| 00:00.36 | wasanzy | http://pastebin.com/ikDQ5FZf |
| 00:04.16 | wasanzy | What I want is, the dial takes the user to the conference, then after x ms, the user is removed from the conference and put back again after the external application finished running successfully |
| 00:07.52 | ChannelZ | well you have a lot going on I can't test with these CURL requests, but what part is not happening the way you want? |
| 00:08.30 | wasanzy | The t extension is not being called |
| 00:08.54 | wasanzy | so the call just drops after 5 secs and that is all |
| 00:10.06 | ChannelZ | well nothing is timing out |
| 00:10.31 | wasanzy | right so the t is just for timing out. |
| 00:10.38 | ChannelZ | You have to have been waiting for something to be dialed for it to be able to time out |
| 00:11.05 | ChannelZ | If autofallthrough is on, your dialplan hits runs out of things to do after the Playback and hangs up. |
| 00:11.44 | wasanzy | so should I turn off autofallthrough? |
| 00:12.32 | ChannelZ | I don't know? Depends if that makes sense for your entire dialplan |
| 00:12.50 | ChannelZ | If you're expecting them to enter something though, you should probably actually do that.. with a WaitExten or something |
| 00:16.10 | wasanzy | am just thinking the wait you mentioned is the way to go. but where to put the wait is the problem now |
| 00:17.05 | ChannelZ | presumably after your Playback on line 27 |
| 00:18.30 | ChannelZ | You dial local/join_conf@voicemenu-menu_level0 with a timeout, and the 'g' flag says 'continue on with the dialplan after the dial exits'. Then you Playback your BILLED_STATUS_CHECK_NOTIFICATION bit, whatever that is, and presumably are then wanting the user to do something. |
| 00:18.55 | ChannelZ | so you WaitExten. And if they don't do anything, the t extension will execute. |
| 00:19.22 | ChannelZ | I guess this is what you want to happen anyway? |
| 00:22.25 | ChannelZ | shrugs |
| 00:23.49 | wasanzy | the line 27 only get called when line 24 is false |
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| 00:28.47 | ChannelZ | mmnoitdoesn't. It will execute when the Dial times out. |
| 00:28.58 | ChannelZ | (I mean also) |
| 00:32.05 | wasanzy | line 27 or the t extension? |
| 00:33.59 | ChannelZ | line 27. Line 26 is the Dial with the 5 second timeout; once it boots them out of the conference to end the call, it actually continues on to line 27 because you also used the 'g' flag in the Dial which says "continue on with the next priority in the dialplan" |
| 00:34.03 | ChannelZ | otherwise they'd just get hung up on |
| 00:35.07 | ChannelZ | So you need to do something interesting after the Dial, whatever that is. It sounds like you really want it to go to the 't' extension, which is probably not the best place for that since 't' is a specific mechanism. |
| 00:39.10 | wasanzy | wow, it worked |
| 00:39.45 | wasanzy | but not exactly as I expected, but is a head way, I think I need to do dial in the t extension as well. |
| 00:42.39 | wasanzy | thank you so much |
| 00:42.49 | wasanzy | I will let you know in case I need more assistant |
| 00:44.05 | [TK]D-Fender | "t" is not for that |
| 00:44.16 | [TK]D-Fender | "t" is for background / waitexten |
| 00:44.32 | wasanzy | [TK]D-Fender: I don't understand |
| 00:44.55 | [TK]D-Fender | "T" has nothing to dow ith your dial |
| 00:45.02 | [TK]D-Fender | do with* |
| 00:48.21 | ChannelZ | basically I think we're suggesting that the 'hand;er |
| 00:48.23 | ChannelZ | oops |
| 00:50.10 | ChannelZ | the 'handler' you have written currently sitting in the 't' extension, should be some other extension which you can jump to (or make it a macro you GoSub to.) the 't' extension has a specific use, it gets called when Background or WaitExten times out, not when *any* operation times out (which is where I think your confusion lies; the 'timeout' of your Dial is really not a timeout but a call limit) |
| 00:51.05 | [TK]D-Fender | "t" is for IVR INPUT only |
| 00:51.23 | [TK]D-Fender | http://pastebin.com/ikDQ5FZf <- Line 8 |
| 00:53.47 | ChannelZ | then you can call that routine from other places, as it seems you are want to do, including from 't' |
| 00:57.07 | wasanzy | am rewriting it |
| 00:58.12 | wasanzy | exten => t,1,GoTo(1,1) |
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| 01:00.53 | [TK]D-Fender | You have no invalid handler, and timout now leads to "1" |
| 01:00.56 | [TK]D-Fender | What's the point? |
| 01:01.13 | *** join/#asterisk u0m3 (~u0m3@92.80.68.195) |
| 01:01.30 | wasanzy | is working |
| 01:01.45 | [TK]D-Fender | You have a menu where all you can do is NOT hit 0,2-9, 8,# and you'll end up in the same place |
| 01:01.53 | [TK]D-Fender | This is a dumb looking menu |
| 01:02.17 | wasanzy | [TK]D-Fender: am doing something like a loop |
| 01:02.47 | wasanzy | so, the caller is remove every x ms in the conference. |
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| 03:25.37 | Kobaz | hmm |
| 03:25.42 | Kobaz | i've never needed to use 't' |
| 03:26.03 | Kobaz | If the Background or WaitExten times out, it just goes to the next line in the dialplan |
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| 04:43.36 | fling | Ok I have mss and medooze mcu installed |
| 04:43.47 | fling | asterisk is configured to connect it via sip |
| 04:44.03 | fling | everything is running but looks like I need to edit some medooze config files⦠|
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| 04:49.11 | jcims | i've tried three different soft phones for linux, none will connect to a local asterisk install...running wireshark they don't even attempt to connect to port 5060 on the host, is there a lowest common denominator sip test client or something? |
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| 05:45.59 | gavimobile | my tech support is telling me there is no solution for them to solve my clid issue. calls used to come in blocked, so they told me to add a paid header. I added it didn't solve the problem. then they told me to remove the paid header and they changed my routes. now my clid shows a 0 or two 00 before my clid number. they are telling me again that I should add a paid header but its not guarenteed because they aren't doing the final termina |
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| 05:58.55 | ChannelZ | all of those things might be true |
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| 06:15.31 | gavimobile | why can't a static did be set |
| 06:19.21 | [TK]D-Fender | <PROTECTED> |
| 06:25.33 | gavimobile | [TK]D-Fender: appoligies for not being clear, I would like to get information on having better control of my caller id. the following dialplan http://pastebin.com/PSiUU3jM apparently isn't sufficent. |
| 06:26.23 | [TK]D-Fender | <PROTECTED> |
| 06:26.29 | [TK]D-Fender | That's what PEER SETTINS are for |
| 06:26.35 | gavimobile | [TK]D-Fender: how can I eliminate my clid in the following example from showing up with a zero in front of the clid and or eliminate it from being blocked |
| 06:27.06 | gavimobile | [TK]D-Fender: I've tried with and without that, however it doesn't solve my issue |
| 06:27.12 | [TK]D-Fender | sendrpid=pai |
| 06:27.29 | [TK]D-Fender | And I don't see the number that is being sent |
| 06:27.41 | [TK]D-Fender | You show dialplan that references variables |
| 06:27.51 | [TK]D-Fender | Do you know how much I trust the contents of those variables? |
| 06:28.15 | [TK]D-Fender | I give you a hint: it's neither positive nor negative |
| 06:28.20 | gavimobile | lol |
| 06:28.47 | gavimobile | just a sec, ill show my variables in just a momnet, also sendrpid=pai gets added to extentions.conf? |
| 06:28.58 | gavimobile | in [general]? |
| 06:29.17 | [TK]D-Fender | <[TK]D-Fender> That's what PEER SETTINS are for |
| 06:29.37 | gavimobile | [globals]* |
| 06:30.31 | [TK]D-Fender | PEER |
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| 06:40.31 | gavimobile | [TK]D-Fender: here are the variables, http://pastebin.com/3x6KibM3 |
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| 06:41.01 | [TK]D-Fender | And the call? |
| 06:46.43 | gavimobile | just a sec, does sendrpid=yes do the same as sendrpid=pai, cause sendrpid is currently set to yes in my sip.conf |
| 06:47.25 | gavimobile | I mean, ill show my cli with the call |
| 06:48.28 | gavimobile | [TK]D-Fender: im using asterisk 11 btw |
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| 08:54.35 | plosak | hello i have a question |
| 08:54.35 | *** join/#asterisk rox (~rox@BSN-142-168-6.static.dsl.siol.net) |
| 08:55.05 | plosak | i am successfully conencting to my SIP providers SBC with Asterisk 13 but when attemptiong an outbound call i recieve the following |
| 08:55.14 | plosak | -- No one is available to answer at this time (1:0/0/0) |
| 08:55.19 | plosak | and the SIP response |
| 08:55.41 | plosak | SIP/2.0 487 Request Terminated |
| 08:55.46 | plosak | Reason: Q.850;cause=31;text="Normal Unspecified" |
| 08:55.58 | plosak | does anyone know if its something fromt he asterisk configuration or the SBC |
| 08:56.23 | rox | Hello, I have a question regarding queue functionality. When a call is waiting in a queue, calls to queue members are being attempted all the time, except when the periodic announcement message is being played. This means, that if a member becomes free during playing of periodic announcement, the call has to wait for the announcement to end, before being connected. Is there a way to make Asterisk attempt connections to members even during peri |
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| 09:19.55 | rox | i guess i'll have to mix it into MOH |
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| 10:35.09 | tomodachi | im using meetme for meetings, kind of like the features, but read that there is a new meeting application called confbridge now that ive upgraded, does it offer better sound quality?! |
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| 10:54.47 | ChannelZ | tomodachi: yes |
| 10:56.01 | ChannelZ | ConfBridge can work in higher samplerates for g722/wideband conferences, can do noise reduction, silence detection.. |
| 10:56.44 | WIMPy | Custom menus |
| 10:57.37 | WIMPy | And maybe multiple confbridges can even use multiple CPUs. Does anyone know for sure? |
| 10:58.39 | ChannelZ | good question. dunno if they are launched as threads or not |
| 11:00.09 | tomodachi | ChannelZ: hmm interesting ive kind of always thought that peer to peer calls has much higher quality than calling into meetme |
| 11:00.15 | tomodachi | i will give it a shot! |
| 11:00.53 | tomodachi | but i only have g711 phones though... |
| 11:01.04 | ChannelZ | Well yes, you are taking multiple audio streams and making mixes for each party.. line noise adds up |
| 11:01.16 | tomodachi | we only have two parties actually |
| 11:01.27 | ChannelZ | not really a conference then |
| 11:01.30 | tomodachi | conf telephone in one office (with up to 5 people) |
| 11:01.37 | tomodachi | and 5 people in another office , mostly |
| 11:02.12 | tomodachi | yeah I guess not occasionaly some one calls in ,but its not the norm really |
| 11:02.15 | WIMPy | Well, confbridge can be nice even for a normal 1-to-1 call, doing noise reduction. |
| 11:02.24 | ChannelZ | yeah. |
| 11:02.47 | tomodachi | yeah maybe reducing background noise that probably is more apparent could help boost the sound quality |
| 11:03.03 | tomodachi | having one phone in scandinavia and the other in china doesnt really do magic with the sound quality |
| 11:03.30 | WIMPy | Get better phones supporting more than just G.711. |
| 11:03.47 | ChannelZ | well it ain't gonna make crappy speakerphones sound any better |
| 11:04.05 | tomodachi | the conference phones are fairly good (i think) |
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| 11:04.26 | tomodachi | with expanded directional microphones, you think they will be enough if i switch to g722? |
| 11:04.54 | WIMPy | Enough? |
| 11:04.57 | ChannelZ | You'll get more clarity for sure if they support it |
| 11:05.01 | WIMPy | If you can switch to G.722, do it. |
| 11:05.15 | ChannelZ | 8khz is like talking into a soup can |
| 11:05.52 | tomodachi | the phones are actually fully analog |
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| 11:05.58 | tomodachi | with an ata converter connected to each one |
| 11:06.26 | ChannelZ | well.. hard to say then |
| 11:06.27 | tomodachi | so i would only swithc those boxes |
| 11:06.35 | tomodachi | my idea at least |
| 11:06.50 | tomodachi | anything anyone can recomend (not to expensive preferably :) |
| 11:07.44 | WIMPy | I don't know any ATA that does more than G.711 and it probably wouldn't help anyway. |
| 11:08.12 | tomodachi | the analog phones are polycom conf phones, surely expensive whenever they were bought |
| 11:08.37 | WIMPy | And now they are outdated. |
| 11:09.42 | tomodachi | guess so thanks for all of your input ill google around and see what i found |
| 11:09.46 | ChannelZ | Cheapest solution would be get them on a softphone or something |
| 11:10.14 | tomodachi | yeah , thats actually a good way of testing it out initially |
| 11:10.30 | WIMPy | Well, those conference phones is what Polyvom is known for ROW. |
| 11:10.43 | tomodachi | ROW? |
| 11:10.58 | WIMPy | Rest Of the World |
| 11:11.01 | tomodachi | ah |
| 11:11.32 | eirirs | hvile av verdenen |
| 11:12.38 | ChannelZ | bork bork bork |
| 11:20.21 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 11:39.28 | plosak | ello i have a question |
| 11:39.29 | plosak | â rox has joined |
| 11:39.29 | plosak | plosak |
| 11:39.29 | plosak | i am successfully conencting to my SIP providers SBC with Asterisk 13 but when attemptiong an outbound call i recieve the following |
| 11:39.29 | plosak | -- No one is available to answer at this time (1:0/0/0) |
| 11:39.29 | plosak | and the SIP response |
| 11:39.29 | plosak | SIP/2.0 487 Request Terminated |
| 11:39.30 | plosak | Reason: Q.850;cause=31;text="Normal Unspecified" |
| 11:39.30 | plosak | does anyone know if its something fromt he asterisk configuration or the SBC |
| 11:39.31 | plosak | rox |
| 11:39.31 | plosak | Hello, I have a question regarding queue functionality. When a call is waiting in a queue, calls to queue members are being attempted all the time, except when the periodic announcement message is being played. This means, that if a member becomes free during playing of periodic announcement, the call has to wait for the announcement to end, before |
| 11:39.32 | plosak | <PROTECTED> |
| 11:40.58 | rox | plosak: could you switch on SIP debug and drop the SIP communication into pastebin? |
| 11:44.17 | plosak | ok |
| 11:53.08 | plosak | hmm i seem to be getting adifferent response now i will have to check it out |
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| 12:20.33 | plosak | http://pastebin.com/66z1hpSv this bis the communication |
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| 12:43.58 | babak | hi, is there any link to TeleYapper or similar scripts ? |
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| 13:34.55 | [sr] | anyone has an yeastar voip gsm? |
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| 14:00.19 | pzn | where can I find documentation about "manager cli events" flow control of calls and differences between asterisk versions? |
| 14:01.43 | pzn | i've seen one asterisk sending "unlink" and other sending "bridge" to the same thing... |
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| 14:37.42 | plosak | http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/index.html try here |
| 14:42.03 | [TK]D-Fender | That link doesn't help him.... |
| 14:43.14 | jacobkiers | pzn: AMI has changed significantly in the past versions. Which versions are you talking about? |
| 14:44.31 | jacobkiers | And more in general: you can find changes between versions on the Asterisk WIKI, especially in the âNew in <version>â sections: https://wiki.asterisk.org/wiki/display/AST/Home |
| 14:46.21 | jacobkiers | Finally, the exact order of AMI events for specific channels differs slightly based on whether the call is answered (or not) (and in the case of ISDN also the cause code / hangup code). |
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| 15:32.47 | Kobaz | merry merry |
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| 16:16.02 | Kobaz | that wasn't too bad |
| 16:16.11 | Kobaz | my ~400 line Fail2Ban clone that actually works |
| 16:16.31 | Kobaz | could make it a little smaller, i do have some extra bits |
| 16:26.56 | tzica | Hello, I have the following setup http://justpaste.it/i6ir and it the begining I want to place a call from PBX A1 through SIP provider |
| 16:27.37 | tzica | what to do to make it happen |
| 16:28.14 | [TK]D-Fender | tzica: "core show application dial" |
| 16:29.29 | tzica | I have started to add an outbound route but I'm receiving an error |
| 16:29.38 | tzica | No matching endpoint found |
| 16:30.23 | Kobaz | it'll be helpful to actually paste your log too |
| 16:31.54 | [TK]D-Fender | tzica: ... |
| 16:31.57 | [TK]D-Fender | ~freepbx |
| 16:31.57 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:31.59 | [TK]D-Fender | ^^^ |
| 16:53.02 | pzn | jacobkiers, I'm using asterisk from ubuntu packages. talking about differences between 11.7 and 11.11, the environment is the same, I just upgraded the ubuntu release and the AMI seems to be finishing calls in a different way, to my AMI event analyser does not work very well... |
| 16:53.22 | pzn | jacobkiers, thanks for your hint, I'll read "new in version" section of docs |
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| 17:03.45 | _pll | Hello, I recently updated my Asterisk server and found that my CDR table was being populated with the unbridged ringings in a Queue instead of the previous behavior, which was only the bridged call that was added to the CDR, is there a way to prevent this? I looked through the configs in stasis.conf but it's not obvious at first glance. |
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| 17:05.36 | [TK]D-Fender | cdr.conf <- |
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| 17:09.29 | _pll | unanswered=yes ; I'm guessing this one is affecting me. I had it on yes for previous versions and it didn't affect Queues. |
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| 17:17.21 | [TK]D-Fender | Possibly due to what is being called (local channels vs direct device), etc |
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| 18:06.12 | _pll | even with unanswered=no I'm getting the whole batch of calls made to all the members in the queue. |
| 18:06.32 | _pll | I restarted Asterisk after changing it by the way. |
| 18:07.19 | _pll | It's not even consistant, only one of the calls will have the dchannel and dst fields but it's random among the no answer and answered. |
| 18:09.17 | _pll | same with duration and billsec, which only the answered call will have but that row won't have the dst and dstcontext most of the time. |
| 18:18.42 | pabelanger | q: has anybody created an analog interface for the raspi for asterisk? |
| 18:18.49 | pabelanger | via USB or something else? |
| 18:26.09 | _pll | core show channels displays location None. |
| 18:26.21 | _pll | whenever they go to the queue. |
| 18:26.42 | [TK]D-Fender | pabelanger: http://www.sangoma.com/products/usbfxo/ |
| 18:27.33 | coppice | Interesting. I thought they had stopped making that |
| 18:29.38 | pabelanger | [TK]D-Fender: thanks |
| 18:31.25 | _pll | I guess I can't use Asterisk 13 yet. Back to Asterisk 11. My only gripe is that GotoIfTime destroys dst,dchannel vars in this version even if I use gosub everywhere. |
| 18:32.09 | _pll | Or am I using it wrong? In Asterisk 13 it works just fine. |
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| 19:30.35 | child | what should I do to have a land line number on use on my voip ? |
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| 19:31.13 | pabelanger | ~itsp-us |
| 19:31.16 | ChannelZ | Buy one |
| 19:31.31 | pabelanger | ~itsp |
| 19:31.31 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 19:31.33 | ChannelZ | Unless you mean an actual land LINE (POTS service) |
| 19:31.44 | pabelanger | ~itsplist-us |
| 19:31.44 | 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 |
| 19:31.57 | child | yeah but how it works? what should I know? to answer the phone calls online to my land number:? |
| 19:32.20 | ChannelZ | They come in as SIP calls like anything else |
| 19:32.53 | ChannelZ | If you have an existing number you want to move to voip, in most places you can port the number from your current telco over to the provider |
| 19:34.14 | child | I dont have a number.. so to have it I should contact the provider any way? ChannelZ |
| 19:34.19 | ChannelZ | If you are indeed in the US, several places like voip.ms and Vitelity are completely "self-service", you can login, buy a DID (a phone number) from a variety of locations, and start receiving calls right now. On thanksgiving! |
| 19:34.39 | doop | The joy of software robots |
| 19:35.10 | ChannelZ | indeed |
| 19:35.11 | child | no Im in Brazil... and the provider dont offer this kind of service |
| 19:36.09 | doop | child: do you have to have a brazilian number? |
| 19:36.25 | child | so my question is if is there a way to use a land line as voip using a hardware on computer or dont know how |
| 19:36.26 | ChannelZ | hmm I don't know about ITSPs in Brazil. |
| 19:36.46 | child | yes doop |
| 19:37.04 | ChannelZ | If you mean an existing analog phone line, you can get it into asterisk yes with an ATA |
| 19:37.38 | child | I want people to call in my land line number, but I just answer in my voip service on mobile phone |
| 19:39.14 | doop | child: in which city are you |
| 19:39.21 | doop | Heres one that offers numbers in manaus |
| 19:39.24 | doop | https://www.voztovoice.net/index.php/en/component/didww/coverage/Brazil/Manaus/55-92/ITSP/Svanto?provider=Svanto |
| 19:39.42 | child | ChannelZ: I'm in Minas Gerais state |
| 19:39.52 | child | too far from Manaus |
| 19:40.05 | doop | Which city child |
| 19:40.10 | child | uberlandia |
| 19:40.30 | doop | Not helpful |
| 19:40.34 | ChannelZ | Assuming your mobile carrier allows 3rd party voip on their data network (some get grumpy about it in the US) yes, you could go from your existing phone line -> ATA -> Asterisk and then forward that call over the internet to your mobile |
| 19:41.21 | child | doop: is Uberlândia the name of my city |
| 19:41.25 | child | L) |
| 19:41.29 | child | :) |
| 19:41.49 | doop | child: in english it seriously sounds like a made up fake name |
| 19:41.53 | doop | http://www.voip-catalog.com/voip_cities_sao-paulo_1.html |
| 19:41.56 | child | ChannelZ: so, I will try to read about it. Do I need have some especific hardware? |
| 19:42.11 | doop | Child: theres SP |
| 19:42.12 | child | doop: its not friend |
| 19:42.23 | doop | Looking now for stuff in minas gerais |
| 19:42.35 | child | https://www.google.com.br/maps/place/Uberl%C3%A2ndia+-+MG/data=!4m2!3m1!1s0x94a4450c10bbbaef:0xae370c93616d5c9c?sa=X&ei=oH53VNKyPNOHsQSD84DoDw&ved=0CBwQ8gEwAA |
| 19:42.38 | child | here we are |
| 19:43.33 | child | ChannelZ: could you advice me some hardware if yes? |
| 19:43.37 | ChannelZ | Yes, the ATA.. Analog Telephone Adapter. There are two types of connections, you need an FXO (to connect to a phone line, as opposed to an FXS which is for connecting an analog phone) |
| 19:44.44 | ChannelZ | The Cisco SPA-3102 has one of each for instance. I can't comment on the international support though |
| 19:46.24 | doop | child: check out http://directcall.com.br |
| 19:46.38 | doop | Your portuguese is presumably better than mine |
| 19:47.20 | child | not in my city |
| 19:47.26 | child | haha |
| 19:47.40 | child | I have searched... we dont have it |
| 19:47.45 | child | its a small city |
| 19:48.52 | child | ChannelZ: with that router I dont need any other hardware? just plug in the server and it works with asterisk and linux? |
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| 19:54.44 | ChannelZ | More or less. It's an ethernet device |
| 19:55.20 | ChannelZ | You plug a phone line in one port, ethernet into another, and configure it to send calls to your asterisk |
| 19:57.11 | WIMPy | pabelanger: IIRC Sangoma have something. Or off course xorcom. |
| 19:57.58 | child | ChannelZ: I see... what if I want to sell this service here ? is there a hardware to do that for more than 50 lines for example? |
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| 20:02.31 | ChannelZ | yes |
| 20:11.49 | WIMPy | child: Before you go to look for hardware, you should probably check the legal side. |
| 20:12.50 | child | yeah... WIMPy Im not looking to buy now.. just wondering the possibilities after have it working |
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| 20:14.46 | child | ChannelZ: thank you very much |
| 20:14.55 | child | thank you too doop |
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| 21:30.33 | [TK]D-Fender | packs up to head home |
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