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| 04:44.02 | mbuf | is there an analog card available with one FSO and FSX port? |
| 04:44.18 | mbuf | most of the cards are 4-ports and quite expensive |
| 04:50.18 | [TK]D-Fender | Most of those cards are modular and you don't have to get 4 ports set up for it |
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| 04:58.09 | mbuf | [TK]D-Fender, so I pay only for one 1 FXO and 1 FXS cards? |
| 04:58.19 | [TK]D-Fender | modules, yes |
| 04:58.31 | [TK]D-Fender | They don't normally make 2-port cards for analog |
| 04:58.46 | [TK]D-Fender | There are cheap external SIP gateways for things like that ... but not normally cards |
| 04:59.22 | mbuf | [TK]D-Fender, could you give an example of SIP gateways available in the market? |
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| 05:00.11 | mbuf | [TK]D-Fender, I just want to store the call info for signal processing |
| 05:00.20 | mbuf | [TK]D-Fender, for a single line |
| 05:01.45 | [TK]D-Fender | clarify that.... |
| 05:02.47 | mbuf | [TK]D-Fender, I have a telephone line at home that I wish to connect to asterisk and a cordless phone to |
| 05:03.02 | mbuf | [TK]D-Fender, to be able to record call information |
| 05:03.08 | mbuf | [TK]D-Fender, for home use |
| 05:03.50 | [TK]D-Fender | ok, that isn't just "call info" that is " take the call in and have another port for some analog phone(s)" |
| 05:04.02 | [TK]D-Fender | Linksys SPA-3102 |
| 05:04.13 | [TK]D-Fender | for example |
| 05:05.18 | mbuf | [TK]D-Fender, yes, I also want to be able to record the calls |
| 05:08.24 | mbuf | [TK]D-Fender, apart from forwarding the calls from the FXS to the FXO, does this device also record the calls and other meta information? |
| 05:13.21 | [TK]D-Fender | no. That's up to you and Asterisk |
| 05:13.28 | [TK]D-Fender | Asterisk "processes calls" |
| 05:26.38 | ChannelZ | WTF.. my phone has decided to just not send DTMF on one line. |
| 06:09.28 | ChannelZ | Bug: "DTMF set for AVT is not sent reliably while using the |
| 06:09.28 | ChannelZ | speaker for the SPA50x phone." Yeah, that would pretty much be it... |
| 06:15.29 | ChannelZ | ARGH so now that's fixed but now the DTMF while in-call is blasting out the phone at 2X volume |
| 06:15.35 | ChannelZ | curses Cisco |
| 06:28.18 | PHunter | Should I leave DONT_OPTIMIZE, BETTER_BACKTRACES and DEBUG_THREADS on until I experience an issue to capture the backtrace and locks, or just start up, snap and revert it back? |
| 06:29.17 | ChannelZ | Their docs aren't even right. "Local DTMF playback duration in milliseconds. Affects only OOB." Then, one sentence later, "With OOB, the DTMF Playback Length does not matter." SMH |
| 06:29.23 | PHunter | At idle, Asterisk seems tame.. but in 8 hours, this server will be getting hammered, and I have to sleep still. |
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| 07:20.50 | awk | Hi guys, hmm, I have some really old Grandstream that lose registration and do not re-register.. is there a http path I can envoke to reboot the phone or register it? It's easy iif they registered to send a sip notify signal... Any suggestions? |
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| 07:43.25 | awk | never mind, I wrote a script to do some curl magic |
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| 07:57.59 | timmo | hi, I was looking for a way to get the RTP ip/port of an active call within Asterisk. I couldnât find a helpful cli command. Does anyone know if this is possible and if so, how? |
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| 09:16.49 | F-G0z | Hey guys! Sometimes, Asterisk use 100% of the CPU and slows down every daemons of my server. When I watch the log I see many messages like 'chan_iax2.c: Max retries exceeded to host x.x.x.x'. For example, yesterday I got ~3 of these messages per seconds and for different IPs. Do you have any ideas how to solve that? Thanks. :) |
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| 09:32.32 | ChannelZ | yeah firewall your machine off |
| 09:33.11 | ChannelZ | (unless this is some other problem and those were legitimate IPs it was just failing to send traffic to) |
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| 09:45.08 | Milos | Is it possible to configure asterisk so that if the remote IP of the client device connecting to the asterisk server is on a local subnet, then it's ok to ring that device, otherwise don't? |
| 09:45.23 | Milos | because the native android app is too shit to say wifi only |
| 09:51.42 | F-G0z | ChannelZ: There is no reason, the IPs were not blocked by the firewall. In the meantime, my web sites were unreachable, SSH was terribly slow because of the CPU usage. Can't I do something? |
| 09:58.59 | F-G0z | fail2ban is installed on the machine, I will set up it with Asterisk. That's a beginning. |
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| 12:04.49 | Ast001 | Hello is there any way to change agent channel status in queue. ? I have agent which is Busy in queue show command and he is not really busy. He is not even logged in in queue. |
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| 12:05.44 | Ast001 | Why is this happening and is there any way to set it except restart of entire Asterisk ? |
| 12:09.17 | Synthase_ | In AMI, yes |
| 12:10.31 | Ast001 | Can you tell me how Synthase ? |
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| 12:11.12 | Ast001 | Which AMI command to use to achieve this ? |
| 12:12.19 | dan_j | Hi, Is it possible to send 484 Address Incomplete from Asterisk? |
| 12:14.57 | Synthase_ | If they aren't even logged into the queue, remove them from it. |
| 12:16.27 | Ast001 | I tried that to delete that queue_member from queue_member_table realtime mysql table, but when I add that interface again in queue by inserting row in queue_member_table he is busy again in queue show |
| 12:16.59 | Synthase_ | You're using pbx_realtime? |
| 12:17.23 | Ast001 | yes I am using realtime engine and my sip accounts are realtime queues are also realtime and queue_members. |
| 12:18.04 | Ast001 | I tried to prune that sip too but when I load it again he is again busy always busy... |
| 12:18.25 | Synthase_ | I avoid that wherever possible, so I'm afraid your issue has far more variables than I can account for. |
| 12:19.11 | Ast001 | I even tried to reload app_queue.so as module but after reload everything was same (busy again). |
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| 12:19.37 | Ast001 | I tried to delete agent from agents.conf too but it seems only restart helps. |
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| 12:20.19 | Ast001 | can not get rid of this asterisk -rx "core show channels" | grep 613 |
| 12:20.19 | Ast001 | Agent/613 613@queue_login_613: Down (None) |
| 12:22.25 | dan_j | Can anyone help me explain whats going on here? http://pastebin.com/DAxawkF9 |
| 12:22.38 | dan_j | The dialplan is simply exten => 558,1,Hangup(484 Address incomplete) |
| 12:22.44 | dan_j | I've also tried exten => 558,1,Hangup(484) |
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| 12:23.05 | dan_j | But the SIP response shows SIP/2.0 603 Declined |
| 12:23.13 | Synthase_ | Ast001: queue remove member SIP/foo from queuename. In AMI, QueueRemove action. |
| 12:23.29 | [TK]D-Fender | dan_j: You don't get to thrown arbitrary ggarbage text in there and extect a SIP packet to report it accordingly |
| 12:23.30 | Ast001 | ok I will try |
| 12:23.48 | Synthase_ | Should ass, first line is for ACLI |
| 12:23.50 | Synthase_ | *add |
| 12:24.10 | dan_j | [TK]D-Fender: I've also tried 484 by itself. |
| 12:24.25 | [TK]D-Fender | dan_j: Show said that was for a SIP status number at all? |
| 12:24.37 | [TK]D-Fender | dan_j: You are making assumption after assumption. |
| 12:24.59 | dan_j | [TK]D-Fender: Whats the causecode mentioned in the wiki? https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Hangup |
| 12:25.50 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause |
| 12:26.08 | [TK]D-Fender | define AST_CAUSE_INCOMPATIBLE_DESTINATION 88 |
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| 12:26.20 | [TK]D-Fender | oops, bad aim |
| 12:26.39 | [TK]D-Fender | 28 address incomplete 484 Address incomplete |
| 12:27.00 | [TK]D-Fender | Actually, not sure you can pass that forward... |
| 12:27.05 | [TK]D-Fender | Go drill the page |
| 12:27.52 | dan_j | Thanks. That works but it looks like the sip phone i'm using doesnt treat it properly. It just hangs up and doesnt ask for more digits. |
| 12:28.26 | dan_j | Thanks for your help. |
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| 12:36.11 | Ast001 | Synthase I get "Unable to remove interface from queue no such queue etc..." I guess it is because queue is not in queues.conf, it is realtime queue and that command seems not to work with realtime queues. |
| 12:37.06 | Ast001 | Anyway I restarted asterisk (/etc/init.d/asterisk stop did not kill it so I kill -9 pid and started and now it is ok. I would like to know reason for this and to solve this without restart if can. I will try to investigate. |
| 12:37.57 | [TK]D-Fender | chan_agent is dead and shouldn't be used anyway |
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| 12:39.55 | Ast001 | I must use it because if I use SIP/888 instead of Agent/888 then incoming call will ring on that sip and caller will hear 3-5 seconds of ringing even if queue member answer answer in second. At least it was the case when I tried some calls thatway. |
| 12:40.50 | Ast001 | and due the fact that I am calling pepole it would not be nice if they hear ringing after answering call and then wait to someone to answer on my side. |
| 12:41.53 | [TK]D-Fender | You've got some other issue you should be loking at then |
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| 12:46.33 | Ast001 | I am not sure I am following you. Are you saying if I put SIP/xxx i queue, then Originate call to someone and when he pick up phone his call connect to context and extension so it enters in Queue he should hear my sip agent wihout any dalay ringing etc ? |
| 12:47.06 | Ast001 | and if not so I have some issue to investigate ? |
| 12:50.11 | [TK]D-Fender | Queues normally play MoH adnd only cut once a member actually answers |
| 12:50.20 | [TK]D-Fender | You aren't being clear on WHEN this delay is happening |
| 12:50.36 | [TK]D-Fender | You are making it sound like it called out.. they DO answer.. the agent talks.. but no audio for 5 seconds |
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| 12:54.26 | Ast001 | no when I use agents caller enters the queue and he is connected to agent at the moment and they talk without problems. When I use SIP/xxx client goes to queue and then hear Ringing until sip agent answered and even if he auto answered 2-3 delay will occure. |
| 12:54.59 | Ast001 | That is why I use AgentLogin and agent channels |
| 12:55.41 | *** part/#asterisk Ast001 (~uros@82.117.198.142) |
| 12:56.37 | WIMPy | dan_j: Use Incomplete() |
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| 12:59.15 | [TK]D-Fender | astSo you have it so your agents don't have to confirm calls? |
| 12:59.26 | [TK]D-Fender | ... And he's gone |
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| 13:03.19 | Ast001 | If I missed some message sorry my client for irc crashed. |
| 13:05.41 | [TK]D-Fender | [08:59][TK]D-FenderastSo you have it so your agents don't have to confirm calls? |
| 13:06.35 | Ast001 | I call client client enters Queue and queue member is Agent/xxx and client is right away connected to Agent/xxx if there are any free agents. |
| 13:06.40 | [TK]D-Fender | And they literally sit in QueueLogin() all the time? |
| 13:07.19 | Ast001 | Agents are sitting in AgentLogin in queue and wait for call. |
| 13:08.01 | [TK]D-Fender | ew.... |
| 13:08.03 | Ast001 | which is maybe 30 seonds or 60 seonds |
| 13:08.12 | Ast001 | because clients comes in waves |
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| 13:16.10 | jepperl | today i suddenly started getting this error on a setup that has been working for quite some time now.. right off the bat, without any further details, does anyone know if something has happended the last couple of days (update to libs, chrome etc.) that might have provoked this error: [2015-04-17 15:10:26] WARNING[31774][C-00000015]: res_rtp_asterisk.c:2071 __rtp_recvfrom: PJ ICE Rx error status code: 37 |
| 13:16.16 | jepperl | 0401 'Unauthorized'. |
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| 13:24.52 | dan_j | WIMPy: Thanks. Same result with the sip phone. Looks like im going to sort out something else. |
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| 13:45.11 | WIMPy | dan_j: Did you enable overlap in sip.conf? Do you use a phone that can do it? |
| 13:47.15 | dan_j | WIMPy: overlap? |
| 13:47.49 | dan_j | No idea if the phone supports it. Its a decent phone but wouldnt be surprised if it didnt. |
| 13:47.55 | dan_j | Panasonic KX-UT133 |
| 13:47.57 | WIMPy | dan_j: allowoverlap=yes |
| 13:48.54 | dan_j | I dont have that field in a realtime db. I'll add it over the weekend and try it out. |
| 13:49.02 | dan_j | a = my |
| 13:49.13 | WIMPy | I think it's a very rare feature. I wouldn't be surprised if the Snoms are the only ones supporting it. |
| 13:49.39 | dan_j | Snoms have some great features but they really look ugly :-) |
| 13:50.09 | WIMPy | But it still needs to be enabled on the phone as well. |
| 13:50.32 | *** part/#asterisk Ast001 (~uros@82.117.198.142) |
| 14:05.50 | dan_j | I cant find anything in the documentation but it might be called something else. Thanks for your help. |
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| 14:18.43 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:30.01 | *** join/#asterisk tompaw (~tompaw@tompaw.xxx) |
| 14:30.04 | tompaw | Hello |
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| 14:30.35 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:31.06 | tompaw | I am building a video wall that will show a current state of a call centre agents. What would be the best approach to achieving an almost real-time update rate? |
| 14:31.39 | tompaw | I was thinking corosync + websockets, but there doesn't seem to be any products that would marry these two together. |
| 14:31.59 | WIMPy | Why just "almost"? |
| 14:32.00 | tompaw | Previously I was simply quering Asterisk every x seconds, but it seems kinda... ancient. |
| 14:32.18 | WIMPy | Use AMI. |
| 14:32.26 | *** join/#asterisk Cust0sL1men (~CustosLim@unaffiliated/cust0slim3n) |
| 14:32.58 | tompaw | WIMPy: you mean like a permanent session that I listen to? |
| 14:33.01 | tompaw | and capture events? |
| 14:33.11 | WIMPy | Yes |
| 14:33.49 | tompaw | Hm... that should be doable with nodejs + socket.io then. |
| 14:34.23 | tompaw | WIMPy: I always felt like the AMI sessions are not meant to be permanent - more like open, do something, close. |
| 14:34.44 | WIMPy | No. Definitely not. |
| 14:35.37 | tompaw | all right, thanks. |
| 14:35.41 | [TK]D-Fender | Ton's of 23rd party panel system have used it for more than a decade. |
| 14:35.49 | [TK]D-Fender | Including FOP, HUDlite, iSymphony, etc |
| 14:36.03 | [TK]D-Fender | (24th party stuff is the shiznit) |
| 14:36.47 | tompaw | So I don't need to bother with stuff like "keyframe-style refreshes", when on top of that permanent session I do a global state pull every minute or so? |
| 14:38.17 | tompaw | Also, damn, it's only been 3 years since we built our call centre dialplan based on 1.8 and now I'm going through 13 docs and it seems like it's hight time we rebuilt all from scratch. |
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| 14:49.02 | marceloamorim | I don't get yet, I register an iax trunk on my pbx and my client pbx, when I do iax2 show peers |
| 14:49.48 | marceloamorim | everything is fine, but when I call I'm getting from chan_iax2.so host x failed to authenticate as user |
| 15:18.39 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-sndrksjubgeamknq) |
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| 15:37.09 | nighty^ | 192.168.0.152 6c3e594c167 00:53:30 0000000160K 0000000004 ( 0.00%) 1.0000 0000000160K 0016777215 (10453.03%) 0.0010 |
| 15:37.17 | nighty^ | nobody notices something wront ? |
| 15:37.20 | nighty^ | nobody notices something wrong ? |
| 15:37.25 | nighty^ | <PROTECTED> |
| 15:37.35 | nighty^ | sip show channelstats |
| 15:37.45 | nighty^ | Asterisk 11.17.1 |
| 15:38.21 | nighty^ | Fix, yes please :) |
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| 16:01.08 | WIMPy | Hmm. There is an 11.17.1, but no 11.17.0? |
| 16:11.41 | *** join/#asterisk superscrat (asanders@nat/digium/x-fqnngmxahkwnkkro) |
| 16:15.30 | nighty^ | no there is 11.17.0 too |
| 16:16.46 | WIMPy | I only see an RC of 11.17.0. |
| 16:24.42 | Qwell | http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-11.17.0.tar.gz |
| 16:24.45 | Qwell | totally exists |
| 16:25.26 | nighty^ | so said I :) |
| 16:25.50 | nighty^ | meanwhile channelstats % are totally wacky |
| 16:26.02 | nighty^ | 192.168.0.148 f8815589-20 01:49:10 0000000327K 0000000007 ( 0.00%) 2.0000 0000000327K 0016777215 (5116.83%) 0.0029 |
| 16:26.12 | nighty^ | it is funny at first |
| 16:26.26 | nighty^ | now it means I have to go fix the code :( |
| 16:26.28 | nighty^ | less funny |
| 16:26.37 | nighty^ | some printf format is wrong |
| 16:26.40 | nighty^ | or something like that |
| 16:26.56 | WIMPy | Hmm. Yes. Strange selection in .. then. |
| 16:28.07 | nighty^ | uhmm bug reporting where ? |
| 16:28.28 | Qwell | issues.asterisk.org |
| 16:29.06 | nighty^ | https://issues.asterisk.org/jira/secure/Dashboard.jspa |
| 16:29.08 | nighty^ | ok |
| 16:29.26 | nighty^ | need to register |
| 16:29.34 | nighty^ | oh my |
| 16:31.21 | nighty^ | uhmmm still trying to find out how to report the bug in there |
| 16:31.31 | nighty^ | it is going beyond my patience threshold |
| 16:32.23 | nighty^ | ok got it |
| 16:32.32 | Qwell | click "Create" |
| 16:32.46 | Qwell | Create issue, even |
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| 16:35.50 | nighty^ | https://issues.asterisk.org/jira/browse/ASTERISK-24973 |
| 16:35.58 | nighty^ | baammm! |
| 16:36.01 | nighty^ | created |
| 16:36.15 | nighty^ | deal with it! |
| 16:36.19 | nighty^ | :) |
| 16:37.01 | nighty^ | somebody triage the shit of it please ! :) |
| 16:37.16 | nighty^ | lol |
| 16:37.21 | nighty^ | severity level Major |
| 16:37.24 | nighty^ | I did not set that |
| 16:37.28 | nighty^ | I swear :) |
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| 16:51.46 | nighty^ | lol |
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| 16:52.01 | nighty^ | Jason is quick to tackle down the Level to minor :) |
| 16:52.18 | Qwell | tbh, I was just testing whether I had access to do that still :p |
| 16:53.15 | file | Qwell, I don't see why you wouldn't have access... :P |
| 16:53.17 | file | to that, at least |
| 16:53.53 | Qwell | shrugs |
| 16:53.55 | nighty^ | Qwell: :) |
| 16:54.29 | file | revokes Qwell's access to reality |
| 16:54.42 | Qwell | That's okay, I never use it anyways. |
| 16:54.56 | nighty^ | reality ? |
| 16:55.15 | nighty^ | man reality uses ME everyday ! |
| 17:05.54 | mjordan | uhm |
| 17:06.01 | [TK]D-Fender | Reality is for people who can't handle drugs... |
| 17:06.14 | mjordan | nighty^: I'm not sure what that issue is. Actually providing a description of the problem would be nice. |
| 17:06.16 | mjordan | As would actual logs |
| 17:07.20 | mjordan | particularly for the channels that are reporting an unreasonably high lost count |
| 17:07.35 | mjordan | a large part of that calculation comes from the RTCP statistics received from the *other* side |
| 17:07.55 | mjordan | so if it is reporting to Asterisk "Hey, I haven't received 1012937 packets", guess what we will show? |
| 17:08.54 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines |
| 17:09.34 | ChannelZ | Did you perchance ChanSpy on either of those channels? |
| 17:12.43 | tompaw | is res_pjsip ready for production involving transfers and confbridges? |
| 17:15.05 | PHunter | Heres life for you.. Enabled Compiler Flags for Backtraces, Disable Optimization and Debug Threads.. issue doesn't show.. |
| 17:15.21 | PHunter | takes a nap at desk. |
| 17:17.55 | *** join/#asterisk russellb (~russellb@redhat/russellb) |
| 17:17.55 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:19.17 | nighty^ | 13:06 <@mjordan> nighty^: I'm not sure what that issue is. Actually providing a description of the problem would be nice. |
| 17:19.28 | nighty^ | run traffic for more than 52 minutes |
| 17:19.38 | nighty^ | and you will see this |
| 17:19.48 | nighty^ | I just updated the ticket |
| 17:19.52 | nighty^ | with this comment |
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| 17:22.09 | mjordan | that's not helpful |
| 17:22.32 | mjordan | nighty^: You need to provide specific instructions for someone to reproduce your problem. |
| 17:22.39 | mjordan | Include your sip.conf, extensions.conf, as well as logs |
| 17:22.48 | mjordan | Go read the issue guidelines I linked |
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| 17:54.32 | nighty^ | mjordan: yeah if you do not want to grep that string in the code and fix the var type |
| 17:54.35 | nighty^ | suit yourself |
| 17:54.54 | nighty^ | mjordan: I am not spending time on this shit |
| 17:55.00 | nighty^ | spent enough time reporting it |
| 17:55.14 | nighty^ | personally I do not really care |
| 17:57.43 | dlynes | Qwell, do you mind if I /msg you for a second? |
| 17:59.06 | dlynes | russellb, So are you heading up the packaging and documentation for Asterisk on Redhat now? I see you're not at Asterisk anymore. |
| 17:59.31 | dlynes | Sorry...not at Digium anymore... |
| 17:59.34 | nighty^ | mjordan: and by the way there is nothing in the logs |
| 18:00.10 | russellb | dlynes: no, not doing much asterisk these days |
| 18:00.20 | russellb | dlynes: I work mostly on OpenStack |
| 18:00.32 | dlynes | russellb, Oh...the backbone of Xenware, right? |
| 18:00.40 | russellb | not exactly |
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| 18:01.16 | russellb | lunch & |
| 18:01.40 | dlynes | ah...thought it was one of the main connecting tissues of Citrix Xenware |
| 18:05.23 | dlynes | russellb, Well, congratulations on the new job. I hope you enjoy it. :) |
| 18:05.53 | dlynes | russellb, I'm sure everyone's going to miss your hand in the documentation on Asterisk |
| 18:18.53 | *** join/#asterisk mutilator (muti@c-68-56-137-33.hsd1.mi.comcast.net) |
| 18:19.50 | mutilator | hey guys, have asterisk setup on this network, someone vpns in, signs into their softphone and makes a call, calls go both directions but audio only goes from softphone -> called party not the other way around |
| 18:20.02 | mutilator | i imagine this is a common issue that i have to flick a switch to fix? |
| 18:20.58 | [TK]D-Fender | There are no switches. |
| 18:21.11 | [TK]D-Fender | Go look where audio was negotiated to go and where things are pointing. |
| 18:21.40 | robmal | Allow udp 10000-20000 on firewall. |
| 18:22.44 | mutilator | i've got the whole call setup but i cant tell what exactly its saying, what i'm seeing it's saying is user@their.lan.ip NAT to vpn.network.ip |
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| 18:23.29 | mutilator | robmal: no firewalls between client and server |
| 18:24.28 | robmal | What kind of vpn? |
| 18:24.43 | mutilator | openvpn, split-tunnel |
| 18:25.14 | robmal | tcpdump in the middle, check if the udp packets are coming from both ways. |
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| 21:41.34 | beardy | Can someone recommend some good (cheap and no nonsense) SIP-trunk providers with numbers in Texas, US? |
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| 21:46.41 | [TK]D-Fender | ~itsplist-us |
| 21:46.41 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.broadvoice.com, http://www.jnctn.com, http://www.sipstation.com, http://vitelity.net, http://voip.ms and http://flowroute.com |
| 21:46.47 | [TK]D-Fender | ^ |
| 21:49.21 | robmal | Oh, cute. |
| 21:49.28 | robmal | ~itsplist-pl |
| 21:49.36 | robmal | Meh, as always. |
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| 21:55.10 | beardy | Thanks [TK]D-Fender. |
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| 21:56.53 | Katty | ALLAH"S PEANUT BUTTER SANDWICHES. |
| 22:00.15 | beardy | I'll have one of those. |
| 22:00.34 | Katty | i hear they're quite tasty with jam. |
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| 22:07.07 | beardy | How's miss Katty today then? |
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| 22:16.24 | file | moo |
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