00:15.35 | *** join/#asterisk mmourani (~Adium@bas1-montreal23-1242526158.dsl.bell.ca) |
00:15.45 | mmourani | hi all |
00:15.50 | mmourani | hope u r doing fine |
00:15.57 | mmourani | I have a quick question |
00:16.03 | mmourani | I know its saturday |
00:16.04 | mmourani | ;) |
00:16.16 | WIMPy | Or sunday |
00:17.12 | *** join/#asterisk serafie (~erin@24.96.64.240) |
00:17.44 | mmourani | I just want to know what do you suggest as a WEBUI stats gathering based on destination, country, vendors, customers, etc… for wholesale open source i can easily install and work on ? |
00:18.19 | mmourani | yeah sunday for some |
00:18.27 | mmourani | :) |
00:18.56 | [TK]D-Fender | <PROTECTED> |
00:18.59 | WIMPy | I guess the best answer is: The one you write yourself. |
00:19.03 | [TK]D-Fender | ^^ |
00:19.07 | [TK]D-Fender | and that is what the answer becomes |
00:19.28 | [TK]D-Fender | Its your custom data. You're going to have to sort it into something meaningful |
00:19.45 | mmourani | i know but it takes a lot of time to redo something from scratch |
00:20.09 | [TK]D-Fender | It isn't done |
00:20.15 | WIMPy | Sometime it takes less time thatn trying out various finished products and configuring them. |
00:20.19 | [TK]D-Fender | So it can't be re-done "from scratch" |
00:20.44 | mmourani | so you suggest i build my own |
00:21.13 | [TK]D-Fender | I'd certainly start be defining what you want to do more precisely so you can have a plan of attack |
00:21.16 | WIMPy | Depending on your need, chances are you have to. |
00:21.30 | [TK]D-Fender | Your data is in your own format and you were just throwing nouns around.... |
00:23.25 | mmourani | thanks for the info |
00:23.32 | mmourani | I will then think about it |
00:26.56 | *** part/#asterisk mmourani (~Adium@bas1-montreal23-1242526158.dsl.bell.ca) |
00:29.22 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
00:29.38 | *** join/#asterisk rogersja (~rogersja@162-192-162-17.lightspeed.rlghnc.sbcglobal.net) |
00:35.39 | *** join/#asterisk classix (salven@64.16.220.132) |
00:38.03 | *** join/#asterisk CeBe (~CeBe@port-92-206-49-84.dynamic.qsc.de) |
00:40.38 | *** join/#asterisk Pullphinger (~Pullphing@c-24-13-69-42.hsd1.il.comcast.net) |
00:41.34 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
00:46.15 | *** join/#asterisk Synx|hm (~Synx@unaffiliated/synx-hm/x-1623004) |
00:46.15 | Synx|hm | . |
00:46.22 | *** part/#asterisk Synx|hm (~Synx@unaffiliated/synx-hm/x-1623004) |
00:46.35 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.29) |
00:47.42 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
00:53.01 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
00:53.45 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
00:54.22 | danfromuk | Hi, I'm trying to understand Masquerading. If a channel is being masqueraded, does that mean the Original channel has ended? |
00:56.47 | *** join/#asterisk bluefirecorp (~gredinger@cpe-98-28-80-91.columbus.res.rr.com) |
01:00.31 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
01:06.30 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
01:25.40 | *** join/#asterisk rogersja (~rogersja@162-192-162-17.lightspeed.rlghnc.sbcglobal.net) |
01:34.28 | *** join/#asterisk rogersja_ (~rogersja@162-192-162-17.lightspeed.rlghnc.sbcglobal.net) |
01:35.09 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.33) |
01:44.40 | phix | hey Penguin, after I got rid of insecure=invite,port I know can't phone my numbers, my VoIP provider tells me I am not available |
01:48.04 | phix | Penguin: I added insecure=invite in now I can call my number from a mobile / landline |
01:52.01 | phix | so I need insecure=invite for my VoIP provider sip account? Or have I configured it wrong? |
01:52.11 | phix | (anyone feel free to jump in here) |
01:52.48 | bluefirecorp | Does anyone have any recommendations for a decent SIP gateway? |
01:53.06 | bluefirecorp | I'm mostly looking for a residential type solution. |
01:53.24 | WIMPy | phix: You found out what you need, so where's the question? |
01:53.27 | bluefirecorp | maybe, 2 trunks (this way, calls can go to voicemail while on the phone) |
01:53.37 | WIMPy | bluefirecorp: Gateway to what? |
01:53.52 | phix | WIMPy: Penguin said it is a secure issue to have insecure=invite or insecure=invite,port |
01:53.59 | phix | security* |
01:54.06 | bluefirecorp | Erm, I'm not so great with terms, but it'd be the public gateway. |
01:54.09 | bluefirecorp | PSTN, I believe ? |
01:54.20 | phix | WIMPy: I am not sure of the security implecations |
01:54.20 | WIMPy | phix: Yes, it it. But often neccessary. |
01:54.28 | phix | what's it mean? |
01:54.40 | WIMPy | bluefirecorp: What kind of interface? FXO? ISDN? |
01:55.15 | bluefirecorp | Erm, ethernet. |
01:55.18 | WIMPy | phix: It means the invite can not only come from what you configured as host=. |
01:55.31 | bluefirecorp | Sorry, I'm really, really new to all this. |
01:55.49 | WIMPy | bluefirecorp: Ok, so what protocoll, if not aleady SIP? |
01:55.57 | dijib | AIX? |
01:56.06 | dijib | IAx |
01:56.08 | dijib | BEER |
01:56.11 | phix | WIMPy: I see, can I set it to match on a wildcard pattern of the domain? |
01:56.26 | phix | WIMPy: I know it will come from *.voip-provider.com |
01:56.34 | dijib | find the ip |
01:56.42 | dijib | or does name resolution work? |
01:56.42 | WIMPy | phix: No. YOu can only permit ip ranges or hosts. |
01:57.13 | WIMPy | Name resolution does work, but no reverse lookups are performed. |
01:57.18 | bluefirecorp | SIP |
01:57.25 | bluefirecorp | :P |
01:57.37 | WIMPy | bluefirecorp: A gateway from SIP to SIP? |
01:58.09 | bluefirecorp | Honestly, I have no idea. Pretty much wanting to setup asterisk as a PBX (I believe that is the right term), for my home phone system, which will be connected to it via SIP. |
01:58.44 | WIMPy | How do you connect to the PSTN? |
01:58.46 | bluefirecorp | Also, I'm planning on using the internet to route my traffic out -- cable internet rather than ISDN, T1, or OC lines. |
02:00.11 | WIMPy | How you connect to the internet doesn't really matter, unless you wanty to connect to the PSTn via your IP connection and that isn't good enough. |
02:00.24 | dijib | your going to need to be running a good QOS implimentation bluefirecorp |
02:00.55 | bluefirecorp | Yep. |
02:01.05 | bluefirecorp | Actually, vlans :) |
02:01.06 | WIMPy | That's a good advice, but only helps in one direction. |
02:01.29 | bluefirecorp | I'm more of a network admin, so networks and such are pretty simple -- even more for a home network. |
02:01.33 | dijib | qos for priority, vlans for segmentation |
02:01.45 | dijib | ok |
02:01.56 | dijib | can you prioritize vlan traffic? |
02:02.01 | bluefirecorp | Of course! |
02:02.09 | dijib | well blow me down |
02:02.30 | dijib | and dont tell me you can do it in ddwrt? |
02:02.44 | bluefirecorp | I'm running pfSense, so I'm not sure. :P |
02:02.53 | bluefirecorp | Let me quickly check. |
02:02.57 | dijib | i should get that running on my esxi |
02:03.02 | dijib | 5.5 yall |
02:03.36 | bluefirecorp | I need to get my storage pool setup. |
02:03.51 | bluefirecorp | I'm thinking of going to DAS rather than NAS/SAN. |
02:04.06 | bluefirecorp | Plugging drives in directly rather than iSCSI/NFS.. =/ |
02:04.31 | dijib | anyways bluefirecorp sip.conf is a pretty simple conf |
02:04.36 | bluefirecorp | Sure, not HA, but my NAS has been giving me nothing but problems since I set it back up, fairly sure it has a few failed drives causing it. |
02:04.47 | dijib | ive been able to set it and forget it, |
02:04.59 | bluefirecorp | dijib: It can be rather confusing once you look at scaling it :) |
02:05.09 | dijib | with minimal IP updates, password changes and version variable changes |
02:05.29 | dijib | what do you mean scaling it? |
02:06.21 | bluefirecorp | More than 1 PBX, or constantly changing accounts (adding and removing people, with extension changing daily), or beyond that. |
02:07.07 | bluefirecorp | Failover, load balancing, etc.. the complications of scaling any network, I suppose. |
02:09.07 | phix | WIMPy: I can have more than one host or ip range line ? |
02:09.28 | WIMPy | phix: AFAIK, yes. |
02:09.32 | phix | <3 |
02:09.37 | phix | thnx again buddy |
02:09.44 | dijib | what kindof deployment is this for? i thought you said it was residential |
02:09.47 | dijib | ? |
02:10.24 | phix | dijib: It could be a lebanese residential, so there would be like 50 ppl living there |
02:11.05 | phix | 10 wives, cousions, uncles, aunties, children, etc |
02:11.10 | dijib | lol |
02:11.24 | bluefirecorp | dijib: It's residential, but at the same time -- my home lab is kinda enterprise. |
02:11.30 | bluefirecorp | Or at least designed to be able to scale. |
02:11.32 | bluefirecorp | (mostly for learning) |
02:11.51 | phix | ah, or he is an enthusist :) |
02:11.55 | bluefirecorp | :) |
02:12.05 | bluefirecorp | Just trying to learn the tricks of the trade. |
02:12.25 | phix | but stil could live in the south and have 10 wives ;) |
02:12.28 | bluefirecorp | And I'm completely ignorant with phone systems. |
02:12.28 | WIMPy | There's only one thing you need to learn: to run away. |
02:12.37 | bluefirecorp | That's what they told me about systems. |
02:12.38 | bluefirecorp | >.> |
02:13.02 | dijib | where the fuck is Penguin |
02:13.04 | dijib | ? |
02:13.12 | phix | dijib: I know right, I have wprds for him |
02:13.15 | phix | words* |
02:13.23 | dijib | i want to give him kisses |
02:13.30 | bluefirecorp | ~book |
02:13.30 | 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 |
02:13.31 | dijib | and get a pcie card back from him |
02:13.46 | dijib | give him a cash cut |
02:13.52 | phix | bluefirecorp: I like that book |
02:13.53 | bluefirecorp | Forgot to bookmark it last time I was here. |
02:14.26 | phix | any new editions yet to cover 1.8 / 1.11? |
02:14.39 | phix | oops, 8 or 11 even |
02:14.41 | phix | lo |
02:15.05 | phix | what? it jumps from 1.8 to 11? |
02:15.15 | dijib | Asterisk 11.6.0 built by root @ swisscomm on a i686 running Linux on 2013-10-29 01:46:29 UTC |
02:15.31 | WIMPy | No, there's 10 in between |
02:15.40 | dijib | hold on is my system clock that far off? |
02:15.44 | dijib | oct 29th? |
02:16.04 | phix | Asterisk 1.8.13.1~dfsg-3+deb7u1 |
02:16.28 | bluefirecorp | lol. |
02:16.43 | bluefirecorp | THERE'S YOUR PROBLESM dijib :P |
02:16.56 | phix | dijib: lol |
02:18.08 | dijib | phix: that was the build date, silly me |
02:18.32 | phix | I had a similar issue with time a few days ago, joining computers into a samba Active Directory Domain and I thought I had all of the time correct, turns out two of the computers would join but I would always get kerberos issues when trying to goto a share or apply GPOs. Turns out the time zones were different on those computers :) |
02:19.01 | dijib | lsass will do that |
02:19.09 | dijib | or is that kerberose.exe |
02:19.12 | dijib | ? who cares |
02:20.49 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
02:21.13 | phix | yeah, who cars about samba and windows on this channel :) |
02:21.22 | dijib | hey phix and bluefirecorp do you guys know how to dial a sip uri? |
02:21.26 | phix | dijib: you know of a good GUI for asterisk? |
02:21.44 | dijib | hmmm yes SeRi was telling me about one wtf was it |
02:21.54 | WIMPy | dijib: Just do it? |
02:22.29 | phix | Dial(SIP/outgoingaccount/blah@hostname:port,,)? |
02:22.59 | WIMPy | There's no "account" if you dial an URL. |
02:23.05 | phix | dijib: SeRi is a whore |
02:23.22 | phix | WIMPy: ok :) |
02:23.38 | phix | Dial(SIP/blah@hostname:port/extension,,) ? |
02:23.59 | WIMPy | yes |
02:24.23 | MaliutaLap | samba? I care ... I'm a ballroom/latin dancer ;) |
02:25.04 | phix | MaliutaLap: :D Can you do the dance of a thousand kerberos packets? |
02:25.22 | dijib | phix: Flash Operator Panel 2 |
02:26.21 | MaliutaLap | phix: no, but I do a wicked body roll :P |
02:26.41 | phix | woooo! |
02:26.49 | phix | dijib: commerical hey |
02:27.02 | dijib | huh? |
02:27.33 | phix | dijib: Flash Operator Panel 2 is commerical software |
02:27.58 | phix | can you extend onto it? |
02:28.15 | dijib | i have no working experience with that commercial procudt |
02:30.28 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
02:30.29 | phix | dijib: ok |
02:30.43 | phix | so that was from a google search? |
02:34.12 | dijib | no that was from ugly bean fucker |
02:34.54 | dijib | he will be in in approx 45min |
02:35.04 | dijib | phix can you dial sip uri? |
02:35.06 | dijib | ? |
02:37.44 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
02:41.15 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.40) |
02:59.27 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
03:03.35 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
03:06.35 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.41) |
03:07.14 | *** join/#asterisk Draecos (~Draecos@58-7-54-134.dyn.iinet.net.au) |
03:07.43 | *** join/#asterisk Draecos (~Draecos@58-7-54-134.dyn.iinet.net.au) |
03:11.58 | *** join/#asterisk hardwire (~hardwire@222-83-237-24.gci.net) |
03:17.14 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
03:38.09 | dijib | bluefirecorp: isnt it all in the contexts? |
03:45.48 | bluefirecorp | Erm? |
03:54.12 | *** join/#asterisk serafie (~erin@24.96.64.240) |
03:54.50 | dijib | teh scaling |
03:56.51 | bluefirecorp | Ah |
03:56.54 | bluefirecorp | Right, right. |
03:57.14 | bluefirecorp | I need to understand the underlying principles of phone systems before I take a look at scaling them. |
03:57.16 | bluefirecorp | :P |
03:58.50 | *** join/#asterisk rogersja_ (~rogersja@162-192-162-17.lightspeed.rlghnc.sbcglobal.net) |
04:03.14 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
04:06.37 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.47) |
04:10.08 | *** join/#asterisk rogersja_ (~rogersja@162-192-162-17.lightspeed.rlghnc.sbcglobal.net) |
04:35.14 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
04:39.15 | *** join/#asterisk rogersja_ (~rogersja@162-192-162-17.lightspeed.rlghnc.sbcglobal.net) |
04:53.08 | Penguin | phix: If you have the field "secret" on the sip peer entry for your ITSP, that's a problem. They will NEVER authenticate against you. If they require you to authenticate against them, use "remotesecret" instead of "secret". Don't use insecure until you know what you're doing. |
04:55.14 | Penguin | Using secret in a peer entry requires that the peer authenticate calls. remotesecret does not require auth on calls inbound to asterisk, but will send auth to the remote peer. |
04:59.08 | WIMPy | Where is the difference? |
05:00.20 | Penguin | secret auths against the remote peer an also requires the remote peer to auth against asterisk. |
05:00.42 | Penguin | remote secret auths against the remote peer but does not require the remote peer to auth against asterisk. |
05:00.42 | WIMPy | Between not using secret and using insecure. |
05:01.04 | Penguin | s/ // |
05:01.29 | Penguin | Now that I'm not sure of. |
05:01.56 | Penguin | I know that using insecure=invite bypasses the auth on incoming calls. |
05:02.13 | WIMPy | Sounds the same to me. |
05:02.19 | Penguin | That's what a lot of people use because they didn't learn about remotesecret. |
05:03.01 | Penguin | But my complaint to phix was that he was using insecure=invite,port for a PHONE. |
05:03.44 | WIMPy | That sounds like a bad idea, indeed. |
05:04.05 | Penguin | Had there been discussion about using the insecure setting for an ITSP, I would have explained all the remotesecret stuff earlier. |
05:06.13 | Penguin | Now you've got me interested to see where the differences are between using remotesecret or secret+insecure. |
05:06.38 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.36) |
05:08.36 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
05:19.10 | *** join/#asterisk mitchrodrigues (~mitchrodr@c-98-244-30-255.hsd1.ca.comcast.net) |
05:30.44 | *** join/#asterisk djgerm (~Adium@64.191.238.196) |
05:30.48 | djgerm | Hello! |
05:31.02 | djgerm | does anybody here use sangoma cards? |
05:32.16 | djgerm | i am having trouble compiling the wanpipe drivers |
05:32.55 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
05:42.22 | *** join/#asterisk redotis (~redotis@nc-76-5-176-123.dhcp.embarqhsd.net) |
05:57.13 | *** join/#asterisk apb (~quassel@174.135.116.22) |
06:10.52 | *** join/#asterisk SGjunior (~sgjunior@modemcable066.0-70-69.static.videotron.ca) |
06:18.10 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
06:19.52 | *** join/#asterisk mitchrodrigues (~mitchrodr@c-98-244-30-255.hsd1.ca.comcast.net) |
06:48.08 | *** join/#asterisk evilman_home (kvirc@89-179-77-66.broadband.corbina.ru) |
06:52.38 | *** join/#asterisk hos7ein (~chatzilla@91.98.33.208) |
07:12.09 | *** join/#asterisk jsjc (~Adium@225.Red-88-12-12.staticIP.rima-tde.net) |
07:14.39 | *** join/#asterisk rogersja (~rogersja@162-192-162-17.lightspeed.rlghnc.sbcglobal.net) |
07:18.35 | *** join/#asterisk djgerm (~Adium@64.191.238.196) |
07:20.19 | *** part/#asterisk djgerm (~Adium@64.191.238.196) |
07:25.35 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
07:41.04 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
07:49.52 | *** join/#asterisk mitchrodrigues (~mitchrodr@c-98-244-30-255.hsd1.ca.comcast.net) |
07:56.44 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
08:05.42 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
08:06.43 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.32) |
08:11.02 | *** join/#asterisk kasanop (~kasanop@128-68-127-68.broadband.corbina.ru) |
08:13.38 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
08:25.51 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
08:30.02 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.127.27) |
08:32.27 | *** join/#asterisk sjs205 (~sjs205@host81-151-252-147.range81-151.btcentralplus.com) |
08:33.23 | *** join/#asterisk vlad_st__ (~vlad_star@109.188.124.67) |
08:35.48 | sjs205 | Morning all, I need to create soe dialplan that basically receives a call, places the caller on hold whi another number is called, caller 2. Some options need to be played to called 2, and when they select a given option the two calls are merged... is this possible? |
08:35.58 | sjs205 | *while |
08:38.24 | *** join/#asterisk v0lZy (~Thunderbi@84-255-194-41.static.t-2.net) |
08:39.45 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
08:46.12 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
08:50.38 | *** join/#asterisk mitchrodrigues (~mitchrodr@c-98-244-30-255.hsd1.ca.comcast.net) |
08:59.20 | *** join/#asterisk sjs205 (~sjs205@host81-151-252-147.range81-151.btcentralplus.com) |
08:59.47 | sjs205 | Did anyone respond to my previous message? sorry, I had a power cut... |
09:00.40 | [TK]D-Fender | sjs205: "core show application dial" <- Macro or Gosub, either will do |
09:01.10 | sjs205 | [TK]D-Fender, cheers, I'#ll look at those now... |
09:06.06 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
09:06.37 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.67) |
09:09.39 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
09:18.13 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
09:46.38 | *** join/#asterisk andy09usa (~andy09usa@unaffiliated/andy09usa) |
09:49.17 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
10:07.18 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.74) |
10:09.40 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
10:13.29 | *** join/#asterisk smirker (~x@CPE-144-137-156-146.lnse3.cha.bigpond.net.au) |
10:20.37 | *** join/#asterisk mitchrodrigues (~mitchrodr@c-98-244-30-255.hsd1.ca.comcast.net) |
10:28.35 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.126.47) |
10:47.59 | sjs205 | Hello again, I've got a problem with macro I've written, it is suposed to dial a call and then play festival to the caller, but it only dials the call, no festival ;) Asterisk doesn't seem to show anything... http://pastebin.com/6KyQ4AVN |
10:48.11 | sjs205 | I'm sure I've missed something really simple :s |
10:48.14 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
10:55.47 | Gugge | sjs205: dial does not end before the call is over. |
10:56.31 | sjs205 | Gugge, ehh? |
10:56.46 | sjs205 | So it is stuck on the first line of the macro? |
10:57.07 | Gugge | its stuck in the Dial untill the callee hangs up |
10:57.45 | sjs205 | So di I need to change line 1 to exten => caller,1,Dial(...,Macro(test,SIP/sjs205)) |
10:57.50 | Gugge | and when he does, the call ends :) |
10:58.06 | sjs205 | would above help Gugge ? |
10:58.13 | Gugge | dont know |
10:58.48 | sjs205 | okay, I'll keep trying... cheers Gugge :) |
11:06.35 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.51) |
11:18.41 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.51) |
11:21.23 | *** join/#asterisk mitchrodrigues (~mitchrodr@c-98-244-30-255.hsd1.ca.comcast.net) |
11:28.46 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.51) |
12:00.15 | *** join/#asterisk Neoti (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
12:18.30 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
12:38.45 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.53) |
12:42.24 | *** join/#asterisk ThoMe (~tm@tm.muc.de) |
12:42.28 | ThoMe | hola! |
12:42.32 | ThoMe | or helo. |
12:43.19 | ThoMe | I use asterisk 11.6 and I connect to a trunk-Provider. My sip show peers said: |
12:43.22 | ThoMe | deutscheTelefon 92.60.208.117 a 5060 OK (28 ms) |
12:43.45 | ThoMe | but i can't recieve calls. outbound no problem but inbound. |
12:44.19 | ThoMe | details: I have no registration datas, only with static ip. my asterisk box is not direct with a static ip, its nat. |
12:44.34 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-117-242.dynamic.qsc.de) |
12:44.58 | ThoMe | when i calling a number and I look with ngrep -W byline port 5060 if I recieve packages then I see nothing. |
12:45.58 | ThoMe | my iptables ulr is: iptables -t nat -A PREROUTING -p udp -m udp -d 82.135.63.218 --dport 5060:5063 -j DNAT --to-destination 192.168.100.4 |
12:46.22 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.53) |
12:51.23 | *** join/#asterisk mitchrodrigues (~mitchrodr@c-98-244-30-255.hsd1.ca.comcast.net) |
13:06.38 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.55) |
13:12.39 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.55) |
13:22.29 | *** join/#asterisk bunglex (5198c9d5@gateway/web/freenode/ip.81.152.201.213) |
13:23.14 | bunglex | hi, wondering if anyone can help me register an SPA3102 as an asterisk trunk - im having problems registering |
13:23.16 | *** join/#asterisk dxd828 (~dxd828@host86-172-58-16.range86-172.btcentralplus.com) |
13:48.20 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
13:52.07 | *** join/#asterisk mitchrodrigues (~mitchrodr@c-98-244-30-255.hsd1.ca.comcast.net) |
13:54.06 | *** join/#asterisk rogersja (~rogersja@162-192-162-17.lightspeed.rlghnc.sbcglobal.net) |
13:55.51 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.55) |
14:01.58 | *** join/#asterisk jploh (~textual@49.144.120.20) |
14:12.22 | *** join/#asterisk esaym153 (~esaym153@216-45-91-132.gvec.net) |
14:12.43 | esaym153 | how do I tell if my remote provider is using the jitterbuffer? |
14:33.27 | *** join/#asterisk evilman_home (kvirc@89-179-77-66.broadband.corbina.ru) |
14:34.16 | *** join/#asterisk SGjunior (~sgjunior@modemcable066.0-70-69.static.videotron.ca) |
14:40.03 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
15:16.13 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
15:18.21 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
15:22.06 | *** join/#asterisk mitchrodrigues (~mitchrodr@c-98-244-30-255.hsd1.ca.comcast.net) |
15:29.27 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
15:44.19 | *** join/#asterisk imox (~imox@91-64-148-46-dynip.superkabel.de) |
15:47.40 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
15:58.37 | *** join/#asterisk Akuma (~Akuma@modemcable085.96-58-74.mc.videotron.ca) |
16:13.59 | *** join/#asterisk bluefirecorp (~gredinger@cpe-98-28-80-91.columbus.res.rr.com) |
16:36.17 | *** join/#asterisk serafie (~erin@24.96.64.240) |
16:39.54 | *** join/#asterisk rogersja (~rogersja@162-192-162-17.lightspeed.rlghnc.sbcglobal.net) |
16:48.25 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
16:48.33 | *** join/#asterisk oquidave (~oquidave@41.202.233.180) |
16:49.00 | oquidave | hello |
16:51.05 | oquidave | if i've a gms-voip gateway connected to asterisk, when a call comes in from cellular user A, when asterisk routes it using the gsm gateway to mobile user B, do i incur any airtime costs? thanks |
16:52.00 | pabelanger | oquidave, ask your providers |
16:52.28 | pabelanger | nothing in life is free |
16:54.26 | oquidave | pabelanger: from my understanding the mobile caller A who is originating the call pays...my confusion is, as the gsm gateway routes the call to another regular mobile user and not a sip user, do i hosting asterisk pbx incur any costs? |
17:06.06 | pabelanger | Yes |
17:08.35 | oquidave | pabelanger: that costs is incured in the simcard dialing out to the receiving mobile user? |
17:09.00 | *** join/#asterisk lvlinux (~lvlinux@c-50-142-148-35.hsd1.tn.comcast.net) |
17:09.07 | pabelanger | oquidave, ask your providers |
17:09.41 | pabelanger | Unless you own the GSM network, you are likely going to incur costs |
17:09.46 | pabelanger | why do you think it would be free? |
17:09.54 | *** join/#asterisk mitchrodrigues (~mitchrodr@c-98-244-30-255.hsd1.ca.comcast.net) |
17:15.05 | *** join/#asterisk tzanger (andrew@wallace.mixdown.ca) |
17:15.09 | Penguin | In debian, how do I see which package owns a given file? |
17:15.15 | *** part/#asterisk tzanger (andrew@wallace.mixdown.ca) |
17:16.46 | *** join/#asterisk serafie (~erin@24.96.64.240) |
17:18.04 | pabelanger | Penguin, apt-cache search "filename" |
17:18.11 | pabelanger | eg: apt-cache search asterisk.conf |
17:18.58 | Penguin | I thought it would be a dpkg command. I was digging through the dpkg man page. |
17:19.49 | Penguin | dpkg-query -l file* also seems to help. |
17:19.50 | Penguin | Thanks. |
17:20.04 | pabelanger | well, there is likely more then one way to skin it |
17:20.23 | Penguin | Yep. apt-cache told me quickly. |
17:24.45 | Penguin | If apt-cache search file name does not return a package, does that mean the file is not owned by a package? |
17:40.25 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
17:43.03 | *** join/#asterisk lvlinux (~lvlinux@c-50-142-148-35.hsd1.tn.comcast.net) |
18:01.18 | ChannelZ | Probably |
18:02.43 | Penguin | That's totally foreign to me. I'm used to every file being owned by a package (except for the ones that users create, of course). So if it's found in /usr/bin/, it should be owned by a package. |
18:09.57 | ChannelZ | Actually I think you want dpkg-query -S for that |
18:12.46 | ChannelZ | 'apt-cache search' only searches package names/descriptions, 'dpkg-query -l' only names. |
18:16.08 | Penguin | I see. |
18:16.35 | Penguin | Let's use vimdiff as an example. |
18:16.44 | Penguin | dpkg-query -S /usr/bin/vimdiff |
18:16.46 | Penguin | dpkg: /usr/bin/vimdiff not found. |
18:17.09 | Penguin | On the other hand, let's use vim.basic as an example. |
18:17.15 | Penguin | dpkg-query -S /usr/bin/vim.basic |
18:17.18 | Penguin | vim: /usr/bin/vim.basic |
18:18.27 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
18:18.44 | Penguin | Now even though vimdiff is a symlink to another file, some package surely had the symlink in it. Wouldn't you think? |
18:21.00 | Penguin | is NOT a Debian person. |
18:26.49 | *** join/#asterisk kasanop (~kasanop@128-68-127-68.broadband.corbina.ru) |
18:27.07 | *** join/#asterisk ^rage^ (~mts@213.21.4.41) |
18:39.30 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
18:42.57 | *** join/#asterisk serafie (~erin@24.96.64.240) |
18:44.52 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-117-242.dynamic.qsc.de) |
18:49.06 | rogersja | what are the maximum character lengths for SIP usernames and secrets? |
18:49.40 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
18:49.58 | Penguin | 1, I'd guess. |
18:50.04 | Penguin | oh |
18:50.05 | Penguin | max |
18:50.15 | Penguin | I thought you said minimum for some reason. |
18:50.47 | Penguin | I've seen some really long ones. |
18:50.56 | rogersja | ha ha 1 would be somewhat limiting, but i suppose its possible |
18:51.14 | Penguin | I just misread. I don't know how. |
18:58.06 | *** join/#asterisk Ice_Strike (~Ice_Black@84.92.51.164) |
18:58.52 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
18:59.40 | bluefirecorp | LOM is the best there ever. |
19:21.00 | *** join/#asterisk SushiB (~Thunderbi@201.170.142.14.dsl.dyn.telnor.net) |
19:48.28 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
19:55.43 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.42) |
20:04.41 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
20:08.54 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.42) |
20:19.31 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.125.39) |
20:23.20 | SushiB | hi all, any one can help me configure an E1 with R2 modify on asterisk, using a sagoma card and version 1.8.20 with Elastix. I configured dahdi-channels.conf and changed the signalling to mfcr2 and the channels, and I add the parameters to the chan_dahdi.conf.wanpipe. |
20:45.41 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.39) |
20:55.38 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
21:07.32 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
21:17.09 | *** join/#asterisk SushiB (~Thunderbi@200.77.217.106) |
21:34.40 | *** join/#asterisk tomii (~tomii@128.54.168.125.sta.wbroadband.net.au) |
22:06.57 | *** join/#asterisk g_r_eek (~g_r_eek@176.92.50.69) |
22:07.29 | *** join/#asterisk elfelvin (~elfelvin@46-65-37-150.zone16.bethere.co.uk) |
22:08.49 | elfelvin | does anyone know where i can get hold of a (cheaper) FXO usb dongle? much like the Sangoma http://www.sangoma.com/products/usbfxo/ |
22:11.49 | elfelvin | maybe the ObiLine (http://www.obihai.com/obiline)? does anyone know if this works with asterisk? |
22:16.48 | WIMPy | AFAIK the only (halfway) directely supported USB interfaces is the Xorcom Astribank series. |
22:19.06 | elfelvin | thanks WIMPy, i would have thought that China would have flooded the market by now but obviously not. im just looking for a cheap FXO -> USB dongle |
22:19.31 | WIMPy | They might have, but that doesn't mean you can use them with Asterisk. |
22:20.02 | WIMPy | Unfortunatly Asterisk doesn't use standard drivers. |
22:21.37 | Penguin | If there is a flood of those devices, what would they be intended to be used with? |
22:26.12 | WIMPy | There are many telephony applications other than Asterisk. |
22:26.21 | Penguin | Name 53. |
22:26.39 | WIMPy | Ask google. |
22:26.53 | Penguin | I really just want a couple examples of what the devices might be used for. |
22:27.38 | WIMPy | NFI. You know that the POTS time has been over for me for two decades. |
22:28.02 | WIMPy | But there's lots of commercial stuff and the linux telephony architecture must be used by somethig as well. |
22:28.04 | Penguin | I also know that you're holding on to every last thread of it. |
22:28.34 | WIMPy | That must have been about 20 years ago. |
22:28.38 | Penguin | oh |
22:28.43 | WIMPy | Almost. |
22:29.20 | *** join/#asterisk rogersja (~rogersja@162-192-162-17.lightspeed.rlghnc.sbcglobal.net) |
22:29.50 | WIMPy | But I will stay with ISDN as long as possible. 1. because the VOIP stuff (at least SIP) is long from being fit for purpose and 2. is extremely expensive. |
22:31.19 | elfelvin | ja there are also alot of 3rd world countries who would also like to benefit from VoIP but dont have ISDN etc, so there is still a large market for POTS |
22:31.35 | elfelvin | but in my case I'm just needing it for a small home project |
22:32.53 | WIMPy | Sure, but I have been used to working technology and have some dificulty with downgrading to some random functionality. |
22:34.02 | WIMPy | I've had some very exciting, but also very unproductive years with Asterisk. |
22:53.03 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.32) |
22:57.49 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.124.60) |
23:03.46 | *** join/#asterisk rogersja (~rogersja@162-192-162-17.lightspeed.rlghnc.sbcglobal.net) |
23:17.01 | rogersja | any voip client recommendations for iOS that support video (H.263, H.263+, H.264)? Zoiper is very unstable right now |
23:17.46 | Kobaz | bria |
23:21.07 | *** join/#asterisk CrashHD (~na@204-195-127-247.wavecable.com) |
23:31.36 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
23:41.48 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.76) |
23:52.19 | SushiB | hi, I'm getting this message when I try to load the chan_dahdi.so module. Signalling request on channel 13 is MCF/R2 but lines is in ISDN PRI signalling... can anyone help me with this? |
23:53.17 | WIMPy | Is that with wanpipe? |
23:54.25 | SushiB | yes |
23:54.51 | WIMPy | Then I guess your wanpipe config and your dahdi config don't match. |
23:56.48 | SushiB | what line should I look from the wanpipe config file? I'm lookin to TE_SIG_MODE is sey to CAS |
23:57.33 | SushiB | and I see the line CommPort is set to PRI |
23:57.48 | WIMPy | Sorry, I don't know wanpipe. |
23:58.17 | WIMPy | It probably shouldn't be set to PRI. I assume PRI implies ISDN. |
23:58.25 | SushiB | ok.. thanks |