be saved as SIP/777.... |
| 05:39.51 | _omer | whereas , actual extension of this person was 100 (SIP/100) in the office. |
| 05:40.29 | [TK]D-Fender | " then his call history will be saved as SIP/777" <- only if you don't tell it to do otherwise |
| 05:41.51 | _omer | ah I got it ... further dialplan amendments needed to save call history as required.... |
| 05:41.59 | _omer | Thanks much! |
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| 05:42.28 | [TK]D-Fender | You're welcome |
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| 07:19.11 | D30 | hey guys,, i a problem with my asterisk recently., some call coming from a local pstn is not properly handled by asterisk, i can see that the call is being detected by asterisk i see from the log that our IVR is playing, but on the callerâs end its just keep ringing.. |
| 07:19.47 | D30 | im not sure if this is a problem with the digium card im using.. a TDM400P card |
| 07:19.55 | D30 | any advice? |
| 07:20.34 | AnonGirl | D30: Answer() |
| 07:20.48 | D30 | AnonGirl: ? |
| 07:20.59 | AnonGirl | D30: Does your dialplan Answer the call? |
| 07:21.21 | D30 | AnonGirl: sometimes the call just went through properly without issue.. |
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| 07:54.16 | ChannelZ | You didn't answer the question |
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| 08:20.47 | [sID] | Do any of you can use the SIP repo? |
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| 08:33.03 | *** join/#asterisk wasanzy (~wasanzy@41-66-254-58-dedicated.4u.com.gh) |
| 08:33.05 | wasanzy | hello |
| 08:33.29 | wasanzy | what do I need to get a toll free Asterisk system working? in terms of hardware and connectivity |
| 08:37.13 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:38.09 | ChannelZ | A computer, an internet connection, and an ITSP |
| 08:38.31 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:38.33 | Rico | ChannelZ: I would said "a computer" only |
| 08:38.37 | Rico | say |
| 08:39.16 | ChannelZ | So you can talk to no one? |
| 08:39.40 | ChannelZ | You can do that without the computer even. |
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| 08:40.48 | ChannelZ | [sID] not sure what that means... |
| 08:41.42 | wasanzy | Rico: so how will the users call from their phone to enter the Asterisk system? |
| 08:42.32 | Rico | wasanzy: did he talk about outgoing calls ? he can just register 2 softphones on his asterisk box |
| 08:42.57 | Rico | and place calls between his softphones |
| 08:44.03 | wasanzy | Rico: am asking for toll free where ppl can call say 0809547852 and it will come to the Asterisk system |
| 08:44.26 | wasanzy | so they will be calling from their phones |
| 08:44.41 | ChannelZ | ..like I said.. |
| 08:45.00 | doop | listen to ChannelZ |
| 08:45.30 | Rico | ok, i did not understood what "toll free" mean |
| 08:46.04 | wasanzy | ChannelZ: I got you there. my worry is users are on different network so how will this be possible for different networks? |
| 08:46.28 | doop | wasanzy a tollfree number should be dialable from any network |
| 08:46.31 | ChannelZ | The internet is one big network |
| 08:46.37 | ChannelZ | As is the telephone system |
| 08:46.46 | wasanzy | ok |
| 08:46.48 | doop | in the country where the tollfree number is located anyway |
| 08:46.53 | wasanzy | thank you |
| 08:47.19 | ChannelZ | You just buy a DID (your toll-free number) from a provider who has them in the country you desire, and people call that number like they would normally |
| 08:47.37 | ChannelZ | You get the calls over the internet into your Asterisk from your ITSP. |
| 08:47.56 | ChannelZ | You could even buy a toll-free number in another country and receive calls from it |
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| 08:50.54 | wasanzy | ChannelZ: oh thank you so much |
| 08:51.11 | *** join/#asterisk Borg (~borg@S0106001f290da141.ca.shawcable.net) |
| 08:52.49 | Borg | Privilege escalation protection disabled! |
| 08:52.49 | Borg | <PROTECTED> |
| 08:53.34 | ChannelZ | eh? |
| 08:53.40 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 08:54.21 | Borg | I'm fairly new to Asterisk and am not understanding the page I was pointed to following that error. :-s |
| 08:54.37 | ChannelZ | You got that error when doing what? |
| 08:54.56 | Borg | when attempting to start the daemon. |
| 08:55.38 | Borg | I've also been looking for a nice GUI management for Asterisk, but have noticed the Asterisk one that was being built was discontinued. :-s |
| 08:55.57 | ChannelZ | What version of asterisk? |
| 08:56.37 | Borg | Asterisk 1.8.32.2 |
| 08:56.37 | Borg | <PROTECTED> |
| 08:57.47 | ChannelZ | You can turn off live_dangerously in asterisk.conf |
| 08:58.27 | Borg | ok. I wasn't sure if that was wise or not because it says it relaxes security. |
| 08:59.28 | ChannelZ | Well it's telling you _now_ that it's relaxed security |
| 09:00.12 | Borg | ah right I see default is set to yes in the description I guess I missed that. thank you! :-) |
| 09:00.35 | ChannelZ | Basically it prevents you from calling certain functions externally (through AMI or ARI, or from the commandline) |
| 09:01.10 | ChannelZ | Although not ARI in your case since it doesn't exist in 1.8 |
| 09:01.31 | Borg | well, what I'm aiming to get is Asterisk integrated to the Openfire XMPP daemon. |
| 09:01.44 | ChannelZ | As for GUIs, there's FreePBX but it's a separate beast and not really supported here |
| 09:01.46 | wasanzy | Another question, with this toll free number, I can do anything like adding the callers in to a conference etc? |
| 09:02.30 | ChannelZ | wasanzy, yeah you can do whatever you want |
| 09:02.40 | Borg | ChannelZ, yeah I saw that. I downloaded it tonight to perhaps test it out, but I don't really have extra systems to just deploy for that purpose so I figured Asterisk would be perfect. |
| 09:02.55 | ChannelZ | Forward the call on to another phone, make them listen to music on hold until the end of time, conference them, whatever |
| 09:03.07 | wasanzy | do I pay monthly charges on the toll free number? |
| 09:03.33 | ChannelZ | Generally yes |
| 09:03.40 | Borg | if you pay for a toll-free number service chances are you will eat your prepaid time or you will be charged as you go yes. |
| 09:03.50 | *** join/#asterisk mesfet (~mesfet@host160-73-static.61-79-b.business.telecomitalia.it) |
| 09:04.04 | ChannelZ | Just depends on the ITSP.. they probably will have a monthly, and then per-minute charge, or perhaps they'll have a XXX minutes-per-month flat fee, etc. |
| 09:05.17 | wasanzy | ok |
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| 09:11.56 | wasanzy | lets say am hosting the Asterisk system in USA and the company who is supposed to received the toll free calls are in Australia, can they use softphone to receive the call? |
| 09:12.25 | doop | why would you do it that way |
| 09:12.53 | doop | put the asterisk system near where its clients are |
| 09:13.01 | wasanzy | doop: I am looking at hosting aspect |
| 09:13.20 | doop | and then you'll have jitter problems all day long |
| 09:13.30 | doop | don't do it that way |
| 09:13.43 | doop | put the asterisk system near where its clients are |
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| 09:14.32 | doop | you can put a second one in the USA and trunk between them if you want to |
| 09:14.38 | ChannelZ | Well the server is going to be remote to someone. |
| 09:14.43 | wasanzy | so the system should be in their office? and I only need one static IP that the ITSP will rount the calls to? |
| 09:15.12 | wasanzy | route |
| 09:15.23 | ChannelZ | It just depends on the quality of the internet connections. |
| 09:16.26 | *** join/#asterisk c0rnoTa (~c0rnoTa@89.17.43.185) |
| 09:16.41 | ChannelZ | If it is a single office these calls will be recieved at, then you can kill a leg of latency with having the server physically local to them. |
| 09:17.12 | wasanzy | what of if I have a quality Internet on the server in USA. won't that work? because the people going to call are from all over the world, specially USA and Australia but the company itself is in Australia |
| 09:17.18 | ChannelZ | But in general, technological hiccups aside, anything can technically be anywhere |
| 09:17.43 | wasanzy | ChannelZ: ok |
| 09:18.51 | ChannelZ | The more places you bounce the traffic to/from, the more latency |
| 09:19.47 | wasanzy | i c |
| 09:20.33 | Borg | yup. hops through many nodes will make a huge difference. |
| 09:21.24 | ChannelZ | But, if your callers are in the US, but you're in Australia, the traffic has to get from A to B eventually |
| 09:22.45 | wasanzy | that is how it is ChannelZ |
| 09:28.20 | ChannelZ | Does the company have an existing PBX for handling the internal phone network? Like, would local Australia calls be coming in some other route (PRI or traditional analog lines) or are you looking to move everything to VoIP? |
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| 09:30.04 | wasanzy | ChannelZ: am looking at moving everything to VoIP |
| 09:32.04 | ChannelZ | So there's something to be said for having the server physically local then to run all of the local devices (the phones at the company as well as local AUS calls) |
| 09:33.06 | ChannelZ | And let it connect back to the US for the US toll free number, rather than making all of the local devices go from AUS to the US and back again. |
| 09:34.41 | ChannelZ | If the company is in AUS but the whole asterisk server is in the US and your route between the two goes down, you lose everything. |
| 09:34.49 | wasanzy | ChannelZ: that is wonderful. but if different numbers are coming to the Asterisk, it means I have to route them all to one sip account? or can I make another phone ring in their office? |
| 09:35.04 | ChannelZ | You can route it any way you want. |
| 09:36.28 | ChannelZ | You could buy a local number in 10 different countries from 10 different ITSPs and bring them into the same system, and make it do anything you want when someone called any one of the numbers. |
| 09:37.21 | wasanzy | ChannelZ: So in all, your advise is to have the system located in their Office instead of USA because of the local phones? |
| 09:37.47 | ChannelZ | If you're still wanting to build a local PBX for the office, yes. |
| 09:38.44 | ChannelZ | I mean you could certainly do both |
| 09:38.52 | wasanzy | ok |
| 09:40.54 | ChannelZ | And on that note I need to go to bed. Have fun y'alls |
| 09:41.47 | wasanzy | I understand the bit about when the route between the two goes down, I lose everything. I think I will have the Asterisk in only one place, either USA or AUS |
| 09:43.02 | wasanzy | ChannelZ: good night thank you |
| 09:45.56 | Borg | hmm ok I seem to be stuck again... |
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| 10:14.27 | *** join/#asterisk yun1989 (5f5c784c@gateway/web/freenode/ip.95.92.120.76) |
| 10:14.33 | yun1989 | hello all |
| 10:14.44 | yun1989 | i have one problem in my dialplan |
| 10:14.52 | [sID] | ok |
| 10:14.53 | yun1989 | http://pastebin.com/8yqnwrFw |
| 10:15.02 | [sID] | wirte |
| 10:15.02 | yun1989 | you can see it in this link |
| 10:15.44 | [sID] | ok, wat the problem ? |
| 10:15.53 | *** join/#asterisk wasanzy (~wasanzy@41-66-254-58-dedicated.4u.com.gh) |
| 10:16.18 | Xaviertoor | [Feb 13 10:08:55] WARNING[52385][C-00000001]: app_confbridge.c:1610 confbridge_exec: Conference menu ) does not exist and can not be applied to confbridge user. |
| 10:17.34 | yun1989 | yes I see it |
| 10:17.46 | yun1989 | you can explain more about error ? |
| 10:18.22 | yun1989 | is one error in confbridge.conf or in extensions.conf ? |
| 10:21.57 | Xaviertoor | It can be in two files! |
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| 10:26.27 | yun1989 | http://pastebin.com/Q6RN5cC0 |
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| 10:26.37 | yun1989 | this is the confbridge.conf |
| 10:26.59 | yun1989 | i add the two confbridge in the top of the file |
| 10:28.11 | yun1989 | http://pastebin.com/KE5UGiDb |
| 10:28.22 | yun1989 | and this is my extensions.conf |
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| 10:45.35 | yun1989 | any sugestion ? |
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| 11:39.01 | anonymouz666 | hi! |
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| 12:48.15 | g-maurizi | can anyone tell me what this error is about? pubsub_on_rx_publish_request: No registered publish handler for event presence ? I am trying to understand SIP SIMPLE messaging and presence, but as far as I can understand asterisk does not conform to RFC for this and does not support it? I have some grandstream GXP2160 phones that have "presence watcher" functionality for BLF keys and "SIP message" |
| 12:48.16 | g-maurizi | functionality that I cant make work with asterisk/freepbx/asteriskNOW? |
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| 14:09.12 | nfi|ermes | some help with gotoif sintax: i wanna check if $temp is empty |
| 14:09.15 | nfi|ermes | http://pastebin.com/gJh72VEp |
| 14:09.23 | nfi|ermes | thanks in advance |
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| 14:20.10 | ghost75 | Assuming that in asterisk config dyndns address with wrong ip is used for externhost. Will it not be reachable by other sip peer until asterisk will reregister or until externrefresh parameter will find new ip after dyndns was fixed? |
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| 14:43.13 | yun1989 | hello all |
| 14:43.23 | yun1989 | i receive this error in CLI |
| 14:43.24 | yun1989 | WARNING[55365][C-00000000]: file.c:830 ast_readaudio_callback: Failed to write frame |
| 14:43.37 | yun1989 | anyone have ideia why ? |
| 14:49.42 | file | the channel was hung up? |
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| 14:54.01 | yun1989 | i think yes |
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| 15:07.17 | mjordan | we may have gotten a tad aggressive on the Playback warnings |
| 15:09.51 | file | mjordan, I think that actually predates... me. |
| 15:10.17 | file | yeah, ti does |
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| 15:10.34 | file | Texas Instruments definitely does |
| 15:12.10 | mjordan | file: unless... you're The Highlander |
| 15:12.17 | file | mjordan, shhhhh |
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| 15:15.34 | Qwell | true story: nobody remembers when file started |
| 15:16.10 | file | Qwell, the past! |
| 15:16.20 | file | or the future, if time wraps around... |
| 15:16.32 | Qwell | or the present, depending on when time wrapped around. |
| 15:16.39 | Qwell | Did you ever think of THAT? |
| 15:16.44 | file | no |
| 15:16.48 | file | but I did think of THAI! |
| 15:16.52 | Qwell | mmm, thai |
| 15:16.58 | file | exactly |
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| 16:18.24 | nfi|ermes | in my dialplan i have this extension : exten => *82.,1,Macro(user-callerid) |
| 16:19.46 | nfi|ermes | when i dial *8234 from X-lite (software softphone) everything works fine but when i dial the same extension (*8234) via a phone connected to Handytone ATA , i get the tone as soon as i dial *82 |
| 16:21.49 | rrittgarn1 | sounds like the ATA has *codes set in it to do different things. I know the Cisco PAP2Ts did... |
| 16:23.42 | lvlinux | yup, your ATA is set to recognize *82 for callerid itself, so it is intercepting it before it gets to * |
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| 16:30.35 | Qwell | more likely, the dialplan on the ATA has *XX or something, so it sends to the PBX once that's dialed. |
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| 16:39.21 | Tuju | argh, my outbound calls are again to:1234@172.16.1.1 - not with FQDN domain name. |
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| 17:42.02 | file | who dat who dat |
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| 18:07.31 | Tuju | why i cannot say "sip set debug invite" ? |
| 18:07.48 | Tuju | very hard to find anything from that REGISTER, OPTIONS flood. |
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| 18:08.11 | lvlinux | i think you can only set sip debug on/off and on/off for specific peer I believe |
| 18:10.02 | Tuju | indeed. and that's the problem. |
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| 18:20.11 | sgriepentrog | Tuju: using tshark works better in that case: tshark -R 'sip.Method==INVITE' |
| 18:20.31 | Tuju | tshark, that's something new. |
| 18:20.35 | Tuju | hmmmm |
| 18:20.41 | sgriepentrog | text (i.e. cli) wireshark |
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| 18:32.18 | [[thufir]] | I'm in the asterisk CLI -- pretty neat. I tried "sip show channels" but it didn't give much detail. how do I show all the config details? |
| 18:32.49 | sgriepentrog | sip show channel x |
| 18:40.22 | [[thufir]] | sgriepentrog: thanks |
| 18:42.21 | Tuju | [[thufir]]: you can tabulate those channel id's too |
| 18:44.50 | [[thufir]] | thanks, guys. I have: http://pastebin.com/GDwb6AtS however, I don't believe it shows all the relevant configuration. or does that show the actual settings? (the settings are wrong, I just want to cat/print/etc them.) |
| 18:47.47 | sgriepentrog | Tuju: trying typing sip show channel and hitting tab. The channel name should pop in, press enter, voila full details. |
| 18:51.18 | [[thufir]] | ok |
| 18:53.32 | [[thufir]] | I know that these settings are incorrect, but these are the SIP settings for a specific channel? http://pastebin.com/dsjggMfi |
| 18:54.47 | Synthase_ | Setting for the endpoint, rather? |
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| 19:02.40 | [TK]D-Fender | [[thufir]]: That is the last meessage from a SIP COMM. That is not a CALL, or PEER SETTINGS. |
| 19:03.02 | [TK]D-Fender | [[thufir]]: It's looking like you have misunderstood what a channel is... |
| 19:04.33 | [[thufir]] | I thought a channel was, like sip trunking. the provider. I have an asterisk, I can reference that. RTFM is fine :) pointers appreciated. It's the "definitive guide", v4 with a starfish I think. |
| 19:05.50 | [TK]D-Fender | [[thufir]]: No. A channel is a comm in progress |
| 19:06.15 | [TK]D-Fender | [[thufir]]: A PEER defines how comms are to be created <- |
| 19:06.25 | [TK]D-Fender | [[thufir]]: those are "settings" |
| 19:06.36 | [[thufir]] | [TK]D-Fender: ahhh, I see. thanks. |
| 19:06.37 | [TK]D-Fender | [[thufir]]: A channel is what is HAPPENING. Which right now ... is nothihng |
| 19:06.43 | [[thufir]] | lol. yup |
| 19:06.44 | [TK]D-Fender | nothing* |
| 19:08.05 | [TK]D-Fender | [[thufir]]: "sip show peer babytel" |
| 19:08.15 | [TK]D-Fender | Peername: babytel |
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| 19:14.18 | AnonGirl | [[thufir]]: SIP trunks don't exist |
| 19:15.07 | [[thufir]] | AnonGirl: you're hurting my brain...in a good way. isn't that what companies sell? SIP trunking? |
| 19:15.40 | AnonGirl | [[thufir]]: yeah... but SIP trunking is an imaginary concept. It's not like IAX, where a trunk is actually a thing |
| 19:15.58 | [[thufir]] | ok, a construct. gotcha. |
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| 19:39.20 | [[thufir]] | is this more reasonable for the configuration of a peer? http://pastebin.com/K6k2Hyux (babytel.ca) |
| 19:41.06 | [TK]D-Fender | Maybe |
| 19:41.12 | [TK]D-Fender | Have you TRIED it? |
| 19:41.20 | [TK]D-Fender | knowing = doing |
| 19:45.58 | [[thufir]] | ok. |
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| 21:02.32 | derekb | hello |
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| 21:03.09 | derekb | curious to see if anyone has had issues with not getting proper caller ID when a call is parked, and picked up from the same phone. |
| 21:04.15 | derekb | seems that if we make an outbound call, park and pick it up at the same phone, the CLID is changed to sw_####_"SIP". internal and inbound calls show correct CLID when parked and picked up. if we set sendrpid=no and trustrpid=no, the CLID will just show the parking lot number. |
| 21:04.20 | derekb | running asterisk 13.2 |
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