| 00:00.41 | pcAngel | is there any way in a sip peer entry to have it automatically register on a second server, when a client connects to that peer entry, and to have all SIP signalling bridged between the server and peer? |
| 00:01.44 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 00:02.05 | ChrisInSydney | g'day all |
| 00:02.13 | ChannelZ | A dingo ate my baby! |
| 00:02.28 | ChannelZ | (or was it 'took'? now I don't remember.) |
| 00:02.37 | ChrisInSydney | ChannelZ: :D |
| 00:02.46 | ChrisInSydney | ate |
| 00:03.05 | ChrisInSydney | if it wasn't, thats the way id like to remember |
| 00:04.07 | WIMPy | pcAngel: The only way I see would be to watch for the registration on AMI and then create a register statement and 'sip reload'. So that's probaly a rather bad idea. |
| 00:04.32 | ChrisInSydney | it was a complete miscarrige of justice that whole thing. Now Ayers Rock or Ularu as it is now known is off limits for campers and they have to stay in a dedicated camping grounds abount 20 mins drive from the "sacred site" |
| 00:05.36 | ChannelZ | It's For The Children. |
| 00:05.48 | pcAngel | I can see that working, and keeping track of if it's already registered to make sure that sip reloads aren't done with a high frequency |
| 00:06.38 | ChrisInSydney | Speaking of Elephants, which we weren't. I've got a cracker of an issue. For some strange reason, a PBX in a Flash system has started doing random 90 second SIP dropouts. Was working fine, No updates or anything. Just started doing it. Been like this for a week. |
| 00:06.44 | pcAngel | alright, I have a few options now to run test cases on tomorrow |
| 00:06.52 | pcAngel | thanks for your help again, WIMPy |
| 00:06.54 | coreyf1513 | pcAngel: asterisk is a b2bua, not a router. to create a front-end registration proxy I recommend looking at a sip router (kamailio/opensips are opensource options). warning - they are difficult to setup. |
| 00:06.59 | ChrisInSydney | I've split the network with the second NIC and a second switch |
| 00:07.16 | ChrisInSydney | does both segments and SIP trunks |
| 00:07.39 | ChrisInSydney | off, then on and fine for a few hours. Active calls go silent |
| 00:07.40 | pcAngel | thanks corey, right now using either of those would create a new single point of failure, otherwise it'd have been my first route |
| 00:07.58 | ChrisInSydney | Any siggestions anyone ?? |
| 00:08.40 | coreyf1513 | pcAngel: sip routers can run stateless in a heartbeat config for redundancy |
| 00:08.48 | WIMPy | ChrisInSydney: Lurk on the line and wait if you see something suspicious. |
| 00:09.59 | pcAngel | the servers are at two different remote sites, and anything I've read on heartbeat has made it seem as if they need to be at the same site, since they check on each other based on which one currently has a local IP address |
| 00:10.59 | pcAngel | The data center our primary site is at has had a DDOS attack take down their network for an hour last month and for an hour in December, so doing single site redundancy no longer keeps us up |
| 00:12.43 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 00:13.01 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
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| 00:18.53 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 00:25.30 | ChrisInSydney | WIMPy: Are you referring to Wireshark ?? |
| 00:27.13 | pcAngel | Chris: try sip set debug peer <peerID> |
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| 00:35.12 | ChrisInSydney | is there a way I can dump the SIP debug messages to a separate file ?? Ast 1.6.2.something |
| 00:35.14 | ChrisInSydney | ?? |
| 00:44.12 | fling | ChannelZ: hey |
| 00:44.25 | fling | my iax2 auth failing somewhy, I can't get why :[ |
| 00:45.42 | fling | CAUSE : No authority found http://dpaste.com/919095/ |
| 00:48.11 | fling | from: http://bpaste.net/show/76501/ to: http://bpaste.net/show/76502/ |
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| 00:50.00 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:51.13 | fling | ChrisInSydney: asterisk -vvvr | tee /tmp/asterisk_output.log |
| 00:51.20 | fling | ChrisInSydney: > sip set debug on |
| 00:51.53 | WIMPy | ChrisInSydney: yes, wireshark might be the easiest to read. |
| 00:54.12 | fling | WIMPy: what am I doing wrong? |
| 00:54.22 | WIMPy | fling: I don't see matching users in your configs. |
| 00:55.40 | fling | WIMPy: user=test secret=123 |
| 00:55.53 | fling | WIMPy: calling from hh_mirror to u-kit_hatchery |
| 00:56.46 | [TK]D-Fender | user != username |
| 00:57.01 | [TK]D-Fender | Any other parameter names you'd like to mix up? |
| 00:57.44 | fling | whoops sorry |
| 00:58.15 | fling | otoh it works for u-kit_barnaul peer wtf |
| 00:58.18 | fling | is fixing |
| 00:58.43 | [TK]D-Fender | Stop using templates for this as well. |
| 00:59.15 | fling | why? |
| 00:59.39 | WIMPy | Now I already closed them. |
| 01:00.03 | [TK]D-Fender | Because when you can find anything adding levels of obsurity to things isn't doing you any good. |
| 01:00.17 | [TK]D-Fender | Make your peers complete to themselves and just be done with it. |
| 01:00.22 | [TK]D-Fender | And remove the commented out junk |
| 01:00.41 | [TK]D-Fender | Top reason for not finding problems in configs : they're loaded with crap |
| 01:01.25 | fling | is removing crap |
| 01:01.43 | artyx | if it smells liek fish its probably just carp |
| 01:02.00 | nightrid3r | some conf files are like theyr creator: full of crap :) |
| 01:02.42 | artyx | While the creator can alleviate that symptom on its own, the config file seldom gets the opportunity |
| 01:03.59 | WIMPy | And don't forget to flush. |
| 01:11.12 | artyx | If i'm usign a sangoma card. theorder of install should be dahdi, wanpipe, asterisk? |
| 01:17.58 | fling | [TK]D-Fender: http://bpaste.net/show/76506/ http://bpaste.net/show/76507/ |
| 01:18.05 | fling | [TK]D-Fender: still the same error |
| 01:18.33 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 01:19.06 | nightrid3r | artyx: you don't need dahdi unless you want to add other cards later |
| 01:19.56 | nightrid3r | so its wanpipe, asterisk |
| 01:22.04 | [TK]D-Fender | fling: Full call debug. Associated dialplan. |
| 01:22.06 | artyx | Well wanpipe requires dahdi to be installed :( i tried it just unpacked |
| 01:22.37 | [TK]D-Fender | [20:19]nightrid3rartyx: you don't need dahdi unless you want to add other cards later <-Wrong |
| 01:23.34 | artyx | first itried unpacking, it errored.... then i tried compiling, a little further, but error.. now i did dahdi install, and am rebuilding wanpipe.. so far so good. but it takes about 7 - 10 mins a try with this slow atom |
| 01:24.19 | artyx | that doesnt even count the bad symlink somewhere deep inside my kernel modules dir ;) |
| 01:25.27 | nightrid3r | <---- stupid |
| 01:25.55 | nightrid3r | forget that i install wanpipe on a distro so dahdi was already there :( |
| 01:26.24 | artyx | Yeh.. i want to try rhel6/centos6 64 bit, asterisk, and selinux with some other frontend |
| 01:26.32 | artyx | call it an excercise in pain management :P |
| 01:27.38 | nightrid3r | wow, extreme :) |
| 01:28.00 | artyx | each one on its own not insurmountable, but everythign together sure adds up :P |
| 01:30.58 | nightrid3r | #include redbull.h :) |
| 01:35.06 | [TK]D-Fender | <PROTECTED> |
| 01:35.13 | *** join/#asterisk deo_ (~deo@222.127.13.226) |
| 01:36.28 | artyx | wanpipe is kicking and screaming [TK]D-Fender |
| 01:36.32 | fling | [TK]D-Fender: http://bpaste.net/show/76509/ http://dpaste.com/919126/ http://dpaste.com/919128/ |
| 01:39.41 | [TK]D-Fender | fling: What are the versions of these 2 servers? |
| 01:40.11 | fling | [TK]D-Fender: from: 11.2.1 to: 11.2.1 |
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| 01:46.00 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 01:49.45 | fling | [TK]D-Fender: but I may call to another peer, 11.2.1 too > http://dpaste.com/919131/ http://dpaste.com/919133/ |
| 01:50.23 | [TK]D-Fender | fling: consolidate down to 1 peer per side and set them identical except for the HOST IP |
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| 01:58.15 | liquidamber | hey guys. I have a DID which is supposed to be able to get 25 calls at a time, but if i get more than a few, it gives a slow busy. the setup is completely vanilla. any ideas where to start? |
| 01:58.46 | liquidamber | i verified with the DID provider that its something with my setup |
| 02:09.07 | [TK]D-Fender | Based on ...? |
| 02:09.38 | liquidamber | yeah, well, nothing :P but since i'm not an asterisk guru i assume its my fault |
| 02:16.06 | [TK]D-Fender | ~assume |
| 02:16.06 | infobot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav It makes an (ass) out of (u) and (me) |
| 02:16.13 | [TK]D-Fender | ^ not a great place to start from |
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| 02:16.23 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 02:16.35 | [TK]D-Fender | liquidamber: A better place to start from is actually looking at CLI for the call attempt even TRYING to come in and seeing what happens |
| 02:17.04 | [TK]D-Fender | liquidamber: If proper debug is enabled and nothign cones in and other calls in/out of it work then it could very likely be provider-side |
| 02:17.25 | [TK]D-Fender | liquidamber: Which..... you should not jump to or even leave as probable until you've actually looked |
| 02:17.27 | liquidamber | thanks, any ideas on how to recreate that condition |
| 02:17.37 | liquidamber | because i dont have 10 phones to call myself with :) |
| 02:17.45 | [TK]D-Fender | liquidamber: Apparently CALLING IN.... can lead to it. So CALL IN. |
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| 03:30.16 | fling | [TK]D-Fender: works with the same [this-peer-name] in iax.conf |
| 03:34.10 | fling | [TK]D-Fender: but why :[ |
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| 04:52.03 | artyx | Anyone still awake? I have this problem with a dahdi card. wondering how i can fix it. If i call in on a dahdi trunk, then let pass to a did/extension. then it goes to voicemail, if i hang up before that prompt finishes, my dahdi line is frozen and no further incoming calls can happen |
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| 05:18.22 | artyx | Also, is there a way to ignore call waiting on a dahdi trunk? (short of having phone co remove it) |
| 05:22.36 | *** join/#asterisk timahvo1 (~rogue@197.179.190.235) |
| 05:23.49 | coreyf1513 | artyx: you can usually disable call waiting for outbound calls with a *XX code, you could add it to your trunk dial.. check your phone book or telco website for special codes, you might be able to call a special number to disable call waiting for the line |
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| 05:27.43 | artyx | I am hoping to block it on inbound calls to the trunk |
| 05:27.57 | artyx | outbound i know about *70 but inbound i think i have to call pstn and have them turn it off |
| 05:28.24 | ChannelZ | around here you have to pay extra for it |
| 05:28.41 | artyx | to turn it off, or on? its part of some package i have for 5$ extra/mo |
| 05:29.50 | ChannelZ | I mean to have call waiting. If you don't want it, call them to stop paying for it if that's the case with you too |
| 05:30.59 | ChannelZ | Either way if it's in a package of other junk you do want, get them to turn it off. |
| 05:31.34 | artyx | so going back to the original question, there is not a way to ignore call waiting on the inbound line? |
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| 05:42.08 | artyx | No. it tells you what the exact hardware is on the system pci bus |
| 05:42.12 | artyx | oops. wrong window |
| 05:42.13 | ChannelZ | You call the telco. Maybe they have a feature code to turn it off but that'd be uncommon (besides the normal per-call outgoing.) |
| 05:42.57 | ChannelZ | Either way they are going to have to tell you or do something about it. |
| 05:45.14 | fling | [TK]D-Fender: iax2 auth works if I put context name into [], like [mycontextnamehere] |
| 05:45.40 | fling | [TK]D-Fender: but it is not working if I will put some random word there |
| 05:47.49 | fling | looks like I'm doing it wrong |
| 05:47.56 | fling | maybe I'm dialing bad |
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| 05:59.19 | ChannelZ | I haven't really been following your problem |
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| 06:50.39 | niklaswe | how can i list aviable confbridge in asterisk console? |
| 06:51.04 | niklaswe | It looke like dont have the confbridge list command I using asterisk 1.8.18 |
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| 07:10.12 | din3sh | Gd mrning all |
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| 07:32.16 | apb1963_ | Hello anyway still awake? |
| 07:32.21 | apb1963_ | err... anyone? |
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| 07:58.33 | WIMPy | Good morning. |
| 07:59.34 | ChannelZ | Is it? |
| 08:00.16 | WIMPy | Not really. Woke up because my nose hurt. Seems I cought the flu. |
| 08:00.35 | ChannelZ | So it's not really good then either. |
| 08:00.57 | WIMPy | What went wrong for you? |
| 08:01.03 | ChannelZ | Sorry, anyway. Sick sucks. |
| 08:01.16 | ChannelZ | Well, it's monday :) |
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| 08:17.14 | *** join/#asterisk joshuahh (~joahahua@27.32.48.149) |
| 08:17.22 | joshuahh | hello just wondering if someone could please give some advise.. i have a vps which is running freepbx on a SIP Trunk.. all is well with regards to incoming calls.. weird issue i am seeing is this, i have a sip client on my iphone and on my laptop.. when i get my iphone (extension 3000) to call the laptop (extension 2000) the laptop rings, but when i click "answer" on the laptop.. it wont actually answer.. i have debug running asterisk -vvvvvvr |
| 08:17.36 | joshuahh | when ever i try to call extension 3000 from extension 2000 (laptop to iphone) nothing shows in the debug... and eventually i get "Call failed" |
| 08:17.46 | joshuahh | also, i try to ring voicemail (333) from the laptop, and it fails.. |
| 08:17.51 | joshuahh | my iphone works perfectly with incoming / outgoing |
| 08:19.33 | kaldemar | sounds like you have misconfigured the client on your laptop. enable sip debug and see if asterisk is getting any messages in from the laptop client. |
| 08:20.02 | joshuahh | kaldemar i did.. but nothing shows up |
| 08:20.10 | joshuahh | only when interactions are made from the iphone |
| 08:20.20 | joshuahh | it shows that it "Registers" |
| 08:20.26 | kaldemar | what shows? |
| 08:21.10 | joshuahh | when i close x-lite it shows : -- Unregistered SIP '2000' |
| 08:21.53 | kaldemar | have you configured it with a proxy that is not your asterisk box? |
| 08:21.55 | joshuahh | -- Registered SIP '2000' at 27.xx.xx.xxxx:5076 |
| 08:21.57 | joshuahh | when i open the app |
| 08:22.25 | joshuahh | domain proxy: "Domain" |
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| 08:37.53 | wdoekes | joshuahh: start tcpdump on your laptop to find out what goes on |
| 08:38.12 | wdoekes | tcpdump -vls0 -nniany port 5060 |
| 08:39.31 | wdoekes | (or windump if you're running windows, or alternative tools.. e.g. wireshark) |
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| 09:02.53 | ChannelZ | is the laptop running its own firewall? |
| 09:13.29 | din3sh | Got SIP response 400 "Bad Request" back from 192.168.6.35:5060 |
| 09:13.36 | din3sh | what does this mean |
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| 09:21.09 | din3sh | Got SIP response 400 "Bad Request" back from 192.168.6.35:5060, what does this mean? |
| 09:22.27 | EmleyMoor | I'm trying to get myself a native fax service. I have proven I can send faxes, using iaxmodem, but receiving is proving difficult. Have tried iaxmedom, t38modem (which won't even send) and using the ReceiveFax app... I am wondering if there is anything I can do, particularly with the latter, to make it work better. |
| 09:22.50 | EmleyMoor | (odd typo there) |
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| 09:26.15 | ChannelZ | din3sh: it means the other end got something it didn't like. |
| 09:27.42 | Addisk | anyone know how to handle 2 PRI sources that provide master timing? |
| 09:27.59 | Addisk | Do i get 2 VoIP Gateways? & link them to the pbx serveR? |
| 09:28.12 | Addisk | there such a device to help me? |
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| 09:44.59 | Rico29 | hi all |
| 09:45.27 | Rico29 | does anyone here have already experienced problems (sip phone screen freeze) with aastra phones ? |
| 09:54.50 | din3sh | c/ear |
| 10:01.23 | EmleyMoor | Rico29: Mine has never done that |
| 10:02.04 | Rico29 | what models of phones are you using ? |
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| 11:12.13 | eirirs | hi |
| 11:12.29 | eirirs | I see asterisk have come out with asterisknow with freepbx gui, how's it compared to the freepbx iso package? |
| 11:13.11 | eirirs | both seems well maintained |
| 11:17.54 | cusco | hi |
| 11:18.26 | cusco | can I trigger some piece of dialplan ONLY when a agent is CONNECTed to the queue'ed call ? |
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| 12:00.38 | WIMPy | Addisk: Use two single port cards. |
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| 12:12.24 | cusco | I can use a macro |
| 12:12.53 | cusco | Queue(Testing,t,,,180,,fromQueue); |
| 12:13.03 | cusco | how do I specify arguments to macro fromQueue ' |
| 12:13.04 | cusco | ? |
| 12:13.47 | WIMPy | Thu shallt not use macor any more. |
| 12:13.57 | cusco | ow.. ? |
| 12:16.11 | *** join/#asterisk bogrd__ (~sandeep@115.119.115.26) |
| 12:17.21 | bogrd__ | Hi, I'm trying to run some fastAGI php scripts on asterisk-11. But I get a broken pipe error. Where am i going wrong? any help is grateful... :) |
| 12:18.01 | kaldemar | bogrd__: your AGI probably does not read responses after commands. |
| 12:18.58 | bogrd__ | kaldemar: hmm.. let me show you the example i tried.. one sec.. |
| 12:20.34 | bogrd__ | kaldemar: this is the script I'm running ( http://paste.kde.org/669080/ ). |
| 12:21.10 | Greenlight | Which side are you getting the broken pipe on ? |
| 12:21.39 | Greenlight | Hmm... that's not fastAGI |
| 12:21.44 | Greenlight | That's just AGI, isn't it |
| 12:22.15 | kaldemar | bogrd__: i don't use or know phpagi. |
| 12:22.23 | Greenlight | You want to call that script with AGI, fastAGI is to call a script from a remote server |
| 12:23.27 | bogrd__ | Greenlight: ya.. I'm new to fastAGI. I used to run AGI before.. I want to change my system to use fastAGI.. I followed this tutorial... http://enricosimonetti.com/2009/04/27/asterisk-fastagi-with-php/ |
| 12:23.47 | bogrd__ | Greenlight: let me look for an example of fastAGI to test.. |
| 12:24.04 | Greenlight | How are you calling it from the dialplan? |
| 12:25.23 | bogrd__ | Greenlight: i'm using fetching it from the mysql db via odbc.. that will finally be AGI("agi://127.0.0.1:4573/sample.php"); |
| 12:26.00 | bogrd__ | bogrd__: the normal AGI is working fine but this fastAGI is giving me these errors! |
| 12:26.21 | Greenlight | You can't use fast agi like that |
| 12:28.00 | bogrd__ | Greenlight: should i use FastAGI(agi://something:port/file.php) ? you mean use FastAGI application instead of just AGI? |
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| 12:29.43 | Greenlight | That php file is designed to be called by AGI. FastAGI is a little different |
| 12:31.16 | bogrd__ | Greenlight: oh is it... when I use FastAGI() in dialplan i'm getting "No Such Application..." Should i select this seperately in menuselect? or.. |
| 12:31.52 | bogrd__ | Greenlight: can you point me to some link where I can learn how to write fastAGI scripts using phpagi library? |
| 12:32.25 | bogrd__ | Greenlight: I looked up the net, mostly I get AGI related articles and very less about fastAGI... :( |
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| 12:36.57 | Greenlight | Why do you need to use FastAGI ? |
| 12:37.15 | Greenlight | Rather than "normal" AGI |
| 12:38.49 | ghost75 | is this like fast cgi ? |
| 12:40.44 | bogrd__ | Greenlight: this is what I have heard and read: when there is high traffic it makes a difference.. normal AGI creates a seperate process for each AGI call.. its heavy on the system and fastAGI optimizes this without creating seperate processes but handles with threads.. |
| 12:41.01 | bogrd__ | Greenlight: yet, I do not have deep knowledge in it.. :) |
| 12:41.21 | ghost75 | same like fastcgi |
| 12:41.26 | bogrd__ | ghost75: yes.. |
| 12:41.34 | Greenlight | Yea, you're right, under heavy load FastAGI is the way to go. Was just checking you high enough load to warrent it |
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| 12:42.47 | Greenlight | FastAGI opens a socket connection. As far as I know you can't use a php script running on a webserver to connect into. When I used FastAGI for example I have a C# app that listens for connections on the FastAGI port |
| 12:43.24 | Greenlight | Then again, I've only used FastAGI as far as "testing" what works and how to use it |
| 12:43.46 | bogrd__ | Greenlight: Not much has been given about using FastAGI in the asterisk definitive guide as well.. yes you are right.. for that we have to configure xinetd for php i think.. not soo sure.. |
| 12:44.33 | Greenlight | Yea - it was only recently that I tried it out, and I found the documentation *very* lacking. To the point I wrote and app to listed and spit out to the console what it received, and worked it out that way. |
| 12:44.48 | *** join/#asterisk davlefouAMD (~david@197.15.73.137) |
| 12:45.04 | Greenlight | It's been a while since I used php, but I *think* there are socket functions in there, you may be able to run the app from php that way |
| 12:45.20 | Greenlight | As in, run the php file from say a shell, and it'll then open a socket and listen for connections. |
| 12:46.31 | bogrd__ | Greenlight: hmmm.. I got your point. Let me check out if I can do anything.. If I do, i'll write a nice blogpost on this so that it can help others trying to do the samething atleast.. :) |
| 12:46.50 | Greenlight | Good plan! |
| 12:46.59 | Greenlight | Even an example would give people a starting point. |
| 12:47.29 | Greenlight | Once you've got things connected, turns out the syntax and stuff from there on is same as normal AGI. It's just getting to that point that can leave you wondering/ |
| 12:48.11 | WIMPy | You can even take it another step and use AMI. Only one socket for all calls. |
| 12:49.02 | Greenlight | Yea, that's the one pitfall I found with FastAGI. A seperate socket connection for each call to FastAGI |
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| 12:50.02 | Greenlight | Like WIMPy said, AMI's a lot more flexable (and fun!) |
| 12:51.23 | Greenlight | On an unrelated note. Is there an overhead to using realtime extensions, and if so what sort of overhead? |
| 12:52.18 | WIMPy | db queries? |
| 12:53.08 | ^rage^ | using python+gevent for fastagi |
| 12:53.22 | Greenlight | At what point is the db hit? |
| 12:53.59 | Greenlight | As it, is it just when the extension registers ? |
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| 13:05.10 | *** join/#asterisk longst (~longst@46-22-127-214.bredband.alvsjo.qonet.se) |
| 13:08.00 | longst | any recommend for step by step asterisk PRI connection guide? |
| 13:08.57 | WIMPy | longst: Plug it in. |
| 13:12.16 | *** join/#asterisk beebeeep (~beebeeep@thanatos.migalin.net) |
| 13:12.55 | longst | some question about /etc/dahdi.conf. I have a TE405P digium card in my Asterisk 1.8.11 installation (CentOS 6.3 32-bit) first of all, this is a quad port PRI, when I boot machine up, only two lamps (port 1 and port 2 lamps) start blinking, the port 3 and port 4 lamps no blinking at all… I am wondering if this is expected behaviour? |
| 13:13.35 | Greenlight | It's been a while since I set them up, but I think the lights only come on for what you've got configured |
| 13:13.37 | WIMPy | If you have only two ports configured, yes. |
| 13:14.20 | Greenlight | I remember all 4 lights glowing on and off in a very Knight Rider sort of way, looks very cool in the data centre |
| 13:14.50 | WIMPy | If you don;t have the driver loaded. |
| 13:14.50 | kaldemar | and then it hits you that the interfaces are actually down. :P |
| 13:15.09 | kaldemar | green lights are nicer to look at. |
| 13:15.29 | Greenlight | :) |
| 13:19.36 | EmleyMoor | Another rule added to my dialplan - any caller ID beginning 001 and not having exactly 13 digits is void |
| 13:20.34 | Greenlight | You getting bogus caller id's ? |
| 13:20.47 | EmleyMoor | Plenty... |
| 13:21.56 | EmleyMoor | I mark any oddball ones as I get them, but general rules that catch loads at a stroke are good. |
| 13:22.06 | Greenlight | Heh. I seen an example somewhere that made me laugh. I was a dialplan that passed the caller id into an AGI script. Apart from if you spoofed your caller id as say "; touch rooted.txt" or something ... |
| 13:23.24 | EmleyMoor | I get plenty of "doubled 0" attempts - that will trap those for the majority of British geographical numbers... |
| 13:23.58 | Greenlight | Remember for the UK there are valid 10 digit numbers too though |
| 13:27.33 | EmleyMoor | Greenlight: Yes... 11 and 12 digits are potentially valid where the first two digits are 01 |
| 13:28.14 | EmleyMoor | Er, 10 and 11 |
| 13:28.41 | EmleyMoor | Fortunately I get very few that are clearly wrong in the UK |
| 13:34.27 | longst | I try to run a "# dahdi_genconf " then it automatic generate >>>>>span=1,1,0,ccs,hdb3,crc4 |
| 13:34.28 | longst | # termtype: te |
| 13:34.29 | longst | bchan=1-15,17-31 |
| 13:34.31 | longst | dchan=16 |
| 13:34.32 | longst | echocanceller=mg2,1-15,17-31 |
| 13:34.34 | longst | # Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" HDB3/CCS/CRC4 |
| 13:34.35 | longst | span=2,2,0,ccs,hdb3,crc4 |
| 13:34.37 | longst | # termtype: te |
| 13:34.38 | longst | bchan=32-46,48-62 |
| 13:34.40 | longst | dchan=47 |
| 13:34.41 | longst | echocanceller=mg2,32-46,48-62 |
| 13:34.43 | longst | # Span 3: TE4/0/3 "T4XXP (PCI) Card 0 Span 3" HDB3/CCS/CRC4 RED |
| 13:34.44 | longst | span=3,3,0,ccs,hdb3,crc4 |
| 13:34.46 | longst | # termtype: te |
| 13:34.47 | longst | bchan=63-77,79-93 |
| 13:34.49 | longst | dchan=78 |
| 13:34.50 | longst | echocanceller=mg2,63-77,79-93 |
| 13:34.52 | longst | # Span 4: TE4/0/4 "T4XXP (PCI) Card 0 Span 4" HDB3/CCS/CRC4 RED |
| 13:34.53 | longst | span=4,4,0,ccs,hdb3,crc4 |
| 13:34.55 | longst | # termtype: te |
| 13:34.56 | longst | bchan=94-108,110-124 |
| 13:34.58 | longst | dchan=109 |
| 13:34.59 | longst | echocanceller=mg2,94-108,110-124 |
| 13:35.01 | longst | # Global data |
| 13:35.02 | WIMPy | ~pb |
| 13:35.02 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 13:35.02 | longst | loadzone= us |
| 13:35.02 | longst | defaultzone= us <<<<< |
| 13:35.40 | longst | I am wondering if it means that four ports are property configured ? |
| 13:36.43 | [TK]D-Fender | longst, That is not saying Asterisk is using them. That's only half of the configs. |
| 13:37.10 | [TK]D-Fender | longst, And PASTEBIN from now on. Do not flood in here |
| 13:38.20 | longst | configuration looks like http://pastebin.com/BkGY9FWB |
| 13:39.42 | longst | and the card lamps behaves like this : http://www.youtube.com/watch?v=JGtWVfjs4pg |
| 13:39.51 | [TK]D-Fender | longst, that is still only HALF of the config |
| 13:39.58 | [TK]D-Fender | longst, CHAN_DAHDI.CONF |
| 13:40.00 | [TK]D-Fender | ^ |
| 13:41.33 | longst | This is my first time working with TE405P card. I borrowed this card from someone. I am wondering if this type of lamps pattern means there are only two ports functional, and the rest of two ports don't work ?? |
| 13:43.20 | [TK]D-Fender | longst, who said they don't work? You haven't shown us the configuration yet |
| 13:43.59 | *** join/#asterisk slayer192 (~chrisc@78.90.85.118) |
| 13:44.11 | WIMPy | He hasn't told us what he wants it to do, either. |
| 13:44.52 | longst | I didn't change chan_dahdi.conf to be honest….I left it default... |
| 13:45.01 | longst | I put it on past board |
| 13:45.03 | [TK]D-Fender | longst, There is no such thing as "default" |
| 13:45.05 | *** join/#asterisk thecoda (~kevin@027c5e55.bb.sky.com) |
| 13:45.11 | [TK]D-Fender | longst, this is your job to configure |
| 13:45.26 | [TK]D-Fender | longst, Do that actual job before wondering if it works. |
| 13:47.02 | longst | OK thanks. I was thinking only require /etc/dahdi/system.conf configuration to make it work… |
| 13:48.05 | [TK]D-Fender | longst, That only says " I have these ports". It doesn't say * is configured to use them. |
| 13:48.37 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 13:49.59 | longst | Good, So if I understood correctly /etc/chan_dahdi.conf defined "how these ports configured to be used" |
| 13:50.57 | WIMPy | The dahdi config is only the physical layer. |
| 13:51.33 | longst | I check out /etc/chan_dahdi.conf it looks http://pastebin.com/BkGY9FWB |
| 13:51.39 | longst | I think it looks good. |
| 13:51.54 | WIMPy | To do WHAT? |
| 13:53.29 | longst | first of all. I think with these configuration, I would be able to see four lamps blinking, after I start machine. what do you think ? |
| 13:55.09 | [TK]D-Fender | <longst> Good, So if I understood correctly /etc/chan_dahdi.conf defined "how these ports configured to be used" <- How != If |
| 13:55.13 | *** join/#asterisk serafie (~erin@nat/digium/x-rkzzunkvkcjgmdmq) |
| 13:56.00 | [TK]D-Fender | <longst> I check out /etc/chan_dahdi.conf it looks http://pastebin.com/BkGY9FWB <--- this is NOT chan_dahdi.conf |
| 13:57.59 | longst | Oh sorry should be this one ...http://pastebin.com/ZWnMgJsQ, this one… |
| 13:59.16 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ybytlclxgqufigym) |
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| 14:00.47 | WIMPy | What ist that? Have you actually taken a look at that file yourself? |
| 14:01.33 | WIMPy | It is almost empty. |
| 14:01.37 | [TK]D-Fender | longst, I don't see anything properly configured in there at all. That is hundred of lines of trash. |
| 14:01.40 | WIMPy | It certainly doesn;t contain a single channel. |
| 14:01.57 | [TK]D-Fender | longst, Go configure your channels. You haven't done your job. |
| 14:02.31 | longst | OK.. thank for your information.. |
| 14:03.50 | longst | I am working with this files. |
| 14:04.33 | longst | by the way, if someone know what "NFAS" means in the context of "Trunk groups are used for NFAS connections." |
| 14:05.15 | [TK]D-Fender | Not ^#%ing Applicable Service..... |
| 14:05.15 | [TK]D-Fender | :) |
| 14:05.20 | [TK]D-Fender | Ignore it. |
| 14:06.45 | WIMPy | I don;t think NFAS exists with E1. |
| 14:07.11 | longst | Oh… in this case would be difficult to understand what [trunkgroups] does... |
| 14:07.48 | [TK]D-Fender | longst, Also unimportant |
| 14:08.40 | WIMPy | It's only used for NFAS, so ignore it. |
| 14:08.44 | Greenlight | Queues can be configured realtime, as well as SIP peers, yes? |
| 14:09.22 | [TK]D-Fender | Greenlight, Yes |
| 14:09.27 | longst | Greenlight, yes |
| 14:09.58 | Greenlight | Excellent. Do know where I might find the current table structure for queues, like the wiki has for sip friends? |
| 14:10.17 | longst | are there any relation between, "dahdi-channels.conf" and "chan_dahdi.conf "? |
| 14:10.21 | WIMPy | In the contrib* directory. |
| 14:10.34 | Greenlight | Ok, thanks |
| 14:13.45 | Greenlight | Hmm in realtime/mysql I see queue_log.sql but no queue.sql annoyingly. Is it hidden away elsewhere? |
| 14:14.26 | [TK]D-Fender | longst, The first is a sample file auto-generated by a system script you could use. |
| 14:14.41 | [TK]D-Fender | longst, And is meant to be "#INCLUDE" 'd in the main. |
| 14:14.55 | [TK]D-Fender | longst, Or.... you could just forget about it and do it right yourself. |
| 14:16.52 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 14:18.10 | leifmadsen | at some point someone really needs to go and build out the sql files for realtime |
| 14:18.25 | leifmadsen | at this point they are basicalyl non-existant, and best way is really looking at the code to figure out the fields required |
| 14:18.27 | leifmadsen | not ideal. |
| 14:18.40 | leifmadsen | there is usually some older or out of date examples flying around the web |
| 14:18.54 | Greenlight | Ahh so I'm not being dumb, they're just not there :) |
| 14:19.04 | WIMPy | Didn't I see them in the contib dir? |
| 14:19.06 | leifmadsen | well, all I can speak towards is them not being there. |
| 14:19.20 | Greenlight | I'm always reluctant to pick up stuff from random locations on web, as sometimes it can be soooo out of date |
| 14:19.23 | leifmadsen | WIMPy: depending on version, they are likely out of date |
| 14:20.02 | Greenlight | When I see the "sqlserver" directory I thought I was onto a winner... alas it was empty :) |
| 14:20.30 | Greenlight | So - not many people use realtime, or ? |
| 14:20.44 | leifmadsen | lots of people use realtime |
| 14:20.57 | leifmadsen | not many people contribute back the sql files after they figure it out it appears |
| 14:21.04 | Greenlight | Gotcha |
| 14:21.06 | leifmadsen | is one of those slackers |
| 14:21.10 | Greenlight | heh |
| 14:21.14 | WIMPy | wonders why it's split up by DBMS. Shouldn;t it be the same SQL? |
| 14:21.28 | Greenlight | The syntax differs a little sometimes |
| 14:21.57 | leifmadsen | ya, different DBs can use different field types etc, or require slight syntax differences |
| 14:22.12 | file | a standard is never a standard. |
| 14:22.14 | WIMPy | likes standards |
| 14:22.24 | leifmadsen | the great thing about standards is there are so many to choose from! |
| 14:22.30 | leifmadsen | paraphrases jsmith like a boss |
| 14:22.33 | Greenlight | Realtime on the surface seems a really neat way to configure things, especially if you want to enable customers to make basic changes etc |
| 14:22.47 | leifmadsen | Greenlight: if you want to allow customers to make changes through an interface |
| 14:22.54 | leifmadsen | allowing them to modify your db is probably a really bad idea |
| 14:23.16 | file | and making your PBX rely so heavily on a DB... is also probably a really bad idea |
| 14:23.19 | leifmadsen | realtime works great; been using it for years |
| 14:23.31 | leifmadsen | has wanted to try out realtime via curl though |
| 14:24.03 | leifmadsen | file: relying on the DB isn't so much a problem if your DB infrastructure is distributed and reliable :) (a whole other problem to solve) |
| 14:24.24 | leifmadsen | moving towards realtime in production does have a learning curve and a lot of extra overhead to learn to contend with |
| 14:24.29 | file | sure but you are also relying on that database being fast and responsive |
| 14:24.34 | leifmadsen | agreed |
| 14:24.56 | leifmadsen | we use read only replicated DBs to the local machine for htat |
| 14:25.04 | Greenlight | Hmmm lots of decisions |
| 14:25.15 | leifmadsen | realtime is a neat idea; it adds a LOT of work |
| 14:25.41 | Greenlight | Which translaters to a LOT of time, which is kinda limited at the moment |
| 14:25.54 | Greenlight | Maybe I'll stick with old school configs for the moment |
| 14:25.59 | leifmadsen | yes, I would suggest that |
| 14:26.46 | longst | I am really a beginner of PRI ISDN board, I am wondering if there is a "quick start" configuration I could use |
| 14:27.05 | Greenlight | I've installed FreePBX at some sites to allow customers to config things them selves, like extra extensions and queues. Problem is that on busy boxes (200 sim calls) reloading the config and asterisk dies 50% of the time |
| 14:27.22 | Greenlight | So I'm seeking a solution that doesn't need to reload the config in Asterisk |
| 14:29.35 | Katty | HI LADS |
| 14:29.45 | WIMPy | wonders if that's really an Asterisk issue. |
| 14:30.02 | Katty | i'll be your asterisk issue in a minute. |
| 14:30.21 | WIMPy | Katty: Is it on fire, yet? |
| 14:30.30 | Greenlight | It's like Asterisk deadlocks, so I'd imagine it *is* an Asterisk issue |
| 14:30.33 | Katty | well i don't smell anything. |
| 14:30.35 | Katty | so that's a good sign. |
| 14:30.46 | Greenlight | How much of it is caused by the overly complex FreePBX crap is another issue |
| 14:31.09 | WIMPy | Quite a but, I guess. |
| 14:31.28 | WIMPy | But it sounds like a good chance to debug what's going on. |
| 14:31.32 | Greenlight | For example, when it "dies". "core show channels", takes about 5 minutes to complete, listing a channel every few seconds |
| 14:31.59 | Katty | i tried freepbx once. |
| 14:32.02 | Greenlight | I'd love to debug it and work it out, but enabling any sort of debugging and the server would just laugh at me |
| 14:32.11 | Katty | i made changes to config files, and couldn't figure out why it wouldn't work. |
| 14:32.16 | Katty | then found out, it wasn't even reading those config files. |
| 14:32.45 | leifmadsen | WIMPy: that has been an issue for a while; when you perform a dialplan reload it can lock the creation of new channels until the dialplan is loaded into memory |
| 14:32.56 | leifmadsen | thats the issue with systems like freepbx that just create duplicate information over and over |
| 14:33.12 | leifmadsen | Greenlight: make sure the console verbosity is turned to zero during reload; it seems to help |
| 14:33.19 | Greenlight | The funny bit is how random it is |
| 14:33.36 | leifmadsen | it's unlikely to be random so much as scripted reloads |
| 14:33.38 | longst | I tried to install two "FreePBX" boxes follow the instructions from "http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html" connected them between PRIs, It worked. And now if I refer the FreePBX "chan_dahdi.conf" it looks like "http://pastebin.com/kfWWL6dm" and it seems there is no specific configuration in FreePBX "chan_dahdi.conf". And now I tried to add "[global] |
| 14:33.39 | longst | #include dahdi-channels.conf" into chan_dahdi.conf. and do a "static-host*CLI> dahdi restart " still seems no progress... |
| 14:33.41 | Greenlight | Like, if it works it reloads in <15 seconds and all is good. Otherwise, I have to kill asterisk after 10 mins |
| 14:34.30 | Greenlight | Guess it's just locks queued after locks and all goes a bit mental |
| 14:34.43 | Greenlight | But yea, reloading a config can be a scarey moment on those systems |
| 14:35.21 | Greenlight | That's why I though realtime sip peers and queues, plus a barebones dialplan would be a nice elegant solution |
| 14:36.00 | Greenlight | Think I'll keep it on the back burner for the moment though till I've the time to play around mor |
| 14:36.15 | Katty | so i heard the pope was resigning |
| 14:36.36 | Greenlight | Yup - clearly related to the horse meat scandal |
| 14:36.59 | Greenlight | isn't a conspiracy thoerist |
| 14:37.07 | Katty | i've not heard anything about the why. |
| 14:38.01 | [TK]D-Fender | Greenlight, You are now... |
| 14:38.25 | EmleyMoor | Katty: Healtd reasons it seems - though he officially cites his age |
| 14:38.32 | EmleyMoor | Health* |
| 14:38.54 | *** join/#asterisk _Corey_ (~chatzilla@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 14:39.02 | WIMPy | I thought Age was a prerequisite? |
| 14:39.11 | [TK]D-Fender | Greenlight, The Vatican has been smuggling in unicorns for centuries as "illegals" and to cover up the whole thing in addition to forcibly cutting off their horns decided to cull the herd a bit. It only LOOKS like horse..... |
| 14:39.21 | WIMPy | Min: Mentally dead, Max: Physically dead. |
| 14:39.31 | *** join/#asterisk Minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
| 14:39.37 | Greenlight | See, I knew it was do with the horse thing :) |
| 14:40.15 | Katty | and here i was hoping it had something to do with the german dungeon porn stock scandle |
| 14:40.20 | EmleyMoor | WIMPy: He is 85 - older than JPII was when he died |
| 14:40.36 | *** join/#asterisk Devon_ (~Devon_@63.214.236.169) |
| 14:40.43 | Greenlight | He only took over a few years back, not like he was going to get any younger |
| 14:40.47 | WIMPy | Katty: Please elaborate |
| 14:41.11 | Greenlight | Age/health doesn't make sense. It's horses/unicorns, or Katty's theory! |
| 14:41.41 | Katty | well...the catholic church owns a german media company called Weltbild |
| 14:41.58 | Katty | which is like amazon |
| 14:42.03 | EmleyMoor | In some ways I hope they choose Nichols next... |
| 14:42.12 | WIMPy | We all know the church is in to the porn business. |
| 14:42.35 | WIMPy | But what's that dungeon story? |
| 14:42.46 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:42.51 | Katty | well they also own half stock in a publishing company named Droemer Knaur |
| 14:42.59 | Katty | which soley produces pornography |
| 14:43.23 | Katty | some of which is the edgy bits. |
| 14:43.38 | Katty | regardless, both publishing company sell porn |
| 14:43.50 | *** join/#asterisk izx (~karthick@shellium/member/karthick87) |
| 14:44.08 | Katty | weltbild being 100% catholic church, and dromer knaur being 50% catholic church |
| 14:45.31 | Katty | there's probably a lot more dirt if you dig. |
| 14:45.38 | Katty | but it was on reddit a few months back |
| 14:46.30 | WIMPy | Like writing HOWTOs for people caught with child abuse? |
| 14:46.35 | WIMPy | Sure |
| 14:46.54 | Katty | that's not limited to the catholic church. |
| 14:47.14 | Katty | my folks were Jehovah Witnesses when i was growing up |
| 14:47.28 | Katty | and i know for a fact that stuff like that happened in their church too....and it was covered up as well. |
| 14:47.31 | WIMPy | No, but they will hide you. |
| 14:48.26 | WIMPy | But if it comes out the catholics can always seek the protective shelter of the vatican. |
| 14:48.39 | Katty | that's pretty lamesauce. |
| 14:49.04 | WIMPy | It's the official backup plan. |
| 14:50.30 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
| 14:51.19 | *** join/#asterisk ariel_ (uid3533@pdpc/supporter/active/abatista) |
| 14:51.28 | ariel_ | hello folks |
| 14:51.51 | Katty | hugs ariel_ |
| 14:52.11 | ariel_ | looking for someone that has connected asterisk, polycom phone to do lookups via ldap off active directory. |
| 14:52.19 | ariel_ | Katty: hi, long time. Hope your doing well. |
| 14:52.25 | ariel_ | hugs Katty |
| 14:52.47 | Katty | ariel_: yesh :> am goodly. how're you dear? |
| 15:00.54 | ariel_ | Fine just trying new setups. Hope winter is not too bad for you this year |
| 15:01.03 | thecoda | What debugging do I need to turn on to see *exactly* what's going on within DAHDI? |
| 15:01.54 | thecoda | polarity reversal and the like… I'm trying to identify why I can't detect an answer over POTS |
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| 15:35.13 | *** join/#asterisk slicknick5181 (~ubuntu@204.195.131.94) |
| 15:35.39 | slicknick5181 | I need some help with in-call DTMF features such as Capp Park |
| 15:35.45 | slicknick5181 | Call Park* |
| 15:36.37 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:42.57 | eirirs | I just started googling for Capp Park |
| 15:43.02 | Faustov | ;) |
| 15:43.27 | Qwell | eirirs: I saw them play live last month. They were great! |
| 15:44.01 | eirirs | ! |
| 15:44.54 | longst | a general question about ISDN PRI. In this DAHDI configuration, http://www.facebook.com/photo.php?fbid=10151387218837906&set=a.10151387218792906.496748.736667905&type=3&theater It is said "T4XXP (PCI) Card 0 Span 1", in this case what meaning for "Span" ? Thank |
| 15:45.17 | Greenlight | The first "port" |
| 15:45.39 | Greenlight | Or "connection" |
| 15:46.13 | Greenlight | Each of your E1's would be a Span |
| 15:47.33 | longst | but any story behind "Span" Is "Span" an abbreviation of some word ? |
| 15:47.50 | Greenlight | I guess cause it spans multiple channels |
| 15:47.59 | Greenlight | Think it's a telephony/isdn term |
| 15:48.06 | WIMPy | no |
| 15:48.45 | *** join/#asterisk imox (~imox@91-66-32-57-dynip.superkabel.de) |
| 15:50.16 | thecoda | Anyone? How can I see exactly what signalling is occurring on my DAHDI card? |
| 15:50.38 | longst | OK, basically, I can say each T1 /E1 port is a "Span", each "Span" should have multiple channels correct? |
| 15:50.50 | Greenlight | longst: Indeed |
| 15:50.51 | thecoda | What DTMF it receives, polarity reversals, etc |
| 15:51.02 | Greenlight | thecoda: Enable pri debug |
| 15:51.05 | WIMPy | thecoda: What exactely do you want to find out? |
| 15:51.12 | WIMPy | longst: Yes. |
| 15:51.16 | thecoda | Why it doesn't detect a remote answer |
| 15:51.27 | Greenlight | thecoda: iirc "pri debug on span 1" |
| 15:52.00 | Greenlight | Oh, wait that's PRI won't show for analogue will it |
| 15:52.14 | WIMPy | No. Analog is evil. |
| 15:52.25 | eirirs | analog is sexy |
| 15:52.35 | Greenlight | Evil with a capital "E" |
| 15:52.45 | thecoda | Absolutely evil |
| 15:52.49 | Greenlight | I hear it's what made the pope quit |
| 15:52.51 | WIMPy | An d a capital VIL |
| 15:53.15 | Qwell | and then the E moved to the end. |
| 15:53.23 | thecoda | I'd move entirely to sip, but still need to deal with the pre-existing number |
| 15:53.33 | Greenlight | Port the number ^^ |
| 15:53.50 | Qwell | print new business cards |
| 15:53.58 | WIMPy | Or get a line that's less than 2 decates old. |
| 15:54.01 | thecoda | It's a home installation |
| 15:54.16 | Qwell | then just post the new number on facebook |
| 15:54.36 | Qwell | "Getting into the 21st century. My new number is ..." |
| 15:54.38 | thecoda | It's also a fallback in case of IT failure :) |
| 15:54.56 | Qwell | I'm sure the IT dept can update your beeper #. |
| 15:55.09 | thecoda | and it gives me free local calls |
| 15:55.15 | Qwell | wow, I'm being cynical this morning. |
| 15:55.23 | leifmadsen | Qwell: this morning? |
| 15:55.26 | _Corey_ | 1. Plug in analog phone. 2. Dial *72 and forward the number to your new SIP service. 3. Play a "I have a new number now" recording on Asterisk. |
| 15:55.30 | Qwell | leifmadsen: Your face. |
| 15:55.35 | leifmadsen | is awesome!@ |
| 15:55.44 | thecoda | I'd say you're being very cynical, given that it's 15:55 |
| 15:55.54 | leifmadsen | 10:55 in the centre of the universe |
| 15:55.56 | Greenlight | He's accross the pond |
| 15:56.03 | Qwell | oh, it's 3pm? I can definitely be more cynical then. |
| 15:56.08 | thecoda | Greenwich is the centre of the universe |
| 15:56.14 | thecoda | hence GMT |
| 15:56.26 | thecoda | Where it's 15:55 |
| 15:56.27 | leifmadsen | you haven't been paying attention to Toronto then |
| 15:57.14 | thecoda | Nope, just to the clock, which now says 15:57 :) |
| 16:01.06 | slicknick5181 | I need some help with in-call DTMF features such as Call Park |
| 16:01.39 | *** join/#asterisk gauravp (~gaurav@c-68-80-206-60.hsd1.pa.comcast.net) |
| 16:02.08 | slicknick5181 | I have a feature set called home-feat and I have call park at *27 but I can not get the phones to do anythinf when I dial this in call |
| 16:02.17 | longst | According to http://www.facebook.com/photo.php?fbid=10151387245397906&set=a.10151387218792906.496748.736667905&type=3&theater It is a screen shot of one "Span" of a TE405P card from DAHDI tools. I am wondering if there are some document explain what different configurations means, for example, Current Alarms, Sync Source IRQ Misses, etc…. |
| 16:02.22 | *** join/#asterisk Sidrov (~Sidrov@173.192.139.246-static.reverse.softlayer.com) |
| 16:02.26 | Sidrov | hello all |
| 16:02.41 | slicknick5181 | Hello |
| 16:02.48 | [TK]D-Fender | slicknick5181, And you've enabled the feature in your dialplan prior to trying to use it? And you've set a Dial() option to allow checking for it? |
| 16:03.36 | Sidrov | anyone knows why asterisk 1.8 is working slower / delaying hangup command with multiple concurent calls on same IVR menu ? |
| 16:03.46 | Sidrov | with 2-3 concurent calls is working fine |
| 16:04.03 | slicknick5181 | [TK]D-Fender, well thats where I did get a little lost I wasn't sure how to enable it in the dialplan but I did put tT in my Dial string |
| 16:04.11 | Sidrov | when number increases, hangup command is executed with an increasing delay |
| 16:04.47 | [TK]D-Fender | slicknick5181, read the sample features.conf for enabling applicationmap, etc |
| 16:04.51 | Greenlight | Sidrov: How are you executing this hangup command: |
| 16:05.09 | [TK]D-Fender | slicknick5181, Though normally You should simply be TRANSFERRING calls to the parking lot, not trying to use it as a "feature" |
| 16:05.27 | Sidrov | goto(hangup) then (hangup) Hangup() |
| 16:05.39 | Greenlight | slicknick5181: Also, if you are wanting to "listen" for DTMF, ensure you pass the appropriate argument to Dial |
| 16:05.59 | Sidrov | cli shows it executed, but call still active for a while then hang |
| 16:06.12 | slicknick5181 | [TK]D-Fender I did but I just could not understand exactly what it was asking me to do |
| 16:06.28 | Greenlight | Sidrov: Guess you could grab a SIP trace,and check what side the delay is at |
| 16:06.55 | gauravp | Hi All, I am in the process of migrating my configuration from Asterisk 8 to 11. My previously working dialplan relied on $CALLERID(dnid) to provide the google voice account associated with an incoming call to route to the appropriate extension. Now however, I see $CALLERID(dnid) evaluating to 0. Couldn't find evidence of that variable being deprecated, but should I be using something else? |
| 16:07.01 | [TK]D-Fender | slicknick5181, Just transfer them like normal to the parking ext |
| 16:07.04 | slicknick5181 | [TK]D-Fender, thats the other thing nothing shows on the CLI when a button is pressed |
| 16:07.26 | slicknick5181 | I'm using an SIP ATA |
| 16:08.18 | Sidrov | Greenlight, delay is from my side, i'm using pocketsphinx to recognise clients commands; as number increases, somehow asterisk gets delayed in commands executing; it just execute them delayed. any setting in asterisk to free up memory ? |
| 16:08.40 | *** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net) |
| 16:08.48 | [TK]D-Fender | slicknick5181, Things don't just "show up". Transfer your call to the parking ext. |
| 16:09.40 | slicknick5181 | [TK]D-Fender, the only way to I have to transfer is using in-call DTMF as I am using a SIP ATA |
| 16:09.43 | AkkerKid | heya guys! If I have asterisk do a query to MSSQL, How do I retrieve more than one column per result? |
| 16:10.31 | Greenlight | Sidrov: No idea what pocketsphinx is. |
| 16:10.48 | *** join/#asterisk lorsungcu (~anonymous@50-78-230-69-static.hfc.comcastbusiness.net) |
| 16:10.57 | Sidrov | Greenlight, it's an opensource voice recognition engine. |
| 16:11.30 | Sidrov | How many concurent calls can asterisk handle with normal configuration 2.2Gb cpu, 1 Gb ram ? |
| 16:11.43 | WIMPy | None |
| 16:11.51 | WIMPy | There is no "normal configuration". |
| 16:11.59 | thecoda | Right, time to enable debug logging |
| 16:12.07 | Sidrov | and how it's behaviour when concurent calls are too many ? |
| 16:12.24 | Greenlight | How many calls are you talking here? |
| 16:12.37 | Greenlight | It sounds like the problem is more related to this opensource application, than to Asterisk |
| 16:12.52 | [TK]D-Fender | <slicknick5181> [TK]D-Fender, the only way to I have to transfer is using in-call DTMF as I am using a SIP ATA <- what model? |
| 16:13.55 | Greenlight | Normally on those, can't you pulse the hangup to do a transfer ? |
| 16:14.51 | slicknick5181 | [TK]D-Fender, I have a USP Connect ATA-172, Just a cheap thing I bout a few Months ago |
| 16:15.07 | Sidrov | Greenlight, 3-10 concurent calls makes the issue; opensource applicaion is not integrated, it's called via agi script, wich provide response and die; BUT, agi executing the command is taking memory from asterisk, as it's a child of asterisk main proccess. maybe this is problem. |
| 16:15.07 | slicknick5181 | Greenlight, Was that directed at me? |
| 16:15.53 | Greenlight | slicknick5181: http://www.welltech.com/product_e_0d.htm That one ? |
| 16:16.06 | [TK]D-Fender | slicknick5181, Check its manual. Most allow you to use the hook-flash for transfers, etc. Either way... just transfer to the parking ext |
| 16:16.38 | slicknick5181 | Greenlight, Yes |
| 16:17.32 | leifmadsen | AkkerKid: you use func_odbc and the multirow method |
| 16:17.36 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 16:17.36 | Greenlight | slicknick5181: Looking at that page, it sounds like it should support it |
| 16:17.37 | thecoda | http://pastebin.com/cprZSy9i |
| 16:17.40 | *** join/#asterisk fredericve (~fes@host-212-68-194-46.brutele.be) |
| 16:17.59 | Sidrov | Greenlight, guess problem is my agi script is loading aprox 10-20Mb memory, and then asterisk is delaying memory relasing |
| 16:18.01 | thecoda | That's a log of me calling my mobile. Why is it not seeing the answer? |
| 16:18.09 | slicknick5181 | [TK]D-Fender, The only thing I find in regards to flash on it's setup screen is flash time. I will check manual |
| 16:18.37 | leifmadsen | AkkerKid: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/getting_funky.html <-- then search for "multirow" |
| 16:18.43 | thecoda | It even detects the DTMF I send back after answering, but my SIP phone still thinks it's ringing |
| 16:18.50 | Greenlight | slicknick5181: Try just pressing hangup quickly, and see if it gives you 2nd line |
| 16:19.14 | *** join/#asterisk Tarso (~Tarso@189.61.55.7) |
| 16:19.36 | fredericve | Hi, I have a setup with 2 SIP peers, both support T38. I want to use T.38 passthrough mode. Is there any way I can force asterisk to send reinvites for T38? I am using asterisk 1.8.13.0. |
| 16:19.44 | slicknick5181 | Greenlight, Pressing talk/flash gives me a dialtone but disconnects the call |
| 16:20.42 | gauravp | My previously working dialplan relied on $CALLERID(dnid) to provide the google voice account associated with an incoming call to route to the appropriate extension. On Asterisk 11 however, I see $CALLERID(dnid) evaluating to 0. |
| 16:21.00 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 16:21.16 | gauravp | ExecIf($["${CALLERID(dnid)}" = "foo@gmail.com"]?Dial(${foo})) |
| 16:22.10 | gauravp | On 11: ExecIf("Motif/+1xxxxxxxxxx-c3e1", "0?Dial(SIP/foo)") in new stack |
| 16:23.18 | *** part/#asterisk sipman (~slane@71-14-128-129.dhcp.ftwo.tx.charter.com) |
| 16:23.31 | leifmadsen | gauravp: then you should look at the value of CALLERID(dnid) and see what it is returning, and why it isn't matching |
| 16:23.33 | slicknick5181 | [TK]D-Fender, Only thing flash related in the flash time, is there a setting that tells asterisk when I press flash to process an xfer |
| 16:23.34 | gauravp | is there an easy way to dump all CALLERID variables for a call so I can see if there's something more appropriate that I can pick up to differentiate between incoming calls from the various xmpp connections? |
| 16:23.36 | leifmadsen | the values are likely not the same as before |
| 16:23.49 | leifmadsen | gauravp: NoOp(${CALLERID(dnid)}) |
| 16:23.57 | leifmadsen | or use DumpChan() |
| 16:25.16 | gauravp | liefmadsen: I actually had NoOp(${CALLERID(dnid)} at the top of my dialplan and it returned NoOp("Motif/+1xxxxxxxxxx-c3e1", "") |
| 16:25.23 | gauravp | will try DumpChan() |
| 16:25.46 | AkkerKid | thanks! @leifmadsen |
| 16:26.42 | leifmadsen | np |
| 16:26.53 | leifmadsen | gauravp: then that is the value that is returned via the function |
| 16:27.04 | leifmadsen | so it doesn't match foo@gmail.con, hence why it returns false |
| 16:29.35 | *** join/#asterisk KB1ZIB (~moos3@cpe-72-224-215-87.maine.res.rr.com) |
| 16:30.08 | rdm | yawns |
| 16:30.27 | slicknick5181 | Greenlight, Do you have any ideas how to make this work? |
| 16:33.39 | gauravp | liefmadsen: nothing useful from DumpChan ... chan_gtalk returned the string 'foo@gmail.com' in CALLERID(dnid), there doesn't seem to be a similar variable with chan_motif |
| 16:33.46 | thecoda | Just did: cat /proc/dahdi/* |
| 16:33.49 | leifmadsen | btw, it's leif :) |
| 16:33.58 | thecoda | and I get : 1 WCTDM/4/0 FXSKS (In use) (EC: MG2 - INACTIVE) |
| 16:34.02 | slicknick5181 | How do I set my CLI to show DTMF |
| 16:34.07 | gauravp | leif: sorry :) |
| 16:34.12 | longst | a question regarding configuration DAHDI device, according to "http://www.cadvision.com/blanchas/Asterisk/DahdiDrivers.html" in the very bottom, "Digium Wildcard TE110P T1/E1 Card 0" configuration screen shot, if some one happen to know what Loop button does? |
| 16:34.12 | thecoda | It *should* be (in use), yes? |
| 16:34.25 | leifmadsen | slicknick5181: modify logger.conf to contain 'dtmf' on the console and then 'logger reload' on the console |
| 16:34.47 | slicknick5181 | Leifmadsen, Thank you very much sir |
| 16:35.11 | thecoda | … I'm not presently in a call |
| 16:35.23 | gauravp | leif: are there any additional variables/parameters that may be available on chan_motif that are not displayed by DumpChan()? |
| 16:35.48 | gauravp | I was previously using dnid to route calls from 3 different google voice accounts to different extensions |
| 16:36.57 | thecoda | anyone? please? |
| 16:37.09 | leifmadsen | gauravp: I'd look at any related functions for motif, and also look at the other available CALLERID() methods like all, name, num, etc |
| 16:38.56 | gauravp | leif: there's CallerIDNum and CallerIDName, but both have the number calling in, not the google voice account proxying in the call |
| 16:39.42 | gauravp | i'll try to NoOp(${CALLERID(all)}) and see if anything is there |
| 16:40.07 | leifmadsen | gauravp: CallerIDNum and CallerIDName are not valid anythings |
| 16:40.16 | leifmadsen | all callerid information is done via the CALLERID() function |
| 16:40.37 | leifmadsen | 'core show function CALLERID' will show you more things you can look at |
| 16:40.49 | *** join/#asterisk blizzow1 (~jburns@75-171-154-172.hlrn.qwest.net) |
| 16:42.21 | slicknick5181 | I enabled DTMF logging and I can see the buttons being pressed but still no xfer to call park or any other feature |
| 16:43.26 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 16:45.51 | *** join/#asterisk din3sh (~din3sh@41.136.83.224) |
| 16:46.55 | gauravp | leif: taking a look, i'll try calling a couple of the arguments to CALLERID() to see if I can find something useful |
| 16:47.25 | gauravp | just thought that DumpChans() would show everything available to CALLERID() |
| 16:48.09 | *** join/#asterisk vlad_starkov (~vlad_star@194.186.53.88) |
| 16:48.55 | leifmadsen | gauravp: no, just the channels vars that are set |
| 16:48.59 | leifmadsen | not functions |
| 16:51.31 | *** join/#asterisk chris_n (~Chris@184.7.21.42) |
| 16:58.21 | thecoda | has run out of new things to try :( |
| 16:58.54 | Katty | time to start trying beers. |
| 16:59.12 | *** join/#asterisk Charlie__ (c15fc7cb@gateway/web/freenode/ip.193.95.199.203) |
| 16:59.17 | Charlie__ | hi |
| 16:59.36 | thecoda | Beer? I already run on Single malt whiskey |
| 17:00.07 | Charlie__ | can anyone tell what could cause those error messages on asterisk 1.8.20.1: channel.c: Exceptionally long voice queue length queuing to Local/394@from-internal-00000510;2 |
| 17:00.17 | Charlie__ | hundreds of them in few seconds.. |
| 17:00.22 | thecoda | has run out of new ways to debug this fubar dahdi config |
| 17:00.52 | Greenlight | Charlie__: What is exten 394 doing ? |
| 17:01.28 | Charlie__ | hi and thanks for your time. extension 394 is calling out using originate call function |
| 17:02.08 | Greenlight | So, in your dialplan, what does 394 try to do? |
| 17:02.09 | Katty | but beer is tasty. |
| 17:02.12 | Katty | whisky not so much. |
| 17:02.20 | Katty | optionally, pear cider is tasty. |
| 17:02.38 | Charlie__ | before those error messages appear? |
| 17:03.14 | Greenlight | Charlie__: I've seen this happening before, there did used to be a few big bugs around it, relating to timing, but think those were all fixed. it's likely because the channel has got in a funny state and isn't processing the voice packets any longer |
| 17:03.47 | Greenlight | Does it stop spamming the messages after a while, or do you need to intervene? |
| 17:03.57 | Charlie__ | yes, it craps the whole asterisk server (reboot or kill & restart of services is required) |
| 17:04.03 | Greenlight | Ouch |
| 17:04.31 | Charlie__ | could this be related to a faulty nic? |
| 17:05.17 | Charlie__ | or is it timing as you suggest? thing is that according to dmesg dahdi detected a timeshift but that could also be down to reboot of the server and ntp kicking in |
| 17:05.21 | Greenlight | I don't *think* so. Hopefully someone more familar with the code than myself can chirp in and indicate the cause of that message. Although seeing the dialplan for that extension would help |
| 17:05.35 | Greenlight | Charlie__: Yes, that would do it |
| 17:05.50 | Greenlight | I've killed Asterisk many a time by doing an ntp update |
| 17:06.12 | Charlie__ | so you think that could be the issue? |
| 17:06.49 | Greenlight | Yea, if the clock on server was changed, then expect the unexpected |
| 17:07.20 | Charlie__ | do you by any chance know to which log does ntp write? |
| 17:07.28 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-okkiaxwvvjpvyyrm) |
| 17:07.34 | Greenlight | Sorry, not off hand |
| 17:08.29 | Charlie__ | ok, will google that |
| 17:09.46 | Charlie__ | thank you for your help greenlight. if you can thing of anything else, let me know ) |
| 17:11.46 | Katty | hum. |
| 17:11.51 | Katty | well i'm all caught up on tickets. now what. |
| 17:13.28 | leifmadsen | Katty: drink! |
| 17:14.12 | leifmadsen | Katty: you could try out Strongbow beer (it is a cider beer) |
| 17:17.48 | cusco | its not beer |
| 17:18.59 | cusco | try Magners |
| 17:21.38 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 17:21.57 | leifmadsen | cusco: not a fan of it, but pretty sure it's still a beer :) |
| 17:22.08 | leifmadsen | disregard, I am wrong again |
| 17:22.57 | leifmadsen | thanks google for making me look like a putz! |
| 17:26.36 | Charlie__ | anyone here? |
| 17:26.51 | leifmadsen | yes |
| 17:26.54 | WIMPy | Define "here". |
| 17:27.06 | leifmadsen | I'm at my desk, which is my current 'here' attribute |
| 17:27.14 | Charlie__ | alive, active, non-bot, non sleeping :) |
| 17:27.30 | Charlie__ | can anyone tell what could cause those error messages on asterisk 1.8.20.1: channel.c: Exceptionally long voice queue length queuing to Local/394@from-internal-00000510;2 |
| 17:27.54 | Greenlight | You checked into the change of server time ? |
| 17:28.01 | Charlie__ | yes |
| 17:28.19 | Charlie__ | did not occur. sync happened only after server reboot |
| 17:28.24 | Greenlight | Ahh okay |
| 17:28.58 | Charlie__ | ntp logs are in /var/log/messages ;) |
| 17:29.38 | Greenlight | Am pretty sure that is caused when whatever's supposed to be connected to the other side of the Local channel isn't taking packets, for some reason. Am sure someone can verify though? In the past there's been issues related to timing sources. What does exten 394 do, and how are calls ending up there ? What happens before those errors in the logs ? |
| 17:31.43 | Greenlight | Wow that's a foobar.... my wholesale SIP carrier just accidentally CC'd all 170 of their wholesale customers openly on an email, instead of using the distribution list. |
| 17:34.57 | Charlie__ | extension 394 is a dynamic agent, calling out using function originate call. last thing before errors is |
| 17:35.01 | Charlie__ | [2013-02-11 12:51:12] VERBOSE[12962] pbx.c: -- Executing [s@macro-dialout-trunk:30] Dial("Local/394@from-interna l-00010e23;1", "CAPI/g9/941:041751816/Bo,300,W") in new stack |
| 17:35.48 | Greenlight | Hmmm... never worked with CAPI channels, so can't advise much, but that's likely where the issue is |
| 17:38.23 | Greenlight | What's the physical setup; what media are you dialling out on ? |
| 17:40.43 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 17:41.22 | Charlie__ | e1 line, connected to e1 card (dialogic diva e1/30). chan_capi is making it work for asterisk, kind of driver |
| 17:41.43 | Greenlight | DAHDI can't be used ? |
| 17:41.51 | Charlie__ | no |
| 17:42.05 | Charlie__ | i'm using dahdi for timing only (dummy) |
| 17:42.10 | Greenlight | I see |
| 17:42.20 | angryuser | hey guys, any yealink users here ? I am trying to provision the T20 phones, it is working fine, but mass firmware upgrade. I've go the simple line : firmware_url = http://centrex.cnsi.fr/1/9.70.0.130.rom under [ firmware ] but after config refresh phone does not download a new one. Any ideas ? |
| 17:42.21 | WIMPy | Charlie__: When does it happen? |
| 17:42.54 | Greenlight | It would seem like the CAPI driver isn't taking packets off from the local channel. |
| 17:43.28 | Charlie__ | usually with high traffic, I think it might be related to originate call function. we had 9 agents calling out using originate call |
| 17:43.47 | angryuser | the rom is on private network, but phones can get them |
| 17:43.59 | WIMPy | Does it happen only at the beginning or end of calls or any time? |
| 17:44.39 | Charlie__ | can't tell... as there were numerous inbound calls and numerous outbound calls all the time. |
| 17:44.56 | *** join/#asterisk ashd (~ashleyd@188-221-47-161.zone12.bethere.co.uk) |
| 17:45.05 | Greenlight | Does the log show the call connecting? DOes it not have timestamps ? |
| 17:45.13 | Charlie__ | if I stick to the extension that was eventually producing errors, it was at the beginning of the call |
| 17:45.24 | Charlie__ | this happened three times today |
| 17:45.32 | Charlie__ | with different extensions |
| 17:45.35 | Greenlight | And each time totally killed the server ? |
| 17:45.38 | WIMPy | It might happen while the call is terminated. In which case it's safe to ignore. |
| 17:45.56 | Greenlight | WIMPy: Yea, but I beleive it;s killing his box each time |
| 17:46.09 | WIMPy | oh |
| 17:46.12 | Greenlight | Indeed |
| 17:46.26 | Charlie__ | symptoms vary, but yes, i had to kill asterisk services and start afresh |
| 17:46.27 | *** join/#asterisk ectospasm (~ectospasm@unaffiliated/ectospasm) |
| 17:46.38 | WIMPy | So what hpenns when it's dead? |
| 17:46.41 | Charlie__ | first time, people could not call in nor out |
| 17:47.27 | Charlie__ | next time, people could call and reach the asterisk, fall into queue, but calls were not distributed to the queue members |
| 17:47.33 | WIMPy | Does that CAPI come with any monitoring tool? |
| 17:50.54 | Charlie__ | e1 card has a driver, interface and few diagnostic tools, yes |
| 17:51.11 | Charlie__ | e1 line as such was tested by telco and is ok |
| 17:51.30 | Charlie__ | another note: we're experiencing those problems after upgrade from asterisk 1.4 |
| 17:51.42 | WIMPy | You need to find out which part gets stuck. |
| 17:51.47 | WIMPy | To? |
| 17:52.19 | Greenlight | 1.8.20 I think he's on now. |
| 17:52.30 | Greenlight | Did you upgrade your CAPI driver ? |
| 17:52.54 | WIMPy | And did you try the other chan_capi? |
| 17:52.56 | Charlie__ | now we have asterisk 1.4.20.1 |
| 17:53.00 | Charlie__ | SORRY |
| 17:53.02 | Charlie__ | 1.8.20.1 |
| 17:53.09 | Charlie__ | :) |
| 17:53.28 | Charlie__ | there's only one chan_capi that would compile against that version, the latest |
| 17:53.41 | Charlie__ | i am not sure if capi is the problem |
| 17:53.50 | WIMPy | No, there are two. |
| 17:54.03 | Charlie__ | are there? |
| 17:54.16 | WIMPy | If the CAPI itself used to work, it has to be chan_capi or Asterisk itself. |
| 17:54.29 | Charlie__ | i've used the one from cytronic and melware, armin being head behind it |
| 17:55.04 | WIMPy | http://www.selasky.org/hans_petter/capi4pbx/ |
| 17:55.06 | Charlie__ | i suspect either asterisk or nic driver (broadcom) |
| 17:55.11 | WIMPy | There is another. |
| 17:55.51 | Charlie__ | do you have any experience with that one from hans petter? |
| 17:56.16 | WIMPy | No. The last time I used CPI is >5 years ago. |
| 17:56.23 | Charlie__ | judging by your name, you're from german speaking world, so you know something about isdn :) |
| 17:56.30 | Charlie__ | ah, ok |
| 17:56.45 | WIMPy | Apart from a simple "does it work" experiment with a FritzBox and remotecapi. |
| 17:57.01 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 17:57.02 | WIMPy | I am |
| 17:57.03 | Charlie__ | ok |
| 17:57.09 | Charlie__ | will certainly have a look |
| 17:57.16 | Charlie__ | thanks for the link and hint |
| 17:57.45 | Greenlight | I suppose one option if you hit a brick wall, is, since you know it works with 1.4.X is to build another box and have that be a SIP<-->CAPI gateway. |
| 17:58.28 | WIMPy | I'd go for a supported card instead if a 2nd PC if I were to go shopping. |
| 17:58.33 | WIMPy | s/if/of/ |
| 17:58.37 | Charlie__ | will not go there |
| 17:58.42 | Charlie__ | i'd rather downgrade |
| 17:59.06 | Charlie__ | diva pri is best there is |
| 17:59.07 | Greenlight | Yea, a DAHDI compatible card would be ideal, but more costly |
| 17:59.28 | WIMPy | I'd pprefer a Linux supported one. |
| 17:59.29 | Charlie__ | but... ok, let's see what we'll find out |
| 17:59.38 | Charlie__ | it is linux supported |
| 17:59.45 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:59.45 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:00.06 | Charlie__ | dahdi cards are way cheaper than diva pri :) |
| 18:00.13 | gauravp | leif: tried a bunch of CALLERID() arguments, but no luck :( |
| 18:00.26 | Charlie__ | ranging from 3.000 to 9000 eur |
| 18:00.27 | leifmadsen | gauravp: sorry to hear that :\ |
| 18:00.28 | WIMPy | Supported BY linux or with support FOR Linux? |
| 18:00.48 | gauravp | i'm now trying to set accountcode in motif.conf and print that from NoOp(${CDR(accountcode)}) ... |
| 18:00.57 | Charlie__ | it's too early to judge what and where is the problem |
| 18:01.01 | Charlie__ | we'll see |
| 18:01.04 | gauravp | thought that would work, but it's empty |
| 18:01.26 | gauravp | although the sample motif.conf seems to indicate you can set accountcode in that manner |
| 18:01.30 | Charlie__ | Wimpy: you're right, it's FOR linux |
| 18:01.41 | Charlie__ | thank you for now |
| 18:02.23 | gauravp | maybe i should look at chan_motif functions like you suggested .. any hints on where i might be able to find what functions exist and how they can be called in a dialplan? |
| 18:02.51 | Greenlight | Right away home to get some dinner ... laters all |
| 18:04.41 | leifmadsen | core show functions |
| 18:04.45 | leifmadsen | that's all I got |
| 18:04.50 | leifmadsen | haven't played with motif too much |
| 18:06.42 | file | recent versions of chan_motif will dial an extension the name of the entry in motif.conf before falling back to "s" |
| 18:07.01 | file | so you can use a single context for multiple accounts inbound |
| 18:08.17 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:09.09 | gauravp | leif: nothing looks promising in the functions .. will keep plugging away at CDR. Never used it before, but any base config required before using CDR? |
| 18:09.49 | file | gauravp, what do you need to do exactly? |
| 18:10.36 | *** join/#asterisk ipiera (~Paul@ipiera.plus.com) |
| 18:11.05 | gauravp | file: so i'm going from chan_gtalk to chan_motif in asterisk 11, and previously i could determine from CALLERID(dnid) the foo@gmail.com google voice account from which an incoming call was coming to direct it to the appropriate extension |
| 18:11.15 | gauravp | with chan_motif, CALLERID(dnid) is empty |
| 18:12.09 | gauravp | per the sample motif.conf provided, it should be possible to set a CDR accountcode in motif.conf, but when i do that, NoOp(${CDR(accountcode)}) is still empty |
| 18:12.23 | file | if your Asterisk 11 is recent then chan_motif will direct the call to an extension named the same as the entry in motif.conf before falling back to "s" |
| 18:12.34 | gauravp | ah.... |
| 18:12.46 | *** join/#asterisk televoip (~kbushong@2620:0:280:12:2e41:38ff:feb1:35b9) |
| 18:12.47 | gauravp | cool, before everything in chan_gtalk went to 's' |
| 18:13.18 | file | I added it based on feedback from users. |
| 18:13.24 | gauravp | i'll give that a go .. |
| 18:13.34 | *** part/#asterisk ipiera (~Paul@ipiera.plus.com) |
| 18:14.27 | file | good, I did not forget to document it in motif.conf.sample! yay |
| 18:14.47 | mjordan | file: did it happen after 11.0.0? |
| 18:14.48 | *** join/#asterisk Uthark (~Uthark@190.0.58.186) |
| 18:14.52 | file | yes |
| 18:14.59 | mjordan | :-D |
| 18:15.15 | mjordan | hesitates to mention UPGRADE.txt |
| 18:15.24 | file | I don't know if I did UPGRADE.txt or not :P |
| 18:15.46 | file | I sorta ... snuck ... it in |
| 18:15.46 | leifmadsen | upgrade all the stacks! |
| 18:19.06 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 18:21.23 | drmessano | lol.. |
| 18:22.25 | gauravp | file: it works (was not in the motif.conf sample btw :p)! might be a good addition here too: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 18:22.45 | file | it's in the 11 branch one I pulled up from the SVN server |
| 18:23.00 | gauravp | i did notice however that when i dropped the call from my asterisk extension, it didn't disconnect the caller |
| 18:23.11 | *** mode/#asterisk [-o CaptainPants] by ChanServ |
| 18:23.42 | gauravp | ie. the caller phone still seemed to think the call was open |
| 18:24.02 | file | I haven't heard of that from anyone else |
| 18:24.08 | gauravp | do i need something after the Dial() ? |
| 18:24.18 | file | nope |
| 18:24.21 | gauravp | ok, i'll test some more |
| 18:25.32 | gauravp | now one more place i was previously using CALLERID(dnid) was to JabberSend a message to myself for incoming calls |
| 18:25.35 | gauravp | that no longer seems to work .. |
| 18:26.46 | file | yeah I got nothin' for that |
| 18:27.02 | file | generally people don't use the same google talk account from multiple systems because it can screw stuff up |
| 18:27.45 | leifmadsen | mjordan: hey! CaptPants is my bot! :) |
| 18:28.19 | gauravp | using the 3 incoming google accounts on only one system, but what i wanted to do was send an xmpp message from the receiving account to my main id (on my cellphone) saying there was an incoming call |
| 18:28.34 | mjordan | this is CaptainPants, totally different |
| 18:28.46 | gauravp | same => n,JabberSend(${CALLERID(dnid)},me@gmail.com,Call from: ${CALLERID(all)}) |
| 18:29.05 | file | ah yeah I still got nothin', should be easy to add though |
| 18:29.07 | gauravp | so me@gmail.com would receive a message from foo, bar and baz@gmail.com when each recd incoming |
| 18:29.27 | file | well unless you added some quick dialplan to set a variable, since you now know what account it came in on |
| 18:30.21 | gauravp | ah, you're right, i'm going to have 3 separate dialplans now for each motif account |
| 18:30.45 | gauravp | i was still thinking of before when i only had 's' and had to redirect calls based on dnid |
| 18:31.01 | gauravp | thanks! |
| 18:31.12 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 18:35.38 | file | some restrictions apply on my IRC messages, please see site for details! get your piece of telephony at asterisk.org |
| 18:37.40 | *** join/#asterisk nightrid3r (~kvirc@62.205.85.51) |
| 18:43.11 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ybytlclxgqufigym) |
| 18:43.16 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ybytlclxgqufigym) |
| 18:43.16 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 18:43.36 | *** join/#asterisk lorsungcu (~anonymous@50-78-230-69-static.hfc.comcastbusiness.net) |
| 18:45.24 | eirirs | hmm |
| 18:46.02 | eirirs | I have a PBX at one subnet, and phones at this subnet works fine, and two GRE tunnels with phones behind those.. they can call out, but incoming calls get busy signal |
| 18:46.07 | eirirs | those phones registers fine |
| 18:46.10 | eirirs | what could it be |
| 18:48.00 | eirirs | I let some Cisco dudes review the config on router and they say it's an asterisk config issue |
| 18:48.37 | Qwell | What does the Asterisk debug say? |
| 18:50.11 | eirirs | -- dialparties.agi: Extension 99 cf is disabled , dialparties.agi: Extension 99 has do not disturb enabled, or followme pre-ring returned busy , ... |
| 18:50.33 | eirirs | with xlite softphone, it works if its on same net as pbx |
| 18:52.13 | eirirs | I have added all three localnets in sip.conf |
| 18:52.33 | *** join/#asterisk bigd1234567890 (~dburgess@63.96.150.226) |
| 18:52.41 | eirirs | and when I try to put the client in TCP mode, debug say 403 unauthorized |
| 18:52.48 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:52.55 | eirirs | while it works fine in tcp at local net |
| 18:55.50 | eirirs | sip show peers shows the peer and says ok with far lower latency time than those in same localnet as pbx |
| 18:57.30 | eirirs | hmm I see in PBX status it says 99@ext-local : SIP/99&Custom:DND99 State:Busy Watchers 0 |
| 18:58.07 | eirirs | possible to change it? |
| 18:59.52 | igcewieling | eirirs: looks to me that the phone may h\ave DND enabled |
| 19:02.02 | eirirs | "My Availability" are greyed out in xlite |
| 19:04.32 | *** join/#asterisk jsmith (~jsmith@fedora/jsmith) |
| 19:04.32 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 19:05.08 | eirirs | with linphone I can change status, but then debug says SIP/2.0 501 Method Not Implemented |
| 19:05.28 | eirirs | maybe the asterisk pbx are too old? the version is Asterisk PBX 1.6.0.26 |
| 19:06.47 | jsmith | eirirs: It's that the SIP channel driver in Asterisk isn't set up to handle SIP presence messages |
| 19:06.59 | *** join/#asterisk Linkforsoad (~Linkforso@2001:1af8:fec1:0:14dc:f243:a9d8:84aa) |
| 19:07.35 | eirirs | jsmith: at this version , or at all newer versions? |
| 19:08.00 | eirirs | can I change the presence status in asterisk CLI ? |
| 19:08.22 | jsmith | eirirs: I think it's for all versions, but I'm not 100% sure. Asterisk has traditionally taken a very call-centric view of the SIP protocol, and not implemented every single RFC that handles non-call-related items in SIP. |
| 19:08.31 | jsmith | eirirs: Not that I'm aware of |
| 19:08.53 | eirirs | hmm ok |
| 19:08.55 | jsmith | eirirs: Presence isn't part of the core SIP standard -- it's an additional set of RFCs |
| 19:09.19 | file | FreePBX does its own thing though, that may be what is getting you |
| 19:09.27 | jsmith | eirirs: (that being said, Asterisk itself has some limited support for presence -- using either device states or the Jabber/XMPP stuff) |
| 19:09.39 | jsmith | Yeah, FreePBX does its own thing |
| 19:09.53 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 19:09.55 | file | and from the bit you pasted you are using FreePBX, and that is indeed what is happening |
| 19:10.17 | [TK]D-Fender | eirirs, * does not respond to presence change requests from devices. |
| 19:14.24 | *** join/#asterisk timahvo1 (~rogue@197.176.192.69) |
| 19:20.33 | *** join/#asterisk bpietro (~bpietro@82.51.236.132) |
| 19:23.13 | *** join/#asterisk coreyf1513 (~cfarrell@ws2.cfware.com) |
| 19:25.41 | eirirs | lol bleh why didn't I come on the idea by using Feature Code... *76 solved it all |
| 19:28.22 | eirirs | facepalm |
| 19:28.43 | Katty | i am a nervous wreck. |
| 19:28.53 | Katty | contacts in 3 hours >.< |
| 19:29.02 | Katty | i am not ready for the eye pokey bits! |
| 19:29.15 | eirirs | lol |
| 19:30.16 | bpietro | hi, I've one simply question: on asterisk site I can see "latest version 11.2.1" and on net I can find .deb package numbered 1.8.10. Is .deb package so tremendously obsolete or it was some numbering switch? |
| 19:31.14 | [TK]D-Fender | Katty, I spent 3/4 of an hour fighting to get them in the first time, in full view of the mall..... a humbling experience to say the least |
| 19:31.42 | [TK]D-Fender | bparker, Debian = decrepit, and Digium doesn (AFAIK) do their own for it yet |
| 19:31.47 | [TK]D-Fender | bpietro, ^ |
| 19:32.00 | Katty | well i don't plan on putting them in, in full view of the mall. |
| 19:32.17 | Katty | they'll probably require me to put them in, on site |
| 19:32.20 | Katty | before i stroll off with them |
| 19:32.37 | Katty | but all my goof-ups are between me and my dr! |
| 19:32.50 | Katty | [TK]D-Fender: any tips for not poking myself in the eye? |
| 19:33.57 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 19:34.23 | [TK]D-Fender | Katty, Just do it slow. The hardest part will be resisting blinking and knocking it out. NExt comes alignment as you fight the urge. |
| 19:34.44 | Katty | i don't think blinking will be an issue |
| 19:34.47 | bpietro | [TK]D-Fender: Iwant install * on little box for testing purposes before installing it (from sources) on production box. I'll compile it for little test box too. Tnx |
| 19:34.50 | Katty | i regularly put on eyeliner...which requires not blinking |
| 19:35.02 | Katty | and i don't mean water line, or above the lid. i put it on the tide line |
| 19:35.02 | [TK]D-Fender | Katty, I meant that I was visible within the store which was in a large mall. Not that I was on the main floor. |
| 19:35.07 | Katty | which is very difficult to get to, for some ;) |
| 19:35.23 | Katty | but yes, slow sounds like a good plan. |
| 19:35.31 | [TK]D-Fender | Katty, This is different. You'll find your body going "NOPE!" when things tough your eye... |
| 19:35.43 | [TK]D-Fender | touch* |
| 19:35.44 | bparker | there are no girls on the internet |
| 19:35.44 | bparker | ! |
| 19:35.48 | igcewieling | Can anyone think of why my asterisk box might have 4984 active SIP dialogs with 1 active call? |
| 19:36.09 | Katty | [TK]D-Fender: yes, indeed. |
| 19:36.11 | igcewieling | The Internet, where the men are men, the women are men, and the children are FBI agents. |
| 19:36.16 | Katty | [TK]D-Fender: like when i accidently poke my eye with the pencil |
| 19:36.19 | [TK]D-Fender | igcewieling, trailing qualifies, channels waiting fonal expiration, etc. |
| 19:36.19 | bparker | precisely |
| 19:36.22 | Katty | [TK]D-Fender: it gets all watery and not pleasant feeling |
| 19:36.26 | chuckf | Katty: my ex gf took about 1.5 hours to get her contacts in the first time |
| 19:36.27 | [TK]D-Fender | Katty, Happens ALL the time ;) |
| 19:36.46 | Katty | chuckf: well that's encouraging |
| 19:36.54 | Katty | chuckf: at a realistic time frame to shoot for |
| 19:37.32 | chuckf | Katty: also some that I've known only took about half hour the first time. It varies |
| 19:37.45 | coreyf1513 | Katty: took me like 10 minutes to put in contacts the first time. i had trouble removing them :( |
| 19:37.48 | chuckf | but don't be shocked if its a really long time |
| 19:38.47 | Katty | nods |
| 19:39.04 | Katty | it takes the boy maybe...30 seconds to get both of his in |
| 19:39.12 | Katty | so i'm encouraged that i'll get used to it, eventually |
| 19:40.03 | chuckf | you will, it'll just take a couple weeks to get that quick |
| 19:41.12 | *** join/#asterisk ra21vi (~ravi@122.177.241.53) |
| 19:41.41 | thecoda | What's the best kind of industrial hardware for thoroughly destroying a recalcitrant asterix box? |
| 19:41.49 | thecoda | s/asterix/asterisk |
| 19:41.57 | ra21vi | Once a caller is connected to an agent from queue, is it possible to query some vars (GETVAR) from callers channel?? |
| 19:42.26 | [TK]D-Fender | Katty, As I said, my first time was massively frustrating, but then they just go in after a few days... |
| 19:42.37 | [TK]D-Fender | thecoda, C4 <- |
| 19:42.59 | ra21vi | I was initially working on AMI to listen to event Bridge, which is fired once a caller in queue is connected to an agent.. |
| 19:43.12 | thecoda | Wouldn't give the quite the hands-on satisfaction I'm craving |
| 19:43.15 | ra21vi | But in my situation, this bridge event is fired more than one time... |
| 19:43.15 | Katty | wibbles uneasily |
| 19:43.33 | igcewieling | thecoda: I think the recommended way is to draw a pentagram around the box, sacrifice a goat and then use C4. |
| 19:43.37 | ra21vi | so I am unable to get the channel name of current call to an agent |
| 19:43.56 | igcewieling | looks at [TK]D-Fender |
| 19:44.11 | thecoda | So no chainsaw? I've topped it up with petrol and *everything* |
| 19:44.43 | igcewieling | thecoda: when was the last time anything with a chainsaw ended well? |
| 19:45.27 | thecoda | igcewieling: Oh, it's already ended *badly*, I'm past that now and onto the payback stage |
| 19:46.37 | thecoda | Wretched thing simply cannot figure out when the other end has answered a call and I've exhausted absolutely every way I can think of to diagnose it and every possible source of help |
| 19:46.52 | thecoda | so now it's revenge time |
| 19:49.38 | igcewieling | thecoda: you're not doing something stupid like expecting an FXO channel to detect answer, are you? |
| 19:50.26 | thecoda | Inbound calls work just fine, it can't detect the remote answer on an outbound call |
| 19:50.30 | mjordan | ra21vi: ASTERISK-18639 |
| 19:50.39 | mjordan | the Bridge event is less than helpful currently |
| 19:51.44 | igcewieling | thecoda: that did not answer my question. |
| 19:51.56 | thecoda | http://pastebin.com/L7mBJ66Y |
| 19:53.00 | igcewieling | thecoda: looks like the answer is "yes", though you are in the UK so I can't be sure. Generally FXO ports (fxs signaling) cannot EVER detect when the far end answers.\ |
| 19:53.09 | thecoda | hmm |
| 19:53.30 | igcewieling | Regular PBXs have the same issue for the most part. |
| 19:54.32 | thecoda | doubly-confusing then, this is the auto-generated dahdi-channels: http://pastebin.com/xPCBRCXd |
| 19:55.05 | ra21vi | mjordan: thank you. It spoiled hot water on my long working plan to develop CTI app :) |
| 19:55.12 | thecoda | I just went on the fact that the config tool had assigned that channel to the from-pstn context |
| 19:55.26 | ra21vi | mjordan: but its better to know issue sooner than being late.. Thanks |
| 19:55.28 | igcewieling | not sure what you are talking about. |
| 19:56.08 | igcewieling | Asterisk users chan_dahdi.conf, it uses NO other config files for DAHDI unless you manually #include them in chan_dahdi.conf |
| 19:56.18 | ra21vi | Now I think someone here will help me. I am out of options. |
| 19:56.26 | ra21vi | Let me describe my situation |
| 19:57.15 | thecoda | igcewieling: The dahdi-channels.conf (2nd file I posted) is what was auto generated by dahdi_genconf |
| 19:57.45 | thecoda | I used it as a guide when I wrote chan_dahdi |
| 19:57.51 | ra21vi | For a small helpdesk IVR, I wrote an AGI script which asks user to enter Ticket No. Then it stores that (in current channel) using SETVAR. Call is then routed to Queue, where agents are waiting. Once an agent gets that call, I want to show the ticket no. usign Screen Pop. |
| 19:58.46 | ra21vi | I was working with AMI to listen on events. Can anyone help me how I can fetch the Ticket ID set in original callers channel before he was sent to queue, in my desktop app using AMI |
| 19:59.11 | thecoda | The "context=from-pstn" is therefore the output of dahdi_genconf, and I took it on faith as being correct :) |
| 19:59.37 | igcewieling | thecoda: unless you #include that file in chan_dahdi.conf it will NOT be used. |
| 20:00.07 | thecoda | Didn't want to include it, it's a legacy format if I understand correctly? |
| 20:00.20 | igcewieling | thecoda: context= is for INCOMING calls. So it has nothing to do with your OUTGOING problem |
| 20:00.23 | thecoda | I simply copied of the relevant settings |
| 20:01.58 | thecoda | sooooo. line 1, FXS, is configured for the from-pstn inbound context, which only makes sense if it's the port attached to my PSTN phone line, and is therefore also the port on which to make outbound calls |
| 20:02.07 | thecoda | or did I miss something really stupid here? |
| 20:02.21 | coreyf1513 | ra21vi: i'm not familiar with how app_queue works but I'm very familiar with AMI. Have you found the caller channel from the agent ami? |
| 20:02.22 | mjordan | ra21vi: listen for the Newchannel event to determine the channel name associated with the caller. Once you have that, listen for the VarSet event to get the Ticket ID for that channel. (https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_VarSet) |
| 20:02.31 | *** join/#asterisk miltux (~miltux@62.1.139.246.dsl.dyn.forthnet.gr) |
| 20:03.21 | igcewieling | thecoda: the only thing you missed is that analog lines generally cannot signal the far end that the call was answered. that is your problem, right? Asterisk doesn't know when an outgoing call is answered by the far wne? |
| 20:03.23 | ra21vi | coreyf1513: Yes some events like Join returns channel ID. I may be wrong a little |
| 20:03.37 | thecoda | igcewieling: exactly :) |
| 20:03.37 | igcewieling | s/wne/end |
| 20:03.52 | ra21vi | mjordan: since the app will be installed on 3-4 desktop, I will have to handle all events traffic to filter the related one. Am i right? |
| 20:04.10 | igcewieling | thecoda: use PRI or SIP to a provider and your problem will go away. WHY do you need asterisk to know that the far end answered? |
| 20:04.28 | mjordan | ra21vi: you'll get lots of VarSet events, so you'll need to know what channel you're looking for |
| 20:04.29 | ra21vi | mjordan: is it how other commercial CTI apps do? Or is there any easy way to access the callers initial channel where Variable was set |
| 20:04.35 | WIMPy | Or more likely a BRI. |
| 20:04.58 | igcewieling | ra21vi: Have you looked at CEL? CEL can generate manager events and may be more reliable for what you want. |
| 20:05.07 | thecoda | Because I wish to make a call over my PSTN line via my asterisk box, local calls are free and often higher quality |
| 20:05.29 | igcewieling | thecoda: you do not need to know that the far end answers to do that. |
| 20:05.33 | ra21vi | igcewieling: No. I have not read about CEL. Will consult the doc :) |
| 20:05.50 | igcewieling | ra21vi: 1.8+ only I think |
| 20:06.07 | igcewieling | thecoda: analog calls are considered answered as soon as dialing is completed. |
| 20:06.22 | ra21vi | mjordan: yes, In my current AMI script which is listening for manager events, it gets a hell of VarSet events. |
| 20:06.57 | ra21vi | igcewieling: oh. Sorry, I am working on 1.8 only. There are some restrictions to choose latest |
| 20:07.17 | ra21vi | igcewieling: since I am using Elastix, its 1.8.. |
| 20:07.19 | coreyf1513 | ra21vi: you might be better to filter out VarSet event and use the ami command Getvar when you need to retrieve the ticket id |
| 20:07.20 | thecoda | igcewieling: I call my mobile, my mobile rings, I answer the call on my mobile, my landline is still playing me the ringing sound, I dial numbers on my mobile, asterisk shows me the DTMF tones on the console, I still just hear ringing on my landline |
| 20:07.29 | igcewieling | ra21vi: 1.8 has CEL |
| 20:07.30 | thecoda | and no voice |
| 20:07.46 | igcewieling | thecoda: ah, you have a different problem. |
| 20:08.11 | ra21vi | igcewieling: ok, let me consult CEL doc. |
| 20:08.19 | igcewieling | thecoda: is this your current chan_dahdi.conf? http://pastebin.com/xPCBRCXd |
| 20:08.31 | ra21vi | Thanks everyone, mjordan coreyf1513 igcewieling :) |
| 20:08.50 | thecoda | igcewieling: yup |
| 20:09.16 | thecoda | igcewieling: No even, this is: http://pastebin.com/L7mBJ66Y |
| 20:09.28 | igcewieling | thecoda: Some places in the UK use a 3-wire analog sort of setup. Is your set up 3-wire or 2-wire? |
| 20:10.25 | thecoda | The former was the auto generated dahdi-channels from genconf |
| 20:10.40 | thecoda | igcewieling: 2-wire to the best of my knowledge |
| 20:11.19 | *** join/#asterisk lorsungcu (~anonymous@24-196-56-142.static.stcd.mn.charter.com) |
| 20:11.58 | igcewieling | thecoda: I would set usecallerid=no, comment out all the CID and polarity settings, and restart asterisk. Just in case they are causing an issue. Not likely, but possible. On the console you should see an Answer messages as soon as Asterisk finishes sending the DTMF down the line. |
| 20:12.22 | *** join/#asterisk b2 (~ion@pdpc/supporter/active/beckb) |
| 20:12.47 | igcewieling | Oh! and set callprogress-no, that is likey your issue. callpogress is an alias for the "screwupmycalls=yes" setting. |
| 20:14.12 | thecoda | usecallerid is for inbound or out? |
| 20:16.00 | igcewieling | thecoda: normally inbound, but since you are having SO many issues, you should comment them all out for testing. |
| 20:16.37 | thecoda | Just the one issue, but it's a bad one :) |
| 20:17.07 | thecoda | going to try turning off call progress first, looks a likely suspect after reading up on it |
| 20:17.52 | thecoda | "If turned on, call |
| 20:17.53 | thecoda | progress attempts to determine answer, busy, and ringing on phone lines. |
| 20:17.54 | thecoda | This feature is HIGHLY EXPERIMENTAL and can easily detect false answers, |
| 20:17.55 | thecoda | so don't count on it being very accurate." |
| 20:18.38 | *** join/#asterisk mrpixel (8ea34ea4@gateway/web/freenode/ip.142.163.78.164) |
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| 20:19.03 | Kobaz | thecoda: ie: disconnect calls randomly |
| 20:19.15 | thecoda | I don't want to mess with callerid having only recently made it work :) |
| 20:21.37 | mrpixel | Looking for specific information on defining the destination port that asterisk (1.8.12.0) uses when doing the re-invite for T.38 negotiation for incoming transmissions. From what I can tell it currently picks a high level port at random but our provider requires it to be 5060 |
| 20:22.06 | mrpixel | I've been searching on google for a few hours now but haven't found anything in specific that would allow me to define this within asterisk |
| 20:22.42 | igcewieling | mrpixel: SIP is 5060, but RTP defaults to 10000 - 20000 and udptl (i.e T.38) defaults to 4000 - 4999 see udptl.conf IIRC |
| 20:22.50 | igcewieling | also see rtp.conf |
| 20:22.53 | mrpixel | k |
| 20:23.15 | mrpixel | I was looking at that |
| 20:23.32 | mrpixel | but the port asterisk chose in last trace was 34499 |
| 20:23.54 | mrpixel | which isn't in a defined port range that I could find |
| 20:25.18 | kaldemar | mrpixel: it is what the peers used as its source port. |
| 20:26.08 | mrpixel | kaldemar: so you're suggesting the carriers equipment made that choice? |
| 20:26.28 | thecoda | igcewieling: Good news and bad news |
| 20:26.39 | thecoda | good news: it works, answering no problem |
| 20:26.51 | kaldemar | it is not asterisk that picks that port, most likely. you'll see it all in sip debug. |
| 20:27.17 | thecoda | bad news: It dials, then pauses, then dials again. The first is obviously from asterisk and the 2nd from the remote |
| 20:27.54 | mrpixel | kaldemar: that was how I felt about the situation but they are suggesting that it is a configuration problem on our end |
| 20:28.01 | thecoda | Any way to suppress the first dialling sound? |
| 20:28.45 | thecoda | s/dialing/ringing |
| 20:28.57 | WIMPy | thecoda: Do you have a "r" in your Dial options? |
| 20:29.16 | thecoda | yup |
| 20:29.18 | thecoda | drop it? |
| 20:29.25 | WIMPy | yes |
| 20:30.50 | thecoda | hmm, one tiny hint of a ring there, but otherwise fine |
| 20:31.17 | thecoda | takes it's time connecting though, any hint of how I might speed that up? |
| 20:33.08 | ra21vi | igcewieling: I am reading CEL doc. But I am not able to get how this is different than simple events. |
| 20:34.41 | ra21vi | igcewieling: according to doc, I can register in cel.conf about which application to monitor for CEL events. But my problem is, once the caller gets into Queue, any agent can pick his call. How can I send event that can be meaningful for that particular agent. |
| 20:36.12 | *** join/#asterisk nny (~Scott@174.107.223.14) |
| 20:36.27 | ra21vi | igcewieling: is there any application which queue uses in back to connect to agent (bridging caller's channel to agents channel)?? |
| 20:39.36 | nny | i have 2 pbxs and when a user wants to dial another user on pbx 2 the dialplan does "Dial(SIP/user@pbx2)". It's complaining of auth mismatch as the auth is user@pbx2 vs pbx1@pbx2. I thought canreinvite=no would resolve this but it's back. Thoughts? |
| 20:40.52 | WIMPy | First use SIP/pbx2/user and second that has nothign to do with reinvites. |
| 20:41.02 | leifmadsen | +1 |
| 20:41.06 | leifmadsen | and this is documented :) |
| 20:42.04 | nny | so the proper dial string is Dial(SIP/pbx2/user)? |
| 20:42.23 | WIMPy | yes |
| 20:42.46 | nny | WIMPy: thank you, i'll test it. It oddly wasn't an issue yesterday and then it was today. |
| 20:44.12 | *** join/#asterisk Dovid (~Dovid@173.63.105.210) |
| 20:44.35 | igcewieling | ra21vi: I don't know much about Asterisk Queues, other than they are a hassle to deal with. However, CEL does generate various BRIDGElike events |
| 20:45.26 | igcewieling | nny: fromuser= in the calling peer will specify the username for outgoing calls from the peer |
| 20:45.49 | ra21vi | igcewieling: oh ok :) .. Actually once the user is sent in Queue, I don't see any option how CEL event will be fired.. So anyway, thank you. Atleast I got a new topic to understand :) |
| 20:46.25 | igcewieling | ra21vi: what I did was write a little script to connect to AMI and spit out all the CEL events so I could figure out what events I needed to process |
| 20:47.40 | nny | igcewieling: i am slightly confused as to why. If I dial a sip peer that isn't an asterisk system it is just dial SIP/XXX@peer, yet when it's 2 asterisk boxes it sends the user as auth? |
| 20:48.41 | nny | igcewieling: and I should designate fromuser=XXX in each pbx's sip.conf defintion? (ex [pbx1] fromuser=pbx1 etc) |
| 20:49.11 | nny | does this information supercede the Dial(SIP/pbx/user) or is that an alternative? |
| 20:50.00 | igcewieling | nny: I can only tell you that I use fromuser= to set the username to connect to the far side asterisk box as |
| 20:51.01 | nny | igcewieling: so I would enter this in the context for each asterisk box |
| 20:52.45 | igcewieling | in the peer definition in sip.conf on the calling PBX. |
| 20:53.20 | igcewieling | as to why you only have an issue when calling an Asterisk box, I suspect other boxes either don't use auth or auth by IP, not by user. |
| 20:54.32 | ra21vi | Is NewChannel event fired when a call is made to asterisk? Does it also being fired in between any intermediate steps like sending to queue or bridge or transferring calls etc? |
| 20:55.28 | *** join/#asterisk Mon|A|rch (~SBean@72.29.180.35) |
| 20:55.45 | WIMPy | Each time a new channel is created. |
| 20:56.47 | ra21vi | WIMPy: oh, so in a call flow, there may be lots of channel created. Got it. Thank you |
| 20:57.09 | WIMPy | If you Dial. |
| 20:57.12 | Mon|A|rch | is there a different name for the asterisk-devel package by any chance? |
| 20:57.18 | Mon|A|rch | other than asterisk-devel |
| 20:58.21 | ChannelZ | asterisk-current.tar.gz :P |
| 20:58.38 | ra21vi | WIMPy: does queues uses Dial app internally to connect a caller to an agent? |
| 20:59.13 | ChannelZ | I doubt there is a -devel package, probably just the source package. There is no Asterisk shared library |
| 20:59.34 | mjordan | ra21vi: not really. There are dial mechanics that both use, but they're implemented separately. |
| 21:00.10 | mjordan | ra21vi: As for Newchannel, its fired on all channel creation, regardless of how that channel was created |
| 21:00.36 | mjordan | it does not apply to transfers/queues/bridges, unless those actions create a new channel (which sometimes they do, depending on what you've done and under what circumstances) |
| 21:00.39 | Mon|A|rch | ChannelZ, ah |
| 21:00.45 | Mon|A|rch | i was just looking for the development headers |
| 21:01.16 | WIMPy | ra21vi: I don't know how exactely, but it has to createchannels to call an agent. |
| 21:01.58 | ra21vi | mjordan: WIMPy : Yes it creates a new channel to Ring Agents and then joins the callers channel with agents channel who picks it |
| 21:03.15 | ra21vi | mjordan: WIMPy : That's why I am getting few channels created in between (due to a bug listening on DTMF) with new uniqueid and channel id |
| 21:06.06 | mjordan | ra21vi: yup. You're going to get Newchannel events quite often, particularly if you have Local channels in the mix. In order to get the Newchannel event with the channel name that you care about, you have to know something about how the channel is being created. For example, if your callers all enter from the same context/exten/priority, you can use that to figure out which Newchannel event you care about |
| 21:09.10 | ra21vi | mjordan: that is a nice tip, never thought of it. Thank you mjordan |
| 21:09.30 | Mon|A|rch | what's the command to display information on a function? |
| 21:09.39 | leifmadsen | core show function <FUNCTION_NAME> |
| 21:09.42 | Mon|A|rch | thanks |
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| 21:31.08 | Mon|A|rch | has anyone used espeak with asterisk? |
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| 22:11.48 | *** join/#asterisk crazed1 (cf4322ee@gateway/web/freenode/ip.207.67.34.238) |
| 22:11.52 | crazed1 | does anyone know what condition causes the error: [2013-02-11 14:07:54] WARNING[26051][C-0000016b] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) |
| 22:16.21 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 22:18.36 | *** join/#asterisk Dovid (~Dovid@173.63.105.210) |
| 22:20.55 | mjordan | crazed1: you attempted to Dial a SIP peer that was unregistered |
| 22:21.04 | mjordan | (most likely) |
| 22:21.29 | crazed1 | thank you mjordan :) |
| 22:21.51 | mjordan | interestingly enough, that's not listed on the hangup cause mappings, so I'll have to update the table :-P |
| 22:22.16 | crazed1 | yay :) i helped! |
| 22:22.47 | mjordan | hm. Well, sort of. In retrospect, it isn't listed because it's an error condition and doesn't map back to a SIP cause code |
| 22:23.15 | mjordan | we can at least put that subscriber absent == unregistered |
| 22:25.39 | crazed1 | During random parts of the day, we have horrendous static, calls drop, silence, this is often accompanied by chan_iax2 errors: 'Received trunked frame befeore first full voice frame', although we use a lot of sip trunks and don't get those errors from sip, but stlil get the static |
| 22:26.38 | crazed1 | i'm pretty sure its the network (or the PBX has bad hardware and i just cant tell) |
| 22:27.19 | crazed1 | its random, not even during heavy load or light, typically its under significant use when it happens, but dropping the load doesn't help at all |
| 22:27.39 | crazed1 | i'm finding almost no corresponding data in the erro logs so it's a bit maddening |
| 22:27.48 | igcewieling | crazed1: does it happen if you switch to using SIP? |
| 22:28.00 | crazed1 | yes it still happens with sip |
| 22:28.17 | igcewieling | For the most part people solve IAX2 problems by switching to SIP. 8-) |
| 22:28.47 | crazed1 | only the outbound calls are over iax and it's happening to all calls :/ |
| 22:41.11 | mjordan | crazed1: if it's happening to both, you may just be suffering from jitter |
| 22:41.22 | mjordan | what version of Asterisk are you on? |
| 22:42.54 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-176-240.red.bezeqint.net) |
| 22:43.00 | crazed1 | 1.8 and 11.0.1 |
| 22:43.25 | crazed1 | we have a colo that transfers us calls via sip, and even those calls are horrid |
| 22:43.37 | mjordan | in 11, you could use FUNC_JITTERBUFFER to put a jitter buffer on the affected channels |
| 22:43.55 | crazed1 | can you explain a little more? |
| 22:43.58 | mjordan | there are jitter buffers in 1.8 as well, but they only exist for the inbound call legs |
| 22:44.22 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_JITTERBUFFER |
| 22:45.03 | mjordan | scratch my statement: the problem in 1.8 is that they are only on the write side, not the read side :-) |
| 22:45.09 | igcewieling | crazed1: does "ifconfig" show anything odd? |
| 22:45.22 | crazed1 | nope ifconfig is all normal |
| 22:46.05 | igcewieling | what sort of jitter does a ping -c 100 ip.of.other.server from your asterisk box show? pings are only a START, they can easily show results which don't translate to SIP/RTP |
| 22:46.51 | crazed1 | its fine now, but so are the calls. i'll try that next time |
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| 22:51.57 | crazed1 | whats the simplest way to apply jitter buffer to all my incoming and outgoing calls? |
| 22:52.53 | crazed1 | fyi i'm not talking about a bit of crinkly noise, i'm talking like 200-1500+ms seconds of silence on (usually) both ends |
| 22:53.24 | mjordan | hm. |
| 22:53.34 | crazed1 | we did have some jitter issues in the past, but i noticed error messages in dmesg that corresponded, now i dont see anymore of those errors |
| 22:53.39 | mjordan | I'm a little hesitant to advocate much without knowing what the traffic/network conditions look like. |
| 22:54.11 | mjordan | You could try a jitter buffer, but if your network conditions are severe it will only do so much to help |
| 22:54.23 | mjordan | and it will only affect media, not signalling |
| 22:54.53 | mjordan | you could put one on an inbound channel easily, using Set(JITTERBUFFER(adaptive)=default) |
| 22:54.55 | crazed1 | well upstairs is conneted to downstairs via a 1 gb pipe. upstairs has a catalyst 4007, downstairs has the pbx's, the router, and a catalyst 6500 |
| 22:55.18 | crazed1 | most of the sales people are downstairs |
| 22:55.21 | mjordan | okay, but I still don't know if you're actually having lots of dropped packets, retransmits, etc. |
| 22:55.41 | mjordan | it's great to say "you shouldn't", but without evidence, I have no idea |
| 22:55.56 | crazed1 | i have checked into the router, and the upstairs switch, but i can't figure how to find the ip of the downstairs switch |
| 22:56.10 | drmessano | cisco? |
| 22:56.21 | crazed1 | i'm upstairs, and i run wireshark and other network monitoring tools, it really doesn't seem like we're using that much network resources, i dont understand it |
| 22:56.25 | mjordan | outgoing you can set using a pre-dial routine on the outbound channel (Dial option b) - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial |
| 22:56.31 | *** part/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 22:56.37 | mjordan | might not be jitter then. |
| 22:56.57 | mjordan | do you have any compiler options enabled, such as DEBUG_THREADS? |
| 22:57.40 | crazed1 | nope =/ |
| 22:58.03 | crazed1 | is there anything i should know about upgrading 11.0.1t 11.2.x? |
| 22:58.17 | crazed1 | 11.0.1 to 11.2.x? |
| 22:58.31 | mjordan | I'd check the change log of course, but there weren't any behavior modifications (those would be in UPGRADE) |
| 22:59.44 | mjordan | huh. Actually there were two :-) |
| 23:00.41 | crazed1 | oh you might be interested in a safe_asterisk bug |
| 23:01.23 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:02.31 | mjordan | is it the one that spits out the error message? |
| 23:02.37 | drmessano | 11.0.1 was the last version that didn't die after 24 hours with SIP TLS enabled |
| 23:02.42 | drmessano | Thats about it.. :) |
| 23:02.46 | crazed1 | this system has cat /proc/sys/fs/file-max = 6514339, but ulimit -n $MAXFILES (6514339) gives the error: ulimit: open files: cannot modify limit: Operation not permitted |
| 23:02.59 | crazed1 | so it would end up using the default (1024)... |
| 23:03.27 | crazed1 | i had to hard code it to 131072 so it wouldn't be stuck with 1024 |
| 23:03.45 | mjordan | crazed1: that sounds like an odd thing with the system |
| 23:03.58 | crazed1 | yea its not a real bug, just a bug on a system like this |
| 23:03.59 | mjordan | drmessano: SIP TLS is broken after 11.0.1?? |
| 23:04.03 | crazed1 | ^^^^ |
| 23:04.06 | drmessano | I have the same issue here on a VM, crazed1 |
| 23:04.25 | drmessano | cannot modify limit |
| 23:04.34 | crazed1 | you can modify the limit, you just can't go very high |
| 23:04.45 | crazed1 | ulimit -n 131072 works fine |
| 23:05.19 | drmessano | mjordan: Yeah, SIP TLS has been giving me fits, and I haven't had enough time to really debug it. I tried recompiling with DEBUG_THREADS, but SIP just seems to STOP working with nothing logged in error |
| 23:06.11 | drmessano | It works fine with 11.0.1, but since one of the 11.2.0 RCs, it's been puke |
| 23:06.22 | crazed1 | what about 11.1.0? |
| 23:06.58 | drmessano | Actually, hang on |
| 23:08.20 | drmessano | It may have been 11.1.0 |
| 23:08.35 | drmessano | I started having problems around christmas, which would have been an 11.1.0 upgrade |
| 23:08.39 | drmessano | Based on release dates |
| 23:08.52 | *** join/#asterisk dpilon (~dpilon@50.138.178.238) |
| 23:09.15 | drmessano | Changed nothing other than upgrading in place, as I have done so many times before.. I've installed every update since |
| 23:09.18 | crazed1 | how do i get the 11.0.1 source? i can't find it on the webdsite |
| 23:09.23 | mjordan | hm. There was a substantial rework of the SIP TCP stack during that timeframe, due to not handling fragmented messages properly (and some other funkiness). |
| 23:09.32 | mjordan | drmessano: if you can get some logs, file a bug report |
| 23:09.56 | mjordan | crazed1: export from svn is probably the easiest. |
| 23:10.50 | drmessano | mjordan: Thats really the problem.. I can't get anything worth posting.. One minute it works, next it doesn't. No bullet fired. Guess I am digging deep enough |
| 23:10.57 | drmessano | not* |
| 23:11.18 | drmessano | Mostly been a time issue.. Box crashes, I reboot the VM, move on to next issue |
| 23:11.24 | drmessano | Ugh |
| 23:11.35 | crazed1 | # svn checkout http://svn.asterisk.org/svn/asterisk/branches/11 asterisk-11.0.1 doesn't work |
| 23:11.39 | drmessano | s/Box crashes/SIP stops working/ |
| 23:11.42 | mjordan | hm. If you can get a log leading up to the freeze up, I can at least take a look. |
| 23:11.54 | mjordan | May not show anything, but it may illustrate something |
| 23:12.58 | mjordan | crazed1: try svn export http://svn.asterisk.org/svn/asterisk/tags/11.0.1 asterisk-11.0.1 |
| 23:13.47 | drmessano | I will try. Honestly, I hate posting bug reports when I have little info to go on. I post "its crashing and it sucks and all I have is this" and I get a hearty WORKSFORME and a CLOSE, which only pisses me off and makes me not want to post again. I can understand the OTHER side, but it mostly seems like a waste :/ |
| 23:14.57 | mjordan | if you can narrow it down to the rev that worked versus not, that's a pretty good indication that something went wrong |
| 23:15.23 | carrar | There is always a lucrative career in dairy farming |
| 23:16.20 | drmessano | How about this.. I will wait until it crashes again, pull whatever I have.. Then downgrade to where I believe it worked, and at least offer that up. I will also namedrop so when someone looks at my bug report and has an itchy CLOSE finger, I can say MJORDAN TOLD ME TO POST WHAT I HAVE NOW SHUTURFACEHOLE |
| 23:17.11 | carrar | name drop all the nick names in the channel right now |
| 23:17.21 | drmessano | lol |
| 23:19.53 | *** join/#asterisk tamiel (~tamiel@90.5.164.196) |
| 23:24.29 | mjordan | drmessano: that works |
| 23:25.27 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 23:26.46 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ybytlclxgqufigym) |
| 23:35.34 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200:221:6aff:feb8:e0b2) |
| 23:39.16 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 23:45.44 | crazed1 | i know this is way beneath you guys, but do you mind telling me the command to upgrade asterisk in place? (without overwriting stuff ofc ;)) |
| 23:49.07 | crazed1 | nvm i'll jus tdo it , thot maybe there was a trick |
| 23:49.15 | crazed1 | that only the cool kits know about |
| 23:57.43 | ChannelZ | just don't "make config" which will trash your configs with the samples |