| 00:04.13 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 00:10.25 | *** join/#asterisk Petchaw (~Pascal@ool-18e42dfd.dyn.optonline.net) |
| 00:30.38 | *** part/#asterisk Petchaw (~Pascal@ool-18e42dfd.dyn.optonline.net) |
| 00:33.05 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:33.06 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:33.16 | dijib | ? |
| 00:33.23 | SeRi | dijib: yo |
| 00:34.02 | *** join/#asterisk b0ot (~DynamicFa@147.177.61.218) |
| 00:34.08 | dijib | eh dude |
| 00:34.10 | dijib | lol |
| 00:34.13 | SeRi | :/ |
| 00:34.14 | dijib | my xmpp is up now |
| 00:34.17 | b0ot | [TK]D-Fender, how do you make asterisk accept all incomming calls if they match |
| 00:34.22 | SeRi | I cant see you dijib |
| 00:34.23 | dijib | whats up? ive got my truck here now |
| 00:34.24 | b0ot | basically remove the requirement for a peer? |
| 00:34.27 | SeRi | send me a msg |
| 00:34.33 | SeRi | on xmmp |
| 00:34.34 | dijib | no? maybe we need to add eachother again |
| 00:34.49 | SeRi | That didnt sound right but what ever |
| 00:34.53 | SeRi | lol |
| 00:34.59 | SeRi | sned me a msg |
| 00:35.04 | SeRi | send* |
| 00:36.39 | dijib | im going to plug your butt |
| 00:37.12 | [TK]D-Fender | b0ot[TK]D-Fender, how do you make asterisk accept all incomming calls if they match <- if "they" match.... isn't that the very definition of "acceptance"? |
| 00:37.34 | [TK]D-Fender | b0ot: And avoid vague use of "they" |
| 00:37.48 | b0ot | right now, asterisk rejects incomming calls from external trunks because they are not peers |
| 00:37.59 | b0ot | I don't want to have setup a peer for each trunk |
| 00:38.15 | [TK]D-Fender | b0ot: Next, stop using "trunk" generically as well |
| 00:38.23 | dijib | give me a min seri |
| 00:38.27 | [TK]D-Fender | b0ot: CHANNELS and TYPES please |
| 00:41.05 | b0ot | in the sip.conf file, I currenlty have one peer called [siptrunk] with type=peer, context=incomming, insecure=invite,port, and host=10.3.1.11 I want to modify my sip.conf file so that any type of incomming call ot the system will use context=incomming (whether it is a vaild peer or not) |
| 00:41.46 | [TK]D-Fender | allowguest=yes |
| 00:41.47 | [TK]D-Fender | ^ |
| 00:42.47 | b0ot | [TK]D-Fender, I already have that |
| 00:42.56 | b0ot | and allowsubscribe=yes |
| 00:43.21 | [TK]D-Fender | Then maybe you should show your actual config and the actual failure |
| 00:43.25 | [TK]D-Fender | ~pb |
| 00:43.26 | 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. |
| 00:43.27 | [TK]D-Fender | ^^^ |
| 00:46.08 | b0ot | pastebin.com/sZ1qKD87 |
| 00:46.18 | b0ot | http://pastebin.com/sZ1qKD87 |
| 00:46.51 | [TK]D-Fender | You've filtered a LOT in there |
| 00:47.09 | [TK]D-Fender | and I don't see a CONTEXT specified at all for [general] |
| 00:47.29 | [TK]D-Fender | ....and that doesn't include the failed call to look at |
| 00:47.47 | [TK]D-Fender | <PROTECTED> |
| 00:48.05 | [TK]D-Fender | Ther is a lot wrong with this first round... |
| 00:48.41 | [TK]D-Fender | araits a new pastebin |
| 00:48.45 | [TK]D-Fender | awaits* |
| 00:49.53 | b0ot | http://mq1Hyuq6 |
| 00:50.01 | b0ot | http://pastebin.com/mq1Hyuq6 |
| 00:50.20 | [TK]D-Fender | Looking for 5021 in default (domain 10.1.1.50) |
| 00:50.25 | [TK]D-Fender | you didn't specify the CONTEXT |
| 00:50.32 | [TK]D-Fender | So it hit [default] |
| 00:50.38 | [TK]D-Fender | [20:47][TK]D-Fenderand I don't see a CONTEXT specified at all for [general] |
| 00:50.55 | [TK]D-Fender | SIP/2.0 404 Not Found <- and clearly no match there |
| 00:50.56 | *** join/#asterisk rneese (~RNeese@pool-108-3-80-177.pitbpa.east.verizon.net) |
| 00:51.38 | *** part/#asterisk rneese (~RNeese@pool-108-3-80-177.pitbpa.east.verizon.net) |
| 00:52.00 | b0ot | bam |
| 00:52.06 | b0ot | added context=incomming |
| 00:52.08 | b0ot | it works now |
| 00:52.58 | b0ot | thanks |
| 00:55.39 | [TK]D-Fender | Always pay close attention to the peer it claims to batch and the "looking for" |
| 00:55.43 | [TK]D-Fender | match* |
| 00:58.41 | b0ot | [TK]D-Fender, do you view any value in the dcap certification and do you happen to know how intensive it goes |
| 00:59.10 | [TK]D-Fender | The question is do employers care about it |
| 00:59.42 | [TK]D-Fender | Because if someone tells me they have a large alphabet after their name.... then maybe they are jsut good at multiple -choice tests |
| 01:00.41 | [TK]D-Fender | Some people can cram and pass but aren't particularly bright and have no retention, and maybe got lucky on a few. |
| 01:01.12 | [TK]D-Fender | I evaluate results so .... to me it matters little. |
| 01:04.40 | b0ot | interesting |
| 01:04.45 | b0ot | the dcap is free isn't it |
| 01:05.04 | [TK]D-Fender | Nope |
| 01:05.21 | [TK]D-Fender | paid testing centers |
| 01:07.40 | b0ot | hmm i thought some level of asterisk cert was free |
| 01:08.50 | [TK]D-Fender | It'd be interesting to hear why you you seemed to think so.... |
| 01:09.03 | [TK]D-Fender | Who else gives free certs? |
| 01:10.04 | b0ot | organizations encouraging people to become certified |
| 01:10.16 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 01:10.41 | b0ot | it seems like an asterisk certification (from a purely certification standpoint) is much less "valuable" than say a ccna/np (voice) |
| 01:10.49 | [TK]D-Fender | so ... basically nobody to validate your expectation? |
| 01:11.33 | [TK]D-Fender | Have you done any job-searching for Asterisk-related jobs? |
| 01:16.01 | *** join/#asterisk jetlag (~jetlag@pool-71-168-200-61.cmdnnj.east.verizon.net) |
| 01:19.35 | *** join/#asterisk jetlag (~jetlag@pool-71-168-200-61.cmdnnj.east.verizon.net) |
| 01:19.47 | b0ot | [TK]D-Fender, I already have a job |
| 01:24.34 | *** join/#asterisk blee (~blee@68.204.1.103) |
| 01:55.52 | dijib | [TK]D-Fender: there is a free asterisk cert.. just not one of much importance |
| 01:57.29 | SeRi | Most jobs do care for certs |
| 01:57.55 | dijib | certs = insurability |
| 01:58.01 | SeRi | conf? |
| 02:08.12 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 02:08.31 | dijib | i might spot in here from time to time SeRi |
| 02:08.51 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 02:35.39 | *** join/#asterisk gnudna (~sklav@unaffiliated/sklav) |
| 02:36.26 | gnudna | Hi guys |
| 02:36.39 | gnudna | got some asterisk questions mostly related to security |
| 02:37.00 | gnudna | any body around for a somewhat quick chat |
| 02:38.24 | gnudna | im wondering what is the best way to keep someone from trying to do sip registrations |
| 02:39.26 | *** join/#asterisk OverOnTheRock (~irc@162.r1.ray.transact.bm) |
| 02:39.38 | gnudna | how can i specify something like permit=192.168.1.0/255.255.255.0 globally in the sip.conf |
| 02:40.00 | gnudna | meaning only local traffic can register as a sip device |
| 02:40.30 | gnudna | i would rather not do it by phone/extension since there are quite a few of them |
| 02:42.47 | *** join/#asterisk mnathani (~mnathani@198-84-231-11.cpe.teksavvy.com) |
| 02:44.47 | *** join/#asterisk OverOnTheRock (~irc@162.r1.ray.transact.bm) |
| 02:51.35 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-zurningfsgdqcjuh) |
| 02:51.35 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:52.21 | gnudna | if i add contactdeny=0.0.0.0/0.0.0.0 and contactpermit=192.168.1.0/255.255.255.0 |
| 02:52.35 | gnudna | will this limit sip registrations to internal devices only? |
| 03:06.23 | gnudna | anybody around? can asnwer the question i asked? |
| 03:08.36 | *** join/#asterisk OverOnTheRock (~irc@162.r1.ray.transact.bm) |
| 03:15.37 | gnudna | is there any updated asterisk security guide all i keep finding is dated back to 2009 |
| 03:18.49 | *** join/#asterisk dgv (~dgv@201.21.188.145) |
| 03:56.21 | *** join/#asterisk classix (~salven@silenceisdefeat.com) |
| 04:00.06 | carrar | gnudna, just ensure thing is plugged into all the ethernet ports and you'll be ok |
| 04:00.17 | carrar | gnudna, just ensure nothing is plugged into all the ethernet ports and you'll be ok |
| 04:00.23 | carrar | heh |
| 04:01.01 | *** join/#asterisk Penguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 04:04.46 | gnudna | funny |
| 04:04.48 | gnudna | :) |
| 04:06.22 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 04:11.47 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 04:11.47 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 04:17.30 | gnudna | so does adding the following to sip.conf actually do anything ->alwaysauthreject = yes -> contactdeny=0.0.0.0/0.0.0.0 -> contactpermit=192.168.11.0/255.255.255.0 |
| 04:17.55 | gnudna | hard to ell looking at the logs if it does anything |
| 04:18.54 | gnudna | i used firewall to block at the moment but i would like to think the above helps somewhat by limiting where devices can register from and not give obvious info about extensions |
| 04:32.32 | *** part/#asterisk gnudna (~sklav@unaffiliated/sklav) |
| 04:35.00 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 04:40.11 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 05:12.50 | *** join/#asterisk ccherrett (~christoph@184.70.186.190) |
| 05:14.40 | ccherrett | http://pastebin.com/VHrmiaQE |
| 05:14.46 | ccherrett | I cannot access voicemail |
| 05:14.49 | ccherrett | any thoughts? |
| 05:16.38 | *** part/#asterisk dgv (~dgv@201.21.188.145) |
| 05:17.53 | nightrid3r | wrong password |
| 05:18.23 | ccherrett | nightrid3r: it is not |
| 05:18.28 | ccherrett | for sure not |
| 05:18.30 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 05:18.46 | nightrid3r | -- Incorrect password '' for user '104' (context = default) |
| 05:18.55 | ccherrett | yeah it is not incorrect |
| 05:19.03 | ccherrett | it is not passing correct info |
| 05:19.26 | ccherrett | I am running a vpn client and server |
| 05:19.38 | ccherrett | on the server side sip phones can check vm |
| 05:19.44 | ccherrett | on the client side they cannot |
| 05:19.56 | ccherrett | they also cannot dial into a conference |
| 05:20.02 | ccherrett | invalid password every time |
| 05:20.48 | nightrid3r | hmmm |
| 05:21.02 | ccherrett | when we took all firewalls out of the equation it worked |
| 05:21.34 | ccherrett | we also connected to another asterisk server from our client side and it worked |
| 05:21.39 | nightrid3r | something goes wrong with dtmf |
| 05:21.44 | ccherrett | so my asterisk server is misconfigured somehow |
| 05:21.50 | ccherrett | nightrid3r: I think so |
| 05:32.42 | [TK]D-Fender | [21:55]dijib[TK]D-Fender: there is a free asterisk cert.. just not one of much importance <- which one? |
| 05:33.30 | [TK]D-Fender | ccherrett: Clearly the DTMF isn't making it <- |
| 05:33.42 | [TK]D-Fender | ccherrett: Wrong mode or firewalling issue |
| 05:34.37 | ccherrett | [TK]D-Fender: I think you are right |
| 05:34.54 | ccherrett | any idea what it would be on openvpn? |
| 05:35.01 | ccherrett | routing of some sort? |
| 05:35.31 | ccherrett | I just set the server to dtmfmode=auto |
| 05:35.35 | ccherrett | but no go |
| 05:36.53 | ccherrett | polycom phones on the client side |
| 05:37.17 | ccherrett | could it be dtmf issues in thier provisioning? |
| 05:39.07 | ccherrett | <dtmf> |
| 05:39.07 | ccherrett | <voIpProt voIpProt.SIP.dtmfViaSignaling.rfc2976="1" tone.dtmf.viaRtp="0"/> |
| 05:39.08 | ccherrett | </dtmf> |
| 05:39.20 | ccherrett | I think we need to do that to our polycoms |
| 05:39.30 | ccherrett | change to SIP INFO (RFC2976) |
| 05:39.43 | ccherrett | from SIP Inbound (RFC2833) |
| 05:41.10 | [TK]D-Fender | You should leave it at rfc2833 |
| 05:41.13 | [TK]D-Fender | and set * accordingly |
| 05:41.25 | [TK]D-Fender | Which is polycom's default |
| 05:42.20 | ccherrett | set *? |
| 05:42.43 | [TK]D-Fender | * = Asterisk |
| 05:43.51 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 05:44.43 | ccherrett | [TK]D-Fender: sorry I am not following. Are you saying in the dtmf section? |
| 05:44.58 | ccherrett | of sip.cfg on tftp file? |
| 05:45.46 | [TK]D-Fender | both |
| 05:45.53 | [TK]D-Fender | Asterisk dside needs to eb set to match |
| 05:45.57 | [TK]D-Fender | be* |
| 05:46.03 | [TK]D-Fender | gah |
| 05:51.51 | *** join/#asterisk salz212 (~chatzilla@182.185.163.97) |
| 05:55.26 | salz212 | Hello all |
| 05:57.17 | *** join/#asterisk evilman_home (kvirc@78-106-20-180.broadband.corbina.ru) |
| 06:01.29 | ccherrett | [TK]D-Fender: oh man palm on face |
| 06:01.35 | ccherrett | * = Asterisk :) |
| 06:01.43 | ccherrett | I did not know what you were saying :) |
| 06:03.43 | ccherrett | [TK]D-Fender: wow we have it working!! |
| 06:08.35 | [TK]D-Fender | ccherrett: Glad to hear |
| 06:14.08 | ccherrett | [TK]D-Fender: so happy, many weeks of hasles |
| 06:14.39 | ccherrett | [TK]D-Fender: hired a company, $1600 and left me in a state of no phones |
| 06:14.52 | ccherrett | spent the last 3 days setting up vpn and reconfiguring |
| 06:14.59 | ccherrett | [TK]D-Fender: if you want a job done |
| 06:15.50 | [TK]D-Fender | Whoever you hired should ahve had a clue. Unfortunately not everyone lives up to expectations |
| 06:15.54 | ccherrett | is sooo pleased :) |
| 06:16.10 | [TK]D-Fender | Hopefully they got the rest of the provisioning right... |
| 06:16.35 | [TK]D-Fender | Proper use of line-keys, labels, MicroBrowser, paging, etc |
| 06:16.44 | ccherrett | [TK]D-Fender: it went from one solution to the next with them |
| 06:16.50 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.34) |
| 06:16.59 | ccherrett | watched a tech in screen set nat=yes nat=no for hours |
| 06:17.08 | [TK]D-Fender | ...sad |
| 06:17.08 | ccherrett | sip reload |
| 06:17.11 | ccherrett | restart phones |
| 06:17.18 | [TK]D-Fender | They clearly have no clue |
| 06:17.36 | ccherrett | yeah it took 3 of them to finally blow everything up |
| 06:18.01 | ccherrett | I hired them to take the load off me, instead I had to learn asterisk vpn and such :) |
| 06:18.18 | ccherrett | [TK]D-Fender: so glad in the end I have some skills to show for it |
| 06:18.25 | ccherrett | that is a big plus |
| 06:19.08 | ccherrett | [TK]D-Fender: I watched them in screen for a few days. It was enough to get me farmiliar, then I could take it over |
| 06:19.19 | ccherrett | ah........ good day :) |
| 06:50.04 | *** join/#asterisk salz212 (~chatzilla@182.185.163.97) |
| 06:55.43 | *** join/#asterisk salz212 (~chatzilla@182.185.163.97) |
| 07:09.07 | salz212 | HI all, I need a little hint regarding originating a call from h extension, I remember I did this before a year a ago now I don't remember how I achieved that. I am referring to a scenario where remote party hangs and caller still has control of originating another call.. like a Goto statement in h extensions creating a sort of loop... |
| 07:18.44 | *** join/#asterisk Nickinator (~user@14-201-136-222.static.tpgi.com.au) |
| 07:43.42 | ChannelZ | well there is the h Dial() parameter |
| 07:44.22 | ChannelZ | there's also F |
| 07:44.31 | ChannelZ | and g |
| 07:47.15 | salz212 | tried all of them not sure what I am doing wrong. Dial(SIP/999XXXNUmber@TrunkProvider,180,TttTL(3000000:60000:30000)Hh) |
| 07:48.35 | ChannelZ | what's with TttT |
| 07:48.49 | ChannelZ | and I also don't know what you're doing |
| 07:51.09 | salz212 | its somebodies code I am trying to rectify.. okay what I am doing or want to do is... 1) Make a call to any number through IVR by DTMF and then Dial 2) In case remote party hangs up I want sent call to the first step of IVR from h extensions.....3) By doing this it will create a loop so a person can dial as many numbers he want without hanging up(caller) |
| 07:53.26 | ChannelZ | oh and I meant the e parameter not h earlier |
| 07:54.24 | salz212 | ok let me try that . as well |
| 07:57.45 | ChannelZ | I'm not sure who the 'e' flag sends to the h exten depending on who hangs up. Otherwise g I believe should make it continue on to the next priority if the called party hangs up in which case you could make it Goto wherever you wanted, etc. |
| 07:57.58 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:01.11 | salz212 | yes I am doing it write as per my knowledge problem is when the remote party hangs legs a also hangs right away. |
| 08:03.58 | ChannelZ | what version of asterisk |
| 08:05.11 | salz212 | its 11.2.1 also tried on 1.8.X |
| 08:05.28 | salz212 | very similar thing is working on 1.6 |
| 08:06.21 | ChannelZ | pastebin your console output with verbose on like 3, because something is wrong.. it works here |
| 08:06.44 | ChannelZ | dunno if all your duplicated parameters are horking it up |
| 08:07.27 | salz212 | ok |
| 08:08.38 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 08:27.20 | salz212 | ChannelZ: one thing specific what I want is to have the new call or OR IVR to be played or route from h extensions.. not from next line so g extension is not really useful. I had to isolate thing from the business logic as its huge and messy so this is just atlest dial plan for POC... http://pastebin.pk/mSaZcljT |
| 08:34.55 | salz212 | a similar thing works on version 1.6 without g extension.. |
| 08:38.34 | ChannelZ | hmm. Seems like both legs of the call are torn down at the point the remote party hangs up which is why you get the 'failed to write frame' on Playback. |
| 08:39.21 | salz212 | yes and here is 1.6 output http://pastebin.pk/xuPl1wGp |
| 08:40.28 | salz212 | fundamentally it is correct but I don't understand it working in 1.6 from h extensions... |
| 08:42.17 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 08:43.08 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 08:45.00 | ChannelZ | Not sure, there must have been some technical change. |
| 08:46.06 | WIMPy | Did I already meantion this month that I'd like to see some new channelstates? |
| 08:46.42 | ChannelZ | Like what? 'Horny', 'Tired', 'Hungry'? |
| 08:47.00 | WIMPy | disconnected and upbutavailable |
| 08:47.09 | WIMPy | (for pickup) |
| 08:47.47 | salz212 | thats what I am after..never thought Asterisk would give me tough time after years.. by the way thanks ChannelZ I am a bit curious to know you real name its been quite a while you are contributing to this community just wondering .. |
| 08:48.02 | ChannelZ | I am Bob |
| 08:48.20 | ChannelZ | It looks like you'll have to use 'g' |
| 08:48.38 | ChannelZ | Though I'm not sure why you're resistant to it |
| 08:49.49 | salz212 | yes :( main reason is that on h extension I am dumping previous calls data e.g billing it... RTCP stats etc etc. it will break older functionalies |
| 08:53.30 | ChannelZ | Well sorry you'll have to get a more technical answer from someone familiar with the code as to what/why it changed or what to do about it besides re-thinking your dialplan a bit |
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| 10:38.18 | phix | Gang! |
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| 13:26.49 | igcewieling | salz212: you know that the RTCP info is only available for calls which do not have directmedia? |
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| 14:38.49 | davlefou | hi, how can install ilbc codec under asterisk 1.8? |
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| 15:05.09 | francisvgarcia | what's up everyone |
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| 15:12.44 | francisvgarcia | can anyone recommend me a good SIP provider in USA for hosting a toll free number there? I need 24 channels for incoming calls. |
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| 15:23.41 | DynamicFail | Any of these codecs better at handling lossy links where packets may be dropped: G.711, G.723.1, G.726, G.729AB |
| 15:23.41 | DynamicFail | VAD, CNG, AEC, PLC, AJB, AGC |
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| 15:24.48 | WIMPy | You should certainly use a stateless codec if you have packet loss. |
| 15:25.09 | WIMPy | But probably you shouldn't do VOIP if you have an issue with packet loss. |
| 15:25.58 | igcewieling | i doesn rk v we |
| 15:26.26 | WIMPy | xctly |
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| 15:45.48 | v0lZy | hi |
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| 18:56.09 | rahulr92 | Hi. Could someone please help me with a phpagi query. I am using $asm->send_request('Originate', array) originate a call. I need to execute some code only after this call ends (based on user response). But presently the remaining php code is executed in parallel. Any way I can disable that? Thanks in advance. |
| 18:57.35 | HolTech | trying to configure a sipura 3000 to *, fxo is working and getting the call, though in my dialplan i want it to send the call to the sipura's fxs port, which is connected already in sip.conf, but im not clear on what the context should be for the fxs |
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| 19:50.23 | bramgn | hi, i'm experiencing problems with asterisk 1.8.20.1 on freebsd 9.1. The SIP module seems to crash periodically, and all SIP registrations are silently dropped.. |
| 19:50.47 | bramgn | when issuing a sip show registry, it shows the last registration timestamp of the moment it crashed |
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| 20:31.37 | [TK]D-Fender | rahulr92: No. An originated call has no ties to whatever triggered it |
| 20:32.19 | [TK]D-Fender | ralhyou'll have to add your logic the the call processing on that channel to clean up after itself or add an identifying piece of information to the channel so you can have your original process track it's progress |
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| 23:04.41 | Katty | GOOD AFTERNOON CUPCAKES. |
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| 23:49.50 | edong23 | i havent eaten afternoon cupcakes, but im glad you enjoyed them. |
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