| 00:07.55 | raspberrypifan | i thought asterisk was a b2bua but freeswitch was a switch |
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| 01:26.46 | Apteryx | Hi there! I recently installed Asterisk using 'apt-get install asterisk' on my Ubuntu laptop. At first I didn't have any problem with sound, but I had not reboot yet. Since then, there was a Linux kernel upgrade, and I changed the configuration of Asterisk a lot, setting autoload=no in modules.conf and only loading what I needed. Today I rebooted, and the sound is gone. Asterisk seems to be starting before pulseaudio and locking the sound devi |
| 01:27.37 | ChannelZ | So does that mean you're using ALSA with the console driver or something? |
| 01:28.25 | Apteryx | ChannelZ: In my modules.conf I'm only loading up chan_alsa.so |
| 01:29.07 | ChannelZ | so if you restart asterisk after the machine boots, it works? |
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| 01:30.30 | Apteryx | I just tried stopping Asterisk, and then doing 'sudo alsa force-reload'. Then I had sound. After, I restarted Asterisk and the sound was blocked again (although at least I could still see my sound card in PulseAudio and not a Dummy Output) |
| 01:31.47 | ChannelZ | Hmm. It's been a very long time since I used chan_alsa |
| 01:32.05 | Apteryx | ChannelZ: What are you using? chan_oss ? |
| 01:32.54 | ChannelZ | No I don't use any local audio.. |
| 01:33.26 | Apteryx | Ok! |
| 01:34.21 | Apteryx | Maybe I screwed up the debian default config. I will try apt-get remove --purge asterisk && apt-get install asterisk (after backuping /etc/asterisk/ ;) and see how the sound is. |
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| 01:37.21 | [TK]D-Fender | no need |
| 01:37.31 | [TK]D-Fender | just stop loading chann_alsa / chan_oss |
| 01:37.44 | [TK]D-Fender | "noload => chan_also.so" <- add to modules.conf |
| 01:39.47 | Apteryx | Ok! I'm not sure why I thought I needed it! Thanks. |
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| 01:40.03 | ChannelZ | Oh. I guess it depends on the interpretation of the question.. is asterisk killing your system's sound, or were you trying to use alsa in asterisk? |
| 01:40.30 | ChannelZ | I was reading it as the latter but it sounds like you meant the former. |
| 01:41.03 | Apteryx | Asterisk is killing my system sound, because I'm still a newb and put load chan_alsa.so in modules.conf with no need for it ;) |
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| 02:16.04 | Apteryx | Is it OK to use some global variables to implement "ringing groups"? For exemple, I have: exten => 14387889919,1,Dial(${MAXIM_DEVICES}&${YUKI_DEVICES}) where MAXIM_DEVICES=SIP/maxim-VostroV130&SIP/maxim-iPhone and YUKI_DEVICES=SIP/yuki-iPhone&SIP/yuki-iPad |
| 02:17.57 | Apteryx | All the devices ring, but I get "Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)" and one way audio. |
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| 02:30.25 | [TK]D-Fender | Apteryx: Clearly not all of them ring |
| 02:30.38 | [TK]D-Fender | Apteryx: because that is telling you that * is not dialing at least one of them . |
| 02:31.05 | [TK]D-Fender | Apteryx: And one-way audio is a networking issue, most commonly due to things like NAT mis-configuration |
| 02:31.16 | [TK]D-Fender | Apteryx: for which you have several settings to make |
| 02:31.21 | [TK]D-Fender | (if applicable) |
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| 02:44.31 | Apteryx | [TK]D-Fender: Ok, I had the one-way audio fixed yesterday, using nat=yes, but I changed to nat=force_rport,comedia today. Will see if it changes anything! Thanks, again. I owe you a couple beers already. |
| 02:45.26 | [TK]D-Fender | those are PART of the settings. There are other for your peers, and others for [general] if your server itself is NAT'd |
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| 02:54.21 | Apteryx | I had commented out the nat=force_rport,comedia for a previous test and forgot... It's working great now :) |
| 02:56.16 | Apteryx | One thing I fail to understand is under which circumnstances I should port forward the RTP ports to my Asterisk server. For now they are not port forwarded and I can receive calls fine. I can call as well. |
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| 03:14.54 | *** join/#asterisk infobot (ibot@rikers.org) |
| 03:14.54 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.10.2 (2014/06/13), 1.8.28.2 (2014/06/13); Standard: Asterisk 12.3.2 (2014/06/13); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 03:38.22 | TazzNZ | w00t - our platform crossed over 750k in calls :D |
| 03:38.27 | TazzNZ | now for 1 mil |
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| 05:24.23 | snadge | TazzNZ, in how long? :P |
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| 06:23.14 | gavimobile | can tftp be set up with subdirectories for the phones to use xxx.xxx.xxx.xxx/model |
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| 06:29.29 | jameswf | Can your phone be set to a path? |
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| 06:31.16 | gavimobile | I don't know jameswf, the manual didn't seem to talk about tftp |
| 06:31.20 | gavimobile | this is a polycom 330 |
| 06:32.12 | gavimobile | also if anyone knows if I need to add tftp:// |
| 06:32.17 | jameswf | gavimobile: this will be a phone limitation not a server one. You need to see what your phone supports |
| 06:33.56 | gavimobile | jameswf: thanks |
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| 06:43.17 | *** join/#asterisk jkroon (~jkroon@kantoor.wdns.uls.co.za) |
| 06:46.47 | jkroon | hi guys, given a rather complex scenario where I've got an application (context) that interrogates the executed on channel using Gosub, and then needs to goto one of two other locations, given as arguments to the Gosub, eg Gosub(...(context^exten^prio,context2^exten2^prio)) ... so now I need to convert the ^ chars to , but giving that to REPLACE or STRREPLACE is going to be problematic ... ? |
| 06:52.41 | gavimobile | jameswf: so if a network has 100 phones and there are 20 different types of phones which require provisioning for each boot, what does one do for best practice assuming that some and some aren't supported for tftp subdirectory usage |
| 06:56.03 | jameswf | They use Mac addresaes |
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| 06:59.34 | gavimobile | could you send me a link for more information on what your talking about jameswf |
| 06:59.57 | gavimobile | I don't understand how mac addresses have anything to do with provisioning |
| 07:00.28 | MaliutaLap | gavimobile: depends on if you are building the phone configs on the fly |
| 07:00.36 | MaliutaLap | gavimobile: I have one site where I do that |
| 07:01.03 | gavimobile | MaliutaLap: my pbx is curently running as the tftp server |
| 07:01.15 | gavimobile | keeping things onsite or offsite doesn't make a difference to me |
| 07:03.01 | MaliutaLap | gavimobile: I don't think you understood what I meant. I have Cisco 7960G phones - I need the MAC address to build the SIP/SEP<MAC>.cnf files on the fly :) |
| 07:03.50 | gavimobile | so there is 1 directory on the tftp server (the root) directory with a bunch of *.cnf files |
| 07:03.53 | gavimobile | ? |
| 07:06.10 | jkroon | gavimobile, basically MAC address maps to provider (ie, type of phone, usually). So if you have the MAC address you can (given some rules) deduce the type of the phone. I've taken to feeding those that requests tftp configs a config that just rewrites them to HTTP, where I can take things like UA string into consideration as well, and build configs on the fly (I map mac addresses to extensions and BLF configs, so I generate configs on the fly t |
| 07:06.10 | jkroon | hen using php running behind apache). |
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| 07:06.30 | jkroon | the MAC address of the phone is a crucial bit of information required for provisioning. |
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| 07:07.20 | eirirs | barcode scanning ftw |
| 07:08.13 | jkroon | yea, having one of those in your desk helps :) |
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| 07:08.30 | eirirs | I got two |
| 07:08.44 | eirirs | one old as fucking but it reads barcodes where the new one cannot |
| 07:08.45 | eirirs | lol |
| 07:09.24 | gavimobile | im looking at my .cfg files |
| 07:12.17 | gavimobile | jkroon: you said I can give some rules.. where do I configure these "rules"? |
| 07:13.29 | gavimobile | I understand that the phone requests from the tftp server xxxmacidxxx.cfg |
| 07:13.45 | gavimobile | but doesn't it also need additional files such as sip.ld |
| 07:14.02 | gavimobile | or in the .cfg file I would change it to sp33/sip.ld |
| 07:15.58 | jkroon | sip.ld ... are you using polycom by any chance? |
| 07:16.16 | jkroon | best person I know of to speak about polycom is chainsaw on #gentoo-voip (usually). |
| 07:16.29 | gavimobile | jkroon: some of my phones are polycom |
| 07:16.50 | jkroon | the other files depends on the brand, make and model of the phone. Some of them don't require it at all. |
| 07:17.05 | gavimobile | but I have more than 1 polycom |
| 07:17.19 | jkroon | so mostly unless you're interested in feeding them firmware updates you can get away with only the MAC based file. |
| 07:17.22 | gavimobile | and each version requires different versions of different files |
| 07:17.41 | jkroon | yea, thus why I say speak with chainsaw - he got it down to a tee for polycom. |
| 07:18.08 | gavimobile | jkroon: from all you told me, I understand that I have the root tftp directory with a bunch of .cfg files and that's fine but that still doesn't patch all the holes |
| 07:18.09 | jkroon | i just cloned a lot of his work for my one polycom client, and i will not pretend to understand half of what we've done. |
| 07:18.23 | gavimobile | jkroon: he's not in :-( |
| 07:18.36 | jkroon | let me see if I can find the notes I took. |
| 07:18.38 | gavimobile | maybe there are others in the room that have implemented provisioning servers |
| 07:18.46 | gavimobile | chanel* |
| 07:20.17 | jkroon | located, let me pastebin quickly. |
| 07:20.48 | jkroon | gavimobile, http://pastie.org/9366781 |
| 07:21.40 | jkroon | IIRC I still did some other magic after that, but that should at least get the firmware portion sorted out, and it sounds like you already have the MAC specific portion going - which was the other issue for me. the XML consumed by polycom really isn't any kind of strict XML ... |
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| 07:30.48 | gavimobile | jkroon: I appreciate the time you took out for me |
| 07:31.02 | gavimobile | reviewing your notes, im not sure what it has to do with me though |
| 07:31.13 | gavimobile | the only thing I can think of is when you mentioned split |
| 07:31.47 | gavimobile | so if I understood correctly, after the bootloader has been installed, I only need a split version which contains minimal files for the phone to boot |
| 07:32.09 | gavimobile | if these files are not existing on the tftp server, the polycom phones will take forever to boot |
| 07:32.44 | jkroon | that's how I managed to get things set up so that the sip.ld files etc loaded correctly no matter what the polycom phone model (and hardware revision) was. |
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| 07:33.34 | jkroon | so first reboot after that took forever to flash etc ... once that was done everything was much faster. |
| 07:44.40 | gavimobile | next question |
| 07:44.52 | gavimobile | can asterisk send vm to multiple emails? |
| 07:45.01 | gavimobile | according to the example in the definitive guide, the answer is yes |
| 07:45.15 | gavimobile | but I just added a third email, and I don't see my mail logs sending to the third email |
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| 09:42.30 | tparcina | What does asterisk-fax actually do? |
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| 11:53.59 | cusco | hey folks |
| 11:54.42 | cusco | I'm looking for a way to set a PRI span down, without going there phisically and detatch the cable .. its on a live system so I would like for other spans to keep on working |
| 11:54.47 | cusco | how can I do this? |
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| 12:08.49 | *** join/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 12:08.56 | Zogot | hey, realtime doesn't support hints? |
| 12:11.08 | *** join/#asterisk wasanzy (~wasanzy@197.159.129.10) |
| 12:11.11 | wasanzy | hello |
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| 12:11.49 | wasanzy | does the application ControlPlayback comes with Asterisk 11.5? how do I know it is installed? |
| 12:12.14 | Chainsaw | core show applications |
| 12:12.33 | cusco | Zogot: it does here |
| 12:12.34 | wasanzy | thank you |
| 12:12.45 | Chainsaw | And indeed, I have it here. core show application ControlPlayback |
| 12:13.01 | Chainsaw | That's 11.10.2 though. I really doubt 11.5 is still secure. |
| 12:14.29 | Zogot | cusco: you are using hints with realtime currently? |
| 12:14.42 | cusco | Zogot: by realtime you mean queue members? yes |
| 12:15.00 | Zogot | nah, realtime as in Asterisk Realtime: https://wiki.asterisk.org/wiki/display/AST/Database+Support+Configuration |
| 12:15.03 | wasanzy | Chainsaw: Thanks, I also have it |
| 12:15.07 | cusco | o.O |
| 12:15.43 | cusco | that is what I said |
| 12:15.50 | cusco | we only have parts in realtime |
| 12:15.57 | cusco | dialplan is not in realtime for example |
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| 12:16.24 | cusco | so we have the family queue members in realtime, that is what I meant |
| 12:17.20 | Zogot | cusco: ah i see, my apologies. i see theres also a thread about it from the mailing list: https://www.mail-archive.com/asterisk-users@lists.digium.com/msg233770.html |
| 12:17.21 | cusco | so..... any way to set a PRI span down, without going there and physically disconect the cable |
| 12:17.24 | cusco | ? |
| 12:18.42 | Stefan27 | in dialplans patterns, how can i include "-" in a character set i expected [\-A][\-A] to match "-A" but it doesnt |
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| 12:26.06 | WIMPy | cusco: What do you want to happen? |
| 12:27.06 | WIMPy | Stefan27: If you want to match "-A" that should be just "-A". |
| 12:27.46 | Stefan27 | no that was just debugging i want [0-9a-zA-Z*#+.%_\-] |
| 12:28.05 | Stefan27 | where \- is supposed to mean - since - with [] has special meaning |
| 12:28.10 | wasanzy | I have this: exten => 1234,n,ControlPlayback(/var/lib/asterisk/sounds/main_menu_en.alaw,4000,*,#,1,0) |
| 12:28.34 | wasanzy | and is giving me this error: [2014-07-08 12:27:56.834] WARNING[16908][C-00000020]: file.c:701 ast_openstream_full: File /var/lib/asterisk/sounds/main_menu_en.alaw does not exist in any format |
| 12:29.06 | file | don't put ".alaw" at the end |
| 12:29.19 | WIMPy | Stefan27: Ok. But you want less than everything? |
| 12:29.23 | wasanzy | oh ok |
| 12:29.27 | Stefan27 | yeah |
| 12:30.04 | WIMPy | Stefan27: Have you tried - as the first inside the []? |
| 12:31.17 | Stefan27 | yeah tried it now, i thought about trying ,-. within the [] but that will include , - and . probably |
| 12:31.23 | Stefan27 | and i dont want , |
| 12:31.39 | Stefan27 | since ascii for , - . is 44 45 46 ? |
| 12:31.54 | WIMPy | Sounds like an idea. |
| 12:32.18 | WIMPy | Argh. My tree is throwing leaves at me. |
| 12:32.21 | Stefan27 | wiki page said \ is supposed to escape special meanings, maybe it's a small glitch |
| 12:33.01 | Stefan27 | The only characters with special meaning within a set are the '-' character, to define a range between two characters, the '\' character to escape a special character available within a set, and |
| 12:33.02 | Stefan27 | the ']' character which closes the set. The treatment of the '\' character in pattern matching is somewhat haphazard and may not escape any special character meaning correctly. |
| 12:33.03 | WIMPy | I think, that a case where voip-info might be usefull. |
| 12:33.04 | Stefan27 | Said the documentation |
| 12:33.54 | WIMPy | Sounds like noone thought of letting - be escaped. |
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| 12:37.30 | wdoekes | Stefan27 and WIMPy: - has issues |
| 12:37.34 | wdoekes | it is autotrimmed |
| 12:37.55 | wdoekes | +1-23-45 == +12345 |
| 12:38.26 | Stefan27 | i see |
| 12:38.50 | wasanzy | file: The controlplayback doesn't seem to be working. I tried with zoiper but I didn't see the sound skipped. Do I need to do something more? |
| 12:38.51 | Stefan27 | trying to make old dialplan code work on new asterisk, maybe I dont need - |
| 12:39.24 | wdoekes | try to avoid it. there have been fixes to the autotrimming in recent years too. so the dialplan may not work as expected |
| 12:39.38 | WIMPy | wdoekes: OIh, right. That's an additional issue. And a very annying one. |
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| 12:43.02 | Stefan27 | can you make variables within patternmatching so that with Y := [0-9a-zA-Z*#+._%] then anywhere exten => _0YX matches '0#5' |
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| 12:43.21 | Stefan27 | within dialplan* |
| 12:43.28 | WIMPy | yes |
| 12:43.54 | WIMPy | Well, you have to exten => ${var} as usual. |
| 12:44.28 | WIMPy | But that doesnt make then dynamic, off course. The variable is expanded when the dialplan is loaded. |
| 12:44.41 | Stefan27 | that's good |
| 12:45.00 | Stefan27 | Y := [0-9a-zA-Z*#+._%] should be a global constant |
| 12:45.07 | Stefan27 | just for readability |
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| 12:47.14 | wasanzy | Hello someone help me with this: exten => 1234,n,ControlPlayback(/var/lib/asterisk/sounds/main_menu_en,3000,*,#,1,0 it is not forwarding the sound, do I meed to press something before? |
| 12:48.18 | WIMPy | Syntax. Playback does only have two arguments. |
| 12:48.59 | WIMPy | Or 1-2 to be exact. |
| 12:49.33 | WIMPy | What do you want the things you added to do? |
| 12:50.21 | WIMPy | OH. I should wake up first. Completeley ignored the "Control". |
| 12:51.00 | wasanzy | WIMPy: So what I have is right? |
| 12:51.30 | WIMPy | Yeah, looks ok. |
| 12:52.01 | WIMPy | Turn up verbose and podssibly debug to see if it has anythign to moan about. |
| 12:52.01 | wasanzy | WIMPy: ok, am using zoiper but the sounds didn't get forwarded |
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| 13:00.05 | wasanzy | WIMPy: the sound plays alright but I it didn't get forwarded |
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| 13:01.18 | WIMPy | Does your DTMF work otherwise? |
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| 13:19.14 | Stefan27 | ok i changed pbx.cs function: _extension_match_core to mimic the behavior of "X" pattern matching just adding a case 'Y': if (((*data < '0') || (*data > '9')) && ((*data < 'a') || (*data > 'z')) && (*data != '-')) ...etc... (in the large switch-block) it seems to work but i didn't change code anywhere else. might this lead to unexpected errors? |
| 13:20.30 | Stefan27 | hope it wont :) |
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| 13:22.07 | Stefan27 | but the special matching chars 'X', 'Z' etc does seem to occur at other places in the source code, where I have not added my new 'Y' |
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| 13:27.18 | Stefan27 | guess ill have to just wait and see unless i can reach the actual coder of the pattern-matching |
| 13:27.54 | WIMPy | Try to ask that one in #asterisk-dev |
| 13:28.30 | Stefan27 | thanks, i probably will in a few minutes |
| 13:29.06 | wasanzy | WIMPy: what is DTMF? |
| 13:30.45 | WIMPy | ~dtmf |
| 13:30.45 | infobot | DTMF: Dual Tone Multi-Frequency. The technical term describing Touch Tone dialing. Basically the combining of two tones, one low frequency and one high frequency. |
| 13:31.03 | wasanzy | oh ok |
| 13:33.54 | WIMPy | Does anyone her by chance know how to remove the front bezel of a hipath 3800? |
| 13:33.59 | wasanzy | is only the * key that is working and that one is fast forwarding |
| 13:37.07 | _Corey_ | WIMPy: I'm told you can pop a screwdriver in a slot in the bottom and it will come off |
| 13:38.30 | WIMPy | No it's obvious when the lid is off. But I first thought you need to remove the front to get the lid off. |
| 13:39.29 | _Corey_ | WIMPy: I was told it's more of an RF shield than a bezel but the guy who knows just left my office. .. ;( |
| 13:39.54 | WIMPy | No. That's plain plastic. |
| 13:41.08 | _Corey_ | WIMPy: Share a photo if you want. I can call him back in... (He used to work for Siemens) |
| 13:41.47 | WIMPy | Thanks, but it's all done. |
| 13:42.04 | WIMPy | I just had to open the case as there was something moving around somewhere. |
| 13:42.28 | WIMPy | But it was just some plastic part. |
| 13:42.49 | _Corey_ | np |
| 13:43.16 | WIMPy | Now I can prepare for the fun part: See how DECT works on these things. |
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| 14:55.41 | c|oneman | what are some tools for troubleshooting 1-way audio or call quality problems? |
| 14:55.44 | c|oneman | on the client side |
| 14:56.19 | [TK]D-Fender | 1-way audio = networking issue, most commonly due to misconfiguration where NAT is involved |
| 14:56.42 | c|oneman | yeah. In this case I think it's a headset issue or something else |
| 14:57.00 | c|oneman | since it's just one person, and they are using our VPN which works fine for everyone else |
| 14:57.59 | [TK]D-Fender | Could always be the client itself |
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| 15:23.49 | cusco | WIMPy: I would like to simulate disconnecting a PRI cable.. telco says that they will forward calls to a different location and I want to test it |
| 15:23.55 | cusco | but I'm not near the equipment |
| 15:24.09 | *** join/#asterisk omenSP (32490489@gateway/web/freenode/ip.50.73.4.137) |
| 15:24.20 | omenSP | Hey there! |
| 15:24.26 | cusco | so I would like to set 2 spans (a single PRI card) down, without affecting other ongoing calls on other spans |
| 15:25.07 | omenSP | The small business I am working for is interested in switching their analog PBX system over to Asterisk.. how would I get started? |
| 15:25.37 | Qwell | ~book |
| 15:25.38 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:26.19 | omenSP | Thank you. |
| 15:27.53 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:28.11 | WIMPy | cusco: If you remove them from the configuration, that should be enough. Unloading the driver certainly helps. |
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| 15:31.19 | omenSP | The book doesn't go into very much detail about converting from an old system to Asterisk and what hardware I need. |
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| 15:38.38 | omenSP | Do I need a PCI-E card that will take our trunk lines? Is that it? |
| 15:40.18 | wasanzy | how do I run this command in Asterisk to determine the length of a wav sound? sox 0_10s_something_8k.wav -n stat | grep -i length |
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| 15:44.23 | WIMPy | omenSP: You can use cards, gateways or switch to an ITSP. |
| 15:45.01 | omenSP | So all I really need is a server with the PCI-E telephony card and I can hook up a phone on the network and it'll work? |
| 15:45.52 | WIMPy | No. that's one option. |
| 15:46.09 | WIMPy | You don't need it. |
| 15:48.35 | omenSP | But that will work? |
| 15:49.10 | omenSP | I thought the system would be a lot more complicated. We currently have a PBX and a phone breakout that routes phone lines to stations. |
| 15:49.26 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 15:50.57 | WIMPy | Well, having a mess of wires is quite easy compared to Asterisk and VOIP phone configuration. |
| 15:51.00 | [TK]D-Fender | omenSP: that depends. |
| 15:51.27 | [TK]D-Fender | omenSP: What is this "phone breakout"? What type & model of phones are attached to that? What kind of physical lines do you currently have? |
| 15:52.04 | omenSP | The phone breakout thing (sorry for my lack of knowledge on the subject), the punchdown block. |
| 15:52.40 | [TK]D-Fender | omenSP: What kind of phones? model? How many? |
| 15:52.47 | [TK]D-Fender | omenSP: What kind fo lines? How many? |
| 15:52.49 | omenSP | And we have around 30 phone lines running to individual offices, extensions controlled through Vertical TeleVantage. |
| 15:52.56 | omenSP | Analog phones. |
| 15:53.01 | omenSP | Like phone stations. |
| 15:53.09 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-vdclrtowugmasvul) |
| 15:53.14 | omenSP | We would be replacing all of them if we change to VoIP |
| 15:53.53 | [TK]D-Fender | omenSP: You SURE they're analog? have you plugged them into a regular line to test? |
| 15:54.08 | [TK]D-Fender | "like phone stations" isn't very definitive |
| 15:54.19 | omenSP | We're running two conductor wire to them, so it's a RJ11 phone. |
| 15:54.39 | [TK]D-Fender | that could mean a lot fo things |
| 15:55.05 | [TK]D-Fender | Mortel Norstar Digital handset run on a signle pair with RJ11 also... and tghey are NOT "analog" |
| 15:55.42 | omenSP | http://www.uniden.com/content/ebiz/uniden/resources/ownersmanuals/TRU9488om.pdf is what I have in my office right now. |
| 15:58.08 | [TK]D-Fender | omenSP: How many phones? |
| 15:58.17 | omenSP | We have around 30 phones in the building. |
| 15:59.19 | [TK]D-Fender | omenSP: Ok, they look like boring analog. Have you tested them? |
| 15:59.31 | omenSP | Tested them with what? |
| 15:59.37 | omenSP | I haven't integrated Asterisk at all yet. |
| 16:00.00 | omenSP | I am wondering what I would need to get in order to do that. We're going to replace the phones with VoIP phones if we integrate Asterisk. |
| 16:00.07 | [TK]D-Fender | I meant on a line DIRECTLY |
| 16:00.24 | [TK]D-Fender | Oh, well if you do actually want to replace them... sure. |
| 16:00.34 | [TK]D-Fender | You COULD reuse them if you wanted |
| 16:01.06 | omenSP | What would I need, besides IP phones, to use Asterisk? |
| 16:01.20 | omenSP | I found a 8-port telephony card for our 8 trunk lines. |
| 16:02.24 | wasanzy | I have this: exten => 1234,n,Set(DUR=sox /var/lib/asterisk/sounds/0248148457_sample.wav -n stat | grep -i length ) |
| 16:02.30 | *** join/#asterisk calum_ (~calum_@cpc67428-harg5-2-0-cust142.7-1.cable.virginm.net) |
| 16:03.09 | wasanzy | and am getting this error: http://paste.scsys.co.uk/407074 |
| 16:03.31 | wasanzy | should I replace the pip with |
| 16:04.12 | [TK]D-Fender | wasanzy: escape it |
| 16:04.18 | [TK]D-Fender | omenSP: That could do. |
| 16:05.43 | omenSP | [TK]D-Fender: So all I need is the server with Asterisk, that card, and I would be able to plug in the trunk lines and have an IP phone plugged in somewhere and I can set the system up and get it working? |
| 16:09.11 | wasanzy | [TK]D-Fender: ok |
| 16:10.25 | [TK]D-Fender | omenSP: Pretty much. Get hardware to connect lines. Get some kind of phones that can talk to*. Configure. Enjoy. |
| 16:11.14 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 16:12.53 | omenSP | Thank you. |
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| 17:01.03 | docelmo | Can anyone tell me if its possibly when asterisk receives a redirect sip message that it can copy all X headers in the 302 message to the new invite of the redirect? |
| 17:01.13 | docelmo | possible.. |
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| 17:33.11 | ipengineer | When running Dial(PJSIP/200&PJSIP/201&PJSIP/203,25) if one of the devices are not available asterisk returns âunable to create channel of type PJSIPâ. Is there a way to have the Dial cmd check if chanIsAvail before trying to send a call so in the event one of these devices are unregistered the others will still ring? |
| 17:33.16 | *** join/#asterisk dwayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 17:33.38 | [TK]D-Fender | The others should ring regardless of the error on one |
| 17:34.05 | [TK]D-Fender | And for your proposed path, no there's no way to do it like that |
| 17:34.10 | ipengineer | [TK]D-Fender: Maybe something else then? This is all I captured: https://gist.github.com/zconkle/6868a175dd4a6067d220 |
| 17:35.42 | [TK]D-Fender | I don;t see an end of channel in there |
| 17:35.49 | [TK]D-Fender | what tells me it didn't continue with the other?> |
| 17:37.01 | ipengineer | It does continue when I test with a different group. It must have been something else then. That error just made me think that was it and once I rebooted all devices they came up and started ringing |
| 17:37.20 | *** join/#asterisk sarlalian (~sarlalian@107.170.239.102) |
| 17:42.24 | wasanzy | am getting wrong result using this: exten => 1234,n,Set(DUR=$[${STAT(s,/var/lib/asterisk/sounds/0248148457_sample.wav)} (- 44) / ( 16000 ) ]) |
| 17:42.29 | *** join/#asterisk ipengineer_ (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 17:42.32 | wasanzy | the result should be 20 |
| 17:43.04 | *** join/#asterisk BakaKuna (~Thunderbi@82-169-251-128.ip.telfort.nl) |
| 17:55.12 | wasanzy | any help? |
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| 18:29.28 | TazzNZ | morning all |
| 18:33.35 | wasanzy | please I need a help |
| 18:34.27 | TazzNZ | ok....what is the issue wasanzy ? |
| 18:35.49 | wasanzy | <PROTECTED> |
| 18:36.15 | wasanzy | the result should be 20 |
| 18:36.25 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-snupxfbgqmmqucwd) |
| 18:36.27 | TazzNZ | that is a tiny wav file ? |
| 18:36.32 | wasanzy | yes |
| 18:36.37 | TazzNZ | what happens if you do ls -la /var/lib/asterisk/sounds/0248148457_sample.wav |
| 18:37.15 | *** join/#asterisk ageis (kevin@ageispolis.net) |
| 18:37.23 | wasanzy | -rw-r--r-- 1 asteriskpbx asteriskpbx 320044 Nov 14 2013 /var/lib/asterisk/sounds/0248148457_sample.wav |
| 18:37.24 | ageis | this sudden SIP retransmission bug is crippling our phone service. |
| 18:37.38 | TazzNZ | that wont return 20 |
| 18:37.42 | ageis | [Jul 8 18:36:47] WARNING[2424]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission 77ba5e3c16f231c3e0cc8c4ef860cbcc@0.0.0.0 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 18:37.43 | ageis | Packet timed out after 6400ms with no response |
| 18:37.43 | ageis | [Jul 8 18:36:47] WARNING[2424]: chan_sip.c:4204 retrans_pkt: Hanging up call 77ba5e3c16f231c3e0cc8c4ef860cbcc@0.0.0.0 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). |
| 18:37.50 | TazzNZ | it should return 320044 |
| 18:38.07 | TazzNZ | ageis: what asterisk version ? |
| 18:38.25 | TazzNZ | and did you zero out that IP ? |
| 18:38.40 | ageis | TazzNN: 11.7 |
| 18:39.01 | ageis | zero out? this problem is showing up suddenly. nothing's changed about the route between the phones and the servers |
| 18:39.15 | TazzNZ | 77ba5e3c16f231c3e0cc8c4ef860cbcc@0.0.0.0 <-- that IP there |
| 18:39.32 | ageis | that looks like the loopback device to me |
| 18:39.40 | TazzNZ | 127.0.0.1 == loopback |
| 18:39.45 | wasanzy | TazzNZ: am trying to get the length of the wav sound, returning the file size is what I don't want. So I want to subtract 44 from file size |
| 18:39.50 | ageis | so what do you mean zero out? |
| 18:40.04 | TazzNZ | wasanzy: s - Returns the size (in bytes) of the file |
| 18:40.26 | TazzNZ | iirc, Asterisk doesn't "understand" WAV files |
| 18:40.45 | TazzNZ | it does have a understanding of wav49 thou |
| 18:40.50 | TazzNZ | *i think* |
| 18:40.58 | ageis | nah wav is fine |
| 18:41.21 | TazzNZ | ageis: so you pasted that message as it is on the console ? |
| 18:41.30 | ageis | TazzNZ: from my full log |
| 18:42.11 | TazzNZ | when you make a call, can you turn on sip debug and see what the headers say about RTP ? |
| 18:42.25 | TazzNZ | there is an INVITE packet |
| 18:42.35 | TazzNZ | PB the contents of that please |
| 18:43.21 | ageis | m=audio 13640 RTP/AVP 0 101 |
| 18:43.21 | ageis | a=rtpmap:0 PCMU/8000 |
| 18:43.21 | ageis | a=rtpmap:101 telephone-event/8000 |
| 18:43.33 | TazzNZ | the whole packet please |
| 18:43.39 | TazzNZ | in pastebin |
| 18:44.06 | *** join/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 18:44.22 | TazzNZ | wasanzy: I don't know how you are going to work out the length of a wav file in * |
| 18:44.49 | TazzNZ | but the STAT function will not do it |
| 18:44.49 | ageis | TazzNZ: https://ageispolis.net/bin/?41978f91d44258e5#AlfQbP2MX1myRnfvC9M8hOhZ7i2TQYoTYmMGt06l+hM= |
| 18:46.29 | [TK]D-Fender | [14:35]wasanzyam getting wrong result using this: exten => 1234,n,Set(DUR=$[${STAT(s,/var/lib/asterisk/sounds/0248148457_sample.wav)} (- 44) / ( 16000 ) ]) [14:36]wasanzythe result should be 20\ |
| 18:46.30 | Katty | OH WHERE IS MY HAIR BRUSH |
| 18:46.36 | Katty | infobot: OHWHEREISMYHAIRBRUSH |
| 18:46.36 | infobot | ohwhereismyhairbrush is, like, NOT FAIR, MY POOR HAIRBRUSH! |
| 18:46.50 | TazzNZ | lol - "hi" Katty |
| 18:46.53 | [TK]D-Fender | wasanzy: Stop trying to do everything in 1 shot. LOOK at the result of your stat before trying to evaluate it, and stop putting useless backets |
| 18:47.00 | Katty | TazzNZ: howdy |
| 18:47.06 | Katty | [TK]D-Fender: are you being nice to the locals, dear? |
| 18:47.32 | [TK]D-Fender | Katty: I now chew before swallowing ;) |
| 18:48.01 | TazzNZ | ageis: what does your network look like ? |
| 18:48.05 | Katty | rolls eyes |
| 18:48.46 | ageis | TazzNZ: we're running Asterisk on a Linode, our network has an Asus router and comcast business connection with a single subnet |
| 18:49.40 | TazzNZ | so what is the IP of the Asterisk box and the phones etc ? |
| 18:50.40 | *** part/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 18:55.10 | ageis | TazzNZ: the phones all have IP 75.144.202.33 the ast box is at 66.175.211.211 |
| 18:55.22 | ageis | wow |
| 18:55.29 | ageis | my externip and localnet wasn't set right |
| 18:55.50 | ageis | could that have anything to do with anything? |
| 18:58.51 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 18:59.31 | TazzNZ | ageis: I'd say :) |
| 18:59.41 | ageis | ok |
| 18:59.58 | TazzNZ | is 172.62 your ITSP ? |
| 19:00.22 | TazzNZ | and you are NAT'ing to the Asterisk box ? |
| 19:00.28 | TazzNZ | so your phones are really: |
| 19:00.48 | TazzNZ | private network -> ROUTER (NAT to 75.144) -> Ast Box at 66.... |
| 19:04.03 | ageis | TazzNZ: now I have a problem with my polycom |
| 19:04.08 | ageis | [Jul 8 18:58:07] WARNING[4024][C-00000003]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) |
| 19:04.09 | *** join/#asterisk jhirley (~chatzilla@50.248.45.129) |
| 19:04.44 | ageis | TazzNZ: I'm not sure where you pulled that 172.62 number |
| 19:05.06 | TazzNZ | whoops - that is meant to be 174 ageis |
| 19:05.13 | TazzNZ | <--- Transmitting (NAT) to 174.62.218.238:5060 ---> |
| 19:05.26 | ageis | that's just one of my devices |
| 19:05.30 | ageis | we have devices in several locations |
| 19:05.32 | ageis | on diff networks |
| 19:05.36 | ageis | that one is not a problem |
| 19:05.41 | ageis | sorry for including that packet |
| 19:05.49 | TazzNZ | all good |
| 19:05.57 | ageis | the main problem is the "Subscriber absent" on my Polycom SoundStation now |
| 19:06.03 | ageis | and I've filled out the conf correctly |
| 19:06.23 | TazzNZ | are you trying to make a call to your polycomm ? |
| 19:06.43 | ageis | yes |
| 19:06.56 | TazzNZ | is it registered ? |
| 19:07.25 | ageis | yes |
| 19:07.32 | ageis | registered, says OK |
| 19:07.39 | ageis | but subscriber absent |
| 19:07.53 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-uitimnxasherhean) |
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| 19:13.29 | TazzNZ | ageis: can you make an outgoing call from it ? |
| 19:14.12 | ageis | not sure, im doing all this remotely |
| 19:15.41 | *** join/#asterisk NoobSaibot (~NoobSaibo@CPE-24-208-40-235.new.res.rr.com) |
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| 19:17.54 | TazzNZ | I'd try that :) |
| 19:24.07 | ageis | TazzNZ: unregistered ATM.. the web interface doesn't even load |
| 19:31.20 | ageis | TazzNZ: ok got it working |
| 19:31.29 | ageis | TazzNZ: do you know anything about setting dscp stuff with iptables? |
| 19:33.01 | NoobSaibot | <PROTECTED> |
| 19:33.52 | NoobSaibot | I got a 3750 PoE switch and a bunch of cisco phones to replace our old Avaya/Grandstream devices, but we don't have a call manager/CUCM license. Is there a handy firmware version that will allow SIP on the cisco phones, or what's my first step? |
| 19:34.14 | TazzNZ | ageis: yeah - what you trying to do ? |
| 19:34.21 | TazzNZ | we might have to take it off channel |
| 19:34.31 | Qwell | NoobSaibot: According to Cisco, you still need a license to use the firmware with some other PBX. |
| 19:34.41 | ageis | TazzNZ: does this look good to you? https://ageispolis.net/bin/?83f6bfc085cd37a3#SeLXPW/vIM49O494yDSXhAQ+bAYgU+tomYXHn45MWZQ= |
| 19:34.44 | TazzNZ | NoobSaibot: yeah - you get a SIP firmware |
| 19:35.08 | NoobSaibot | Qwell: I thought the phones came with their own licesnes, unless they're factory replacements. They're all original phones, just no call manager. |
| 19:35.13 | TazzNZ | which - iirc, is quite pricey |
| 19:35.51 | TazzNZ | NoobSaibot: you will still have to buy the firmware |
| 19:35.54 | TazzNZ | per device |
| 19:35.58 | Qwell | NoobSaibot: You'd think so, wouldn't you? |
| 19:36.05 | NoobSaibot | Thanks TazzNZ, i'll check into the other firmware and figure out the licensing issue. |
| 19:36.17 | TazzNZ | NoobSaibot: hence I stay away from Cisco and Oracle :) |
| 19:36.18 | Qwell | I would suggest different phones. |
| 19:37.08 | NoobSaibot | Yeah, we had a different office close and we got their stuff, so now i'm supposed to 'use' it. They just ripped everything down with no documentation whatsoever, just a bunch of boxes of stuff showed up in my server room. |
| 19:37.35 | TazzNZ | NoobSaibot: keep the boxes, sell the phones :D |
| 19:38.02 | NoobSaibot | good idea, actually... |
| 19:38.26 | Qwell | "Phones? I don't know what you're talking about, boss. Those boxes were all empty." |
| 19:38.37 | TazzNZ | hi5 Qwell |
| 19:40.16 | TazzNZ | oh NoobSaibot - use the money from the sale to buy better phones :D |
| 19:43.22 | *** join/#asterisk panicman (~panicman@223.27.112.16) |
| 19:43.54 | panicman | Hi all, I've a setup with 3 outgoing trunk, i want to load balance for outgoing |
| 19:44.02 | panicman | can anyone help please |
| 19:44.21 | TazzNZ | panicman: sure - can you show us your extentions.conf |
| 19:44.22 | panicman | using CLI mode |
| 19:44.55 | panicman | exten => _X.,1,GotoIf($[${RAND(0,99)} >= 50]?,6|7) |
| 19:44.55 | panicman | exten => _X.,2,Dial(SIP/TP1/${EXTEN:0},30) |
| 19:44.55 | panicman | exten => _X.,3,Hangup() |
| 19:44.55 | panicman | exten => _X.,4,Dial(SIP/TP2/${EXTEN:0},30) |
| 19:44.55 | panicman | exten => _X.,5,Hangup() |
| 19:44.55 | panicman | exten => _X.,6,Dial(SIP/TP3/${EXTEN:0},30) |
| 19:44.55 | panicman | exten => _X.,7,Hangup() |
| 19:44.56 | panicman | its working for TP1 & TP2 |
| 19:45.05 | TazzNZ | I meant in a pastebin :) |
| 19:45.15 | panicman | i don;t know how to do it sir |
| 19:45.20 | TazzNZ | ~pb |
| 19:45.20 | 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. |
| 19:45.21 | panicman | new here |
| 19:45.40 | panicman | let mecheck sir |
| 19:46.53 | panicman | http://pastebin.com/yzLyxe5L |
| 19:47.38 | panicman | TazzNZ -- Got it ! |
| 19:48.09 | TazzNZ | ok ? |
| 19:48.18 | panicman | waiting for your kind update |
| 19:48.38 | TazzNZ | ah ok |
| 19:49.03 | panicman | :) |
| 19:49.10 | *** join/#asterisk bitwize (~S2TS@c83-253-248-92.bredband.comhem.se) |
| 19:52.02 | TazzNZ | panicman: are you sure it's working for TP1 and 2 |
| 19:52.12 | panicman | yes |
| 19:52.15 | TazzNZ | I would have thought it would work for TP1 and 3 |
| 19:52.20 | *** join/#asterisk NoobSaibot_ (~NoobSaibo@CPE-24-208-40-235.new.res.rr.com) |
| 19:52.24 | panicman | _X.,1,GotoIf($[${RAND(0,99)} >= 50]?,6) |
| 19:52.42 | TazzNZ | that is not what you showed |
| 19:52.53 | panicman | no difference sir |
| 19:53.03 | TazzNZ | GotoIf($[${RAND(0,99)} >= 50]?,6|7) <-- really ? |
| 19:53.29 | panicman | i mean, _X.,1,GotoIf($[${RAND(0,99)} >= 50]?,6|7) is not working |
| 19:53.37 | panicman | what I mean |
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| 19:54.46 | panicman | any idea sir ! |
| 19:55.37 | [TK]D-Fender | panicman: No comma after the question mark |
| 19:55.39 | TazzNZ | I think that GotoIf is not doing what you think it should |
| 19:55.40 | [TK]D-Fender | ^^^^ |
| 19:56.14 | [TK]D-Fender | and no pipe either... |
| 19:56.21 | [TK]D-Fender | THAT should be a comma |
| 19:56.41 | [TK]D-Fender | GotoIf($[${RAND(0,99)} >= 50]?6,7) <--- |
| 19:56.46 | panicman | let me try sir |
| 19:56.48 | TazzNZ | yeah panicman, you going to need 2 gotoif's |
| 19:57.21 | TazzNZ | personally, I would have done a round-robin |
| 19:57.36 | TazzNZ | RAND is so......radnom |
| 19:57.41 | TazzNZ | random even |
| 19:58.55 | [TK]D-Fender | Actually it's pseudo-random which is often not quite random enough |
| 19:58.59 | TazzNZ | panicman: uhm - that [TK]D-Fender suggested will hangup some calls |
| 19:59.08 | TazzNZ | it should read |
| 19:59.21 | panicman | can u tell me how to do the round robin sir |
| 19:59.24 | TazzNZ | GotoIf($[${RAND(0,99)} >= 50]?4,6) |
| 19:59.41 | panicman | Channel 'SIP/ss-0000a634' sent into invalid extension ' 6' in context 'ss7', but no invalid hand |
| 20:00.01 | TazzNZ | you have a space |
| 20:00.09 | TazzNZ | ' 6' != '6' |
| 20:00.46 | TazzNZ | ageis: whoops - forgot about you |
| 20:01.04 | ageis | let me update that |
| 20:01.05 | ageis | one sec |
| 20:01.14 | TazzNZ | ageis: cool :) |
| 20:01.16 | ageis | TazzNZ: https://ageispolis.net/bin/?e20907ab55ade7eb#Jck33zWF8vMl5xPbBeoHBN8RkJKth5kNvS9nhueT07Q= |
| 20:01.37 | ageis | 24 is decimal for hexidecimal 0x18, high throughput and low delay |
| 20:01.43 | ageis | right? |
| 20:01.54 | TazzNZ | ageis: I would only worry about adjusting the RTP traffic |
| 20:02.06 | TazzNZ | the SIP traffic is a very small amount of the call data |
| 20:02.08 | ageis | so the --sport 10000:20000 ? |
| 20:02.19 | TazzNZ | and doesn't impact the quality or dropped packets |
| 20:02.25 | ageis | ok... but the rules look good anyhow? |
| 20:02.31 | TazzNZ | yeah - if that is what is set in rtp.conf |
| 20:02.58 | TazzNZ | is this linux box your firewall ? |
| 20:03.16 | TazzNZ | and you want to "shape" RTP traffic ? |
| 20:04.17 | ageis | no this is the asterisk box |
| 20:04.36 | ageis | I don't have any DSCP settings in rtp.conf |
| 20:04.50 | TazzNZ | right - so you want to set this in the ip header so that the rest of the networking devices can shape the traffic |
| 20:04.53 | ageis | I have some tos=0x18 and tos_sip=cs3 and tos_audio=ef in sip.conf |
| 20:05.10 | TazzNZ | there is no DSCP settings in rtp.conf, I was refering to the ports |
| 20:05.18 | ageis | ohhhh |
| 20:05.29 | ageis | actually, they start at 5000 and end at 31000 |
| 20:05.31 | ageis | so I should fix that |
| 20:05.38 | ageis | thanks |
| 20:05.43 | TazzNZ | ageis: yeah - that setting in sip.conf is enough for asterisk to set the settings, you don't need iptables then |
| 20:05.46 | ageis | 10000:20000 is godo you think? |
| 20:05.52 | ageis | TazzNZ: yeah, i read that |
| 20:06.05 | TazzNZ | 10000:20000 is the default and more than enough |
| 20:06.22 | ageis | okay. what about rtpchecksums and rtcpinterval ? |
| 20:07.03 | TazzNZ | rtcpinterval I would leave that at the default unless you have a spefic reason to change it |
| 20:07.27 | TazzNZ | are you trying to solve an issue here or just fine tuning ? |
| 20:07.55 | ageis | fine tuning to give high priority and performance. we've having some issues with dropping in the middle of calls |
| 20:09.03 | TazzNZ | tbh, I would leave most of the settings alone until you can say that a setting is the cause of an issue |
| 20:09.38 | TazzNZ | when the calls drop, do you have a long period of no audio ? |
| 20:09.44 | TazzNZ | like 30 seconds ? |
| 20:09.57 | TazzNZ | does it affect internal and external calls ? |
| 20:10.04 | TazzNZ | or only one of the above ? |
| 20:11.40 | ageis | short periods of no audio |
| 20:11.48 | ageis | umm |
| 20:11.50 | ageis | incoming calls |
| 20:12.06 | panicman | sorry back, not working , its sending me to 6 , here is the log--- sent into invalid extension '4' in context |
| 20:12.32 | panicman | GotoIf($[${RAND(0,99)} >= 50]?4,8) |
| 20:12.37 | panicman | this is the current settings |
| 20:12.45 | TazzNZ | panicman: why 8 ? |
| 20:12.54 | panicman | 3 trunk |
| 20:13.01 | panicman | tp1 tp2 tp3 |
| 20:13.16 | panicman | oh |
| 20:13.26 | TazzNZ | ah wait |
| 20:13.34 | panicman | panic:( |
| 20:13.45 | TazzNZ | make that |
| 20:13.52 | TazzNZ | GotoIf($[${RAND(0,99)} >= 50]?4:8) |
| 20:14.01 | TazzNZ | if 4 is the start of TP2 |
| 20:14.07 | TazzNZ | and 8 is the start of TP3 |
| 20:14.26 | [TK]D-Fender | Where |
| 20:14.30 | [TK]D-Fender | Where's TP1? |
| 20:14.38 | TazzNZ | TP1 won't work for now |
| 20:14.44 | [TK]D-Fender | \o/ |
| 20:14.47 | TazzNZ | we only have 1 gotoif |
| 20:14.49 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 20:14.55 | TazzNZ | we need 2 :) |
| 20:16.23 | panicman | Tp1 & Tp3 is working, TP2 is not working :( |
| 20:16.54 | panicman | can u give me the total extension config if possible |
| 20:16.59 | TazzNZ | ok - now panicman - please paste the correct config |
| 20:17.13 | TazzNZ | since the one you pasted, didn't even have 8 in it |
| 20:17.27 | panicman | ok sir |
| 20:20.09 | panicman | http://pastebin.com/gCLqQZgb |
| 20:21.00 | TazzNZ | VIEWS: 25 | EXPIRES: NEVER <-- 25 views already....wow |
| 20:21.05 | TazzNZ | anyways |
| 20:21.23 | panicman | ohhh |
| 20:24.49 | panicman | http://pastebin.com/5v9UBsTp |
| 20:25.12 | panicman | TP1 & TP3 working only |
| 20:25.16 | panicman | TP2 is not working |
| 20:28.26 | TazzNZ | http://pastebin.com/GSDPT9K4 |
| 20:28.29 | TazzNZ | that is what I would do |
| 20:28.37 | TazzNZ | it is untested so test it first |
| 20:28.44 | TazzNZ | not on your production system |
| 20:29.00 | TazzNZ | at least, that is my second choice |
| 20:29.44 | panicman | let me check both :) |
| 20:29.46 | panicman | thanks a lot |
| 20:29.54 | TazzNZ | both ? |
| 20:30.46 | panicman | no no |
| 20:30.52 | panicman | wrong type :) |
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| 23:45.14 | mroe | is there a service that I can use to test my inbound calling? |
| 23:48.59 | [TK]D-Fender | yes. Several kinds |
| 23:49.03 | [TK]D-Fender | Grab a cell phone |
| 23:49.09 | [TK]D-Fender | call from a telco land-line |
| 23:49.13 | [TK]D-Fender | etc |
| 23:50.42 | mroe | [TK]D-Fender: oddly enough I have neither |
| 23:51.05 | [TK]D-Fender | You have noboday ANYWHERE near you you can contact to tel them to call you>? |
| 23:51.11 | [TK]D-Fender | No friend you can e-mail? |
| 23:51.55 | mroe | I just sent out a few messages, but I'm guessing I'm going to need to test this for a while and I was wondering if there was a site/service that would call you with a pre-recorded message |
| 23:52.44 | [TK]D-Fender | They'd be understandably rare so that they don't get abused to harass others |
| 23:52.54 | mroe | I guess |
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| 23:58.55 | mjordan | set up two Asterisk instances |
| 23:58.57 | mjordan | call yourself! |
| 23:59.25 | mroe | mjordan: I'm also testing my NAT, so that will be difficult |
| 23:59.51 | mjordan | well, you will exercise your NAT settings by pointing two Asterisk servers at each other |