| 00:00.27 | BenC[UK] | http://pastebin.com/3ucVJpy2 |
| 00:00.30 | p3nguin | If I was still using SIP, I would see that you are calling me from 3145551212 because apparently SIP redirect works correctly and retains the original caller id number. |
| 00:01.04 | BenC[UK] | I have this dialplan... I originate a call to it using the AMI, sending in a lead id, the idea then if for my script to iterate through the calls thats been set up for that lead |
| 00:01.20 | BenC[UK] | it makes the first call, it connects, and I get added to the queue |
| 00:01.29 | BenC[UK] | however, something is hitting my hangup event |
| 00:02.31 | BenC[UK] | http://pastebin.com/qKVw3ngw is the output on the cli |
| 00:03.59 | BenC[UK] | when the call goes to the sales queue, I use the caller id to load the customer details on the screen for an agent.. then at the end of the call, the agent can choose a disposition |
| 00:05.06 | BenC[UK] | once the call has finished, I need the script to start again |
| 00:05.20 | BenC[UK] | so I have been trying to use the "h" extension |
| 00:05.22 | p3nguin | I guess I need to learn how to use their bug reporting thingy so I can tell them call forwarding is busted. |
| 00:05.34 | BenC[UK] | but its getting called before the end of the call |
| 00:06.03 | ChannelZ | p3nguin: oh. I didn't know what your setup was. But your outgoing trunk is SIP yes? (or IAX I guess you've said in the past) |
| 00:09.26 | p3nguin | It's IAX2, but that part isn't really important. The difference between working and non-working is SIP on the Cisco phone and SCCP on the Cisco phone. |
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| 00:11.41 | sbrath | Anyone know if it's possible to compile Sangoma wanpipe drivers with dahdi 2.5? |
| 00:13.49 | p3nguin | Even back when I had SIP on the phones, I used an IAX2 trunk. Caller ID on forwarded calls was still intact. |
| 00:28.25 | sbrath | well for anyone who wants to know.... dahdi 2.5.x + sangoma wanpipe (any version) - does now a happy couple make.... had to back up to 2.4.x |
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| 00:29.43 | sbrath | pop quiz. can anyone postulate a reason that a PIAF asterisk 1.6 box + sangoma T1 would have working DTMF on friday but have it stop working on Monday? And since monday the volume on the T1 is low.... |
| 00:30.42 | sbrath | and that as of that monday, also checking voicemail causes asterisk to restart after you put in your voicemail password... |
| 00:36.30 | p3nguin | Other than the fact that PiaF is a piece of trash? Nope, can't really come up with any magical things to make stuff quit working. |
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| 01:23.32 | hesco_home | Trying to build asterisk from source, starting with dadhi, I'm encountering the error described here: http://pastebin.com/VkMfGE4k , which also lays out the strategies I've tried to date. Can anyone offer some advice about how to push past this point? |
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| 01:30.28 | Karen_m | p3nguin, are you there? |
| 01:31.13 | Karen_m | p3nguin, yesterday you mentioned that you would show me your asterisk config for faxes, could i possibly see it? If you provided it, my power went out for a bit, today. |
| 01:37.11 | Karen_m | can someone explain the 'h' extension? |
| 01:37.13 | Karen_m | http://lists.digium.com/pipermail/asterisk-users/2010-January/243730.html |
| 01:37.36 | Karen_m | if you look there, he uses s,n,Hangup(), and then starts using; exten => h,1,NoOp(....)... |
| 01:37.44 | Karen_m | does Hangup() call extension 'h'? |
| 01:37.52 | carrar | ~book |
| 01:37.52 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
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| 02:03.53 | p3nguin | karen_m: Go into VoIP.ms portal to your account settings, and change the Device type seting from ATA/softphone to Asterisk/IP PBX. |
| 02:04.13 | p3nguin | karen_m: Then your phone calls will stop going to 's' when they are supposed to be going to your phone number. |
| 02:06.59 | Karen_m | p3nguin, yes that was brilliant of you guys to figure that out. I actually read that last night and changed it. my /200 now gets ignored lol, but that's great! |
| 02:07.13 | Karen_m | i changed it to the did number today and it's working *great* |
| 02:07.20 | Karen_m | i just am trying to figure out the config for faxes |
| 02:13.53 | p3nguin | karen_m: See if this works for you: http://pastebin.com/6RQV9nEx |
| 02:15.02 | Karen_m | wow that is brilliant! |
| 02:15.36 | p3nguin | I redid the configuration when I changed Asterisk computers. I didn't have fax modules for a while, and now that I installed them again, I haven't had a single successful fax... so I'm not sure how well this new configuratio works. If you have problems, I'll dig out the old config from backups -- one that I did get successful faxes on, so I know it works. |
| 02:16.06 | p3nguin | But I feel like this should work fine. |
| 02:17.34 | p3nguin | Every fax I get now ends in APPERROR. |
| 02:17.45 | p3nguin | When watching sip debug, I see timeouts. |
| 02:18.03 | Karen_m | why do you think that is? |
| 02:18.05 | p3nguin | I haven't put much time into fixing it yet, but I really would like to get it fixed soon. |
| 02:18.08 | Karen_m | is that a flaw with ReceiveFAX? |
| 02:18.28 | p3nguin | It could be a problem with my asterisk version, with res_fax, or with res_fax_digium. |
| 02:18.43 | p3nguin | Those are the things that are different from when it used to work for me. |
| 02:18.52 | Karen_m | p3nguin, did you get in that private club leifmadsen was offering you? |
| 02:19.01 | p3nguin | heh yeah |
| 02:19.12 | p3nguin | You know what it was, right? |
| 02:19.12 | Karen_m | ??? :) |
| 02:19.14 | Karen_m | no |
| 02:19.18 | p3nguin | Google+ |
| 02:19.20 | Karen_m | i am curious |
| 02:19.31 | Karen_m | ohhhhhh lol |
| 02:19.35 | p3nguin | I added him in my circle a while ago, but he hadn't added me back. |
| 02:20.18 | p3nguin | He finally got around to it, and now I'm more special than before. |
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| 02:22.17 | Karen_m | the golden circle, where peasants cannot visit :) |
| 02:22.43 | p3nguin | They can look, but they can't touch. |
| 02:23.04 | Karen_m | lol |
| 02:24.15 | tonsofpcs | lookie, no touchie? |
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| 02:36.01 | p3nguin | karen_m: Here's the old way: http://pastebin.com/RkLWfJSa |
| 02:36.25 | p3nguin | karen_m: I recreated it in case you wanted to see how they differed. |
| 02:36.49 | p3nguin | As you can see, there's no change in the actual reception of the fax. |
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| 02:48.56 | Karen_m | p3nguin, is there a place that you know if, that is trustworthy that sends faxes/test faxes for free? |
| 02:49.04 | Karen_m | kind of like that smtp checker for email |
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| 02:57.26 | ChannelZ | I could probably send you one |
| 02:57.30 | p3nguin | karen_m: There used to be a few, but they've either disappeared or become bothersome. If you'll /notice me your fax number, I'll try sending one shortly. I'll find something to send. |
| 02:57.38 | p3nguin | or channelz will do it. |
| 02:57.41 | p3nguin | Either way. |
| 02:58.46 | ChannelZ | I haven't been paying attention, are you using t.38? |
| 02:59.58 | p3nguin | Grr, I need to eat some supper before it's too darn late for supper. I hate eating too close to bed time, because then I wake up even more hungry than I would have been if I had just not eaten before bed. |
| 03:00.09 | Karen_m | p3nguin, where do you set this? ${DB(email/fax-manager)} |
| 03:00.52 | Karen_m | ChannelZ, whatever regular fax machines use. |
| 03:01.36 | ChannelZ | well but you're using SIP channels right? |
| 03:08.15 | Karen_m | yes, correct |
| 03:08.26 | Karen_m | and i'm using chicago.voip.ms -> chicago gigenet box |
| 03:08.41 | Karen_m | sorry, received a phone call on my newly setup did :) |
| 03:08.43 | Karen_m | this is amazing |
| 03:08.49 | Karen_m | why don't people use voip more!?! lol |
| 03:09.17 | Karen_m | hey p3nguin ,did you go eat yet? |
| 03:09.42 | ChannelZ | did you get a Fax For Asterisk license? |
| 03:09.51 | p3nguin | karen_m: Anything that says DB() is in the AstDB (Asterisk database). |
| 03:10.11 | p3nguin | 1.8 has res_fax built in... does it still need something else with a license? |
| 03:10.15 | Karen_m | I am not using an AstDB yet, so can I just hardcode the value with it? |
| 03:10.23 | p3nguin | You can. |
| 03:10.27 | ChannelZ | Oh, I dunno, maybe it doesn't |
| 03:10.44 | Karen_m | i never purchased a license, i am just using ReceiveFAX and SendFAX kind of thing |
| 03:10.45 | p3nguin | I found a Totino's party pizza. It's on the Pizzazz as we speak. |
| 03:11.00 | Karen_m | is there a way to validate the extensions.conf before I try a dialplan reload? |
| 03:11.19 | p3nguin | If it's trashed, pbx_config should complain. |
| 03:11.34 | Karen_m | ok, that's the only way? i didn't see it on the help dialplan, that's why i wa sasking |
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| 03:11.51 | p3nguin | For the most part, you'll have to do it with your own eyes. |
| 03:12.20 | Karen_m | if I change that $DB{...} to an email, does it have to be escaped? |
| 03:12.24 | Karen_m | the, \@ part? |
| 03:12.46 | Karen_m | karen.m...@gmail.com or .. karen.m...\@gmail.com |
| 03:12.56 | Karen_m | i'm worried about breaking my current running DID's/extensions :) |
| 03:14.23 | Karen_m | no complaints yet! dialplan reloaded :) |
| 03:14.26 | ChannelZ | While the modules are included (well, res_fax is, res_fax_digium isn't) I thought you still had to have a license. You can get 1 free one |
| 03:14.34 | p3nguin | karen.m@gmail.com is fine |
| 03:14.56 | p3nguin | I don't know; that's why I asked. |
| 03:15.09 | p3nguin | I have my fax for asterisk stuff for my 1.4 system. |
| 03:15.36 | p3nguin | Once I get to 1.8, I'll have to learn what I need for my fax. |
| 03:16.51 | p3nguin | karen_m: My fax config has dependency of tiff2pdf and mutt. Make sure you have them. |
| 03:17.23 | p3nguin | libtiff provides tiff2pdf |
| 03:18.20 | p3nguin | I got tired of having TIFFs in my email and not having a decent viewer that I could conveniently use, so my solution was turn 'em into PDFs. |
| 03:18.52 | ChannelZ | p3nguin: I think it pretty much works the same way as before |
| 03:19.21 | p3nguin | Install res_fax from within asterisk, then register res_fax_digium from the web? |
| 03:19.25 | Karen_m | how do you get a free license? |
| 03:19.35 | p3nguin | Buy it on digium.com. |
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| 03:19.55 | p3nguin | http://store.digium.com/productview.php?product_code=804-00007 |
| 03:21.44 | ChannelZ | si |
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| 03:23.10 | Karen_m | nice! |
| 03:23.55 | Karen_m | with sip.conf, what links the [general] to the next block? for example, right now it's [voipms] that always calls context 'voipms-inbound' ... I created a new subaccount, and then a new [block], and that block calls a different context, but it always calls [voipms] |
| 03:24.07 | Karen_m | even though I have my username set to the 123456_fax |
| 03:25.07 | ChannelZ | there is no link |
| 03:25.27 | p3nguin | No link, no association. I already covered this. |
| 03:25.49 | Karen_m | how do I set 123456_fax ( the did routed to this), to use .. fax-in |
| 03:25.54 | p3nguin | You only need one single account for your asterisk system. |
| 03:26.28 | Karen_m | oh you just route it to [mycontext] and then run includes ? |
| 03:26.32 | p3nguin | Once you route the new DID for fax to the only account you're using on your asterisk, I'll show you how to deal with it. |
| 03:26.40 | ChannelZ | Does anyone know what a proper Digium card serial number looks like? I have like 5 different things written down off various stickers and shit from my TDM800 but the website hates all of them |
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| 03:30.04 | p3nguin | karen_m: http://pastebin.com/Piqv4Egj See line 40. |
| 03:30.11 | ChannelZ | argh piece of crap |
| 03:30.25 | p3nguin | karen_m: And then 62-79. |
| 03:32.33 | hesco_home | I'm in the midst of building a new asterisk server. I'm concerned about what impact it will have on my phone service when I have two servers registering to my providers as available to process my inbound phone calls. It won't be forever, but their will be overlap when I'm still testing the new one and before I have turned off the old one. Can anyone comment of best practices for such a situation, please? |
| 03:32.51 | Karen_m | i ruined my sip.conf and now all calls are 403, forbidden :( |
| 03:32.56 | Karen_m | even though it says registered |
| 03:33.15 | ChannelZ | turn one off while you're testing the other |
| 03:33.42 | ChannelZ | registration has little to do with being able to make/receive calls |
| 03:34.13 | hesco_home | so my system will be down during testing I take it. |
| 03:34.29 | Karen_m | WARNING[8382]: chan_sip.c:19339 handle_response_invite: Received response: "Forbidden" from |
| 03:34.42 | ChannelZ | hesco_home: you can only register to an account from one place at a time |
| 03:34.54 | ChannelZ | hesco_home: whichever one registers last is which one your ITSP will know you by |
| 03:34.59 | hesco_home | ok, that is helpful. |
| 03:35.26 | p3nguin | karen_m: You need only one register line for each ITSP, and you need only one peer for that ITSP, too. |
| 03:35.33 | hesco_home | and if they keep updating their registrations, it will ping-pong back and forth between the two, I guess. |
| 03:36.08 | Karen_m | p3nguin, can you show me what your sip.conf looks like without hte passwords of course? |
| 03:36.09 | ChannelZ | hesco_home: actually I take that back.. other SIP proxies and things might be able to handle multiple registrations and do something special like failover, but who knows |
| 03:37.07 | p3nguin | While some ITSPs using SIP proxies allow multiple registrations, assume they don't because asterisk does not allow more than one. The last one to register to asterisk gets the calls. |
| 03:37.37 | hesco_home | but the safest I guess will be to minimize the variables during testing. sounds like a late night process then. |
| 03:38.11 | Karen_m | so i've got multiple register lines under [general] and then only one [voipms] box? |
| 03:38.18 | p3nguin | karen_m: http://pastebin.com/tER2jGnY |
| 03:38.38 | hesco_home | The issue right in front of me at the moment relates to errors for a `make all` while trying to build dadhi. Some idea of what I've tried so far is at: http://pastebin.com/x3W8uga5 |
| 03:38.45 | p3nguin | karen_m: If you have more than one register line for voipms, decide which one you are going to keep and delete the other. |
| 03:38.57 | p3nguin | karen_m: one asterisk box, one account, one register statement. |
| 03:39.21 | Karen_m | oh that's what the problem is then! |
| 03:39.22 | Karen_m | lol |
| 03:39.31 | Karen_m | i had the polycom a few weeks ago, and was seperating things |
| 03:44.22 | p3nguin | After you stop using the sub account that you felt you needed, be sure to go to DID management and route the fax number to your correct account instead of the unneeded sub account. |
| 03:55.13 | Karen_m | darn, failed fax |
| 03:55.15 | Karen_m | WARNING T.30 ECM carrier not found |
| 03:55.31 | Karen_m | says that 3 times, and then Hangup, unknown code 0 |
| 03:55.43 | Karen_m | time i better start googling lol |
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| 03:56.28 | Karen_m | is there a way from asterisk -r, to print out what 'rtpkeepalive' is set to? |
| 03:56.36 | Karen_m | someone commented on that setting in regards to this |
| 03:58.16 | p3nguin | I have i686 VIA Nehemiah CentaurHauls processor... should I be using i686 or generic for modules such as res_fax_digium? |
| 03:58.34 | kaldemar | Karen_m: sip show settings |
| 03:59.03 | p3nguin | The benchmark app chokes and dies, so I can't find out that way. |
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| 05:07.30 | coppice | The Nehemiah is the VIA C3. This is almost but not quite an i686, and can't be relied on to act as one |
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| 05:24.19 | ChannelZ | p3nguin: i686 should be fine |
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| 06:01.51 | brdude | Does, Dial(IAX2/guest@misery.digium.com/s@default) still work? |
| 06:02.06 | brdude | I've been trying and am only getting a busy signal. |
| 06:02.20 | brdude | I've also tried Dial(IAX2/guest@misery.digium.com/s) |
| 06:02.28 | brdude | no luck there either. |
| 06:07.33 | kaldemar | brdude: it works |
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| 06:08.20 | brdude | daldemar, do I have to have the port forwarded to my asterisk box for it to work? |
| 06:08.34 | brdude | kaldemar * |
| 06:09.42 | kaldemar | no, unless your firewall is behaving badly. enable iax2 debug and see if you get any messages back from misery. |
| 06:12.16 | brdude | That's weird i just tried a sip test number and it works fine. |
| 06:15.09 | kaldemar | it has little to do with a test call over iax2 working. |
| 06:17.24 | brdude | kaldemar I was just wondering if it was my firewall |
| 06:18.04 | kaldemar | enable debug and see if it is. |
| 06:20.20 | brdude | kaldemar, I'm on 1.8 and the only comands I can find to turn iax2 debug are "iax2 set debug on" and that doesn't work. |
| 06:20.58 | kaldemar | how does it not work? |
| 06:21.22 | kaldemar | "No such command"? |
| 06:21.29 | brdude | yep |
| 06:21.35 | brdude | says no such command |
| 06:22.03 | kaldemar | then you don't even have the module loaded. what does "module load chan_iax2.so" say? pastebin the result. |
| 06:23.16 | brdude | Unable to load module chan_iax2.so |
| 06:23.17 | brdude | Command 'module load chan_iax2.so' failed. |
| 06:24.15 | kaldemar | "core set verbose 10" and try again. you should get more output. |
| 06:25.22 | brdude | I set verbose to 50 and I'm still only getting those 2 lines |
| 06:26.23 | brdude | now module reload chan_iax2.so works with no warning. |
| 06:30.28 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
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| 06:30.51 | schmidts | good morning |
| 06:32.19 | brdude | morning |
| 06:32.43 | *** join/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it) |
| 06:33.26 | *** join/#asterisk Karen_m (~karen@d50-99-60-236.abhsia.telus.net) |
| 06:33.42 | Karen_m | darn lost connection, missed scrollback buffer for reading/learning :( |
| 06:34.14 | brdude | kaldemar, that's weird when I try to reload chan_sip.so it gives me a message saying it went fine but when i try to reload chan_iax2.so I don't get anything back. |
| 06:34.49 | *** join/#asterisk nix8n82-phone (~AndChat@24.143.28.16) |
| 06:35.36 | brdude | rebooting system to see what happens |
| 06:36.30 | brdude | Karen_m the only thing you missed it me having a problem with IAX2 not working on my system. |
| 06:38.37 | Karen_m | brdude, i think p3guin uses iax2 |
| 06:38.43 | Karen_m | he's brilliant, and helped me a lot |
| 06:38.50 | Karen_m | await his return lol |
| 06:39.25 | kaldemar | brdude: how did you install asterisk? |
| 06:39.42 | Karen_m | brdude, remove iax2 config, reload chan_iax2, put config settings in one at a time between each reload and see which one is broke? |
| 06:40.14 | brdude | kaldemar, I installed from source following the asterisk book 3rd edition. |
| 06:40.50 | kaldemar | did you do a "make samples" after "make install"? have you modified iax.conf? |
| 06:41.02 | brdude | Karen_m, it's not the iax2 config per see. I'm actually trying to dial (IAX2/guest@misery.digium.com/s@default) and I'm having no luck |
| 06:41.51 | brdude | kaldemar, I dind't run 'make samples" but I do have a iax.conf file that I copied from the book |
| 06:41.59 | irroot | schmidts morning see you joined us today :P not playing angry birds ?? |
| 06:42.07 | brdude | here let me pastebin it. |
| 06:42.50 | brdude | Wait, I just rebooted the server and now "module reload chan_iax2.so" works. |
| 06:43.16 | brdude | Still can't make call the digium server though. |
| 06:43.31 | kaldemar | pastebin CLI output for a call. |
| 06:43.32 | brdude | let me see if i can turn debug for iax on now. |
| 06:44.51 | schmidts | irroot hey yesterday it was a national holiday ;) |
| 06:46.21 | brdude | kaldemar, thanks for the help it works now. |
| 06:46.42 | brdude | I had to turn off iptables since I still don't have it configured. |
| 06:47.11 | brdude | But the thing with the chan_iax2.so not reloading is still very weird. |
| 06:47.29 | brdude | Oh well it works now and hopefully will keep working. |
| 06:48.01 | WIMPy | Probably it was loaded, but wit a non-working config. |
| 06:48.51 | brdude | WIMPy, but it wasen't letting me reload it at all. Or at least it wasen't giving me any feed back when I tried to. |
| 06:49.06 | brdude | Couldn't even set it to debug. |
| 06:49.30 | WIMPy | load or reload? |
| 06:50.14 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 06:50.49 | brdude | for load it gave me a message saying it couldn't load it. |
| 06:51.11 | brdude | and when I tried to reload it would not give me any feedback. |
| 06:51.30 | WIMPy | You can't load a module that's already laded. |
| 06:52.13 | WIMPy | And if reload doens't say anything it didn't find errors reloading its config. |
| 06:52.35 | WIMPy | But just because it doesn't have errors, doesn't mean it will work. |
| 06:53.43 | brdude | Yeah, but now it at least tells me that it's realoding. |
| 06:54.25 | brdude | and before if i tried to reload chan_sip.so it would also tell me that, that was being reloaded. |
| 06:54.39 | brdude | Oh well, it works now. |
| 06:54.52 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 06:55.16 | WIMPy | That's a little misleading BTW. It only tells the module to reload its configuration which that module may or may not do. |
| 06:55.30 | WIMPy | If you want to be 100% sure, you have to unload and load. |
| 06:56.54 | brdude | WIMPy, thanks if it starts acting up again I'll unload it and then try to reload. |
| 07:04.27 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 07:06.22 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 07:09.33 | penguin | any php questions |
| 07:09.42 | penguin | immore then willing to helpout |
| 07:10.05 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-139-212.ks.ks.cox.net) |
| 07:10.47 | penguin | any php questions anyone |
| 07:11.01 | penguin | more then happy to answer or help out |
| 07:11.20 | WIMPy | This is #asterisk, not #php |
| 07:12.10 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 07:17.42 | ChannelZ | I will entertain a short PHP question |
| 07:19.48 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 07:23.54 | irroot | broadsoft rep just called .... not intrested in patented crap |
| 07:24.18 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:28.03 | wdoekes2 | s/patented/proprietary/ |
| 07:28.09 | wdoekes2 | good morning, btw |
| 07:31.34 | *** join/#asterisk Disident (~Disident@85.62.99.220) |
| 07:32.43 | *** join/#asterisk oej (~olle@office.ipvision.dk) |
| 07:32.54 | irroot | morning |
| 07:33.26 | WIMPy | Moin irroot |
| 07:33.47 | irroot | yo WIMPy morning been abducted by the busy |
| 07:34.10 | WIMPy | No aliens? |
| 07:38.13 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 07:40.50 | irroot | WIMPy customers count ?? :P |
| 07:48.12 | *** join/#asterisk MarKsaitis (~MarKsaiti@94-30-69-205.xdsl.murphx.net) |
| 07:50.32 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 07:51.45 | *** join/#asterisk Nasga (~Nasga@186.212.10.93.rev.sfr.net) |
| 07:52.08 | schmidts | hey guys, does anyone knows which numer has the Zimbabwe Mobile Telecel or Econet? |
| 07:52.28 | schmidts | i have this numer +26373 and i have to know which one of the above this could be ;) |
| 07:53.10 | WIMPy | Are you sure, they don't know number portability in Zimbabwe? |
| 07:54.23 | schmidts | Wimpy even if they do it, my rating doesnt know it :D |
| 07:55.00 | *** join/#asterisk Guest8383 (~Geek@unaffiliated/cain) |
| 07:56.18 | schmidts | my problem is a customer was hacked over the weekend and they generated several thousands to zimbabwe and sierra leone. i know the price we have rated this calls, but i cant match my purchase rate for this cause i only have a list with names buth without numbers :( |
| 07:56.32 | WIMPy | http://countrycode.org/zimbabwe |
| 07:57.58 | *** join/#asterisk KNERD (~KNERD@adsl-99-72-119-220.dsl.hrlntx.sbcglobal.net) |
| 08:00.10 | schmidts | Wimpy thanks but +263 - 73 isnt in this list :( |
| 08:01.29 | WIMPy | So it's probably a landline? |
| 08:02.12 | schmidts | no i have found several pages where its listed as a mobile number |
| 08:02.31 | irroot | +263 my neighbours :P |
| 08:02.34 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 08:03.16 | *** join/#asterisk k3asd` (~k3asd@static-217-133-87-244.clienti.tiscali.it) |
| 08:03.32 | k3asd` | hi |
| 08:03.40 | schmidts | irroot AFAIK was this one of the newer methods of hacking, you get some carriers and let them pronounce a number range which doesnt exists, hack some systems and let them call this numbers and after a week this number is deleted |
| 08:04.25 | irroot | +26373 is mobile |
| 08:05.37 | irroot | yeah indeed can go smack them up a bit just accross the border :P |
| 08:05.49 | schmidts | irroot thank you very much! do you even know which carrier? |
| 08:06.02 | irroot | looks like econet |
| 08:06.11 | schmidts | FUUUU____ |
| 08:06.25 | irroot | did some work for them when i was in harare 2 years back |
| 08:06.38 | schmidts | i think i am going to kill my product manager now :( |
| 08:07.05 | schmidts | our purchase rate for econet is 0,37?/min but we sell this for 0,26?/min :( |
| 08:07.15 | WIMPy | Do you pay more than you bill your customers? |
| 08:07.32 | WIMPy | BWLer... |
| 08:08.13 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:08.44 | schmidts | Wimpy not my fault, i only do the technical stuff nit the prices |
| 08:08.51 | schmidts | s/nit/not |
| 08:09.22 | WIMPy | sure |
| 08:09.40 | schmidts | :P |
| 08:12.44 | schmidts | irroot looks i have some luck today, its not econet its telecel |
| 08:12.55 | irroot | cool |
| 08:13.05 | irroot | im trying to double check |
| 08:14.08 | irroot | the guys there have more than one cell phone the interconnects dont work or are so expensive its better to phone on net |
| 08:14.35 | KNERD | what is jabber/gtalk equalevent to SIP's "p-asserted-identity "? |
| 08:14.38 | irroot | also different providers have different packages and they take advantage of best one at the right time |
| 08:30.59 | *** join/#asterisk SantiamTech (alex@173-164-96-37-Oregon.hfc.comcastbusiness.net) |
| 08:43.06 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85f2.bcn.adamo.es) |
| 08:49.31 | *** join/#asterisk Poincare (~jefffnode@2001:470:cb24:4::2) |
| 08:57.25 | *** join/#asterisk Polysics (~luca@host210-142-static.228-95-b.business.telecomitalia.it) |
| 08:57.27 | Polysics | hello |
| 08:57.38 | Polysics | time to go back to work on that complicated almost-queue thing |
| 08:58.07 | Polysics | the idea is that i need to be able to hold a caller somewhere while i find someone that will answer to him, then connect them |
| 08:58.21 | Polysics | the receivers are to be chosen using a complicated logic so i cannot just use queues |
| 08:58.37 | Polysics | what would be the ideal course of action, speaking in broad Asterisk functions? |
| 08:58.52 | Polysics | i wll implement that in Adhearsion once i find out the functions i need |
| 08:59.03 | Gugge | start musiconhold |
| 08:59.10 | Gugge | and then do whatever you need to do in the dialplan :) |
| 08:59.27 | Gugge | whenever you have found your destination, dial it |
| 09:00.07 | WIMPy | How does Adhearsion work? Or what is that? |
| 09:00.33 | Polysics | WIMPy, Adhearsion is a great Ruby framework for developing voice (Asterisk mostly) applications |
| 09:00.56 | Polysics | it takes a lot of complexity out of otherwise difficult functions |
| 09:01.21 | WIMPy | That's what the hompage says as well, Not very informative. |
| 09:01.37 | WIMPy | Is it for AGIs? Does it use AMI? Does it generate dialplans? |
| 09:02.01 | Polysics | WIMPy, it is for AGIs, can receive, send and parse AMI events and commands, it does not generate dialplans |
| 09:02.46 | Polysics | i am using it to implement this system where translators get connected to customers, that have bought credit, for services |
| 09:03.00 | WIMPy | Ok, if you doing it AGI, just Dial. |
| 09:03.25 | Polysics | what if the receiving person is busy at the moment? |
| 09:03.26 | WIMPy | If your using AMI, start MOH and redirect the caller to the target. |
| 09:03.53 | Polysics | i basically need to create a "mini-queue" tehre |
| 09:04.05 | WIMPy | Isn't that what you would check in your proggie? |
| 09:04.21 | Polysics | i think i could just slap the caller in a db table, and when the receiver hangs up, he gets dialed again |
| 09:04.43 | WIMPy | Then do it the AMI way and let them sit there until you have a free translator. |
| 09:04.45 | Polysics | WIMPy, yes, but ideally if the receiver is busy i would like to queue the caller in some way |
| 09:04.57 | Polysics | which AMI function am i looking for to do that? |
| 09:05.00 | Polysics | originate? |
| 09:05.05 | WIMPy | there = in MOH. |
| 09:05.17 | WIMPy | redirect |
| 09:05.43 | WIMPy | You just let the dilplan end in MusicOnHold(). |
| 09:06.00 | Polysics | sounds easy enough |
| 09:06.02 | WIMPy | Then when you have a vacant translator, redirect the caller there. |
| 09:06.07 | Polysics | let's try building it .-D thanks! |
| 09:06.26 | Polysics | doesn't redirect require the receiving channel to be up? |
| 09:06.38 | WIMPy | no |
| 09:07.31 | WIMPy | You just have to watch for the event that tells you that someone has reached that MOH extension/priority so you know you have a caller. |
| 09:07.52 | Polysics | one thing i never figured out: how do i get the full documentation for a specific command or event, for the version of * i am using? |
| 09:08.00 | WIMPy | Or you do that in another way, but I think that makes it more complicated. |
| 09:08.32 | WIMPy | core show [application|function] ... , manager show ... |
| 09:09.00 | WIMPy | For events I don't have an answer. |
| 09:10.26 | *** join/#asterisk aglenday (~aglenday@59.167.161.74) |
| 09:10.28 | Polysics | the new wiki looks informative |
| 09:11.05 | Polysics | so, basically Redirect makes the current channel "dial" the extension specified |
| 09:11.16 | WIMPy | yes |
| 09:11.47 | Polysics | in the case i am using some sort of FollowMe functionality, which i have been making in the last few days, how do i join two channels that are already both there? |
| 09:11.59 | WIMPy | Well, you actually have to send it to the dialplan. |
| 09:12.17 | WIMPy | bridge |
| 09:12.24 | Polysics | followme = i call a cellphone, when someone picks up i play "press 1 to accept call", if call is accepted i join the two |
| 09:12.47 | Polysics | if call is not accepted, i call the next cellphone in the list (if available) |
| 09:12.53 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 09:13.12 | WIMPy | So for redirect you need to have an extension that does the dial, but you can set cariables, off course. |
| 09:16.04 | Polysics | i carry all the info i need in the extensions, so i would say i am set for taht |
| 09:24.09 | *** join/#asterisk e-fon_patrick (~patrick@fwj00.e-fon.ch) |
| 09:24.52 | *** join/#asterisk felipe_ (~felipe@unaffiliated/felipe) |
| 09:25.17 | e-fon_patrick | good morning guys, hopefully i joined the right channel ;) may i ask you a question? |
| 09:25.48 | kaldemar | ~ask |
| 09:25.49 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 09:28.05 | e-fon_patrick | thx. I upgraded one of our asterisk nodes from 1.6.1.20 to 1.6.2.18. are there any changes in RT queue handling, as you're aware of? in upgarde.txt are no infos regarding this. |
| 09:29.42 | e-fon_patrick | for ex: a queue member logged in to one queue displays in asterisk 1.6.1.20 as SIP/0435000491 (realtime) (Not in use) and on 1.6.2.18 as SIP/0435000491 (realtime) (Unavailable) on the same database |
| 09:30.41 | e-fon_patrick | any help is really appreciated |
| 09:31.56 | e-fon_patrick | in fact, the 2 servers are really similar, just upgraded one of them. they're connection to the same3 database |
| 09:31.57 | WIMPy | That was discussed on issues.asterisk.org I think. |
| 09:32.30 | e-fon_patrick | thx for the hint WIMPy |
| 09:36.29 | *** join/#asterisk mirko_brankovic (~mirko_bra@212.200.146.253) |
| 09:37.39 | mirko_brankovic | hello |
| 09:38.37 | *** join/#asterisk wonderworld (~ww@port-92-201-212-27.dynamic.qsc.de) |
| 09:39.12 | mirko_brankovic | Anyone has some experiance with ChanSpy() app.? |
| 09:41.00 | wdoekes2 | sure |
| 09:41.04 | wdoekes2 | ~ask |
| 09:41.04 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 09:43.29 | mirko_brankovic | I've installed Ast1.8.5.0 and while spying on channel i can't leave ChanSpy app. even if i put x(0) option |
| 09:46.20 | mirko_brankovic | My question is: Why only x() and c() options of ChanSpy() application is not working for me ? |
| 09:46.39 | kaldemar | does DTMF for for you in any other use? |
| 09:47.38 | mirko_brankovic | yes on # I can change volume (from -4 to 4) as it is default |
| 09:49.37 | mirko_brankovic | I set it up as ChanSpy(,g(SPYGROUP),qSx(0)) and SPYGROUP is set when caller is making a Dial, but x option seems not to be working |
| 09:51.58 | kaldemar | put dtmf option in the console line in logger.conf and do "logger reload" in CLI. then you'll see debug lines whenever asterisk detects dtmf. at least you'll know that dtmf is working properly. |
| 09:53.51 | mirko_brankovic | ok, thanks, I'll try that |
| 10:16.57 | *** join/#asterisk debuggerboy (~debuggerb@117.202.115.251) |
| 10:21.17 | *** join/#asterisk bintut (~bintut@cm158.kappa8.maxonline.com.sg) |
| 10:32.32 | mirko_brankovic | [Aug 16 12:20:29] DTMF[20369] channel.c: DTMF begin '0' received on SIP/... |
| 10:32.34 | mirko_brankovic | [Aug 16 12:20:29] DTMF[20369] channel.c: DTMF begin ignored '0' on SIP/... |
| 10:33.30 | mirko_brankovic | I guess this means dtmf is working fine, maybe I didn't handle correctly situation when 0 is pressed |
| 10:35.22 | *** join/#asterisk Merlin (merlin@216.105.114.10) |
| 10:35.38 | WIMPy | There's no DTMF end, wich is the only relevant message as far as I understand. |
| 10:36.04 | Merlin | does anyone have information on or know about the current status of tty/tdd support in Asterisk? voip-info.org has information as of 2006 which is ridiculously out of date |
| 10:37.24 | WIMPy | You generally have a hard time with non IP data today. |
| 10:38.46 | mirko_brankovic | [Aug 16 12:20:31] DTMF[20369] channel.c: DTMF end '0' received on SIP/..., duration 102 ms |
| 10:38.48 | mirko_brankovic | [Aug 16 12:20:31] DTMF[20369] channel.c: DTMF end passthrough '0' on SIP/... |
| 10:39.07 | WIMPy | ok |
| 10:39.46 | mirko_brankovic | my mistake :) |
| 10:41.02 | *** join/#asterisk debuggerboy (~debuggerb@117.202.115.251) |
| 10:46.13 | *** part/#asterisk Disident (~Disident@85.62.99.220) |
| 11:00.14 | WIMPy | Do I get it right that app_pppd was only available for Asterisk 1.2? |
| 11:08.46 | *** join/#asterisk debuggerboy (~debuggerb@117.196.139.94) |
| 11:10.45 | *** join/#asterisk dandate2 (~dan@124.6.157.210) |
| 11:11.19 | dandate2 | so i just got a g729 license from digium for the first time. but when i run wget http://downloads.digium.com/pub/telephony/codec_g729/\benchg729/x86-32/benchg729-1.0.7-x86_32 -O benchg729 per the howto i get 404 error? |
| 11:12.04 | leifmadsen | dandate2: good chance the documentation is out of date (you probably need a newer version |
| 11:12.31 | leifmadsen | also there is an extra backslash in your link |
| 11:12.37 | leifmadsen | http://downloads.digium.com/pub/telephony/codec_g729/benchg729/x86-32/ |
| 11:12.39 | dandate2 | yeah i noticed that too, its in the howto |
| 11:12.41 | dandate2 | like that |
| 11:12.43 | leifmadsen | and I was right about the version |
| 11:12.43 | dandate2 | ill try without it |
| 11:12.56 | leifmadsen | you should just look at the link and get the right one :) |
| 11:13.12 | dandate2 | ok i see they on 1.0.8 now |
| 11:13.13 | leifmadsen | I just copied up-to-and-including the codec_g729 part |
| 11:13.21 | dandate2 | right |
| 11:13.28 | dandate2 | how do i tell if i'm running 32bit or 64bit? |
| 11:14.38 | leifmadsen | uname -a |
| 11:14.53 | leifmadsen | it depends on what OS flavour you installed |
| 11:15.13 | leifmadsen | Linux sleezy.leifmadsen.internal 2.6.40-4.fc15.x86_64 #1 SMP Fri Jul 29 18:46:53 UTC 2011 x86_64 x86_64 x86_64 GNU/Linux |
| 11:15.15 | leifmadsen | for example |
| 11:15.22 | leifmadsen | that tells me I'm running 64-bit |
| 11:15.50 | dlynes | leifmadsen, hello, leif...any idea what's happening with issue 17288? there hasn't been any update on it for a while |
| 11:16.42 | dandate2 | wget http://downloads.digium.com/pub/telephony/codec_g729/\ |
| 11:16.42 | dandate2 | <PROTECTED> |
| 11:16.45 | dandate2 | whoops |
| 11:17.02 | leifmadsen | dlynes: if there has been no update, then nothing else has happened |
| 11:17.49 | dlynes | leifmadsen, ok...just asking you about that one, because you had posted a question, the original poster posted a number of replies, and then nothing... |
| 11:18.12 | leifmadsen | thanks for the note, I'll take a look |
| 11:18.26 | dlynes | leifmadsen, ok |
| 11:18.27 | dandate2 | how to verify im using asterisk 1.4? |
| 11:18.36 | dlynes | dandate2, core show version at the asterisk cli |
| 11:18.37 | leifmadsen | dandate2: asterisk -rx "core show version" |
| 11:18.51 | leifmadsen | dandate2: or 'show version' in earlier versions of asterisk |
| 11:20.46 | leifmadsen | dlynes: unfortunately there is no one assigned to that issue, and no one else seems to be terribly interested in it |
| 11:21.24 | dlynes | leifmadsen, ah...so I guess my best option for that one currently is to modify my dialplan to compensate, I guess? |
| 11:21.32 | mirko_brankovic | can anyone tell me why ChanSpy (...x(0)) option is not exiting that app on 0 DTMF (using ast 1.8.5.0) |
| 11:21.43 | WIMPy | Ah, one of the bad cause codes. |
| 11:22.02 | dlynes | WIMPy, nod |
| 11:22.13 | leifmadsen | dlynes: pretty much -- the alternative to either fix it yourself, find someone else to fix it for you, or hire someone to fix it |
| 11:22.59 | WIMPy | Yesterday we had someone who got "no user responding" when an already answered call was released. |
| 11:23.24 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 11:23.24 | dlynes | leifmadsen, yeah...I'm able to fix it myself...not that difficult...just means I can't use ubuntu updates on it anymore, and I'll have to patch any updates before I activate them |
| 11:25.00 | leifmadsen | dlynes: ya, the patch that was submitted for that change was just removing code, so that is why it is not being accepted as that. Someone would have to fix it differently |
| 11:25.22 | leifmadsen | so the effort on that issue is higher than just looking at the patch, someone would have to do some coding |
| 11:25.59 | kaldemar | mirko_brankovic: ChanSpy(,g(SPYGROUP),qSx(0)) <-- syntax error |
| 11:26.27 | mirko_brankovic | what is the error? |
| 11:26.40 | leifmadsen | too many commas |
| 11:26.47 | leifmadsen | don't separate flags with commas |
| 11:26.48 | kaldemar | mirko_brankovic: your g() and the rest of the options are separated by a comma. |
| 11:27.03 | mirko_brankovic | aha right, thanks |
| 11:27.51 | mirko_brankovic | ok this is how it is in my ael macro: ChanSpy(,Sqsx(0)g(HOTTV)) |
| 11:28.43 | mirko_brankovic | so 0 dtmf should exit app_ChanSpy without need to catch it right? |
| 11:31.30 | kaldemar | nothing to catch really. 0 will exit the ChanSpy application, nothing more. |
| 11:31.59 | mirko_brankovic | that is what i thought too but not vorking :( |
| 11:32.47 | mirko_brankovic | Connected to Asterisk 1.8.4.4 that is the version i tried it :) |
| 11:33.14 | kaldemar | how is it not working? still does not exit the application? |
| 11:34.27 | mirko_brankovic | yes, that extra ',' was a typo here not in my macro |
| 11:36.38 | kaldemar | mirko_brankovic: show a CLI output of a call with dtmf debug prints enabled. |
| 11:38.39 | Merlin | does anyone have information on or know about the current status of tty/tdd support in Asterisk? voip-info.org has information as of 2006 which is ridiculously out of date |
| 11:41.07 | mirko_brankovic | <PROTECTED> |
| 11:41.09 | mirko_brankovic | <PROTECTED> |
| 11:41.11 | mirko_brankovic | [Aug 16 13:38:34] NOTICE[20552]: app_chanspy.c:479 start_spying: Attaching SIP/... to SIP/... |
| 11:41.13 | kaldemar | pastebin |
| 11:41.13 | mirko_brankovic | That is where i pressed 0 but there was no output |
| 11:41.16 | kaldemar | ~pb |
| 11:41.17 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 11:44.09 | kaldemar | if there's no output when you press 0 and logger has dtmf for console configured, then asterisk does not get DTMF from your phone. |
| 11:45.35 | mirko_brankovic | logger.conf: |
| 11:45.39 | mirko_brankovic | console => notice,warning,error,dtmf |
| 11:45.41 | mirko_brankovic | ;console => notice,warning,error,debug |
| 11:45.43 | mirko_brankovic | messages => notice,warning,error,dtmf |
| 11:46.15 | mirko_brankovic | so that is a bigger problem than :/ |
| 11:47.28 | kaldemar | and logger is reloaded with "logger reload"? |
| 11:47.37 | mirko_brankovic | yes |
| 11:48.02 | mirko_brankovic | app_read is working fine, accepting dtmf |
| 11:48.40 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 11:49.33 | mirko_brankovic | Verbosity is at least 3 |
| 11:49.35 | mirko_brankovic | should i increase that or 3 is good to see dtmf info in console |
| 11:52.32 | dlynes | mirko_brankovic, what's your dtmf set to? inband, rfc2833, or info? |
| 11:52.58 | dlynes | mirko_brankovic, and what codec are you using? |
| 11:53.11 | mirko_brankovic | sip |
| 11:53.25 | dlynes | mirko_brankovic, sip is not a codec, and it's not a dtmf setting |
| 11:53.30 | dlynes | mirko_brankovic, it's a protocol |
| 11:54.05 | mirko_brankovic | ok , where to check dtmf settings |
| 11:54.10 | dlynes | mirko_brankovic, sip.conf? |
| 11:54.25 | kaldemar | mirko_brankovic: verbosity does not affect logger output on DTMF. |
| 11:54.30 | dlynes | mirko_brankovic, i'm guessing you didn't set this system up? |
| 11:54.46 | dlynes | mirko_brankovic, or you're using freepbx, or something similar? |
| 11:54.54 | mirko_brankovic | yes i did but i'm still a bigginer |
| 11:55.25 | dlynes | mirko_brankovic, ah...anyways...look for the following lines: dtmf=xxxx, disallow=xxxx, allow=xxxx |
| 11:55.32 | mirko_brankovic | inband |
| 11:55.52 | dlynes | mirko_brankovic, and what are your disallow/allow lines saying? |
| 11:56.31 | mirko_brankovic | dissallow=all /allow=alaw |
| 11:56.31 | dlynes | mirko_brankovic, reason I'm asking is because inband only works on clear channel codecs such as g726 and g711 |
| 11:56.38 | mirko_brankovic | using only alaw files |
| 11:56.51 | dlynes | mirko_brankovic, disallow, or dissallow? |
| 11:57.18 | mirko_brankovic | disallow=all |
| 11:57.20 | mirko_brankovic | sorry typo :) |
| 11:57.21 | dlynes | ok |
| 11:57.29 | dlynes | just wanted to make sure it was a typo |
| 11:58.12 | mirko_brankovic | well not sure about codecs :/ |
| 11:58.12 | kaldemar | set dtmfmode to rfc2833. |
| 11:58.14 | dlynes | well, i've gotta run, but I thought I'd ask those questions, in case those weren't asked alrady |
| 11:58.39 | mirko_brankovic | ok thanks ) |
| 11:59.20 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 11:59.48 | mirko_brankovic | ok i'll try with rfc2833 |
| 12:00.13 | jacc0 | hi all |
| 12:00.50 | jacc0 | is anybody in here known with the Vox-sopho ? does it have sip support? |
| 12:00.52 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 12:04.56 | mirko_brankovic | kaldemar, it's not working with rfc2833, not even app_read |
| 12:06.00 | WIMPy | You probably have to set that on the phone as well. |
| 12:11.45 | *** part/#asterisk mirko_brankovic (~mirko_bra@212.200.146.253) |
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| 12:22.20 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 12:22.21 | phix | :D |
| 12:24.57 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
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| 12:33.59 | irroot | Asterisk is a bad system according to a customer who insists there must be no VM / Fwding and all calls go to "reception" the recptionist found the DND button and callers complained ... |
| 12:34.29 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 12:35.09 | *** join/#asterisk robl^laptop (~robl@pdpc/supporter/active/robl) |
| 12:35.55 | atheos | <PROTECTED> |
| 12:35.59 | phix | irroot: they are right! asterisk is a bad system for customers that think it is a bad system :) |
| 12:36.11 | irroot | lol |
| 12:36.11 | wdoekes2 | what?! asterisk cannot overrule the dnd button? |
| 12:36.19 | chuckf | sounds like the client needs a new receptionist |
| 12:36.46 | *** join/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-218-224.w86-204.abo.wanadoo.fr) |
| 12:37.18 | merlin8282 | Moin ! I have a problem : the variables passed via AGI seem to be empty... |
| 12:37.19 | phix | then again those customers also think public health care is a bad thing, the teaparty is the gratest thing that has every happend to politics, and climate change is a myth |
| 12:37.20 | atheos | the complaints I hear typically follow "This doesn't work EXACTLY like the 6 figure CCM system you replaced it with. What are we paying you for?" |
| 12:37.52 | Gugge | "something different" |
| 12:38.04 | Gugge | would be the obvious answer :) |
| 12:38.13 | merlin8282 | 1.6.2.9-2+squeeze2, when I do echo $agi_arg_1 it's empty. The AGI is called with many parameters: exten => 71,n,AGI(findfreeconf.sh,dynaconf,60,69) |
| 12:38.31 | robl^laptop | CCM? like a Nortel CS2000? |
| 12:38.38 | irroot | facepalm and headbang |
| 12:38.41 | atheos | cisco call manager |
| 12:38.44 | phix | atheos: :) exactly meaning they dont have to bash their head against their table? |
| 12:39.03 | robl^laptop | atheos: ahhh! ok, sorry I joined late in the conversation. ;-) |
| 12:39.05 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 12:39.08 | atheos | phix, they never did the head bashing. They just paid a company to do it for them. |
| 12:39.19 | irroot | the thing is they went with asterisk cause it was much cheaper and more open than the other systems allowing there software to intergrate |
| 12:40.08 | phix | hehe |
| 12:40.12 | robl^laptop | returns to cursing at the Nortel / Avaya and the CS2000 |
| 12:40.56 | phix | those other systems can do everything as well, you just need to spend your profit margin on them to ge teh features and maintain then |
| 12:41.00 | phix | them |
| 12:44.20 | *** join/#asterisk k3asd` (~k3asd@static-217-133-87-244.clienti.tiscali.it) |
| 12:44.48 | irroot | phix indeed now they want custom reports we have already given them full TMS linking in the billing nodes accross the country [we billed by node <50km is local] and lot of other statistics they had bugger all before a 10yo Samsung 500 |
| 12:45.24 | phix | so they want more for less then complain about it? sounds normal to me |
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| 12:46.48 | *** part/#asterisk mirko_brankovic (~mirko_bra@212.200.146.253) |
| 12:46.55 | p3nguin | coppice: Are you suggesting that 'generic' might be a better choice for res_fax_digium on my VIA CPU than the 'i686' module? |
| 12:47.05 | *** join/#asterisk mutex7c (~mutex7c@82.113.99.57) |
| 12:48.33 | *** join/#asterisk mutex7c (~mutex7c@82.113.99.57) |
| 12:53.21 | kaldemar | merlin8282: how are you reading the variables? |
| 12:54.35 | merlin8282 | kaldemar: I think I just get it; for some reason the variables were not given to the script (in asterisk console, I did only see AGI(scriptname), but no variable... |
| 12:54.41 | merlin8282 | now it seems to work |
| 12:55.27 | merlin8282 | it's ok. Forget it (and thanks for all the fish ;) |
| 13:01.50 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 13:01.56 | coppice | p3nguin: the C3 has a couple of i686 instructions missing. some code runs OK. some doesn't |
| 13:02.12 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-nhjpkmmlnpxkktwj) |
| 13:02.18 | coppice | p3nguin: presonally, I'd run spandsp. it works better :-) |
| 13:05.48 | jacc0 | ping leifmadsen |
| 13:05.56 | leifmadsen | jacc0: pong? |
| 13:06.04 | jacc0 | leifmadsen:https://issues.asterisk.org/jira/browse/ASTERISK-18093?focusedCommentId=181095#comment-181095 |
| 13:07.53 | jacc0 | I don't care if you close it :) |
| 13:08.46 | jacc0 | I'm already so happy; no crashes, deadlocks and memleaks after removing res_timer_timerfd.so :) |
| 13:09.01 | jacc0 | on 20+ production envirements :) |
| 13:09.09 | leifmadsen | jacc0: I need you to test the latest changes -- there is a branch that is supposed to fix timerfd |
| 13:09.21 | jacc0 | okay, will do |
| 13:09.28 | leifmadsen | someone has a task right now to fix it |
| 13:09.30 | leifmadsen | let me get you the info |
| 13:10.05 | Gugge | jacc0: did you just remove res_timer_timerfd and install dahdi? |
| 13:10.15 | *** join/#asterisk Merlin (merlin@evendata.net) |
| 13:12.15 | *** join/#asterisk DagMoller (~aguirre@unaffiliated/dagmoller) |
| 13:12.41 | DagMoller | can anyone telme why this ocurs? VERBOSE[24983] chan_dahdi.c: -- Moving call from channel 3 to channel 6 |
| 13:13.21 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 13:13.34 | WIMPy | Probably channel 3 was seized from both sides at (almost) the same time. |
| 13:14.34 | leifmadsen | jacc0: https://reviewboard.asterisk.org/r/1361/ |
| 13:14.49 | DagMoller | WIMPy, have any solution? |
| 13:15.10 | WIMPy | The solution is what has happend. |
| 13:15.42 | DagMoller | this never ocurs before... |
| 13:16.17 | DagMoller | its start 2 weeks from now |
| 13:16.58 | WIMPy | Well, you need to have "good" timing for this to happen. Why does it bother you? |
| 13:17.33 | *** join/#asterisk asilva (~andre@2801:88:1000:2::12) |
| 13:17.48 | asilva | Hey, does anyone have information or study topics for dCAP ? |
| 13:18.41 | DagMoller | WIMPy, because this never ocurs by passing 2 yers |
| 13:18.55 | robl^laptop | asilva: check the asterisk training offers on digium's site. that should give you an idea on what topics are covered |
| 13:18.59 | jacc0 | Gugge: I've always been installing dahdi; but some version made res_timer_timerfd default timing |
| 13:19.24 | WIMPy | DagMoller: More calls? Or someone changed the channel allocation policy? |
| 13:19.52 | DagMoller | WIMPy, nothing changes |
| 13:20.21 | asilva | robl^laptop, tkz! |
| 13:21.13 | robl^laptop | asilva: basically need to know everything covered in the Asterisk Advanced course |
| 13:21.20 | WIMPy | DagMoller: Unless you use a version where that procedure is broken, there's nothing to worry about. |
| 13:22.17 | *** join/#asterisk VoipForces (~Adium@69.165.199.195) |
| 13:22.26 | asilva | tkz, i'll find the topics and study what i have more doubt about it, i'll try to get ready for astricon, take the exam there! |
| 13:22.53 | robl^laptop | speaking of dCAP, I noticed that it still lists the 2nd ed of the Asterisk book. Is dCAP still targeting 1.4 now and not 1.8? |
| 13:22.56 | VoipForces | Hi all, anyone has hints on how to have asterisk detect tones for disconnected numbers when a call is issued via the AMI ? |
| 13:24.02 | irroot | VoipForces ?? hangupcause ?? |
| 13:24.14 | VoipForces | its not hanging up... |
| 13:24.20 | WIMPy | VoipForces: Get in to the digital age. |
| 13:24.46 | VoipForces | WIMPy: What do you mean? |
| 13:25.02 | *** join/#asterisk SpiderMon (SpiderMon@c-66-176-31-249.hsd1.fl.comcast.net) |
| 13:25.24 | SpiderMon | OT: can ne1 suggest a good filesharing service |
| 13:25.41 | WIMPy | Get a connection to the world that tells you ewhat's going on instead of listening to tones. |
| 13:26.16 | VoipForces | WIMPy: This is a SIP trunk with AllStream that I am doing interop test on⦠|
| 13:26.41 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 13:26.44 | WIMPy | Then it might be a good idea to try another ITSP. |
| 13:26.49 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:27.41 | p3nguin | coppice: I don't know if there's a good fax implementation for 1.4 involving spandsp. |
| 13:28.06 | *** part/#asterisk DagMoller (~aguirre@unaffiliated/dagmoller) |
| 13:28.16 | p3nguin | So I'd still like to know if you are suggesting that the generic modules would be better than the i686 modules. |
| 13:29.11 | SpiderMon | ne1 using AT&T SIP service. And can ne1 suggest a reliable fax server app that can handle lets say 10-20 fax extensions |
| 13:29.17 | *** join/#asterisk makmak78 (~makmak78@83.145.38.138) |
| 13:29.31 | makmak78 | hello! anyone familiar with queues and dtmf |
| 13:29.57 | p3nguin | ~ask |
| 13:29.57 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:30.00 | asilva | robl^laptop, that's a good question, can't tell for sure! |
| 13:30.03 | p3nguin | makmak78: ^^^^^ |
| 13:30.50 | makmak78 | when i connect my agent to agentlogin and waiting for call, if i press a dtmf digit (rfc2833) it queues until i reviceve a call |
| 13:31.03 | makmak78 | it just loops and loops until i get a call |
| 13:31.29 | p3nguin | When you say "it loops," what are you talking about? |
| 13:31.34 | makmak78 | if i should press * which is the hangup command, it waits until i get a call and then hangs it upp right away |
| 13:32.41 | makmak78 | http://pastebin.com/qdDLzR3w |
| 13:34.04 | VoipForces | WIMPy: you know any that gives result codes for ALL disconnected numbers? |
| 13:34.51 | WIMPy | Probably not in the right country. |
| 13:35.04 | VoipForces | WIMPy: Canada |
| 13:35.20 | WIMPy | But that's an interesting thing to test. |
| 13:36.53 | makmak78 | in this scenario the agent is connected to queue via agentlogin and waiting for call, if then pressing * (command to hangup queue call) BEFORE getting any queue call, the next queue call to this agent is hungup instantly because the dtmf command seems to wait until a call is connected |
| 13:40.03 | *** part/#asterisk eject_ck (~eject_ck@62.205.134.210) |
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| 13:51.03 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:56.46 | WIMPy | p3nguin: I have to correct myself. UI seem to send calls from 36 different IPs by now. Configuring 36 peers starts to get tedious :-( |
| 13:57.27 | p3nguin | Oh wow. How do they expect a user to deal with that?! |
| 13:58.39 | WIMPy | Dunno. But having to list IPs doesn't look clever here. |
| 13:59.27 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:11.37 | p3nguin | I'm glad I don't use that company for services. |
| 14:13.17 | Qwell | SRV? |
| 14:14.45 | *** join/#asterisk BenC[UK] (~bcummins@cpc2-lock3-2-0-cust77.6-1.cable.virginmedia.com) |
| 14:18.14 | *** join/#asterisk robl^ (~robl^@pdpc/supporter/active/robl) |
| 14:20.26 | BenC[UK] | Guys, I am struggling ...I am trying to originate a dialplan using the AMI, sending in a "Lead ID", and then use the dialplan to pull the first number to call for that lead, then call the number, and if it answers, then send them to a queue, if it doesnt answer, it updates the status, and moves on to the next number for that lead |
| 14:20.49 | Merlin | BenC[UK]: why aren't you using AGI for that? |
| 14:21.06 | BenC[UK] | Merlin: well, I need to way to start the call originally |
| 14:21.16 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 14:21.17 | Merlin | call file? |
| 14:21.19 | BenC[UK] | once in the dialplan, I am using AGI to handle the call |
| 14:21.25 | Merlin | drop a call file |
| 14:21.32 | BenC[UK] | ok, I could use a call file - but isnt it essentially the same as an originate |
| 14:22.00 | Merlin | it's just easier to use call files and AGI to initiate calls rather than AMI |
| 14:22.00 | p3nguin | It isn't? |
| 14:22.15 | atheos | pretty much, only less reliable |
| 14:22.27 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 14:22.30 | BenC[UK] | ok, thats fine - I can use a call file, let me try that, and see what happens |
| 14:22.32 | atheos | and less convenient for planning calls in the future |
| 14:22.50 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 14:23.44 | BenC[UK] | in a call file, do I have to specify a channel? |
| 14:24.26 | kaldemar | BenC[UK]: yes you do. what's the problem with AMI originate? |
| 14:24.38 | BenC[UK] | The issue I am getting.. |
| 14:24.54 | BenC[UK] | my dialplan starts.. it calls an agi script to get the number to call |
| 14:24.57 | kaldemar | i'd use AMI rather than a call file. |
| 14:25.18 | p3nguin | In the case of incoming calls, will SRV be looked up for every configured peer and the call will be matched against the list of addresses PER CALL, or on loading of chan_sip and then subsequently at the intervals specified in the record? |
| 14:25.19 | Merlin | it's so much more complicated to implement a reliable connection to AMI |
| 14:25.20 | BenC[UK] | which is fine, then I call the Dial(SIP/myprovider/1234567890) |
| 14:25.34 | Merlin | it drops so easily, TCP doesn't handle the disconnects well |
| 14:25.46 | Merlin | i would honestly stay away from AMI if not absolutely necessary |
| 14:25.47 | BenC[UK] | my phone rings.. |
| 14:26.00 | BenC[UK] | I need it to send to a queue if I answer |
| 14:26.18 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:26.29 | BenC[UK] | if its busy, or I dont answer etc, I need to update the database, and then call the next number for the lead |
| 14:26.55 | kaldemar | BenC[UK]: you originate to Local/exten@context and then the other end to an extension that goes to the queue. |
| 14:27.37 | kaldemar | and exten@context handles all you wish to do in case of busy, no answer or what ever you want to act on. |
| 14:29.21 | BenC[UK] | if its busy though, I need to make another call |
| 14:29.29 | BenC[UK] | and then when that answers, send them to the queue |
| 14:29.59 | BenC[UK] | the second call just seems to hangup though |
| 14:30.12 | *** join/#asterisk Polysics (~Luca@host71-72-dynamic.245-95-r.retail.telecomitalia.it) |
| 14:30.21 | Polysics | hello |
| 14:30.27 | Polysics | i have been given a SIP address |
| 14:30.31 | Polysics | how do i call that? :-D |
| 14:30.47 | p3nguin | Dial(SIP/number@host) |
| 14:31.10 | p3nguin | That's for a SIP URI call. |
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| 15:06.09 | BenC[UK] | is there a sample dialplan around for trying multiple outbound carriers? |
| 15:06.42 | p3nguin | You need a sample for that? |
| 15:06.46 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 15:07.55 | Kobaz | Dial(foo); if (("${DIALSTATUS}" != "ANSWER") && ("${DIALSTATUS}" != "BUSY") ... etc) { Dial(bar); } |
| 15:08.02 | p3nguin | If you want to try multiple carriers on a single call, you can use sequential Dial()s. List them in order that you want to try them. If one fails, the next should be used. |
| 15:08.38 | Kobaz | Actually if it's ANSWER and the 'g' option isn't used, it will exit anyway, so you don't have to check for that |
| 15:08.55 | Kobaz | i'm usually using g though |
| 15:12.21 | BenC[UK] | of course, being dumb |
| 15:12.25 | BenC[UK] | thanks |
| 15:15.53 | tully` | Is there any way to call an EAGI script but not have asterisk wait for it to finish? |
| 15:17.20 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:17.24 | *** join/#asterisk frd91gt (~frd91gt@209-124-48-194.static.123.net) |
| 15:17.26 | tully` | I'm piping the call to the script but if the script hangs for any reason asterisk wont run the hangup routine and i end up with tons of zombie scripts |
| 15:18.53 | *** join/#asterisk cerienjean (~iper@95.138.77.91) |
| 15:25.55 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 15:33.14 | kaldemar | tully`: make the script die on SIGHUP or make it not hang. make sure that you don't have AGISIGHUP channel variable set to "no" in your dialplan before the AGI execution. |
| 15:34.29 | leifmadsen | has anyone every configured a polycom phone to provide a different ringback based on dialplan configuration? (I'm talking about the ringing you hear in the headset, not the ringing of the device itself) |
| 15:35.47 | Qwell | ingoing or outgoing? |
| 15:35.51 | Qwell | ingoing? wtf |
| 15:35.56 | Linuturk | is my T1 card sharing an interrupt? http://pastebin.com/4Qcx6GWw |
| 15:36.04 | p3nguin | ingoing! |
| 15:36.16 | p3nguin | outcoming calls, of course. |
| 15:36.26 | Qwell | Linuturk: It is on 15 by itself |
| 15:36.48 | Linuturk | Qwell: when I do an lspci -vb, it shows several other devices on 15 |
| 15:36.58 | Linuturk | two nics, and the SAS |
| 15:37.01 | Qwell | pastebin |
| 15:37.16 | *** join/#asterisk irroot (~irroot@197.173.101.201) |
| 15:37.23 | tully` | kaldemar, is that the only way to kill the script? I'm using php and pcntl_signal and it's not proving to be very reliable in my testing. |
| 15:37.27 | WIMPy | /proc/interrupts only lists IRQs claimed by drivers. |
| 15:37.41 | Linuturk | http://pastebin.com/FGhqURuX Qwell |
| 15:38.25 | Linuturk | http://pastebin.com/NASSZeJL Qwell IRQ misses on the dahdi 1 too |
| 15:39.11 | Linuturk | this might explain the series of issues I've been having, such as funky faxes, certain numbers not dialing |
| 15:39.27 | Linuturk | dmtf related, I think |
| 15:39.37 | ChannelZ | Qwell: Can you tell me what a serial number on a TDM800 looks like? I wrote down just about every number on every sticker last time I had my machine open but the website doesn't like anything |
| 15:40.06 | Qwell | It's the one that starts with "S/N:" |
| 15:40.11 | WIMPy | ChannelZ: Is the card too old? |
| 15:40.26 | ChannelZ | ? Maybe. 2007 |
| 15:40.55 | Qwell | Should be like 9 alphanum digits. |
| 15:41.01 | ChannelZ | (I bought it Nov 2007 that is) |
| 15:41.08 | Qwell | Last 6 would be num |
| 15:41.52 | ChannelZ | hmm. Yeah I have a BFF###### number |
| 15:42.04 | Qwell | that'd be it |
| 15:42.09 | ChannelZ | at least I think it's BFF, it could be 3FF but I tried that too and no worky |
| 15:42.28 | Linuturk | Qwell: am I right about them sharing the same IRQ? |
| 15:42.32 | ChannelZ | but the green and red modules I'm pretty sure are BFF* too |
| 15:42.35 | Qwell | ChannelZ: What are you trying to do? |
| 15:42.52 | ChannelZ | I was trying to reg it to my account since I apparently never did |
| 15:44.08 | WIMPy | Linuturk: It is some level of shared. In theory it shouldn't matter if the other devices aren't used, but that has caused trouble for me before anyway. |
| 15:44.29 | ChannelZ | back in a bit, I have to dash to work |
| 15:48.19 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 15:53.12 | Linuturk | WIMPy: well, since the other devices are network ports and sata drives |
| 15:53.22 | Linuturk | I imagine it is hammering that IRQ pretty hard |
| 15:56.18 | WIMPy | It they are in use, they should show up under /proc/interrupts. |
| 15:57.16 | Linuturk | well, I know the sata controller is in use |
| 15:57.31 | Linuturk | this server has 4 nics though |
| 15:57.53 | Linuturk | i'm only using two |
| 15:58.09 | Linuturk | I will try to force it onto it's own IRQ tonight |
| 15:58.21 | p3nguin | its |
| 15:58.23 | p3nguin | not it's |
| 15:59.52 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 16:00.52 | WIMPy | Well, there is not sata appearing, but I'd expect that to use MSI anyways. |
| 16:01.11 | WIMPy | So it's a little hard to say what's really going on. |
| 16:01.35 | Linuturk | well, the irq misses have to give weight to an irq conflict, yes? |
| 16:02.03 | WIMPy | No, it can be just too slow. |
| 16:02.30 | WIMPy | That can happen because of other interrupts as well. |
| 16:03.02 | WIMPy | I'd force the pri cards IRQ to another cpu. |
| 16:03.18 | *** join/#asterisk wasanzy (~emmanuel@196.201.43.55) |
| 16:03.35 | WIMPy | 17 missed IRQs isn't that much. In what time was that? |
| 16:04.01 | Linuturk | 4 weeks, 3 days, 17 hours, 28 minutes |
| 16:04.09 | xbp | morning |
| 16:05.03 | WIMPy | One missed IRQ every 2nd day is surely not causing any trouble. |
| 16:05.15 | merlin8282 | mmm... Any idea why ChannelRedirect only works when the call initiator has no audio ? |
| 16:05.32 | wasanzy | hi room |
| 16:06.00 | Linuturk | WIMPy: well, I've been having several issues, and I was hoping this was the cause. (faxing issues and trouble connecting to certain 1800 numbers.) |
| 16:06.22 | Linuturk | https://issues.asterisk.org/jira/browse/PRI-124 WIMPy if you remember |
| 16:06.29 | xbp | Linuturk : sounds like provider or dialplan issues |
| 16:06.35 | wasanzy | Can I use a land-line phone plug to asterisk box, to provide IVR? which means asterisk will from then be answering alls on behalf of the land-line phone |
| 16:06.39 | xbp | ya you changing pri providers? |
| 16:06.47 | WIMPy | It wouldn't explain more than one bad line in a fax every 2nd day. |
| 16:06.48 | xbp | clock sourcing was one of our issues recently |
| 16:06.53 | xbp | slipping frames etc |
| 16:07.01 | xbp | att to vz |
| 16:07.03 | xbp | bbiaf |
| 16:07.09 | Linuturk | xbp: haven't changed in quite a while |
| 16:07.16 | Linuturk | just the new server is what has changed |
| 16:07.24 | Linuturk | I'm using the same signalling as the old system |
| 16:07.25 | xbp | ah ok |
| 16:07.31 | xbp | ya <3 |
| 16:07.39 | WIMPy | That issue was related to an early media bug. |
| 16:07.42 | merlin8282 | I explain: I do a ChannelRedirect (in a n-way calling macro), but the party is sent to the conference only when the caller and the 3rd party have opened the conference. In other words, the party has to wait (hearing no moh) for the caller and the 3rd party to come to the conference. |
| 16:07.53 | xbp | i thought you m ight have been battling that ole wnk wnk pri issue i was contending with last week remotely for a customer in joplin missouri |
| 16:08.11 | xbp | lovely issues when you're 1800 miles away <-: |
| 16:08.15 | wasanzy | any idea from any one? |
| 16:08.26 | Linuturk | the only other change was, I removed two channelbanks and a dedicated echo canceller, replacing them with a sip gateway and the new ec on the t1 card. |
| 16:09.07 | Linuturk | WIMPy: yeah, that's what we figured last time (early media) but why would it suddenly come back without a change in the server? |
| 16:09.42 | WIMPy | Because your provider chaneg something? Or the provider hosting those 800 numbers? |
| 16:11.04 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:11.39 | wasanzy | <PROTECTED> |
| 16:12.39 | Linuturk | maybe my lbo is off? |
| 16:12.43 | Linuturk | line build out |
| 16:12.46 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 16:12.47 | Linuturk | I never understood that |
| 16:13.09 | Linuturk | is that the distance from the modem to the server? or from the server to the telco's office? |
| 16:13.17 | wasanzy | hmmmm |
| 16:15.11 | WIMPy | You only have a physical connection to the NT. |
| 16:15.24 | Linuturk | NT? |
| 16:15.52 | WIMPy | The box you connect the card to. |
| 16:16.04 | WIMPy | Might actually be a modem. |
| 16:16.30 | Linuturk | well, we've got fiber coming in, and their modem box thing gives me ethernet and a pri |
| 16:17.17 | merlin8282 | mmm, in fact I seem to have this problem: https://issues.asterisk.org/view.php?id=17117 |
| 16:17.31 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 16:19.17 | *** join/#asterisk nix8n82-phone (~AndChat@24.143.28.16) |
| 16:19.45 | Linuturk | WIMPy: will you be on after 5 PM Central Time ? |
| 16:20.05 | Linuturk | I can provide all the debug information for this issue, and I'm hoping you can help me |
| 16:20.06 | WIMPy | When is that? |
| 16:20.33 | Linuturk | 1700 US Central Time |
| 16:21.01 | chazzam | -6 GMT? |
| 16:21.04 | WIMPy | - how many hours from UTC? |
| 16:21.05 | chazzam | or is it -5 now |
| 16:21.09 | WIMPy | Ah |
| 16:21.12 | *** join/#asterisk mocker (~AndChat@32.144.6.34) |
| 16:21.32 | chazzam | currently round 11:30 |
| 16:21.35 | WIMPy | Yes, I will probably be here then. |
| 16:21.52 | Linuturk | ok |
| 16:21.59 | Linuturk | thanks |
| 16:22.03 | leifmadsen | CDT == -0500 GMT |
| 16:22.21 | Linuturk | so, you think trying to swap the IRQ is an exercise in futility |
| 16:22.34 | chazzam | I couldn't remember if we were in daylight savings or not |
| 16:22.42 | leifmadsen | yep |
| 16:22.51 | Linuturk | which means the issue will probably be found in the SIP or PRI debug |
| 16:23.33 | WIMPy | Do you ever see things like HDLC aborts? |
| 16:23.45 | *** join/#asterisk niekie (quasselcor@CAcert/Assurer/niekie) |
| 16:24.11 | Linuturk | yes |
| 16:24.21 | Linuturk | I grepped my full log for that, and I see them |
| 16:24.42 | Linuturk | http://pastebin.com/QA1NbyfH |
| 16:24.52 | WIMPy | If you see more than 17 of them for the last 4 weeks, you've got another problem. |
| 16:25.54 | WIMPy | That means between the card and the telcos switch. |
| 16:26.03 | Linuturk | yeah, there are tons of them |
| 16:26.21 | Linuturk | [Jul 17 05:55:03] NOTICE[17248] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 |
| 16:26.23 | WIMPy | So you get IP over the same fiber? Any paket loss there? |
| 16:26.24 | Linuturk | that too |
| 16:26.37 | WIMPy | Definitely bad. |
| 16:27.38 | Linuturk | ping to google does pretty good |
| 16:27.44 | Linuturk | 0 packet loss so far |
| 16:28.25 | WIMPy | So the part from the card to the modem is on top of the list so far. |
| 16:28.53 | Linuturk | http://pastebin.com/4AsqCgVd << /etc/dahdi/system.conf |
| 16:28.57 | Linuturk | anything off there? |
| 16:29.38 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 16:29.51 | Linuturk | http://pastebin.com/nxzghFcB << missed a line in the copypasta |
| 16:30.07 | Linuturk | ignore the root too :P |
| 16:30.35 | Linuturk | http://pastebin.com/4TjxbR2B << there we go |
| 16:32.31 | WIMPy | If the framing or coding was wrong I'd expect it to work a lot worse. |
| 16:32.37 | WIMPy | i.e. not |
| 16:33.16 | WIMPy | I've never experimented with the LBO thing. |
| 16:33.30 | WIMPy | Have you tried to replace the cable? |
| 16:33.50 | Linuturk | I have not. they aren't standard ethernet cables, are they? |
| 16:34.18 | WIMPy | A full 8-wire patch cable fits. |
| 16:34.31 | Linuturk | http://pastebin.com/ngCQSNR4 << chan_dahdi.conf btw |
| 16:34.42 | Linuturk | a standard patch cable? |
| 16:34.44 | Linuturk | no crossover? |
| 16:34.49 | WIMPy | yes |
| 16:35.11 | WIMPy | chan_dahdi definitely above the problem level. |
| 16:35.18 | WIMPy | No crossover. |
| 16:35.36 | Linuturk | so, we might be looking at a bad cable? |
| 16:35.37 | WIMPy | And a PRI corssover is different than an ethernet corssover. |
| 16:35.38 | Linuturk | lol |
| 16:35.49 | WIMPy | Shit happens. |
| 16:35.59 | WIMPy | Definitely worth a try. |
| 16:36.14 | Linuturk | weird how the other server was working just fine, but then again, it had the echo canceller in between |
| 16:36.40 | WIMPy | That should not make any difference. |
| 16:36.59 | WIMPy | Certainly not for the D-channel. |
| 16:37.01 | *** join/#asterisk darkskiez (~darkskiez@62-50-207-4.client.stsn.net) |
| 16:38.05 | Linuturk | wait, shouldn't the timing source be the teleco? |
| 16:38.11 | Linuturk | telco* |
| 16:38.41 | Linuturk | doesn't 0 mean this card, instead of listening for the other side? |
| 16:38.48 | WIMPy | Did I miss something? |
| 16:38.53 | WIMPy | yes. |
| 16:38.59 | Linuturk | http://pastebin.com/4TjxbR2B |
| 16:39.14 | Linuturk | <PROTECTED> |
| 16:39.37 | WIMPy | Right. That should be it. |
| 16:40.04 | Linuturk | so, it should read: span=1,1,0,ESF,B8ZS |
| 16:40.16 | WIMPy | Yes. |
| 16:40.28 | WIMPy | And that fits to your issues. |
| 16:41.22 | Linuturk | since this card has a hardware echo canceller, I don't have to specifically set it in system.conf, do I? |
| 16:41.33 | Linuturk | that also explains why it gradually gets worse |
| 16:41.39 | Linuturk | as the clocks get more and more out of sync |
| 16:41.40 | WIMPy | Err, what? |
| 16:41.47 | WIMPy | yes |
| 16:41.54 | *** join/#asterisk breardo (~root@216-192-29-134.rrcc.mnscu.edu) |
| 16:41.57 | breardo | hey all |
| 16:42.05 | Linuturk | the echo cancel is totally unrelated. I want to make sure it's using the hardware echo cancel on the t1 card :) |
| 16:42.14 | WIMPy | That configuration probably shouldn't be possible at all. |
| 16:42.31 | Linuturk | i wonder why it's working . . . |
| 16:42.49 | WIMPy | You can't configure that, other than by givong parameters to modprobe. |
| 16:43.06 | WIMPy | Well, it isn't working :-) |
| 16:43.13 | Linuturk | lol |
| 16:43.15 | Linuturk | true |
| 16:43.23 | breardo | I'd like to use *8 to answer other extensions calls, but I'd like to be able to choose which one in the event both are rining... can *8 be modified to allow me to choose which extension? for example, *8101 (for ext 101) *8112 (for ext 112) ? |
| 16:44.01 | WIMPy | That's because the timing is configured in dahdi/system.conf but signalling in asterisk/chan_dahdi.conf, although they must match. |
| 16:44.33 | p3nguin | Excelent question. I've never tried to use *8 to pickup a call when more than one phone was ringing. |
| 16:44.51 | breardo | what i've read says it will randomly choose an extension.. |
| 16:44.59 | breardo | not ideal for what im looking for.. unfortunately :) |
| 16:45.52 | p3nguin | Do you often have multiple phones in your callgroup which are ringing at the same time, which you'd like to hijack? |
| 16:46.15 | breardo | its for an office, and a lady likes to be able to answer two other extensions when they are not in their offices.. |
| 16:46.26 | Katty | has anyone seen mister blackwell? |
| 16:46.30 | p3nguin | I guess they'd actually need to be in the pickupgroup in this case. |
| 16:46.35 | Qwell | Katty: never heard of him |
| 16:46.37 | p3nguin | I haven't seen James in months. |
| 16:46.39 | Katty | Qwell: jblack |
| 16:46.42 | Qwell | Katty: never heard of him |
| 16:46.47 | Katty | infobot: seen jblack |
| 16:46.57 | infobot | jblack <~jblack@71.181.209.104> was last seen on IRC in channel #asterisk, 259d 22h 26m 34s ago, saying: 'Does asteriskgui walk over manual configuration changes?'. |
| 16:46.59 | Kobaz | one more box moved to 1.8 |
| 16:47.06 | Linuturk | this really explains the faxing issue, as faxes are much more sensative to timing than voice calls |
| 16:47.15 | p3nguin | He hasn't even been on freenode in three weeks. |
| 16:47.26 | Katty | i found him on gtalk |
| 16:47.34 | Linuturk | and, it explains the early media problem, as the order of the protocol messages really matter when accepting the call |
| 16:47.39 | Katty | also good morning mister parker |
| 16:47.43 | Qwell | Katty: never heard of him |
| 16:47.53 | Katty | that's nice. |
| 16:48.01 | WIMPy | Linuturk: Yes, But now you fond it and everything should be well. And dahdi gets a negative point on my sympathy list again. |
| 16:48.01 | Katty | how're you dear? |
| 16:48.03 | Qwell | Katty: still planning on moving? |
| 16:48.13 | Katty | Qwell: yes'r |
| 16:48.23 | Qwell | How's that going? |
| 16:48.28 | Katty | Qwell: it's taking longer than i thought, and things are a smidgen more complicated with the boy |
| 16:48.45 | WIMPy | Linuturk: Actually the order shouldn't matter much. |
| 16:49.28 | WIMPy | All steps up to CONNECT are optional. |
| 16:49.57 | WIMPy | And they should only cause a STATUS message when received afterwards. |
| 16:49.59 | robl^laptop | breardo: yes, you can do that. I disable it in features and add it to dial plan. something like exten => _*8XXXX,1, Pickup(${EXTEN:2}@PICKUPMARK) |
| 16:50.16 | Katty | Qwell: why do you ask? are you planning a dr visit to st. louis? :P |
| 16:50.21 | WIMPy | But who know, how picky one or the other implementation is. |
| 16:50.25 | breardo | ahhh cool robl... |
| 16:50.41 | breardo | didnt even know you could do that :) haha |
| 16:51.54 | Katty | goodness me, where are my manners. |
| 16:52.25 | Linuturk | http://pastebin.com/6pgfQD2v << WIMPy dahdi_genconf came up with this :) I'm going to swap them out tonight, and restart the server. |
| 16:52.27 | Katty | GOOD AFTERNOON #ASTERISK |
| 16:52.32 | Qwell | Katty: You have to give me a better excuse than that. :p |
| 16:52.35 | Katty | distributes hugs |
| 16:52.37 | breardo | OH, HELLO KATTY. THANK YOU FOR ACKNOWLEDGEMENT. :) |
| 16:52.50 | Katty | i can't help it. mister Qwell is distracting |
| 16:52.53 | Katty | hugs breardo |
| 16:52.57 | breardo | smiles inside |
| 16:53.09 | Katty | hugs p3nguin |
| 16:58.06 | ChannelZ | freshens his lipstick |
| 16:58.59 | carrar | hi KattyRooooo |
| 16:59.16 | WIMPy | Linuturk: Any theory, how you got that 0 in there? |
| 17:00.48 | Katty | hugs carrar |
| 17:00.53 | Katty | hugs ChannelZ |
| 17:03.17 | Linuturk | WIMPy: not sure. could have been me manually editing for some reason. the old system had that echo canceller, and the old system was responsible for timing to two channel banks |
| 17:03.20 | Linuturk | that might have been it |
| 17:04.41 | carrar | woohoo |
| 17:06.21 | WIMPy | Linuturk: Yes, left the wrong span over from the old config, I guess. |
| 17:06.33 | ChannelZ | What's new an exciting Katty? |
| 17:07.04 | Linuturk | WIMPy: ok, I changed my system.conf to this: http://pastebin.com/19ngyPrC and did a dahdi restart in the asterisk cli |
| 17:07.33 | WIMPy | Linuturk: I think you have to dahdi_cfg. |
| 17:08.09 | WIMPy | Probably dahdi_cfg -s;dahdi_cfg. |
| 17:08.17 | Linuturk | http://pastebin.com/pJb3bVhQ << dahdi_cfg -vvvv |
| 17:08.58 | Katty | ChannelZ: hmm...lets' see |
| 17:09.02 | WIMPy | So let's see if that works on the fly. |
| 17:09.02 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 17:09.05 | Katty | ChannelZ: well i go on vacation starting next week on wed |
| 17:09.13 | Katty | ChannelZ: this saturday i officiate at a local roller derby game |
| 17:09.17 | Katty | ChannelZ: zombie theme ^_^ |
| 17:09.34 | Linuturk | span is still up |
| 17:09.39 | Katty | ChannelZ: next saturday i go to paducah ky to help with /their/ roller derby game. |
| 17:09.40 | Linuturk | the echo cancel is on now |
| 17:09.46 | Linuturk | it seems |
| 17:09.57 | Linuturk | those 1 800 numbers still won't go through |
| 17:11.18 | *** join/#asterisk TimeRider (~steve@host-92-27-131-175.static.as13285.net) |
| 17:11.55 | Linuturk | WIMPy: I guess the issue isn't with the timing source then, b/c those numbers still aren't going through. I hope this helps the faxing issues |
| 17:12.50 | WIMPy | That will certainly help with fax. |
| 17:13.40 | WIMPy | For the 800 numbers you might try to upgrade. There was an early media issue recently. |
| 17:14.13 | ChannelZ | Nice. |
| 17:14.34 | Linuturk | I'm on 1.8.4.4 right now. last time I went to 1.8.5 it caused the system to give congestion after about 12 hours of use. WIMPy that might have been the timing issue though . . . |
| 17:15.13 | WIMPy | Possibly |
| 17:16.06 | Linuturk | newer versions of dahdi is available too |
| 17:16.12 | Linuturk | got an upgrade night, it seems |
| 17:17.39 | WIMPy | Now the interesting question remains: Can you have multiple timing sources on one card? |
| 17:17.52 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 17:18.14 | Linuturk | this card only has 1 port . .. how would I check for multiple timing sources? isn't that impossible for a 1 port card? |
| 17:18.53 | WIMPy | No the question is for multi-port cards. |
| 17:19.22 | WIMPy | I came to think about that last time we discussed a timing source issue. |
| 17:19.40 | Linuturk | :-D |
| 17:19.56 | Linuturk | these 1800 numbers weren't a problem on 1.8.5 |
| 17:20.12 | Linuturk | and, until today, they weren't a problem on 1.8.4.4 |
| 17:20.24 | WIMPy | Good. So you know the solution to that problem as well. |
| 17:20.29 | Linuturk | hopefully the congestion problem was the timing issue |
| 17:20.36 | Linuturk | and not a regression in 1.8.5 |
| 17:21.01 | Linuturk | thank you for your help WIMPy |
| 17:21.22 | WIMPy | I didn't spot it. You did that yourself. |
| 17:22.03 | Linuturk | but you made me start thinking about timing |
| 17:22.11 | *** join/#asterisk wasanzy (~emmanuel@196.201.43.55) |
| 17:22.12 | Linuturk | let's call it 50/50 |
| 17:22.20 | wasanzy | helloo |
| 17:23.14 | Katty | hai |
| 17:23.39 | p3nguin | Is there any way to see what date I added someone to a g+ circle? |
| 17:24.01 | wasanzy | about my previous question, I did some research and I think this phone will help me build an IVR system and provide ss7 services as well?: http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps10033/ps10039/data_sheet_c78-502079.html |
| 17:25.07 | ChannelZ | oh no not ss7 again |
| 17:25.10 | WIMPy | wasanzy: I did mention that SS7 phones don't exist, didn;t I? |
| 17:25.46 | Qwell | How does a phone have anything at all to do with either SS7 or IVR? |
| 17:25.57 | ChannelZ | Qwell: We've been asking this question for a week |
| 17:25.58 | p3nguin | Calling, dontcha know? |
| 17:27.12 | wasanzy | I mean can I connect the phone to asterisk box to enable me setup IVR system? ss7 is out |
| 17:27.15 | WIMPy | Well, it should theoretically be possible to do a SS7 phone. Definitely senseless but probably possible. |
| 17:27.41 | Qwell | wasanzy: Your question makes no sense. A phone has absolutely nothing to do with IVR. |
| 17:27.46 | WIMPy | Asterisk doesn't care how you connect to it when using your IVR. |
| 17:27.50 | Qwell | Can I use a toaster to ride my bicycle? |
| 17:27.59 | ChannelZ | wasanzy: YES |
| 17:28.17 | wasanzy | that is an IP phone |
| 17:28.25 | WIMPy | Qwell: No, but if it's a really big toaster, you might be able to use it as bike stand :-) |
| 17:28.35 | ChannelZ | You can make phone calls, and receive phone calls, and put people on hold, and EVERYTHING! |
| 17:28.41 | p3nguin | or put some wheels on it and ride it. |
| 17:28.42 | Qwell | WIMPy: 8 slot is the way I roll. |
| 17:29.05 | WIMPy | Industrial toaster? |
| 17:29.21 | p3nguin | Toast for the masses. |
| 17:29.41 | ChannelZ | Go big or go home |
| 17:29.46 | p3nguin | word |
| 17:30.00 | jaytee | does the SS7 phone come with standard or upgraded warp nacelles? |
| 17:30.03 | WIMPy | See, that's where we get back to SS7 :-) |
| 17:31.14 | Katty | likes ss7 |
| 17:31.57 | breardo | QUICK.. REVERSE POLARITY!! |
| 17:32.02 | breardo | DIVERT POWER!! |
| 17:32.31 | Katty | watches stuff fry with the reverse polarity |
| 17:32.33 | breardo | you know when Geordi opened up that little cabinet next to the warp core, the ship was screwed |
| 17:32.46 | ChannelZ | error: writers out of ideas. transporter has malfunctioned |
| 17:32.47 | breardo | you know a "Captain, we have a coolant breach" was coming |
| 17:32.51 | WIMPy | Lets get back to SS5 and blow some whistles. |
| 17:32.58 | breardo | and he'd roll under the door in a CLASSIC geordi laforge manouever |
| 17:33.12 | beardy | SS7.. phone? |
| 17:33.26 | wasanzy | I am only looking for a way to build an IVR system now, forgetting about the ss7 for now, but am finding it difficult to understand how that will work with out attaching any hardware phone or sot of |
| 17:33.32 | ChannelZ | beardy: Don't fight it, just go with it |
| 17:33.36 | jaytee | is it just me or did Riker with a beard always come off as a dick as compared to Riker without a beard? |
| 17:33.43 | breardo | ohh man now I got my TNG appetite all worked up |
| 17:33.44 | ChannelZ | wasanzy: use a softphone |
| 17:33.55 | Katty | wasanzy: ah, i think i understand what you're getting at |
| 17:33.56 | breardo | no way.. Riker with beard was the epitome of classic riker |
| 17:33.56 | ChannelZ | wasanzy: or do you mean "how to hook it up to the phone system"? |
| 17:34.01 | breardo | Riker was a dick without the beard |
| 17:34.08 | beardy | fires thrusters and drifts away |
| 17:34.14 | Katty | wasanzy: you can build an IVR system without having a phone to test |
| 17:34.16 | breardo | thanks for the trek talk :) |
| 17:34.38 | Katty | wasanzy: you can get the basics setup, and just use a NoOp for the time being |
| 17:34.43 | ChannelZ | breardo: LLAP |
| 17:35.05 | ChannelZ | Katty: or use a softphone... |
| 17:35.14 | Katty | softphone is also an option |
| 17:36.22 | WIMPy | You could even use the console. |
| 17:36.39 | WIMPy | But sending DTMF is a little uncool there. |
| 17:36.53 | beardy | For an easier understanding you can use a softphone to test your IVR or whatever you're building, but that's not necessary either really, but it does give the "here is my phone", "here is my IVR/PBX/starship" logic. |
| 17:37.01 | wasanzy | I think the best word is hook it up, because I want to be able to call my local line and have the asterisk answer it and get to the IVR menu |
| 17:37.51 | ChannelZ | wasanzy: get an ATA with an FXO if you want to use an analog phone line |
| 17:39.00 | Qwell | wasanzy: again, phones have absolutely nothing to do with an IVR. |
| 17:39.16 | *** join/#asterisk rdahlin_1 (~rdahlin_1@2001:16d8:cc97:1:f8d0:d52:1b2f:56d7) |
| 17:39.29 | wasanzy | ChannelZ: you are right, and you remember I posted some sangoma A200 analog card |
| 17:40.35 | ChannelZ | not me |
| 17:41.01 | ChannelZ | but that'd probably work just lovely regardless |
| 17:41.18 | ChannelZ | oh.. nevermind I misread |
| 17:41.39 | beardy | does have an IVR howto started long ago, guesses he should finish it sometime. |
| 17:41.51 | Qwell | ~book |
| 17:41.52 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 17:41.54 | Qwell | everything you need to know |
| 17:41.56 | wasanzy | Qwell: am confuse because without a phone how can I use eg my cell phone have the IVR system prompt it's menu? |
| 17:42.10 | Qwell | ... |
| 17:42.14 | Qwell | Go read the book again. |
| 17:43.14 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:43.23 | ChannelZ | wasanzy: you don't need a phone, you need an adapter to connect a phone LINE so your Asterisk "becomes the phone" |
| 17:43.24 | *** join/#asterisk jkroon (~jkroon@dsl-241-242-129.telkomadsl.co.za) |
| 17:43.28 | beardy | wasanzy: You don't need a telephone to connect a phone line to an Asterisk system. |
| 17:47.20 | acidfoo | phone line ----> fxo card ---> asterisk |
| 17:47.42 | *** join/#asterisk rdahlin_1 (~rdahlin_1@2001:16d8:cc97:1:f8d0:d52:1b2f:56d7) |
| 17:47.57 | acidfoo | i'm sure he'd like to know more about what he needs that what he doesn't |
| 17:48.18 | Qwell | <infobot> Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 17:48.18 | Qwell | <Qwell> everything you need to know |
| 17:48.41 | leifmadsen | reading is hard |
| 17:48.48 | leifmadsen | just sayin' |
| 17:49.02 | Katty | i wonder how much we read ina lifetime |
| 17:49.06 | Katty | how many pages worth of /stuff/ |
| 17:49.11 | Qwell | 6 |
| 17:49.12 | acidfoo | leifmadsen, i've heard that writing is harder ;P |
| 17:49.46 | leifmadsen | meh |
| 17:49.48 | leifmadsen | just make stuff up |
| 17:50.08 | Qwell | it doesn't even have to be accurate |
| 17:50.16 | Qwell | just report it in the errata when people complain |
| 17:50.57 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 17:50.57 | acidfoo | I can see it.. buy the errata and get a copy of the book for free ;) |
| 17:51.18 | beardy | A proven concept. |
| 17:51.38 | Katty | wonders what the errata is |
| 17:52.40 | ChannelZ | we are |
| 17:53.30 | p3nguin | Well, here I go again. |
| 17:53.52 | ChannelZ | Back to the nacho experiment? |
| 17:58.48 | *** join/#asterisk hobodave_ (~hobodave@pdpc/supporter/professional/hobodave) |
| 18:08.18 | breardo | can I use PickUP to answer a call coming in across a DHADI channel to a SIP phone, from another SIP phone? |
| 18:22.02 | *** join/#asterisk fenlander (~fenlander@82.152.81.57) |
| 18:25.12 | *** join/#asterisk techknowlust (~patrick@li257-211.members.linode.com) |
| 18:25.37 | *** join/#asterisk zotrix (~zotrix@144-37.dsl.aichyna.com) |
| 18:27.00 | techknowlust | is it bad practice to have mailbox's numbered the same as their extensions ? or is that even possible ? |
| 18:32.41 | *** join/#asterisk cerberus_za (~coert@196.209.230.147) |
| 18:37.06 | atheos | techknowlust - you can use any numbering scheme you like. Matching extensions to mailbox numbers is the most convenient. |
| 18:41.11 | breardo | can I use PickUP to answer a call coming in across a DHADI channel to a SIP phone, from another SIP phone? |
| 18:43.05 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 18:49.02 | ChannelZ | breardo: yes |
| 18:49.14 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
| 18:56.37 | *** join/#asterisk garryfre (~gfreemyer@mxgw.videomaker.com) |
| 18:59.09 | techknowlust | I'm suddenly getting "we're sorry but due to techincal difficulties" messages when dialing inbound to asterisk |
| 18:59.28 | techknowlust | I've turned the verbosity in the CLI up and see nothing. Any ideas what's the best way to debug this ? |
| 18:59.41 | ChannelZ | figure out why your ITSP isn't sending you calls |
| 18:59.53 | ChannelZ | Do you register to them or do they know you by static IP? |
| 18:59.59 | garryfre | I have this Asterisk system that keeps recording this beeping this phone system emits to the astarisk server if the person leaving a message forgets to press the # key. I've searched and searched for a way to have it sense this beeping and stop recording it, and I've found absolutely nothing. Please anyone have any hints, a clue by four? I'm completely beat and defeated. |
| 19:01.03 | ChannelZ | garryfre: so the person hangs up and Asterisk continues to record voicemail? |
| 19:01.21 | garryfre | channelz yes. |
| 19:01.42 | ChannelZ | How is the call coming in to Asterisk? Apparently the channel is not signalling that it has terminated |
| 19:02.40 | *** join/#asterisk mickecarlsson (~Micke@h10n3c1o1101.bredband.skanova.com) |
| 19:03.12 | garryfre | From our phone service, into asterisk to an fxo port, its not emiting silence or a dial tone when the caller hangs up, just this beeping is being recorded until max message length is reached |
| 19:03.53 | ChannelZ | Hmm. When the caller hangs up is there a delay between that and the beeping or does it start immediately? |
| 19:04.33 | garryfre | I do not fully understand what to call the system that's calling in. This was just dropped into my lap, and its' feeling like a 3000 pound brick .... hmm let me see ... |
| 19:05.16 | techknowlust | ChannelZ: Yep I register with them and the registrations seem active. It's from FreePBX's online store |
| 19:05.20 | ChannelZ | Is it another PBX or something that is calling in? |
| 19:05.40 | techknowlust | I'm trying to call the number on my cell |
| 19:05.41 | garryfre | Yes, I think it is another pbx I'm guessing |
| 19:05.51 | ChannelZ | hmmm |
| 19:06.14 | garryfre | I'm thinking maybe I have to program it to stop that beeping. |
| 19:06.18 | ChannelZ | well if it's not signalling correctly or otherwise doing this beeping on purpose there's not much you can do about it on the Asterisk side |
| 19:06.30 | ChannelZ | you'd have to figure out what the hell this other system thinks it's doing |
| 19:06.35 | garryfre | ok, thanks. at least it tells me where not to look |
| 19:07.10 | garryfre | I think I know exactly what it thinks it's doing ... it's driving me crazy .... er nm, its too late for that. It's keeping me crazy. :) |
| 19:07.26 | ChannelZ | Sounds like if it is another PBX that the lines are configured for some special case but I can't imagine what |
| 19:07.46 | garryfre | If the caller hits the # key it works as expected. |
| 19:07.58 | ChannelZ | sure, you're telling * to stop recording |
| 19:08.23 | ChannelZ | It will do that on its own if the channel hangs up, or depending on your silence detection settings in voicemail.conf, but if neither of those things is happening.... |
| 19:09.01 | garryfre | Yep, damn thing is immediate. My problem is this asterisk is in for testing while the existing pbx is running because the server it's on is dying. |
| 19:09.28 | ChannelZ | techknowlust: well assuming you really are receiving no INVITEs (sip set debug on) then start looking at the network.. either they're not sending, sending to the wrong place, or you're not receiving maybe because of a firewall or router change, etc. |
| 19:09.54 | garryfre | yep, what I figured ... server keeps listening, pbx keeps being roadrunner ... meep meep! |
| 19:10.33 | ChannelZ | garryfre: Sorry I'm not experienced with any other PBXes so I can't even make a guess as to why it's doing what it's doing, like if it was some common method used in the past for some purpose |
| 19:12.17 | garryfre | Channels yep, and I mentioned the purpose --- a prophetic desire to drive me nuts before I even got this job. :) Least I do enjoy listening to tt-weasels and the one about all members of the household are currently assisting other telemarketers. |
| 19:12.23 | ChannelZ | Just sounds like some custom signalling method on the other PBX for those channels you've connected to Asterisk. |
| 19:14.17 | ChannelZ | Dig around in its configuration (if you can) and see if there is anything interesting |
| 19:15.51 | garryfre | Yep, next step is to read this four inch thick tome called "Panasonic DBS Digital Business System." |
| 19:19.03 | ChannelZ | well you already know what the BS stands for |
| 19:21.09 | *** join/#asterisk hobodave_ (~hobodave@pdpc/supporter/professional/hobodave) |
| 19:21.42 | garryfre | LoL |
| 19:22.43 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 19:23.19 | *** join/#asterisk vfabi (~fabi@host-static-109-185-192-44.moldtelecom.md) |
| 19:28.07 | *** join/#asterisk beardy (~beardy@unaffiliated/beardy) |
| 19:30.11 | *** join/#asterisk hobodave_ (~hobodave@pdpc/supporter/professional/hobodave) |
| 19:30.49 | *** part/#asterisk frd91gt (~frd91gt@209-124-48-194.static.123.net) |
| 19:31.53 | techknowlust | can sip register statements be in files that are included in sip.conf ? |
| 19:32.04 | techknowlust | or must they be present in the main sip.conf file ? |
| 19:35.21 | prgmrchris | anyone have any good recommendations for software simiar to fonality's hud? im looking for something simple that just accesses AMI and I can put on another box and allow it access to AMI. Fonality seems to have abandoned hud and I'd rather not use it because its mac support is awful. If there is anything web based that would nice. |
| 19:35.27 | *** join/#asterisk hobodave_ (~hobodave@pdpc/supporter/professional/hobodave) |
| 19:35.39 | prgmrchris | I thought about FOP but user to user chat is missing from that and clients like that |
| 19:36.33 | ChannelZ | techknowlust: I think the requirement is that they have to appear under [general] as they are unrelated to any particular peer |
| 19:36.52 | ChannelZ | techknowlust: probably that works from an include so long as they are |
| 19:39.22 | SuperNull | Anyone know if asterisk 1.2 had cdr_mysql support ? i have a realllyyyy old install that needs mysql accounting |
| 19:39.46 | techknowlust | ChannelZ: is it possible to have 2 generals ? Can I define one in sip.conf and add to it with an include ? |
| 19:40.01 | ChannelZ | techknowlust: no idea |
| 19:40.15 | ChannelZ | I'll guess 'no' but you can always try |
| 19:40.58 | atheos | techknowlust - yes, it's part of asterisk addons |
| 19:41.24 | atheos | sorry techknowlust - I meant to say SuperNull |
| 19:42.46 | SuperNull | thanks atheos.. looks like the module just isnt loading.. trying preload to force |
| 19:43.07 | atheos | SuperNull - make sure you have all the proper dependencies for mysql |
| 19:43.09 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 19:43.38 | atheos | you would, assuming you're going to run mysql on the same box. |
| 19:43.44 | SuperNull | should its using realtime with mysql |
| 19:44.38 | SuperNull | preload got it i think.. |
| 19:44.45 | *** join/#asterisk oej (~olle@195.41.130.3) |
| 19:44.47 | SuperNull | just wasn't showing in show modules.. |
| 19:46.39 | *** join/#asterisk d_preston215 (~chatzilla@static-76-161-250-54.t1.cavtel.net) |
| 19:46.44 | d_preston215 | Any idea why setting outbound caller id to ' hidden ' on a trunk without any Dialing rules still shows the telephone number? |
| 19:48.09 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:53.25 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 19:53.45 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 19:56.52 | leifmadsen | [2011-08-16 15:56:01] NOTICE[30842]: chan_sip.c:22616 handle_request_invite: Call from '' (79.172.209.97:5060) to extension '+011441214001361' rejected because extension not found in context 'default'. |
| 19:56.54 | leifmadsen | nice |
| 19:58.01 | ChannelZ | I got all kinds last night |
| 19:58.29 | ChannelZ | but I had ones like #441412950142 |
| 19:59.13 | *** join/#asterisk MrOli (43cbade2@gateway/web/freenode/ip.67.203.173.226) |
| 19:59.14 | paulc | Someone wants to call the UK |
| 19:59.32 | paulc | Glasgow.. randomly! |
| 20:01.36 | merlin8282 | Anyone has an idea why AGI stops after returning the result of <SIP/21-0000000e>AGI Rx << EXEC MeetMeCount 61,CFEMPTY ? After this I do a "GET VARIABLE CFEMPTY" in the script, but nothing is received (in the asterisk console)... |
| 20:04.13 | merlin8282 | (of course I do read the results in the script) |
| 20:10.52 | *** join/#asterisk webmonkey (~webmonkey@184.9.254.96) |
| 20:12.10 | merlin8282 | forget it, found my error. |
| 20:14.34 | jaytee | setup |
| 20:18.53 | jaytee | set |
| 20:23.30 | *** join/#asterisk NathanWheeler (4016ebcd@gateway/web/freenode/ip.64.22.235.205) |
| 20:25.35 | NathanWheeler | Hey, power got cut to our asterisk box earlier today, and now incoming callers cannot be heard by their recipients... any idea what might have happened and how to fix it? |
| 20:27.09 | NathanWheeler | outbound calls, and calls from one extension to another work fine... also, voicemails from external callers are not recorded, voicemails from an internal extension are recorded but blank |
| 20:32.12 | jaytee | NathanWheeler, how do calls come in? SIP? PRI? Analog? |
| 20:33.12 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 20:33.34 | NathanWheeler | SIP as far as I know... (I'm an asterisk noob) |
| 20:33.55 | NathanWheeler | @jaytee I know they aren't analog |
| 20:35.07 | jaytee | one way audio issues typically in most cases are caused by firewall nat issues. ports for RTP need to be open. Could they have been configured and not saved? is the * server internet facing or is it also behind a firewall? |
| 20:35.55 | *** join/#asterisk oej (~olle@195.41.130.3) |
| 20:36.20 | NathanWheeler | the server is internet facing, and the box has been rebooted before without issue as far as I know |
| 20:37.18 | ChannelZ | well there's a lot of 'as far as I know' going on.... make sure |
| 20:37.41 | jaytee | the odd piece here is that you said internal calls and outbound calls work fine but internal calls to voicemail are blank. |
| 20:38.02 | ChannelZ | * requests the remote end send their RTP traffic to the ports listed in /etc/asterisk/rtp.conf - if those are blocked incoming (or the remote end is unable to send to them on their end) that's one reason |
| 20:38.43 | NathanWheeler | @ChannelZ it would take months to get through the things I "think" I know about this network... I thought the damn box was plugged into a battery backup, but apparently our previous admin didn't feel like doing that |
| 20:39.30 | leifmadsen | NathanWheeler: did the external IP change and the externaddr= setting in sip.conf not match anymore? |
| 20:40.07 | NathanWheeler | give me a sec to check on that one |
| 20:40.37 | ChannelZ | wonders |
| 20:46.33 | NathanWheeler | @liefmadsen - nope they match... |
| 20:47.52 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 20:51.55 | MrOli | Hello All! Wondering if anyone had this problem before. Asterisk 1.8 + polycom SIP phones + PRI to PSTN. Calls out to PSTN work fine. Calls in to PSTN work fine. We can receive a call from PSTN, put on hold, place outbound call to PSTN, and transfer call.. works fine. However, when you set a "permanent forward" on the polycom, when an inbound call comes in from PSTN, Polycom tells Asterisk to dialout and transfer, which works, but n |
| 20:53.14 | ChannelZ | I'm already confused |
| 20:58.28 | leifmadsen | MrOli: part of your post was cut off btw |
| 20:58.34 | leifmadsen | "...and transfer, which works, but n" |
| 20:58.52 | WIMPy | leifmadsen, ChannelZ: I've been getting them for weeks. Always to *441412* or *441214* but with very interesting substitututions for the first *. |
| 20:59.26 | leifmadsen | nice |
| 20:59.31 | WIMPy | And all of them end with retransmission timeouts. |
| 21:00.07 | WIMPy | Looks like they don't even listen for an answer. |
| 21:00.37 | leifmadsen | yep same thing here |
| 21:01.37 | MrOli | oooops. here it is again... |
| 21:02.01 | MrOli | Hello All! Wondering if anyone had this problem before. Asterisk 1.8 + polycom SIP phones + PRI to PSTN. Calls out to PSTN work fine. Calls in to PSTN work fine. We can receive a call from PSTN, put on hold, place outbound call to PSTN, and transfer call.. works fine. |
| 21:02.17 | MrOli | However, when you set a "permanent forward" on the polycom, when an inbound call comes in from PSTN, Polycom tells Asterisk to dialout and transfer, which works, but no audio ! I have been able to replicate the issue at another install that has SIP trunking to PSTN... |
| 21:02.25 | MrOli | Help ? :) |
| 21:03.23 | ChannelZ | WIMPy: It must be some off-the-shelf script which is just probing and catalogging.. because I get a transmission error after each one too |
| 21:04.23 | WIMPy | It must be a botnet. I see them on different servers, including ones on dynamic IPs and they're coming from all sorts of IPs. |
| 21:04.29 | paulc | MrOli: how is the redirect happening? Like does the redirected call land in a context? you see the call going out on PSTN? |
| 21:05.09 | WIMPy | Isn;t it kind of stupid to do forwarding on phones if ou have Asterisk running? |
| 21:05.27 | MrOli | paulc: the call definitely goes out, we pick it up.. but no audio on either end. |
| 21:06.02 | paulc | MrOli: Can you pastebin your console output with debug and verbose set to 3, of the call coming in and going out but being dead audio |
| 21:06.46 | MrOli | the key info on the console is: Got SIP response 302 "Moved Temporarily" back from 192.168.111.202:5060 |
| 21:06.55 | WIMPy | But I think I read something about client transfers, either here or on Jira. |
| 21:07.10 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 21:07.11 | MrOli | paulc: then Now forwarding SIP/carrier-000000b6 to 'Local/14445556666@sa-phones' (thanks to SIP/7053-000000b7) |
| 21:07.39 | MrOli | paulc: i guess the only difference with doing everything manually is the "Local/" part |
| 21:07.55 | paulc | So what does Local/14444... do? You said PSTN but it's not, it's SIP trunking (or maybe I misread?). |
| 21:08.05 | paulc | What is your "canreinvite" setting for your SIP provider? |
| 21:08.38 | MrOli | paulc: definitely "no" for the phones... let me check for the SIP provider... (btw, this problem also happens with PRI) |
| 21:08.39 | Katty | NUGGET |
| 21:09.02 | Katty | this is a customer page, would Nugget please meet his party at the front of the store. |
| 21:09.05 | *** join/#asterisk hobodave_ (~hobodave@pdpc/supporter/professional/hobodave) |
| 21:09.09 | MrOli | paulc: canreinvite=no |
| 21:09.17 | MrOli | paulc: for the SIP provider in sip.conf |
| 21:09.26 | ChannelZ | It's probably something weird with the Local channel.. why is it doing that? |
| 21:09.40 | ChannelZ | and in 1.8 it's "directmedia" now, by the way |
| 21:09.43 | paulc | MrOli: When you say "also happens with PRI" can you explain? PRI call in --> phone --> call forwarding... to where? SIP provider? other PSTN trunk? |
| 21:10.24 | paulc | MrOli: There's something about local channels and /n so they don't optimise themselves out of the flow.. but it's fuzzy in my head right now.. |
| 21:10.42 | MrOli | paulc: when the connection to the PSTN is a PRI, the problem also occurs: inbound oubound works fine. can transfer calls out.. but when the polycom does the forward, then no audio once the call is connected |
| 21:10.44 | paulc | MrOli: Why not implement call forwarding in the dialplan, as WIMPy suggested? That gives you the ultimate control without having to worry about 302s etc |
| 21:11.42 | MrOli | paulc: users want to use the keys on the phone to do the forward; gives them a visual feedback on the display.. and i guess since it's right here in front of them with a big key saying "Forward" i can't blame them |
| 21:11.49 | paulc | MrOli: depending on your dialplan, you could have the target extension (ie forward destination) call Answer then do the outdial, in the sa-phones context. but it's not the ideal way of doing things. |
| 21:11.53 | WIMPy | As I already said: I think I read about that not working somewhere, so I'd try another Asterisk version. Or better do it on the server, off course. |
| 21:12.24 | MrOli | is there a way to tell asterisk not to use Local/ channel to handle the 302 ? |
| 21:12.41 | WIMPy | You should be able to configure the phone so it sends ar request to Asterisk instead. |
| 21:12.44 | ChannelZ | Why is it being dialed as a local channel to begin with? |
| 21:12.51 | *** join/#asterisk BMJ (~bjohns@c-98-251-113-67.hsd1.ga.comcast.net) |
| 21:12.52 | *** mode/#asterisk [+o BMJ] by ChanServ |
| 21:13.12 | MrOli | Channelz: precisely ! is that a default/changeable behavior in Asterisk ? |
| 21:13.15 | leifmadsen | yes |
| 21:13.27 | leifmadsen | that's how it redirects the call to the dialplan |
| 21:13.32 | ChannelZ | I guess I never paid attention |
| 21:14.06 | MrOli | leifmadsen: does it mean we need to add a new piece of dialplan to handle "302" requests from SIP phoens ? |
| 21:14.26 | MrOli | leifmadsen: to bypass this default behavior of asterisk using Local/ channel ? |
| 21:14.36 | leifmadsen | that's not possible |
| 21:14.39 | leifmadsen | to my knowledge |
| 21:14.52 | MrOli | sad |
| 21:15.08 | leifmadsen | it's redirecting to the appropriate context for outbound dialing right? |
| 21:15.19 | ChannelZ | Let me test mine, it was working last I knew. |
| 21:15.27 | Nugget | eyes Katty |
| 21:15.30 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 21:15.31 | MrOli | leifmadsen: it does, because that's the only way it knows where to dial out... |
| 21:15.33 | leifmadsen | ya I've used it on my polycoms before it seems to work.... |
| 21:15.42 | paulc | MrOli: Try adding an Answer to the extension that the redirect matches, to see if "media flowing through Asterisk" helps |
| 21:15.54 | paulc | It's totally the wrong way to solve the problem, but may aid in troubleshooting |
| 21:15.54 | MrOli | yes, i remember using it fine a while ago.. maybe it's a new behavior on 1.8 |
| 21:16.06 | MrOli | paulc: i definitely tried that |
| 21:16.13 | MrOli | paulc: didn't help.. :( |
| 21:16.34 | paulc | How about an answer, followed by a Playback - does the inbound call hear that audio? |
| 21:16.45 | leifmadsen | might want to do an rtp debug to see how or where the audio is actually flowing |
| 21:16.55 | MrOli | i'm still so confused about this "Local/" stuff.. i definitely need to read up on that and understand exactly what those are and how they work. I know Leif love those :) |
| 21:17.06 | MrOli | rtp debug is a great idea ! |
| 21:17.14 | MrOli | i'm going to test that out ! |
| 21:17.49 | MrOli | is that something to do at the packet level (tcpdump) or asterisk console ? |
| 21:17.51 | ChannelZ | worked here.. albeit not PRI. |
| 21:18.09 | WIMPy | No audio or one-way audio? |
| 21:18.21 | MrOli | WIMPy: no audio eitherway |
| 21:18.35 | ChannelZ | PSTN->DAHDI->Asterisk->SIP->Phone (forwarded)->DAHDI->PSTN |
| 21:19.05 | ChannelZ | Although it sounded a mess, a bit echoey, not sure what that's all about. |
| 21:20.00 | WIMPy | That's why you should do it on the server, not on the phone. |
| 21:21.10 | MrOli | WIMPy: i agree with you. a little dialplan to handle call forwards and leveraging AstDB is great for that, and also to be able to change/remove forwards remotely |
| 21:21.33 | WIMPy | Yes. |
| 21:21.39 | MrOli | WIMPy: but users have already gotten the manual of the phone and figured out that feature; it will be hard to change behavior now :( |
| 21:21.53 | WIMPy | And it should be possible to do that with the phones keys/menu. |
| 21:22.14 | WIMPy | i.e. reconfigure the keys. |
| 21:22.21 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 21:22.44 | Nugget | Katty: pout |
| 21:23.07 | ChannelZ | At te end of the day there shouldn't be much of a difference though. |
| 21:23.14 | leifmadsen | MrOli: you could detect the forward request and ignore it :) |
| 21:23.23 | timeshell | Where is the appropriate place to set the txgain/rxgain in 1.8.x? I keep changing it in chan_dahdi,conf, but it doesn't seem to do anything there. When I set it in CLI using set dahdi hwgain / swgain I can set it correctly. |
| 21:23.30 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:23.35 | timeshell | But it seems to completely ignore the chan_dahdi.conf |
| 21:23.43 | leifmadsen | <PROTECTED> |
| 21:23.44 | leifmadsen | <PROTECTED> |
| 21:23.44 | leifmadsen | <PROTECTED> |
| 21:23.44 | leifmadsen | <PROTECTED> |
| 21:23.44 | leifmadsen | <PROTECTED> |
| 21:23.45 | WIMPy | There should: By default more than 40ms delay. |
| 21:23.56 | leifmadsen | (that's from the Dial() application documentation) |
| 21:24.17 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 21:24.19 | ChannelZ | Because after the phone redirects and Asterisk dials the local channel, I wind up with "-- Native bridging DAHDI/1-1 and DAHDI/4-1" |
| 21:24.22 | leifmadsen | might be possible to disable the forwarding options n the polycom phones, although to be honest I have never tried (or looked) |
| 21:24.37 | leifmadsen | the Local channel should optimize itself out of the path anyways once it connects the calls |
| 21:24.38 | MrOli | leifmadsen: that's the help for the DIAL command ? |
| 21:24.47 | leifmadsen | MrOli: yep -- core show application dial |
| 21:24.47 | ChannelZ | 1 being what I dialed in on and 4 being what it re-dialed out of |
| 21:24.58 | WIMPy | ChannelZ: Ok, if that works it should be the same. |
| 21:25.17 | ChannelZ | Yeah. The phone isn't actually placing a new call or anything. |
| 21:25.43 | leifmadsen | MrOli: then use DumpChan() to see what channel variables are available to you -- you might have a way of detecting that a forward request was attempted, and if so, to determine what to do with the call after |
| 21:26.01 | leifmadsen | ok gotta go start dinner |
| 21:26.03 | leifmadsen | is away |
| 21:26.10 | ChannelZ | have fun |
| 21:26.13 | MrOli | leifmadsen: and play something like "please use *1234 to fordward calls"... |
| 21:26.15 | MrOli | hmmk |
| 21:26.17 | MrOli | leifmadsen: thanks a lot |
| 21:26.28 | MrOli | leifmadsen: bon appétgit |
| 21:26.35 | leifmadsen | MrOli: well only the caller would hear that prompt :) |
| 21:26.36 | WIMPy | For the pri it might help to enable native bridging, as ChannelZ has working. |
| 21:26.55 | leifmadsen | MrOli: but you could then look in the database and see if there is data to forward the call |
| 21:26.59 | MrOli | WIMPy: native bridging is b-to-b-channel ? |
| 21:27.09 | WIMPy | yes |
| 21:27.24 | MrOli | okidok |
| 21:27.31 | WIMPy | That won't help with the itsp, obviousely. |
| 21:27.41 | MrOli | yes |
| 21:27.58 | jkroon | WIMPy, /n to Local/ prevents it optimizing itself out. |
| 21:28.05 | MrOli | i'm going to start with looking at RTP packets as leif suggested, seems to be the option that requires the least changes :) |
| 21:28.08 | jkroon | might be what you're looking for :) |
| 21:28.30 | MrOli | i guess this is done at nic card level ? (tcpdump) |
| 21:28.36 | jkroon | close |
| 21:28.45 | WIMPy | But I'd take a closer look at the phones manual. It should be possible to tell it to dial some feature code when it wats to do forwarding. |
| 21:28.56 | jkroon | for all practicaly purposes MrOli the answer is yes. |
| 21:29.30 | MrOli | jkroon: thanks :) will try now... |
| 21:29.42 | MrOli | dumps packets.. YAAYY |
| 21:30.51 | jkroon | MrOli, you can also use the asterisk CLI for debugging rtp stream issues. |
| 21:31.06 | jkroon | "rtp set debug on" iirc. there are ways to limit it to specific IPs. |
| 21:31.29 | WIMPy | Or as I already suggested: Try another Asterisk version. |
| 21:32.09 | MrOli | jkroon: ah this sounds more convenient! thanks a lot! |
| 21:33.46 | timeshell | in asterisk, there needs to be a dahdi show hwgain/swgain as well as hwgain so one gain see what the current value is. |
| 21:33.59 | timeshell | sorry, as well as SET |
| 21:41.34 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 21:42.11 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 21:43.50 | ChannelZ | Actually it seems like the easiest thing to test is to setup a dummy exten that does a Dial() back out and see if that even solves your problem |
| 21:44.22 | Katty | i need an opinion |
| 21:44.26 | Katty | what's the better derby name |
| 21:44.29 | *** join/#asterisk war9407 (war@c-71-62-61-74.hsd1.va.comcast.net) |
| 21:44.34 | Katty | Colonel Lynn Nixx or Sergeant Squirrelie |
| 21:47.50 | *** join/#asterisk pdtpatri1k (~ptaylor@mainstwan.farheap.com) |
| 21:48.34 | ChannelZ | I like Sergeant |
| 21:48.52 | pdtpatri1k | Question .. i'm getting these errors.. is there an app i need to compile/instal ?? http://pastebin.com/iDv7Xswh |
| 21:49.28 | ChannelZ | what is /tmp/localhost |
| 21:49.33 | ChannelZ | that seems wierd |
| 21:50.04 | pdtpatri1k | that is where the .wav files are being stored based on the php script |
| 21:50.11 | Qwell | ls -l /tmp/localhost/en/cache_tviLkNuIsWKEJgICX+xLAi==.* |
| 21:50.14 | Qwell | Show us. |
| 21:50.49 | *** join/#asterisk pdtpatrick_ (~pdtpatric@mainstwan.farheap.com) |
| 21:50.52 | *** part/#asterisk pdtpatrick_ (~pdtpatric@mainstwan.farheap.com) |
| 21:51.12 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 21:51.49 | *** join/#asterisk pdtpatrick_ (~pdtpatric@mainstwan.farheap.com) |
| 21:51.54 | pdtpatrick_ | Here you go |
| 21:51.55 | pdtpatrick_ | http://pastebin.com/2xy8qe5G |
| 21:52.39 | Qwell | that isn't what I asked for |
| 21:53.05 | pdtpatrick_ | ur just globbing no ? |
| 21:53.12 | WIMPy | Are they propperly formatted riff files? Where do they come from? |
| 21:53.24 | Qwell | copy/paste the command I gave |
| 21:54.03 | pdtpatrick_ | root@vxiprod:/tmp/localhost/en# ls -l /tmp/localhost/en/cache_tviLkNuIsWKEJgICX+xLAi==.* |
| 21:54.03 | pdtpatrick_ | -rw-r--r-- 1 root root 0 2011-08-16 14:42 /tmp/localhost/en/cache_tviLkNuIsWKEJgICX+xLAi==.wav |
| 21:54.08 | Qwell | 0 |
| 21:54.24 | ChannelZ | That's a lovely name |
| 21:54.53 | pdtpatrick_ | lol yeah i didn't write the php script.. i'm just trying to get the system back up. But saw the error in asterisk and trying to see if there's something that can be done |
| 21:54.58 | WIMPy | that makes a no |
| 21:55.03 | Qwell | You can tell Asterisk to play a valid sound file. |
| 21:55.54 | pdtpatrick_ | you're saying the .wav file is not a valid sound file? |
| 21:56.07 | Qwell | you're saying a 0 byte file is valid? |
| 21:56.09 | ChannelZ | what does 'file' say about one of them? |
| 21:56.26 | ChannelZ | oh..yes heh |
| 21:56.46 | ChannelZ | Shortest song EVAR |
| 21:57.11 | Qwell | Somebody accidentally ran silence suppression on 4'33". |
| 21:57.11 | pdtpatrick_ | Quell they are not all 0... |
| 21:57.12 | pdtpatrick_ | root@vxiprod:/tmp/localhost/en# file cache_ZtvO3K6NR4sDf0lDjKoRA5\=\=.wav |
| 21:57.12 | pdtpatrick_ | cache_ZtvO3K6NR4sDf0lDjKoRA5==.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 21:57.13 | WIMPy | Oh, let's start the null vs. nothing debate :-) |
| 21:57.29 | ChannelZ | it's probably banging into the ones that are and barfing |
| 21:57.30 | Qwell | pdtpatri1k: cache_ZtvO3K6NR4sDf0lDjKoRA5==.wav wasn't in that pastebin. |
| 21:58.38 | pdtpatrick_ | here's all the files |
| 21:58.38 | pdtpatrick_ | http://pastebin.com/ZZ0TrSRd |
| 21:59.08 | Qwell | well it's pretty obvious which ones are broken |
| 21:59.21 | ChannelZ | And low and behold the one you originally pasted is one that is 0 bytes |
| 21:59.33 | Qwell | ChannelZ: you're like 5 minutes behind. :p |
| 21:59.58 | ChannelZ | No just wondering why this is still a mystery 5 minutes later :) |
| 22:00.12 | Qwell | pdtpatrick_: Here's my point. You've only shown us one error, in one file. That file happened to be 0 bytes. That's obviously not a coincidence. |
| 22:03.09 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 22:03.21 | pdtpatrick_ | Thanks |
| 22:03.55 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 22:12.25 | Merlin | does anyone know how Asterisk's support for TTY/TDD works on the Zaptel driver? The Wiki has information as of 2006, which is like asterisk 1.4, but the code doesn't appear to have changed as of 1.8. i'm wondering how asterisk passes the TDD data to a SIP channel, where does it appear in the SIP packet, if anywhere? |
| 22:34.21 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 22:36.35 | WIMPy | Has anyone ever tried (or thought about) using queues with custom devstates to dynamically control where a call should go while waiting for an answer? Do you think that works? |
| 22:38.09 | WIMPy | I need someting where agents can take multiple calls, which queues can't do if I understand that correctly. |
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| 23:08.03 | citywok | WIMPy: you can send multiple calls to a single agent, just set ringinuse=yes |
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| 23:10.11 | WIMPy | Yes, but I don't (yet) see how I can make it try those with the least active calls first. |
| 23:10.34 | WIMPy | That is multiple agents, but not all of them. |
| 23:10.47 | leifmadsen | use Local channels to determine if you should call them or not |
| 23:11.00 | leifmadsen | I'm not sure I quite follow the logic or what you're trying to do |
| 23:11.48 | WIMPy | What's the idea with using local channels? |
| 23:12.26 | chazzam | use GROUP_COUNT functionality to determine how many calls they can take via local channels |
| 23:12.34 | chazzam | will have to /n the local channels though |
| 23:12.36 | WIMPy | I need to try a few agents simultaneousely to quickly find one that is able to take the call. |
| 23:12.52 | chazzam | you would have to just have ringall as the mode then I would think |
| 23:13.02 | leifmadsen | ya |
| 23:13.11 | leifmadsen | use ringall, or use a Local channel to call various groups of people |
| 23:13.12 | chazzam | or code your own ring strategy |
| 23:13.31 | WIMPy | I'm only 3/4 way through reading, but I can't configure how many active calls an agent can have, can I? |
| 23:13.45 | leifmadsen | if you use Local channels, then you can call the Local channel then you can choose how many people (and what people) to call |
| 23:13.58 | leifmadsen | WIMPy: what do you mean "how many"? |
| 23:14.09 | chazzam | same => n,Set(GROUP(QUEUE)=${LKUP_EXTEN}) |
| 23:14.11 | chazzam | same => n,GotoIf($[${GROUP_COUNT(${LKUP_EXTEN}@QUEUE)} > 1]?busy) |
| 23:14.12 | WIMPy | Yes, but I can't modify that while they're ringing. |
| 23:14.15 | leifmadsen | use a Local channel in combination with a GROUP() |
| 23:14.22 | leifmadsen | modify it before you ring them |
| 23:14.26 | chazzam | you can change that > 1 to determine how many calls they could get in a local channel |
| 23:14.32 | *** join/#asterisk hobodave_ (~hobodave@pdpc/supporter/professional/hobodave) |
| 23:15.17 | chazzam | same => n(busy),Busy(20) |
| 23:15.22 | WIMPy | My idea was that I would use ringall for only not-in-use channels and then control the currently called agents via devstates. |
| 23:15.51 | chazzam | so have all agents log-in using local channels |
| 23:15.55 | chazzam | have ringinuse=no |
| 23:16.22 | chazzam | do you check on devstate, and if you decide its busy then call Busy() instead of dialing hte agent in your dialplan that handles the other end of the local channel |
| 23:16.40 | leifmadsen | WIMPy: you either call all the agents (strategy=ringall) that are on (ringinuse=yes) or not on (ringinuse=no) a call |
| 23:16.48 | leifmadsen | I don't think you need DEVSTATE() at all |
| 23:17.35 | chazzam | flees for the day |
| 23:18.05 | WIMPy | The thing is I neet to call agents that are already talking, but I don't want to ring all, 1) in order not to disturb too many and 2) to get the next call to the next set of agents at the same time. |
| 23:18.10 | WIMPy | need |
| 23:18.29 | leifmadsen | so use a Local channel then |
| 23:18.42 | leifmadsen | it's the only way to control logic prior to delivery |
| 23:18.46 | leifmadsen | (but after the queue) |
| 23:18.54 | leifmadsen | maybe you don't even want to use a Queue() |
| 23:19.26 | WIMPy | But I can't add another agent then without cancelling that dial and place a new one, right? |
| 23:19.34 | WIMPy | Maybe. |
| 23:19.39 | WIMPy | Well, probably. |
| 23:19.44 | leifmadsen | huh? |
| 23:20.12 | WIMPy | But that seems to be a solution to change the called parties dynamically. |
| 23:20.14 | leifmadsen | Use a Local channel for each group of agents you want to call |
| 23:20.23 | leifmadsen | that's what you're trying to do |
| 23:20.34 | leifmadsen | I think you're talking to generally now for a real answer |
| 23:20.54 | leifmadsen | you probably need to actually write out the desired call flow |
| 23:21.12 | WIMPy | If I have a ringall to only not-in-use agents and one previousely in-use agent becomes not-in-use, we he get a ring then? |
| 23:21.25 | *** join/#asterisk hobodave_ (~hobodave@pdpc/supporter/professional/hobodave) |
| 23:21.31 | leifmadsen | but basically the answer is, "some combination of Local channels, Queue()'s, DEVSTATE(), dialplan logic, queues.conf and queuerules.conf magic" will do what you want |
| 23:21.51 | leifmadsen | WIMPy: not for that attempt |
| 23:21.56 | WIMPy | Oh |
| 23:22.06 | leifmadsen | he will for the next attempt, which should be only a few seconds later |
| 23:22.15 | leifmadsen | I don't ring agents for more than about 1-2 rings |
| 23:22.41 | WIMPy | Well, it can take a little time for them to answer. |
| 23:23.04 | leifmadsen | sure... about 2 rings worth |
| 23:23.27 | leifmadsen | otherwise, that previously in use agent will not become part of the group of not in use people being rung until the next attempt |
| 23:23.31 | WIMPy | No, I need more time. |
| 23:23.49 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:23.55 | WIMPy | I had hoped queues were more dynamic. |
| 23:24.17 | leifmadsen | once a call is placed, there is no dialplan logic or call logic going on |
| 23:24.30 | leifmadsen | it won't just start ringing additional agents while it's ringing other agents |
| 23:24.36 | WIMPy | That sounds like somethig I could do within the dialplan with repeded Dial()s. |
| 23:24.51 | leifmadsen | exactly why I was saying you might not even want to use Queue() |
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| 23:25.13 | leifmadsen | if you're ringing multiple agents, do you *really* need to start ringing other agents during that cycle? |
| 23:25.27 | leifmadsen | I can think of how you can do it though |
| 23:25.34 | WIMPy | I thought they would allow a more dynamic call distribution. Looks like I expected too much there. |
| 23:25.34 | leifmadsen | but it's so fuckin' hacky :) |
| 23:25.51 | leifmadsen | queue's work based on time, and execute things within time slots |
| 23:25.58 | WIMPy | Yes, that would be sensible. |
| 23:26.20 | leifmadsen | action is taken based on the state of things just prior to executing something during that time slot |
| 23:26.36 | WIMPy | If I get a better match, I want to call there, without terminationg other rings. |
| 23:26.39 | leifmadsen | when it ends, it re-evaluates and exectues based on the status prior to the next time slot |
| 23:26.49 | leifmadsen | ok, then you need to use multiple Local channels |
| 23:27.00 | leifmadsen | ok, try to follow this logic.... |
| 23:27.02 | leifmadsen | (it gets messy) |
| 23:27.26 | WIMPy | Yes, I got the messy feeling already... |
| 23:27.38 | leifmadsen | Dial(Local/check1@ringall&Local/check2@ringall&Local/check3@ringall) |
| 23:27.56 | leifmadsen | then in each of those extensions, spread out your checks with a Wait() |
| 23:28.06 | leifmadsen | Wait(5), Wait(10), Wait(15), etc.... |
| 23:28.23 | leifmadsen | then after the wait, check the status of channels, and determine who to call (based on DEVSTATE() and GROUP_COUNT()) |
| 23:28.44 | leifmadsen | if someone became free during the check2 or check3, then you can start ringing them without cancelling the other calls |
| 23:28.46 | WIMPy | Yes, doing it interleaved might be a good idea. |
| 23:30.23 | leifmadsen | just have as many interleaved channels as you want for the cycle |
| 23:30.31 | leifmadsen | lather, rinse, repeat |
| 23:30.46 | leifmadsen | the initiating local channel could even be called from the Queue() if you really wanted |
| 23:30.53 | WIMPy | is thinking... |
| 23:31.38 | WIMPy | Ok, it is possible to redirect a ringing channel. |
| 23:31.42 | leifmadsen | yes |
| 23:31.50 | leifmadsen | I think it's Redirect() or Transfer() in the dialplan |
| 23:31.55 | leifmadsen | something like that |
| 23:32.02 | leifmadsen | redirect from which side I guess? |
| 23:32.33 | WIMPy | So I think I forget about queues and just dial the first n channels then dynamically redirecting them to others as it seems appropriate, either for a better match or due to timeout. |
| 23:33.26 | WIMPy | There had to be a catch... I might lose channels if someone is unavailabel. |
| 23:34.15 | leifmadsen | lose channels? |
| 23:34.38 | WIMPy | It I Dial the destinations directly. |
| 23:34.46 | *** join/#asterisk hobodave_ (~hobodave@pdpc/supporter/professional/hobodave) |
| 23:34.48 | leifmadsen | please elaborate |
| 23:34.50 | WIMPy | So I have to use local channels. |
| 23:35.24 | WIMPy | The Idea was to just Dial(sip/agent1&sip/agent2&sip/agent3) |
| 23:35.45 | WIMPy | Then whe I decide, redirect the channel to agent1 to agent2 and so on. |
| 23:36.13 | leifmadsen | how are you deciding? |
| 23:36.29 | WIMPy | Which is fine, unless say agent2 is (temporarily) unreachable. Then the cahnnel is gone and can't be redirected any more. |
| 23:36.42 | leifmadsen | until you try it again |
| 23:36.48 | leifmadsen | in the next time slot |
| 23:37.02 | leifmadsen | because the DEVSTATE() will not be Not In Use |
| 23:37.02 | WIMPy | Indee, but that's what I'm trying to avoid. |
| 23:37.15 | leifmadsen | I don't understand how you'll ever avoid it |
| 23:37.35 | leifmadsen | the only possible way then, is to write a program to monitor channel states in the AMI |
| 23:37.48 | leifmadsen | then program the monitoring and redirecting from there |
| 23:37.51 | leifmadsen | you'll never do it in the dialplan |
| 23:38.02 | WIMPy | I think it should work, but I'm not sure the guy that needs it, needs it as much as it's work :-) |
| 23:38.19 | WIMPy | Yes, that was the idea. |
| 23:38.26 | leifmadsen | to solve your problem is certainly going to take more than a few minutes, that's for sure :) |
| 23:38.36 | WIMPy | yes |
| 23:38.37 | leifmadsen | I'd quote him 10-15 hours |
| 23:38.58 | leifmadsen | it'd be quite the custom solution |
| 23:39.02 | WIMPy | I guess that might be more. |
| 23:39.15 | leifmadsen | depends how configurable it needs to be I guess :) |
| 23:39.19 | WIMPy | The selection logic doesn't come for free, either. |
| 23:39.40 | leifmadsen | aye |
| 23:39.51 | leifmadsen | sounds like a good week or two project if done right |
| 23:40.21 | WIMPy | Well, let's see how important it is for them. |
| 23:41.05 | carrar | moo |
| 23:41.21 | WIMPy | But it sounds like an interesting thing anyway :-) |
| 23:41.21 | carrar | It's only money! |
| 23:57.50 | WIMPy | What happens if I configure a queue to only call one agent at a time and the calls fails for whatever reason. Will the next run start immediately then or will it wait for the configured timeout |
| 23:57.52 | WIMPy | anyway? |
| 23:59.00 | WIMPy | is thinking about the possibility to reverse the ting and put a local cahnnel in front and put the caller in to multiple queues. |