| 00:05.10 | *** join/#asterisk m0sphere (m0sphere@S01060018e78c9cff.cg.shawcable.net) |
| 00:21.23 | *** join/#asterisk fling (~fling@fsf/member/fling) |
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| 01:41.33 | *** part/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 01:41.34 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
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| 02:18.50 | outtolunc | Required IE '50' for security event type '12' not present (can it get any more cryptic) ;) |
| 02:29.24 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:29.37 | outtolunc | closer .. Required IE '50' for security event type '12' (InvalidPassword) not present |
| 03:40.04 | m0sphere | outtolunc, just a guess, but IE 5.0? |
| 03:40.08 | m0sphere | other than that, i have no clue |
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| 03:57.00 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 03:59.07 | m0sphere | for any of you using festival/text2wav TTS, which voice do you believe is best? |
| 04:03.47 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
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| 04:48.10 | *** join/#asterisk talntidd (4c1cd93d@gateway/web/freenode/ip.76.28.217.61) |
| 04:49.45 | talntidd | http://pastebin.com/VDm7d06t |
| 04:49.59 | talntidd | can someone assist me? background() is saying the file doesn't exist. Everything else says otherwise... |
| 04:50.47 | talntidd | actually, this one has the extensions context in it too: http://pastebin.com/9L0KkNbv |
| 04:51.01 | *** join/#asterisk m0sphere (m0sphere@S01060018e78c9cff.cg.shawcable.net) |
| 04:52.15 | [TK]D-Fender | talntidd: And you should know better... you NEVER specify the extension with Playback & Background |
| 04:52.54 | [TK]D-Fender | silly goose |
| 04:55.24 | talntidd | hrrrm? I don't follow you |
| 04:58.35 | [TK]D-Fender | -- Executing [s@closed:3] BackGround("IAX2/north-ella-116", "/var/lib/asterisk/sounds/custom/Week1_20131125Thanksgiving.gsm") in new stack |
| 04:58.46 | [TK]D-Fender | You told it the EXACT filename |
| 04:58.48 | [TK]D-Fender | that is not allowed |
| 04:59.05 | [TK]D-Fender | * picks the extension automatically |
| 05:00.23 | talntidd | so I need to tell it /custom/Week1.... ? |
| 05:00.36 | talntidd | OH! |
| 05:00.38 | talntidd | my god |
| 05:00.40 | talntidd | i'm stupid |
| 05:00.43 | talntidd | the FILE extension |
| 05:00.47 | talntidd | not the phone extension |
| 05:00.50 | talntidd | fuck me. |
| 05:01.09 | talntidd | thanks dude. *smacks face* |
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| 05:27.34 | *** join/#asterisk ziz212 (~chatzilla@203.115.12.246) |
| 05:29.02 | ziz212 | Hi, My ISP blocked Voip and I couldnt use internet to connect to my Asterisk server. What would be the solution for that ? Can I use TLS and SRTP for this ? |
| 05:31.07 | [TK]D-Fender | VPN. |
| 05:31.14 | [TK]D-Fender | And hope you don't get caught. |
| 05:31.53 | ziz212 | Cant we use TLS and SRTP |
| 05:31.56 | ziz212 | for this |
| 05:33.00 | [TK]D-Fender | Did you try? |
| 05:33.17 | m0sphere | ziz212, did they block the server or the port? |
| 05:34.06 | m0sphere | if they blocked the server, demand to know why |
| 05:34.07 | ziz212 | Yes I have tried. But in that environment I couldn't up the TLS connection to the server. |
| 05:34.14 | m0sphere | if they blocked the port, change the port? |
| 05:34.22 | ziz212 | They have blocked the ports |
| 05:36.01 | ziz212 | the thing is I have tried TLS and SRTP in LAN but not working on internet. So I searched by directly calling them. They point out the reason becuse they blocked the ports. 5060, 5061, and many more |
| 05:36.34 | ziz212 | I have tried to change the ports 5061 to 15061 .. but it is not working |
| 05:37.34 | ziz212 | Do I need to change the RTP ports also |
| 05:38.37 | ziz212 | which is 10000 to 20000 |
| 05:38.54 | [TK]D-Fender | CHANGE THE PORTS |
| 05:39.14 | [TK]D-Fender | You are asking questions you already have the answer to. |
| 05:39.32 | [TK]D-Fender | And you're not showing the failure |
| 05:39.35 | [TK]D-Fender | or configs |
| 05:39.37 | [TK]D-Fender | or anything |
| 05:39.43 | [TK]D-Fender | just like the last time you asked |
| 05:40.13 | talntidd | welp, my application is finally done. :) |
| 05:40.25 | talntidd | thanks TK. I tried for... like 2 hours.. on that freaking extension thing. :( |
| 05:44.51 | *** join/#asterisk redotis (~redotis@nc-76-5-176-123.dhcp.embarqhsd.net) |
| 05:45.24 | ziz212 | [TK]D-Fender: I tried to prepare a doc to share. but after that I felt that it is not secure not because of you all . I know you all helping to community. but if i share it it will publish all over there. I can give you access to teamviwer and you can see any information you need.. Pls understand my my situation. Pls come to teamviwer and help me to solve my issue. |
| 05:45.56 | [TK]D-Fender | use a private pastebin if you have to. |
| 05:46.27 | [TK]D-Fender | But I'm not wasting my time on teamviewer. |
| 05:46.43 | [TK]D-Fender | If you want help, provide real information and stop being difficult |
| 05:47.07 | ziz212 | I can give you any access to my server |
| 05:47.36 | [TK]D-Fender | pick some private PB service |
| 05:47.38 | ziz212 | Frankly saying I dont have private pastebin |
| 05:47.47 | [TK]D-Fender | I'm not logging in and digging up proof for you |
| 05:49.47 | redotis | I'm running "asterisk -rx "channel originate SIP/4500 extension 999@voiceline" to try to dial people automatically into conference 999 but in meetme the callerid num shows up as unknown |
| 05:49.59 | redotis | Anyone have a suggestion |
| 05:50.17 | redotis | the number doesn't display but the name from sip.conf actually does |
| 05:50.34 | ziz212 | I am sorry to bother you for that. But I don't have any option with me and I am going here and there for helps .. I know you have good reputation and have helped me past for some problems.. If you could do this by considering my request .. I am grateful to you. |
| 05:50.49 | redotis | crap |
| 05:50.53 | redotis | i think i see the problem |
| 05:53.22 | redotis | nevermind...derp |
| 05:53.25 | [TK]D-Fender | redotis: channel originate can't set the callerid. |
| 05:53.38 | redotis | yeah my callerid was setup wrong in the sip.conf |
| 05:53.40 | redotis | derp |
| 05:53.41 | redotis | thanks man |
| 05:53.41 | [TK]D-Fender | redotis: You'd have to dial a local channel that can set it or use another method |
| 05:54.03 | redotis | thank you |
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| 06:11.57 | Penguin | *GASP* |
| 06:12.01 | Penguin | You're unignored? |
| 06:16.19 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 06:20.19 | outtolunc | much better.. Required IE '50' (Challenge) for security event type '12' (InvalidPassword) not present |
| 06:24.11 | fonewiz | Anyone have any experience setting up Asterisk on Digital Ocean Vm's or Amazon EC2? Primarily looking for Digital Ocean info. Been running Asterisk for years on dedicated hardware but the benefits of virtualization are becoming harder and harder to ignore in terms of redundancy, scalability etc. |
| 06:25.18 | fonewiz | Wondering if I just do a plain vanilla install of Asterisk or if I need to set anything special before letting the make install rip |
| 06:25.52 | fonewiz | I read somewhere that I should set Build Native=NO in menu select but only read this in one case so, not sure.. |
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| 07:04.35 | *** join/#asterisk D30 (~deo@203.177.9.66) |
| 07:05.05 | D30 | hi all, can someone share some ideas.. whats actually the meaning of this? |
| 07:05.14 | D30 | DAHDI/3-1 - im pertaining to 3-1 |
| 07:05.25 | D30 | i know its on channel 3 but what is 1?? |
| 07:06.10 | kaldemar | call 1. |
| 07:06.27 | D30 | kaldemar: can you elaborate ? whats call 1? |
| 07:06.31 | outtolunc | think of it as tech/dev-occurrence |
| 07:06.32 | D30 | call #1 ? |
| 07:06.37 | kaldemar | if you'd put the first call on hold and make another one, you'd see 3-2. |
| 07:06.49 | D30 | ahh okay kaldemar got it ;) |
| 07:06.53 | D30 | thanks outtolunc |
| 07:06.59 | kaldemar | don't worry about it too much. |
| 07:08.15 | D30 | just wondering kaldemar hehe anyways thanks :) |
| 07:08.18 | *** join/#asterisk elcontrastador (~textual@173-12-219-184-Fresno.hfc.comcastbusiness.net) |
| 07:09.44 | elcontrastador | Is there any way to supress console output of a particular module without unloading it? |
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| 07:39.09 | Joel_re | hey, Im trying to build a game around asterisk, where in phone keypad input is accepted as controls |
| 07:39.38 | Joel_re | what I have so far is keypad inputs recognized from each caller |
| 07:39.47 | Joel_re | Im trying to figure whats the right way ahead |
| 07:40.09 | elcontrastador | I'm not able to get the 't' extension to work in my call tree. Fails with Auto fallthrough, channel 'SIP/<sip phone name>' status is 'UNKNOWN' |
| 07:40.18 | Joel_re | I plan to have Agi<->nodejs<->websockets<->html5 browser app |
| 07:40.24 | elcontrastador | never hits 't' |
| 07:40.27 | Joel_re | does this sound right? |
| 07:48.26 | *** join/#asterisk gryphon (~gryphon@82.140.120.164) |
| 07:48.38 | ChannelZ | elcontrastador: Are you using Background or WaitExten? |
| 07:49.00 | elcontrastador | both...i got it working just now |
| 07:50.01 | elcontrastador | i need sleep... :-) |
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| 08:36.26 | ziz212 | [TK]D-Fender: Can I post the pastbin to your private chat? |
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| 08:48.46 | ziz212 | My ISP blocked Voip and I couldnt use internet to connect to my Asterisk server. What would be the solution for that ? Can I use TLS and SRTP for this ? |
| 08:49.21 | ziz212 | kaldemar: Can you help for this? |
| 08:52.36 | kaldemar | why are you targeting me with that? |
| 08:59.33 | tomodachi | what software are people using for storing info of call lenghts etc in a easily presentable way. |
| 08:59.53 | tomodachi | i just want some simple call statistics like number of calls, lenght from who to who etc |
| 09:02.31 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 09:03.00 | tomodachi | ziz212: just set up a vpn and route your iax / sip trunk through it |
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| 09:15.14 | Gugge | ziz212: change ISP |
| 09:15.40 | Gugge | and remember to telle the ISP why you left them. |
| 09:25.34 | *** join/#asterisk D30 (~deo@203.177.9.66) |
| 09:27.02 | ziz212 | Chaning ISP is not that much of easy |
| 09:27.27 | ziz212 | I have tried this by changing the port also |
| 09:27.36 | ziz212 | I got the blocked port list |
| 09:27.40 | ziz212 | and did that |
| 09:28.02 | ziz212 | But still TLS and SRTP is not working |
| 09:30.31 | ziz212 | Cant we use SRTP with TLS though this blocked line ? |
| 09:32.38 | tomodachi | ziz212: is it just one phone? or do you have many clients? |
| 09:33.03 | ziz212 | Hoping to put more than 10 clients |
| 09:33.07 | ziz212 | with sip trunks |
| 09:41.31 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:58f9:26e4:24ad:7a71) |
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| 09:48.38 | ziz212 | Can you help me for this ? |
| 09:49.20 | ziz212 | I have changed the port to 5080 and still it is not workin |
| 09:49.24 | ziz212 | working? |
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| 12:15.40 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:52.41 | SupaYoshi | does anyone know here, how i can debug paging / intercom? |
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| 14:35.22 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 14:52.20 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 15:17.14 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 15:38.44 | eXcAliBuR | I'm getting back into a project that I started a while back, and I have a digium phone that fails to configure using DPMA, says error fetching config from proxy |
| 15:38.52 | eXcAliBuR | I did a factory reset |
| 15:38.55 | eXcAliBuR | and that didn't work |
| 15:39.12 | eXcAliBuR | I don't believe the phones are set to do anything by mac address, but i'm not sure |
| 15:39.38 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:40.05 | eXcAliBuR | my notes that i did before, aren't very complete |
| 15:40.16 | eXcAliBuR | they have the ip address of the pbx box, and login stuff |
| 15:40.18 | eXcAliBuR | :/ |
| 15:40.27 | eXcAliBuR | lol |
| 15:42.30 | eXcAliBuR | also I have a very nice voicemail setup, "the person at extension *** is unavalible" then it hangs up |
| 15:42.41 | eXcAliBuR | kinda funny |
| 15:48.42 | eXcAliBuR | this is what i see when it's attempting to do it's thing |
| 15:48.43 | eXcAliBuR | http://pastebin.com/ttzyeyYL |
| 15:59.31 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 16:03.10 | gaps | channel not getting released immediately after attended transfer in Asterisk 11.2.1 |
| 16:16.25 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
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| 16:39.39 | *** join/#asterisk esperegu (~quassel@ip-213-124-221-122.ip.prioritytelecom.net) |
| 16:40.56 | esperegu | Hi. I keep getting a 'The Number You have Dialed is not In Service.' anyone knows how to debug this? |
| 16:41.25 | WIMPy | Dial a number that exists. |
| 16:41.51 | WIMPy | Maye you're sending it in the wrong format? |
| 16:42.20 | Chainsaw | esperegu: That's the KPN version of the message. ISDN or analog? |
| 16:42.50 | esperegu | it is played to me by asterisk |
| 16:42.57 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 16:43.02 | esperegu | I call with my mobile to asterisk box |
| 16:43.05 | Chainsaw | raises eyebrow |
| 16:43.06 | Greenlight | So where in your dialplan are you playing it |
| 16:43.08 | esperegu | (sip registration) |
| 16:43.10 | Greenlight | And why |
| 16:43.55 | [TK]D-Fender | esperegu: Show us the call |
| 16:43.55 | Greenlight | Just because you called from your mobile doesn't mean Asterisk is playing it |
| 16:45.04 | esperegu | this is what asterisk saids on commandline when I call: |
| 16:45.05 | esperegu | http://dpaste.com/1482537/ |
| 16:45.33 | [TK]D-Fender | esperegu: First.. that is FREEPBX... |
| 16:46.18 | [TK]D-Fender | esperegu: And that call is not matching a trunk pointing to the right context and doesn't look like you have an INBOUND ROUTE to match it either |
| 16:46.24 | [TK]D-Fender | ~freepbx |
| 16:46.24 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:46.27 | [TK]D-Fender | ^^ |
| 16:46.41 | [TK]D-Fender | Please join their channel for continued support |
| 16:47.37 | esperegu | how to see if I have freepbx? command line saids: # asterisk -v |
| 16:47.37 | esperegu | Asterisk 1.8.11.1-1digium1~lucid |
| 16:48.05 | Greenlight | ... |
| 16:48.06 | tm1000 | esperegu: did you configure asterisk by hand |
| 16:48.11 | tm1000 | or through a web interface |
| 16:48.29 | esperegu | tm1000: it supposed to just work. I use http://www.linuxmce.org |
| 16:49.01 | esperegu | tm1000: I only filled in a sip account. It used to work but I changed provider and now it does not work anymore |
| 16:49.13 | [TK]D-Fender | esperegu: You didn't set your trunk/inbound route up right |
| 16:49.23 | [TK]D-Fender | esperegu: And this is not a GUI support channel |
| 16:49.38 | tm1000 | hm |
| 16:49.44 | tm1000 | looks like they ripped off some freepbx dialplan |
| 16:49.50 | Greenlight | Indeed |
| 16:49.54 | esperegu | [TK]D-Fender: I know. thats why I am on the console |
| 16:50.01 | [TK]D-Fender | they used to use a forked version of it. |
| 16:50.14 | tm1000 | [TK]D-Fender: thanks |
| 16:50.15 | esperegu | tm1000: I know they used to use freepbx but got rid of it |
| 16:50.23 | [TK]D-Fender | esperegu: Yes, and since you're at the console qwe can see your trunk/inbound route was not set up right otherwise ti'd have matched the call |
| 16:51.55 | [TK]D-Fender | tm1000: Or some extra stuff around it.... feel free to drudge up every time n3glv wretched all over at the mere mention of it (almost entirely his doing). |
| 16:53.17 | tm1000 | [TK]D-Fender: I just didnt know |
| 16:53.42 | [TK]D-Fender | tm1000: I thought it had dropped off the planet.. apparently not. |
| 17:01.53 | gaps | channel not getting released immediately after attended transfer in Asterisk 11.2.1 |
| 17:02.59 | Greenlight | gaps: Is that a question ? |
| 17:03.42 | gaps | Greenlight: yes |
| 17:03.52 | esperegu | looks like this is the part that gets run: http://dpaste.com/1482541/ |
| 17:04.02 | esperegu | that seems normal/ok ? |
| 17:04.14 | Greenlight | gaps: Can you elaborate on exactly what you're asking ? |
| 17:04.19 | [TK]D-Fender | esperegu: I've already told you your trunk is not being matched. Stop looking in the dialplan for this |
| 17:05.19 | [TK]D-Fender | gaps: We are at 11.6.0 Please test against current. |
| 17:05.30 | gaps | Greenlight: I did a attended transfer... example - A calls B, B calls C and did a attended transfer of A and C, but still the channel is occupied |
| 17:05.57 | Greenlight | Which channel, and what do you mean "occupied" ? |
| 17:06.12 | esperegu | [TK]D-Fender: is that set in here: http://dpaste.com/1482544/ ? |
| 17:06.33 | Greenlight | To rule out any possible bugs, I'd advise to upgrade to current then test again. |
| 17:06.35 | [TK]D-Fender | esperegu: Why are you messing direct in MySQL? |
| 17:06.52 | esperegu | [TK]D-Fender: I don't know how else to change it? |
| 17:07.00 | gaps | Greenlight: I mean the A and C channel are external numbers and the channel is still active till any one of the leg disconnects |
| 17:07.00 | [TK]D-Fender | esperegu: in FreePBX like everything else |
| 17:07.18 | esperegu | [TK]D-Fender: there is no FreePBX as far as I know |
| 17:07.18 | gaps | Greenlight: actually it supposed to disconnect once B transfer the call to C right? |
| 17:07.20 | Greenlight | Which channel ? |
| 17:07.24 | Greenlight | Lets see the call |
| 17:07.30 | [TK]D-Fender | esperegu: http://wiki.linuxmce.org/index.php/Telecom |
| 17:07.33 | esperegu | [TK]D-Fender: only a small webinterface to change the sip address |
| 17:07.36 | [TK]D-Fender | esperegu: 4. Configure Asterisk using FreePBX. This is LMCE's telecom brain |
| 17:07.48 | Greenlight | And what do you mean "active". If the call is still in progress, of course the channel is active. |
| 17:07.55 | Greenlight | I'm really not understnading the queston. |
| 17:07.58 | [TK]D-Fender | esperegu: that is FreePBX dialplan, and instantly recognizable as such |
| 17:08.03 | gaps | Greenlight: let me explain |
| 17:08.53 | gaps | Greenlight: I have a SIP trunk configured in Asterisk, there is a incoming call to the SIP trunk and it is routed to one extension, the extension is transferring the call to external Toll Free number and disconnected from his end |
| 17:08.57 | drmessano | LinuxMCE has a version built against Ubuntu 12.04 that is "Pre-Alpha"? |
| 17:09.09 | esperegu | [TK]D-Fender: http://forum.linuxmce.org/index.php/topic,12923.msg93130.html#msg93130 |
| 17:09.17 | drmessano | fires up his VaporWare 2000 and scans |
| 17:09.27 | drmessano | Yep, dead |
| 17:09.34 | gaps | Greenlight: In this scenario, once the extension disconnects the call if I see the log the call is still ongoing... |
| 17:09.36 | Greenlight | gaps: So until that call is hungup, there will be at least two active channels for it |
| 17:10.30 | Greenlight | So all THREE parties have hungup, and you still have active channels s? |
| 17:10.34 | [TK]D-Fender | esperegu: It is clearly a fork at best |
| 17:11.17 | gaps | Greenlight: actually I disconnected from my extension end... so the transferred numbers are external... one is the incoming call and the other one is Toll Free... |
| 17:11.38 | gaps | Greenlight: basically the billing minutes is calculated with my VoIP trunk... |
| 17:11.56 | [TK]D-Fender | esperegu: first they are set as "nat=yes" this is wrong, and it should be "no" |
| 17:11.56 | gaps | Greenlight: for the transferred call until those party hangups |
| 17:12.20 | Greenlight | Ok |
| 17:12.27 | Greenlight | Still not seeing the problem |
| 17:12.37 | drmessano | [TK]D-Fender, http://goo.gl/Al3PeL |
| 17:12.46 | Greenlight | You call party A, and then transfer to party C |
| 17:12.49 | [TK]D-Fender | esperegu: pastebin another call with "sip set debug on". We need to confirm what the inbound cal llooks like. |
| 17:12.55 | gaps | Greenlight: yes |
| 17:13.02 | gaps | Greenlight: sorry |
| 17:13.06 | Greenlight | So, until that call ends, the channels will remain. |
| 17:13.17 | gaps | Greenlight: actually A is an incoming call |
| 17:13.33 | gaps | Greenlight: myself am B, and transferring the call to C |
| 17:13.40 | Greenlight | So, if A is incoming, and C is "free", why you ask about billing ? |
| 17:14.05 | gaps | Greenlight: after I transfer the call to C, A and C are connected.... |
| 17:14.11 | Greenlight | Yes... |
| 17:14.35 | gaps | Greenlight: whats happening is until C or A disconnects the billing seconds is calculated in my outgoing minutes... |
| 17:14.47 | *** join/#asterisk anonymouz666 (~anonymouz@187.76.181.102) |
| 17:14.56 | Greenlight | But you've called a FREE NUMBER |
| 17:15.08 | esperegu | [TK]D-Fender: http://dpaste.com/1482557/ |
| 17:15.09 | Greenlight | And the call is still in progress |
| 17:15.37 | gaps | Greenlight: sometimes it might be Toll Free and some times its a DID number too |
| 17:15.54 | gaps | Greenlight: actually how it supposed to be? |
| 17:15.54 | Greenlight | Ok, in which case you still have an active call, and so get billed. |
| 17:15.58 | [TK]D-Fender | esperegu: ping sip.cheapconnect.net |
| 17:16.04 | [TK]D-Fender | esperegu: from your server's CLI |
| 17:16.16 | gaps | Greenlight: in both cases toll free and in DID as well... |
| 17:16.33 | esperegu | # ping sip.cheapconnect.net |
| 17:16.33 | esperegu | PING sip.cheapconnect.net (78.40.244.252) 56(84) bytes of data. |
| 17:16.33 | esperegu | 64 bytes from sip01.mtsip.nl (78.40.244.252): icmp_seq=1 ttl=54 time=16.7 ms |
| 17:16.35 | Greenlight | gaps: No, obviosuly, you should not get billed for a toll free number... |
| 17:16.59 | [TK]D-Fender | change the "host" line to jsut the IP |
| 17:17.06 | gaps | Greenlight: so if it is DID number then its fine? |
| 17:17.28 | Greenlight | What do you mean? What exactly is the problem you're having; or how are things not matching your expectations ? |
| 17:17.32 | [TK]D-Fender | esperegu: then apply yoru changes retest and pastebin upon failure |
| 17:18.11 | gaps | Greenlight: actually my question is once I do the transfer the call is out of my control correct? |
| 17:18.33 | Greenlight | gaps: No |
| 17:18.36 | esperegu | [TK]D-Fender: \o/ .... u ta bomb! |
| 17:19.10 | esperegu | [TK]D-Fender: what is the issue? |
| 17:19.18 | esperegu | [TK]D-Fender: so I can tell the provider. |
| 17:19.21 | [TK]D-Fender | esperegu: reverse DNS lookup was failing on that alternate resolving name we saw in the ping |
| 17:19.38 | gaps | Greenlight: so in which scenario the call is out of control... ? in terms of transfer |
| 17:19.50 | esperegu | [TK]D-Fender: thats not fixable I suppose when they have multiple machines ? |
| 17:19.52 | [TK]D-Fender | esperegu: which isn't entirely their problem |
| 17:20.01 | [TK]D-Fender | esperegu: Things get complicated then |
| 17:20.33 | esperegu | [TK]D-Fender: another way to solve it on my side? |
| 17:21.24 | [TK]D-Fender | esperegu: change how you process your inbound calls. |
| 17:21.35 | [TK]D-Fender | esperegu: This is GUI stuff... you are not in control of your full system. |
| 17:21.42 | Greenlight | gaps: YOu mean the transfer happens at the remote side? |
| 17:23.23 | gaps | Greenlight: o its happening in my end i mean Asterisk end |
| 17:24.35 | Greenlight | gaps: Nearly always anything like this, the call is still going to be going through your Asterisk box. There are ways where you can do remote "transfers" but they're generally not supported by ITSPs |
| 17:24.59 | esperegu | hmmm. seems I get no incomming voice yet. maybe that nat setting? |
| 17:25.15 | Greenlight | As far as your ITSP is concerned, there are two calls going to/from you. If you've connected them, after a transfer, that's your concern, not theirs. |
| 17:25.20 | [TK]D-Fender | esperegu: easily believable |
| 17:27.43 | esperegu | [TK]D-Fender: the softphone on my server works but the cisco not. (no inbound sound) |
| 17:28.46 | *** join/#asterisk SushiB (~Thunderbi@200.77.217.106) |
| 17:29.16 | [TK]D-Fender | esperegu: You need to look at the calls with SIP debug... |
| 17:29.37 | gaps | Greenlight: so the billing will calculate from us even though we disconnect from our end.. thats what you mean right? |
| 17:29.53 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 17:30.08 | Greenlight | gaps: Yes |
| 17:30.21 | gaps | Greenlight: how about blind transfer... ? |
| 17:30.24 | Greenlight | And, you are still connected to them |
| 17:30.31 | gaps | Greenlight: ok |
| 17:30.52 | Greenlight | Party A <--> Asterisk. Party C <--> Asterisk. |
| 17:30.57 | Greenlight | That's what your ITSP sees. |
| 17:32.03 | gaps | Greenlight: once Asterisk transfer it will be A <---> C right? |
| 17:32.21 | Greenlight | No. |
| 17:32.28 | WIMPy | Only *IF* it transfers the call. |
| 17:32.42 | WIMPy | Which it probably doesn't (can't) do. |
| 17:32.49 | Greenlight | SIP 401 Redirect ? |
| 17:33.07 | gaps | Greenlight: so even in blind transfer its the same? |
| 17:33.18 | Greenlight | gaps: Yes. |
| 17:33.26 | gaps | Greenlight: ok |
| 17:33.30 | WIMPy | Greenlight: It *might* work. |
| 17:33.48 | Greenlight | WIMPy: I admire your optimism :) |
| 17:34.01 | drmessano | The millions of times this discussion has come up.. does anyone know of an actual instance where a provider DOES support the transfer and take the remote Asterisk out of the loop? |
| 17:34.11 | drmessano | It all sounds like theory to me |
| 17:34.20 | WIMPy | Oh, and even if it works, you never know for how long. |
| 17:34.27 | Greenlight | drmessano: I have to agree |
| 17:34.45 | Greenlight | The billing side of things just makes it to complex to bother supporting. |
| 17:35.21 | drmessano | I realize we all want to be careful and say "Well, SIP supports it.. the provider COULD support it, but they PROBABLY don't"... but in reality, don't they pretty much "DONT"? |
| 17:35.23 | Greenlight | For example, what if you transfer to a more expensive number than the one they called you on. Who foots that bill? You? Your ITSP? The caller? |
| 17:36.18 | gaps | Greenlight: its add in my bill |
| 17:36.39 | Greenlight | gaps: So, why bother ? |
| 17:36.47 | WIMPy | It's really the only realistic option to bill it as two calls. |
| 17:36.55 | drmessano | Yep |
| 17:37.10 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-psqegghbakqrxmjq) |
| 17:37.11 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 17:37.49 | WIMPy | And as customers might not like that, best to not support transfers in the first place. |
| 17:37.55 | eXcAliBuR | error fetching config... but if i select old config it works |
| 17:38.10 | eXcAliBuR | how can i get dpma to let the phone use the new user? |
| 17:38.36 | eXcAliBuR | i have all auth disabled |
| 17:42.04 | drmessano | I guess my point is that this conversation usually goes around in circles due to the careful wording, but in reality "PROBABLY does not" only provides false hope and even more "Will it go round in circles" action |
| 17:42.36 | drmessano | When someone is clueless, "It may not work" is weeks worth of false hope.. |
| 17:43.07 | Greenlight | drmessano: Yea, you're probably like. Curse our accuracy! |
| 17:43.12 | Greenlight | *probably right |
| 17:43.20 | drmessano | lol |
| 17:43.45 | drmessano | I'm more prone to say "That's not gonna work", providers don't support it.. and be wrong 1 time out of 1000 |
| 17:43.48 | *** join/#asterisk morcao (c1885cb9@gateway/web/freenode/ip.193.136.92.185) |
| 17:44.27 | morcao | hi guys ? |
| 17:44.43 | Greenlight | That's all fine and well until he stumbles over a post regarding a Transfer and SIP 401, and things "ah ha!" |
| 17:45.17 | Greenlight | morcao: Hi |
| 17:45.54 | morcao | I have one question, Is possible communicate between Asterisk and PSTN without Digium ? |
| 17:46.00 | drmessano | Greenlight, that goes back to my original statement.. in 8 years of frequenting this channel, and the hundreds of times its been asked, I dont ever recall it resolving as "ah ha!", but maybe I missed something |
| 17:46.13 | drmessano | Which is why I asked, if ever... |
| 17:46.44 | navaismo | morcao, use an ITSP, a gateway(SIP-FXO) |
| 17:46.52 | Greenlight | Or a non-Digium card. |
| 17:46.53 | morcao | it's free ? |
| 17:47.02 | drmessano | I was wrong once too, but then I found out I was mistaken |
| 17:47.06 | *** join/#asterisk nicknam1232 (5c15e940@gateway/web/freenode/ip.92.21.233.64) |
| 17:47.18 | Greenlight | ^^ |
| 17:47.26 | navaismo | there is no such thing as Free for a PSTN connect |
| 17:48.02 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 17:48.06 | drmessano | It's not possible to Asterisk without Digium. IJS |
| 17:49.37 | morcao | navaismo: then how the current systems that use asterisk make communication? |
| 17:50.07 | Greenlight | ... they pay... |
| 17:50.50 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 17:51.03 | morcao | and what form they use? |
| 17:51.18 | Greenlight | Huh? |
| 17:51.28 | Greenlight | What form of... payment.. ? |
| 17:51.31 | morcao | Digium ? seems to me not |
| 17:51.59 | anonymouz666 | morcao: are you from ptt? |
| 17:52.01 | morcao | which system? |
| 17:52.08 | morcao | yes |
| 17:53.24 | Greenlight | AN ITSP for example |
| 17:53.27 | Greenlight | ~itsp |
| 17:53.27 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 17:55.10 | Greenlight | You could also get a hardware card to connect directly to the PSTN |
| 17:55.25 | Greenlight | ANd there's other ways too |
| 17:55.57 | morcao | I read in some forums that it was possible but the number of PSTN network settings had to be this wrong then? |
| 17:56.12 | Greenlight | WHat ? |
| 17:56.26 | Greenlight | You read that what was possible, exactly ? |
| 17:58.12 | morcao | http://www.youtube.com/watch?v=hS5uUtUZpVI |
| 17:58.36 | Greenlight | [05:55pm] <Greenlight> ANd there's other ways too <-- That's one of these |
| 17:58.37 | morcao | in this case with is possible? |
| 17:59.15 | [TK]D-Fender | morcao: Are you asking if what that video is showing is possible...? |
| 17:59.22 | navaismo | in the video they use a dongle |
| 17:59.28 | morcao | yes |
| 17:59.44 | Greenlight | A dong;e isn't free though. |
| 18:00.39 | [TK]D-Fender | morcao: Do you think the video is a fake? |
| 18:00.52 | navaismo | or free? |
| 18:02.26 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 18:02.30 | Greenlight | Would be quite cool to hook up a RaspPi into like say a remote control car, along with a camera, and control it over the 3G |
| 18:02.45 | navaismo | yes you can |
| 18:03.54 | eXcAliBuR | how do i know if my phones are using UDP or TCP? |
| 18:04.02 | morcao | not seem |
| 18:04.05 | [TK]D-Fender | eXcAliBuR: Look at the call |
| 18:04.14 | Greenlight | morcao: What ? |
| 18:04.18 | rrittgarn | If i make an outgoing call, call gets connected, then I try to blind xfer the call to another internal extension. I get a failed transfer message on the phone that tried to blind xfer. Any thoughts as to why this might be happening? Works with incoming calls, just not outgoing connected calls. |
| 18:04.47 | morcao | I believe it is possible without paid systems |
| 18:04.50 | [TK]D-Fender | rittnot without debug... which we should have before asking. |
| 18:04.51 | Greenlight | Heh seems there's a whole site about it: http://pi-cars.com/ |
| 18:04.56 | [TK]D-Fender | rrittgarn: not without debug... which we should have before asking. |
| 18:04.58 | Greenlight | morcao: Yes, you are correct. |
| 18:07.47 | morcao | then how ITSP are paid, from the settings of the asterisk? |
| 18:08.02 | [TK]D-Fender | morcao: No. |
| 18:08.05 | [TK]D-Fender | morcao: YOU pay them |
| 18:08.20 | Greenlight | imagines credit card details in sip.conf |
| 18:08.21 | [TK]D-Fender | morcao: You give them credit card info to BILL you etc. |
| 18:09.27 | Penguin | Many are prepaid. You put credit on your account and debit it with each call. |
| 18:10.04 | eXcAliBuR | should I use tcp or udp? |
| 18:10.12 | Penguin | Depends on your needs. |
| 18:10.16 | [TK]D-Fender | eXcAliBuR: Yes |
| 18:10.17 | eXcAliBuR | internal network |
| 18:11.22 | morcao | After buying this card would have to do more in asterisk? |
| 18:11.58 | Penguin | What card are you buying? |
| 18:12.41 | [TK]D-Fender | morcao: What "more" are you talking about? You haven't told us anything |
| 18:12.54 | morcao | I have no idea said they needed a bill paid |
| 18:13.08 | Penguin | Pay by PayPal if you'd like. |
| 18:13.21 | Penguin | Pay by bitcoin if you'd like. |
| 18:13.37 | Penguin | Pay by gold bullion if you'd like. |
| 18:13.46 | morcao | what you want me to say? |
| 18:14.45 | [TK]D-Fender | morcao: What card are you talking about? You are having trouble describing complete scenarios to us... |
| 18:14.55 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 18:16.59 | morcao | sorry for my confusion |
| 18:18.30 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 18:18.35 | morcao | so where can I get an account that allows me to make calls to all devices? |
| 18:20.09 | [TK]D-Fender | morcao: Separate you idea of "account" and "devices". An account through some service provider has no relationship to any other devices you have. |
| 18:21.03 | eXcAliBuR | how do i trouble shoot while i'm getting error fetching config from proxy? |
| 18:21.23 | [TK]D-Fender | eXcAliBuR: what is this "fetching config" you're referring to? |
| 18:21.37 | Penguin | morcao: An account with an ITSP allows you to send and receive calls between your VoIP phones and the PSTN. |
| 18:21.50 | eXcAliBuR | it's on my digium phones |
| 18:22.37 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 18:23.10 | morcao | Penguin: yes and how get one ? |
| 18:28.22 | [TK]D-Fender | morcao: go CHOOSE a provider that services the area you want to call at a price you feel is good. |
| 18:28.50 | morcao | but if I report only the asterisk to the PSTN do not have to pay for it? right? |
| 18:30.06 | morcao | I have interest in this area because it is my job end of course not to build a business |
| 18:30.15 | [TK]D-Fender | morcao: No. you PAY for PSTN access |
| 18:31.02 | morcao | could be free was better;) |
| 18:31.42 | [TK]D-Fender | morcao: Who gives you free cell phone service? This stuff isn't free. SOMEBODY is paying for it and it's 99% of time you. |
| 18:31.55 | [TK]D-Fender | or more like 99.999999% |
| 18:33.06 | morcao | thanks for your attention |
| 18:33.08 | eXcAliBuR | HELP!!! HELP!!! HELP |
| 18:33.10 | eXcAliBuR | !!! |
| 18:33.18 | eXcAliBuR | (~._.~) |
| 18:33.43 | morcao | you helped me |
| 18:33.44 | morcao | ;) |
| 18:41.25 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 18:41.29 | *** join/#asterisk af_ (~af@93-43-6-79.ip89.fastwebnet.it) |
| 18:55.58 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
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| 19:05.23 | *** join/#asterisk karl-s (~karl.s@pool-173-51-92-61.lsanca.fios.verizon.net) |
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| 19:27.37 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 19:30.07 | *** join/#asterisk kuruption (kuruption@vato.is.a.big.black.cock.addikt.org) |
| 19:30.19 | karl-s | does anyone know if G.729 passthrough requires directrtpsetup? |
| 19:30.59 | [TK]D-Fender | it doesn't |
| 19:31.32 | karl-s | thx. |
| 19:32.19 | [TK]D-Fender | Passthrough simply means no transcoding. Doesn't amtter if it changes wrappers like IAX>SIP just just oges through your server under the same protocol(since each leg is independent anyway) |
| 19:32.54 | eXcAliBuR | this is making me very grumpy |
| 19:35.55 | eXcAliBuR | [Nov 25 14:33:49] WARNING[1775]: phone_users.c:1016 process_networks: Unknown network option 'udp_ka_interval' |
| 19:36.00 | eXcAliBuR | why.... it's in the manual |
| 19:36.16 | eXcAliBuR | :[ |
| 19:37.48 | navaismo | latest firmware? |
| 19:38.03 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 19:39.36 | eXcAliBuR | yes |
| 19:39.40 | eXcAliBuR | just updated it |
| 19:41.28 | eXcAliBuR | i don't want to call digium |
| 19:41.29 | eXcAliBuR | :( |
| 19:41.32 | eXcAliBuR | they don't like me |
| 19:44.23 | pabelanger | eXcAliBuR, better off calling them for support, we don't have source to see what is the issue |
| 19:44.29 | Penguin | If you called them and said, "HELP!!! HELP!!! HELP!!!" like you did here, I can understand why. |
| 19:48.50 | *** join/#asterisk slicknick5181 (cfff7249@gateway/web/freenode/ip.207.255.114.73) |
| 19:49.10 | eXcAliBuR | I should throw them all in the garabe |
| 19:49.17 | eXcAliBuR | garbage* |
| 19:50.00 | slicknick5181 | Asterisk 11 I would like to be able to barge on a channel DAHDI and SIP and be able to listen and speak to both parties |
| 19:50.48 | *** join/#asterisk slicknick5181 (cfff7249@gateway/web/freenode/ip.207.255.114.73) |
| 19:51.13 | slicknick5181 | Sorry I accidently closed if anybody answered my inquiry |
| 19:52.57 | Penguin | ChanSpy() does that. |
| 19:53.20 | [TK]D-Fender | slicknick5181: Bridge() <- |
| 19:53.33 | [TK]D-Fender | Chanspy only whispers to one end... |
| 19:54.06 | Penguin | <PROTECTED> |
| 19:54.06 | Penguin | <PROTECTED> |
| 19:54.18 | [TK]D-Fender | OOOH, new options! |
| 19:54.23 | Penguin | new in 1.8 |
| 19:54.24 | [TK]D-Fender | I'll take that! |
| 19:55.29 | slicknick5181 | I would like to be able to dial 15005 for example and be connected to a specific extension if on an active call |
| 19:55.48 | Penguin | You don't connect to extensions. Extensions execute commands. |
| 19:56.23 | slicknick5181 | Well correct. I mean a specific Device |
| 19:56.38 | Penguin | If extension 15005 executes ChanSpy(SIP/device), then you'll spy on that device. |
| 19:56.42 | karl-s | slicknick5181, you are going to have to write custom dialplan that uses the function DEVICE_STATE |
| 19:56.43 | slicknick5181 | channel DAHDI 1 Is what I want to bridge with |
| 19:58.05 | [TK]D-Fender | then go for it |
| 19:58.58 | Penguin | You don't need to bridge with it. Just spy on it. |
| 20:00.38 | slicknick5181 | Well I was reading a bit on Chanspy and it appears I have the options there I'm just not sure where to put the options |
| 20:00.52 | Penguin | core show application ChanSpy |
| 20:01.22 | *** join/#asterisk fonewiz (~fonewiz@cpe-173-174-255-129.satx.res.rr.com) |
| 20:02.20 | slicknick5181 | gives 5 categories but "not available" sows underneath |
| 20:03.09 | Vendigroth | Hi! Does anyone know if there is a difference between doing Answer() in the dialplan or through AGI? I seem to be getting different behavior. I've tried asking about this here before, but no one seems to have any ideas. |
| 20:07.59 | Vendigroth | <PROTECTED> |
| 20:08.24 | Vendigroth | ^have it built with no xmldoc. |
| 20:08.31 | navaismo | Vendigroth, what is the different behavior? |
| 20:09.44 | slicknick5181 | Thatnk you guys very much!! |
| 20:09.46 | *** part/#asterisk slicknick5181 (cfff7249@gateway/web/freenode/ip.207.255.114.73) |
| 20:12.10 | Vendigroth | if I first go through AGI to decide if i want the call answered, and then do AGI-Answer I get a single "incompatible voice frame" notice for slin. (I know its not a big deal). But if the Answer() is in dialplan I get no such notice. |
| 20:12.47 | [TK]D-Fender | Vendigroth: Pastebin the 2 calls for comparison |
| 20:14.38 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 20:17.38 | Vendigroth | with Notice: http://pastebin.com/cGLaB9m9 Without:http://pastebin.com/f6nKWyx9 I have pri/AGI/verbose/debug on on those. can make some with less options turned on. |
| 20:27.03 | navaismo | weird |
| 20:27.19 | [TK]D-Fender | Vendigroth: -- Executing [5852879987@inbound_call:2] EAAGI("DAHDI/i1/-f", "/usr/asher/callflows/inbound_mgr.pl") in new stack |
| 20:27.30 | [TK]D-Fender | Vendigroth: first ... what is EAAGI? |
| 20:27.58 | navaismo | weird at that ^ |
| 20:27.58 | [TK]D-Fender | Vendigroth: EAGI is with a voice channel which could be SLIn as oppoed to more "codec based on things being more raw at that point (I might imagine) |
| 20:28.05 | Vendigroth | yeah.. that. Its basically an alias for AGI. for compatability. |
| 20:31.15 | Vendigroth | as far as asterisk is concerned it is ordinary AGI things (enhanced variable in res_agi is 0) |
| 20:33.17 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
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| 21:49.07 | *** join/#asterisk sp3ck (~sp3ck@unaffiliated/sp3ck) |
| 21:52.29 | sp3ck | Hi, i use asterisk with freepbx everything seems to be fine but the wakeup calls |
| 21:52.47 | sp3ck | the log file says : file.c:663 ast_openstream_full: File wakeup-call does not exist in any format |
| 21:53.08 | sp3ck | the .gsm files, for wakeup calls, are there |
| 21:53.15 | sp3ck | what can i do to fix this |
| 21:53.44 | sp3ck | (permissions are 661 asterisk:asterisk) |
| 21:53.49 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 21:53.59 | sp3ck | ? |
| 21:54.55 | Penguin | How about the permissions on the directory above the file? |
| 21:55.14 | sp3ck | give me a sec |
| 21:56.03 | sp3ck | i dont think there is a problem because conference rooms announcements work well...so the files are in the same directory |
| 21:56.51 | navaismo | same language folder? |
| 21:56.59 | sp3ck | drwxrwxr-x 7 asterisk asterisk 4096 Nov 25 23:51 asterisk |
| 21:57.00 | sp3ck | drwxrwxr-x 5 asterisk asterisk 53248 Aug 9 18:53 sounds |
| 21:57.54 | sp3ck | language folder? ^^ |
| 21:58.23 | Penguin | like asterisk/sounds/en/ |
| 21:58.47 | sp3ck | there is no "en" folder in my asterisk/sounds folder |
| 21:59.33 | sp3ck | my asterisk verion is: Asterisk 1.8.10.1~dfsg-1ubuntu1 built by buildd @ vernadsky on a i686 running Linux on 2012-04-24 12:44:43 UTC |
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| 22:01.08 | sp3ck | is there a way to see which file path asterisk tries to reach so i can start somehow debug this issue? |
| 22:02.16 | karl-s | uncomment ;full in /etc/asterisk/logger.conf |
| 22:02.29 | karl-s | then check the full log in /var/log/asterisk/full |
| 22:02.34 | karl-s | after doing a reload |
| 22:03.00 | Penguin | logger reload, that is. |
| 22:03.32 | karl-s | or reboot your server |
| 22:03.35 | karl-s | they all work :) |
| 22:03.40 | Penguin | That's ridiculous. |
| 22:03.54 | karl-s | i always like the shotgun approach |
| 22:04.40 | anonymouz666 | I am lucky then because I don't use your telephony services :-) |
| 22:04.41 | Penguin | I could crash my car into a building so the insurance company will buy me a new one, too, but I'd rather just trade it in. |
| 22:07.13 | sp3ck | ok, i had full enabled already from fpbx |
| 22:07.33 | anonymouz666 | Penguin: Please don't crash |
| 22:07.34 | sp3ck | shotgun approach? hahahah |
| 22:09.03 | Penguin | anonymouz666: I was only trying to illustrate that merely getting to the end result isn't the important part, but how you get the result matters. |
| 22:09.20 | sp3ck | http://pastebin.com/rKT62frT |
| 22:10.14 | Penguin | It looks like it is only looking for ulaw and h264. |
| 22:11.19 | sp3ck | wakeup-call module looks for ulaw and h264 and conference module looks for gsm? |
| 22:12.02 | sp3ck | let's say wakeup mod wants ulaw .. how can i make it read the gsm? |
| 22:15.15 | Penguin | I'm not sure about that, but you could use asterisk's file convertor to make the ulaw files. |
| 22:15.41 | Penguin | file convert in-file.gsm out-file.ulaw |
| 22:25.59 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 22:28.09 | sp3ck | i've tried sox -V wakeup-call.gsm -r 8000 -c 1 -t ul -w wakeup-call.ulaw but i'm doing something wrong |
| 22:28.16 | Penguin | (1615.40) <Penguin> file convert in-file.gsm out-file.ulaw |
| 22:28.18 | Penguin | ^ |
| 22:28.21 | sp3ck | is this the correct syntax? |
| 22:28.24 | *** join/#asterisk ywcahello (3289886a@gateway/web/freenode/ip.50.137.136.106) |
| 22:28.30 | Penguin | In the asterisk CLI... |
| 22:28.53 | Penguin | I said "asterisk's file convertor" |
| 22:29.45 | Penguin | Although I can't see why asterisk wouldn't simply transcode between ulaw and gsm in the case of your files being gsm and the call being ulaw. That seems like pretty basic functionality. |
| 22:30.06 | *** join/#asterisk dxd828 (~dxd828@host-92-24-127-29.ppp.as43234.net) |
| 22:30.18 | ywcahello | Experiencing an issue with Asterisk 1.8: In the dialplan, running CALLERID(all) = "Skype Contact <skype-name>" results in the CALLERID(num) being set to "skypename". The dashes are removed. Makes sense for a numeric DID but not for a Skype name. Does anyone know of a workaround? |
| 22:30.55 | Penguin | DId you try correctly setting the callerid information? |
| 22:31.12 | Penguin | CALLERID(all)=Your Name <your number> |
| 22:31.16 | Penguin | No quotes anywhere. |
| 22:31.35 | ywcahello | oh, yes, I'm not using quotes. I mistyped in my question |
| 22:31.50 | ywcahello | I'm seeing similar behavior with CALLERID(num)=skype-name |
| 22:32.09 | Penguin | I've never tried to put a hyphen in the number field, so I don't know. |
| 22:32.43 | ywcahello | The issue seems to be limited to the CALLERID function. When calls come in over a Skype channel, dashes aren't removed. |
| 22:33.03 | ywcahello | Hmm, trying to figure out the best outlet to get a solution to this. Opening a bug on the bug tracker doesn't seem appropriate. |
| 22:35.17 | ywcahello | Just found this link: https://issues.asterisk.org/jira/browse/ASTERISK-16528 . Looks like I'll need to find a workaround solution or patch it myself. |
| 22:35.18 | LieutPants | [ASTERISK-16528] [Status: Closed] Enable to use '-' sign in CallerID - https://issues.asterisk.org/jira/browse/ASTERISK-16528 |
| 22:35.35 | ywcahello | haha, funny timing |
| 22:35.54 | Penguin | Funny timing of what? |
| 22:37.28 | ywcahello | LieutPants posted the same link I did within a second or so |
| 22:37.43 | Penguin | That's a response, not a coincidence. |
| 22:39.27 | Penguin | ASTERISK-1041 |
| 22:39.28 | LieutPants | [ASTERISK-1041] [Status: Closed] [patch] video format description bug into an INVITE message - https://issues.asterisk.org/jira/browse/ASTERISK-1041 |
| 22:39.31 | Penguin | See? |
| 22:39.31 | ywcahello | Oh, ok. It's been about 15 years since I hung out on IRC. |
| 22:39.40 | ywcahello | got it |
| 22:39.49 | navaismo | ASTERISK-22897 |
| 22:39.50 | LieutPants | [ASTERISK-22897] [Status: Open] WebSocket connection from JsSIP or SIPML5 generate a segmentation fault(core dumped) - https://issues.asterisk.org/jira/browse/ASTERISK-22897 |
| 22:39.52 | sp3ck | Penguin, I found it... (that's why i always hated guis). The sounds folder is /usr/share/asterisk/sounds/en and there wasn't any recordings for wakeup-calls oin there so I copied them from /var/lib/asterisk/sounds |
| 22:40.21 | Penguin | That was mentioned. |
| 22:40.30 | navaismo | o/ |
| 22:43.40 | sp3ck | Penguin, I thought /var/lib/asterisk/sounds was the folder of the "running" recordings... :S |
| 22:43.49 | sp3ck | Penguin, navaismo karl-s thanx for help. |
| 22:44.27 | Penguin | When you install from packages, the paths seem to be different. |
| 22:45.58 | sp3ck | yeah... |
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