| 00:00.28 | drfreeze | Have a client that is missing voicemails. They think that their hangup cases when checking voicemail are related to the missing voicemail |
| 00:04.48 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-15-179.bois.qwest.net) |
| 00:16.41 | *** join/#asterisk albertoandrade (~albertoan@186.206.5.67) |
| 00:29.58 | paolosupino | p3nguin: will paste bin in a short while…. |
| 00:33.24 | kessius | hi, how to make asterisk listen on port 75793 |
| 00:34.38 | *** join/#asterisk michael-i (~anonymous@c-24-7-126-118.hsd1.ca.comcast.net) |
| 00:37.26 | p3nguin | kessius: Configure a channel to listen there. Which channel do you want to listen on that port? |
| 00:40.22 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-qcsfwjwlflayvvui) |
| 00:40.54 | kessius | <PROTECTED> |
| 00:43.00 | kessius | p3nguin, on asterisk sip.conf set bindport 5060, 75793 |
| 00:43.19 | *** join/#asterisk JunK-Y (~junky@pdpc/supporter/active/junk-y) |
| 00:44.20 | [TK]D-Fender | fprior: Ok, where are the debugs and configs? |
| 00:45.21 | paolosupino | p3nguin: the paste in http://pastebin.com/JJcLf9Zg resulted in a private number being displayed. |
| 00:46.46 | michael-i | This may not be the most appropriate forum but does anyone have a go-to recommendation for a PRI-SIP gateway? I don't want to use a card in the Asterisk server itself, would rather have a gateway in between. |
| 00:49.19 | [TK]D-Fender | michael-i: All very expensive by comparison. AudioCodes Mediant series, or perhaps Quintum.... Mediatrix ones were a little iffy in the firmware dept. |
| 00:49.48 | [TK]D-Fender | michael-i: And that question is just fine for here |
| 00:50.07 | michael-i | [TK]D-Fender: expensive is ok, it just has to work well |
| 00:51.19 | JunK-Y | mediatrix! |
| 00:51.51 | michael-i | from someone named junky ;) ;) I'm scared |
| 00:52.41 | kessius | someone [TK]D-Fender how to make the asterisk listem port 75793 - i do bindport = 5060,75973,5050 - the * accepted these parameters - how to test if * listem 75793 |
| 00:52.54 | michael-i | isn't that out of the port range?! |
| 00:53.05 | kessius | Please I need help |
| 00:53.42 | lanning | on linux as root "netstat -anpu" |
| 00:56.13 | p3nguin | kessius: You can't do that. chan_sip can only listen on one port. |
| 00:56.32 | p3nguin | And yes, 75793 is not a valid port. |
| 00:56.47 | p3nguin | Valid ports are 0-65535. |
| 00:59.47 | kessius | is true tha range are 0-65535. - even with 127.0.0.1:75793 |
| 01:00.02 | p3nguin | 127.0.0.1:75793 <----- invalid |
| 01:00.20 | p3nguin | And chan_sip will only listen on one port, anyway. |
| 01:01.36 | *** join/#asterisk DaPrivateer (~matt7229@71-9-155-174.static.oxfr.ma.charter.com) |
| 01:03.06 | kessius | p3nguin - I want to implement this - http://www.personal.psu.edu/wcs131/blogs/psuvoip/2011/12/ |
| 01:06.32 | michael-i | kessius: you're confusing a port with a sip peer/uri |
| 01:06.48 | michael-i | you only need to bind to 5050 as far as I can see |
| 01:11.42 | *** part/#asterisk vinhdizzo (~vinh@dhcp-v022-171.mobile.uci.edu) |
| 01:12.48 | *** join/#asterisk SaRSAeOL (~sarsaeol@66-113-78-49.rev.ibsinc.com) |
| 01:13.12 | SaRSAeOL | has anyone played with the new Digium IP phones? |
| 01:13.33 | SaRSAeOL | i have a few D70s i am configuring that are getting abysmal voice quality |
| 01:13.59 | [TK]D-Fender | kessius: You cannot bind to multiple ports. |
| 01:14.05 | kessius | how to make for configure * link or bind a channel with a port 127.0.0.1:0-65535 - |
| 01:14.23 | [TK]D-Fender | kessius: You can't. End of story. |
| 01:16.55 | *** join/#asterisk voipnation_ (18b2d41a@gateway/web/freenode/ip.24.178.212.26) |
| 01:17.30 | voipnation_ | Hi everyone. First time user here! |
| 01:19.46 | *** join/#asterisk SaRSAeOL (~sarsaeol@66-113-78-49.rev.ibsinc.com) |
| 01:20.49 | p3nguin | Long time listener, first time caller? |
| 01:22.12 | p3nguin | kessius: No where on that page does anything say anything about port 75793. |
| 01:22.56 | SaRSAeOL | would this be the wrong room for digium hardware inquiries? |
| 01:22.57 | p3nguin | It does mention EXTENSION 75973, though. |
| 01:23.24 | p3nguin | It isn't a room, it's an IRC channel. |
| 01:23.37 | SaRSAeOL | pardon, is this the wrong channel |
| 01:23.41 | p3nguin | And it isn't necessarily the wrong channel. It's just the wrong time of day. |
| 01:24.03 | p3nguin | There are like five people paying attention right now. |
| 01:24.09 | p3nguin | You and I are two of those. |
| 01:24.39 | SaRSAeOL | lol okay i can live with that |
| 01:24.42 | SaRSAeOL | thanks p3nguin |
| 01:25.06 | p3nguin | You can also contact Digium support, but they won't answer at this time of day, either. |
| 01:25.22 | SaRSAeOL | yeah that was plan A |
| 01:25.34 | SaRSAeOL | ;) |
| 01:25.59 | voipnation_ | what hardware are you speaking of SaRSeOL? |
| 01:26.05 | SaRSAeOL | D70 |
| 01:26.11 | SaRSAeOL | on of their new IUP phones |
| 01:26.12 | SaRSAeOL | IP* |
| 01:26.52 | voipnation_ | ahhh... can't help as I haven't had a chance to order one yet |
| 01:27.22 | SaRSAeOL | they look nice… have had horrendous audio though… hoping its some config option i have overlooked |
| 01:27.51 | voipnation_ | Horrendous how? |
| 01:28.06 | SaRSAeOL | digital distortion |
| 01:28.45 | voipnation_ | what version of asterisk are you running? |
| 01:28.50 | [TK]D-Fender | SaRSAeOL: What codec? Call within the local lan? What is the other end using? What are netowrking conditions like? What alternative settings have you tried? |
| 01:29.22 | [TK]D-Fender | SaRSAeOL: Who shot J.R.? What is the average airspeed velocity of an unladen swallow? What would you do for a Klondike bar? |
| 01:29.34 | SaRSAeOL | if A is digium phone and B is polycom, calling A to B produces robotic voice of A speaker |
| 01:29.39 | SaRSAeOL | asterisk 1.8 |
| 01:29.47 | SaRSAeOL | ulaw |
| 01:29.55 | SaRSAeOL | within LAN |
| 01:30.08 | voipnation_ | What about B to A? |
| 01:30.16 | SaRSAeOL | B to A is crystal clear |
| 01:30.28 | voipnation_ | have you got a pcap of the call yet? |
| 01:30.51 | SaRSAeOL | [TK]D-Fender: African or Asian swallow |
| 01:30.52 | SaRSAeOL | ? |
| 01:31.00 | SaRSAeOL | i do |
| 01:31.05 | SaRSAeOL | signaling looks normal |
| 01:31.20 | SaRSAeOL | i think it may have to do with how the digium phone is setting up the RTP stream |
| 01:31.27 | [TK]D-Fender | SaRSAeOL: Thorough.... good. What codec? |
| 01:31.27 | voipnation_ | if you decode the RTP stream can you hear the digitized problem? |
| 01:31.48 | SaRSAeOL | [TK]D-Fender: I've tried g711u and a |
| 01:31.57 | kessius | p3nguin friends - I have a fxo hardware-pci and only configure the extensions.conf [inbound_fxo] -how to * receive call if not 127.0.0.1:? - it is mult ports |
| 01:31.58 | SaRSAeOL | voipnation_: yes |
| 01:32.28 | [TK]D-Fender | SaRSAeOL: check the packetization rate between them... that is about the only thing I might wonder about at this point. In an enclosed LAN you really shouldn't have these kinds of issues |
| 01:32.59 | [TK]D-Fender | SaRSAeOL: If RTP flows through * and you have timing issues for something like a VM (only thing I can think of) then perhaps as well... but even then low odds |
| 01:33.10 | SaRSAeOL | [TK]D-Fender: sort of my frustration exactly, the packetization is at 20ms |
| 01:33.29 | voipnation_ | have you tried .030 |
| 01:33.52 | SaRSAeOL | so 30 ms |
| 01:33.52 | SaRSAeOL | ? |
| 01:33.55 | voipnation_ | ya |
| 01:34.00 | SaRSAeOL | ill give it a shot |
| 01:34.03 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 01:34.14 | [TK]D-Fender | SaRSAeOL: Ok, all sounding standard... I'm suspecting you'll need to call Digium support very directly for this. Because there is virtually no way I'd suspect the Polycom side of this without something solid pointing to it.. |
| 01:35.06 | SaRSAeOL | [TK]D-Fender: yeah and I've tried polycom to polycom with the exact same config with zero issues |
| 01:35.26 | SaRSAeOL | [TK]D-Fender: digium support is first on my to do for the morning |
| 01:35.41 | voipnation_ | Have you tried calling a Polycom outside the LAN? |
| 01:36.16 | SaRSAeOL | not a polycom, but I've tried to a pstn line with the same result… for what its worth |
| 01:36.56 | voipnation_ | I am unsure how to private message as this is like day 0 of using IRC. But you could try calling me. I have a 670 sitting here. |
| 01:37.45 | SaRSAeOL | sent you a PM |
| 01:38.13 | [TK]D-Fender | SaRSAeOL: Multiple different end points with same result on Digium side = Digium Phone issue. Simple scientific process |
| 01:38.33 | p3nguin | voipnation_: /query nickname |
| 01:38.47 | [TK]D-Fender | SaRSAeOL: And those phones are so very new that there isn't really an experience base to go around. |
| 01:39.09 | voipnation_ | Correct but it sure is fun troubleshooting generic RTP issues |
| 01:39.26 | SaRSAeOL | [TK]D-Fender: i'm leaning heavily in that direction as well |
| 01:39.43 | SaRSAeOL | there isn't even a comprehensive user guide out there |
| 01:39.47 | SaRSAeOL | is frusterating |
| 01:40.07 | voipnation_ | what model of polycom are you using |
| 01:40.10 | [TK]D-Fender | SaRSAeOL: I admit I was looking forward to getting my hands on one for evaluation.... |
| 01:42.10 | kessius | p3nguin michael-i thank you, really i'm confusing a port with a sip peer/uri |
| 01:43.10 | voipnation_ | Yea that sounded very trashy lol |
| 01:43.47 | SaRSAeOL | lol yeah so my initial review for the digium phones is not stellar, hopefully support will be able to shed some light on the situation |
| 01:44.25 | voipnation_ | It almost sounds like a busted mic. Definitely not a typical sounding RTP issue. Never heard that one before. |
| 01:46.38 | SaRSAeOL | even worse that it is happening over all the digium phones i have tried (3) |
| 01:46.51 | SaRSAeOL | which makes me think config over hardware |
| 01:47.43 | [TK]D-Fender | Or firmware... |
| 01:48.15 | voipnation_ | have you tried removing all codecs but g729 |
| 01:48.48 | p3nguin | kessius: No, you are confusing port number with EXTENSION. |
| 01:49.02 | p3nguin | That page talks about EXTENSION 75973. |
| 01:49.22 | voipnation_ | It really seems like two different compressions of the same codec |
| 01:49.34 | p3nguin | Compressions? |
| 01:49.40 | *** join/#asterisk ios_sos (~nbeard@24-181-146-94.static.dlth.mn.charter.com) |
| 01:49.48 | p3nguin | ulaw isn't compressed. |
| 01:50.20 | kessius | p3nguin - michael-i for Incoming calls will ring sip:75973@127.0.0.1 - how to configure sip.conf or only extensions |
| 01:51.00 | voipnation_ | p3nguin, i understand that |
| 01:54.48 | voipnation_ | [TK]D-Fender do you think it could be comfort noise causing the bad audio? |
| 01:55.33 | SaRSAeOL | voipnation_: comfort noise? |
| 01:55.51 | [TK]D-Fender | voipnation_: No.. because VOICE is distorted, not just silence between. |
| 01:56.52 | [TK]D-Fender | kessius: Make your peer to match your incoming call and point it to a context. then make your extension to match what they dialed |
| 01:57.26 | kessius | p3nguin - michael-i - register = username:password@127.0.0.1:75973 |
| 01:57.26 | kessius | <PROTECTED> |
| 01:59.07 | voipnation_ | Anyone here had a chance to test the Grandstream 3175? Having very poor results with it, just wondering others thoughts. |
| 01:59.51 | voipnation_ | Online reviews seem decent. I wouldn't give this phone half a star /5 so there is no way everyone is having the same results im having. |
| 02:05.06 | drfreeze | Anyone know how to get a hangup event when checking voicemail? |
| 02:05.08 | drfreeze | Have a client that is missing voicemails. They think that their hangup cases when checking voicemail are related to the missing voicemail |
| 02:06.13 | SaRSAeOL | [TK]D-Fender: and voipnation_ the problem is gone by disabling support for g726-32… even though the call was not using it and it was at the lowest priority… it must be a bug in firmware |
| 02:06.33 | SaRSAeOL | even though I'm still wondering how its not my fault |
| 02:06.38 | SaRSAeOL | lol |
| 02:06.50 | SaRSAeOL | hugs [TK]D-Fender and voipnation_ |
| 02:07.09 | [TK]D-Fender | SaRSAeOL: "If a tree falls in the forest, and there is noone around to hear it .... is the man still wrong?" |
| 02:07.19 | [TK]D-Fender | - women |
| 02:07.28 | *** join/#asterisk scubes13 (~scubes13@cpe-024-168-196-000.sc.res.rr.com) |
| 02:07.54 | voipnation_ | What codec could we use to hear said tree? |
| 02:08.44 | voipnation_ | Sarsaeol... Good to hear... sorry we couldn't help |
| 02:08.56 | voipnation_ | although i will take credit for it anyway |
| 02:09.48 | p3nguin | kessius: sip:75973@127.0.0.1 is EXTENSION 75973. Extensions are configured in extensions.conf. |
| 02:13.56 | kessius | p3nguin: then I can configure only extensions.conf [75973] |
| 02:16.05 | kessius | p3nguin : or [default] exten=>75973,1... / and on sip.con peer and context |
| 02:16.37 | p3nguin | [75973] would be a context named 75973. |
| 02:16.52 | p3nguin | Choose a better context, then create extension 75973. |
| 02:17.12 | p3nguin | I'm not going to teach you asterisk configuration today. |
| 02:17.27 | p3nguin | You can read the book just like everyone else who needs to know how to configure it. |
| 02:27.51 | *** join/#asterisk voipnation (~voipnatio@24-178-212-26.static.ftwo.tx.charter.com) |
| 02:29.07 | voipnation | so Sars, I got disconnected. Is all well now? |
| 02:29.18 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 02:34.56 | SaRSAeOL | voipnation: indeed it is, conferring the remainder of the phones so i can get the h3ll outta here |
| 02:35.07 | SaRSAeOL | damn auto correct… configuring* |
| 02:36.31 | SaRSAeOL | then tomorrow i get the fun part of loading the Digium Phones Module for Asterisk |
| 02:36.36 | SaRSAeOL | all kinds of cool features with that |
| 02:38.39 | voipnation | I hear that. This stuff will consume your life if you let it LOL.. I let it unfortunately |
| 02:39.20 | voipnation | When I wake, to when I sleep. Always messing around with my PBX. Learning, tweaking, destroying all kinds of stuff :) |
| 02:40.51 | kessius | <PROTECTED> |
| 02:41.41 | voipnation | Anyone have any experience with compiling perl |
| 02:43.49 | p3nguin | kessius: That is very broken. |
| 02:43.56 | voipnation | im trying to install something called Time.Duration.pm and have no clue where to put it |
| 02:44.07 | voipnation | time/duration.pm |
| 02:44.20 | p3nguin | What is it? |
| 02:44.53 | p3nguin | It seems like something that belongs in /usr/share/perl5/vendor_perl/ -ish. |
| 02:45.07 | voipnation | http://search.cpan.org/~sburke/Time-Duration-1.02/Duration.pm |
| 02:45.26 | p3nguin | /usr/share/perl5/vendor_perl/Time/ on my box. |
| 02:45.48 | p3nguin | I don't have Duration.pm, though, only Zone.pm. |
| 02:46.13 | [TK]D-Fender | kessiusp3nguin : because not words - http://pastebin.com/5aQqiMYy <- I presume you mean "doesn't work" |
| 02:46.16 | voipnation | K thx. Ill try and figure it out. Someone I know wrote a custom program to view tenant CDR's and without it i get errors |
| 02:46.39 | p3nguin | There. -r--r--r-- 1 root root 14412 Aug 18 2007 /usr/share/perl5/vendor_perl/Time/Duration.pm |
| 02:46.39 | [TK]D-Fender | kessius: exten=>7597,1,Dial(SIP/1101,60,wWrtT) <- and as far as you've described... this SHOULDN'T work |
| 02:47.08 | [TK]D-Fender | kessius: 7597 != 75973 |
| 02:47.09 | p3nguin | I just isntalled perl-time-duration |
| 02:47.52 | p3nguin | That's an old module. |
| 02:49.02 | p3nguin | I guess on CentOS, it is perl-Time-Duration. |
| 02:50.25 | p3nguin | Not sure which distro you are using. |
| 02:51.36 | voipnation | yes running centos |
| 02:51.47 | *** join/#asterisk mintos (~mvaliyav@114.143.165.128) |
| 02:51.53 | voipnation | I suppose i just navigate to that directory and explode the tarball there? |
| 02:52.02 | voipnation | I was thinking it would be a little more involved |
| 02:54.03 | voipnation | I don' evne have a perl directory under /usr/share |
| 02:54.04 | p3nguin | no |
| 02:54.11 | p3nguin | yum -y install perl-Time-Duration |
| 02:54.22 | p3nguin | We have package managers for a reason. |
| 02:55.08 | p3nguin | If you haven't installed rpmforge, you probably need to do that first. |
| 02:55.27 | p3nguin | Lots of useful stuff in rpmforge that you'll want over time. |
| 02:56.15 | p3nguin | Not sure how to do that? See http://wiki.centos.org/AdditionalResources/Repositories/RPMForge |
| 02:58.07 | p3nguin | ~rpmforge |
| 02:58.07 | infobot | RPMforge is a collaboration of Dag and other packagers. They provide over 5000 packages for CentOS, including wine, vlc, mplayer, xmms-mp3, and other popular media tools. It is not part of Red Hat or CentOS but is designed to work with those distributions. http://wiki.centos.org/AdditionalResources/Repositories/RPMForge |
| 02:59.31 | voipnation | THanks so much. Now that I jogged my memory. Not having the right repo was where I left off last time. |
| 03:02.31 | voipnation | maybe it will have something needed to benchmark my raid configuration |
| 03:18.59 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 03:28.36 | *** join/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 03:29.09 | *** part/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 03:36.04 | *** join/#asterisk ruben23 (~user@112.205.68.8) |
| 03:36.12 | ruben23 | hi guys |
| 03:37.05 | ruben23 | hi guys how do i optmized asterisk for voip calls..? any idea..? |
| 03:38.06 | kessius | [TK]D-Fender, thank you - this was ok -7597 != 75973 - but when call 75973 not appears in console * - |
| 03:38.52 | [TK]D-Fender | kessius: Where do we see your SIP DEBUG for your failed call? |
| 03:39.47 | [TK]D-Fender | ruben23: Google up "Asterisk QoS" |
| 03:40.22 | kessius | <PROTECTED> |
| 03:40.38 | [TK]D-Fender | kessius: place your call |
| 03:42.46 | kessius | <PROTECTED> |
| 03:43.09 | p3nguin | You should have read the book. |
| 03:43.43 | [TK]D-Fender | kessius: What part of "there is no port over 65535" do you not understand about the TCP/IP stack? |
| 03:44.09 | [TK]D-Fender | kessius: And you are jumping topics again |
| 03:44.41 | [TK]D-Fender | kessius: First you're asking about dialplan, then you're asking about REGISTRATION which is completely different. Pick something to fix and stick with it |
| 03:44.46 | kessius | <PROTECTED> |
| 03:44.51 | [TK]D-Fender | kessius: You are going around in cirecles and getting nothing finished |
| 03:45.01 | [TK]D-Fender | circles* |
| 03:46.23 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 03:48.06 | kessius | [TK]D-Fender sip set debug on to 75973 |
| 03:49.22 | p3nguin | You can't sip set debug on an extension. |
| 03:49.35 | [TK]D-Fender | kessius: NO. Do not attempt to restict your debugging. look at ALL traffic |
| 03:49.46 | [TK]D-Fender | p3nguin: It's also a peername on his config |
| 03:50.04 | [TK]D-Fender | p3nguin: But still bad to restrict right now |
| 03:50.49 | p3nguin | Well, that guide said extension 75973 on host 127.0.0.1. I didn't see any mention of any peers. |
| 03:52.29 | [TK]D-Fender | p3nguin: He mentioned it and was visible in one of his pastebins... |
| 03:52.39 | [TK]D-Fender | p3nguin: http://pastebin.com/5aQqiMYy |
| 03:52.50 | p3nguin | I quit paying attention to him after a while. |
| 03:52.58 | p3nguin | It's too irritating to say the same things over and over and over. |
| 03:53.02 | [TK]D-Fender | p3nguin: Yeah, I can relate... |
| 03:54.36 | ruben23 | guys i ahve asterisk server on my network and im concerned about, when i try to register phone extensions on it im getting latency 100ms above for those extensions but this is only on local network..any idea what could be the reasons for it..? when i tried ping test form server to PC i get below 1 ms..but on asterisk CLI i get 100ms |
| 03:55.07 | p3nguin | You can't measure latency of an extension. Extensions have no round-trip times to be measured. |
| 03:55.24 | *** part/#asterisk paolosupino (~paolo-sup@net-2-38-113-188.cust.dsl.vodafone.it) |
| 03:55.29 | p3nguin | Also, Asterisk CLI does not have any ping tools. |
| 03:55.41 | p3nguin | Asterisk is not capable of ICMP at the network layer. |
| 03:56.21 | p3nguin | If you are using the qualify time for a PEER, you cannot compare it to ping time of a HOST. |
| 03:57.04 | p3nguin | Asterisk qualify time of a PEER measures the response to a SIP OPTIONS packet in the application layer. |
| 03:57.30 | p3nguin | Why must everyone think these two things are the same? |
| 03:57.47 | Kobaz | because it makes sense to assume that if you don't know how sip works |
| 03:57.57 | p3nguin | oh :/ |
| 03:57.58 | Kobaz | people think response time = icmp ping |
| 03:58.17 | Kobaz | some people don't know any other way |
| 03:58.36 | p3nguin | s/ other way/thing/ |
| 03:59.01 | Kobaz | heh |
| 03:59.06 | p3nguin | I soooooo wish infobot would do corrections on other people's text. |
| 03:59.30 | p3nguin | That would make things so much more amusing. |
| 03:59.44 | Kobaz | it does on #asterisk-dev |
| 03:59.53 | p3nguin | OH?! |
| 03:59.53 | Kobaz | oh |
| 03:59.56 | Kobaz | other people |
| 03:59.58 | *** join/#asterisk gajini (~root@61.12.17.171) |
| 04:00.00 | Kobaz | it doesnt do other peolpe |
| 04:00.00 | Kobaz | heh |
| 04:00.00 | p3nguin | awww |
| 04:00.07 | p3nguin | You ruined my hopes. |
| 04:00.11 | Kobaz | heh |
| 04:00.19 | Kobaz | dashed all hopes! |
| 04:00.54 | [TK]D-Fender | Actually... its not so much a SIP thing as it is an Asterisk thing. It could have used ICMP as a validation tool, but that is often filtered, etc. Also as SIP is an application layer routed protocol it's fail at the proxy level. But that is not apparent to people new to larger setups. |
| 04:01.09 | [TK]D-Fender | it'll* |
| 04:01.35 | [TK]D-Fender | Kobaz: h-o-p-e-s <- THERE I FIXED IT |
| 04:01.36 | Kobaz | epic fail at the proxy level |
| 04:01.58 | p3nguin | I would have guessed that the reason SIP packets are used for measuring it was that the phones are using SIP, so that's a common denominator. |
| 04:02.30 | [TK]D-Fender | p3nguin: Routing is the real reason.... |
| 04:02.33 | Kobaz | one more feature |
| 04:02.36 | Kobaz | that's all i need |
| 04:02.37 | Kobaz | do de do |
| 04:06.00 | ruben23 | p3nguin:thanks for the info you give so the 100 ms for qualify is somehow ok..? |
| 04:06.25 | Kobaz | basically that tells you that the peer is reachable |
| 04:06.34 | Kobaz | brain dead servers might even be 2000ms |
| 04:06.42 | Kobaz | but your actual sip call quality would be fine |
| 04:06.45 | Kobaz | it depends on the server |
| 04:07.09 | Kobaz | it tells you how long the remote host took to respond to your sip options request |
| 04:08.23 | [TK]D-Fender | ruben23: Yes |
| 04:10.15 | ruben23 | thanks guys. |
| 04:14.01 | p3nguin | I do not worry over 100 ms qualify time. |
| 04:14.21 | Kobaz | spilled milk is not much to worry about either |
| 04:14.34 | p3nguin | Not when you have a wife to clean it up, anyway. |
| 04:14.53 | ruben23 | or helper.. |
| 04:14.58 | p3nguin | Oh crap, I thought she was walking over here to read what I just typed. |
| 04:15.05 | Kobaz | manservant |
| 04:15.11 | p3nguin | But she was headed somewhere else. |
| 04:15.15 | p3nguin | timing |
| 04:21.43 | p3nguin | This is interesting psychology. I deployed an office with multi-line phones. I configured two line keys on each phone: one line key showed a display name of the 10-digit phone number next to the button, the other line key showed the internal extension number used to reach the phone. I just checked the CDRs from a specimen phone, and most of the time when dialing an external number, she was using the line key showing the ... |
| 04:21.49 | p3nguin | ... 10-digit phone number. |
| 04:22.08 | p3nguin | Both line keys had the exact same capabilities. The only difference was the display name next to the button on the phone. |
| 04:22.36 | Kobaz | heh |
| 04:22.39 | Kobaz | it |
| 04:22.41 | Kobaz | 's all the same |
| 04:22.48 | Kobaz | people do what they are used to |
| 04:22.56 | Kobaz | "grab an outside line" |
| 04:23.13 | p3nguin | Now I move on to phase two of the test. I removed the second line key, so now there is only one line key active and it shows the internal extension number as the display name next to the button. |
| 04:23.38 | Kobaz | everyone is going to think they can't dial outside anymore |
| 04:23.41 | p3nguin | I'll see how they react. |
| 04:24.03 | p3nguin | Only the receptionist has more than one active button now. |
| 04:24.33 | Kobaz | "let's watch their reaction as we swap their favorite milk chocolate with iodine and food coloring" |
| 04:24.44 | p3nguin | External calls ring into the second line key which shows the 10-digit number. Calls from other phones in the office will ring into the first line which shows the internal number. |
| 04:25.15 | Kobaz | what if they are already on the phone on line 1 |
| 04:25.18 | Kobaz | and they get an internal call |
| 04:25.24 | p3nguin | voicemail |
| 04:25.53 | p3nguin | I had to shut off call waiting because people don't understand what a ringing phone means. |
| 04:26.10 | p3nguin | Now each phone only gets one call at a time. |
| 04:26.16 | Kobaz | hah |
| 04:26.23 | p3nguin | If you're on the phone, a call to your phone goes directly to busy vm. |
| 04:26.27 | p3nguin | People are so silly. |
| 04:26.44 | Kobaz | i set up a new office and they want direct dials for everyone |
| 04:26.55 | Kobaz | no main ivr, no attendant phone |
| 04:28.01 | p3nguin | The complaint I got was, "When I call, it just rings and rings and rings before finally saying the person is on the phone." Okay, so they want no ringing if the person is on the phone. Got it. No more call waiting. |
| 04:28.28 | Kobaz | well just make it ring like two times if someone is on the phone |
| 04:28.33 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 04:28.40 | p3nguin | If people would answer their beeps and use hold like a normal person, it wouldn't have been an issue. |
| 04:28.59 | p3nguin | I suppose I could do that. |
| 04:29.17 | p3nguin | Check the channels and adjust the ring time. |
| 04:29.33 | p3nguin | I actually hadn't considered that. |
| 04:30.08 | p3nguin | The complaint was about ringing just to ultimately end up in busy vm, so I eliminated the ringing aspect of it. |
| 04:30.50 | p3nguin | Next they will probably complain about ringing and unavailable vm when the person isn't on the phone but just doesn't answer. |
| 04:32.15 | p3nguin | At that point, I may tell them to take a walk to the other person's office before calling them just to make sure he or she is there to answer. *sigh* |
| 04:34.11 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:49.05 | drfreeze | p3nguin: funny stuff |
| 04:49.40 | drfreeze | people have no clue about how phone systems operate |
| 04:50.02 | drfreeze | or impractical most of their suggestions are |
| 04:50.23 | *** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:5:0:558:12) |
| 04:51.36 | *** part/#asterisk Marquel (~Marquel@unaffiliated/marquel) |
| 04:52.13 | *** part/#asterisk ruben23 (~user@112.205.68.8) |
| 04:58.37 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 05:02.14 | Kobaz | hah |
| 05:02.14 | Kobaz | yeah |
| 05:02.18 | Kobaz | i have a good one for that |
| 05:02.32 | Kobaz | so there's a three phone ring group for the operator at this place |
| 05:02.39 | Kobaz | 5pm two of the people leave |
| 05:03.10 | Kobaz | the front desk person forwards their phone and goes to the main office and sits at a cube at the forward destination |
| 05:03.28 | Kobaz | then she complains people are getting voicemail instead of ringing forever at the operator when she gets up for breaks |
| 05:03.43 | Kobaz | if you don't want people getting voicemail, either dont forward your phone, or turn off voicemail |
| 05:03.45 | Kobaz | "yeah but" |
| 05:04.15 | *** join/#asterisk din3sh (~din3sh@ADSL-TPLUS-111-164.telecomplus.net) |
| 05:07.02 | *** join/#asterisk din3sh (~din3sh@ADSL-TPLUS-111-164.telecomplus.net) |
| 05:09.21 | din3sh | morning all |
| 05:16.21 | MrOli | morning din3sh |
| 05:22.23 | din3sh | though its nitetime in US |
| 05:22.24 | din3sh | :p |
| 05:22.35 | MrOli | yup |
| 05:23.36 | ChannelZ | It's PANTS-OFF DANCE-OFF TIME! |
| 05:24.23 | din3sh | hehe |
| 05:24.28 | din3sh | its 10am here |
| 05:31.18 | *** part/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 05:41.56 | *** join/#asterisk tm1000 (~tm1000@provisioner.net) |
| 05:46.50 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 05:49.27 | din3sh | having a problem Got SIP response 486 "Busy Here" back from sip peeers with no DND on phones |
| 05:51.48 | MrOli | is the sip peer sending the 486 a hardware ip phone ? |
| 05:51.55 | din3sh | yes |
| 05:53.05 | MrOli | what's the model# ? |
| 05:53.35 | din3sh | atcom |
| 05:54.25 | din3sh | there's no DND activated |
| 05:55.32 | MrOli | and it was working fine before, or it's always been like that ? |
| 05:55.56 | din3sh | was working ok |
| 05:56.05 | MrOli | what's the model# ? |
| 05:56.09 | din3sh | now its more frequent |
| 05:56.14 | din3sh | atcom AT620 |
| 05:58.28 | MrOli | downloading manual |
| 05:59.25 | MrOli | the manufacturer must have a modem to host their website.. 30mns to download a 2MB PDF file! |
| 06:00.25 | din3sh | woooo |
| 06:00.25 | din3sh | whats your bandwidth? |
| 06:00.25 | din3sh | speed i mean |
| 06:03.11 | *** join/#asterisk din3sh (~din3sh@41.212.203.135) |
| 06:03.13 | MrOli | 20MB |
| 06:03.20 | din3sh | wow! |
| 06:03.30 | din3sh | still 30mins to download the manual |
| 06:07.52 | p3nguin | 20 MegaBytes of what? |
| 06:08.07 | din3sh | i have been unable to replicate the same scenario with same settings |
| 06:08.09 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 06:08.20 | p3nguin | I'd guess the phone is in use. |
| 06:08.37 | p3nguin | If there is no DND(busy), then it must be in use. |
| 06:08.39 | din3sh | i've set call-limit to 6 |
| 06:08.43 | MrOli | p3nguin: i was thinking of that, with a call limit issue |
| 06:08.54 | p3nguin | Doesn't mean the phone supports more than one call. |
| 06:09.08 | p3nguin | Check the call waiting option. |
| 06:09.18 | din3sh | ok let me check |
| 06:10.46 | p3nguin | I'm not familiar with that phone, but some phones allow users to turn call waiting on and off. |
| 06:11.52 | MrOli | din3sh: are you able to login to the web configuration interface of the phones ? |
| 06:13.11 | din3sh | yes can log in |
| 06:17.19 | MrOli | if you go under SIP/ Advanced SIP settings... |
| 06:17.53 | MrOli | is "ban anon call" checked ? |
| 06:17.59 | MrOli | what do you have under " forward type" ? |
| 06:19.11 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:19.13 | schmidts | good morning |
| 06:20.01 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 06:20.45 | din3sh | MrOli: ban anon call is not checked |
| 06:21.01 | MrOli | din3sh: also under " Call Service Setting" , there are several options that might play a role: "no disturb", "enable call waiting", and "accept any call" |
| 06:21.04 | din3sh | Foward type = off |
| 06:22.25 | din3sh | no disturb = not checked |
| 06:22.53 | din3sh | enable call waiting =checked |
| 06:23.04 | din3sh | accept any call =checked |
| 06:23.38 | din3sh | also i tried 2 simultaneous calls on a test phone jst now, it receives the 2nd call alright |
| 06:24.11 | din3sh | on a 3rd call, i get ERROR[17115]: chan_sip.c:3283 update_call_counter: Call to peer '7665' rejected due to usage limit of 2 |
| 06:24.26 | din3sh | due to the call-limit 2 i have set in sip_buddies |
| 06:24.43 | din3sh | means phone can receive more than 1 call |
| 06:24.59 | din3sh | the sip 486 is no on a production box |
| 06:26.16 | din3sh | am unable to replicate the sip 486 at my office :/ |
| 06:28.55 | din3sh | I get sip 480 busy here if i set DND on the phone, not 486 error |
| 06:28.56 | din3sh | :s |
| 06:32.14 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 06:35.14 | din3sh | :'( |
| 06:39.36 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:bd5f:fbbf:7e33:fffd) |
| 06:46.10 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 06:48.54 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1006) |
| 06:55.57 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 06:57.07 | *** join/#asterisk mcaj (mcaj@nat/novell/x-nhtnhtmctmfffbxs) |
| 07:01.56 | *** join/#asterisk ndemir (~ndemir@85.105.28.77) |
| 07:02.14 | schmidts | din3sh DND produces mostly a busy |
| 07:02.49 | ndemir | Hello. I have an asterisk server. I want to call outbound via SIP with my analog phones. Can i do this with a FXS card? |
| 07:03.19 | din3sh | what would be the cause of the 486 busy if DND is not active? |
| 07:03.21 | *** join/#asterisk Boohbah (~Boohbah@gateway/tor-sasl/boohbah) |
| 07:04.06 | *** part/#asterisk Boohbah (~Boohbah@gateway/tor-sasl/boohbah) |
| 07:04.32 | schmidts | din3sh sorry my fault 486 is busy here, 480 means temporary unavailable so 486 is right |
| 07:05.03 | schmidts | and you can get it back when the phone is in use and you dont have call waiting deactivated |
| 07:05.33 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 07:07.18 | *** join/#asterisk lucky (~lucky@unaffiliated/lucky) |
| 07:08.07 | ndemir | Hello. I have an asterisk server. I want to call outbound via SIP with my analog phones. Can i do this with a FXS card? |
| 07:08.19 | schmidts | ndemir yes |
| 07:08.34 | ndemir | schmidts: thanks |
| 07:09.37 | lucky | Hi folks, I realize this is a bit off topic but I was hoping someone might know.. are there any commercial providers or services that will enable regular SMS to a DID pass through to SIP clients normally, or some other similarly workable approach? |
| 07:13.31 | ChannelZ | http://www.vitelity.com/services/sms |
| 07:15.09 | ChannelZ | Although I'm not positive by what means they deliver the service |
| 07:17.12 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 07:18.51 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:19.22 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:25.56 | *** join/#asterisk wonderworld (~ww@dsdf-4db54223.pool.mediaWays.net) |
| 07:30.24 | *** join/#asterisk bulkorok (~bulkorok@217.110.197.225) |
| 07:31.17 | *** join/#asterisk joecool (~joecool@no-sources/joecool) |
| 07:31.55 | joecool | is there any way to get better debug info out of reload_conf... I uninstalled and reinstalled the feature code module in freepbx and it broke something |
| 07:32.09 | joecool | attempting to reload conf in freepbx throws out |
| 07:32.10 | joecool | Reload failed because retrieve_conf encountered an error: 20 |
| 07:32.14 | beardy | Where do you think you are? |
| 07:32.43 | beardy | #freepbx is ----> over there |
| 07:33.53 | joecool | ok |
| 07:34.45 | din3sh | :) |
| 07:35.51 | joecool | on the topic of asterisk still, reload_conf is a part of asterisk |
| 07:35.57 | joecool | can I get better output out of it? |
| 07:36.02 | joecool | -bash-3.2$ ./retrieve_conf --debug |
| 07:36.03 | joecool | Checking for PEAR Console::Getopt..OK |
| 07:36.03 | joecool | Aborting reload because extension conflicts or bad destinations |
| 07:36.06 | joecool | ^^ this does not help me |
| 07:48.41 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 07:48.42 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 07:48.55 | *** join/#asterisk darkskiez (~mhb@fsf/member/darkskiez) |
| 07:59.26 | Gugge | joecool: reload_conf? where is that in asterisk? |
| 08:00.43 | joecool | retrieve_conf I meant |
| 08:01.17 | joecool | (it was a typo, the prompt i pasted showed the correction) |
| 08:02.58 | din3sh | The device state of this queue member, SIP/xxxx, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. |
| 08:03.08 | din3sh | where is upgrade.txt? |
| 08:03.13 | din3sh | sorry for stupid question |
| 08:04.03 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 08:11.50 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:19.59 | Gugge | joecool: retrieve_conf seems like something that uses php / pear ... what makes you think it is part of asterisk? |
| 08:20.31 | Gugge | din3sh: in the source tgz |
| 08:20.32 | joecool | its location |
| 08:20.38 | joecool | in /var/lib/asterisk/bin/ |
| 08:20.48 | Gugge | joecool: well, it has nothing to do with asterisk |
| 08:21.22 | joecool | it's a rather stupid location to be thrown then if it has nothing to do with asterisk |
| 08:21.28 | Gugge | #freepbx is rather stupid |
| 08:21.50 | Gugge | s/#// |
| 08:23.47 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:25.09 | din3sh | why is freepbx not good? |
| 08:26.06 | *** part/#asterisk ndemir (~ndemir@85.105.28.77) |
| 08:27.13 | Gugge | Its probably fine. I dont use it. And imho its messy. |
| 08:27.36 | Gugge | And this channel does not support it. #freepbx does. |
| 08:27.50 | joecool | it's friggin insecure too |
| 08:27.53 | din3sh | elastix also messy? |
| 08:29.17 | din3sh | doesnt have any freepbx in prod :) |
| 08:32.26 | *** join/#asterisk Dovid (~Dovid@office.mypbxmanager.net) |
| 08:43.48 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:44.58 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 08:48.21 | *** join/#asterisk Tim_Toady (~moi@130.43.51.201.dsl.dyn.forthnet.gr) |
| 09:18.48 | *** join/#asterisk brian98 (~brian98@188.141.12.34) |
| 09:38.39 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 09:41.46 | din3sh | how to use loadtest.tcl? |
| 09:46.40 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 09:48.49 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 09:49.45 | jacc0 | hi all! |
| 09:50.56 | jacc0 | how can I use values from database as [globals] in extensions.conf? |
| 09:52.00 | wdoekes2 | jacc0: if they're mutable, they shouldn't be global. if they're not mutable, why are they in the db? |
| 09:52.52 | jacc0 | they are mutable (DTMFToneLenght and InterDTMFTimeout) |
| 09:53.32 | jacc0 | but I don't want to access the database everytime I send dtmf, and not even in with every incomming connection |
| 09:54.11 | jacc0 | It would generate a lot of useless load on mysql |
| 09:54.33 | jacc0 | because the are only configured once; when the project gets deployed |
| 09:55.16 | wdoekes2 | if they're configured once, I wouldn't call them mutable |
| 09:56.43 | wdoekes2 | perhaps you need to write them in your xyz.conf when deploying (#include extensions_site_specific.conf) or load the extensions.conf from the db (static realtime) |
| 09:57.39 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 09:59.57 | kaldemar | jacc0: or "#exec <script>" under [globals], where <script> is something that outputs those variables. maybe. |
| 10:04.53 | jacc0 | :) damn, kaldemar |
| 10:05.10 | jacc0 | nice sollution |
| 10:05.39 | jacc0 | :) |
| 10:05.49 | din3sh | i get this when answering a call on Snom360 [logger.c: RTCP SR transmission error, rtcp halted] |
| 10:05.54 | din3sh | any idea y? |
| 10:07.40 | jacc0 | make sure all your SIP phones have silence suppression set to OFF |
| 10:16.10 | jacc0 | din3sh: http://www.tek-tips.com/viewthread.cfm?qid=1341194 |
| 10:17.25 | din3sh | with the same phone, when it transfers to another phone and then re-transfer to a third phone |
| 10:17.39 | din3sh | caller hears only MOH, callee hears caller talking |
| 10:17.49 | din3sh | might be related to rtcp prob ? |
| 10:18.36 | jacc0 | try disabeling canreinvite or directmedia |
| 10:18.42 | jacc0 | in sip.conf |
| 10:19.22 | din3sh | no directmedia in 1.4.x |
| 10:19.32 | jacc0 | true, |
| 10:19.39 | jacc0 | it's canreinvite in 1.4 |
| 10:19.54 | jacc0 | correct me if i'm wrong |
| 10:20.11 | din3sh | yes |
| 10:20.20 | din3sh | set it to no for that peer? |
| 10:23.27 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 10:32.33 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 10:35.54 | *** join/#asterisk kl4m (~kl4m@gw2.noc1.sys-tech.net) |
| 10:37.36 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 10:45.10 | jacc0 | yes |
| 10:45.20 | jacc0 | maybe set it to no for all clients |
| 10:46.06 | din3sh | :/ |
| 10:46.08 | din3sh | i tried |
| 10:46.16 | din3sh | canreinvite already no |
| 10:46.20 | din3sh | not helping |
| 10:46.29 | din3sh | still with the rtcp error |
| 10:55.01 | *** join/#asterisk hariom (~hariom@115.118.173.77) |
| 10:59.34 | *** join/#asterisk ermali82 (~ermalkris@213.207.42.45) |
| 11:01.17 | ermali82 | does anyone have success with UUI "USERUSERINFO" ? |
| 11:22.06 | *** part/#asterisk hariom (~hariom@115.118.173.77) |
| 11:22.32 | jacc0 | did you do a 'sip reload' after setting canreinvite =no ? |
| 11:23.52 | din3sh | it was already running with canreinvite=no |
| 11:24.03 | din3sh | its jst a notice i guess |
| 11:24.11 | din3sh | coz it doesnt break the call flow |
| 11:24.35 | kaldemar | ermali82: did you modify source to enable it and recompile? |
| 11:33.58 | din3sh | my main problem remains double transfers, callee can only hear moh |
| 11:34.09 | din3sh | even if call has been transfered |
| 11:34.37 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 11:34.50 | *** part/#asterisk lwbnet (~luke@194.71.111.4) |
| 11:36.15 | jurij1 | after restoring 32bit freepbx backup to 64bit i get: WARNING[4999] loader.c: Unable to open modules directory '/usr/lib/asterisk/modules'. |
| 11:36.26 | jurij1 | how do i set path to /usr/lib64/...? |
| 11:41.19 | *** join/#asterisk usrb1n (~pixfix@174.163.216.87.static.jazztel.es) |
| 11:46.13 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:46.13 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:49.42 | *** join/#asterisk tompaw (~tompaw@slave30.tesserakt.eu) |
| 11:52.24 | wdoekes2 | jurij1: astmoddir in asterisk.conf ? |
| 11:53.30 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 11:58.31 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 11:59.31 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-15-179.bois.qwest.net) |
| 12:06.42 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:06.59 | *** join/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 12:07.16 | *** join/#asterisk bmg505 (~leon@196-209-123-205.dynamic.isadsl.co.za) |
| 12:09.29 | ermali82 | kaldemar: modifyed sig_pri.c to enable USERUSERINFO but still cant set the value to a variable |
| 12:10.09 | ermali82 | by enablind debuging "pri intense debug span X" i see the useruserinfo |
| 12:13.11 | WIMPy | Since when is UUS supported? |
| 12:14.25 | *** join/#asterisk gusto (~gusto@nrbg-4dbe1fdb.pool.mediaWays.net) |
| 12:14.30 | gusto | ts |
| 12:14.43 | gusto | i had a partial success on my be-converged voip |
| 12:15.20 | gusto | today i managed to get through ... but when i tried to call my telephone i got "die von ihnen gewaehlte rufnummer ist VON IHREM ANSCHLUSS NICHT ERREICHBAR" |
| 12:15.42 | gusto | so ... my asterisk runs on openwrt so it does not have sound bullshitting installed |
| 12:15.52 | *** join/#asterisk albertoandrade (~albertoan@186.212.37.227) |
| 12:15.55 | WIMPy | So that was not a geo number? |
| 12:15.55 | gusto | so i am sure that it was the telefonica provider |
| 12:16.05 | gusto | ha? |
| 12:16.08 | gusto | what is GEO? |
| 12:16.16 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 12:16.19 | WIMPy | geographical |
| 12:16.28 | WIMPy | Not a service number. |
| 12:16.42 | gusto | w8 |
| 12:16.45 | gusto | i ll try |
| 12:16.57 | gusto | doesnt help |
| 12:17.30 | gusto | w8 |
| 12:17.32 | *** join/#asterisk albertoandrade (~albertoan@186.212.37.227) |
| 12:17.33 | WIMPy | BTW: Where did you get an open account from Telefonica? |
| 12:17.40 | *** part/#asterisk jurij1 (~jurij@213.215.248.66) |
| 12:17.43 | gusto | i try to reconfigure it once again, because i got BAD REQUEST URI NOW |
| 12:17.45 | gusto | w8 |
| 12:19.00 | gusto | shit |
| 12:19.02 | gusto | does not help |
| 12:19.09 | gusto | maybe i should try another number |
| 12:19.26 | gusto | WIMPy: i got that through my dsl provider - endesha |
| 12:19.47 | gusto | the do sell dsl accounts and you get that VoIP for free ... either you use it or not |
| 12:22.14 | WIMPy | Heared that name for the first time. I only know the Alice accounts where you don't get your account data. |
| 12:25.07 | bobb_WU | can someone give me advice on how to recompile dahdi for a new kernel? |
| 12:25.14 | leifmadsen | make |
| 12:25.46 | leifmadsen | one you have the new kernels development modules installed, just run 'make install' again |
| 12:25.53 | bobb_WU | i'm on gentoo so i'm using emerge |
| 12:26.06 | gusto | WIMPy: i did have alice dsl too |
| 12:26.06 | bobb_WU | i'm running the new kernel and have tried recompile (through emerge) like 3 times |
| 12:26.19 | leifmadsen | then that's not the question you asked -- use emerge to manage the modules then |
| 12:26.44 | leifmadsen | sounds like you might want to check with gentoo to figure out why emerge can't help you then |
| 12:26.50 | gusto | WIMPy: that is cool, you get this shitty big dsl modem from bautzen and you connect your phone directly to there, but i am not sure if that is VoIP .. it maybe just Telephony over DSL |
| 12:27.25 | gusto | i am going to try another provider, maybe i get luck with getting at least ringing |
| 12:27.25 | leifmadsen | gusto: I think that's just called PSTN :) |
| 12:27.30 | WIMPy | gusto: No. It's SIP. |
| 12:27.41 | bobb_WU | yeah i'll head into gentoo's room to ask |
| 12:27.42 | bobb_WU | thanks leif |
| 12:27.47 | gusto | leifmadsen: no, because it is connected to the DSL modem |
| 12:28.19 | gusto | leifmadsen: it is not connected to the analogue phone, i got two phones simultaneously and could call myself ... one on analogue PSTN and then that alicedsl |
| 12:28.28 | WIMPy | leifmadsen: We don't have much PSTN left here. |
| 12:28.37 | gusto | WIMPy: where? |
| 12:28.38 | leifmadsen | gusto: I was just entertained by the thought of "telephony over DSL" |
| 12:28.47 | WIMPy | Germany |
| 12:28.52 | gusto | leifmadsen: would be a good idea |
| 12:29.07 | leifmadsen | I think that's just either PSTN or VoIP :) |
| 12:29.10 | gusto | WIMPy: bullshit, here in germany there is everything done by analogue telephony, in slovakia too |
| 12:29.13 | WIMPy | In DK they use VoDSL. |
| 12:29.19 | gusto | yes |
| 12:29.22 | gusto | that is a cool idea |
| 12:29.31 | gusto | VoDSL :-D |
| 12:30.14 | gusto | because then you do not have all this shit like ATMinsideEthernetinsidePPPinsideEthernet and that all incubating IP |
| 12:30.16 | WIMPy | Then tell me where you get an analogue telephone line? DTAG is the only one an they will try to sell you VOIP as well. |
| 12:30.39 | leifmadsen | orders himself up some internet over ethernet |
| 12:30.47 | gusto | i got an analogue telephone from DT till february |
| 12:30.52 | din3sh | :D |
| 12:30.53 | gusto | and in slovakia i still have one |
| 12:30.55 | gusto | or more |
| 12:31.05 | gusto | and ISDN is also still around |
| 12:31.06 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85fd.bcn.adamo.es) |
| 12:31.26 | gusto | i mean the problem is the monopol of DTAG/GMBH in germany |
| 12:31.27 | bobb_WU | ah i figured it out |
| 12:31.36 | gusto | no surprise no one other sells it |
| 12:31.38 | WIMPy | Yes, tey still exist, but it you want a new connection there are only few options left. |
| 12:31.48 | gusto | well, there is only one |
| 12:31.58 | gusto | either you want a phone or you do not |
| 12:32.14 | WIMPy | You can still get real ISDN from Vodafone. |
| 12:32.22 | gusto | so so |
| 12:32.24 | gusto | :-D |
| 12:32.32 | leifmadsen | bobb_WU: for posterity sake (logging) can you say what you ran into and how you fixed it? |
| 12:33.11 | gusto | in the meantime i sent a mail to my provider that i get this voice message that i can not call any number |
| 12:33.14 | gusto | however |
| 12:34.06 | WIMPy | BTW: Are there any plans or has anyone heard any roumors about plans to support outgoing overlap dial in chan_sip? |
| 12:34.20 | gusto | btw. did you also noticed that the most VoIP providers are idiots? i mean ... i subscribed to some prepaid providers but their systems do not seem to be configured right as well |
| 12:34.20 | leifmadsen | outgoing overlap? |
| 12:34.30 | WIMPy | At least Telefonica supports it so that would be really great. |
| 12:34.31 | leifmadsen | what is the difference between incoming overlap? |
| 12:34.44 | gusto | WIMPy: what? |
| 12:35.00 | bobb_WU | sure thing leif |
| 12:35.09 | WIMPy | leifmadsen: Incoming works, outgoing is not implemented. |
| 12:35.19 | leifmadsen | ah, ok got it |
| 12:35.20 | bobb_WU | i updated my kernel yesterday and had to update the link or something |
| 12:35.25 | leifmadsen | took me a minute to think |
| 12:35.44 | bobb_WU | the command was "eselect kernel list" then i had to choose the right number with "eselect kernel set 3" |
| 12:35.56 | bobb_WU | but that -3- was the correct number for my system |
| 12:36.14 | din3sh | i have a problem whereby double legged attended transfers are not bridged, caller continues to hear MOH, while callee can hear caller talking |
| 12:36.19 | din3sh | plzz helppppp :/ |
| 12:36.21 | bobb_WU | at that point, re-emerge dahdi and it made the drivers in the correct spot |
| 12:36.26 | gusto | btw. when someone calls my number ... how do i tell asterisk in the dialplan to send it to my ata-adapter/phone? |
| 12:36.27 | din3sh | bangs his head on his desk |
| 12:37.11 | gusto | din3sh: i am banging my head on my desk for three months already - since i got in contact with VoIP /asterisk/ |
| 12:37.26 | ectospasm | din3sh: what version of Asterisk, what technologies (SIP/DAHDI/etc.) are you using to bridge the attended transfer? |
| 12:37.32 | gusto | but i make progress |
| 12:37.42 | din3sh | 1.4.42 |
| 12:37.46 | gusto | everyday i come a bit forward .. now i heard something |
| 12:37.54 | gusto | din3sh: is that your version of asterisk? |
| 12:37.59 | WIMPy | gusto: Same for everyone. Maybe som day it will become usable :-) |
| 12:38.07 | din3sh | happens on both SIP-SIP-SIP, DAHDI-SIP-SIP |
| 12:38.28 | leifmadsen | din3sh: pretty sure that was just an issue with 1.4 that wasn't fixed |
| 12:38.29 | ectospasm | Asterisk 1.4 will be EOL April 21st. |
| 12:38.32 | leifmadsen | could be wrong |
| 12:38.36 | gusto | yes |
| 12:38.42 | gusto | i am using asterisk18 |
| 12:38.45 | din3sh | arrgh |
| 12:38.51 | gusto | and everyone should use asterisk18 by now |
| 12:38.53 | din3sh | 1.8 or 18? |
| 12:38.54 | din3sh | :p |
| 12:38.56 | gusto | 1.8 |
| 12:38.57 | leifmadsen | 1.8 :) |
| 12:39.10 | gusto | asterisk18 is the package name to openwrt/opensuse too/ |
| 12:39.20 | ectospasm | same on AsteriskNOW |
| 12:39.59 | din3sh | ok ok |
| 12:40.13 | din3sh | i got it, i gotta move to asterisk18 |
| 12:40.24 | jacc0 | or asterisk 10 |
| 12:40.36 | din3sh | asterisk20 |
| 12:40.41 | ectospasm | remember, 1.8 is Long Term Support (LTS) release, and will be fully supported longer than 10. |
| 12:40.54 | din3sh | uhuh |
| 12:40.57 | gusto | yes |
| 12:41.00 | bobb_WU | my odbc functions aren't registered in asterisk :/ how do i force them to try and register so i can see the output in the CLI? |
| 12:41.03 | gusto | and asterisk18 is easy to find |
| 12:41.04 | ectospasm | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 12:41.12 | leifmadsen | bobb_WU: module load func_odbc.so |
| 12:41.22 | gusto | i honestly do not know where to get asterisk 10.0 .... exept by compiling it |
| 12:41.26 | leifmadsen | bobb_WU: or module reload func_odbc.so if you've made changes |
| 12:41.30 | bobb_WU | just says it can't load the module |
| 12:41.52 | leifmadsen | gusto: http://packages.asterisk.org/ |
| 12:41.52 | *** join/#asterisk asteriskser_ (c584b45c@gateway/web/freenode/ip.197.132.180.92) |
| 12:42.00 | ectospasm | does the module exist, bobb_WU? |
| 12:42.04 | leifmadsen | bobb_WU: do you have res_odbc.so loaded? |
| 12:42.13 | leifmadsen | did you compile odbc related modules? |
| 12:42.16 | leifmadsen | guesses not |
| 12:42.27 | gusto | hahahahahaha |
| 12:42.28 | leifmadsen | ls /usr/lib/asterisk/modules/*odbc*.so |
| 12:42.29 | gusto | centos :-D |
| 12:42.38 | leifmadsen | gusto: not sure why that is funny |
| 12:42.40 | bobb_WU | well yeah it was a working system before i took it down for upgrades |
| 12:42.59 | bobb_WU | odbc is in my make file and i just recompiled the psqlodbc package |
| 12:43.02 | leifmadsen | pabelanger: change management! :) |
| 12:43.02 | gusto | leifmadsen: because there is only the 5 release |
| 12:43.10 | leifmadsen | looks at Qwell for that |
| 12:43.11 | gusto | who does still use the centos5 release |
| 12:43.12 | din3sh | upgrades on a working system? |
| 12:43.14 | gusto | that is old |
| 12:43.18 | leifmadsen | gusto: lots of people including myself |
| 12:43.21 | din3sh | don't fix whats not broken |
| 12:43.22 | gusto | :-P |
| 12:43.23 | din3sh | :D |
| 12:43.26 | leifmadsen | 5.8 was released like 2 weeks ago |
| 12:43.28 | gusto | i have centos6 on my server |
| 12:43.35 | leifmadsen | good for you, we're all very proud of you |
| 12:43.37 | bobb_WU | i was on 2.6 kernel and 1.6 asterisk |
| 12:43.57 | bobb_WU | i'm working on one system so i can image and upgrade all of our nodes (10 or so) |
| 12:43.58 | gusto | and honestly, i do not like it too much, i would be far more happy with a freebsd jail or something ... but i did not try it yet |
| 12:44.11 | asteriskser_ | what are problems relating to asterisk scalability i read alot about, but i really don't see anyproblem can some one please shed some light on that issue ? |
| 12:44.22 | ectospasm | bobb_WU: upgrading shouldn't be without headache then, depending on what version of 1.6 you're coming from, there could be two or three major revisions in between... |
| 12:44.28 | leifmadsen | asteriskser_: please ask a less vague question |
| 12:44.34 | asteriskser_ | ok |
| 12:45.17 | bobb_WU | i'll try to re-compile asterisk since its not finding modules- maybe the kernel eselect trick will fix it |
| 12:45.28 | gusto | btw. do you know the difference between german and slovak post? |
| 12:45.48 | asteriskser_ | presence for example we can do that in asterisk , why do we need scf then ? |
| 12:45.58 | gusto | the german one delivers too late and the slovak one too soon |
| 12:46.17 | gusto | the consumer is pissed anyway |
| 12:46.32 | leifmadsen | asteriskser_: because you don't understand what Asterisk SCF does -- Asterisk SCF has the capability to do live call failover for instance |
| 12:46.35 | WIMPy | Too soon? How does that work? |
| 12:46.54 | din3sh | delivers before you even post |
| 12:46.59 | fprior | Hi, Is there any way to tell Asterisk not to generate additional headers X-Asterisk-HangupCause and X-Asterisk-HangupCauseCode ? |
| 12:47.24 | gusto | WIMPy: like i said, i sent myself a package and i did not manage to sleep over and it was there |
| 12:47.48 | gusto | WIMPy: and according to DHL the package was still on route while i already had it |
| 12:47.50 | WIMPy | And what's bad about that? |
| 12:47.56 | asteriskser_ | leifmadsen: what is i understand about scf is extensibility i.e adding more modules easily , availability which we can do with asterisk redundancy |
| 12:48.00 | gusto | WIMPy: because i did not expect it |
| 12:48.04 | WIMPy | That's normal. |
| 12:48.17 | asteriskser_ | leifmadsen: but my real question is |
| 12:48.35 | WIMPy | 's got the impression DHL only syncs the data once a day. |
| 12:48.36 | leifmadsen | asteriskser_: in many cases, Asterisk SCF may not be required, but Asterisk SCF is more of a framework and platform for a large number of calls. Asterisk is a PBX that you can distribute. |
| 12:48.38 | gusto | WIMPy: and it was just luck that they put it down behind my fence and there was good wather so i took it over while wanting to get something to eat |
| 12:48.55 | leifmadsen | you might even use them together |
| 12:49.01 | leifmadsen | and in many cases probably would |
| 12:49.28 | asteriskser_ | leifmadsen: yes , but the issues of handling large number of calls can be done with kamilio for instance right ? |
| 12:49.59 | leifmadsen | asteriskser_: sure, but its a SIP proxy |
| 12:50.10 | leifmadsen | it doesn't handle live call failover |
| 12:50.46 | leifmadsen | "Asterisk SCF is a framework that allows developers to create real-time communications applications that include voice, video and text and that meet the demands of a full range of uses, from embedded applications to enterprise and carrier solutions. Asterisk SCF is architected to provide the highest levels of availability, scalability, extensibility, fault-tolerance and performance." |
| 12:50.48 | gusto | is it a problem when registry gets another IP address than peer does? i mean one provider SRV record resolves to many ip addresses and so i have two different ip addresses for one provider record for peer and for registry |
| 12:51.24 | bobb_WU | leifmadsen: the modules weren't there so selecting the newest kernel and recompiling fixed my odbc issues. now to fix the database issues and maybe i can make a call |
| 12:51.34 | gusto | WIMPy: you know what the biggest problem is with telefonica/reselers and their VoIP? |
| 12:51.38 | WIMPy | gusto: You may not match an incoming call. |
| 12:52.04 | WIMPy | As I said: I only know the Alice thing. |
| 12:52.10 | gusto | WIMPy: its that you can not contact telefonica directly and the reseller does not know what you re talking about (because they are incompetent) |
| 12:52.30 | gusto | WIMPy: what do you mean by incoming call matching? |
| 12:52.32 | asteriskser_ | leifmadsen: ok please correct me if i am wrong , if i am only intersted in voice but with large scale and i will only use sip i can use asterisk and it will scale well right ? |
| 12:52.42 | WIMPy | I think that will be the same everywhere. |
| 12:52.46 | leifmadsen | asteriskser_: maybe |
| 12:52.56 | leifmadsen | it depends how you implement it and what you mean by scale |
| 12:53.02 | WIMPy | Although the Alice support has been quite good so far. |
| 12:53.08 | asteriskser_ | i mean over 10000 calls |
| 12:53.18 | gusto | nooo ... alice ... i doubt it |
| 12:53.18 | asteriskser_ | distrbuted stats |
| 12:53.22 | leifmadsen | I'm working on a system that has 50 physical boxes and handles over 250 different PBXs |
| 12:53.27 | gusto | WIMPy: what do you mean by incoming call matching? |
| 12:53.42 | leifmadsen | asteriskser_: it's infinitely scalable depending on how you architect such a system |
| 12:53.42 | WIMPy | gusto: If the call comes from another IP that you got for the peer, it won't match. It then depends on you allowing guest calls. |
| 12:53.47 | *** join/#asterisk Carlos_PHX1_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 12:54.07 | leifmadsen | asteriskser_: one physical box will not handle 10,000 simultaneous calls (probably) |
| 12:54.34 | asteriskser_ | leifmadsen: the only reason i am asking this questions is i find in many slides of asticon the issue of scalabilty raised |
| 12:54.35 | WIMPy | So unless they authenticate, wich I guess they don't, you either need to create one peer per IP or accept guest calls. |
| 12:54.38 | leifmadsen | asteriskser_: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Clustering.html |
| 12:54.42 | gusto | WIMPy: and do you think that that could cause that voice message? wouldnt that end up in an asterisk debug error? |
| 12:55.04 | WIMPy | That's incoming only. |
| 12:55.06 | leifmadsen | asteriskser_: sure, and those people probably didn't architect their systems correctly or didn't have the ability to understand how |
| 12:55.06 | asteriskser_ | leifmadsen: that is assuming no transcoding |
| 12:55.20 | leifmadsen | you haven't been clean what you mean by scalable |
| 12:55.26 | leifmadsen | s/clean/clear/ |
| 12:55.35 | gusto | WIMPy: where do i adjust guest calls/ |
| 12:55.45 | gusto | ? |
| 12:55.55 | gusto | is going to the toilet first |
| 12:56.10 | leifmadsen | if you have a box that handles 250 calls and have 40 boxes you're at 10,000 calls |
| 12:56.30 | WIMPy | gusto: allowguest |
| 12:56.52 | *** part/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 12:58.08 | leifmadsen | asteriskser_: the problem with your question is there are many many variables associated with your question, so it's not possible to answer with a yes or no. It depends entirely on what you're doing, and how you're architecting the system. Transcoding is just one aspect. How many calls per second? how fast are you setting up calls? how long are calls lasting? is there recording? are you using ramdisks? are you evenly dist |
| 12:58.08 | leifmadsen | ributing calls amongst the systems? how many systems do you have? -- I could go on |
| 12:58.33 | leifmadsen | Is Asterisk scalable? Of course it is. You just have to know how to architect the system. |
| 12:59.36 | asteriskser_ | leifmadsen: thank you that is the answer that i was looking for , because i didn't see anything in the way of scaling asterisk however those slides that are floating around on the web made me worried |
| 13:00.34 | leifmadsen | asteriskser_: I have no idea what slides you're talking about -- you haven't watched my presentations on Asterisk clustering then I guess |
| 13:00.50 | asteriskser_ | leifmadsen: can i have link please |
| 13:01.03 | asteriskser_ | leifmadsen:*the |
| 13:01.08 | leifmadsen | asteriskser_: also when you search on Google, be very careful about what the dates are that come back. In many cases stuff from 2006 comes back before new information because it has been clicked on more times. |
| 13:01.08 | *** join/#asterisk ios_sos (~nbeard@24-181-146-94.static.dlth.mn.charter.com) |
| 13:01.57 | asteriskser_ | leifmadsen: thanks for the heads up |
| 13:02.10 | leifmadsen | asteriskser_: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Clustering.html -=- http://www.astricon.net/videos/Asterisk-Powered-Distributed-Call-Centers.html |
| 13:02.22 | leifmadsen | http://www.astricon.net/videos/Clustering-and-Scaling-Asterisk-with-Kamailio.html |
| 13:03.11 | asteriskser_ | leifmadsen: thanks |
| 13:05.44 | bobb_WU | how do i check my database connections through the CLI? i'm having connection issues even though i'm pretty sure i have the right parameters defined. |
| 13:07.10 | bobb_WU | the user account has permission on the view (psql asterisk -U username) |
| 13:08.16 | leifmadsen | odbc show |
| 13:08.28 | *** join/#asterisk serafie (~erin@nat/digium/x-uclatgfkobfollzo) |
| 13:08.49 | bobb_WU | hmm it says its connected |
| 13:09.31 | bobb_WU | going for a quick smoke while postgres reboots |
| 13:15.10 | *** join/#asterisk Flumdahl (n30@shell.auth.se) |
| 13:15.17 | Flumdahl | how do i get reports to work so i can see the incoming call logs there? latest freepbx |
| 13:16.02 | gusto | was putting the toilet under heavy load |
| 13:16.11 | gusto | WIMPy: allowguest in sip.conf? |
| 13:16.26 | bobb_WU | "odbc show" has my local connect working, but when the query executes from the dialplan, the CLI says it can't connec |
| 13:16.30 | bobb_WU | *connect |
| 13:17.29 | [TK]D-Fender | gusto, Yes |
| 13:18.32 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-aqtnszpedzustkqk) |
| 13:18.32 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:18.40 | gusto | SIP/2.0 487 Request Terminated |
| 13:18.52 | gusto | does this look familiar to someone? |
| 13:19.03 | gusto | [TK]D-Fender: i am going to try that out |
| 13:19.08 | [TK]D-Fender | gusto, Single line messages like that with no sense of context don't tell us anything. |
| 13:19.32 | gusto | ; - allowguest (default enabled) |
| 13:19.53 | gusto | [TK]D-Fender: of course, but there is not more yet |
| 13:19.58 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 13:20.00 | [TK]D-Fender | gusto, One side gave up for whatever reason and just said "I'm done". We don't see what it is in response to. |
| 13:20.10 | gusto | the debug doesnt give much context |
| 13:20.12 | [TK]D-Fender | gusto, There is more... you're jsut not showing us |
| 13:20.21 | [TK]D-Fender | ~pb |
| 13:20.21 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 13:20.22 | [TK]D-Fender | ^^^^^^^ |
| 13:20.45 | gusto | i ll paste it when it happens again, i ll try that guest first |
| 13:21.07 | ectospasm | not to mention a pcap will go a long way in debugging SIP as well. |
| 13:21.14 | gusto | ah .. i have it enabled ... because it is not disabled anywhere |
| 13:21.29 | [TK]D-Fender | no need for pcap. * SIP debug is what counts |
| 13:21.47 | [TK]D-Fender | And is self contained and tells you more |
| 13:22.19 | Flumdahl | i have cdr reports module installed |
| 13:23.21 | gusto | so |
| 13:23.28 | [TK]D-Fender | Flumdahl, #freepbx , #asterisknow <- |
| 13:23.30 | [TK]D-Fender | ~freepbx |
| 13:23.30 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 13:23.38 | gusto | the guest one wasnt it and i just tried out to let down the firewall |
| 13:23.51 | gusto | i put it up again, because that did not help either |
| 13:24.02 | [TK]D-Fender | gusto, You are guessing and not showing. |
| 13:24.14 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:24.16 | gusto | [TK]D-Fender: w8, i ll get that paste |
| 13:25.24 | *** part/#asterisk bobb_WU (~bobb_WU@206.74.211.14) |
| 13:25.35 | *** join/#asterisk bobb_WU (~bobb_WU@206.74.211.14) |
| 13:25.35 | *** part/#asterisk bobb_WU (~bobb_WU@206.74.211.14) |
| 13:25.54 | *** join/#asterisk bobb_WU (~bobb_WU@206.74.211.14) |
| 13:26.47 | *** part/#asterisk Flumdahl (n30@shell.auth.se) |
| 13:33.15 | gusto | ah |
| 13:33.22 | gusto | as i see the authentification worked |
| 13:33.32 | gusto | should i include that in the pastebin as well? |
| 13:34.03 | gusto | oh |
| 13:34.04 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 13:34.18 | gusto | there is nothing to paste any more ... the 487 is not there any more |
| 13:34.46 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 13:34.52 | gusto | seems like my account is deactivated from the providers side ... nothing to do about it ... i already reported it |
| 13:35.17 | [TK]D-Fender | maybe whatever firewall you threw up in front is blocking things entirely. Hard to say as we have no details about what you're doing, where the call is really coming from, what else you have connecting to your server, etc. |
| 13:35.21 | gusto | but according to the log it looks very good |
| 13:35.52 | gusto | [TK]D-Fender: well, i get that voice error, so the connection to my provider stands |
| 13:36.12 | *** join/#asterisk asteriskser__ (c58456b0@gateway/web/freenode/ip.197.132.86.176) |
| 13:36.13 | gusto | [TK]D-Fender: also in the log ... i never seen such a lot of success messages from asterisk in my life |
| 13:36.15 | [TK]D-Fender | well if it not a file your server is playing back to you.... then it's them. |
| 13:36.31 | gusto | hahahahahaha |
| 13:36.40 | [TK]D-Fender | And I don't see a clear distinction on where you hear it. Incoming? Outgoing? |
| 13:36.49 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 13:36.52 | gusto | [TK]D-Fender: i already told everyone that my asterisk runs on opewrt ... there are no voice messages installed |
| 13:37.09 | gusto | [TK]D-Fender: and the voice error message is in german - where would asterisk get it from ? :-D |
| 13:37.27 | *** join/#asterisk darkmantis (~darkmanti@gemini.cybershade.org) |
| 13:37.27 | [TK]D-Fender | gusto, yes, and I don't see a proper full description of the call. |
| 13:37.48 | gusto | [TK]D-Fender: when i call a number ... it says in german- from my provider's side- also the log says that everything is OK, that i can not call that number from my account |
| 13:37.50 | [TK]D-Fender | is that outside calling in? Does it actually attempt to arrive at *? Is this a mesage on your calling out? |
| 13:38.51 | gusto | [TK]D-Fender: i am calling a number, but it does not ring, instead provider's installed voice says to me in german, that i am not allowed to call that number |
| 13:39.12 | [TK]D-Fender | gusto, Again, is that you calling OUT your sever to their service? |
| 13:39.13 | darkmantis | Hi Guys |
| 13:39.16 | gusto | [TK]D-Fender: we are talking about asterisk, so me calling outside ... not trying to call myself |
| 13:40.12 | gusto | [TK]D-Fender: i am calling my grandmother's PSTN over their voip service |
| 13:40.22 | gusto | [TK]D-Fender: or my own mobile phone number |
| 13:40.33 | gusto | [TK]D-Fender: throws the same messages on both |
| 13:40.50 | gusto | i would say ... i mastered it now |
| 13:41.56 | gusto | just calling myself and forwarding it to my ata is not done yet |
| 13:42.00 | [TK]D-Fender | <gusto> [TK]D-Fender: i am calling my grandmother's PSTN over their voip service <- this is what I was waiting to hear |
| 13:42.03 | gusto | that will be the next thing to do |
| 13:42.39 | [TK]D-Fender | gusto, So you call them, they give you audio and nd say there is some other problem. SIP isn't the problem. So either your number is, or their service doesn't want to reach them |
| 13:43.21 | gusto | [TK]D-Fender: yes, they are idiots, thats no news ... maybe the be-converged server is not configured properly |
| 13:43.48 | [TK]D-Fender | gusto, Have you double-checked the number you are passing them VS what format they say you should use? |
| 13:43.56 | gusto | [TK]D-Fender: or every number is blocked for my account so that i can not call anyone |
| 13:44.12 | gusto | [TK]D-Fender: yes, i tried several formats |
| 13:44.31 | [TK]D-Fender | gusto, Starting to sound like yuor provider is refusing. Ask them. |
| 13:44.39 | gusto | [TK]D-Fender: and the number is right because i was calling from my old telephone that was connected to PSTN before |
| 13:44.49 | gusto | [TK]D-Fender: yes, i already reported it |
| 13:46.07 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:46.07 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:46.16 | DarkMantis | Can anyone tell me what the best authentication security is for IAX/Asterisk or is that not relevant for here? |
| 13:46.20 | gusto | i am going to move to another providers i subscribed to ... so that i can use my working asterisk now |
| 13:46.42 | Chainsaw | DarkMantis: As always the most secure way to run a service is to firewall it off from the big bad internet. |
| 13:46.55 | Chainsaw | DarkMantis: And whitelist the IPs you want to talk to. |
| 13:46.56 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
| 13:46.59 | gusto | DarkMantis: the best security is not to use VoIP at all, or use a provider who does give you prepaid option |
| 13:47.09 | gusto | :-D |
| 13:47.20 | DarkMantis | I have got firewalls in place (hardware and software) however, I wold have thought there would be an authentication method (TLS or w/e) that could be used? |
| 13:47.45 | Chainsaw | DarkMantis: The downside of encryption is that it adds latency. |
| 13:48.02 | gusto | i was like smashed about that digest auth he does really everytime / so on in and out / like MD5 sums of some credentials ... hm ... impressively naive :-D |
| 13:48.07 | Chainsaw | DarkMantis: And humans are particularly sensitive to delays in audio (particularly if delayed echos result). |
| 13:48.35 | gusto | yes, the solution would be to let it do only authentification through TLS or something |
| 13:48.38 | DarkMantis | Chainsaw: Okay, so is there no recommended security type? |
| 13:48.41 | gusto | and not encrypting the whole thing |
| 13:48.57 | gusto | i would recommend TLS |
| 13:49.10 | bobb_WU | Chainsaw: are you the maintainer of the Gentoo asterisk packages? |
| 13:49.12 | DarkMantis | Oh yeah, sorry the only bit we intend to encrypt is the authentication. Nto so much the calls themselves |
| 13:49.18 | Chainsaw | bobb_WU: That is correct. |
| 13:49.25 | gusto | but no idea how to configure it, but i ve seen in the sip.conf that asterisk18 seems to support that |
| 13:49.28 | bobb_WU | thanks for all your help! |
| 13:50.04 | gusto | tss ... ppl still using gentoo :-D |
| 13:50.15 | DarkMantis | gusto: I am new to all the SIP/Asterisk/IAX so forgive my ignorance. I think the server I am workign on currently runs an older version of Asterisk |
| 13:50.16 | gusto | thaught that all moved to archlinux |
| 13:50.43 | gusto | DarkMantis: i am new to asterisk as well, so we are in the same boat |
| 13:50.45 | Chainsaw | gusto: I've only used it since 2003 or so. |
| 13:50.51 | Chainsaw | gusto: (Gentoo, not Asterisk) |
| 13:51.10 | gusto | Chainsaw: i did use gentoo a long time ago as well but moved away from it |
| 13:51.16 | Chainsaw | bobb_WU: I hope the dozen or so extra patches make it easier to use :) |
| 13:51.22 | DarkMantis | I use arch where possible too gusto :P |
| 13:51.24 | gusto | Chainsaw: it was cool when it came around |
| 13:51.34 | Chainsaw | (Queue Digium developers telling me off for carrying downstream patches) |
| 13:51.37 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:51.40 | gusto | DarkMantis: i do not use arch, but i think arch is the better choice |
| 13:51.50 | DarkMantis | Ah fair enough ^_^ |
| 13:52.00 | gusto | DarkMantis: i know someone who uses it |
| 13:52.19 | gusto | DarkMantis: i am more into FreeBSD, you know? |
| 13:53.00 | bobb_WU | chainsaw: it works with init.d scripts and makes the software maintenance so much easier, just wanted to thank you for all your hard work |
| 13:53.02 | DarkMantis | I knwo of it, but I can't say I have honestly ever used it |
| 13:53.07 | gusto | and NetBSD and OpenBSD from time to time ... also Dragonfly, but i did not manage to get it work as a native installation, maybe only in virtualbox |
| 13:53.33 | Chainsaw | bobb_WU: You are most welcome. I'm glad it is enjoyed :) |
| 13:53.38 | gusto | init.d ... well |
| 13:53.46 | bobb_WU | lol i don't know about enjoyed |
| 13:53.53 | bobb_WU | its being a huge headache right now |
| 13:54.12 | gusto | in respect to asterisk it could use also systemd, as long as you find a -nodaemon option |
| 13:54.12 | Chainsaw | Telephone systems are like copiers. People only talk about them when they fail. |
| 13:54.26 | Chainsaw | Which makes supporting them a bit of a thankless task. |
| 13:54.28 | [TK]D-Fender | DarkMantis, "older version" need to get a specific # associated to it, and that will be the limiting factor |
| 13:55.19 | DarkMantis | It is running 1.4 |
| 13:55.26 | gusto | well, i enjoyed that opinion of some guy back then in the centuries of inventors like Edison ... i have no idea who it was but he was mentioned by Obama in his speach lately |
| 13:55.38 | gusto | who said - telephone is a great invention, but who would want to use one? |
| 13:56.19 | gusto | me too, as soon as i get this VoIP s*** to work, i am not going to use it |
| 13:57.00 | gusto | maybe i build my own infrastructure with ATA adapters so that i can call asterisk to asterisk my friends and family and so on, but i will have to buy some more ata adapters for that |
| 13:58.45 | gusto | VoIP may be cool when you want to be reachable from PSTN telephones, and than it is not important what plan you have, maybe a prepaid account is enough with a local number in, i found a provider who can give me one, so that someone can call me ;-) or better not |
| 13:59.02 | DarkMantis | Sorry, what's PSTN? |
| 13:59.10 | gusto | DarkMantis: analogue telephone |
| 13:59.13 | DarkMantis | Ah right |
| 13:59.14 | DarkMantis | XD |
| 13:59.29 | gusto | DarkMantis: they call it here like that, it was new to me till yesterday too |
| 13:59.47 | gusto | who would imagine that i would call an analogue phone like PSTN .. no one |
| 13:59.56 | gusto | however ... not calling at all saves a lot of money |
| 14:01.17 | gusto | well, and when nothing else works, one can still find that guy who wrote the RFC's for VoIP and beat him up :-D |
| 14:01.26 | bobb_WU | how do i fix an "SQL Exceute error -1: HY000 Error: Permission denied for relation new "? |
| 14:01.40 | bobb_WU | i logged in to postgres with that user account and was able to query the view |
| 14:01.40 | gusto | yes |
| 14:01.51 | gusto | that seems like an permission arror inside the sql database |
| 14:02.03 | bobb_WU | i copied and pasted the password into the unixodbc parameter files |
| 14:02.12 | gusto | i would check if the user you are trying to use has permissions to do his job |
| 14:02.13 | bobb_WU | extended permissions to localhost in the pg_hba.conf file |
| 14:04.44 | gusto | so ... i am done for today |
| 14:12.43 | *** join/#asterisk kessius (~cassio@201.21.173.58) |
| 14:14.09 | bobb_WU | any advice on how to troubleshoot that error? |
| 14:14.28 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 14:18.56 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 14:18.59 | qakhan | all what is the right sequence of installaing asterisk, unimrcp, pocketsphinx and asterisk connector bridge ??? |
| 14:38.28 | aberrios | anyone compiled wanpipe 3.5.25 with dahdi 2.6.0 okay? having issues here. |
| 14:45.13 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 14:46.04 | aberrios | nvm |
| 14:50.04 | TSM | im getting audio distortion on a connection that one of our users has at their house, using either G722 or u/alaw I hear a razzing on the upper frequencies of the users voice |
| 14:52.00 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 14:53.30 | *** join/#asterisk gonewage (~gonewage@mail.sni-law.com) |
| 14:54.01 | *** join/#asterisk RogersJA (~RogersJA@S0106001346cbd191.gv.shawcable.net) |
| 14:54.02 | *** join/#asterisk wonderworld (~ww@dsdf-4db5edca.pool.mediaWays.net) |
| 14:54.33 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:54.46 | Chainsaw | TSM: Have you considered the possibility of it being microphone/client-specific? |
| 14:55.20 | *** join/#asterisk theHub (~theHub@69.177.93.21) |
| 14:55.35 | TSM | could be, the phone was taken from the office yesterday when it was working fine |
| 14:59.43 | *** join/#asterisk scubes13 (~scubes13@rrcs-70-60-211-241.midsouth.biz.rr.com) |
| 15:01.03 | Chainsaw | TSM: Could be something as mundane as the ultrasonic noise from a nearby CRT television being introduced into the signal. If the filter is not perfect, which it never is... |
| 15:01.22 | TSM | grrr true, ile tell the person to move the phone |
| 15:15.12 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 15:18.51 | jaytee | I just installed fail2ban yesterday on centos 5.7 running asterisk 1.6.2 and it doesn't seem to be adding anything to the asterisk jail. I've checked the configs against a working version on another pbx and I don't see anything different other than the ignoreip ranges. I've restarted the fail2ban daemon and iptables but no luck. Not sure what to check next. Anyone have a suggestion? |
| 15:22.34 | p3nguin | I'd like to see filter.d/asterisk.conf and jail.conf. |
| 15:23.20 | jaytee | ok, please hang on a sec while I pastebin them |
| 15:24.53 | jaytee | http://pastebin.com/n7gApK9Q < that's jail.conf |
| 15:26.02 | jaytee | http://pastebin.com/TbgpymxK < /etc/fail2ban/filter.d/asterisk.conf |
| 15:27.26 | ChannelZ | I don't think any of those Registration lines will match (what version of Asterisk?) |
| 15:27.43 | ChannelZ | I use NOTICE.* chan_sip\.c.* Registration from .* failed for '<HOST>:[0-9]+' - (Wrong password|No matching) |
| 15:27.54 | ChannelZ | and NOTICE.* chan_sip\.c.* Call from '.*' \(<HOST>:[0-9]+\) to extension '.*' rejected .* extension not found |
| 15:28.28 | jaytee | it's running Asterisk 1.6.2.18.2 |
| 15:28.32 | p3nguin | What scenario are you testing where it is failing to ban? |
| 15:29.06 | p3nguin | Registrations, unauthed calls, etc? |
| 15:29.30 | jaytee | I've used a softphone from outside with a bad password and also with a non existent sip account |
| 15:29.49 | p3nguin | Yes, but what scenario? |
| 15:29.54 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v022-171.mobile.uci.edu) |
| 15:32.13 | p3nguin | Telling me that you are using a specific type of device doesn't tell me what you are doing with it. |
| 15:32.16 | jaytee | I've tested registration using an incorrect password and I've also tested trying to register with an invalid account. Those two scenarios should match the wrong password regex and the No matching peer found. |
| 15:32.41 | p3nguin | Okay, registrations should certainly be tested and banned if needed. |
| 15:32.57 | jaytee | which is how I've tested it in the past |
| 15:34.26 | p3nguin | grep Registration /var/log/asterisk/messages |
| 15:34.31 | p3nguin | Do you see anything? |
| 15:38.58 | *** join/#asterisk orioni (~chatzilla@46.183.121.133) |
| 15:43.57 | jaytee | ok, now I'm getting bans and I found the main part of my problem. |
| 15:44.47 | jaytee | I had one outside ip address I was testing with was already in the ignoreip range. my bad (slaps self with a trout) |
| 15:45.28 | jaytee | and when I test from another it's banning when I try to register a valid account that doesn't match the ACL list. |
| 15:46.36 | jaytee | so if I remove the ban and remove the deny/permit on the test account and use a wrong password it will probably match and ban from that too. |
| 15:47.58 | TSM | why does this not evaluate to 1, '$[${EPOCH} <= ${RGSTART}]' |
| 15:48.29 | TSM | or the other way '$[${RGSTART} <= ${EPOCH}] |
| 15:48.34 | [TK]D-Fender | TSM, perhaps you should output those 2 variables first and look at them first. |
| 15:50.17 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 15:56.54 | *** join/#asterisk Hive (~Hive@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 16:00.57 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 16:04.50 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 16:08.21 | jaytee | p3nguin, after commenting out the deny/permit, removing the successful ban on the IP that matched the ACL regex I've retested and it's still not banning on bad password attempts |
| 16:08.44 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 16:10.32 | *** join/#asterisk VinceAntw (~VinceAntw@91.176.80.180) |
| 16:12.02 | p3nguin | grep "Registration.*failed" /var/log/asterisk/messages |
| 16:17.27 | p3nguin | Is there any way to limit the time an application or function can operate? I'm using the CURL() function to fetch a URL, but I need to limit the time it can run so that if the URL does not return after, say, 2 seconds, it will give up and move on rather than waiting for a definitive response from the web server. |
| 16:19.53 | p3nguin | If it is successful, it should return in under 1 second; if there is a problem, it can take up to 30 seconds to report the failure. I can't wait the 30 seconds for a failure. That's far too long for the call to sit and wait for something to happen at that step of dial plan. |
| 16:20.21 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 16:21.03 | *** join/#asterisk din3sh (din3sh@41.136.247.121) |
| 16:23.44 | jaytee | p3nguin, got it all working now. Had to fix a mangled regex for the Wrong password expression. Thanks for helping! |
| 16:24.02 | jaytee | I'm definitely getting old :-( |
| 16:30.36 | *** join/#asterisk Defraz (~Defraz@69.20.176.132) |
| 16:32.04 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 16:36.52 | *** join/#asterisk TimeRider (~steve@92.41.65.4.threembb.co.uk) |
| 16:37.35 | *** join/#asterisk VinceAntw (~VinceAntw@91.176.80.180) |
| 16:39.45 | VinceAntw | Hi everybody, can anyone assist me with a DAHDI question? |
| 16:40.03 | VinceAntw | is there any way to see if a DAHDI channel has congestion? |
| 16:40.30 | WIMPy | What does that mean? |
| 16:43.39 | VinceAntw | if I make a call with a phone connected to a dahdi analog card and the other party hangs up it should play the hangup sound (in my country it is 425,500 0,500) but instead it plays much shorter beeps (which I think is congestion) |
| 16:44.07 | *** join/#asterisk gusto (~gusto@nrbg-4dbe1fdb.pool.mediaWays.net) |
| 16:44.12 | gusto | so so |
| 16:44.17 | gusto | ready to test :-) |
| 16:51.16 | jaytee | one of my clients wants to be able to allow collect calls. Is this possible with VOIP using an ITSP like Flowroute? |
| 16:53.52 | p3nguin | Not to my knowledge. The next best thing is a call-back system. |
| 16:54.03 | p3nguin | Or just use a toll-free number. |
| 16:56.12 | p3nguin | Toll-free would be a lot cheaper than a collect call anyway. |
| 17:02.04 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v022-171.mobile.uci.edu) |
| 17:16.25 | *** join/#asterisk twodogs (~twodogs@telok.vitiate.me) |
| 17:30.45 | *** join/#asterisk sp00kz (~ilubj00@unaffiliated/sp00kz) |
| 17:32.42 | p3nguin | Okay, I think I figured out how to limit the length of time for an app or function. No thanks to anyone. |
| 17:32.48 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v022-171.mobile.uci.edu) |
| 17:33.55 | *** join/#asterisk CyBeRxIxO (~CyBeRxIxO@190.41.182.228) |
| 17:33.58 | jaytee | p3nguin, thanks. I'd started thinking about the toll-free number after I asked. Makes sense and should be cheaper. My client is a law firm and they get collect calls from the local jail but toll free should work for them. |
| 17:34.19 | CyBeRxIxO | Hi, any asterisknow support here? |
| 17:34.23 | p3nguin | I had a feeling it was jail related. |
| 17:35.01 | p3nguin | Many jails do collect calls OR calling cards (which use toll-free access numbers), so a toll-free number should work just fine. |
| 17:35.05 | CyBeRxIxO | excuse me, i got a case anyone up to help me with advices? |
| 17:36.46 | CyBeRxIxO | does iax2 works fine between elastix and asterisknow? |
| 17:37.47 | CyBeRxIxO | if it is does g729 digium paid codec works on them? |
| 17:38.31 | p3nguin | Both use Asterisk as back ends, so yes IAX2 works between them. |
| 17:38.51 | p3nguin | And yes g.729 codec works over IAX2. |
| 17:39.34 | *** join/#asterisk timahvo1 (~rogue@41.80.213.231) |
| 17:40.58 | *** join/#asterisk fobus912 (~fobus912@41.143.23.173) |
| 17:41.01 | fobus912 | Hi All |
| 17:41.26 | fobus912 | does anyone know about an Open Source STUN Server that support authentication ? |
| 17:43.31 | *** join/#asterisk joecool (~joecool@no-sources/joecool) |
| 17:57.09 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 17:59.43 | CyBeRxIxO | p3nguin thanks for ur answer |
| 18:02.42 | CyBeRxIxO | having 20 sip station, 2 pstn voip trunk lines and 1 iax2..... does it make sense on puttin g729 licence on the "pstn voip trunk"? |
| 18:07.02 | p3nguin | If you need to save bandwidth at the sacrifice of voice quality, sure. |
| 18:07.38 | *** join/#asterisk vfabi (~fabi@178.76.76.61) |
| 18:07.57 | *** join/#asterisk TheCompWiz (~TheCompWi@wsip-68-109-200-102.mc.at.cox.net) |
| 18:08.39 | TheCompWiz | anyone know how to get asterisk to not-care if (when using tls + sip) the common-name on the client certificate doesn't match the client's IP (or whatever else) |
| 18:11.42 | Hive | Would someone please enlighten me as to when a SIP channel is established and when it is destroyed? My server shows what feels like too many active SIP channels :| |
| 18:12.16 | TheCompWiz | afaik... you should see 1 channel per dialog. |
| 18:12.23 | *** join/#asterisk singler (~singler@beta.kirneh.eu) |
| 18:13.20 | Hive | So when a call is bridged such as the answer() command is issued to an incoming call |
| 18:13.35 | Hive | a sip channel is established, but when is it destroyed? |
| 18:13.40 | TheCompWiz | bridged between two sip peers? or dahdi -> sip? or ??? |
| 18:14.25 | *** join/#asterisk VultureZ (~Chuck@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 18:14.45 | TheCompWiz | basically a "channel" is made everytime the pbx needs to talk to the device over sip. Generally, while a call is active there is at least 1 open dialog for each involved peer. |
| 18:15.10 | VultureZ | TheCompWiz, is a channel established for registration purposes? |
| 18:15.10 | TheCompWiz | when the pbx starts talking to the phone... a channel is opened first... when it's done talking... the channel is closed. |
| 18:15.18 | TheCompWiz | it can momentairily |
| 18:15.23 | VultureZ | okay |
| 18:15.30 | VultureZ | I bet that is what Hive is questioning |
| 18:17.01 | TheCompWiz | for "registrations" you'll see something like xx.xxx.xxx.xxx (None) random-characters 0x0 (nothing) No Rx: REGISTER <guest> |
| 18:17.43 | Hive | Is there a command that I need to issue in the hangup context of calls to close out the channels? My server shows '926 active SIP dialogs' which seems like too many |
| 18:18.13 | *** join/#asterisk Tim_Toady (~fuzzy@130.43.51.201.dsl.dyn.forthnet.gr) |
| 18:20.04 | TheCompWiz | how many endpoints? |
| 18:22.33 | Hive | 45 |
| 18:22.46 | TheCompWiz | that is too many. |
| 18:22.49 | TheCompWiz | sip trunk? |
| 18:22.55 | pigpen | anybody got a Digium D70 phone out there? I got a question. |
| 18:23.17 | TheCompWiz | digium makes phones? |
| 18:23.23 | pigpen | yeah. |
| 18:23.31 | Hive | by end point you mean sip peers? |
| 18:23.32 | TheCompWiz | never seen one... interesting. |
| 18:23.33 | pigpen | Should be released to the general public mid April. |
| 18:23.34 | _Corey_ | pigpen: I have one here |
| 18:23.49 | TheCompWiz | Hive: attached devices that aren't trunks. |
| 18:24.14 | pigpen | _Corey_, using the side buttons, are they ready to show in use/idle yet? |
| 18:24.24 | _Corey_ | pigpen: Sure, that works |
| 18:24.40 | _Corey_ | The extended status stuff works nicely too... "Extended away" etc |
| 18:25.15 | Hive | TheCompWiz, Oh are you saying that the '926 active SIP dialogs' is too many? Or that 45 peers is too many? It's odd, my 'core show channels' shows an accurate number of open channels, but 'sip show channels' has that huge number |
| 18:25.40 | TheCompWiz | 926 active sip dialogs for 45 endpoints. |
| 18:25.40 | TheCompWiz | at most.... you should have 1-2 per endpoint. |
| 18:25.55 | pigpen | _Corey_, how do you add it? I am guessing you are not using the "contacts" app to add. |
| 18:26.02 | Hive | Yeah thats how I feel, was hoping to get some light shed on why that's happening |
| 18:26.52 | pigpen | _Corey_, I have added a few (SIP exten and a hint exten) with no luck. |
| 18:27.02 | pigpen | I took the quick route and just setup quickly with the web app. |
| 18:27.11 | pigpen | but will do the provisioning when I have more time. |
| 18:27.19 | _Corey_ | pigpen: I'm using it with the latest AsteriskNow ISO and the FreePBX digium module... is that what you're using? |
| 18:27.42 | TheCompWiz | Hive: got some sort of proxy or nat in the middle? |
| 18:28.10 | pigpen | _Corey_, no. I am using Asterisk 10.1.2 and a 1.6.1.12 box |
| 18:28.24 | pigpen | so my guess is you are provisioning it. |
| 18:28.42 | pigpen | probably with the DPMS app |
| 18:28.52 | _Corey_ | pigpen: Yeah, I'm using the DPM |
| 18:28.57 | Hive | No, direct connection, and sip.conf has bindaddr=0.0.0.0 |
| 18:29.07 | _Corey_ | pigpen: I can probably look at the config it's producing to see what you should have though |
| 18:29.15 | pigpen | yeah. I used a DPMS the other day, but it was an AR15 platform. ;-) |
| 18:29.16 | TheCompWiz | Hive: what kind of endpoints? |
| 18:29.47 | Hive | Polycom 50x series |
| 18:30.45 | TheCompWiz | Hive: I've got at least 50 501s on my network... and I'm not having that many sip channels... (I also have 550s and 33X phones) |
| 18:30.58 | TheCompWiz | no clue what to tell ya. |
| 18:31.09 | _Corey_ | pigpen: Are you just looking for the contacts xml file format? |
| 18:31.09 | TheCompWiz | Hive: what firmware versino? |
| 18:31.15 | TheCompWiz | *version? |
| 18:31.42 | pigpen | _Corey_, yeah, if I can add the "buddy watch" polycom equivilent. |
| 18:32.02 | *** join/#asterisk SaRSAeOL (~sarsaeol@66-113-78-49.rev.ibsinc.com) |
| 18:33.20 | _Corey_ | pigpen: It looks like it's documented here: http://docs.digium.com/phones/phones-module-for-asterisk-users-guide-beta.pdf |
| 18:33.31 | _Corey_ | Have you checked that out yet? |
| 18:33.55 | pigpen | Hive, I ran across something some time back. Having a PRI, if I didn't account for every DID, and if one was called that was not counted for, it would loop up a crap load of channels. Mind you, I have about 500 DID numbers. |
| 18:34.39 | pigpen | _Corey_, no, not yet, but thanks. I opened the box, 5 min of playing, 10 min of looking at DPMS, then got in here. I need to get some usage on this to make sure it is a product to move forward with. |
| 18:34.59 | pigpen | thanks bty...i am rushing it a bit. Got too much stuff to do today. |
| 18:35.04 | Hive | TheCompWiz, we are running the most recent firmware for those phones |
| 18:35.14 | p3nguin | I still keep getting a stuck channel periodically and nothing I can do short of restarting asterisk gets rid of it. Any ideas how to get rid of it? channel request hangup does not touch it, I've redirected it to extensions which run Hangup(), etc. I just want to destroy stuck channels easily without restarting. |
| 18:35.22 | Hive | pigpen, i think you might be onto something... |
| 18:35.38 | pigpen | yeah, it is not fun when that happens. |
| 18:35.45 | pigpen | just keeps looping the channels. |
| 18:35.55 | Hive | lol... any way to clear out some of these channels? a restart? |
| 18:36.11 | TheCompWiz | Hive: a restart would do it.... |
| 18:36.15 | pigpen | you can hang them up. but stop/start works better. |
| 18:36.26 | pigpen | in the 1.6 and prior, sometimes the channel would get locked. |
| 18:36.59 | TheCompWiz | long-term... it behaves like a memory leak... (each active channel consumes resources) .... |
| 18:37.13 | Hive | Hmm, ok definitely going to look into this |
| 18:37.29 | Hive | Thanks for your help guys :) |
| 18:38.13 | _Corey_ | pigpen: Aha... I've been using the D70 off and on since Christmas and probably full-time since February. Very few complaints so far. |
| 18:38.58 | TheCompWiz | Hive: I've got a box with ~120 users.... been running for months now without a reload and it's only got 4 active channels right now. |
| 18:39.10 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 18:39.14 | Hive | Thats what my core show channels looks like |
| 18:39.27 | Hive | I saw this 900 something from sip show channels and figured something was awry |
| 18:39.29 | TheCompWiz | I'm talking about 'sip show channels' |
| 18:40.31 | TheCompWiz | ... wish someone knew how to help me... /sigh |
| 18:40.57 | pigpen | _Corey_, yeah, I got to get my headset working. I have a GE Netcom 9350... |
| 18:41.33 | TheCompWiz | does the digium phone support EHS? |
| 18:41.38 | pigpen | yeah |
| 18:41.43 | TheCompWiz | shiney... |
| 18:41.48 | TheCompWiz | mgiht have to get one to play with. |
| 18:41.54 | pigpen | good luck. |
| 18:41.56 | Qwell | TheCompWiz: You should get 400 to play with. |
| 18:41.58 | TheCompWiz | lol |
| 18:42.15 | TheCompWiz | Qwell: pfft.... if I don't like 'em... who is gonna send me money back? |
| 18:42.23 | _Corey_ | I've been told the EHL support isn't quite available yet ;) |
| 18:42.24 | Qwell | TheCompWiz: _Corey_ will |
| 18:42.29 | TheCompWiz | WOOOOHOOO! |
| 18:43.13 | TheCompWiz | Qwell: you know you wanna help me with TLS fun.... |
| 18:43.14 | _Corey_ | minus a nominal (~100%) restocking fee, sure |
| 18:43.21 | TheCompWiz | :P |
| 18:43.52 | CyBeRxIxO | guys price per g729 codec? |
| 18:44.04 | TheCompWiz | same as girls price per g729 codec. |
| 18:44.11 | CyBeRxIxO | :o free? |
| 18:44.24 | *** join/#asterisk AlfE_ (~quassel@83-215-36-12.bruck.dyn.salzburg-online.at) |
| 18:44.27 | p3nguin | Was that your quoted price? |
| 18:44.29 | TheCompWiz | nope. |
| 18:44.52 | CyBeRxIxO | i need to get 3 g729 licence |
| 18:45.02 | CyBeRxIxO | what i have to do? |
| 18:45.04 | p3nguin | You probably just need to get one license. |
| 18:45.22 | CyBeRxIxO | i have 2 psnt/voip trunk and 1 iax2 trunk |
| 18:45.23 | CyBeRxIxO | need 3? |
| 18:45.27 | TheCompWiz | CyBeRxIxO: the process consists of... contacting digium... asking them for a quote... and then throwing money at them. |
| 18:45.30 | p3nguin | Just one license. |
| 18:45.34 | TheCompWiz | (especially as they OWN g729.) |
| 18:45.41 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 18:45.49 | Qwell | no, no, no, no, no, we most definitely do not "own" it. |
| 18:45.50 | _Corey_ | lol, they don't own g729 |
| 18:45.54 | p3nguin | It's really not even that difficult. |
| 18:46.06 | CyBeRxIxO | just one licence? what happeb if there are calls on the same time |
| 18:46.12 | p3nguin | Just go to the web site and buy one license for as many channels as you need. |
| 18:46.18 | _Corey_ | some guy sitting on a bunch of patents in a cushy evil-looking chair somewhere "owns" g.729 |
| 18:46.57 | CyBeRxIxO | what you mean p3nguin |
| 18:47.05 | p3nguin | I mean exactly what I said. |
| 18:47.14 | CyBeRxIxO | i need the licence and support to install it |
| 18:47.28 | p3nguin | Go to digium.com. Buy a license for as many channels as you intend to support. |
| 18:47.50 | CyBeRxIxO | you mean 1 licence for 3 channels? |
| 18:47.50 | p3nguin | I'll do it for you, but you'll pay my regular hourly rate. |
| 18:47.52 | TheCompWiz | I thought digium owned the patents/copyrights for g729... /shrug |
| 18:48.15 | p3nguin | If you need three channels, go buy a license for three channels. |
| 18:48.18 | *** join/#asterisk Russ (~russ@209-147-130-219.nat.asu.edu) |
| 18:48.24 | _Corey_ | TheCompWiz: If I remember correctly, these guys actually hold the patent: http://www.sipro.com/ |
| 18:48.45 | Qwell | _Corey_: lots of companies own patents |
| 18:48.50 | CyBeRxIxO | i currently have elastix last version |
| 18:48.54 | Qwell | there's just one that collectively sells licenses to use them |
| 18:48.54 | CyBeRxIxO | do i need asterisknow? |
| 18:52.58 | _Corey_ | Qwell: Yeah, I don't know who Digium pays for what but I've heard g.729 licensing isn't too crazy |
| 18:53.32 | _Corey_ | Now, "visual voicemail" on the other hand... ;) |
| 18:55.45 | CyBeRxIxO | "The Digium G.729 codec and Digium product registration tools are supported on Linux x86 and x86_64 environments only." |
| 18:55.55 | CyBeRxIxO | means not on i386? |
| 18:56.03 | p3nguin | i386 IS x86 |
| 18:56.14 | TheCompWiz | _Corey_: I just use the google voice setup. Free... it's visual... and they even give the option to email/text a transcription of the voicemail. |
| 18:56.38 | CyBeRxIxO | p3nguin im about to buy the licence |
| 18:56.48 | CyBeRxIxO | but i'd need some help |
| 18:57.02 | p3nguin | x86 encompasses, but is not necessarily limited to, 386, 486, 586, and 686. |
| 18:57.09 | gusto | thats cool |
| 18:57.48 | gusto | so i tried another voip provider and i get the same error ... but now i know that it is sipcode 603 what means the same as with that be-converged |
| 18:57.57 | p3nguin | What help is it that you need to buy the license? |
| 18:58.12 | CyBeRxIxO | u told me i need 1 licence of 3 channels |
| 18:58.19 | p3nguin | yes....... |
| 18:58.19 | CyBeRxIxO | my system is: |
| 18:58.21 | _Corey_ | TheCompWiz: The patent troll who claims to have a "ownership" of "visual voicemail" is currently trying to shake down dozens of companies (mostly with iPhone apps) |
| 18:58.35 | CyBeRxIxO | 20 sip client in lan, 2 trunk pstn/voip and 1 iax2 |
| 18:58.43 | CyBeRxIxO | u sure i need 3 channels? |
| 18:58.47 | Gugge | CyBeRxIxO: you need a license for the concurrent number of channels that needs to decode/encode g729 |
| 18:58.50 | p3nguin | License as many channels as you want to use. |
| 18:58.56 | TheCompWiz | _Corey_: lol... too funny. I hate patent trolls. |
| 18:59.18 | p3nguin | If you want to transcode 5 channels using g.729 at the same time, buy 5 channels. |
| 18:59.27 | _Corey_ | TheCompWiz: yeah, don't get me started... One of my customers actually pulled the feature from their iPhone app |
| 18:59.37 | p3nguin | If you want to transcode only 3, buy 3. |
| 19:00.34 | CyBeRxIxO | if my configuration says "disallow=all allow=g729" but all are channels are being used? what happens |
| 19:00.43 | CyBeRxIxO | on the next call |
| 19:00.57 | p3nguin | It depends on if you will need to transcode or not. |
| 19:01.15 | p3nguin | You can use g729 end to end, without transcoding, without the license. |
| 19:01.36 | p3nguin | If you transcode, you'll use a channel out of your licensed decoders/encoders. |
| 19:01.44 | [TK]D-Fender | CyBeRxIxO, Your 4th call will drop like a rock |
| 19:01.50 | CyBeRxIxO | XD |
| 19:02.03 | CyBeRxIxO | cant tell that the the rich boss |
| 19:02.21 | [TK]D-Fender | CyBeRxIxO, If he's rich, have him pay for the number of channels you need to support |
| 19:02.23 | p3nguin | If he is so rich, buy enough channels so it isn't a problem. |
| 19:02.32 | [TK]D-Fender | Acutally.. if you want your solution to work... same thing |
| 19:02.44 | CyBeRxIxO | ok ok got it |
| 19:02.46 | CyBeRxIxO | but... |
| 19:03.03 | CyBeRxIxO | what happen if i lost the comp |
| 19:03.10 | CyBeRxIxO | and install a new one |
| 19:03.16 | CyBeRxIxO | my licence lost? |
| 19:03.29 | [TK]D-Fender | CyBeRxIxO, there is a limited transfer capability |
| 19:03.30 | p3nguin | If you call between two phones and do not have to transcode, you will not use up one of your encoders/decoders. |
| 19:04.26 | CyBeRxIxO | when asterisk mess up and need to reinstall it |
| 19:04.35 | CyBeRxIxO | how to put the licence again? |
| 19:04.56 | CyBeRxIxO | i mean, is that posible? or i need to buy a new one |
| 19:06.48 | *** part/#asterisk gonewage (~gonewage@72.2.130.205) |
| 19:09.10 | CyBeRxIxO | when i recive call on psnt/voip trunk g729 apply automatic? or depend of configuration of spa3102? |
| 19:09.50 | p3nguin | Everything is dependent on configuration. |
| 19:10.04 | p3nguin | ~magic |
| 19:10.05 | infobot | Forms based RAD language. URL: http://www.magic-sw.com |
| 19:10.22 | CyBeRxIxO | spa3102 show "g729a" is that the codec im buying? |
| 19:10.32 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:10.56 | p3nguin | As I've said several times already, you only need a license to transcode. |
| 19:12.53 | *** join/#asterisk drudge` (~drudge@unaffiliated/drudge/x-837452) |
| 19:12.55 | *** part/#asterisk drudge` (~drudge@unaffiliated/drudge/x-837452) |
| 19:19.23 | *** join/#asterisk paolosupino (~paolo-sup@net-2-38-113-188.cust.dsl.vodafone.it) |
| 19:20.13 | paolosupino | can anyone point me to a URL that as a clear how to on setting up a confbridge in asterisk 1.8? |
| 19:20.35 | p3nguin | core show application ConfBridge |
| 19:24.28 | *** join/#asterisk Russ (~russ@209-147-130-219.nat.asu.edu) |
| 19:24.45 | CyBeRxIxO | i asume then once i put the g729 paid licence to configure all devices on "disallow=all allow=g729" |
| 19:25.11 | CyBeRxIxO | am i right? |
| 19:25.34 | p3nguin | No. |
| 19:25.48 | p3nguin | You only set that on peers that you want to always use g729. |
| 19:28.05 | *** part/#asterisk VinceAntw (~VinceAntw@91.176.80.180) |
| 19:30.21 | CyBeRxIxO | i want all to use always g729 |
| 19:30.42 | p3nguin | Why? Why would anyone want to do that for phones on a LAN attached to Asterisk? |
| 19:31.41 | p3nguin | You said yourself that you have 20 phones on a LAN attached to Asterisk. |
| 19:32.11 | p3nguin | You should use the cheapest (in terms of processing) codec possible for those. |
| 19:32.41 | p3nguin | cheapest and best quality |
| 19:32.54 | p3nguin | That will probably be ulaw or alaw. |
| 19:34.08 | p3nguin | Or, if you're inclined to try it, testlaw. |
| 19:34.11 | _Corey_ | g722 sounds pretty nice ;) |
| 19:34.37 | *** join/#asterisk WindBack (~quassel@190.220.135.165) |
| 19:35.31 | p3nguin | But it is more expensive, and he probably doesn't have phones that even support it. |
| 19:36.07 | gusto | how do i assign a register to a diaplan? |
| 19:36.13 | p3nguin | You don't. |
| 19:36.19 | gusto | because when i call my number it always goes to default |
| 19:36.22 | _Corey_ | p3nguin: g722 is free if you have a device that supports it |
| 19:36.26 | p3nguin | Register statements go in sip.conf. |
| 19:36.33 | gusto | [Apr 4 19:35:31] NOTICE[3558]: chan_sip.c:22081 handle_request_invite: Call from '' (62.52.148.87:5060) to extension '4991131042466' rejected because extension not found in context 'default'. |
| 19:36.36 | p3nguin | "cheapest (in terms of processing)" |
| 19:36.47 | _Corey_ | ah, gotcha |
| 19:36.53 | CyBeRxIxO | thanks p3nguin |
| 19:36.58 | CyBeRxIxO | got it |
| 19:37.01 | WindBack | Hello.. I don't find any documentation which explains me the functionality of "sip show mwi" cli command. Can anybody explain me it |
| 19:37.14 | _Corey_ | p3nguin: I haven't been paying attention :) |
| 19:37.18 | p3nguin | It's okay. |
| 19:37.35 | p3nguin | I understand how IRC works. |
| 19:38.27 | rrittgarn | Say the asterisk process just dies... where are the logs for that crash? /var/log/asterisk/messages ? |
| 19:39.05 | p3nguin | If it dies, it can't write logs. |
| 19:39.27 | p3nguin | If it logged anything prior to dying, I'd expect to see it in the full log. |
| 19:40.18 | CyBeRxIxO | so to use the g729 codec all i need is: "on pstn/voip trunks disallow=all allow=g729" and on spa3102 "preferred codec=g729a (wich is the only g729 named" and everything will work fine, am i right? |
| 19:40.53 | p3nguin | Do you want to always use g729 between your asterisk and the ITSP? |
| 19:41.03 | [TK]D-Fender | CyBeRxIxO, You don't need to set a preference on the SPA. If *'s peer sys G729, then that is the end of it |
| 19:41.16 | CyBeRxIxO | yes im currently having cuts on calls |
| 19:41.47 | CyBeRxIxO | qualify is bad |
| 19:41.55 | CyBeRxIxO | very bad |
| 19:42.04 | [TK]D-Fender | And that may have nothing to do with codec |
| 19:42.17 | p3nguin | probably doesn't |
| 19:42.18 | [TK]D-Fender | Jitter = cuts |
| 19:42.42 | CyBeRxIxO | during the call there are moments that cant hear nothing |
| 19:43.14 | p3nguin | If you haven't checked your available and used bandwidth, why are you so sure changing to g729 is going to help? |
| 19:43.28 | rrittgarn | p3nguin, if i have more than 3 callers in conf bridge it ends the asterisk process... nifty bug.. Asterisk SVN-branch-10-r360139 |
| 19:43.29 | p3nguin | g729 is used to reduce the used bandwidth of calls over a link. |
| 19:43.59 | CyBeRxIxO | i have 2mb internet for more than 20 users |
| 19:44.07 | CyBeRxIxO | i think that is my calls problem |
| 19:44.41 | CyBeRxIxO | sometimes call are normal, sometimes dont |
| 19:45.09 | CyBeRxIxO | iax2 is better than pstn but still qualify is real bad |
| 19:45.31 | p3nguin | You can use trunking with iax2 to reduce the bandwidth of 20 calls. |
| 19:46.25 | p3nguin | How many concurrent calls to the PSTN are you having at peak? |
| 19:46.27 | CyBeRxIxO | what you mean on "use trunking" |
| 19:46.33 | CyBeRxIxO | 2 or 3 |
| 19:46.37 | p3nguin | IAX2 supports trunking. |
| 19:46.51 | p3nguin | trunk=yes |
| 19:46.58 | p3nguin | SIP does not do trunking. |
| 19:47.04 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 19:47.27 | CyBeRxIxO | iax2 is trunking between 2 elastix on diferent places |
| 19:47.33 | CyBeRxIxO | that is done |
| 19:47.39 | p3nguin | Did you specify trunk=yes? |
| 19:47.43 | p3nguin | on both peers? |
| 19:49.17 | CyBeRxIxO | just checked, yes that is the config |
| 19:49.29 | CyBeRxIxO | trunk=yes on both peers |
| 19:49.47 | p3nguin | Okay, if trunk=yes on both sides, trunking is enabled and that will help save bandwidth of multiple calls. |
| 19:49.53 | CyBeRxIxO | type=friend |
| 19:49.54 | CyBeRxIxO | trunk=yes |
| 19:49.54 | CyBeRxIxO | qualify=1000 |
| 19:49.54 | CyBeRxIxO | disallow=all |
| 19:49.54 | CyBeRxIxO | allow=ulaw |
| 19:50.03 | p3nguin | But for only two or three calls, your savings will not be that great. |
| 19:50.10 | CyBeRxIxO | is that enough on a good iax2 trunk? |
| 19:50.39 | p3nguin | That's all there is for trunking. Set trunk=yes on both sides and trunking is enabled. |
| 19:51.18 | CyBeRxIxO | why still qualify of calls on iax2 is still bad |
| 19:51.36 | CyBeRxIxO | how can i make it better? g729 licence? |
| 19:52.19 | gusto | what i do not understand is that when i put in peer config context=<context> it still does not look for context, but it looks still to 'default' |
| 19:52.45 | gusto | so i absolutely do not understand that error message |
| 19:52.55 | p3nguin | Two ulaw calls should only use about about 160 kb/s without trunking. |
| 19:53.04 | gusto | <PROTECTED> |
| 19:53.15 | [TK]D-Fender | CyBeRxIxO, If you have jitter issues, FIX YOUR LINK |
| 19:53.36 | p3nguin | Either your call is not matching the peer you configured, or you aren't correctly setting the context value. |
| 19:53.47 | [TK]D-Fender | gusto, Because it clearly isn't matching your peer. Something you'd see if you paid attention to the SIP DEBUG of the call attempt |
| 19:53.56 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 19:54.42 | gusto | [TK]D-Fender: so what do i do about it? |
| 19:54.56 | [TK]D-Fender | gusto, Look at the actual call and fix your peer |
| 19:55.23 | gusto | [TK]D-Fender: what peer? the one where the call comes from? |
| 19:55.31 | [TK]D-Fender | yes |
| 19:55.59 | gusto | [TK]D-Fender: where do i fix it? in reigistry? |
| 19:56.05 | gusto | register => |
| 19:56.07 | [TK]D-Fender | sip.conf <- |
| 19:56.16 | gusto | of course, but where exactly |
| 19:56.20 | [TK]D-Fender | YOUR PEER |
| 19:56.35 | gusto | so inside the peer config |
| 20:03.49 | gusto | Looking for 655161 in default (domain 77.190.31.219:5060) |
| 20:04.44 | gusto | it ssems like my asterisk server is rejecting the call because he does not find 655161 |
| 20:04.47 | gusto | well |
| 20:05.14 | gusto | should i say that [myprovider] is in context=655161 ? |
| 20:05.20 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:05.20 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:06.56 | [TK]D-Fender | gusto, No, it is looking in [default] and not in the context you specified in your peer because it is not identified as COMING from your peer. |
| 20:08.20 | gusto | [TK]D-Fender: so i go to [default] in extensions.conf and put there what? like exten => 655161,1,Dial(myphone) ? |
| 20:08.44 | [TK]D-Fender | gusto, No, you go fix your peer and make sure that you have a match in the context it points to. |
| 20:09.50 | gusto | [TK]D-Fender: so register => .../EXTENSION must be the same as [PEER] context=EXTENSION ? |
| 20:10.19 | *** join/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 20:10.35 | asteriskmonkey | as meetme been dropped in 1.10? |
| 20:11.50 | [TK]D-Fender | asteriskmonkey, No |
| 20:11.58 | [TK]D-Fender | gusto, No, that is not your PEER |
| 20:12.22 | [TK]D-Fender | gusto, and no, context != extension. |
| 20:13.04 | *** part/#asterisk kl4m (~kl4m@gw2.noc1.sys-tech.net) |
| 20:13.21 | asteriskmonkey | I buillt asterisk 10.3.1 on freebsd and it seems to be missing meetme |
| 20:13.31 | [TK]D-Fender | gusto, A register statement is also not a peer. That is nowhere I told you to look for these past half a dozen times I've been telling you the same thing. |
| 20:13.34 | gusto | ok, so he is looking for 655161 in default ... how do i make him look else? or is it OK, that he looks in default? |
| 20:13.47 | [TK]D-Fender | asteriskmonkey, installed from where? |
| 20:13.56 | [TK]D-Fender | gusto, Fix your peer |
| 20:14.13 | asteriskmonkey | ports |
| 20:14.19 | gusto | [TK]D-Fender: but i do not know what should be wrong with my peer, my peer works |
| 20:14.22 | asteriskmonkey | /usr/ports/net/asterisk10 |
| 20:14.29 | [TK]D-Fender | gusto, No, it clearly doesn't |
| 20:14.46 | [TK]D-Fender | gusto, otherwise it'd be looking where the context in there points to |
| 20:15.11 | [TK]D-Fender | asteriskmonkey, And do you see it in the source in tehre? |
| 20:15.20 | gusto | [TK]D-Fender: i have some code for you, w8 |
| 20:15.27 | [TK]D-Fender | asteriskmonkey, Because that is "ports", that isn't "Asterisk official packaging" |
| 20:15.46 | gusto | [TK]D-Fender: http://www.personal-voip.de/index.php?page=wiki&wikiid=support:konfigurationen:asterisk |
| 20:16.22 | asteriskmonkey | have to check all the build files ill check there thanks for th yes/no about it being still there |
| 20:16.56 | [TK]D-Fender | gusto, Means nothing to me. You aren't looking at your call, I do't see your actual configs and you seem to be exhibiting a very severe learning disability. the term "fix your peer" simply does not seem able to sink in. |
| 20:18.01 | asteriskmonkey | is confbridge better than meetme? |
| 20:18.23 | [TK]D-Fender | asteriskmonkey, Yeah, it hasn't gone anywhere yet. ConfBridge hasn't quite caught up in functionality to the point of dropping MeetMe yet. Perhaps very soon as I've heard there has been a lot of progress |
| 20:19.47 | gusto | [TK]D-Fender: http://pastebin.com/riRLv8WH |
| 20:21.23 | gusto | [TK]D-Fender: i can not see what is wrong with my peer here |
| 20:22.26 | [TK]D-Fender | gusto, I don't see you looking at the call with SIP debug enabled. |
| 20:23.12 | [TK]D-Fender | gusto, And I can see that you did not even DEFINE what context calls from your peer should go to. |
| 20:23.33 | p3nguin | If the call doesn't come from 46.182.250.50, it will not match the peer. |
| 20:24.14 | p3nguin | It could be matching, but without setting the context, it will go to the context set in general. |
| 20:24.26 | gusto | [TK]D-Fender: http://pastebin.com/TC21Q5hW |
| 20:25.03 | gusto | p3nguin: you may be right |
| 20:25.10 | p3nguin | Of course. |
| 20:25.12 | gusto | p3nguin: that seems to be the problem |
| 20:25.26 | p3nguin | Found peer 'personalvoip' for '015223817325' from 46.182.250.50:5060 |
| 20:25.29 | p3nguin | The peer was matched. |
| 20:25.38 | gusto | ah |
| 20:25.38 | [TK]D-Fender | He never specified a context |
| 20:25.39 | p3nguin | You should have set the context. |
| 20:26.00 | gusto | but that does not happen with be-converged, but that is another issue, OK |
| 20:26.11 | [TK]D-Fender | gusto, And we have no idea how you configered tthat other one |
| 20:26.13 | gusto | because sometimes the requests come from ip addresses i never seen before |
| 20:26.19 | p3nguin | The same applies to all peers. You set a context PER PEER or it uses the default context set in general. |
| 20:26.24 | [TK]D-Fender | gusto, And does not matter as this one clearly does not have the context specified |
| 20:26.42 | [TK]D-Fender | Looking for 655161 in default (domain 77.190.31.219:5060) <--- so it fell back to [default] which is somethin you should never allow like that. |
| 20:26.48 | gusto | [TK]D-Fender: yes, yes, we are talking about this first and that second we can talk about when this works, or maybe then the other will work again, when i find the problem |
| 20:27.06 | [TK]D-Fender | We found the problem. |
| 20:27.14 | gusto | so so |
| 20:27.44 | gusto | but you say that context specification does not solve the problem |
| 20:27.52 | [TK]D-Fender | So fix your peer. Set the context that you failed to set (which the link you gave us SHOWED THEM DOING |
| 20:28.33 | gusto | i already tried that with context set before, and it did not help, i ll do it again and i ll paste the debug |
| 20:29.40 | gusto | well, this time we got it |
| 20:29.44 | p3nguin | I've never seen it take someone three months to configure a couple peers and a few extensions. |
| 20:29.47 | gusto | hmm ... ok |
| 20:29.53 | gusto | :-D |
| 20:29.57 | gusto | me neither |
| 20:30.05 | [TK]D-Fender | Couple? This is one. Not "couple" or "few" |
| 20:30.11 | gusto | but i am still making progress |
| 20:30.22 | p3nguin | He's been working on two peers. |
| 20:30.30 | p3nguin | telnr and personalvoip |
| 20:30.39 | [TK]D-Fender | \o/ |
| 20:30.46 | [TK]D-Fender | Ok, checkout time here, BBIAB |
| 20:30.49 | p3nguin | THREE MONTHS |
| 20:31.08 | p3nguin | I would have fired you after the third DAY. |
| 20:31.11 | gusto | but i was not trying three months in line |
| 20:31.22 | gusto | yes, i am trying the third day :-D |
| 20:31.27 | gusto | but over three months |
| 20:31.28 | gusto | :-D |
| 20:33.36 | pabelanger | ~itsp |
| 20:33.36 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 20:33.43 | pabelanger | ~itsp-us |
| 20:34.41 | pabelanger | ~itsplist-us |
| 20:34.41 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 20:35.29 | pabelanger | any feedback about flowroute? good / bad? |
| 20:35.43 | p3nguin | good |
| 20:37.55 | gusto | p3nguin: please take a look at this one http://pastebin.com/08GBNWLY |
| 20:38.03 | *** join/#asterisk danlench (~daniell@pool-71-113-225-119.herntx.dsl-w.verizon.net) |
| 20:38.09 | gusto | p3nguin: that is more important than that other shit |
| 20:38.23 | gusto | and context is specified, but falls into default |
| 20:38.30 | p3nguin | No matching peer for '015223817325' from '62.52.148.39:5060' |
| 20:38.33 | gusto | thats what i was talking about |
| 20:38.37 | gusto | yes |
| 20:38.37 | p3nguin | Has nothing to do with context. |
| 20:38.41 | gusto | so |
| 20:38.45 | gusto | here we are |
| 20:38.45 | p3nguin | No matching peer = default context. |
| 20:38.49 | gusto | ok |
| 20:39.09 | gusto | is not that big of an idiot |
| 20:39.30 | gusto | p3nguin: so what can be done about that? |
| 20:39.35 | p3nguin | What is the host you set for that peer? |
| 20:39.39 | p3nguin | host= what? |
| 20:39.53 | gusto | sipproxy.endesha.be-converged.com |
| 20:40.25 | p3nguin | There's no DNS for that host. |
| 20:40.27 | gusto | pcscf-brln-de01.endesha.be-converged.com. 21 IN A 62.52.148.39 |
| 20:40.30 | gusto | it is |
| 20:40.34 | gusto | SRV record |
| 20:40.49 | gusto | dig _sip._udp.sipproxy.endesha.be-converged.com srv |
| 20:40.49 | p3nguin | okay |
| 20:41.10 | p3nguin | Okay, they have four hosts. |
| 20:41.19 | gusto | and that is what i am talking about the whole day, that i have this 4 ip addresses and i can not know which one hits me |
| 20:41.28 | p3nguin | Pastebin your peer. |
| 20:41.31 | p3nguin | I'll show you how. |
| 20:41.35 | gusto | ??? |
| 20:41.39 | gusto | what peer now? |
| 20:41.47 | p3nguin | Pastebin the peer for be-converged. |
| 20:41.55 | gusto | you mean the sip.conf [be-converged]? |
| 20:41.58 | p3nguin | exactly |
| 20:42.04 | gusto | no, problem, w8 |
| 20:42.36 | p3nguin | again with the "weight" |
| 20:43.07 | gusto | p3nguin: http://pastebin.com/eGr58EGx |
| 20:44.21 | gusto | maybe i will have to adjust the other VoIP provider as well, because that time it was "just luck" that he matched the peer |
| 20:45.17 | gusto | i could swear that he did not do it before ... because personalvoip has also more ip addresses but resloves only to one, but they have some more, they have a list of raw ipv4 addr's on website |
| 20:46.05 | asteriskmonkey | [TK]D-Fender : http://pastebin.com/DE4KByKz |
| 20:46.19 | asteriskmonkey | should do i just enable that and take our the replace line? |
| 20:46.26 | gusto | he is not here any more |
| 20:46.32 | p3nguin | http://pastebin.com/GGvRsQx1 |
| 20:47.06 | danlench | Morning, I have an existing pbx vodavi punchblock analog install with 21 internal lines and 4 external. New to all this and have been trying to research it for weeks. We want to maybe go voip internal and keep our 4 PSTN lines. either way we want to go asterik. |
| 20:47.30 | gusto | p3nguin: aaah, i ve seen something like that already somewhere in documentation/forum, but for another use |
| 20:47.46 | p3nguin | That will allow all of the hosts to match the peer. |
| 20:47.54 | gusto | of course, i did not get that idea to use it for it |
| 20:48.02 | gusto | p3nguin: yes, i understand |
| 20:48.15 | p3nguin | Give it a try. |
| 20:48.37 | danlench | my question is, how to get the PSTN into the computer and then can i use a normal NIC to push it out over the network |
| 20:49.00 | gusto | so i do not have to make a "peer" config for every one i use just [peer](take over properties) |
| 20:49.01 | asteriskmonkey | danlench you need an anaolg card |
| 20:49.06 | SaRSAeOL | danlench: digium makes pci cards that have fox ports |
| 20:49.11 | SaRSAeOL | fxo* |
| 20:49.14 | asteriskmonkey | or just use sip from a provider |
| 20:49.45 | p3nguin | danlench: You can use an FXO card in the computer, or you can use a SIP gateway device like an SPA-8000. |
| 20:50.21 | p3nguin | Err... is the 8000 FXS only? |
| 20:50.33 | gusto | p3nguin: wow, i have now a lot of peers :-D |
| 20:51.04 | p3nguin | If the SPA-8000 is FXS only, then "like" was the keyword in my sentence. |
| 20:51.20 | gusto | well, it works ... now it matches the peer |
| 20:51.36 | gusto | and throws A LOT OF MESSAGES |
| 20:51.43 | gusto | for every peer conf |
| 20:51.57 | danlench | p3nguin: its starting to sink in, slowly. really new to pbx though, ugg |
| 20:52.04 | gusto | ok, what do we do now? |
| 20:52.21 | gusto | extensions.conf |
| 20:52.26 | danlench | and our current system is dying |
| 20:52.31 | p3nguin | danlench: There are devices which have 4 or 8 FXO ports. Apparently the SPA-8000 isn't one like I thought it was. |
| 20:52.48 | gusto | FXO ports are ports for analogue telephones? |
| 20:52.52 | SaRSAeOL | yes |
| 20:52.55 | SaRSAeOL | outside lines |
| 20:52.58 | p3nguin | gusto: Once you match peers, then you create extensions in the context set in the peers. |
| 20:53.05 | SaRSAeOL | there are fxo and fxs |
| 20:53.13 | gusto | p3nguin: yes, that will be the "easier" part now |
| 20:53.14 | gusto | :-D |
| 20:53.30 | danlench | ok right. FXO connected to the PSTN and then the NIC to the internal switch right (basically) |
| 20:53.37 | p3nguin | yes |
| 20:53.52 | danlench | p3nguin: thanks, this helps alot. |
| 20:54.03 | danlench | too many acronyms |
| 20:54.04 | p3nguin | If you have a gateway device, it will have the analog ports and speak SIP over Ethernet. |
| 20:54.21 | p3nguin | Otherwise, the card in the Asterisk box will take care of all the work. |
| 20:54.48 | danlench | Session Initiation Protocol? |
| 20:54.52 | p3nguin | ~sip |
| 20:54.52 | infobot | sip is, like, Session Initiation Protocol, http://www.cs.columbia.edu/sip/ (see RFC 3261) It's HIP to be SIP! |
| 20:55.04 | danlench | k |
| 20:55.11 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:55.28 | p3nguin | We use SIP in Asterisk more than other VoIP protocols. |
| 20:55.39 | *** join/#asterisk Defraz (~Defraz@69.20.176.132) |
| 20:55.58 | danlench | then the gateway box will go into the switch and the asterix box will talk to it |
| 20:56.06 | p3nguin | But Asterisk is capable of other channel technologies if they are required. |
| 20:56.12 | p3nguin | Yes, exactly. |
| 20:56.47 | p3nguin | You'd configure a peer (or several peers) in asterisk for the gateway to talk with asterisk. |
| 20:57.01 | danlench | so its just like an internet gateway but for telephony |
| 20:57.06 | p3nguin | Yes. |
| 20:57.23 | danlench | and get new cool phones ;) |
| 20:58.04 | p3nguin | Then any calls from IP phones would go through asterisk, and if the calls are supposed to go to the PSTN, Asterisk will send the call via the analog gateway device. |
| 20:58.22 | p3nguin | If configured to do so, of course. |
| 20:58.58 | gusto | p3nguin: so i say exten => 655161,1,Goto(SIP/myphone)? |
| 20:58.59 | danlench | ok, all internal calls go through asterik and external (PSTN) to gateway |
| 20:59.16 | p3nguin | Check pricing on a card with four FXO modules and then compare it to prices of gateways with FXO ports. |
| 20:59.29 | danlench | ok, thx p3nguin |
| 20:59.39 | danlench | see y'all soon |
| 20:59.47 | p3nguin | gusto: Dial(), not Goto(). |
| 20:59.53 | *** part/#asterisk danlench (~daniell@pool-71-113-225-119.herntx.dsl-w.verizon.net) |
| 21:00.12 | p3nguin | gusto: If you have a peer in sip.conf named "myphone", then extension 655161 would Dial() the peer (the phone). |
| 21:00.27 | gusto | yes yes |
| 21:00.30 | gusto | i ll try it first |
| 21:02.03 | gusto | [Apr 4 21:01:34] WARNING[3740]: app_dial.c:2198 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 21:02.12 | gusto | for all providers |
| 21:02.15 | rrittgarn | whats the best way to see what is causing a crash/quit? I've got a confbridge set up, and whenever I get more than three calls it looks like it gracefully quits... mid call... only when the third user hits # |
| 21:02.35 | p3nguin | What do you mean by "for all providers" ? |
| 21:03.18 | *** join/#asterisk dijib (~root@bas10-kitchener06-1176138838.dsl.bell.ca) |
| 21:03.37 | dijib | anybody know how to disable MWI? |
| 21:04.09 | gusto | p3nguin: i tried both |
| 21:04.18 | leifmadsen | dijib: don't have a mailbox= setting for your peer |
| 21:04.24 | p3nguin | gusto: Show me. |
| 21:04.31 | gusto | p3nguin: i get the same error message, i am now looking at the debug output |
| 21:04.35 | p3nguin | Show me how you are doing it. |
| 21:04.43 | p3nguin | Show me the extension. |
| 21:04.58 | p3nguin | Show me something instead of just saying it doesn't work. |
| 21:05.23 | fprior | Hi all, here again with the case of the spa400 and * 1.8 . There are news on http://pastebin.com/fVC9JtV2 , [TK]D-Fender |
| 21:05.30 | gusto | p3nguin: http://pastebin.com/LV3GrVYz |
| 21:05.57 | p3nguin | I don't see the verbose output in there. |
| 21:06.18 | p3nguin | When you show me something like that, you should have core set verbose 3 in addition to sip set debug on. |
| 21:07.25 | paolosupino | can I put the language parameter in sip.conf in specific labels or do I have to put it in the [general] section? |
| 21:07.26 | p3nguin | By that, all I know is that the call is trying to find extension 655161 in context home. |
| 21:07.54 | p3nguin | You didn't show me what extension 655161 is doing. |
| 21:08.21 | gusto | p3nguin: http://pastebin.com/UWNiQF6B |
| 21:08.33 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 21:08.51 | p3nguin | SIP/linksysata is offline. |
| 21:09.09 | p3nguin | sip show peers |
| 21:09.11 | gusto | it is not |
| 21:09.15 | p3nguin | You'll see it isn't there. |
| 21:09.16 | gusto | but maybe it is not registred |
| 21:09.21 | p3nguin | Exactly. |
| 21:09.22 | gusto | i am restarting it, w8 |
| 21:09.27 | p3nguin | Not register = not online. |
| 21:09.35 | p3nguin | (as far as asterisk is concerned) |
| 21:09.42 | gusto | yes yes |
| 21:09.44 | gusto | <PROTECTED> |
| 21:10.04 | p3nguin | If you don't see an IP address for it in sip show peers, it isn't online. |
| 21:10.14 | gusto | well |
| 21:10.17 | gusto | now it got well |
| 21:10.21 | p3nguin | Even if you do see an IP address, there is still a chance that it has gone away. |
| 21:10.52 | gusto | yes yes, it sometimes does not re-register |
| 21:11.23 | gusto | it is because i am often trying new configs now and when i am reloading sip, then it takes some time till that linksysata adapter realizes that |
| 21:11.30 | gusto | and maybe he even doesnt |
| 21:11.32 | gusto | however |
| 21:11.38 | gusto | i ll try the other one |
| 21:12.26 | gusto | what i do not understand is why he is going with that same extension on both |
| 21:12.49 | gusto | i mean it works vor personalvoip and endesha-be-converged with only one extension |
| 21:13.00 | gusto | 655161 |
| 21:13.04 | p3nguin | Set the register timeout to a lower value, like 60. |
| 21:13.06 | p3nguin | instead of 3600. |
| 21:13.07 | gusto | interesting, isnt it? |
| 21:13.18 | gusto | well, that is not that big of a problem |
| 21:13.55 | p3nguin | You can call extension 655161 via both providers? |
| 21:14.09 | p3nguin | And extension 655161 always Dial()s SIP/linksysata ? |
| 21:14.16 | gusto | yes |
| 21:14.29 | gusto | and outside calling still does not work |
| 21:14.44 | gusto | so what we managed to get to work is only when someone calls me |
| 21:14.49 | p3nguin | I would think you'll get the same result with both providers if the end point is the same. If it is offline, it's gone and cannot be reached. |
| 21:15.04 | p3nguin | Fix your extensions for outbound dialing. |
| 21:15.22 | gusto | easier to say than done |
| 21:15.26 | p3nguin | Create a context for outbound extensions. |
| 21:15.28 | gusto | not today any more |
| 21:15.32 | p3nguin | No, it's really as easy as it sounds. |
| 21:15.33 | gusto | i have ones |
| 21:15.35 | gusto | i am not dumb |
| 21:15.40 | p3nguin | isn't sure |
| 21:15.46 | gusto | and actually it works, but i get 603 from the other side |
| 21:16.02 | gusto | but now i am happy that we got at least this 50% running |
| 21:16.06 | p3nguin | 603? |
| 21:16.09 | p3nguin | What's a 603? |
| 21:16.15 | gusto | no idea |
| 21:16.25 | p3nguin | There's more to a message than just a number. |
| 21:16.31 | p3nguin | What are the words associated with it? |
| 21:17.39 | gusto | no idea |
| 21:17.45 | gusto | i had to look it up as well |
| 21:17.52 | gusto | but that error is not thrown to me |
| 21:17.56 | gusto | it is on the other side |
| 21:18.01 | gusto | and i do not see the whole message |
| 21:18.10 | rrittgarn | kernel: [483891.365841] asterisk[11155]: segfault at 0 ip b76d75b0 sp ad352698 error 4 in libc-2.11.3.so[b7664000+140000] |
| 21:18.46 | rrittgarn | any tips? |
| 21:19.04 | p3nguin | 603 Decline |
| 21:19.15 | p3nguin | They won't accept the call. Show me your extension for dialing out. |
| 21:19.21 | p3nguin | I have a feeling you're doing it wrong. |
| 21:19.33 | p3nguin | I think you're sending the call TO them as opposed to THROUGH them. |
| 21:19.56 | p3nguin | Show me the extension using one of those providers. |
| 21:20.22 | p3nguin | just the line with the Dial() should be enough. |
| 21:20.56 | *** join/#asterisk krotos (~d3v1l@87.13.68.165) |
| 21:20.59 | krotos | hi all guy |
| 21:21.04 | p3nguin | 2295 calls processed |
| 21:21.06 | p3nguin | Not bad. |
| 21:22.02 | sp00kz | over how long? |
| 21:27.08 | gusto | so |
| 21:27.29 | gusto | [Apr 4 21:23:00] NOTICE[3558]: chan_sip.c:22081 handle_request_invite: Call from '' (193.106.16.101:5060) to extension '655161' rejected because extension not found in context 'default'. |
| 21:27.47 | gusto | like i said, there was an unreported IP, i put it in there later |
| 21:27.57 | gusto | but someone can call me, that one works |
| 21:28.32 | p3nguin | I forgot to look at the uptime before I restarted asterisk. It would have been several days. |
| 21:28.51 | gusto | 04/04/12 23:19:15 < p3nguin> They won't accept the call. Show me your extension for dialing out. |
| 21:28.54 | gusto | w8 |
| 21:28.58 | p3nguin | weight again |
| 21:29.08 | gusto | p3nguin: exten => _49.,1,Dial(SIP/personalvoip,,) |
| 21:29.09 | gusto | exten => _01522.,1,Dial(SIP/personalvoip,,) |
| 21:29.16 | p3nguin | That's why it doesn't work. |
| 21:29.36 | gusto | but when i do personalvoip/extension) it behaves the same way |
| 21:29.36 | p3nguin | Should be Dial(SIP/personalvoip/${EXTEN}) |
| 21:29.41 | gusto | so so |
| 21:29.51 | gusto | and what should be the extension? the callbacknumber? |
| 21:30.07 | p3nguin | Dial(SIP/personalvoip) means you are sending the call TO personal voip. You don't want to send it TO them, you want to send it VIA them. |
| 21:30.17 | gusto | yes |
| 21:30.21 | gusto | so? |
| 21:30.23 | p3nguin | Show me a phone number you wish to call. |
| 21:30.38 | gusto | ehm ... in the logs there are some |
| 21:30.42 | gusto | but ill give you one, w8 |
| 21:30.44 | p3nguin | Just type one. |
| 21:31.11 | gusto | "015223817325" |
| 21:31.31 | gusto | over that _1522. extension |
| 21:31.47 | p3nguin | That would match exten => _01522.,1,Dial() |
| 21:31.56 | gusto | ok |
| 21:31.59 | gusto | and then? |
| 21:32.11 | p3nguin | In that case, ${EXTEN} would = 015223817325 |
| 21:32.19 | p3nguin | You would send 015223817325 to the provider. |
| 21:32.23 | gusto | because _1522. matches only the number but then what to do with it |
| 21:32.30 | p3nguin | Is that the correct format for phone numbers? |
| 21:32.36 | gusto | i do not care |
| 21:32.40 | gusto | w8 |
| 21:32.40 | p3nguin | You have to care. |
| 21:32.52 | gusto | you said something that may be importatn |
| 21:32.54 | gusto | important |
| 21:32.57 | p3nguin | If you don't send the correct number format, they will not take the call. |
| 21:33.19 | gusto | because i do not call everytime the same number, so is ${EXTEN} something like a variable? |
| 21:33.25 | p3nguin | yes |
| 21:33.37 | p3nguin | ${EXTEN} is the extension that you called. |
| 21:33.43 | leifmadsen | it contained what the pattern matched |
| 21:33.45 | krotos | i've got a boring client that broke my head saying that there are bad quality on phone call. Is out of my network, and only thing i can do is ping, or mtr to his host. |
| 21:33.47 | gusto | so is ${EXTEN} everytime the number for example _1522. matches to? |
| 21:33.54 | krotos | there is a tool like mtr for measuring jitter? |
| 21:33.58 | p3nguin | If you called 015223817325 and it matched _1522. then it is 015223817325 |
| 21:34.04 | leifmadsen | Asterisk 101... |
| 21:34.14 | p3nguin | If you called 01522111111111 and it matched _1522. then it is 01522111111111 |
| 21:34.19 | gusto | ok ok |
| 21:34.23 | gusto | we give it a try |
| 21:34.34 | p3nguin | leifmadsen: He can't read the book or something. |
| 21:34.46 | p3nguin | Learning impaired. |
| 21:34.48 | leifmadsen | http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics-SECT-3.html#asterisk-DP-Basics-SECT-3.6.3 |
| 21:35.06 | leifmadsen | gusto: read the Dialplan Basics chapter.... |
| 21:35.26 | leifmadsen | http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics.html |
| 21:35.29 | p3nguin | As [tk]d-fender put it, he's "exhibiting a very severe learning disability." |
| 21:35.43 | gusto | ok, one works |
| 21:36.06 | gusto | hmm |
| 21:36.11 | gusto | i would put it other way |
| 21:36.23 | [TK]D-Fender | ${EXTEN} holds whatever was used in the pattern match you're in. |
| 21:36.31 | gusto | take my sentences / clarifications / definitions and put them into some documentation |
| 21:36.32 | [TK]D-Fender | Not necessarily what was initially called |
| 21:36.49 | gusto | 04/04/12 23:33:46 < gusto> so is ${EXTEN} everytime the number for example _1522. matches to? |
| 21:37.00 | [TK]D-Fender | .... |
| 21:37.08 | [TK]D-Fender | that made no sense |
| 21:37.20 | p3nguin | (1633.19) <gusto> because i do not call everytime the same number, so is ${EXTEN} something like a variable? |
| 21:37.23 | p3nguin | (1633.25) <p3nguin> yes |
| 21:39.58 | krotos | hi guy, there are some situation where is helpfull use/enable jitterbuffer on asterisk |
| 21:40.12 | krotos | and other situation where is not helpfull? |
| 21:42.14 | leifmadsen | p3nguin: ya I can't see what TK says |
| 21:42.58 | leifmadsen | krotos: not useful when you don't have jitter as it needs to introduce latency |
| 21:43.06 | leifmadsen | since it's a buffer |
| 21:43.49 | krotos | leifmadsen: so, if in my situation i've got a lot of user with 30/50 ping, and instead of this only 6 with wimax connection(ping so hig..jitter too) |
| 21:44.33 | [TK]D-Fender | On /ignore am I? \o/ |
| 21:55.30 | *** join/#asterisk CyBeRxIxO (~efernande@190.41.182.228) |
| 21:55.48 | CyBeRxIxO | hi again |
| 21:55.49 | CyBeRxIxO | xD |
| 21:56.19 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:59.45 | CyBeRxIxO | my current problems are that i get randomly seconds of silent during the call, and when it starts the person im calling cant listen to my untill 3-5seconds after call started... can anyone help me? im getting the g729 licence but im not sure if that will solve my problem |
| 22:01.12 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 22:09.35 | *** join/#asterisk Korolev (~Korolev@201.225.187.210) |
| 22:11.07 | paolosupino | I'd like to extend a very big thank you for leifmadsen, [TK]D-Fender, p3nguin, kaldemar and everyone else in the channel who planted fear in me of asking stupid question and caused me to look for the correct information before asking… It also made me find the solutions alone :-) |
| 22:16.44 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 22:21.10 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 22:32.10 | pigpen | hi all. I have a new Digium D70 phone. I am running Asterisk 10.1.2, so the DPMA is not happy with it as of yet (at least that is what they say in the docs) |
| 22:33.09 | pigpen | Can anybody that is currently using this phone with the DPMA grab an example config file, likely in the ftp/tftp/http/https directory as a mac_address.cfg file and post it for me so I can provision it manually? |
| 22:33.19 | leifmadsen | pigpen: well the Asterisk 10 phone branch was just branched today |
| 22:33.36 | pigpen | leifmadsen, you are the bearer of good news. |
| 22:33.50 | leifmadsen | I have zero idea as to the status of said branch |
| 22:33.51 | leifmadsen | but it does exist |
| 22:34.04 | pigpen | not that I am a complete idiot, but where would I grab it? |
| 22:34.52 | pigpen | I got this in the standard email: http://downloads.digium.com/pub/telephony/res_digium_phone/ |
| 22:34.56 | pigpen | but it only lists 1.8.11 |
| 22:35.03 | leifmadsen | subversion is the only place |
| 22:35.10 | pigpen | k. under 10.x |
| 22:35.12 | pigpen | ? |
| 22:35.32 | pigpen | or is this under a separate tree? |
| 22:35.36 | leifmadsen | sigh |
| 22:35.37 | leifmadsen | http://svn.asterisk.org/svn/asterisk/branches/ |
| 22:35.42 | leifmadsen | gotta go, later |
| 22:35.53 | pigpen | tks. yeah, I have svn setup. |
| 22:36.01 | pigpen | didn't know if it was a seperate tree other than 10 |
| 22:38.11 | pigpen | oh...seperate branch...10-digiumphones/ |
| 22:42.52 | CyBeRxIxO | does my sip t9 yealink afect voice qualify? is the cisco better? |
| 22:49.30 | gusto | p3nguin: thanks for your help, i would not find out that fast w/o your help |
| 22:50.11 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-136-96.ks.ks.cox.net) |
| 22:50.18 | gusto | is tired and goes to sleep |
| 22:50.31 | *** join/#asterisk ketas (~ketas@ketas6-sixxs.si.pri.ee) |
| 22:51.12 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 22:56.23 | mjordan | pigpen: the 10-digiumphones branch was just created today, so it hasn't gone through the same level of testing that the 1.8-digiumphones branch has (yet). If you decide to play around with it, let me know how it works for you |
| 22:56.28 | *** join/#asterisk albertoandrade (~albertoan@186.206.5.67) |
| 22:56.35 | mjordan | pigpen: fyi, it will mirror, at this point in time, 10.4.0-rc1 |
| 23:00.18 | *** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net) |
| 23:01.04 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 23:15.25 | *** join/#asterisk ketas (~ketas@ketas6-sixxs.si.pri.ee) |
| 23:24.23 | *** join/#asterisk CyBeRxIxO (~efernande@190.41.182.228) |
| 23:24.42 | CyBeRxIxO | my current problems are that i get randomly seconds of silent during the call, and when it starts the person im calling cant listen to my untill 3-5seconds after call started... can anyone help me? im getting the g729 licence but im not sure if that will solve my problem, any advices? |
| 23:24.46 | CyBeRxIxO | ty for reading |
| 23:28.17 | p3nguin | Did you ever check the bandwidth like I suggested? Two ulaw calls will only use up about 160 kb/s, so that's not very much used out of your 20 mbps service. |
| 23:28.32 | p3nguin | Changing to g729 isn't going to solve that jitter you have. |
| 23:33.47 | CyBeRxIxO | my internet bandwidth is 3Mbps at 25% guarante |
| 23:34.25 | CyBeRxIxO | i kno, poor bandwidth. when i was in spain i had 20mb but here in south americ that is a pain |
| 23:34.36 | CyBeRxIxO | 5mb is max bandwidht |
| 23:35.19 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 23:59.42 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-aqtnszpedzustkqk) |