| 00:01.54 | Ice_Strike | CE[29877]: chan_sip.c:22318 handle_request_register: Registration from '<sip:20x@87.76.3x.xxx;transport=UDP>' failed for '94.192.11x.xx' - No matching peer found |
| 00:01.57 | Ice_Strike | good sign eh :) |
| 00:02.02 | p3nguin | Good. |
| 00:02.15 | p3nguin | Now you need to create a peer in sip.conf for your phone. |
| 00:02.47 | Ice_Strike | I thought I did |
| 00:02.52 | Ice_Strike | let me pastebin |
| 00:03.20 | Ice_Strike | http://pastebin.com/tGXzBcBD |
| 00:05.18 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:05.18 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:07.03 | p3nguin | http://pastebin.com/VXxfgytR |
| 00:09.05 | Ice_Strike | thanks mate |
| 00:12.53 | Ice_Strike | working :) |
| 00:16.33 | p3nguin | Also... |
| 00:16.37 | p3nguin | ~devices |
| 00:16.37 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 00:16.57 | p3nguin | Don't learn bad habits early. |
| 00:17.51 | Ice_Strike | p3nguin got extension test working |
| 00:17.52 | Ice_Strike | :) |
| 00:17.57 | p3nguin | Nope, you don't. |
| 00:18.09 | p3nguin | You have a device with the name of "test" working. |
| 00:18.33 | p3nguin | Phones are not extensions; extensions are not phones. |
| 00:18.47 | Ice_Strike | p3nguin I mean extention.conf file |
| 00:18.54 | Ice_Strike | eg: exten => 111,3,Playback(demo-echotest) |
| 00:19.17 | Ice_Strike | I can hear when I dial 111 |
| 00:19.32 | p3nguin | Good. Now learn about priorities. |
| 00:19.52 | Ice_Strike | Yep, thanks for help though |
| 00:20.02 | p3nguin | Using numbered priorities everywhere will surely cause headaches later. |
| 00:20.34 | Ice_Strike | you mean use n instead |
| 00:20.57 | p3nguin | yes |
| 00:20.59 | p3nguin | http://pastebin.com/Piqv4Egj |
| 00:21.50 | Ice_Strike | nice thanks |
| 00:22.13 | Ice_Strike | let me explain what I am trying to archive |
| 00:23.32 | Ice_Strike | http://pastebin.com/sfHGcX3f |
| 00:24.12 | Ice_Strike | Not sure if I should use AGI or AMI ? |
| 00:24.22 | Ice_Strike | I saw some PHP libaries for that |
| 00:24.39 | p3nguin | That can be done in dial plan without any scripting. |
| 00:24.49 | p3nguin | But you'll use AGI if you want to script it with PHP. |
| 00:24.57 | p3nguin | AMI is the manager interface. |
| 00:25.14 | Ice_Strike | using dial plan to do loop via database? |
| 00:25.22 | p3nguin | yes |
| 00:25.24 | Ice_Strike | I think I use PHP, that is what I am most familar with |
| 00:26.58 | p3nguin | I would do it with regular dial plan and leave PHP for something more... relevant to PHP. |
| 00:27.32 | p3nguin | But you can do 99% of it in PHP and then use a very simple extension to launch the script with AGI(). |
| 00:27.48 | Ice_Strike | So If I press 123 from phone, it will continue looping many numbers from the datababase? It dailplan should hangup automatically and then auto dial to next number.. |
| 00:28.50 | Ice_Strike | Ok, I need to check mysql connection for dial plan and like a while loop feature |
| 00:39.43 | Ice_Strike | p3nguin i've got php AGI working :D |
| 00:39.51 | Ice_Strike | well just testing bit. |
| 00:40.15 | Ice_Strike | Im off to bed, thanks for the help. |
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| 02:52.13 | atan | Running into a qwirk when I try to direct a voip call from phone->asterisk->voipms->asterisk->different phone something about failed to authenticate device. Almost like it's trying to pass along my SIP information. Not sure how to prevent but I bet I missed a setting on my SIP peer |
| 02:52.31 | atan | http://pastebin.com/AzzLWMgk even :-) |
| 02:56.30 | Rahoul | Hello, are there any public sip conference lines? (party lines) |
| 02:59.35 | ectospasm | Rahoul: you could set one up with Asterisk pretty simply, but I don't know of anything like that off the top of my head. |
| 03:01.50 | Rahoul | hey ectopasm, yeah I know how to set it up, but a few years back there were some public open party lines where voip discussions took place (sipphone for example). |
| 03:02.14 | Rahoul | I was wonderig if there are still any type of services with uri for people to call :) |
| 03:11.36 | p3nguin | atan: Why is the ITSP involved at all? |
| 03:12.39 | atan | p3nguin, the as of yet the number being dialed isn't setup locally. I don't "know" it's with me just yet. |
| 03:13.00 | atan | p3nguin, what's going on here is VOIP phone tries to call number that just so happens to also be local to that asterisk box when it tries to terminate |
| 03:14.18 | atan | adding Insecure : port,invite to the calling device seems to resolve the error I saw there but the call doesn't actually come back to the server |
| 03:15.54 | p3nguin | I can't see the problem, so I can't fix it. |
| 03:16.06 | p3nguin | There shouldn't be a problem calling yourself via ITSP. |
| 03:16.13 | p3nguin | I'll test it just to be sure. |
| 03:16.23 | atan | <PROTECTED> |
| 03:17.24 | p3nguin | It works fine for me. |
| 03:17.38 | atan | voip phones->asterisk->voipms->pstn works fine. So does pstn->voipms->asterisk->voip phones. It's only happening if I try to call a device that's also registered to the same asterisk box O_O when you try to send the call back I get Got SIP response 486 "Busy Here" back |
| 03:17.45 | p3nguin | I'm not sure why I'm getting goofy caller id numbers showing up on the inbound call, but the call completes just fine. |
| 03:18.09 | p3nguin | Sounds like the phone your call is directed to is busy. |
| 03:18.16 | p3nguin | DND, maybe |
| 03:18.49 | atan | I threw playback(tt-monkeys) into the dialplan and the number accepts a call from pstn, but not voip still busy. No reason for it to be busy though. |
| 03:21.18 | p3nguin | I still haven't seen it. |
| 03:21.29 | p3nguin | I keep waiting for the pastebin, but it just hasn't arrived. |
| 03:21.52 | ChannelZ | stares longingly into the trash |
| 03:22.42 | atan | Okay that didn't do anything as expected. Need to clean up my dialplan and make it deal simple one sec. Right now it's trying to open like 30 outbound lines which clearly isn't right. |
| 03:22.42 | p3nguin | Okay, this is really weird. Calls from my phone are from "VOIP CALLER" with a random phone number. |
| 03:23.41 | atan | p3nguin, what do you have insecure set to for your SIP connect to the ITSP, and what's insecure set to on your peer? |
| 03:24.23 | p3nguin | Not set, i.e. "no" |
| 03:24.55 | p3nguin | I have only one peer entry for the ITSP. |
| 03:25.15 | p3nguin | For phones, you never set insecure anyway. |
| 03:32.49 | p3nguin | Okay, figured out that little issue. |
| 03:33.34 | ectospasm | I had trouble tonight because it appears Asterisk 1.8.11-cert1 doesn't have pbx_ael.so enabled by default... |
| 03:34.53 | p3nguin | You use AEL? |
| 03:35.05 | ectospasm | ...just on my personal system. |
| 03:35.18 | ectospasm | looks cleaner than .ini IMO. |
| 03:35.41 | ectospasm | ...but I do most of my stuff in extensions.conf |
| 03:35.44 | p3nguin | It might be cleaner because it is so much more restrictive. |
| 03:36.14 | ectospasm | I think it's cleaner because it's not as tedious to construct. |
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| 03:46.54 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 03:47.11 | joako | I am missing the "core reload" command.... how do I enable it? |
| 03:48.18 | ectospasm | joako: should be just "reload" |
| 03:49.04 | ectospasm | joako: I stand corrected. What version of Asterisk are you using? |
| 03:49.32 | joako | ectospasm, Asterisk 1.4.44 |
| 03:50.04 | ectospasm | joako: I don't have a 1.4 system to connect to, but my guess is "core reload" isn't a valid option for 1.4. Try "reload" |
| 03:50.43 | ectospasm | joako: just so you know, Asterisk 1.4 is EOL (End of Life) since April 21, 2012. |
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| 03:55.00 | p3nguin | Don't "reload" ... |
| 03:55.03 | p3nguin | Reload what you need to reload. |
| 03:55.27 | p3nguin | Need to reload sip config? sip reload |
| 03:55.34 | p3nguin | Need to reload dial plan? dialplan reload |
| 03:55.38 | p3nguin | et cetera |
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| 04:20.39 | rtfm | question |
| 04:20.53 | rtfm | Should we buy the devel version, what language is the source code in? |
| 04:23.17 | ectospasm | devel version of what? Asterisk? |
| 04:26.50 | rtfm | yeah |
| 04:27.07 | kaldemar | you can just download it from http://downloads.asterisk.org/pub/telephony/asterisk/ |
| 04:27.51 | rtfm | call center stats? |
| 04:27.53 | rtfm | ok |
| 04:28.01 | ectospasm | rtfm: it's open source, and it's written mostly in C. |
| 04:29.07 | kaldemar | what "call center stats"? are talking about some 3rd party application? |
| 04:29.38 | rtfm | http://www.asternic.biz/downloads/lite |
| 04:29.53 | rtfm | k |
| 04:29.54 | ectospasm | rtfm: Asterisk generally doesn't provide call center stats, you'll need a third-party app for that. |
| 04:31.46 | rtfm | k |
| 04:34.20 | rtfm | Have another question, might be unrelated or a network issue, just want to isolate this - A callee hits the IVR, but the time taken from IVR to phone extentions is quite long (20ms to 1000ms) latency. Could this be a problem with the acd? |
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| 04:36.11 | joako | Actuallly I am just a bit confused.... |
| 04:36.34 | joako | The system in question is Asterisk 1.6.2.23 and neither the "reload" or "core reload" commands are recognized. |
| 04:40.08 | kaldemar | rtfm: time for what? audio latency? what has ACD to do with this? |
| 04:40.25 | rtfm | yeah |
| 04:40.58 | rtfm | hm |
| 04:42.16 | kaldemar | how did you measure the audio latency? |
| 04:47.14 | rtfm | kaldemar uh, talking to these vendors |
| 04:47.19 | rtfm | latency within the network |
| 04:47.55 | rtfm | not audio, I'm just trying to figure out how this works. |
| 04:49.08 | p3nguin | joako: Like I said... |
| 04:49.12 | p3nguin | (2255.03) <p3nguin> Reload what you need to reload. |
| 04:49.12 | p3nguin | (2255.27) <p3nguin> Need to reload sip config? sip reload |
| 04:49.12 | p3nguin | (2255.34) <p3nguin> Need to reload dial plan? dialplan reload |
| 04:49.44 | joako | p3nguin, I used to --- and still can on other systems -- use the "reload" command to reload everything |
| 04:49.52 | p3nguin | Well, stop it. |
| 04:50.17 | joako | e.g. if I edit users.conf why type "reload users" "reload diaplan" and "reload voicemail" when I can do the same with 1 command |
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| 04:50.31 | ectospasm | p3nguin: so give him a reason why. I'd like to know, too... |
| 04:50.32 | p3nguin | ~users.conf |
| 04:50.32 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 04:51.27 | p3nguin | It's just silly to want to reload an entire system when generally only an area or two need to be reloaded. It's like rebuilding the engine to change a spark plug. |
| 04:51.52 | joako | I put my cheap white bread in users.conf and I use the other config files for the fine baguettes! |
| 04:52.12 | ectospasm | ...it usually doesn't hurt to do a full reload, though. Unless a total Asterisk restart is necessary... |
| 04:52.18 | joako | .... but all this talk about bread still doesn't explain why the "core reload" commadn is missing. |
| 04:52.31 | p3nguin | is going to make a French dip in a minute. |
| 04:52.46 | ectospasm | au jus? |
| 04:52.53 | p3nguin | Of course. |
| 04:53.28 | kaldemar | joako: how did you install? did you modify modules.conf? |
| 04:53.51 | *** join/#asterisk F|shie (~chatzilla@182.177.14.90) |
| 04:54.01 | p3nguin | I had a couple Jack Link's beef sticks for supper, so now I'm getting hungry. |
| 04:54.34 | ectospasm | just made a simple ham&swiss sammich |
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| 05:14.17 | F|shie | hello all, is there anyway to detect dtmf tones within a eagi script? |
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| 05:24.54 | joako | F|shie, Honestly I can say I don't know the answer. But I would assume the answer is most certainly: yes |
| 05:27.59 | F|shie | joako: that much i know :P i believe agi functionality cannot be used within an eagi? |
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| 07:31.51 | youjelly | hello o/ |
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| 09:26.27 | chrisq | is there a way forward a call from one asterisk to another using Dial() and retain the realm/domain? |
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| 09:31.40 | Ice_Strike | I am using a Zoiper software phone and it is not clear when making a call. |
| 09:31.46 | Ice_Strike | There is an error: |
| 09:31.47 | Ice_Strike | channel.c:3168 __ast_read: Dropping incompatible voice frame on SIP/test-0000001d of format ulaw since our native format has changed to 0x2 (gsm) |
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| 09:34.45 | Ice_Strike | Oh its working now |
| 09:35.28 | Ice_Strike | I just restarted software |
| 09:41.47 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 09:47.09 | Ice_Strike | Anyone tried using VoipBuster to connect in Asterisk server? |
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| 10:25.08 | *** join/#asterisk gusto (~gusto@88.128.203.255) |
| 10:25.36 | gusto | hey, how are you doing folks? |
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| 11:30.09 | Ice_Strike | I am good |
| 11:30.12 | Ice_Strike | you? |
| 11:30.33 | gusto | well |
| 11:30.34 | Ice_Strike | I am having problem with CALLERID |
| 11:30.45 | gusto | ah |
| 11:30.49 | Ice_Strike | Number is not appearing |
| 11:30.51 | gusto | i am not even that far |
| 11:30.54 | Ice_Strike | see http://pastebin.com/w8xTbXeS |
| 11:31.05 | gusto | but am going to fix that as well sometimes |
| 11:31.13 | gusto | as soon as i get my dsl connection back |
| 11:35.38 | kaldemar | Ice_Strike: you're most likely setting it to a value that your ITSP does not allow you to. |
| 11:36.15 | gusto | what values are there? |
| 11:37.03 | gusto | i saw that such thing was set to n.america on my pap2t device per default, then i set it to something more common |
| 11:37.11 | gusto | but i did not try it out |
| 11:37.17 | Ice_Strike | kaldemar which voip provider would you recommend that allow me to set callerid |
| 11:37.35 | gusto | the most ones do |
| 11:37.36 | Ice_Strike | im using voipbuster |
| 11:53.02 | gusto | dont they have also capacity problems sometimes? |
| 11:55.40 | gusto | do voip providers also use DNS resolving of that telephone numbers so that they do not need to connect through PSTN when a VoIP number calls another VoIP number they can connect to each other directly over the internet, right? |
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| 12:54.03 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 12:54.04 | [sr] | howdy |
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| 12:56.01 | [sr] | howdy |
| 12:56.11 | [sr] | ops, sorry for the echo |
| 12:56.30 | [sr] | people, the dialparties.agi script, is supposed to be part of * source ? |
| 12:57.49 | [sr] | ops no, it's freepbx |
| 12:57.51 | [sr] | sorry |
| 12:57.51 | [sr] | :P |
| 12:59.28 | gusto | ?? |
| 13:05.16 | [sr] | one thing |
| 13:05.24 | [sr] | i want to compile the ogg format support |
| 13:05.42 | [sr] | i already have libogg-dev on the system, but still not available :s |
| 13:05.52 | gusto | ogg format support that sounds interesting |
| 13:06.33 | [sr] | yes, format_ogg_vorbis from the format interpretters |
| 13:07.51 | gusto | do you mean CELT? |
| 13:09.30 | [sr] | if you tell me what CELT is... sorry, dont know |
| 13:09.49 | gusto | http://en.wikipedia.org/wiki/CELT |
| 13:10.29 | gusto | that's the only one codec i found in ekiga, that is supported, directly supported vorbis i have not seen in any softphone and of course neither somewhere else |
| 13:12.06 | gusto | but i see that CELT works only between 32 and 96 kHz, that would be overkill for VoIP, according to wikipedia |
| 13:14.24 | gusto | i would be more interested in speex, that seems to be a better sounding codec to me |
| 13:16.25 | gusto | [sr]: how did you came to that idea to use ogg/vorbis to asterisk? |
| 13:16.37 | gusto | is curious |
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| 13:23.08 | [sr] | gusto: just want to compil :P |
| 13:30.25 | gusto | well, you just could disable that vorbis plugin, or not? |
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| 13:44.59 | [sr] | yes |
| 13:50.20 | [sr] | brb |
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| 14:08.22 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 14:21.03 | Ice_Strike | How to automatically press 1234 after dailing a number? |
| 14:21.07 | Ice_Strike | like this: |
| 14:21.14 | Ice_Strike | exten => 501,n,Dial(SIP/+447xxxxxxx@voipms) |
| 14:21.20 | Ice_Strike | exten => 501,n,wait(2) |
| 14:21.31 | Ice_Strike | then auto press digit 1234 |
| 14:21.33 | Ice_Strike | ? |
| 14:25.24 | gusto | well, to my understanding it would not go that far to even for the wait statement, because you ve already dialed something |
| 14:25.35 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 14:26.51 | Ice_Strike | no |
| 14:27.10 | Ice_Strike | when calling a number, the operatar say press 1234 |
| 14:27.21 | Ice_Strike | i want asterisk dail plan to do it automatically |
| 14:31.01 | gusto | yes, that was something i tried too, but i gave up |
| 14:41.30 | *** join/#asterisk RogersJA (~RogersJA@S0106001346cbd191.gv.shawcable.net) |
| 14:43.36 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 15:06.08 | *** join/#asterisk Russ (~russ@ip68-4-185-173.oc.oc.cox.net) |
| 15:21.08 | jpsharp | Ice_Strike: Use the SendDTMF command after Wait. |
| 15:24.13 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 15:29.06 | Ice_Strike | jpsharp like this? exten => 501,n,SendDTMF(1234) |
| 15:32.53 | Ice_Strike | doesnt seem to work |
| 15:33.51 | *** join/#asterisk tomwish (~Tom@75-172-177-253.phnx.qwest.net) |
| 15:40.41 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 15:40.41 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:50.50 | *** join/#asterisk timahvo1 (~rogue@197.179.253.139) |
| 16:02.05 | chrisq | anyone using ${SIPFROMDOMAIN} to set outgoing sip domain? |
| 16:02.21 | chrisq | it doesn't seem to work for me when using it in my dialplan |
| 16:10.37 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 16:10.40 | [sr] | u gauys are lucky |
| 16:10.42 | [sr] | i'm back! |
| 16:10.45 | [sr] | guys |
| 16:14.03 | gusto | and? |
| 16:14.19 | gusto | what is *.pt [sr] ? |
| 16:14.22 | gusto | portugal? |
| 16:14.26 | [sr] | yes |
| 16:24.40 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.130.16) |
| 16:24.46 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 16:51.19 | chrisq | what i want to manipulate is the sip header "To" which is stripped of the domain part when forwarded to a different asterisk server from a dialplan |
| 16:56.40 | pabelanger | chrisq: setup a sip proxy |
| 16:59.00 | chrisq | pabelanger: asterisk can not do this? |
| 17:00.00 | [sr] | where do i configure the video part of SIP? having this: ignoring 'video' media offer because port number is zero |
| 17:00.01 | pabelanger | chrisq: if you have to modify sip headers, you want a sip proxy, not asterisk. Asterisk has limited support for this, EG: SIP_HEADER function |
| 17:00.28 | *** join/#asterisk nameless` (~nameless`@mal35-2-82-228-176-119.fbx.proxad.net) |
| 17:00.32 | nameless` | hi there |
| 17:00.50 | nameless` | is is possible to authentify users through PAM module ? |
| 17:01.29 | pabelanger | nameless`: for what? |
| 17:01.51 | chrisq | pabelanger: yes, i figured out as much about that function. is there another way to dial to a specific context on the other asterisk server? |
| 17:02.39 | pabelanger | chrisq: yes, just setup a new peer on the other server |
| 17:02.47 | pabelanger | then define the contect= |
| 17:02.51 | pabelanger | context= |
| 17:03.29 | chrisq | pabelanger: ok, thanks, and this doesnt create a huge overhead? |
| 17:04.00 | pabelanger | I couldn't tell you. Its going to require some work |
| 17:04.29 | chrisq | cause i guess i could also dial() and prefix the username with domain_domname_ or something |
| 17:05.09 | pabelanger | what are you trying to do |
| 17:05.14 | chrisq | like dial(SIP/astpeer/<domain>_username) |
| 17:05.35 | atan | What's the deal with forcing Asterisk to play a ringing tone while using Dial()? As it is now, DIAL(SIP/123,30,r) the r isn't cutting it. Phone hears blank. |
| 17:05.49 | chrisq | pabelanger: i have an asterisk server on the net which forwards calls to other asterisks which are in different locations |
| 17:06.16 | chrisq | sp the SRV record is pointing at this "hub", which then forwards it to the correct asterisk |
| 17:06.17 | nameless` | pabelanger: i'd like user@domain.org to be authentified with through PAM |
| 17:06.29 | nameless` | pabelanger: instead of having the password in a file, i want it to be authentified using PAM |
| 17:06.35 | atan | Hmm uppercase R seems to do the trick though. Interesting. |
| 17:06.39 | pabelanger | chrisq: either use dundi or setup a SIP proxy |
| 17:06.59 | pabelanger | nameless`: best you can get it ldap |
| 17:07.13 | p3nguin | atan: R is not valid, so all you did was get rid of the r option. |
| 17:07.41 | nameless` | pabelanger: and maybe with RADIUS and radius using pam ? :) |
| 17:07.49 | p3nguin | In other words, stop trying to use the r option when you don't need to use it. |
| 17:08.07 | atan | Hmm. Noted. |
| 17:08.11 | p3nguin | Dial(SIP/device,timeout) |
| 17:08.34 | chrisq | pabelanger: my problem is that these asterisk "end point" servers are on dhcp, will dundi solve this too? |
| 17:08.47 | p3nguin | Servers on DHCP? |
| 17:09.22 | *** join/#asterisk timahvo1 (~rogue@41.81.106.96) |
| 17:09.22 | pabelanger | nameless`: there is no radius support for SIP, unless you write something. Best you get is file, database or ldap |
| 17:09.27 | chrisq | p3nguin: this is why i have a dual quadcore 1u box on the net, to connect them all |
| 17:10.00 | pabelanger | chrisq: no, dundi is just the routing portion. It does not care about DHCP |
| 17:10.08 | nameless` | pabelanger: for SIP ? i'm talking about asterisk |
| 17:10.30 | pabelanger | nameless`: what are you trying to authenticate |
| 17:11.15 | chrisq | pabelanger: ok, thanks for all the help |
| 17:11.38 | nameless` | pabelanger: i own a server at domaine.foo, i'd like to provide my user user@domain.foo the ability to call each other (user1 with user2 etC.) |
| 17:11.53 | pabelanger | nameless`: call how? |
| 17:12.03 | nameless` | pabelanger: using random sip client like jitsi |
| 17:12.22 | nameless` | pabelanger: since i already set up a lot of services all of them using the same login/password with pam |
| 17:12.34 | nameless` | pabelanger: i wanted the sip password user@domain.foo to be the same |
| 17:12.43 | pabelanger | nameless`: Right, SIP. Asterisk only supports the following authenticate: there is no radius support for SIP, unless you write something. Best you get is file, database or ldap |
| 17:13.57 | nameless` | pabelanger: ok, too bad |
| 17:15.15 | atan | If Dial() gets an ANSWER and the call is ended it stops there in the dial plan right (hangup)? Unless you specify that it should go on |
| 17:15.27 | pabelanger | If you want to centralize your authentication for SIP, use LDAP. Then use can use LDAP's interface for authentication modules |
| 17:15.33 | atan | ^ as such something like GotoIf($[${DIALSTATUS} = ANSWER]?hangup); would be nonsense, right? |
| 17:15.53 | nameless` | pabelanger: i already use PAM for all my services |
| 17:16.07 | pabelanger | atan: yes, Dial() will block unless you tell it to continue in the dialplan |
| 17:16.11 | nameless` | pabelanger: i can't move to LDAP anymore unless i spend a hell of time reconfiguring the services |
| 17:16.46 | pabelanger | nameless`: Then you need to write something to add pam support into asterisk. Make sure you submit the patch upstream |
| 17:17.18 | nameless` | pabelanger: did you get to the code already ? how much time do you estimate ? |
| 17:17.28 | nameless` | pabelanger: i can do that yes, but i'd prefer not to go this way :) |
| 17:20.00 | pabelanger | nameless`: not sure |
| 17:20.04 | [sr] | people |
| 17:20.11 | [sr] | is there anyone here from dahdi dev's? |
| 17:20.24 | [sr] | ok going -dev |
| 17:20.33 | pabelanger | ~ask |
| 17:20.33 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:20.45 | p3nguin | atan: If you want to know if the DIALSTATUS was ANSWER, check it in extension h. |
| 17:21.08 | atan | p3nguin, actually by it closing the call off I can remove unneeded lines in my dialplan. :D score |
| 17:21.18 | p3nguin | Even better! |
| 17:21.45 | atan | Next on my list to mess with is voicemail via odbc. Going by http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/I_section12_tt1578.html I have the storage setup correctly but is it possible to define voicemail.conf entirely in MySQL? Box & all, not just the messages? |
| 17:22.19 | atan | Correction. Possible I assume so. Might you know a guide or place I could turn to get it setup? :-) |
| 17:22.39 | [sr] | pabelanger: just 'd like to know if cards like "HFC-S ISDN BRI card" would be supported soon, i already see a: zaphfc- |
| 17:23.28 | pabelanger | [sr]: supported by who? |
| 17:23.54 | [sr] | pabelanger: dahdi |
| 17:24.28 | [sr] | i know there is a dahdi-HFC project, but if this would be supported by the main dev of dahdi, better |
| 17:25.16 | pabelanger | [sr]: Yes, which manufacture? Your question is for product management. Since most of dahdi's development is driven by actual products |
| 17:26.17 | [sr] | pabelanger: i have 2 of this cards right now, one i have no idea the manufactor, lspci only show's me that info (HFC-S ISDN BRI card), but i have another that has: "ASUSTeK Computer Inc. ISDNLink P-IN100-ST-D" |
| 17:27.22 | pabelanger | [sr]: I wouldn't expect support soon(tm), unless somebody like Digium or Xorcom is working on a new board to support it. Community development for DAHDI is different then Asterisk |
| 17:28.10 | [sr] | pabelanger: is there a specific place to give info about it? i mean, at least help on dev as testing, since i'm not a C coder |
| 17:28.55 | pabelanger | [sr]: Have you tried the latest DAHDI drivers and setup BRI? |
| 17:29.06 | [sr] | i have latest, 2.6.1 |
| 17:29.38 | [sr] | its not supported, the driver part says: zaphfc- |
| 17:29.46 | [sr] | no "+" plus sign |
| 17:31.20 | pabelanger | right, so you can ask in asterisk-dev (IRC or mailing list) and ask about up coming support. But like I said, unless you have an existing patch you want to merge, development is mostly done by Digium and Xorcom |
| 17:31.54 | pabelanger | which means contacting their product management teams |
| 17:32.26 | [sr] | that would be an headache! |
| 17:33.45 | *** join/#asterisk angryuser_laptop (~angryuser@2a01:e34:ee09:8850:e913:d01c:9357:a33d) |
| 17:33.57 | pabelanger | well, that's how they make their money. Board sales / development |
| 17:34.13 | [sr] | ya i understand that part |
| 17:36.29 | [sr] | but a few versions back, like 2.5 this zaphfc didn't appear, so i guess there's about 50% of the support |
| 17:50.27 | *** join/#asterisk tomwish (~Tom@75-172-177-253.phnx.qwest.net) |
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| 18:22.37 | [sr] | pabelanger: small help, how do i configure this? chan_sip.c:9142 process_sdp: ignoring 'video' media offer because port number is zero |
| 18:25.24 | pabelanger | [sr]: have the device send a port number? |
| 18:25.42 | *** join/#asterisk bytemaster (~tyrtyrty@cpc3-basl7-0-0-cust252.basl.cable.virginmedia.com) |
| 18:26.30 | [sr] | no camera |
| 18:26.38 | [sr] | hum i get i figure it out |
| 18:35.58 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 18:39.50 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 18:45.04 | atan | Right now when a SIP peer Dial()s a call to my ITSP it's volunteering information about my SIP device. Like, sending the auth of my device. Is there a way to tell Asterisk to kindly not forward this information and only use what was provided in the sip friend to place the call? |
| 18:47.26 | [TK]D-Fender | atan: fromuser <- |
| 18:48.25 | atan | [TK]D-Fender fromuser is set within the sip.conf peer[] but it's still sending it. As a result when dialing internal numbers (well, DIDs that are "internalish") it seems to provide additional details it need not |
| 18:48.35 | [TK]D-Fender | show us |
| 18:48.42 | atan | [TK]D-Fender, or are you suggesting fromuser be set on each SIP device that is placing the call? |
| 18:49.08 | [TK]D-Fender | no, on your ITSP peer |
| 18:51.35 | *** join/#asterisk asterisk-Tester (~ramy@80.79.159.206) |
| 18:53.46 | atan | [TK]D-Fender, here's the bulk of it I think this is everything if I missed something please advise. http://pastebin.com/iaCmqD3K |
| 18:54.35 | *** join/#asterisk Ice_Strike (Ice_Strike@94-192-112-241.zone6.bethere.co.uk) |
| 18:54.37 | atan | [TK]D-Fender, keeping in mind here all calls except those being routed back function just fine. It's only if one happens to call a DID on the same account. |
| 18:56.55 | Ice_Strike | Why SendDTMF does not execute after Dial command? |
| 18:57.07 | Ice_Strike | see example http://pastebin.com/hZ96uU9t |
| 18:57.28 | Ice_Strike | When making a call, the operater say enter digit number |
| 18:57.34 | [TK]D-Fender | atan: new PB. Do not mask ANYTHING except the SECRET, and show the complete call with SIP DEBUG enabled. You get 1 shot at this.... |
| 18:57.38 | Ice_Strike | but does not seem to work? |
| 18:58.01 | [TK]D-Fender | Ice_Strike: Of course not, because if it gets to the 2nd priority the DIal has ENDED |
| 18:58.53 | Ice_Strike | [TK]D-Fender Ahh ok, what do I need to do to solve my issue? |
| 18:59.27 | Ice_Strike | I like automatically enter keys when operater say enter a number |
| 18:59.46 | Ice_Strike | basically after 3 seconds when making a call |
| 18:59.56 | [TK]D-Fender | Ice_Strike: "core show application dial" <- |
| 19:00.33 | Ice_Strike | hmmm |
| 19:01.03 | Ice_Strike | yes I have tried that before |
| 19:01.25 | Ice_Strike | Dial(SIP/+4479999999999@voipms,,D(wwwwwwwww1w2w3w4)) |
| 19:01.28 | Ice_Strike | like that |
| 19:01.59 | [TK]D-Fender | better |
| 19:03.08 | Ice_Strike | yes I have done that but does not work? - operater keep saying enter the numbers |
| 19:04.18 | [TK]D-Fender | maybe you're not waiting enough.... call yourself to see |
| 19:07.25 | WIMPy | [sr]: dahdi_hardware detecting HFC-S cards isn't really new. |
| 19:10.01 | atan | [TK]D-Fender ugh it's messy because there are more devices connecting and the sip debug is catching them all. |
| 19:10.09 | atan | Actually it's too long it moves too fast. |
| 19:10.35 | [sr] | WIMPy: hum, but not for all cards right |
| 19:14.31 | *** join/#asterisk urvg4 (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 19:15.27 | atan | [TK]D-Fender, will http://pastebin.com/sbhCY9yB suffice? Does that include everything you need to get an idea what's going on? |
| 19:17.22 | WIMPy | [sr]: What "all cards"? |
| 19:17.54 | [sr] | WIMPy: the HFC cheap cards... the one's used 12years ago for internet connection |
| 19:18.11 | [TK]D-Fender | No sip debug on your outbound call, things masked where I told you not to. Incomplete configs. As I said you got 1 shot. This just wasted it. Maybe someone else will look at this.... |
| 19:18.23 | *** join/#asterisk philipp64|laptop (~chatzilla@66-232-81-197.ctcweb.net) |
| 19:19.32 | atan | I fail to understand why we're not allowed to mask identifiable details from public pastebins. Is there a reason for this? |
| 19:21.45 | *** join/#asterisk HyperNerdV2_ (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 19:22.42 | WIMPy | [sr]: They are all the same and support situation hasn't changed, except for the name. |
| 19:23.03 | WIMPy | Instead of zaphfc it's dahdi-hfcs now. |
| 19:23.21 | [sr] | ya but that project uses an old version of dahdi |
| 19:25.02 | Ice_Strike | [TK]D-Fender Yes I have dailed manually via software phone and it work. |
| 19:25.13 | WIMPy | Yes, new versions are becoming incompatible att the time :-( |
| 19:25.25 | WIMPy | s/att/all/ |
| 19:25.27 | Ice_Strike | But does not work with D(wwwwwwwww1w2w3w4) :( |
| 19:25.38 | Ice_Strike | Is it possible to debug D(wwwwwwwww1w2w3w4) ? |
| 19:25.49 | Ice_Strike | to make sure it is really working |
| 19:29.33 | Ice_Strike | When I do: Dial(SIP/+4479999999999@voipms,,D(wwwwwwwww1w2w3w4)) |
| 19:29.49 | Ice_Strike | the tone is blank for a couple seconds. |
| 19:29.59 | pabelanger | It won't work |
| 19:29.59 | Ice_Strike | silent |
| 19:30.02 | pabelanger | not with SIP |
| 19:30.07 | Ice_Strike | damn :( |
| 19:30.14 | pabelanger | SendDTMF |
| 19:30.23 | Ice_Strike | Yes I have done SendDTMF that dont work too! |
| 19:30.42 | Ice_Strike | When making a call, the operater say enter the numbers |
| 19:30.52 | Ice_Strike | I want asterisk to enter the number automatically |
| 19:31.54 | [TK]D-Fender | pabelanger: This should have nothing to do with SIP |
| 19:32.17 | Ice_Strike | <PROTECTED> |
| 19:32.59 | [TK]D-Fender | Ice_Strike: Show me the proper call and config with D() |
| 19:33.22 | [TK]D-Fender | Ice_Strike: and no sense showeing him the call of senddtmf after.. as I said that is useless there |
| 19:33.30 | pabelanger | D() won;t work. He wants to sendDTMF after the channel is answered |
| 19:33.31 | Ice_Strike | [TK]D-Fender you mean original number? |
| 19:33.56 | [TK]D-Fender | Dial(SIP/peer/numer,,D()) |
| 19:34.02 | *** join/#asterisk beardy (~beardy@unaffiliated/beardy) |
| 19:34.10 | [TK]D-Fender | fix your dialplan apply. Paste new call & config |
| 19:34.32 | [TK]D-Fender | and you'd better have SIP DEBUG enabled for it. |
| 19:35.28 | Ice_Strike | I have PM you. |
| 19:35.42 | Ice_Strike | This is what I actually have |
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| 20:33.19 | atan | If you user is defaultuser=123 name=123 how can you access the 123 from the dialplan? Looking here https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables I see ${CHANNEL} which gives me slightly too much information. I just want the name not the -uuid thing. |
| 20:34.20 | gusto | well no |
| 20:34.48 | gusto | the user is only the user you use for the peer connection, you can put in the peer into your dialplan, and in fact you should have |
| 20:35.10 | gusto | `book |
| 20:35.14 | gusto | ~book |
| 20:35.14 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:35.16 | gusto | so |
| 20:35.39 | gusto | so, i find this documentation the only usable one |
| 20:37.21 | atan | Just need to figure out which section this is in :-) haven't had the joy of reading the whole thing yet :D |
| 20:40.22 | Ice_Strike | Why does D() does not work? |
| 20:40.23 | Ice_Strike | exten => 501,n,Dial(SIP/+44797xxxxxx5@voipms,,D(wwwwwww0w7w8w5w4w2w2w2w2w2w2w)) |
| 20:41.35 | Ice_Strike | it seem D() is not executing when the line is open |
| 20:47.22 | *** part/#asterisk asterisk-Tester (~ramy@80.79.159.206) |
| 20:52.39 | bluregard | hello all |
| 20:53.18 | bluregard | I'm wondering, when doing a database write, is it better to hand it off to an AGI script or use func_odbc writesql? Anyone have an opinion on this? |
| 20:54.56 | jpsharp | Depends on your system load. Firing up an AGI forks off another process, so if you've got a heavily loaded machine you could forkbomb yourself. |
| 20:56.23 | bluregard | yeah |
| 20:59.19 | Ice_Strike | psharp I am having problem with D() and SendDTMF |
| 20:59.33 | Ice_Strike | It is not doing what I wanted to do. |
| 20:59.58 | Ice_Strike | When making a call, after 2 seconds, it should enter the number automatically |
| 21:00.04 | Ice_Strike | exten => 501,n,Dial(SIP/+44797xxxxxx5@voipms,,D(wwwwwww0w7w8w5w4w2w2w2w2w2w2w)) |
| 21:00.08 | Ice_Strike | dont work |
| 21:00.21 | Ice_Strike | even using SendDTMF |
| 21:02.45 | WIMPy | Ice_Strike: Use Option A. |
| 21:04.07 | Ice_Strike | Option A? |
| 21:04.57 | Ice_Strike | A(x): Play an announcement (x.gsm) to the called party. |
| 21:05.19 | WIMPy | That would be after answer. |
| 21:06.03 | Ice_Strike | Do you mean like this |
| 21:06.08 | Ice_Strike | exten => 501,n,Dial(SIP/+44797xxxxxx5@voipms,,A(wwwwwww0w7w8w5w4w2w2w2w2w2w2w)) |
| 21:06.24 | WIMPy | No, you have to use files. |
| 21:06.45 | Ice_Strike | Why do I need to use files?? |
| 21:07.00 | Ice_Strike | I want asterisk to enter the numbers when it making a call |
| 21:07.12 | WIMPy | You just quoted the documentation yourself. |
| 21:07.31 | Ice_Strike | You said use Use Option A. |
| 21:07.57 | WIMPy | Yes, and that needs files. |
| 21:08.31 | WIMPy | I don;t know of any direct way. |
| 21:10.38 | pabelanger | Just use M or U, and create a context with senddtmf |
| 21:12.40 | Ice_Strike | hmm let me try that |
| 21:12.56 | Ice_Strike | this giving me a heachache now, been tring to get it work almost all day! |
| 21:16.40 | Ice_Strike | Ok so is it like this: |
| 21:16.41 | Ice_Strike | exten => 501,n,Dial(SIP/+44797xxxxx@voipms,,M(SendNumber)) |
| 21:16.48 | Ice_Strike | [SendNumber] |
| 21:17.23 | Ice_Strike | exten => 501,1,SendDTMF(wwwwwwwwwwwwwww0ww7ww8ww5ww4ww1ww1ww1ww1ww1ww1w |
| 21:17.34 | Ice_Strike | thats right @pabelanger? |
| 21:18.01 | pabelanger | no |
| 21:18.25 | pabelanger | *CLI> core show application Dial |
| 21:18.49 | Ice_Strike | thanks! |
| 21:31.43 | Ice_Strike | @pabelanger I have read it but I am almost correct |
| 21:31.53 | Ice_Strike | so its [macro-SendNumber] |
| 21:32.40 | pabelanger | right |
| 21:32.41 | pabelanger | Also |
| 21:32.44 | pabelanger | *CLI> core show application Macro |
| 21:32.48 | pabelanger | will explain how to use it |
| 21:33.57 | Ice_Strike | pabelanger like this? http://pastebin.com/mKy0tUrh |
| 21:35.11 | pabelanger | Yes, looks right |
| 21:36.52 | Ice_Strike | @pabelanger I have tested it.. it is like long pause/silent for 10 seconds but did not use SendDTMF it seem |
| 21:37.46 | pabelanger | pb your console output |
| 21:37.59 | Ice_Strike | debug on? |
| 21:38.48 | Ice_Strike | sip set debug on ? |
| 21:39.59 | gusto | life |
| 21:41.10 | Ice_Strike | @pabelanger .. |
| 21:42.33 | gusto | some say, life is like a flower, i say life is like a DVB signal, everytime you need it it fades away |
| 21:43.11 | gusto | are you still solving that call issue? |
| 21:43.38 | pabelanger | ~collectdebug |
| 21:43.39 | infobot | collectdebug is, like, a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 21:54.23 | p3nguin | ice_strike: Dial(SIP/voipms/0044797xxxxx) |
| 22:01.42 | *** join/#asterisk devdvd (~Jason@c-67-163-163-124.hsd1.wv.comcast.net) |
| 22:02.09 | Ice_Strike | @pabelanger I have PM you a log |
| 22:02.55 | devdvd | hey, i just upgraded to 10.4.0 from 1.6.2.9 and now I have issues where when the remote end terminates the call, the system never knows about it, so my sip phone connected to asterisk will never hang up on its own. Ideas? |
| 22:04.29 | devdvd | the path out is (sip phone -> asterisk -> sip trunk -> pstn |
| 22:07.45 | devdvd | however, i find if the call goes the other way (comes in from the pstn to my sip extension) |
| 22:07.49 | devdvd | if the remote end hangs up |
| 22:07.51 | devdvd | it works just fine |
| 22:11.00 | *** join/#asterisk tomwish (~Tom@75-172-177-253.phnx.qwest.net) |
| 22:18.59 | Ice_Strike | pabelanger you there? |
| 22:24.28 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:58.18 | gusto | you know what, i like only the pap2t ata adapter, i ve tried that spa112 that i have also bought, that is crap |
| 22:58.38 | gusto | it does not allow me to configure it right and always does what i do not want |
| 22:58.53 | gusto | and no idea how to upgrade the firmware of that device |
| 23:24.14 | *** join/#asterisk Russ (~russ@ip68-4-185-173.oc.oc.cox.net) |
| 23:34.20 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 23:40.18 | *** join/#asterisk mazpe (addd995a@gateway/web/freenode/ip.173.221.153.90) |
| 23:40.33 | mazpe | i'm getting a WARNING[9773] db.c: Unable to open Asterisk database '/var/lib/asterisk/astdb': Permission denied |
| 23:41.04 | gusto | you have it on openwrt? |
| 23:41.17 | mazpe | currently /var/lib/asterisk/ is owned by root... should it be asterisk:asterisk ? |
| 23:41.45 | gusto | do not care so much by whom it is owned, when it gets hacked, you are pissed anyway |
| 23:41.51 | WIMPy | Unless you're running Asterisk as root... |
| 23:41.59 | WIMPy | LOL |
| 23:42.19 | gusto | and on openwrt, it is most likely root, right? |
| 23:42.26 | gusto | where we come to my first question |
| 23:44.08 | WIMPy | As long as it isn;t Windows, I don't see the difference. |
| 23:44.23 | gusto | however |
| 23:44.56 | gusto | astdb gets created on start, so either he has no directory, or it has not enough permissions (the programm) to create it inside the existing one |
| 23:45.02 | gusto | not much of a deal |
| 23:45.59 | mazpe | -rw-r--r--. 1 root root 8192 May 5 18:44 astdb |
| 23:46.15 | mazpe | its running on centos |
| 23:46.21 | gusto | however |
| 23:46.42 | gusto | and the permissions for /var/lib/asterisk? |
| 23:47.13 | mazpe | drwxr-xr-x. 11 root root 4096 May 1 10:45 asterisk |
| 23:47.50 | gusto | and look up 'ps aux | grep -i asterisk' to see as what user you are running the asterisk daemon |
| 23:48.01 | mazpe | the error that i'm getting |
| 23:48.04 | mazpe | [May 5 18:47:22] WARNING[9792] db.c: Unable to open Asterisk database '/var/lib/asterisk/astdb': Permission denied [May 5 18:47:22] WARNING[9792] db.c: Database unavailable |
| 23:48.13 | mazpe | ok |
| 23:48.21 | gusto | but, first of all ... that error ... it is not really an error, i ignored that also |
| 23:48.44 | mazpe | running as root |
| 23:48.52 | WIMPy | It will likely give you some trouble. |
| 23:48.58 | gusto | so ... it does not make any difference when you have it ... i think that astdb is even a module, and if so, you can kick it out and solve it thus |
| 23:49.09 | gusto | i did not see any |
| 23:49.23 | mazpe | hmmmm |
| 23:50.11 | WIMPy | Even if you don;t use it in your dialplan, it's used for SIP registrations, dundi and some other stuff. |
| 23:51.37 | gusto | who cares about sip registrations, it reads its config from the sip.conf file and that is it |
| 23:51.55 | gusto | and actually it is better to have him not to remember anything he has done, than to let him do |
| 23:52.05 | WIMPy | Yes, and if you reload them, it forgets where to find its peers. |
| 23:52.13 | gusto | no |
| 23:52.27 | WIMPy | We just had that the other day. |
| 23:52.29 | gusto | that is pure speculation |
| 23:53.07 | WIMPy | database show |
| 23:53.49 | file | if you *restart* Asterisk it won't be able to retrieve the existing registrations, but if you reload it won't touch 'em |
| 23:54.14 | *** join/#asterisk jaddison (~Adium@184.66.16.213) |
| 23:54.35 | gusto | and what is the problem? |
| 23:54.40 | gusto | it can register again |
| 23:54.44 | WIMPy | Maybe it was a core reload, but I think it was just about sip reload. |
| 23:54.52 | WIMPy | Sure it can. |
| 23:54.55 | file | it's just a period of time where the device is unreachable |
| 23:54.59 | WIMPy | But it will be unreachable until then. |
| 23:55.17 | file | which can vary depending on configuration of the maximum/minimum expiration time |
| 23:55.36 | gusto | ah, you mean the peers behind you? |
| 23:55.39 | gusto | i do not care |
| 23:55.50 | WIMPy | Most probaly your phones. |
| 23:55.51 | gusto | that would not be peers, but rather devices |
| 23:55.52 | jaddison | It seems easy to find reasonably cheap and decent outgoing gateways, but harder to find incoming that are cheap and decent. I'm thinking per minute costing at the moment. It's all a bit hazy for me overall. Any tips on pricing or suggestions of providers? It's a low volume, calling card type of business, to start with. I'd appreciate your input, thanks. |
| 23:55.55 | gusto | yes yes |
| 23:55.57 | gusto | now i understand |
| 23:56.18 | gusto | but i set it to 60 seconds ... and so ... i do not have to care about that |
| 23:59.09 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |