| 00:12.17 | haroldp | thanks for your help |
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| 01:13.10 | raspberrypifan | i cant get asterisk now to boot in a VM |
| 01:13.13 | raspberrypifan | it says things are missing |
| 01:13.20 | raspberrypifan | adn hten it hangs on the printer |
| 01:13.22 | raspberrypifan | section |
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| 01:15.22 | WIMPy | Printer? |
| 01:15.40 | raspberrypifan | yes it says loading printer driver or something to that and it doesnt do anything |
| 01:15.41 | raspberrypifan | after that |
| 01:15.46 | raspberrypifan | im gonna try rerunning it right now |
| 01:16.04 | raspberrypifan | i just downloaded the asterisk now 64 bit, im on a mac running vmware fusion |
| 01:19.07 | raspberrypifan | first issue it says that GCC does not exist |
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| 01:29.43 | raspberrypifan | is my dvd corrupt? |
| 01:29.49 | raspberrypifan | WIMPy |
| 01:31.20 | WIMPy | I have NFI what it needs GCC for, but not having a working GCC can be related to virtualisation and an OS that doesn't cope with your configuration. |
| 01:31.38 | raspberrypifan | hm |
| 01:31.59 | raspberrypifan | its stuck at starting virtual printing demaon again |
| 01:32.42 | WIMPy | N idea what that's good for either. |
| 01:32.49 | WIMPy | No... |
| 01:33.10 | raspberrypifan | hm |
| 01:33.13 | raspberrypifan | im gonna try the 32 bit version |
| 01:33.20 | WIMPy | Why don't you just install your favourite OS and then install Asterisk on it? |
| 01:33.29 | raspberrypifan | my favorite os is os x |
| 01:33.33 | raspberrypifan | hwo do i install it on os xlol |
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| 01:34.06 | WIMPy | From source, I guess. |
| 01:34.22 | raspberrypifan | hm |
| 01:34.24 | raspberrypifan | bad idea |
| 01:34.37 | WIMPy | Why? |
| 01:34.44 | raspberrypifan | its gonna leave files all over my drive |
| 01:35.18 | WIMPy | that's the idea of installing software. |
| 01:36.02 | raspberrypifan | but its gonna have file dysentery |
| 01:36.20 | WIMPy | What? |
| 01:36.39 | raspberrypifan | spread all kinds of files al over the place |
| 01:36.42 | raspberrypifan | and be unable to delet all of them |
| 01:36.48 | raspberrypifan | why isnt there a package for os x |
| 01:37.27 | WIMPy | Have you checked if there is one? |
| 01:38.26 | raspberrypifan | yes ive looked |
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| 02:32.42 | ChannelZ | You're only allowed to use Facetime on OSX |
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| 05:03.59 | KNERD | do faxes go over IAX2, or is it SIP only? |
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| 05:31.48 | ChannelZ | Depends as what |
| 05:32.53 | ChannelZ | If you're talking bleepy bleepy standard fax as audio, it'll go across pretty much anything so long as the codec doesn't destroy it (so, ulaw/alaw) or your jitter/packet loss doesn't destroy it |
| 05:34.18 | ChannelZ | If you're talking t.38, I think that aught to work as well. |
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| 08:21.41 | neeby_goosey | How do I control asterisk behaviour in case of failed blind transfer? |
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| 08:23.43 | neeby_goosey | I mean what place in dialplan is triggered by that? |
| 08:26.34 | michael_work | i think there is some parameter or value of callback or something. i use my own dialplan for that to i'm free to implement it as i want, same to you |
| 08:27.44 | michael_work | btw if you transfer without answering it would be forward and asterik do not allow to call back same exten it was forwarded from |
| 08:28.14 | michael_work | which is logic as it might create infinitive loop in case if phone is set to forward and not person doing it |
| 08:29.46 | neeby_goosey | By transfering without answering you mean that the device we tried to transfer to doesn't answer? Or device that triggered transfer? |
| 08:30.00 | michael_work | device that trigger |
| 08:30.04 | michael_work | A calls B |
| 08:30.13 | michael_work | B transfer C without answering |
| 08:30.20 | michael_work | B => forwarded |
| 08:30.52 | michael_work | A calls B, B answer and transfer to C without talking to C => blind transfer |
| 08:31.14 | michael_work | in second case B can call A as well and after B transfer |
| 08:31.54 | neeby_goosey | Thanks, I get your point now. But still, I don't understand how to keep the call between A and B in case B->C transfering fails. |
| 08:33.00 | neeby_goosey | By default, the call ends. |
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| 08:33.35 | neeby_goosey | Do you know the solution? |
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| 08:36.48 | michael_work | your own dialplan? |
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| 08:37.45 | Zogot | ahoyhoy |
| 08:39.23 | neeby_goosey | michael_work, I have my own dialplan, what's next? |
| 08:39.46 | michael_work | set some parameter |
| 08:40.06 | neeby_goosey | My problem is I have no idea where the call goes in dialplan after it has been unsuccessfully transfered ( |
| 08:40.23 | michael_work | after you use Dial() check the DIALSTATUS and if it's one you want to trasfer back goto callback dialplan or something like that |
| 08:40.39 | michael_work | use g option and check DIALSTATUS |
| 08:40.47 | neeby_goosey | That's something, thank you. |
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| 10:01.44 | neeby_goosey | I noticed that System('curl <some-url>'); command freezes the dialplan when <some-url> is inaccessible. So is there any failsafe way to 'curl' from dialplan? |
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| 12:36.06 | neeby_goosey | Ok, it seems that a CURL function with a bunch of timeout CURLOPTS will do just fine. |
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| 12:41.03 | marceloamorim | Guys, yesterday I was wondering if I can handle it better with my codecs, I`m working at an ISP and bandwidth never was a problem, so I always used ulaw and alaw, but if I have any an sip account when the internet isn´t good enough, I should try to work with some codec that use less bandwidth? |
| 12:41.57 | bulkorok | you can try it with g729 (some costs usually…) |
| 12:42.25 | neeby_goosey | What about free gsm? |
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| 12:43.05 | marceloamorim | My think about this is, if I have one ip-phone set with g726 or gsm or anothe codec and I tried call to another phone with ulaw, I´ll have issue with the audio? |
| 12:43.52 | neeby_goosey | Depends on what are codec preferences on PBX and another phone. |
| 12:46.07 | marceloamorim | I usually set on phones ulaw and alaw, and on my sip.conf I set all codecs avaiable |
| 12:47.12 | neeby_goosey | Then, provided you have all the neccessary asterisk modules loaded, asterisk would convert one audio format stream to another in real time. |
| 12:48.05 | marceloamorim | so its fine if one phone I have gsm and another phone I have ulaw |
| 12:48.34 | neeby_goosey | Yes. Here are the modules used: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-CHP-5-SECT-1.html#Architecture_id292600 |
| 12:48.41 | marceloamorim | asterisk will works fine and convert the audio |
| 12:48.44 | marceloamorim | cool |
| 12:49.15 | neeby_goosey | But the convertion is generally not recommended because it takes you machine's resources. |
| 12:49.23 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 12:49.38 | neeby_goosey | Better set everything up to use the same codec. |
| 12:49.44 | [TK]D-Fender | That's very generic... |
| 12:50.05 | [TK]D-Fender | Unless you have a tragicall weak server or process a lot of simultaneous calls it hardly matters |
| 12:50.16 | neeby_goosey | Well, yes. |
| 12:53.41 | marceloamorim | my phone just have g711A(alaw) g711U(ulaw), g723, g726,g729 and iLBC |
| 12:53.56 | marceloamorim | so I usually set just ulaw and alaw |
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| 13:00.52 | marceloamorim | MOS for g729 is 3.92 |
| 13:01.18 | marceloamorim | better than g726 |
| 13:01.43 | marceloamorim | nice, maybe I should try, less bandwidth and good MOS |
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| 13:02.51 | bulkorok | g729 is espacially developed for low bandwith and high rtt |
| 13:03.11 | bulkorok | it can be used with rtt up to 300ms ! |
| 13:03.24 | bulkorok | for one direction ;-) |
| 13:03.43 | marceloamorim | cool, I´ll try this |
| 13:04.03 | bulkorok | do it… sounds way more better than gsm |
| 13:05.09 | neeby_goosey | I wish g729 was free... |
| 13:05.58 | neeby_goosey | That is a really good codec. |
| 13:06.07 | marceloamorim | gsm is used for most teamspeak that I used, for like 100 users on the same room |
| 13:06.10 | bulkorok | some years and he is... |
| 13:06.17 | marceloamorim | g729 isn´t free? |
| 13:06.21 | bulkorok | nope |
| 13:06.23 | marceloamorim | o.O |
| 13:06.30 | bulkorok | 10$ per en/decoder |
| 13:06.41 | marceloamorim | but, I have this codec on my phone |
| 13:06.55 | marceloamorim | maybe they pay for this |
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| 13:07.54 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:08.00 | bulkorok | they do |
| 13:09.14 | marceloamorim | so I can use for "free", because this price is built-in on the value of the phone |
| 13:09.17 | bulkorok | http://en.wikipedia.org/wiki/G729#Licensing |
| 13:09.23 | bulkorok | you can |
| 13:10.26 | marceloamorim | nice, I should use this because I paid for this without know =) |
| 13:11.09 | bulkorok | usually all hardware phones can do g729 |
| 13:11.18 | neeby_goosey | A better link: http://www.voip-info.org/wiki/view/Asterisk+G.729+Licensing |
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| 13:23.57 | As001 | Hello I saw strange behavior of Asterisk 1.8.23. It emit multiple Bridge events in AMI for one call. I saw 1000 of them for one single call and I don't know why is that happen. That does not happen always but few times a day. Is it connected with dtmf and https://issues.asterisk.org/jira/browse/ASTERISK-18639 ? |
| 13:24.43 | As001 | Can agent who press numbers on xlite softphone to trigger leave and new bridge for the same connection ? |
| 13:27.31 | *** join/#asterisk StaRetji (~vencuga@87.250.52.14) |
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| 13:30.51 | [TK]D-Fender | AMI bridge doesn't listen for DTMF |
| 13:31.35 | [TK]D-Fender | And the versions listed in that report are really bad, and you are 4 releases behind on your own branch which isn't the same |
| 13:35.31 | StaRetji | hello, what is default setting for ;maxcalls = 10; Maximum amount of calls allowed. |
| 13:35.38 | StaRetji | i hope it is not 10? |
| 13:35.39 | StaRetji | lol |
| 13:35.58 | StaRetji | is there cli command to print this value? |
| 13:38.13 | As001 | [TK]D-Fender why then so many Bridge events for same call ?I will consider upgrade in 1.8 branch as soon as I am sure it will work with our software. |
| 13:39.16 | *** join/#asterisk wouter (~wouter@barbershop.grep.be) |
| 13:39.28 | wouter | I'm trying to figure out why asterisk isn't logging failed logins |
| 13:40.14 | wouter | when I did 'asterisk -r', I was immediately swamped with failed login attempts, to the extent that my console was useless in trying to debug anything |
| 13:40.19 | wouter | but nothing is showing in the logs |
| 13:40.32 | wouter | (offending IP has been firewalled away now, but still) |
| 13:40.46 | wouter | (asterisk 1.8.13.1 as packaged by Debian, fwiw) |
| 13:41.02 | snmp | ~pb |
| 13:41.02 | 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. |
| 13:41.08 | snmp | yeah) |
| 13:41.20 | wouter | is that to me? |
| 13:41.40 | snmp | wouter: what doyou know of logger.conf |
| 13:41.56 | wouter | didn't know it existed... looking at it now :) |
| 13:42.05 | snmp | yeap) |
| 13:42.32 | [TK]D-Fender | [09:23]As001Hello I saw strange behavior of Asterisk 1.8.23 <--- THIS is 4 versions out. I did not suggest to change branches |
| 13:42.50 | [TK]D-Fender | As001: And 1.8 doesn't get new features so it shouldn't change compatability |
| 13:44.17 | marceloamorim | yeah, I didn´t find codec_g729 on the codecs, doesn´t matter if the phone has the option, the asterisk doesn´t have =( thx for the link guys |
| 13:45.04 | As001 | yes I understand what you meant to change to newer 1.8 version but I tried with 1.8.25 and had problems for some reason I couldn't find, while 1.8.23 is working without problems. |
| 13:45.21 | As001 | except this Bridge... |
| 13:45.38 | As001 | I will try 1.8 current in few days. |
| 13:47.29 | *** join/#asterisk BlackDex_ (~BlackDex@ori.vyus.nl) |
| 13:47.38 | BlackDex_ | Hello there |
| 13:48.10 | BlackDex | I want to have a specific sip client to call out with a dedicated outgoing number |
| 13:48.14 | BlackDex | how can i configure that? |
| 13:49.10 | StaRetji | is it possible that asterisk drops calls with 503 when is overloaded? Need to know that, thanks |
| 13:51.13 | neeby_goosey | BlackDex, you must inform your provider about this intention, and if they accept it then simply use one of the appropriate Dial application options, e.g. Dial(SIP/VoIP-provider/88003001020,,f(<dedicated-outgoing-number>)); |
| 13:51.45 | neeby_goosey | Is that what you want? |
| 13:52.06 | BlackDex | we have an ISDN card and a number range from 00 till 99 to use |
| 13:52.38 | leifmadsen | StaRetji: unless you setup a call counter or something, asterisk doesn't know what "overloaded" means |
| 13:52.48 | BlackDex | i can set the caller-id for forwarded calls and such, but i can't get it to have a specific sip client to use a dedicated outgoing number |
| 13:53.01 | [TK]D-Fender | BlackDex: "core show function CALLERID" <- set it before you dial |
| 13:53.25 | BlackDex | and how to do that for a specific SIP client? |
| 13:54.42 | StaRetji | thanks leifmadsen, I am getting lot of 503 calls while isp trunks ha 1000 connetctions free |
| 13:54.55 | [TK]D-Fender | BlackDex: Its your dialplan. It does what you tell it to |
| 13:55.02 | StaRetji | and I have just about 100 calls |
| 13:55.46 | *** part/#asterisk As001 (~uros@82.117.198.142) |
| 13:56.10 | BlackDex | [TK]D-Fender: i think i have found it thx |
| 13:56.36 | StaRetji | peers uses g729 while I use asterisk.lv g729 codec ( are these codec same, or maybe asterisk is doing codec translation? sorry if question is stupid :( |
| 14:00.08 | jameswf | StaRetji: Are you in a room staffed by the nice folks who you are stealing from asking about pirated software? |
| 14:00.54 | StaRetji | jameswf: what do you mean? |
| 14:01.15 | StaRetji | asterisk.lv is not legal? If you are saying that, i am removing it right now |
| 14:01.46 | jameswf | StaRetji: g.729 is licenced software. You may buy a licence and download the proper binaries from Digium |
| 14:02.22 | jameswf | StaRetji: http://www.digium.com/en/products/software/g729-codec |
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| 14:03.53 | StaRetji | omg, I thought the ones i used are free mate, ok, removing it right now, going back to ulaw/alaw |
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| 14:38.01 | dan_j | Hi, how do I reload the dundi keys without a full reload? |
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| 15:04.36 | *** join/#asterisk rhineheart_m (~chatzilla@unaffiliated/rhineheartm/x-283746) |
| 15:05.19 | rhineheart_m | hello.. after changing the codecs in asterisk sip settings.. I can no longer make an outgoing call.. but I can call sip phones within the network without any problem.. any idea? thanks. |
| 15:07.25 | MaliutaLap | rhineheart_m: the ITSP doesn't support the codecs you are allowing out to it ... that's my thought |
| 15:07.32 | newtonr | yup |
| 15:07.41 | newtonr | rhineheart_m, should be easy to tell with a SIP trace |
| 15:08.03 | MaliutaLap | rhineheart_m: or check the support info from your ITSP and check what codecs they support |
| 15:09.31 | rhineheart_m | I tried calling the ptsn using another landline and I can hear the IVR... |
| 15:10.01 | rhineheart_m | making outgoing calls will show up "hang up" message only.. |
| 15:16.55 | [TK]D-Fender | Look at the actual call and see what is actually happening |
| 15:17.15 | rhineheart_m | how to do that exactly? |
| 15:17.43 | [TK]D-Fender | * CLI with SIP debug enabled. |
| 15:18.12 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
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| 15:23.01 | rhineheart_m | ok...in *cli now.. what to issue command to read the logs to determine what causes the termination of outbound route? |
| 15:24.07 | [TK]D-Fender | "core set verbose 10" |
| 15:24.11 | [TK]D-Fender | "sip set debug on" |
| 15:24.13 | [TK]D-Fender | Place a call. |
| 15:24.20 | [TK]D-Fender | pastebin the entire thing from beginning to end |
| 15:24.21 | [TK]D-Fender | ~pb |
| 15:24.22 | 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. |
| 15:24.23 | [TK]D-Fender | ^^^^^ |
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| 15:37.10 | rhineheart_m | http://pastebin.com/Wzzkh06z |
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| 15:42.34 | rhineheart_m | [TK]D-Fender: http://pastebin.com/Wzzkh06z |
| 15:44.47 | [TK]D-Fender | I don't see it terminating you... |
| 15:44.55 | [TK]D-Fender | I see your SIP device choosing to end the call |
| 15:45.00 | [TK]D-Fender | 531 |
| 15:51.38 | rhineheart_m | yeah.. I just noticed it too.. it allows using my csip in my android...I'll try using an ip phone |
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| 15:58.13 | rhineheart_m | lastly, what's the difference of this.. NAT yes no never route (under the NAT Settings) ? |
| 15:58.13 | rhineheart_m | if the server is located in the same lan segment with the sip clients... do I still need NAT to be in yes? |
| 15:58.13 | *** join/#asterisk workingcats (~workingca@212.122.48.77) |
| 15:58.13 | lvlinux | rhineheart_m: NO! only set NAT if there is NAT |
| 15:58.49 | [TK]D-Fender | Doesn't really matter |
| 15:59.17 | rhineheart_m | as far as I know.. please correct me though if I'm wrong... NAT usually should be enabled if the sip server is connected directly to a modem right? |
| 15:59.30 | [TK]D-Fender | No. |
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| 15:59.49 | rhineheart_m | or let's say..if firewall is enabled? |
| 16:00.00 | [TK]D-Fender | NAT is specified on multiple levels, under [general] and under your individual peers |
| 16:00.21 | [TK]D-Fender | And leave the term "firewall" out.. that only implies filtering rules. |
| 16:00.29 | [TK]D-Fender | not transforms |
| 16:01.08 | *** part/#asterisk hecatae (~philip@221.ttcbilling.com) |
| 16:02.17 | rhineheart_m | ok.. why did you say it doesn't matter? |
| 16:02.46 | [TK]D-Fender | Because it doesn't on that level. |
| 16:03.52 | rhineheart_m | so in my case.. can I just choose never? |
| 16:04.42 | rhineheart_m | does it has effect to call quality esp on a bw restricted environment? |
| 16:05.03 | rhineheart_m | where some of the sip clients are connected to the network wirelessly? |
| 16:05.35 | [TK]D-Fender | No effect at all |
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| 16:12.05 | rhineheart_m | ok. Thanks for the clarification.. I thought it has something to do with the natting/hopping.. |
| 16:12.28 | rhineheart_m | unlike making it like in a bridge mode |
| 16:14.06 | [TK]D-Fender | If they are on the same boring subnet it doesn't cause * to do anything any different if you did the rest right |
| 16:23.23 | lvlinux | yeah maybe not, but it will confuse you later if you look at your config and see nat=yes :-D |
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| 16:24.49 | rhineheart_m | thanks for your time guys.. I appreciate it :) |
| 16:24.58 | rhineheart_m | will go ahead now. |
| 16:28.39 | *** join/#asterisk cusco (~tralala@2001:41d0:1:6caf::cafe) |
| 16:28.44 | cusco | yellow |
| 16:32.52 | lvlinux | hai |
| 16:32.55 | cusco | :) |
| 16:32.59 | lvlinux | hehe |
| 16:33.26 | cusco | a queue has members configured with local interfaces but SIP/xyz state_interface |
| 16:33.38 | cusco | why does a queue tries to ring a member that is in_use |
| 16:33.41 | cusco | for instance .. |
| 16:33.49 | cusco | what settings should I look at? |
| 16:33.56 | cusco | or a member that is paused |
| 16:33.59 | [TK]D-Fender | Show us the queue and device states |
| 16:34.08 | cusco | by show you mean... |
| 16:34.12 | cusco | queue show? |
| 16:34.23 | [TK]D-Fender | yes, as well as hint dumps, channel dumps, etc |
| 16:34.36 | [TK]D-Fender | Would also be nice to know what version you're even running... |
| 16:34.57 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 16:35.03 | cusco | well :( i'm afraid to answer that one... heh |
| 16:35.08 | cusco | still on 1.6.2 |
| 16:35.48 | cusco | we tried migrating to 11 a few months back, had to rollback, and haven't had time to plan new migration |
| 16:36.19 | cusco | so.. can I still provide those details you mentioned, or should I stop asking now? |
| 16:36.22 | cusco | :p |
| 16:36.46 | [TK]D-Fender | Don't make me ask and then give nothing.... |
| 16:36.56 | cusco | hold ... |
| 16:38.21 | cusco | we're not using hints but how to check that? |
| 16:38.32 | cusco | is there a cli command to dump hint? |
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| 16:39.24 | [TK]D-Fender | core show hints |
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| 16:41.53 | cusco | [TK]D-Fender: http://paste.debian.net/98314/ |
| 16:42.18 | cusco | 691 was not in use at the time of queue show |
| 16:42.46 | cusco | in the Local dialplan we do a ChanIsAvail() |
| 16:42.59 | cusco | and return busy() |
| 16:43.08 | cusco | if other than not in use |
| 16:43.27 | cusco | queue is real_time |
| 16:43.31 | [TK]D-Fender | I don't see a confirmation of state devices used, code to set those, the actual device state proving what's going on. |
| 16:44.11 | [TK]D-Fender | And go set up hints for the devices so I can prove what * sees for them at least |
| 16:44.40 | cusco | ok let me read about that.. meanwhile would you like a select to the table that holds queue_members? |
| 16:44.56 | [TK]D-Fender | Bring it all at once in one pastebin |
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| 16:50.33 | SuperNull | Anyone using Cisco Routers as a registrar /sbc with asterisk ? |
| 16:50.38 | cusco | ok I'm having trouble to understand hints .. I should use hints instead of SIP/xyz in state interface? |
| 16:50.52 | cusco | hint:xyz@context ? |
| 16:53.25 | cusco | then shows as invalid in "queue show" |
| 16:54.06 | cusco | ah modules loaded before .. ok |
| 16:54.10 | *** join/#asterisk scouture (~scouture@unaffiliated/scouture) |
| 16:54.54 | cusco | hmm.. still shows invalid |
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| 16:58.31 | [TK]D-Fender | chan_sip, etc should be loaded before app_queue |
| 16:58.35 | [TK]D-Fender | preload <- |
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| 17:04.57 | cusco | [TK]D-Fender: so I can just reload app_queue |
| 17:04.58 | cusco | right? |
| 17:05.12 | cusco | I'm missing something regarding hits |
| 17:05.19 | cusco | its probably not just set the state interface |
| 17:06.04 | cusco | gah |
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| 17:36.53 | navaismo | stupid earthquakes |
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| 18:01.29 | jwwwww_ | Hello. |
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| 18:01.59 | navaismo | moshimoshi |
| 18:02.02 | leifmadsen | preload should almost never be needed; the modules have ordering controls |
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| 18:06.14 | darkdrgn2k | im looking for a way to find a "vanity" 10digit dial nubmer, whats the best way, services etc about looking for one? |
| 18:06.32 | ChkDigit | Any reason why a hangup() would leave a sip phone "on hook" |
| 18:06.38 | ChkDigit | ? |
| 18:06.43 | ChkDigit | Errr off hook. |
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| 18:14.31 | BCrookAtRA | I'm trying to play a recording to my callers that contains all ten touchtones (the goal is to trick autodialer robots into thinking we pressed whatever digit they say to press to be removed). But the touchtones always come out sounding clipped as if they were too loud, no matter how quiet I make them, or howlong i play them. Does asterisk 'intercept' touchtones somehow? CIT tones play normally as expected even though they are significntly lo |
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| 18:28.34 | [TK]D-Fender | [14:14]BCrookAtRAI'm trying to play a recording to my callers that contains all ten touchtones (the goal is to trick autodialer robots into thinking we pressed whatever digit they say to press to be removed). But the touchtones always come out sounding clipped as if they were too loud, no matter how quiet I make them, or howlong i play them. Does asterisk 'intercept' touchtones somehow?... |
| 18:28.35 | [TK]D-Fender | ...CIT tones play normally as expected even though they are significntly lo <- your audio file is what it is. |
| 18:29.00 | [TK]D-Fender | either your file is too loud or it's getting amplified out of your channel |
| 18:29.08 | file | what what oh |
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| 18:45.12 | lvlinux | if i downgrade my dahdi version, do i need to recompile asterisk? |
| 18:45.34 | BCrookAtRA | I've halved the volume of the touchtones several times now, and they are nearly indistinguishable when plaid as an mp3 or looked at as a waveform |
| 18:45.44 | [TK]D-Fender | lvlinux: depends on chan_dahdi |
| 18:45.47 | BCrookAtRA | they still crackle though and sound like they're clipping when asterisk plays them to a caller |
| 18:46.21 | [TK]D-Fender | MP# is alrady 1 lossy level and you've not given us any detail what you're sending it over |
| 18:46.52 | lvlinux | [TK]D-Fender: i compiled and installed dahdi 2.9 and asterisk 11.9, then tried to compile wanpipe, and I had to go with a version of wanpipe that only supported dahdi 2.7, so I had to go down to 2.7. |
| 18:47.23 | [TK]D-Fender | Might have to recompile... go try |
| 18:47.44 | lvlinux | if * loads dahdi fine and doesn't complain about anything, can i assume it's ok? |
| 18:47.53 | [TK]D-Fender | If it works... it's OK |
| 18:48.02 | lvlinux | k thanks |
| 18:48.11 | lvlinux | i'll just try it and see |
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| 19:29.24 | cusco | so.. I've been reading on the hints.. I'm using AEL... I created a test extension: hint(SIP/${EXTEN:3}) _999XXX ... Im checking that it is not valid.. how do you go around this? |
| 19:37.33 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 19:47.15 | cusco | [TK]D-Fender: do I have to create a hint for each queue member? |
| 19:48.34 | [TK]D-Fender | I don't do AEL |
| 19:48.39 | cusco | I understand |
| 19:48.48 | cusco | if it were in extensions.conf tho |
| 19:48.58 | cusco | how would you go about it? |
| 19:51.08 | [TK]D-Fender | you don't do "variableRead the docs, I do't knwo the syntax, except to ssa you should be doing each EXPLICITLY, not with variables & patterns |
| 19:52.14 | cusco | ok.. the subscribecontext can be a completelly different context from the rest of my dialplan |
| 19:52.22 | cusco | err... |
| 19:52.23 | cusco | nevermind |
| 19:52.54 | cusco | so I need to decleare each hint explicitly for each device |
| 19:52.59 | cusco | right? |
| 19:57.36 | [TK]D-Fender | I don't even care about subscriptions, it's a case of proving * CAN track the device at all |
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| 20:00.50 | dan_j | If I want to match any character or number for an extension, is there anything better than _. ? |
| 20:01.08 | dan_j | I seem to recall that you get warnings when using _. |
| 20:01.50 | dan_j | _[a-zA-Z0-9]. ? |
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| 20:07.50 | [TK]D-Fender | checkout time, BBL |
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| 20:33.12 | dan_j | Just using dundi for the first time. When using a dialplan to perform a lookup, it seems to take a long time till the dialled extension starts ringing. But when I perform a lookup using the CLI, I get a result straight away. |
| 20:33.19 | dan_j | Any ideas what it could be? |
| 20:33.29 | dan_j | I'm connecting the two asterisk servers via IAX |
| 20:33.36 | dan_j | The other calls are sip |
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| 20:42.58 | *** join/#asterisk gushi (~danm@prime.gushi.org) |
| 20:43.29 | gushi | Hey all -- can anyone tell me how I can get asterisk to give me more context on this error? "[May 8 11:16:08] WARNING[27869] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)" |
| 20:43.47 | gushi | I'd love to know, for example, where in my dialplan it's trying to do that? |
| 20:44.02 | gushi | or what peer it's trying to dial? |
| 20:44.19 | lvlinux | gushi: core set verbose 9 |
| 20:44.27 | navaismo | sip set debug on |
| 20:48.49 | gushi | lvlinux, that helped...apparently it's because some of our extensions (softphones, mainly) aren't connected, so it's harmless, apparently. |
| 20:49.17 | lvlinux | gushi: yep that'll do it :-) |
| 20:49.55 | gushi | There's no syntax to specify the dial in that will prevent that error, is there? |
| 20:50.05 | dan_j | When dialling a peer normally, you can do Dial(SIP/${EXTEN}/${DID}), but when dialling a peer using dundi, all you can do is Goto(lookupdid,${EXTEN}) |
| 20:50.07 | gushi | well, warning, properly speaking |
| 20:50.15 | dan_j | Is there any way to get the ${DID} passed through too? |
| 20:50.27 | dan_j | Or any other variables like timeouts etc. |
| 20:52.12 | lvlinux | gushi: there probably is a way to check if a phone is there, and adjust the dial string, but I don't think it would be worth messing with just to get rid of the warning. |
| 20:52.51 | gushi | Okay -- we migrated from 1.4 to 1.8 recently, and I've been grepping for all the various WARNINGS, and any I can make go away are nice. |
| 20:53.27 | gushi | and most of our extensions dial like five different clients (desk sets, home numbers, cell forwards, softphones, etc). |
| 20:53.55 | dan_j | It will usually mean that one of the sip phones you are trying to dial is currently offline. |
| 20:54.04 | gushi | so there's kinda a lot of this error. Especially because I tend to insist on "if you have two devices, like a smartphone and an iphone, that's two different identities" |
| 20:54.31 | dan_j | insist? its necessary if you want both to ring |
| 20:54.47 | dan_j | but if one is offline and you try to dial it, you'll get that warning |
| 20:55.00 | gushi | not everyone does. Some people just want it so they can call home when they're at a conference. |
| 20:55.17 | gushi | but I'm being a jerk because I want to avoid the problem. |
| 20:55.55 | gushi | it also lets me be more safe about the "stolen laptop problem" |
| 20:56.16 | gushi | so yeah, net result, a lot of this warning :) |
| 20:56.32 | dan_j | yes. basically |
| 20:58.10 | gushi | And yeah, I suppose I could handle all the dialing through a macro or even an agi that explodes the original string and checks if they're all online, and only dials the ones that are...but eww. |
| 20:58.43 | lvlinux | yep |
| 20:59.59 | gushi | adds a grep -v to his nightly cronjobs |
| 21:01.29 | gushi | So, is there any way to shut off the warning about mismatched NAT and NONAT settings? That prints itself ONCE PER PEER? |
| 21:01.59 | *** join/#asterisk Zogot (~Adium@90-145-116-55.bbserv.nl) |
| 21:04.39 | lvlinux | yeah, change the config so they aren't mismatched :-) |
| 21:07.54 | gushi | I can maybe understand that paranoia if my sip identities are simple 4-digit extensions (which I already see people trying to register as, and which I'll be solving with fail2ban) |
| 21:08.46 | gushi | I can even understand printing the warning once if any extension mismatches the standard, but in my case, things are set that way for a reason. Cisco 7960's are touchy behind NAT, and when settings mismatch. |
| 21:14.43 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 21:22.10 | *** join/#asterisk jonno11 (~jonno11@amigopod.rave.ac.uk) |
| 21:22.37 | dan_j | I'm trying to use a gosub but depending on the exten that passed to the gosub, i get this 'Attempt to reach a non-existent destination for Gosub' |
| 21:22.55 | dan_j | I've tried using the 'i' extension to execute a 'return' but that doesnt appear to be supported. |
| 21:23.01 | dan_j | whats the correct way of doing it? |
| 21:25.14 | dan_j | alternatively, is there any way to get a gosub to automatically return, even when return is not executed? |
| 21:30.38 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
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| 21:41.53 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 21:54.38 | *** join/#asterisk swiss__ (~swiss__@70.184.40.66) |
| 21:55.25 | swiss__ | hey all, trying to use Read() to get a dtmf key press from a user and i cant get it to recognize a key press, any ideas? |
| 21:55.33 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
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| 22:00.02 | navaismo | dtmfmode? |
| 22:00.17 | swiss__ | the one i set in sip.conf correct? |
| 22:00.49 | navaismo | let me use my magicians powers to know which one you are using |
| 22:00.56 | navaismo | concentrates |
| 22:01.08 | navaismo | focus pocus |
| 22:01.14 | swiss__ | no need just want to make sure i understand the question |
| 22:01.29 | navaismo | rfc2833 maybe? |
| 22:01.35 | swiss__ | if you mean the one in sip.conf ive tried auto and inband |
| 22:01.43 | navaismo | oh wait! |
| 22:02.11 | navaismo | i forgotten to ask-->from where you are dialing, sip internal device, pstn or..? |
| 22:02.24 | swiss__ | sip internal device |
| 22:02.36 | navaismo | ok try to use rfc2833 |
| 22:02.44 | swiss__ | let me give that a shot |
| 22:04.23 | swiss__ | negative |
| 22:04.41 | swiss__ | allow me to send you my various file through the magic of pastebin unless you can use your telepathy on those as well |
| 22:04.47 | navaismo | time to see the cli with the dtmf debg enabled |
| 22:04.58 | swiss__ | 10-4 |
| 22:05.01 | navaismo | yeah via pastebin |
| 22:08.07 | swiss__ | http://pastebin.com/USeWEYWk |
| 22:08.09 | swiss__ | magic |
| 22:08.40 | navaismo | dont you love that? |
| 22:08.50 | swiss__ | to no end |
| 22:10.17 | navaismo | ok now a call with the dtmf log enabled, you can enable the dtmf log via logger.conf adding dtmf to the console line, then reload the logger in the cli with logger reload |
| 22:12.42 | swiss__ | done, now to call |
| 22:16.20 | *** part/#asterisk hecatae (~Philip@host-92-27-124-62.static.as13285.net) |
| 22:18.32 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 22:18.32 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 22:20.16 | swiss__ | no luck |
| 22:20.27 | swiss__ | i dont get anything no matte how hard i push buttons |
| 22:20.35 | swiss__ | usually doing things harder fixes it |
| 22:26.55 | navaismo | didi you get the cli output? pastebin it if so |
| 22:27.14 | swiss__ | cli didnt return any dtmf i mashed |
| 22:27.31 | swiss__ | also tried inband in addition to rfc |
| 22:27.31 | navaismo | did you enabled the log? |
| 22:27.52 | swiss__ | yup |
| 22:27.55 | swiss__ | nothing there either |
| 22:28.29 | navaismo | enable that in the full log and see if there is the dtmf record |
| 22:28.40 | swiss__ | did that and negative |
| 22:28.53 | swiss__ | whats odd is i had this working yesterday for a brief time |
| 22:29.02 | navaismo | sip device right? like softphone |
| 22:29.08 | navaismo | or is an ata |
| 22:29.29 | swiss__ | one is softphone the other is a cell |
| 22:31.50 | *** join/#asterisk rhineheart_m (~chatzilla@unaffiliated/rhineheartm/x-283746) |
| 22:33.45 | navaismo | not sure if in the rtp debug you can see if the dtmf is coming to asterisk |
| 22:34.08 | rhineheart_m | hello.. I have this issue. Making outgoing calls using an at.com ip phone after changing the codecs in sip settings is no longer possible. I can make outgoing call using csip softphone in android only.. Here's my pastebin. Thanks.. http://pastebin.com/kedDB2aH |
| 22:36.26 | swiss__ | on a lark i also tried fiddling with the Read() setup but no luck. let me check the rtp |
| 22:37.03 | navaismo | rhineheart_m, lines 770-772 hangup cause 58-->BEARERCAPABILITY_NOTAVAIL do you have g729 licenses? |
| 22:37.10 | swiss__ | hmmm i cant find anything related to rtp |
| 22:37.35 | navaismo | rtp set debug on |
| 22:37.42 | swiss__ | ah that |
| 22:37.42 | navaismo | which asterisk version |
| 22:37.56 | swiss__ | 1.8.10.1 |
| 22:38.11 | rhineheart_m | navaismo: do I need to purchase it? |
| 22:38.21 | navaismo | g729 isnt free |
| 22:38.28 | navaismo | ~g729 |
| 22:38.29 | infobot | [~g729] G.729(.a /.ab /.b) is a patent-encumbered ITU-standard voice codec operating at 8kbps offering quality similar to GSM. For Asterisk to transcode G.729 licenses (per channel) must be bought from http://store.digium.com |
| 22:39.47 | rhineheart_m | what does it mean with per channel? |
| 22:39.54 | rhineheart_m | is this the one? http://www.digium.com/en/products/software/g729-codec |
| 22:40.14 | navaismo | yep, each channel use a licence |
| 22:40.23 | navaismo | s/licence/license |
| 22:40.44 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 22:40.49 | swiss__ | no luck |
| 22:41.03 | *** join/#asterisk u0m3_ (~u0m3@92.80.96.59) |
| 22:41.21 | rhineheart_m | navaismo: per channel means per ip phone? |
| 22:42.02 | navaismo | ~channel |
| 22:42.02 | infobot | hmm... channel is This refers to the group of resellers that supply most companies with software, hardware, and support. The channel is a force to be reckoned with, and it competes directly against companies like Gateway 2000 and Dell. A channel can also be a content container, like a television channel. You may be watching TV channels on the Internet someday. |
| 22:42.15 | navaismo | weird |
| 22:42.25 | rhineheart_m | lol.. I like you infobot |
| 22:42.36 | navaismo | a normal call use two channels, |
| 22:43.11 | rhineheart_m | Digium's G.729 Codec for Asterisk is licensed on a per-channel basis. A channel is defined as a single connection from an endpoint to an Asterisk application, or a bi-directional call between two endpoints attached to Asterisk. |
| 22:44.56 | navaismo | ok that |
| 22:45.11 | navaismo | why not only disable the codec? |
| 22:46.23 | rhineheart_m | of this following codec..which is free and considered to only use a small bandwidth? G711u |
| 22:46.24 | rhineheart_m | G711a |
| 22:46.26 | rhineheart_m | G722 |
| 22:46.27 | rhineheart_m | G723 |
| 22:46.29 | rhineheart_m | G726 |
| 22:46.30 | rhineheart_m | sorry for the multiple lines.. |
| 22:47.13 | swiss__ | anger! if i disable the GotoIF line then it works, i can get dtmf feed back |
| 22:47.22 | navaismo | im on a test? because im feeling nervous |
| 22:48.00 | rhineheart_m | navaismo: is that supposed for me? |
| 22:55.49 | rhineheart_m | navaismo: I disabled g279 and enabled ulaw.. I can now make outgoing calls.. thanks. :) |
| 22:59.23 | navaismo | nice |
| 22:59.24 | swiss__ | so ive got it narrowed down |
| 22:59.55 | swiss__ | if i include the GotoIF line it doesnt work, remove that and i can get dtmf back |
| 22:59.56 | *** join/#asterisk timahvo1 (~rogue@197.237.174.64) |
| 23:00.50 | navaismo | cli output please |
| 23:02.24 | *** join/#asterisk rhineheart_m (~chatzilla@unaffiliated/rhineheartm/x-283746) |
| 23:02.31 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 23:05.15 | swiss__ | http://pastebin.com/RHfxX5tY |
| 23:07.26 | navaismo | voip,ms use rfc2833 too thst weird really |
| 23:08.09 | swiss__ | i set it back to inband on a hunch and thats where we are |
| 23:09.14 | rhineheart_m | of this following codec..which is free and considered to only use a small bandwidth? G711u G711a G722 G723 G726. Thanks, |
| 23:10.53 | navaismo | http://en.wikipedia.org/wiki/Comparison_of_audio_formats |
| 23:12.15 | rhineheart_m | thanks but it didn't show any bandwidth comparison.. |
| 23:12.52 | navaismo | http://www.cisco.com/c/en/us/support/docs/voice/voice-quality/7934-bwidth-consume.html |
| 23:13.42 | navaismo | ~bandwidth |
| 23:13.42 | infobot | [~bandwidth] This is a measure in bits per second of the amount of data that can be sent over a particular cable, interface, or bus. |
| 23:13.47 | rhineheart_m | that helps.. thanks.. |
| 23:14.21 | navaismo | ~lmgtf?codec+bandwidth+comparison |
| 23:14.27 | navaismo | ~lmgtf |
| 23:14.35 | navaismo | ¬¬ |
| 23:14.52 | navaismo | ~lmgtfy |
| 23:14.53 | infobot | i heard lmgtfy is http://lmgtfy.com/ |
| 23:15.03 | navaismo | ~lmgtfy?bandwidth |
| 23:15.10 | navaismo | I never get it :( |
| 23:15.20 | navaismo | <PROTECTED> |
| 23:15.35 | navaismo | give up |
| 23:16.10 | navaismo | ~sudo |
| 23:16.11 | infobot | [~sudo] Better than su, according to talon. It allows a permitted user to execute a command as the superuser or another user, as specified in the sudoers file. Or can allow you to do silly things like run X11 apps with root privileges; also good in scripts with "username ALL = NOPASSWD: /some/program", or http://www.aplawrence.com/Basics/sudo.html, or good for ordering sandwiches, or not pseudo |
| 23:16.37 | navaismo | ~sudo make me sandwich |
| 23:16.37 | infobot | make me sandwich: sudo make navaismo a sandwich. |
| 23:17.49 | rhineheart_m | you're great navaismo. :D |
| 23:18.01 | rhineheart_m | I will make sandwich for you.. |
| 23:18.28 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:18.49 | navaismo | no im not, [TK]D-Fender is fabulous |
| 23:19.04 | swiss__ | so any ideas on the macro issue? |
| 23:19.29 | navaismo | not me |
| 23:19.53 | [TK]D-Fender | Yeah, I'll look.... psatebin it all up |
| 23:20.00 | [TK]D-Fender | pastebin* |
| 23:20.01 | [TK]D-Fender | ~pb |
| 23:20.02 | 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. |
| 23:20.06 | [TK]D-Fender | ^^^^ |
| 23:20.47 | swiss__ | http://pastebin.com/RHfxX5tY |
| 23:20.50 | swiss__ | output^^ |
| 23:21.53 | [TK]D-Fender | then you hav a DTMF mode issue, not a macro issue |
| 23:22.15 | swiss__ | but it has to be macro because thats the only change that makes a difference |
| 23:23.03 | [TK]D-Fender | "core show application read" <- |
| 23:23.39 | rhineheart_m | [TK]D-Fender: thanks for helping. I appreciate it :D |
| 23:23.53 | [TK]D-Fender | rhineheart_m: Can't remember the issue... but you're welcome |
| 23:24.26 | swiss__ | yeah ive read the documentation |
| 23:25.27 | swiss__ | more than once |
| 23:25.47 | swiss__ | and im fairly certain the syntax is correct |
| 23:27.04 | [TK]D-Fender | pastebin it.... |
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| 23:29.33 | *** join/#asterisk Rumbles (~Rumbles@cpc65400-shef11-2-0-cust198.17-1.cable.virginm.net) |
| 23:29.55 | mushroomed | Hi, I have my devide registered already but when I pick up the handset I have no tone |
| 23:30.05 | swiss__ | http://pastebin.com/w33jyQQj |
| 23:30.14 | Rumbles | Hi :) |
| 23:30.15 | mushroomed | Also when I call that extension I get --> Everyone is busy/congested at this time |
| 23:30.44 | Rumbles | just wondering, is there a limit to the number of remote UNIX conenctions asterisk would accept? and if so, can it be increased? specifically in 1.2.x :( |
| 23:31.34 | ChannelZ-Wk | How do you know it's properly registered? |
| 23:32.10 | ChannelZ-Wk | And the dialtone is locally generated on the device so if it's not providing it then it thinks something is wrong I guess |
| 23:32.14 | mushroomed | ChannelZ: Because I see it in my *CLI as a connected peer |
| 23:32.48 | ChannelZ-Wk | At the right IP? |
| 23:32.53 | mushroomed | ChannelZ: Yes~ |
| 23:33.39 | ChannelZ-Wk | Then without seeing some SIP debug, it sounds like your device is horribly confused about something. |
| 23:33.47 | mushroomed | It's a Grandstream GXP1405 |
| 23:33.57 | mushroomed | ChannelZ: How can I make SIP debug? |
| 23:34.04 | ChannelZ-Wk | sip set debug on |
| 23:38.19 | mushroomed | ChannelZ: http://pastebin.com/1mN401pB |
| 23:44.33 | [TK]D-Fender | [19:22][TK]D-Fender"core show application read" <- |
| 23:44.39 | [TK]D-Fender | ^ Paastebin |
| 23:45.38 | swiss__ | http://pastebin.com/iSV8WXdE |
| 23:45.56 | file | looks around |
| 23:46.57 | mushroomed | ChannelZ: Any news? |
| 23:48.04 | [TK]D-Fender | swiss__: Ok, your call description in thre looks wird, could you place those as 2 calls.... |
| 23:48.31 | [TK]D-Fender | swiss__: the comments are worrded oddly and you broke up the debug in pieces with it |
| 23:50.50 | swiss__ | alright hold on |
| 23:51.03 | swiss__ | let me do a fresh test and send that along |
| 23:52.30 | mushroomed | [TK]D-Fender: Can you help me please? |
| 23:54.49 | [TK]D-Fender | mushroomed: I don't see "congested" anywhere in that pastebin |
| 23:55.00 | mushroomed | [TK]D-Fender: -- Got SIP response 486 "Busy Here" back from |
| 23:55.07 | mushroomed | That's what I get |
| 23:55.17 | leifmadsen | looks at file. Intently. |
| 23:55.23 | mushroomed | And this --> == Everyone is busy/congested at this time (1:1/0/0) |
| 23:55.42 | [TK]D-Fender | that isn't in this last pastebin |
| 23:55.59 | [TK]D-Fender | [19:38]mushroomedChannelZ: http://pastebin.com/1mN401pB <- not here.... |
| 23:56.28 | file | leifmadsen, hi |
| 23:56.34 | leifmadsen | file: well hello to you sir |
| 23:56.43 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 23:57.42 | file | leifmadsen, would you like some mushroom and cognac risotto? |
| 23:57.44 | mushroomed | [TK]D-Fender: http://pastebin.com/PdHXZNYt |
| 23:57.57 | leifmadsen | file: is it a trap? |
| 23:58.00 | swiss__ | http://pastebin.com/KUzEpd7u |
| 23:58.04 | file | leifmadsen, might be |
| 23:58.09 | ChannelZ-Wk | sorry I had to wander off |
| 23:58.10 | leifmadsen | I'll pass, just in case |
| 23:58.12 | leifmadsen | sounds delicious |
| 23:58.17 | leifmadsen | I will obtain a beer instead |
| 23:58.29 | mushroomed | [TK]D-Fender: ChannelZ-Wk: http://pastebin.com/PdHXZNYt --> Line 419 |
| 23:58.34 | leifmadsen | I suppose you were not offering, but rather just asking if I wanted some |
| 23:58.42 | leifmadsen | ASSUME NOTHING WITH MR. FILE |
| 23:58.43 | file | correct! |
| 23:59.26 | ChannelZ-Wk | See line 398 |
| 23:59.39 | ChannelZ-Wk | Your phone is rejecting the call. Is DND on or something? |