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| 00:17.26 | ankurgupta0001 | navasimo,I have sip application on iphone |
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| 00:29.03 | ankurgupta0001 | MWI not working when moved voicemail storage to mysql database,please help |
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| 05:21.55 | dongs | any decent asterisk manager apps for android? call status/transfer/etc |
| 05:22.26 | dongs | something that looks liek a keysystem and not a bunch of white text on black background |
| 05:22.33 | dongs | i.e. stuff peopel can use |
| 05:30.43 | Wiretap | knock one up |
| 05:30.47 | Wiretap | AMI isn't that painful |
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| 05:32.12 | dongs | Wiretap: are you for real? its 2013 |
| 05:32.34 | Wiretap | dongs, yes, I am |
| 05:32.34 | dongs | I mean, back in 2001 or so I would consider doign that for about 5 minutes |
| 05:33.41 | kaldemar | is programming so last season nowadays? |
| 05:34.31 | dongs | kaldemar: whats wrong wiht wanting a working solution? |
| 05:34.39 | dongs | I didn't say I want it for free. |
| 05:34.42 | dongs | I just want shit that works. |
| 05:35.32 | dongs | https://play.google.com/store/apps/details?id=com.asteriskdroid.amgr&feature=also_installed i mean, whaT?! |
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| 06:28.09 | srp_ | Hi, I'm thinking of using starpy of scripting AMI and AGI. I wanted to know if supports asterisk-11.2.1 |
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| 06:34.19 | ChannelZ | hrrm. Why is make menuconfig |
| 06:34.21 | ChannelZ | oops |
| 06:35.16 | ChannelZ | why is 'make menuconfig' on one of my machines a blue BG with a grey panel, whereas on another it's this disgusting hot pink/purple background? |
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| 06:54.29 | Kalidarn | is there a command I can give to asterisk to make it core dump |
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| 06:54.45 | Kalidarn | well crash, that will produce a core dump |
| 06:55.36 | Kalidarn | I'm running it with the -g switch, so it should dump it's core, i'm just not sure where it will dump it. |
| 06:55.42 | Kalidarn | ls / |
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| 07:28.31 | fling | how to make asterisk to play short beeps when everyone is busy? |
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| 07:33.36 | razorgank | i've been trying to trace my response ok to an INVITE sent by signalling and it's not getting to the destination, any tools to trace up to what point where my packets reach? |
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| 07:39.16 | fling | ChannelZ: what about make nconfig? |
| 07:41.08 | fling | ChannelZ: also that box probably has different terminal color settings… |
| 07:42.16 | kaldemar | fling: play when? for who? |
| 07:42.45 | kaldemar | razorgank: "sip set debug on" |
| 07:43.04 | kaldemar | razorgank: see where it gets sent. |
| 07:44.23 | fling | kaldemar: for ata phone |
| 07:44.48 | fling | kaldemar: when the called number is busy :p |
| 07:45.20 | razorgank | kaldemar: yup i'm using that, but i want to know up to which point in the route where my packets are getting blocked, is traceroute enough? force it to port 5060 and since by default it uses udp as probes? |
| 07:45.21 | kaldemar | fling: so someone dials using the ata, and you want the caller to hear a beep if the destination is busy? |
| 07:45.26 | fling | exten => kaldemar,1,Hello() |
| 07:45.34 | fling | kaldemar: yes |
| 07:46.09 | kaldemar | fling: inspect DIALSTATUS variable after your Dial app, and play a beep in case it has BUSY in it. |
| 07:46.12 | fling | I know, ata should playback it, not asterisk |
| 07:46.40 | fling | because we may send busy over sip, right? |
| 07:46.55 | fling | or is is simplier to playback beeps? |
| 07:47.08 | kaldemar | fling: yep. but you can of course answer and do what you want. |
| 07:47.10 | fling | s/is\ is/is\ it/ |
| 07:47.30 | fling | beeps |
| 07:47.37 | fling | kaldemar: thanks :] |
| 07:48.21 | kaldemar | razorgank: you can always try traceroute but the results are not necessarily the same. |
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| 07:51.31 | razorgank | kaldemar: i'm having a timeout problem since my transmission of the 200 ok response are not getting through, and i'm not getting an ACK from signalling, i also tried to use netcat but there's no response also. could it be a firewall or a routing problem? |
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| 07:52.19 | kaldemar | razorgank: it could be anything between you and the destination. your guess is just as good as mine. |
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| 07:54.05 | razorgank | kaldemar: thanks bro maybe the best way is to setup an asterisk on a laptop and go onsite to test |
| 07:54.07 | kaldemar | razorgank: check that you don't have insanely low values set in sip.conf for timert1 or timerb. |
| 07:54.36 | fling | kaldemar: where to get this beeping sound? is it in core sounds? |
| 07:55.23 | kaldemar | razorgank: those (or timerb if you have it set) affect how long asterisk waits for a response to an INVITE. |
| 07:55.37 | kaldemar | fling: what beeping are you looking for? a proper busy tone or something else? |
| 07:56.35 | razorgank | kaldemar: i can connect to asterisk if i'm using a sip client(from a pc) the problem was when the connection to a media gateway, so mobile phones could connect to my asterisk box, i can register the mgw, but the signalling is the problem |
| 07:56.39 | kaldemar | fling: Busy(), Playtones(busy), Playback(beep)... |
| 07:56.41 | fling | kaldemar: yes! busy tone |
| 07:56.51 | fling | thanks! |
| 07:57.11 | kaldemar | razorgank: are you sure a NAT is not a problem? |
| 07:58.05 | razorgank | kaldemar: i'm just new to nat, but we are using a private address, and am not using nat, would it matter? |
| 07:59.14 | fling | razorgank: noone is new to nat :P |
| 07:59.33 | kaldemar | razorgank: depends on the overall scenario. if there is a NAT between your asterisk and the gateway, it matters. |
| 08:02.30 | razorgank | kaldemar: my setup before was with nat, on sip.conf and defined my local and public ip, should i also define their ips in the sip.conf? |
| 08:03.16 | kaldemar | razorgank: you need to define their IP only if they don't register to you. |
| 08:03.50 | kaldemar | pastebin a CLI output with sip debug enabled for a call, it's quick to see if something is wrong in configs. |
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| 08:07.25 | razorgank | kaldemar: here's the logs from our last testing http://pastebin.com/26xMxpjU |
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| 08:11.52 | kaldemar | razorgank: looks like they are sending to you to a 10.128 address, and you're sending out a 125. address. |
| 08:25.02 | wdoekes | razorgank: you say you're not using nat, then why do you send out the external IP? remove externalip=/externalhost= from config |
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| 08:34.26 | PbxMan | morning |
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| 08:38.24 | razorgank | wdoekes: sorry that was an old log, my bad. i've removed the externalip from my sip.conf before, i'm looking for the current logs right now but for now i can only generate the last pcap from the last call http://i.imgur.com/sKFXFBB.jpg |
| 08:39.44 | wdoekes | razorgank: the image isn't really descriptive.. tcpdump -r my.pcap -vls0 |
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| 08:50.37 | srp_ | hi, I need some help in starpy. is this the right place to ask? |
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| 08:59.16 | razorgank | wdoekes: i can see now, in my sip headers when i'm transmitting the OPTIONS method, the public ip is defined in the from field, but it should be the private ip, how should i force asterisk to use the private ip? does it have something to do with nat? |
| 08:59.41 | Wiretap | razorgank: you need to tell asterisk what your localness are |
| 08:59.43 | Wiretap | localnets* |
| 09:00.58 | razorgank | Wiretap: thanks, should i set the sip entry of nat=yes for that particular mgw? |
| 09:01.32 | Wiretap | why would you want nat ON if there's no NAT? |
| 09:01.50 | Wiretap | NAT is only to be used when you are traversing NAT |
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| 09:10.15 | wdoekes | srp_: possibly. you can ask |
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| 09:14.00 | srp_ | wdoekes: I wanted to know if starpy supports asterisk-11.2.1 |
| 09:15.23 | wdoekes | why wouldn't it? make sure you take the github version though |
| 09:16.06 | srp_ | wdoekes: okay. just wanted to know as i found some -old python libs supporting only 1.8.. thanks anyway :) |
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| 09:22.58 | srp_ | wdoekes: I also would like to know if startpy allows me to create new extensions programatically via python. any ideas on that? |
| 09:23.44 | wdoekes | if nothing else, you can CLI-command your way into 'dialplan add ...' |
| 09:24.11 | wdoekes | but I'd prefer ARA for manipulating the dialplan |
| 09:25.54 | srp_ | wdoekes: i am trying to automate the process of creating extensions. As asterisk-gui or freepbx can create extensions on the web interface, i think it should be possible in starpy. |
| 09:26.55 | wdoekes | freepbx writes dialplan to static files |
| 09:28.16 | wdoekes | and the fact that freepbx does things, does not mean that an asterisk interface lib (starpy) should be able to. you don't expect starpy to include a web interface, do you? |
| 09:28.29 | srp_ | wdoekes: no no :) |
| 09:29.12 | srp_ | wdoekes: i'm building a rest api to fully control asterisk.. i am using starpy for that :) |
| 09:29.59 | wdoekes | srp_: starpy won't give you persistent storage. you'll need a file or db backend |
| 09:30.29 | wdoekes | points to ARA some more |
| 09:32.12 | srp_ | wdoekes: yes. thanks :) i wanted something that can create users on sip.conf / users.conf |
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| 11:47.47 | Greenlight | I need some sort of SIP proxy to generate CDR's so that I can bill customers. I'm currently using Asterisk for this, but it's not really idea. I've looked at OpenSIPs and Kamilio - would folks recommend one of them (or something else?)? |
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| 12:17.25 | leifmadsen | Greenlight: generally most people seem to head towards kamailio |
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| 12:29.55 | gavimobile | im considering purchasing a digium card with 1 fxo. I will connect my pots line to the fso for incomming calls only. if I get an incomming call on that line and someone else calls, will the asterisk make the line available? |
| 12:30.26 | gavimobile | I don't think this has anything to do with outgoing calls but I use rapidvox which has unlimited channels for outgoing calls |
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| 12:32.42 | kaldemar | gavimobile: 1 FXO => one call. |
| 12:33.17 | Greenlight | leifmadsen: Thanks, I'll check it out |
| 12:33.34 | gavimobile | kaldemar: can't asterisk move the call to another channel an open that line somehow? |
| 12:33.41 | srp_ | I am using sipML 5 as a client to connect to asterisk 11.2.1. If I try to register with asterisk in the same machine, it gets registered. If I try to register from other system to the same asterisk server, I get this error 'Disconnected: Failed to connet to the server'. Please help |
| 12:38.53 | kaldemar | gavimobile: you don't have another channel if you only have one FXO. |
| 12:38.59 | leifmadsen | gavimobile: there is no other open line... it's a physical connection |
| 12:39.13 | gavimobile | I see |
| 12:39.37 | leifmadsen | you don't magically get more call capacity on the same line because asterisk is attached to it :) |
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| 12:40.19 | Greenlight | Some lines support "hold", whereby you can be notified of a call waiting, and place call 1 on hold, and answer call 2. All on a single analogue line - do you mean something like that/ |
| 12:40.19 | gavimobile | my goal is to leave my current istp, so if I scratch out that idea about getting a card and I transfer my did to another voip (somehow), I just need to request a few channels ie.5 to be allowed to receive 5 concurrent incomming calls |
| 12:40.49 | gavimobile | Greenlight: that solutin won't work, because I need more than 2 concurrent calls |
| 12:40.53 | leifmadsen | Greenlight: fyi, that's called call waiting |
| 12:41.15 | leifmadsen | gavimobile: well a physical line isn't going to be what you want then |
| 12:41.28 | leifmadsen | unless you get multiple phone lines, you won't be able to handle multiple calls |
| 12:41.32 | gavimobile | unless if I can get 2 analog lines and place 2 people on hold, but I would need the people on hold to hear the moh from the queue. is that possible? |
| 12:41.40 | leifmadsen | no |
| 12:41.43 | Greenlight | I always thought Call Waiting was just the notification, never realised swapping the calls was covered by that description as well |
| 12:42.10 | gavimobile | ok, so its clear that I should stick with voip for incomming calls to allow me to have this flexability |
| 12:42.19 | leifmadsen | when you hook-flash after a call waiting notification, the phone company is holding the line for you, not your pbx, so you can't send them audio |
| 12:42.31 | Greenlight | Yea, indeed |
| 12:42.37 | gavimobile | leifmadsen: thanks for that clarification |
| 12:43.05 | gavimobile | I hate my istp with a passion |
| 12:43.14 | Greenlight | There's plenty more out there |
| 12:43.19 | gavimobile | but im my country I don't have much choices |
| 12:43.32 | gavimobile | plus I want to take advantake of local 1800 calls, emergency calls etc |
| 12:43.50 | gavimobile | a benefit of the analog line will allow me to make *NXXX calls |
| 12:44.26 | gavimobile | its not worth buying a card so I can make *NXXX calls |
| 12:46.50 | *** join/#asterisk CRCinAU_ (~CRCinAU@another.bloody.irc.session.from.crc.id.au) |
| 12:46.53 | leifmadsen | use an ATA then... ? |
| 12:47.20 | CRCinAU_ | I'm wondering if anyone has seen this on asterisk 11.2.1: res_fax.c:1585 receivefax_t38_init: error generating CED tone |
| 12:48.05 | gavimobile | leifmadsen: like a sipura/cisco box? |
| 12:48.17 | leifmadsen | like an ATA :) |
| 12:48.25 | gavimobile | how will that help me? |
| 12:48.30 | CRCinAU_ | the only reference I've found in google for this seems to be links to pastebins :( |
| 12:48.33 | gavimobile | technically |
| 12:49.34 | CRCinAU_ | The full messages I'm seeing are here: http://pastebin.com/T7ydEkAw |
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| 12:58.01 | mjordan | CRCinAU_: your problem starts here: [Jan 29 23:40:39] WARNING[3133][C-00000005]: indications.c:157 playtones_alloc: Unable to set 'SIP/spin-00000005' to signed linear format (write) |
| 12:58.18 | mjordan | CRCinAU_: that implies that we couldn't switch the format of the SIP channel to signed linear |
| 13:00.08 | coppice | well, switching a T.38 channel to signed linear is really hard |
| 13:00.15 | CRCinAU_ | I'm not sure what this has to do with a system that only enables alaw, ulaw and 729 though? :\ |
| 13:00.33 | mjordan | it just needs a translation path |
| 13:00.38 | CRCinAU_ | it should it be doing that in the first place? |
| 13:00.58 | mjordan | yes, it needs to do that. |
| 13:01.14 | CRCinAU_ | signed linear = slin right? |
| 13:01.17 | mjordan | yes |
| 13:01.46 | CRCinAU_ | I can see a path in a 'core show translation' |
| 13:01.56 | *** join/#asterisk n3hxs (~ed@pool-108-16-94-10.phlapa.fios.verizon.net) |
| 13:01.57 | CRCinAU_ | it shows a figure for alaw -> slin etc etc |
| 13:02.09 | coppice | you can see a patch from slin to T.38? :-\ |
| 13:02.17 | coppice | s/patch/path |
| 13:02.31 | CRCinAU_ | ie: |
| 13:02.33 | CRCinAU_ | asterisk*CLI> core show translation paths slin |
| 13:02.34 | CRCinAU_ | --- Translation paths SRC Codec "slin" sample rate 8000 --- |
| 13:02.34 | CRCinAU_ | <PROTECTED> |
| 13:02.34 | CRCinAU_ | <PROTECTED> |
| 13:02.34 | CRCinAU_ | <PROTECTED> |
| 13:02.36 | CRCinAU_ | <PROTECTED> |
| 13:02.43 | CRCinAU_ | t38 isnt' mentioned there at all |
| 13:02.51 | CRCinAU_ | nor would I really expect it to? |
| 13:02.51 | mjordan | t38 isn't a format |
| 13:03.02 | mjordan | not in this sense. |
| 13:03.04 | coppice | of source not, but your call seems to be in T.38 mode |
| 13:03.12 | coppice | s/source/course |
| 13:03.20 | mjordan | also, that's slin to alaw, not alaw to slin |
| 13:03.36 | mjordan | what is the output of core show translation path alaw |
| 13:03.57 | CRCinAU_ | asterisk*CLI> core show translation paths alaw |
| 13:03.58 | CRCinAU_ | --- Translation paths SRC Codec "alaw" sample rate 8000 --- |
| 13:04.04 | CRCinAU_ | <snipped a bit> |
| 13:04.08 | CRCinAU_ | <PROTECTED> |
| 13:04.11 | mjordan | kk |
| 13:04.32 | mjordan | not sure why then, other than your log shows an error setting the write format. A DEBUG log would probably show more |
| 13:05.41 | CRCinAU_ | hmmm - let me looksee.... |
| 13:05.58 | CRCinAU_ | just via a 'core set debug xx'? |
| 13:06.03 | CRCinAU_ | or something specific? |
| 13:06.21 | CRCinAU_ | I'll admit, my knowledge of the internals of asterisk is pretty minimal.... |
| 13:07.45 | mjordan | CRCinAU_: you should set 'debug' as one of the logger levels going to a file (otherwise you'll flood your CLI) |
| 13:07.51 | mjordan | in logger.conf |
| 13:07.51 | *** join/#asterisk TarCert (c1a9b80e@gateway/web/freenode/ip.193.169.184.14) |
| 13:07.54 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.197.215) |
| 13:08.10 | CRCinAU_ | just sending a test fax from my machine to test. |
| 13:08.16 | CRCinAU_ | PSTN -> SIP provider for fax. |
| 13:08.28 | CRCinAU_ | ok - same deal |
| 13:08.44 | CRCinAU_ | it doesn't even generate the fax tone on the asterisk sie |
| 13:08.45 | CRCinAU_ | side even |
| 13:09.13 | CRCinAU_ | same output though -> |
| 13:09.14 | CRCinAU_ | [Jan 30 00:08:05] WARNING[3220][C-00000000]: indications.c:157 playtones_alloc: Unable to set 'SIP/spin-00000000' to signed linear format (write) |
| 13:09.17 | CRCinAU_ | [Jan 30 00:08:05] ERROR[3220][C-00000000]: res_fax.c:1585 receivefax_t38_init: error generating CED tone on SIP/spin-00000000 |
| 13:09.20 | CRCinAU_ | [Jan 30 00:08:05] ERROR[3220][C-00000000]: res_fax.c:1926 receivefax_exec: error initializing channel 'SIP/spin-00000000' in T.38 mode |
| 13:09.42 | CRCinAU_ | then it hangs up |
| 13:09.53 | CRCinAU_ | so I get a busy tone back on the PSTN |
| 13:10.30 | mjordan | CRCinAU_: without a debug log showing what happened when it tried to set the write format on the SIP channel, there isn't much I can tell you |
| 13:10.35 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 13:10.44 | CRCinAU_ | that is with: core set debug 50 |
| 13:11.00 | mjordan | nope :-) |
| 13:11.08 | CRCinAU_ | asterisk*CLI> core set debug 50 |
| 13:11.09 | CRCinAU_ | Core debug is at least 50 |
| 13:11.17 | mjordan | I believe you, but DEBUG isn't being sent to the CLI |
| 13:11.23 | mjordan | you have to configure that in logger.conf |
| 13:11.25 | CRCinAU_ | oh. |
| 13:11.30 | mjordan | and I'd recommend that you don't send it to the CLI |
| 13:11.34 | CRCinAU_ | I thought it defaulted to CLI. |
| 13:11.37 | mjordan | nope |
| 13:12.16 | CRCinAU_ | is this sufficient: debug => debug |
| 13:12.18 | CRCinAU_ | ? |
| 13:12.20 | mjordan | no |
| 13:12.28 | mjordan | you need all of them - a debug log should contain all message levels |
| 13:12.37 | CRCinAU_ | hmmm ok |
| 13:12.38 | mjordan | debug being one of them. Use the example of 'full' in the logger.conf |
| 13:12.45 | CRCinAU_ | let me tinker for a little |
| 13:12.53 | mjordan | I'd do this: generate a debug log showing the fax attempt. Get a pcap as well. Post an issue on the issue tracker, including your sip.conf and extensions.conf |
| 13:13.06 | mjordan | I'm not sure this is a bug, but it feels odd that it can't set the write format |
| 13:13.18 | mjordan | and it'll take some info to figure out why it's failing |
| 13:13.33 | *** join/#asterisk peektoseen (~peektosee@193.105.235.1) |
| 13:14.08 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-hdmisgiurexixeqp) |
| 13:14.10 | CRCinAU_ | oh. got it going... |
| 13:14.13 | CRCinAU_ | let me run a fax. |
| 13:20.22 | CRCinAU_ | ok |
| 13:21.41 | CRCinAU_ | I've got the debug log down to ~20l |
| 13:21.41 | CRCinAU_ | err 20k |
| 13:21.41 | CRCinAU_ | stuff I see as interesting: |
| 13:21.41 | CRCinAU_ | [Jan 30 00:14:49] DEBUG[3199][C-00000002] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. |
| 13:21.41 | CRCinAU_ | If I understand right, this means t.38? |
| 13:22.04 | CRCinAU_ | [Jan 30 00:14:49] DEBUG[3199][C-00000002] chan_sip.c: We're settling with these formats: (ulaw|alaw|g729) |
| 13:22.22 | CRCinAU_ | thats what I have set in my sip.conf to only allow ulaw/alaw/g729 |
| 13:22.54 | CRCinAU_ | [Jan 30 00:14:49] DEBUG[3199][C-00000002] chan_sip.c: *** Joint capabilities are (ulaw|alaw|g729) |
| 13:22.57 | CRCinAU_ | [Jan 30 00:14:49] DEBUG[3199][C-00000002] chan_sip.c: *** Our capabilities are (ulaw|alaw|g729) |
| 13:23.00 | CRCinAU_ | [Jan 30 00:14:49] DEBUG[3199][C-00000002] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw |
| 13:23.03 | CRCinAU_ | thats good. |
| 13:23.04 | CRCinAU_ | we settled on alaw.... |
| 13:23.39 | CRCinAU_ | [Jan 30 00:14:50] DEBUG[3246][C-00000002] chan_sip.c: SIP answering channel: SIP/spin-00000002 |
| 13:23.44 | CRCinAU_ | it answered the call. |
| 13:24.16 | CRCinAU_ | [Jan 30 00:14:50] DEBUG[3246][C-00000002] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|alaw|g729) |
| 13:24.25 | CRCinAU_ | still likes ulaw/alaw/g729 |
| 13:24.59 | CRCinAU_ | but then this happens: |
| 13:25.00 | CRCinAU_ | [Jan 30 00:14:50] DEBUG[3246][C-00000002] chan_sip.c: Oooh, format changed to g729 |
| 13:25.22 | leifmadsen | CRCinAU_: please don't flood the channel |
| 13:25.24 | leifmadsen | ~pb |
| 13:25.24 | 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. |
| 13:25.50 | CRCinAU_ | 20kb to pastebin? :P |
| 13:26.01 | leifmadsen | it's more the number of lines |
| 13:26.02 | CRCinAU_ | I'm trying to cherry pick :) |
| 13:26.11 | leifmadsen | more than 3 lines == pastebin |
| 13:26.26 | CRCinAU_ | wait whut. |
| 13:26.36 | leifmadsen | that's general irc etiquette |
| 13:26.47 | CRCinAU_ | it seems that asterisk settles on alaw via AST_CODEC_CHOOSE |
| 13:26.55 | CRCinAU_ | but then its saying the codec changed to g729 |
| 13:27.01 | file | because the remote side sent g729. |
| 13:27.30 | CRCinAU_ | then it says the fax stuff doesn't support g729 - for obvious reasons |
| 13:27.46 | CRCinAU_ | then it complains that it can't find a translation from g729 -> slin |
| 13:28.30 | CRCinAU_ | where does it even get slin from? o_O |
| 13:28.40 | leifmadsen | signed linear is internal |
| 13:28.50 | CRCinAU_ | Hmmmm |
| 13:29.34 | CRCinAU_ | it seems we go: alaw -> g729 (can't find g729 -> alaw) -> (can't find g729 to slin) |
| 13:29.48 | file | it's not. |
| 13:30.01 | file | the SDP negotiation resulted in 3 formats being permitted, ulaw alaw and g729 |
| 13:30.07 | file | each side is free to use any of those at any time |
| 13:30.11 | CRCinAU_ | yeah - and I see: |
| 13:30.12 | CRCinAU_ | [Jan 30 00:14:50] DEBUG[3246][C-00000002] chan_sip.c: Oooh, format changed to g729 |
| 13:30.13 | file | the remote side that is sending you media decided it wanted to use g729 |
| 13:30.30 | CRCinAU_ | then the warning: [Jan 30 00:14:50] WARNING[3246][C-00000002] channel.c: Unable to find a codec translation path from (g729) to (alaw) |
| 13:30.53 | CRCinAU_ | then this: [Jan 30 00:14:52] WARNING[3246][C-00000002] channel.c: Unable to find a codec translation path from (g729) to (slin) |
| 13:31.00 | file | right. |
| 13:31.08 | file | the remote side ended up using g729, and Asterisk switched |
| 13:31.35 | CRCinAU_ | hmmmmm. |
| 13:31.50 | CRCinAU_ | asterisk*CLI> core show translation paths g729 |
| 13:31.50 | CRCinAU_ | --- Translation paths SRC Codec "g729" sample rate 8000 --- |
| 13:31.56 | CRCinAU_ | <PROTECTED> |
| 13:32.15 | file | g729 is a licensed codec, it's not included with Asterisk |
| 13:32.24 | CRCinAU_ | so. I guess I can try disabling g729.... |
| 13:32.32 | file | yes. |
| 13:32.36 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 13:32.47 | CRCinAU_ | although wouldn't it make sense that asterisk then says "Whoops. I can't do that" and reinvite back to the second choice? |
| 13:33.16 | file | it's hard to know that. |
| 13:33.30 | CRCinAU_ | Hmmm. |
| 13:33.32 | file | Asterisk can pass through that media happily, and if your audio files are transcoded to g729 it can play back files |
| 13:33.36 | file | same goes for moh |
| 13:34.01 | CRCinAU_ | I've actually been using 'codec_g729-ast110-gcc4-glibc-pentium4.so' for quite some time. |
| 13:34.10 | *** join/#asterisk deo (~deo@112.198.90.178) |
| 13:34.18 | file | no comment. |
| 13:34.19 | *** join/#asterisk hwt (hwt@curb.thorkildssen.com) |
| 13:34.24 | CRCinAU_ | I wonder why it isn't appearing in the translation path how it used ot. |
| 13:34.31 | hwt | hey, is there a way to get asterisk to reply with a 403 from within the dialplan? |
| 13:34.44 | CRCinAU_ | well, I have bought g729 licenses in the past.... however now Digium refuse to relicense them to me without paying again |
| 13:34.53 | CRCinAU_ | apparently because I changed my hardware too many times for my asterisk server. |
| 13:35.14 | file | hwt, Hangup(21) |
| 13:35.15 | leifmadsen | usually you just need to call and ask for the license to be reset |
| 13:35.38 | file | that's only permitted up to a certain point |
| 13:35.45 | CRCinAU_ | leifmadsen: yeah - I emailed a while ago - then they refused to do it. |
| 13:36.06 | CRCinAU_ | it got to a point where each reboot, due to stupidities in the licensing, it would render the license invalid |
| 13:36.08 | hwt | file: excellent, thanks |
| 13:36.19 | CRCinAU_ | it screwed with the MACs of the ethernet adapters |
| 13:36.32 | CRCinAU_ | and that caused it to fail each time. |
| 13:36.53 | CRCinAU_ | so after a few months of this happening on and off, they just refused to reset the license anymore |
| 13:37.03 | CRCinAU_ | now I don't really care and use other methods. |
| 13:38.15 | CRCinAU_ | this was in the early days of udev though... so it gave me lots of issues with ethernet adapters at the time |
| 13:38.27 | CRCinAU_ | ie eth0 and eth1 would randomly swap on boot. |
| 13:38.35 | CRCinAU_ | then boom, licenses invalid |
| 13:38.38 | leifmadsen | I like stories. |
| 13:38.45 | CRCinAU_ | reboot again, invalid again. |
| 13:39.05 | CRCinAU_ | me too... ;) |
| 13:39.24 | file | how long ago was that? |
| 13:40.29 | CRCinAU_ | probably around asterisk 1.2 days? maybe 1.4? |
| 13:40.37 | file | ah |
| 13:40.41 | CRCinAU_ | a long time ago :) |
| 13:41.13 | file | years ago I believe the licensing was tweaked to fix that issue... that's so long ago I can't remember |
| 13:41.16 | CRCinAU_ | The reason I got from Digium was "Sorry, we can't be sure you're not pirating the licenses. denied" |
| 13:41.34 | file | aye |
| 13:41.51 | CRCinAU_ | yeah - I can't remember exactly either.... but needless to say, I just worked around it and figured it was worth it for less hassle. |
| 13:42.39 | CRCinAU_ | hahaha |
| 13:42.41 | CRCinAU_ | <PROTECTED> |
| 13:42.43 | CRCinAU_ | whoops. |
| 13:43.03 | CRCinAU_ | durr on my part |
| 13:44.48 | CRCinAU_ | heeey... thats more like it.... |
| 13:45.09 | CRCinAU_ | now to see what happens on a fax again... it should probably kick out g729 in favour of alaw when ReceiveFax kicks in I'd assume. |
| 13:45.38 | CRCinAU_ | otherwise, I'll just nail g729 out in somewhere that may/may not affect normal outbound calls on that peer.... |
| 13:46.52 | CRCinAU_ | bingo. |
| 13:47.10 | CRCinAU_ | [Jan 30 00:46:24] NOTICE[3276][C-00000003]: channel.c:4251 __ast_read: Dropping incompatible voice frame on SIP/spin-00000003 of format alaw since our native format has changed to (g729) |
| 13:47.20 | CRCinAU_ | [Jan 30 00:46:32] WARNING[3276][C-00000003]: res_fax.c:1669 receivefax_t38_init: channel 'SIP/spin-00000003' timed-out during the T.38 negotiation. |
| 13:47.24 | CRCinAU_ | then it goes back to alaw. |
| 13:47.45 | CRCinAU_ | then starts its T.30 negotiation. |
| 13:48.39 | CRCinAU_ | Thanks for the punt in the right direction... |
| 13:49.36 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 13:49.38 | CRCinAU_ | I wouldn't have expected that a missing translation to/from g729 would cause it to fail... especially as I would only think that alaw would have been selected |
| 13:50.18 | CRCinAU_ | another thought.... Can I change the available codecs in the dialplan? |
| 13:50.51 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 13:50.52 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:52.01 | kaldemar | CRCinAU_: set SIP_CODEC_INBOUND or SIP_CODEC |
| 13:52.04 | CRCinAU_ | as I use the same peer for outgoing calls - but only incoming faxes... I'd like to allow g729 to be used for outgoing calls, but restrict incoming to alaw... |
| 13:52.18 | CRCinAU_ | kaldemar: awesome. I'll google that :) |
| 13:52.22 | kaldemar | for outgoing there is SIP_CODEC_OUTBOUND. |
| 13:52.47 | CRCinAU_ | the only restriction being I have to do it before an Answer on incoming? |
| 13:53.12 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.197.215) |
| 13:53.36 | kaldemar | CRCinAU_: yes. |
| 13:53.48 | CRCinAU_ | that makes perfect sense. |
| 13:55.25 | *** join/#asterisk serafie (~erin@nat/digium/x-qedhgjbexwlhtcyv) |
| 13:55.25 | CRCinAU_ | Theres no examples on the web page, so I assume its set like: exten => s,n,Set(${SIP_CODEC_INBOUND}=alaw) |
| 13:55.51 | kaldemar | your usage of Set is incorrect. |
| 13:55.59 | CRCinAU_ | oh? |
| 13:56.12 | kaldemar | the variable. |
| 13:56.44 | CRCinAU_ | oh - of course. |
| 13:57.06 | CRCinAU_ | hahah that'd confuse things... Set(g729=alaw) or similar ;) |
| 13:57.10 | CRCinAU_ | facepalms |
| 13:57.49 | CRCinAU_ | probably shouldn't be screwing with this at 1am after a few beers... but meh ;) |
| 13:58.10 | *** join/#asterisk bchia (~Adium@micc-208-115-78-243.smartcity.com) |
| 13:59.08 | CRCinAU_ | I just get to make myself look like a complete n00b :) |
| 13:59.08 | CRCinAU_ | and I keep getting about 20 emails every 5 minutes saying people are destroying my Ingress resonators and links. Grrr. |
| 14:03.56 | CRCinAU_ | Hmmm. |
| 14:03.56 | CRCinAU_ | now its failing again after adding the SIP_CODEC_INBOUND=alaw. |
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| 14:29.18 | carrar | *Y*A*W*N* |
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| 15:42.59 | rue_house | can anyone help me with a mgcp problem? my second gateway wont work |
| 15:48.22 | [TK]D-Fender | rue_house, Almost nobody uses MGCP, let alone those that frequent #asterisk. You should be asking thins on the mailing list.... |
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| 16:10.31 | Weezey | Hey @leifmadsen, (or anyone), I'm setting up a new system using realtime, is there any way without registration to have peers qualify so it shows something other than UNKNOWN in sip show peers ? |
| 16:10.47 | leifmadsen | qualify=no |
| 16:11.01 | leifmadsen | otherwise, the peers have to respond to the OPTIONs requests |
| 16:11.14 | Weezey | cool |
| 16:11.26 | leifmadsen | you'll need host=<something specific> other than host=dynamic or else the peers have to register |
| 16:13.07 | Weezey | K, I did that now, I'm getting "Unmonitored", is there something else to show them or is it just impossible with Realtime? |
| 16:18.11 | rogers | Seem to be having a problem with internal calls - when you call an extension, it will ring and then go to voicemail, however the phone keeps ringing and ringing. What might be the issue?> |
| 16:18.45 | WIMPy | Networking issue or bad phone. |
| 16:19.42 | rogers | wimpy what networking issue? happening to all phones |
| 16:21.15 | Weezey | is there a router/firewall between asterisk and the phones? |
| 16:21.43 | igcewieling | "customer IT vendor: I don't know what you are talking about a firewall, I only have a router doing NAT" |
| 16:23.39 | Weezey | liefmadsen, I don't get it though, if I use sip.conf, qualify=yes and I see ms times for the peers but if I do realtime qualify=yes gives me UNKNOWN and qualify=no gives me Unmonitored. If I'm stuck with that, I am, but if there's a way to fix I'd like to as it's an easy way to identify an issue. |
| 16:23.57 | Weezey | also I can't spell leifmadsen |
| 16:23.57 | *** join/#asterisk sewellj (~chatzilla@2607:0:3:1::4) |
| 16:24.33 | leifmadsen | qualify does work with realtime peers after they register |
| 16:24.40 | leifmadsen | we use it all the time |
| 16:24.40 | *** join/#asterisk mountainm2k (~msturtz@gw.booyahnetworks.com) |
| 16:26.39 | rogers | yeah 2 locations, there is a VPN between them |
| 16:26.51 | rogers | ipsec vpn netgear and cisco |
| 16:27.03 | mountainm2k | Hi: I have a bunch of extensions that "forward" to off-site phone numbers. For these calls, I don't care if I never get them back, IE, I don't need any ring-no-answer, etc. But I would like to "transfer" them with my carrier, and get them completely off my PRI, instead of tieing up two lines. |
| 16:27.38 | [TK]D-Fender | mountainm2k, "transfer=yes" <- chan_dahdi.conf |
| 16:27.39 | mountainm2k | I know the virtual office companies that you hire to screen your calls can do this -- but how? |
| 16:27.48 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 16:28.04 | saint_ | greetings all |
| 16:28.22 | saint_ | can someone tell me why chan_sip.so would not load ? |
| 16:28.39 | saint_ | I am installing asterisk on a brand new box. Followed the asterisk book from asteriskdocs.org ... |
| 16:28.56 | saint_ | created my sip.conf , but when i start asterisk, it will not load |
| 16:29.05 | igcewieling | leifmadsen: If I understand you correctly, qualify only works with Realtime if the peers register, but not if they do not register (i.e. static IP addresses)? |
| 16:29.05 | saint_ | in the make menuconfig, i can see that chan_sip is checked |
| 16:29.13 | mountainm2k | Ah, OK. Once I do that, anything else? Right now they are forwarding via Dial |
| 16:29.30 | *** part/#asterisk Linkforsoad (~Linkforso@2001:1af8:fec1:0:fd54:886b:9e6d:3235) |
| 16:29.30 | leifmadsen | igcewieling: that's not what I said |
| 16:29.30 | igcewieling | saint_: "module load chan_sip.so" in the CLI, look for errors. |
| 16:29.35 | mountainm2k | Do I need to call a different function in order to Transfer off-net? |
| 16:29.40 | saint_ | and chan_sip.so is present in /usr/lib/asterisk/modules/ |
| 16:29.58 | [TK]D-Fender | mountainm2k, that's fine as long as you aren't using any dial options that force * to stay in the middle |
| 16:30.02 | WIMPy | mountainm2k: Not after answering the call. |
| 16:30.03 | igcewieling | leifmadsen: cool. I was concerned there for a moment. |
| 16:30.11 | leifmadsen | <leifmadsen> you'll need host=<something specific> other than host=dynamic or else the peers have to register |
| 16:30.30 | [TK]D-Fender | mountainm2k, And of course your carrier has to support this |
| 16:30.36 | mountainm2k | Yeah, i don't answer() in this case -- in my dialplan, I call a macro that just does the Dial |
| 16:30.39 | igcewieling | leifmadsen: I need a bigger window, that had scrolled off. thanks. |
| 16:30.52 | [TK]D-Fender | mountainm2k, Doesn't matter if you answer |
| 16:30.57 | mountainm2k | Yeah, well, if they don't I'll bug them to fix it. :-) It's Level3, I gotta believe they will. |
| 16:31.09 | [TK]D-Fender | mountainm2k, "transfer=yes" will automatically 2BCT on bridging |
| 16:31.12 | [TK]D-Fender | ~2bct |
| 16:31.12 | infobot | [~2BCT] 2BCT (2 B-Channel Transfer) allows a call coming in over DAHDi and back out again to the same telco to be handed off freeing the channels from your circuit. To enable this (if your carrier supports it) add "transfer=yes" to your channel configurations. |
| 16:31.14 | mountainm2k | Ah, OK -- so just Dial with no other options? |
| 16:31.19 | WIMPy | mountainm2k: then you could ue Transfer() as well. |
| 16:31.24 | saint_ | igcewieling: that worked. the book says "module reload chan_sip.so" .... by s/reload/load , it worked. |
| 16:31.25 | [TK]D-Fender | mountainm2k, Nothing that would interfere |
| 16:31.32 | [TK]D-Fender | mountainm2k, like those requiring DTMF |
| 16:31.37 | mountainm2k | How will I know if its working? |
| 16:31.39 | igcewieling | mountainm2k: are you a direct Level 3 customer? If so be VERY careful, L3 will bill the call to the trunk TN if they don't recognize the callerid as a TN on your account. |
| 16:31.45 | mountainm2k | Yeah, I don't have any digitmaps on this |
| 16:31.51 | saint_ | igcewieling: thanks for the trick. now my issue is that if I stop asterisk and start it again, the module is not loaded automatically. is there anything to do anywhere for that ? |
| 16:31.54 | igcewieling | saint_: look in /etc/asterisk/modules.conf for a noload => chan_sip.so |
| 16:32.07 | rogers | weezy/wimpy what might be the issue with VPN between? |
| 16:32.12 | WIMPy | mountainm2k: By the call ariving at the destination. |
| 16:32.43 | mountainm2k | igcewieling: Yes, I'm a direct Level3 customer... Not sure what you mean though -- in this case they are all local calls anyway, so the billing doesn't matter |
| 16:32.51 | saint_ | igcewieling: nah, it s not in there .. |
| 16:33.18 | saint_ | igcewieling: drives me nuts, because the 1st installation that i did and which is in prod now has no issues with that .. |
| 16:33.21 | Weezey | leifmadsen: yeah, me too, I'm trying to set it up for things that don't register though. which in sip.conf works, but realtime doesn't show me their last ms. |
| 16:33.51 | saint_ | igcewieling: i m so not in the mood to start again from scratch with this one ... |
| 16:34.12 | igcewieling | saint_: stop asterisk. start asterisk as "asterisk -cvvv" dig through the output until you see why chan_sip.so is not loading |
| 16:35.49 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 16:35.55 | igcewieling | mountainm2k: We were getting billed for calls by Level 3 which we had no record of. Turns out in the billing CDRs we get from Level 3 the ANI was set to the "default billing TN" on the account when we sent out calls with an ANI which was not on our level 3 account. They do NOT support Diversion: headers like VZB does. |
| 16:36.05 | mountainm2k | [TK]: Can I safely assume that transfer=yes would go in zapata.conf -- since that's what I'm running? |
| 16:36.29 | mountainm2k | Actually, I have that option in all my channels *except* the PRI... Which is strange, because the other ZAP channels are analog... |
| 16:36.50 | WIMPy | mountainm2k: You should go and upgrade to a recent version before doing anything else. |
| 16:37.04 | saint_ | igcewieling: thanks for the int . it said "modules.conf loaded" but just after it said it had an error on line 1. Line 1 started with a . instead of a ; for my comment. |
| 16:37.11 | saint_ | igcewieling: i fixed that, and it's working.... uha ! |
| 16:37.25 | mountainm2k | Hah -- don't I know it... I have a whole new server sitting here to replace this one (6+ years old), and haven't had time to work on it... |
| 16:37.43 | mountainm2k | new T1 card and everything... |
| 16:38.47 | Weezey | rogers: missing packets that tell the phones to stop ringing. |
| 16:39.19 | rogers | Weezey, is there a name for that packet? |
| 16:39.50 | [TK]D-Fender | mountainm2k, Yes |
| 16:40.14 | mountainm2k | igcewieling: In my case, I don't think I care, because these forwards would be billed to my account in any case, if they were a chargable call... |
| 16:40.40 | igcewieling | mountainm2k: *nod* in this specific case, but you should know for the future |
| 16:40.43 | mountainm2k | How can I tell if its actually working, will the debug log say "transfer off net accepted" or something normal like that? |
| 16:40.59 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:41.24 | WIMPy | If it doesn't work, the call will most probably drop. |
| 16:41.27 | mountainm2k | igcewieling: Do you have certain outbound calls billed to seperate accounts on the same system? |
| 16:42.26 | igcewieling | mountainm2k: We take the CDRs and bill the end customer, they are all billed to our master account. |
| 16:42.58 | igcewieling | But when the CDRs we get back from Level 3 has a TN which is not assigned to any customer, it makes it hard to bill the call back to the end user. |
| 16:45.44 | [TK]D-Fender | mountainm2k, You'll see the channels hang up. |
| 16:45.57 | [TK]D-Fender | mountainm2k, Perhaps core debug would say more |
| 16:47.09 | Weezey | rogers, there's a constant flow of UDP packets between your asterisk box and your SIP phones, if there's a VPN between them, you have to be sure it's not eating any of them. |
| 16:47.30 | Weezey | rogers, if you answer the phone can you hear the callers? can they hear you? |
| 16:48.16 | rogers | yeah calls work fine |
| 16:48.23 | rogers | seems to be an issue with voicemail only |
| 16:48.30 | rogers | also the vocemail notification light keeps blinking |
| 16:48.35 | rogers | even though theres no voicemails |
| 16:48.43 | rogers | eventually it will go away after undetermined amount of time |
| 16:52.12 | *** join/#asterisk timahvo1 (~rogue@41.81.125.120) |
| 17:02.02 | rogers | ideas? |
| 17:05.28 | *** join/#asterisk phyrexianslug (~phyrexian@S0106000c29543ac5.wp.shawcable.net) |
| 17:06.01 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:07.57 | rogers | wow looks like when I call out, and the other end hangs up, my side still thinks its connected |
| 17:08.10 | phyrexianslug | trunk type? |
| 17:09.04 | rogers | SIP |
| 17:09.31 | rogers | I think its an internal issue phyrexianslug, im having problems with voicemail notifications, and ringing when it has gone to voicemail |
| 17:11.31 | phyrexianslug | tried sip debug yet? |
| 17:12.12 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 17:12.47 | phyrexianslug | your provider should send you a hangup when their sip->pots (or whatever tech they use) |
| 17:13.44 | phyrexianslug | http://www.gl.com/images/maps-sip-web-sip-call-flow.gif you should get a "bye" from them at hangup |
| 17:14.04 | phyrexianslug | if it's not arriving, or the upstream router is doing anything funky, you might not get it |
| 17:14.55 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.48.38) |
| 17:18.38 | rogers | yeah I think some new firewall rules are causing it (vpn between phones and asterisk box) |
| 17:18.58 | rogers | I just forced nat off on the asterisk side for the two extensions, and the call worked anyway |
| 17:19.05 | rogers | still no voicemail notification working |
| 17:20.13 | igcewieling | rogers: remember your localnet settings should include both local and remote networks which are on the other end of the VPN (assuming there is no nat involved between your local and remote networks) |
| 17:20.51 | rogers | yeah thats gotta be it! |
| 17:20.59 | rogers | I dont have the remote side in the local networks |
| 17:21.03 | rogers | can I add multiple? |
| 17:23.03 | *** join/#asterisk oej_ (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 17:25.07 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:25.07 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:27.36 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.173.160) |
| 17:27.40 | igcewieling | rogers: yes. |
| 17:28.47 | rogers | juat add multiple localnet= lines? |
| 17:29.18 | pabelanger | Hmm |
| 17:29.26 | [TK]D-Fender | rogers, Yes |
| 17:29.32 | pabelanger | just started getting: PRI got event: HDLC Abort (6) on D-channel on channel 1 for a site |
| 17:31.51 | rogers | thanks guys, that was the problem |
| 17:31.57 | igcewieling | pabelanger: Asterisk detected an error on your PRI. This could be caused by an issue with your PRI card or (more likely) an error on the line. |
| 17:32.24 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1004) |
| 17:33.42 | *** join/#asterisk DoSJustin (~justin@vpn.bctconsulting.com) |
| 17:34.58 | pabelanger | igcewieling, Yup, been down this path before will Bell. Seems every few months, something fcuks up with this site |
| 17:36.23 | pabelanger | Great, getting CRC errors from the line |
| 17:36.55 | WIMPy | pabelanger: Send them a bottle of K60 :-) |
| 17:37.34 | pabelanger | http://pastebin.com/bZWiH9G9 |
| 17:37.36 | pabelanger | grr |
| 17:53.03 | *** join/#asterisk nacho2k (~Thunderbi@r190-64-14-98.ir-static.anteldata.net.uy) |
| 17:55.21 | nacho2k | hi guys!, a long time ago, I used to buy linksys pap2 for fxo, could anybody tell me what should I buy for getting similar devices? |
| 17:55.52 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:59.04 | navaismo | uhmm pap2t is for fxs right? |
| 18:00.24 | [TK]D-Fender | yes |
| 18:00.41 | [TK]D-Fender | nacho2k, How many ports? |
| 18:03.09 | ghost75 | pap2t is old btw |
| 18:03.12 | nacho2k | [TK]D-Fender, sorry I mean fxs, I need 2 ports |
| 18:03.34 | [TK]D-Fender | nacho2k, jsut for regular phones? |
| 18:03.41 | nacho2k | [TK]D-Fender, yes |
| 18:03.54 | [TK]D-Fender | nacho2k, Then the PAP2T-NA is still just fine |
| 18:04.07 | ghost75 | if you dont need t.38 |
| 18:05.52 | [TK]D-Fender | ghost75, Already established <- |
| 18:06.00 | ghost75 | ? |
| 18:06.06 | nacho2k | great, I dont need to use fax.. |
| 18:06.12 | nacho2k | thanks!! |
| 18:06.37 | [TK]D-Fender | ghost75, <[TK]D-Fender> nacho2k, jsut for regular phones? <nacho2k> [TK]D-Fender, yes |
| 18:06.52 | [TK]D-Fender | ghost75, "regular phones" = "clearly not fax machines" |
| 18:06.53 | ghost75 | and? |
| 18:08.04 | navaismo | XD |
| 18:09.01 | ghost75 | anyone used the spa112 with t.38 ? |
| 18:09.40 | [TK]D-Fender | ghost75, If he's only using regular phones then he doesn't need T.38. |
| 18:10.50 | ghost75 | i am asking not for him, i asked for me |
| 18:11.01 | nacho2k | another device I have to buy is a router with 2 fxs, any suggestion? I 'd prefer openwrt compatible but I didnt find any.. |
| 18:14.06 | [TK]D-Fender | <ghost75> i am asking not for him, i asked for me <- would help if we could tell that it wasn't about him ... and that it was worded as a question. But OK.... |
| 18:14.58 | [TK]D-Fender | nacho2k, SPA-122 |
| 18:15.32 | [TK]D-Fender | for dual port plus router |
| 18:15.47 | [TK]D-Fender | the SPA-112 replaced the PAP2 and also supports T.38 |
| 18:15.55 | [TK]D-Fender | So that solves all of them in one show it seems |
| 18:16.40 | nacho2k | [TK]D-Fender, great ! thanks for the info! I will buy some of them! |
| 18:17.14 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 18:19.51 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 18:29.29 | *** part/#asterisk mountainm2k (~msturtz@gw.booyahnetworks.com) |
| 18:31.09 | *** join/#asterisk WinstonNEWS (~WinstonNE@92-67-190-109.dsl.ovh.fr) |
| 18:31.13 | WinstonNEWS | Hi |
| 18:31.31 | navaismo | sup! |
| 18:34.13 | *** join/#asterisk tamiel (~tamiel@208.66.27.62) |
| 18:36.05 | WinstonNEWS | I have a problem with my server Asterisk 1.6.2.9-2 + squeeze6 I installed the asterisk-mp3 exten => 333,1(nbr2call),Read(NBR2CALL,'/var/lib/asterisk/sounds/miseenrelation',6,,2,20) and when I do, an error displays [Jan 29 19:10:49] WARNING[27975]: file.c:650 ast_openstream_full: File '/var/lib/asterisk/sounds/miseenrelation' does not exist in any format |
| 18:36.05 | WinstonNEWS | [Jan 29 19:10:49] WARNING[27975]: file.c:953 ast_streamfile: Unable to open '/var/lib/asterisk/sounds/miseenrelation' (format 0x1 (g723)): No such file or directory |
| 18:36.38 | WinstonNEWS | this file is a mp3 and fonctionnaly on the version 1.4 asterisk |
| 18:37.25 | sawgood | Hi: using this http://pastebin.com/S6KpPpNY (can someone tell me the syntax to have the call 1st answered by a process asking for a PIN number) and then continue on with the rest of the code? |
| 18:37.38 | igcewieling | WinstonNEWS: try exten => 333,1(nbr2call),Read(NBR2CALL,/var/lib/asterisk/sounds/miseenrelation,6,,2,20) |
| 18:37.39 | sawgood | Asterisk 1.8.20.0 on CentOS 5.5 |
| 18:37.52 | WinstonNEWS | on debian |
| 18:38.30 | sawgood | the code is from the web (called cidspoof.agi) ... since 2004 or so I think |
| 18:39.56 | igcewieling | WinstonNEWS: When you use excessive quoting the terrorists win. |
| 18:47.18 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 18:47.35 | WinstonNEWS | igcewieling, yes thank i have other problem :( |
| 18:47.37 | WinstonNEWS | ast_read_textfile: Error can't stat '/var/www/connect.txt' |
| 18:47.54 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
| 18:48.10 | WinstonNEWS | exten => s,n,ReadFile(user='/var/www/connect.txt',20) |
| 18:49.05 | wdoekes | drat.. terrorists win again |
| 18:52.42 | kaldemar | WinstonNEWS: drop the single quotes around the file. |
| 18:53.11 | kaldemar | sawgood: core show application Authenticate, core show application Read |
| 18:53.24 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 18:53.33 | igcewieling | WinstonNEWS: In Asterisk quotes are literal, so you are setting the variable user to be quote /var/www/connect.txt quote which is unlikely to be a valid filename. |
| 18:53.46 | *** join/#asterisk cheasee (~cheasee@vie-188-118-246-206.dsl.sil.at) |
| 18:53.49 | cheasee | hi |
| 18:54.05 | WinstonNEWS | ok |
| 18:55.11 | jmetro | i dont think there is any directoy above / labeled " so its most likely impossible. |
| 18:56.00 | Dovid | hi all. is there any way of convertaing voicemail messaes from wav to wav49? |
| 18:56.11 | wdoekes | jmetro: that makes it a relative path starting with the dir "'" |
| 18:57.00 | igcewieling | Dovid: Yes, using sox or using the Asterisk CLI "convert" command. |
| 18:57.05 | Dovid | thanks |
| 18:57.17 | igcewieling | the convert command will only work if the .wav already works in asterisk |
| 18:58.33 | Dovid | yes. i have an issue where i saet up 3 boxes using NFS. one is set to WAV49 the other two are wav |
| 18:58.44 | Dovid | the issue is when asterisk deletes a message it keeps the wav file. |
| 18:59.26 | igcewieling | *nod* you need them all to have the same settings for that sort of thing. |
| 18:59.41 | cheasee | ive got a gigaset phone connected to a pap2 adapter using my asterisk installation. on the gigaset, when having an acitve call and trying to put a second caller into a conference, the gigaset types *95* which doesnt work. now on the pap2 i found "Vertical Service Activation" but there is nothing about conference and no code *95 (but *91 *92 and *93) - how do i get this working? |
| 19:05.29 | cheasee | is there something i have to set on the asterisk installation? or is the pap2 the problem? im not sure where to look for the solution |
| 19:07.19 | [TK]D-Fender | <WinstonNEWS> exten => s,n,ReadFile(user='/var/www/connect.txt',20) <- stop putting quotes. they are not legal |
| 19:07.32 | [TK]D-Fender | WinstonNEWS, You have shown multiple lines like this.... |
| 19:07.41 | *** join/#asterisk navaismo (~navaismo@189.191.239.96) |
| 19:08.01 | [TK]D-Fender | WinstonNEWS, And that looks liek SET syntax where you should be providing a var name etc. |
| 19:08.59 | [TK]D-Fender | WinstonNEWS, correction, just the uotes seem bad right now... |
| 19:09.03 | [TK]D-Fender | quotes* |
| 19:13.00 | leifmadsen | fyi, Asterisk: The Definitive Guide 4e (based on Asterisk 11 LTS) is due to be submitted to O'Reilly on Thursday morning. Those who are interested in reviewing the book on OFPS have a last shot to do it tomorrow. The book review is available at http://ofps.oreilly.com/titles/9781449332426/ |
| 19:13.09 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-ejxbpipookvtgbun) |
| 19:22.06 | sawgood | http://pastebin.com/9Zbaf8es (can someone offer a tip for my dialplan process) |
| 19:28.33 | navaismo | sawgood, so if you enter correctly the passwd the call drops? can you show us the cli output with agi debug? |
| 19:29.14 | sawgood | navaismo: I've never used agi debug (sorry) so this will be my 1st ... |
| 19:29.38 | sawgood | Here is the only output shown at the Asterisk CLI when the call drops (with no sip set debug running) |
| 19:30.12 | navaismo | to enable the agi debug only execute: agi set debug on |
| 19:30.12 | sawgood | -- Auto fallthrough, channel 'SIP/Asterisk-box-00000038' status is 'UNKNOWN |
| 19:30.17 | sawgood | doing it now |
| 19:31.26 | sawgood | with agi set debug on (nothing more comes up on the screen) before the call drops (expect for the one line I showed you above) |
| 19:32.31 | navaismo | pb |
| 19:32.35 | navaismo | complete call |
| 19:33.16 | sawgood | http://pastebin.com/pWekaH2U |
| 19:33.32 | sawgood | I did not have sip set debug on for this call (only agi set debug on) |
| 19:34.57 | igcewieling | looks to me like asterisk is not seeing the line after Authenticate. do a dialplan show and verify what it shows matches what you configured. |
| 19:35.59 | igcewieling | "dialplan show DISA-Services1" |
| 19:36.17 | navaismo | yes, and your first pb show exten => 4155551213,n,Wait(1) but first is 4155551212 |
| 19:36.37 | igcewieling | Also there is no way in hell what you pasted as your dialplan is what is being shown in your CLI output. |
| 19:36.59 | igcewieling | exten => 4155551213 will NEVER match 9258850171 |
| 19:37.21 | igcewieling | navaismo: good catch |
| 19:38.51 | *** join/#asterisk solitude88 (~solitude8@wsip-24-234-107-109.lv.lv.cox.net) |
| 19:39.32 | sawgood | http://pastebin.com/RY0KAnRK (this is the extensions.conf running on the box now) |
| 19:40.20 | [TK]D-Fender | sawgood, Line 19 |
| 19:40.24 | [TK]D-Fender | sawgood, major FUBAR |
| 19:41.52 | sawgood | If line 13 and 14 are uncommented (removing lines 15-19) the process works (no authentication) |
| 19:43.11 | [TK]D-Fender | sawgood, Line 19 [TK]D-Fender> sawgood, major FUBAR |
| 19:43.13 | [TK]D-Fender | ^ |
| 19:43.32 | navaismo | ^^ |
| 19:43.46 | [TK]D-Fender | http://pastebin.com/RY0KAnRK <---- |
| 19:44.37 | sawgood | what to fix? |
| 19:45.16 | [TK]D-Fender | sawgood, I just gave you the line number. What do YOU see there? |
| 19:46.14 | sawgood | Line 19 = exten => 9258850171,2,AGI(cidspoof.agi) |
| 19:46.34 | [TK]D-Fender | and what looks wrong there relative to your OTHER lines? |
| 19:46.34 | navaismo | priority! |
| 19:46.45 | [TK]D-Fender | Step ***2*** perhaps? |
| 19:48.12 | sawgood | Well, I've tried changing exten => 9258850171,2,AGI(cidspoof.agi) to exten => 9258850171,n,AGI(cidspoof.agi) (no luck) |
| 19:48.31 | [TK]D-Fender | sawgood, Show us corrected dialplan (LOADED), and the call to match |
| 19:48.54 | sawgood | oh, is it going to work (let me try) |
| 19:50.01 | sawgood | oh, its working now! (dialplan reload) |
| 19:50.08 | sawgood | goes away |
| 19:50.28 | igcewieling | sawgood: before going away read the asterisk book |
| 19:50.54 | sawgood | I read it all day ... its just you people are so tempting! |
| 19:51.02 | sawgood | I offer pizza, but they don't want it! |
| 19:52.49 | igcewieling | Generally the only time you don' t use "n" for your priority is for the first line (i.e. priority 1) |
| 19:53.48 | sawgood | igcewieling: thank you! |
| 19:57.36 | WIMPy | n doesn't always work. |
| 19:57.55 | WIMPy | Not if you do creative dialplans :-) |
| 20:01.56 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200:221:6aff:feb8:e0b2) |
| 20:04.43 | igcewieling | WIMPy: my dialplans must not be creative. I am considering "n(start)" the same as "n" |
| 20:05.17 | WIMPy | ? |
| 20:05.36 | WIMPy | doesn't get that comment |
| 20:06.23 | igcewieling | In my statement "Generally the only time you don' t use "n" for your priority is for the first line (i.e. priority 1)", I consider priority "n(start_or_whatever)" to be an "n" priority. I never need any priority other than 1 |
| 20:06.48 | WIMPy | Oh |
| 20:07.19 | WIMPy | That's not too creative :-) But you can have multiple extensions share some priorities. |
| 20:07.38 | igcewieling | That is just asking for trouble. |
| 20:08.07 | WIMPy | You can do it with s and n, but I wouldn't try that without the classic manually numbered priorities. |
| 20:08.15 | WIMPy | Why? |
| 20:08.15 | igcewieling | makes things way too complex for easy troubleshooting. Yes, I know FreePBX uses that "feature" and it makes it very very very hard to debug. |
| 20:08.24 | WIMPy | It's extremely handy. |
| 20:08.42 | igcewieling | It is extremely lazy in my opinion. |
| 20:08.58 | WIMPy | That's the idea. |
| 20:09.08 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.147.159) |
| 20:18.33 | igcewieling | I thought the idea was to have a PBX which is easy to troubleshoot and maintain? |
| 20:20.15 | WIMPy | Not sure how Asterisk fits in there. But my idea is less configuration=easier. |
| 20:22.39 | WinstonNEWS | i have other problem |
| 20:22.53 | WinstonNEWS | moh started and i not fonctionaly |
| 20:23.05 | WinstonNEWS | StartMusicOnHold("SIP/incoming-00000000", "default") |
| 20:23.19 | WinstonNEWS | Wait("SIP/incoming-00000000", "2") |
| 20:23.22 | WinstonNEWS | Dial("SIP/incoming-00000000", "SIP/technique,120,tTm") |
| 20:23.50 | [TK]D-Fender | WinstonNEWS, MusicOnHold is a BLOCKING app. It will not start in the background. That is not what it does. |
| 20:24.36 | WinstonNEWS | hum background is not dial + music |
| 20:24.39 | [TK]D-Fender | WinstonNEWS, and I'm pretty sure you don't want to allow "incoming" the right to transfer themselves... |
| 20:24.50 | [TK]D-Fender | WinstonNEWS, "core show application dial" <---- |
| 20:24.59 | [TK]D-Fender | WinstonNEWS, It has an option for MoH instead of ringing.... |
| 20:25.25 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 20:26.32 | saint_ | if I select to load the WAV sound files , and if I have my clients setup as allow=ulaw in sip.conf, will asterisk still be able to play those sound files over ? |
| 20:27.45 | WinstonNEWS | its good i have lost to answer :p |
| 20:29.26 | *** join/#asterisk DoSJustin (~justin@vpn.bctconsulting.com) |
| 20:30.08 | *** join/#asterisk ideaman55 (~ideaman55@173-10-29-217-BusName-utah.ut.hfc.comcastbusiness.net) |
| 20:31.30 | *** join/#asterisk nny (~Scott@174.107.223.14) |
| 20:36.11 | [TK]D-Fender | saint_, Yes |
| 20:36.44 | *** join/#asterisk jsjc (~Adium@124.Red-83-55-40.dynamicIP.rima-tde.net) |
| 20:37.03 | saint_ | [TK]D-Fender: does it take any special ressource or license for that ? |
| 20:37.12 | [TK]D-Fender | saint_, No |
| 20:37.14 | saint_ | [TK]D-Fender: if not, what is the point of having different file formats ? |
| 20:37.35 | saint_ | [TK]D-Fender: like why would one chose ulaw instead of mp3 or wav ? |
| 20:38.06 | nny | hmm. Have an issue here I am looking into. Have some numbers that upstream plays "Number not in service" to normally (say from a cell). Through both VoIP providers the calls just ring eternally. wireshark shows it stops at 183 Progress (and Ringing to the sip client). Sip debug also here: http://pastebin.com/ZqDQz3CU |
| 20:38.06 | nny | I am considering adding a timeout to outbound dial statements. This should not affect my current plan it is (designed to try 2 providers if the first one fails, see http://pastebin.com/uREBrMX3 ). I am skeptical about assinging a generic (call could not be completed) to all calls over 30 seconds. |
| 20:38.06 | nny | Thoughts? |
| 20:40.23 | nny | I should note that the debug is at ~ 1 minute when I closed it (and ran sip show channels) it just rings the whole time. The provider never intercepts to state the number is invalid or otherwise. I am assuming that both providers use the same upstream route for this number |
| 20:46.31 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 20:46.31 | [TK]D-Fender | saint_, What if some of your calls come in ULAW ... and some come in as GSM? What then? |
| 20:47.04 | saint_ | that is why i asked if asterisk would be able to play files over |
| 20:47.05 | [TK]D-Fender | <saint_> [TK]D-Fender: like why would one chose ulaw instead of mp3 or wav ? <-- immediately there is a question of how much PROCESSOR LOAD it takes to convert from X to Y. |
| 20:47.10 | *** join/#asterisk zhando (~user@c-67-161-122-77.hsd1.wa.comcast.net) |
| 20:47.51 | zhando | looks like * doesn't like systemd enumeration of ethernet nics? |
| 20:47.53 | saint_ | [TK]D-Fender: so if i have one connection to/from google voice, and if i take in account what you said (what if a call comes in ULAW).. how the heck do i know what format i should install ? |
| 20:48.58 | [TK]D-Fender | saint_, Well if all your calls are in ulaw .. then what format do you think would be best for your files? |
| 20:49.24 | saint_ | saint_: ulaw, agreed. but how do i know that all calls are in ulaw ? |
| 20:49.41 | saint_ | [TK]D-Fender: is it something like sip calls are all ulaw for example ? |
| 20:50.49 | [TK]D-Fender | saint_, No... it's a question of knowing what codec your provider supports and you want to use. |
| 20:51.14 | zhando | nevermind... the real question is why does the asterisk dir go away in /run in arch linux.. recreate it manually with the right permissions and everything's fine. |
| 20:51.15 | *** join/#asterisk timahvo1 (~rogue@41.90.177.106) |
| 20:51.48 | saint_ | [TK]D-Fender: because i just removed all the ulaw sounds , and installed wav ones , and the moh is still playing ... |
| 20:52.32 | saint_ | [TK]D-Fender: the only difference is the type of moh that was played. robotic on ulaw, nice on wav. but it's still playing when i call from outside. |
| 20:54.11 | *** join/#asterisk Inf0r (uid2810@gateway/web/irccloud.com/x-cwgrlwvqohfjrycf) |
| 20:55.23 | Inf0r | i need an inbound route script that will ring but not answer a specific did. i don't want it to go to voicemail and i don't want to have an extension registered for this. is this possible? Thanks |
| 20:56.12 | WIMPy | So you just want it to do nothing pretending to be ringing or what? |
| 20:56.49 | [TK]D-Fender | Inf0r, exten => 12345678,1,Ringing |
| 20:57.02 | [TK]D-Fender | Inf0r, exten => 12345678,2,wait(500) |
| 20:57.08 | [TK]D-Fender | Inf0r, exten => 12345678,3,Goto(1) |
| 20:57.11 | [TK]D-Fender | Done |
| 20:57.21 | Inf0r | wimpy, ya for a googlevoice number |
| 20:57.47 | Inf0r | id like it to ring for 10 seconds and then drop the call |
| 20:57.53 | saint_ | Inf0r: if you do that, google voicemail will pick up at some point.. you know that right ? |
| 20:58.12 | WIMPy | You should be able to modify [TK]D-Fenders answer yourself. |
| 20:58.33 | Inf0r | ya i want google voicemail to pickup, but apparently it requires a did to be associated with it |
| 20:58.33 | [TK]D-Fender | exten => 12345678,1,Ringing exten => 12345678,2,wait(10) exten => 12345678,3,Hangup |
| 20:58.36 | *** join/#asterisk datarecall (~data@loxely.illusivecreations.com) |
| 20:58.52 | saint_ | Inf0r: i m just curious.. wha'ts the point of having a google number if you don t pickup ..? |
| 20:59.28 | Inf0r | i just want it for a voicemailbox, but i guess you have to have a separate did associated with your google voice number? |
| 21:00.05 | WIMPy | Why do you want googles VM if you've got Asterisk? |
| 21:00.09 | nny | hmm anyone have thoughts on my quandry. I think it's pretty much provider based but just double checking |
| 21:00.51 | saint_ | Inf0r: if you want it for a mailbox, why don't you turn ON the DO NOT DISTURB in google voice ? All calls will go directly in the mailbox.. |
| 21:00.51 | Inf0r | its an emergency mailbox in case asterisk is down |
| 21:01.05 | Inf0r | saint let me see |
| 21:02.05 | saint_ | I still have issues with those digium phones that I cant provision. |
| 21:02.15 | saint_ | technical support has been .... useles if i can say .. |
| 21:02.46 | saint_ | if I ran my business like that, i would go bankrupt in no time with my customers. |
| 21:04.43 | Inf0r | saint, ya that works |
| 21:04.46 | Inf0r | facepalm |
| 21:04.57 | Inf0r | thanks for the help anyways fender and all |
| 21:05.02 | *** part/#asterisk Inf0r (uid2810@gateway/web/irccloud.com/x-cwgrlwvqohfjrycf) |
| 21:06.36 | WIMPy | saint_: Is there any information anywhere on how to provision them at all? |
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| 21:10.25 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 21:14.16 | saint_ | WIMPy: there is. I followed it. I sent to the tech support guy all the files he needed (they have a script that takes it all), and all he could come up with was that I should have renamed my [network] in my digium file, which i did. still not working. |
| 21:14.21 | *** join/#asterisk mimecar (~mimecar@unaffiliated/mimecar) |
| 21:14.27 | *** part/#asterisk mimecar (~mimecar@unaffiliated/mimecar) |
| 21:15.20 | Qwell | saint_: What is the issue you're having? |
| 21:18.19 | *** join/#asterisk omikron4 (~Thunderbi@unaffiliated/omikron4) |
| 21:19.15 | saint_ | Qwell: The phones are saying COMMUNICATION ERROR TImed out contacting proxy. / and the asterisk logs are showing this: http://pastebin.com/TxX19zBj |
| 21:19.32 | saint_ | Qwell: I'm trying from scratch with a brand new intall that i'm doing now. |
| 21:19.36 | *** part/#asterisk omikron4 (~Thunderbi@unaffiliated/omikron4) |
| 21:20.07 | Qwell | what's the xxxx all about? |
| 21:21.05 | saint_ | qwell asterisk IP public address |
| 21:24.28 | Qwell | and what is at 192.168.1.117? |
| 21:25.48 | *** join/#asterisk pichurris (~marcos@189.134.208.202) |
| 21:26.40 | nny | should i repost my question or is it just one of those things that is too odd to answer? |
| 21:27.32 | pichurris | I can not log admin account in the web interface, does asterisk has a default user and pass? |
| 21:27.52 | Qwell | pichurris: Asterisk does not have a web interface. |
| 21:28.06 | pichurris | Qwell, I just install something GUI |
| 21:28.12 | Qwell | So ask them. |
| 21:30.07 | nny | pichurris: the gui is not native to asterisk, this channel is for native asterisk only. I'd suggest investigating what GUI you are using and find the appropriate channel |
| 21:30.10 | jmetro | asterisk doesnt have a user or password on its own unless you mean the user account that YOU setup on the server for asterisk. |
| 21:30.29 | pichurris | nny, thank you ! |
| 21:31.06 | pichurris | nny, I'm just learning and following some online tutorials sry. |
| 21:34.26 | Qwell | saint_: Hold on. The xxxx is Asterisk's IP? Are you sure about that? |
| 21:35.01 | Qwell | Where's the SIP debug to go along with this? What is the proxy in the way that you didn't mention? |
| 21:36.02 | saint_ | Qwell: no no , my bad , i m sorry |
| 21:36.09 | saint_ | xxx is my public IP address |
| 21:36.18 | saint_ | the digium phone is on 192.168 private network |
| 21:36.25 | saint_ | the Asterisk had a public IP address |
| 21:36.52 | Qwell | and what address are you telling the phone to connect to? |
| 21:37.07 | saint_ | some IP on the internet. it's not visible in this trace |
| 21:37.12 | saint_ | that 's directly from asteirks |
| 21:37.21 | Qwell | How am I supposed to help you if it's not visible? |
| 21:38.27 | saint_ | well.. it's on the internet.. what do you want me to tell you ? asterisk does not show it in the traces no matter what .. |
| 21:39.00 | Qwell | The phone is obviously hitting Asterisk. |
| 21:39.32 | saint_ | yeah, i know |
| 21:39.46 | Qwell | One of these things cannot, therefore, be true. |
| 21:40.17 | saint_ | Qwell: i m going to change the architecture for now. i m going to put asterisk in the same private network, so nobody can say it's because of that, and try to have it work this way. |
| 21:40.19 | saint_ | we'll see how it goes. |
| 21:41.54 | saint_ | hahahaha... the spam.wav file is hillarious |
| 21:47.54 | Qwell | doesn't know why he bothers sometimes |
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| 21:57.28 | jmetro | saint_ try tt_allbusy |
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| 22:21.00 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 22:23.31 | danfromuk | Hi. Is it a requirement that phones stop ringing when the queue periodic announcement is playing? Currently, timeout is set to 60 as I want the phones to ring for 60 seconds. But if its set the 60, the period-announce-frequency isn't checked till the timeout. |
| 22:24.20 | [TK]D-Fender | danfromuk, Yes |
| 22:25.00 | danfromuk | Shame. Ok, thanks |
| 22:25.28 | WIMPy | ouch |
| 22:36.15 | sawgood | exten => _1NXXNXXXXXX,1,Dial(SIP/Asterisk-Services1/${EXTEN}) (on outbound calls, how can I change ${EXTEN} from x402 showing (to the actual DID number) (I tried this line exten => _1NXXNXXXXXX,1,Dial(SIP/DISA-Services1/9258850169) (call fails with a "603 delcline" from the ITSP |
| 22:37.55 | navaismo | what? |
| 22:38.23 | navaismo | you want to change the destination number or the CID? |
| 22:38.49 | sawgood | outbound CID to show a hard coded number from extensions.conf everytime |
| 22:39.20 | sawgood | I'm sorry (I need a break) my bad |
| 22:39.23 | navaismo | ¿? |
| 22:39.23 | sawgood | sorry! |
| 22:40.01 | navaismo | Look at SET and CALLERID(all|num|name) |
| 22:40.14 | sawgood | ty |
| 22:42.04 | navaismo | and take a big look on the ~book |
| 22:42.11 | navaismo | ~book |
| 22:42.12 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:42.48 | sawgood | navaismo: I know you mention reading the book (thank you) I own and have read every copy ever made (a few times) ... |
| 22:43.27 | navaismo | well we discover news things reading again a book ;) |
| 22:43.33 | sawgood | I do agree |
| 22:43.56 | jmetro | its like a tv show you loved as a kid, you watch it again as an adult and realize how ... wrong it was |
| 22:44.11 | navaismo | haha |
| 22:45.33 | *** join/#asterisk janmate (~janmate@89.173.160.127) |
| 22:45.50 | sawgood | Eight-Is-Enough (remember that one)? |
| 22:45.55 | sawgood | like that was real! |
| 22:45.58 | *** join/#asterisk vfabi (~fabi@host-static-89-41-121-42.moldtelecom.md) |
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| 22:55.50 | ChannelZ | sawgood: well there's that mormon couple with their child army... |
| 22:56.01 | volga629 | Hello Everyone, what mean small letter "a" in sip show peers command ? |
| 22:56.12 | volga629 | asterisk 11 |
| 22:56.30 | ChannelZ | or whatever they are |
| 22:57.07 | ChannelZ | volga629: what column? |
| 22:57.19 | navaismo | volga629, in which column? |
| 22:57.43 | volga629 | Forcerport |
| 22:58.37 | drmessano | ACL |
| 22:58.40 | drmessano | Not Forceport |
| 22:59.53 | drmessano | http://lists.digium.com/pipermail/asterisk-users/2009-September/238029.html <--- |
| 22:59.54 | ChannelZ | dunno what little a is versus big A |
| 23:00.15 | ChannelZ | Perhaps he has a terrible font. |
| 23:00.21 | drmessano | Could be |
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| 23:05.36 | volga629 | thank s |
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| 23:19.42 | rue_work | anyone good with mgcp.conf? |
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| 23:37.21 | ChannelZ | Hmm. Why do the orange Download buttons on asterisk.org/downloads have a 'W' on them? Is this Wordpress faff? |
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| 23:49.44 | ChannelZ | actually I guess this is Drupal. |
