| 00:01.26 | wtf911 | ah the call goes through now but not with the number i set |
| 00:02.04 | BekarfelASUS | should the 1 for CALLERID be n too? |
| 00:02.26 | WIMPy | The lines need to be numbered, starting from 1. |
| 00:02.29 | Ice_Strike | voipstunt does not allow callerid without validatin |
| 00:02.45 | WIMPy | n is a shortcut for next, i.e. the previous line +1 |
| 00:02.48 | Ice_Strike | voipstunt, voipchip and voipbuster is the same company |
| 00:02.50 | BekarfelASUS | ahh |
| 00:03.04 | Ice_Strike | voipcheap* |
| 00:03.21 | wtf911 | would you happen to know of a free one that does? |
| 00:03.34 | Ice_Strike | voip.ms |
| 00:03.44 | Ice_Strike | and flowroute.com |
| 00:09.10 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 00:13.39 | *** join/#asterisk teknoprep (~chrisrawl@unaffiliated/teknoprep) |
| 00:13.52 | teknoprep | hi all i am at a loss |
| 00:13.57 | teknoprep | i get outgoing calls |
| 00:14.06 | teknoprep | but i still don't get how to code inbound calls |
| 00:14.31 | teknoprep | i have a very simple setup? i have one ITSP.. with one Extension |
| 00:14.47 | teknoprep | outbound works great? the extension is called BlueCloud |
| 00:15.05 | WIMPy | Every call consists of an incomming and an outgoing channel. |
| 00:15.06 | teknoprep | i don't know how to route ALL calls to that no matter what the CID or DID is |
| 00:15.20 | teknoprep | WIMPy, yeah i understand SIP really well... |
| 00:15.31 | teknoprep | WIMPy, i don't get asterisk dial plans is what my issue is |
| 00:15.41 | WIMPy | No, Asterisk, not SIP. |
| 00:16.02 | WIMPy | Did you try the book? |
| 00:16.08 | BekarfelASUS | ~book |
| 00:16.08 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 00:16.10 | BekarfelASUS | or smth |
| 00:16.12 | teknoprep | WIMPy, i am good on reading the book |
| 00:16.16 | BekarfelASUS | win! |
| 00:16.26 | teknoprep | i have a very specific need? one extension one ITSP |
| 00:16.28 | teknoprep | thats it |
| 00:16.38 | WIMPy | There are other books, BTW. |
| 00:16.40 | teknoprep | not alot on the internet about it |
| 00:16.51 | p3nguin | One extension? That won't allow very much to happen. |
| 00:17.16 | p3nguin | teknoprep: Didn't I already tell you how to "route" calls? |
| 00:17.23 | teknoprep | shouldn't this be? exten => s,1,Dial(Local/BlueCloud,60) |
| 00:17.29 | teknoprep | p3nguin, didn't work |
| 00:17.33 | p3nguin | You did it wrong. |
| 00:17.38 | teknoprep | i agree |
| 00:17.41 | teknoprep | i did do it wrong |
| 00:17.48 | WIMPy | s only matches calls without destination. |
| 00:17.50 | teknoprep | otherwise i am sure it would work |
| 00:17.59 | p3nguin | Or your request was inaccurate. |
| 00:17.59 | teknoprep | _X. |
| 00:18.14 | p3nguin | s only matches s |
| 00:18.19 | WIMPy | And that local channel doesn;t look very convincing, either. |
| 00:18.37 | teknoprep | why would the local channel not look convinving ? |
| 00:18.50 | p3nguin | Mostly because it is incomplete. |
| 00:18.58 | WIMPy | s matches calls without destination or to "s". |
| 00:19.16 | p3nguin | It's SIP, so the destination would be s or it doesn't match s. |
| 00:19.24 | WIMPy | Firsst of all, I doubt, you want to use a local channel. |
| 00:19.55 | WIMPy | Are you sure, you can't have no destination with sip? |
| 00:20.00 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-zmguxowyhuaxnbhv) |
| 00:20.03 | teknoprep | yes i can use SIP |
| 00:20.06 | p3nguin | You cannot have "no destination" in SIP. |
| 00:20.08 | teknoprep | SIP/BlueCloud would be it |
| 00:20.44 | p3nguin | If you are trying to send all calls from one place to another peer, you use the channel tech that the other system uses. If BlueCloud uses SIP, use SIP/BlueCloud/${EXTEN} in your Dial(). |
| 00:20.50 | WIMPy | And what if yo dial sip:@domain.com? |
| 00:21.08 | p3nguin | exten => _X.,1,Dial(SIP/BlueCloud/${EXTEN}) |
| 00:21.24 | teknoprep | thats what i have p3nguin |
| 00:21.39 | teknoprep | its not working? calls come in and just fail immediatly |
| 00:21.40 | p3nguin | This extension says to "forward" all calls to a peer named BlueCloud. |
| 00:21.58 | p3nguin | Then you should be showing me the evidence. |
| 00:22.00 | p3nguin | sip set debug on |
| 00:22.01 | p3nguin | core set verbose 3 |
| 00:22.02 | teknoprep | i'll pastbin my peer for BlueCloud.. which on this server is called eZuce? |
| 00:22.08 | p3nguin | Make the call. Pastebin the output. |
| 00:22.46 | teknoprep | one other thing |
| 00:23.09 | teknoprep | for the context=inbound-ezuce this should be set on the ITSP settings in the SIP.conf |
| 00:23.22 | teknoprep | then i create a [inbound-ezuce] inside extensions.conf |
| 00:23.31 | p3nguin | okay |
| 00:23.39 | teknoprep | and this is where i then create the exten = everything i want to do to the call |
| 00:23.43 | teknoprep | is this correct assumption |
| 00:23.58 | p3nguin | So far, it sounds like you are correct in describing it. |
| 00:24.30 | p3nguin | Pastebin the peer for the ITSP and the peer for the other system. |
| 00:24.47 | teknoprep | can't pastebin the peer for the other side |
| 00:24.54 | teknoprep | but it is registered... |
| 00:24.58 | p3nguin | Because of what? |
| 00:25.08 | teknoprep | its not asterisk based |
| 00:25.12 | teknoprep | oh. |
| 00:25.17 | teknoprep | you want both sides of asterisk |
| 00:25.19 | teknoprep | will do |
| 00:25.20 | p3nguin | In your sip.conf, you have to have a peer entry for both places. |
| 00:25.51 | teknoprep | pastebin.com/tcaKnnzL |
| 00:26.13 | WIMPy | Nexst time, please send a complete URL. |
| 00:26.44 | p3nguin | What's wrong with what he sent? |
| 00:27.02 | WIMPy | I can't click on it. |
| 00:27.03 | p3nguin | If your browser doesn't know how to deal with the address, get a better browser. |
| 00:27.18 | p3nguin | I can't click URLs either, but you don't see me complaining. |
| 00:27.27 | WIMPy | What does the browser have to do with that? |
| 00:27.34 | p3nguin | Nothing. |
| 00:28.21 | teknoprep | p3nguin, pastebin.com/GHNqRCHg |
| 00:28.36 | p3nguin | empty |
| 00:31.45 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-plumqmlifdtsmzis) |
| 00:32.23 | p3nguin | teknoprep: That paste is empty. I need to see the peer definitions. |
| 00:32.54 | teknoprep | i am having a bit of an issue one sec please |
| 00:33.02 | teknoprep | i can't paste from an RDP session on my mac |
| 00:33.07 | teknoprep | or copy |
| 00:33.28 | teknoprep | http://pastebin.com/hPcxkvLC |
| 00:33.29 | teknoprep | there we o |
| 00:33.45 | p3nguin | Use curlpaste, pastebinit, or something similar. |
| 00:33.46 | BekarfelASUS | his copypasta isn't aldenté yet |
| 00:34.26 | p3nguin | You don't have a peer definition for BlueCloud. |
| 00:34.34 | teknoprep | its eZuce |
| 00:34.46 | p3nguin | Then why were you dialing BlueCloud above? |
| 00:34.47 | WIMPy | And register => goes to [general] |
| 00:34.50 | teknoprep | this setup is slightly different but i have the context name changed to eZuce |
| 00:34.56 | teknoprep | i don't have access to the BlueCloud box |
| 00:35.00 | p3nguin | I don't care. |
| 00:35.09 | p3nguin | You still have to have a peer entry for it. |
| 00:35.40 | teknoprep | no if i am not using it in my exten => |
| 00:35.52 | teknoprep | exten => _X.,1,Dial(SIP/eZuce/${EXTEN}) |
| 00:36.01 | p3nguin | That's different from what you said earlier. |
| 00:36.04 | teknoprep | and the context that this is in -> exten => _X.,1,Dial(SIP/BlueCloud/${EXTEN}) |
| 00:36.25 | p3nguin | Exactly. You are dialing a peer named BlueCloud. |
| 00:36.27 | teknoprep | i did also mention about the naming convention for this earlier also.. but i don't want to argue |
| 00:36.28 | p3nguin | But you don't have one. |
| 00:36.58 | p3nguin | No peer named BlueCloud means that Dial() will die a horrible death. |
| 00:36.59 | teknoprep | bah fuck i don't have paste |
| 00:37.02 | teknoprep | its eZuce |
| 00:37.17 | teknoprep | i am logged in remotely ? to a system without SSH access |
| 00:37.25 | p3nguin | Then stop dialing SIP/BlueCloud/${EXTEN} |
| 00:37.28 | teknoprep | so i can not directly copy and paste? its such a hassle |
| 00:37.40 | teknoprep | i am no longer dialing that? i am dialing? exten => _X.,1,Dial(SIP/eZuce/${EXTEN}) |
| 00:37.42 | teknoprep | i promise |
| 00:37.43 | p3nguin | If you are on the terminal, use curlpaste or pastebinit like I said. |
| 00:37.44 | teknoprep | lol |
| 00:38.05 | p3nguin | Where does the call come FROM? |
| 00:38.17 | p3nguin | nexvortex? |
| 00:38.17 | teknoprep | nexvortex |
| 00:38.18 | teknoprep | yes |
| 00:38.25 | p3nguin | Fix the register statement. |
| 00:38.33 | p3nguin | (1200.14) <p3nguin> Register statements can only go in the general section, too. |
| 00:38.44 | p3nguin | I said that hours ago. |
| 00:40.23 | p3nguin | nexvortex is not behing a nat, so change nat=yes to nat=no. Change canreinvite=nonat to directmedia=no. |
| 00:40.31 | p3nguin | s/behing/behind/ |
| 00:43.09 | *** join/#asterisk internat (~nf@60.241.102.25) |
| 00:44.36 | teknoprep | nothing |
| 00:44.42 | teknoprep | i do get ringing after 14 seconds tho |
| 00:44.57 | *** join/#asterisk _370z-hsv (~E63AMG@207.98.192.200) |
| 00:45.02 | p3nguin | Still waiting for the debug. |
| 00:45.15 | _370z-hsv | penguin, I thought this room would be dead |
| 00:45.21 | _370z-hsv | because Diablo3 comes out in 4 hours heh |
| 00:45.24 | p3nguin | (1921.57) <p3nguin> sip set debug on |
| 00:45.24 | p3nguin | (1922.01) <p3nguin> core set verbose 3 |
| 00:45.27 | p3nguin | (1922.08) <p3nguin> Make the call. Pastebin the output. |
| 00:46.07 | teknoprep | i think i had an issue with NAT in my firewall |
| 00:46.19 | teknoprep | one sec and if that doesn't work i'll post up the trace |
| 00:54.44 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 00:58.31 | *** join/#asterisk Lepon (Lepon@60-241-109-208.static.tpgi.com.au) |
| 00:59.13 | teknoprep | the initial invite isn't hitting the system? wtf |
| 00:59.14 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 01:00.08 | teknoprep | http://pastebin.com/Q6nNTZQ1 |
| 01:00.10 | teknoprep | now it is |
| 01:00.27 | teknoprep | No matching peer for '6107413324' from '66.23.129.253:5060' is that right ? |
| 01:03.31 | Lepon | Hey all, wondering if I could get some help with debuging my FastAgi script in asterisk 1.6. Basically the setup is I have asterisk 1.6 running debian and am generating some calls from a windows server using the management interface. Once the call is connected it execute a fastagi script (also running on my windows box) to control the flow of the call. This all works fine, generates the calls |
| 01:03.32 | Lepon | controls them through fastagi, until I try and do a set extension command. When i do that my call just terminates. What I am trying to do is get some debug output from asterisk to see what is happening when my fastagi script executes to determine why it is dying. Hope that all makes sense. |
| 01:04.01 | *** join/#asterisk danfromuk (~IceChat77@2.26.250.118) |
| 01:07.46 | wtf911 | hey p3nguin |
| 01:10.58 | *** part/#asterisk danfromuk (~IceChat77@2.26.250.118) |
| 01:11.29 | wtf911 | i'm using flowroute.com with this line... |
| 01:11.30 | wtf911 | exten => _XXXXXXXXXXX,n,Set(CALLERID(num)=2095551214) |
| 01:13.03 | wtf911 | and the caller id is showing up as unavailable |
| 01:15.58 | Lepon | Any thoughts would be much apperciated. |
| 01:16.01 | wtf911 | i've tried putting a 1 in front |
| 01:18.15 | Lepon | wtf911 - no idea about your provider but I know the provider we use restricts what callerid values we can set for outgoing calls and if we set it to something they don't allow it goes out as private. For external calls of course not internal. |
| 01:18.53 | wtf911 | i'm going to try setting p asserted identity i'll let you know |
| 01:27.26 | wtf911 | hmm |
| 01:30.20 | wtf911 | not the fix |
| 01:32.49 | wtf911 | i tried a new york number and a north carolina number, both landlines, and no dice still getting unavailable |
| 01:33.33 | Lepon | have you ever had an outgoing callerid working? maybe they block them all together? |
| 01:33.53 | Lepon | im defiantly not an asterisk expert though, just going on what i have experienced |
| 01:34.40 | wtf911 | nope i'm new to all of this |
| 01:34.51 | wtf911 | but i know flowroute does caller id passthrough |
| 02:17.29 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 02:20.32 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 02:21.16 | *** join/#asterisk Yourname` (~Administr@unaffiliated/yourname/x-837320) |
| 02:22.05 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 02:23.12 | Yourname` | Call going out from Remote agent over NAT, out the firewall, on the internet.. to another router, then to port-forwarded asterisk, then out again over the internet through another sip trunk.. and no audio. What could be the issue? |
| 02:32.51 | *** join/#asterisk nickgaw (~nick@70-139-55-30.lightspeed.hstntx.sbcglobal.net) |
| 02:33.40 | nickgaw | Hi, What section do I look for in sip.conf to configure asterisk so it can answer calls at another sip number and then forward those calls to my internal asterisk system? |
| 02:35.50 | *** join/#asterisk F2Knight (~ben@71-222-66-159.ptld.qwest.net) |
| 02:39.12 | ectospasm | Yourname`: looks like you've got double-NAT issues. |
| 02:39.33 | Yourname` | ectospasm: Anyway to fix it? |
| 02:39.54 | F2Knight | is double natted. |
| 02:40.15 | ectospasm | Yourname`: maybe: http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:40.51 | Yourname` | ectospasm: Following that one left and right, but nada. All ports open everywhere. My nat, pbx nat. |
| 02:41.10 | ectospasm | yeah, SIP wasn't designed to traverse NAT. Sorry. |
| 02:53.39 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 02:53.39 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 02:58.53 | Yourname` | All good.. |
| 02:59.52 | Yourname` | ectospasm: Here's another thing. How come that the Asterisk server that's behind the NAT with perfectly fine forwarded ports NOT able to do proper calls when i set externip/localnet? |
| 03:05.58 | JamKo | Yourname`: Could be your firewall sucks, or has some kind of sip alg running. It could also be the remote client's firewalls. |
| 03:06.22 | Yourname` | JamKo: I'd like to think so, but cisco isn't known for sucky firewalls :(\ |
| 03:06.39 | JamKo | Yourname`: depends who is configuring it. |
| 03:06.42 | JamKo | :) |
| 03:06.45 | Yourname` | lol true |
| 03:07.13 | JamKo | Yourname`: you should get that asterisk box on a class A ip. |
| 03:07.28 | JamKo | Yourname`: that cuts your problems with nat in half. |
| 03:07.45 | JamKo | Yourname`: nat on both ends is going to be a nightmare. |
| 03:07.55 | JamKo | nat on one end is bad enough. |
| 03:10.21 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 03:10.37 | Yourname` | I know, FML :( |
| 03:10.48 | Yourname` | I can't do either, both of them are out of control.. :S |
| 03:11.13 | *** join/#asterisk elkng (~elkng@unaffiliated/elkng) |
| 03:13.30 | JamKo | Yourname`: It's a mess. The problem is getting that RTP stream to punch through the correct ports. The SIP control ports generally are easy enough to get going, but the RTP is batshit. |
| 03:14.43 | Yourname` | I've done rtpstart=10000, rtpend=10700 and opened those ports everywhere as udp. remote nat, office nat and sip trunk iptables.. |
| 03:15.07 | Yourname` | JamKo: Good thing is calls between remote ext <-> office pbx (both NAT) are doing well.. |
| 03:15.26 | Yourname` | it's just when it goes remote > office pbx > sip trunk is the issue.. |
| 03:16.37 | Yourname` | In the beginning office pbx had trunk settings as fromdomain=IP of the sip trunk and fromuser. That had no audio at all. Removed those, and audio is there, but choppy and illegible. |
| 03:16.43 | JamKo | Yourname`: at least you are using iptables. That gives you half a chance. Get packet captures. That will show you where the ports are going wrong. |
| 03:17.14 | Yourname` | Now there's trustrpid and sendrpid in there which I'm reading about to see how it would work if I remove them.. |
| 03:17.33 | Yourname` | JamKo: What do you think I narrow the problem by first stopping iptables for now? |
| 03:18.08 | JamKo | Stopping iptables on the asterisk box, or is it running as a router somewhere in between? |
| 03:19.46 | Yourname` | JamKo: That's the sip trunk the calls are going to from the pbx |
| 03:21.26 | JamKo | Yourname`: if the sip trunk you are referring to is asterisk, and you have it behind a cisco wall, then yes I would definitely shutdown netfilter at this point. |
| 03:22.34 | Yourname` | JamKo: One sec. Remote agent -> cisco -----internet------> cisco-based-router -> asterisk ------internet --------> centos server that is running a2billing/asterisk (has iptables) |
| 03:23.46 | JamKo | Yikes man. Is there a firewall in front of that centos box? |
| 03:25.18 | Yourname` | JamKo: Nope. Had iptables. Just tried without iptables, and still choppy calls. |
| 03:25.22 | *** part/#asterisk BekarfelASUS (~here@173.85.204.173) |
| 03:25.40 | ectospasm | IP tables wouldn't affect choppy audio on calls. |
| 03:25.45 | ectospasm | check your network latency |
| 03:25.59 | Yourname` | ectospasm: Latency is low low. Like under 20ms |
| 03:26.05 | JamKo | yea chop is a jitter issue most likely. |
| 03:26.18 | ectospasm | which segment is 20ms? You've got at least three. |
| 03:26.21 | JamKo | or latency as ectospasm is stating. |
| 03:28.19 | JamKo | Yourname`: as long as iptables nat policy is accept, and you have the correct filter table rules set to allow communication, it should be ok. A nat/firewall issue is calls fail, no audio, or one way audio. |
| 03:28.20 | ectospasm | ...also, what codec are you using? If you're using G.711, that could also be contributing to the poor audio |
| 03:28.35 | Yourname` | ectospasm: ulaw |
| 03:28.43 | JamKo | good call ectospasm . |
| 03:28.43 | Yourname` | JamKo: Gotcha. |
| 03:28.46 | ectospasm | try a lower bitrate codec |
| 03:28.53 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 03:29.00 | JamKo | yea ulaw is the worst. get off of that. |
| 03:29.04 | ectospasm | ...like GSM, or G.729 if you have the licenses for it. |
| 03:29.18 | Yourname` | If I had the license for g729, I'd so go for it |
| 03:29.54 | Yourname` | PS I want you to know. At first I had issues all around. Calls from remote sip to remote sip exten via the same pbx were choppy, inbound/outbound choppy, staticky, etc.. |
| 03:30.10 | Yourname` | But then I shortened udp ports and made sure they were open on the firewall and that worked. |
| 03:30.18 | Yourname` | Removed ALL the choppy issues |
| 03:30.28 | Yourname` | Now it's just the sip trunk calls that are choppy. |
| 03:31.01 | JamKo | doesnt sound right. chop should not be related to ports. Its either going through the right port or it's not. |
| 03:31.31 | JamKo | should say "going to" the right port. |
| 03:31.58 | JamKo | Your only issue is chop? |
| 03:32.18 | Yourname` | JamKo: Yeah, as of now |
| 03:32.30 | Yourname` | Might not be the ports like ya'll say |
| 03:32.35 | Yourname` | Maybe it's the IP settings? |
| 03:32.40 | ectospasm | enable jitter buffer in sip.conf |
| 03:32.47 | Yourname` | ectospasm: Already done |
| 03:32.50 | ectospasm | ...choose a low bitrate codec (GSM is free) |
| 03:33.22 | ectospasm | Digium sells G.729 channels for $10/channel, you might explore that option as well... |
| 03:33.44 | ectospasm | 10USD, btw. |
| 03:34.29 | Yourname` | For now I'm gonna have to try and fix this atleast a little before going there.. |
| 03:34.30 | Yourname` | lol |
| 03:34.36 | Yourname` | Gonna try gsm now |
| 03:34.51 | ectospasm | note that you may be unhappy with GSM's quality |
| 03:35.51 | *** join/#asterisk greenwolf (4b7d97c2@gateway/web/freenode/ip.75.125.151.194) |
| 03:36.00 | Yourname` | lol so what will I be fixing then? |
| 03:37.42 | Yourname` | Tried it |
| 03:37.43 | ectospasm | bandwidth issues which may be causing the problem. |
| 03:37.46 | Yourname` | <PROTECTED> |
| 03:37.47 | Yourname` | <PROTECTED> |
| 03:37.47 | Yourname` | <PROTECTED> |
| 03:37.54 | ectospasm | ~pastebin |
| 03:37.54 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 03:37.56 | Yourname` | That's what I see on that sip trunk (a2billing) |
| 03:38.41 | ectospasm | that means you've created a fixed jitter buffer on the SIP/voxbeam connection. |
| 03:38.57 | ectospasm | ...and it was subsequently destroyed when the channel was destroyed. |
| 03:39.24 | Yourname` | Yea lol, but I'm saying jitterbuffer is enabled and still it's being a brat |
| 03:40.45 | ectospasm | ...if the choppy audio isn't jitter, then the jitter buffer won't do you any good. |
| 03:41.03 | Yourname` | Right |
| 03:41.11 | ectospasm | you could always try the adaptive jitter buffer. |
| 03:41.17 | Yourname` | Any other ideas what this could be? gsm, ulaw = both.. |
| 03:41.39 | ectospasm | endpoints attached to Asterisk >> SIP trunk are OK? |
| 03:41.57 | ectospasm | i.e., not remote endpoints |
| 03:42.50 | Yourname` | ectospasm: The pbx has jitterbuffer enabled, and it's fixed. Want me to switch to adaptive? |
| 03:43.25 | ectospasm | can't hurt |
| 03:43.30 | Yourname` | ectospasm: Good question. My apologies for forgetting to mention that. The endpoints connected to the pbx are also not able to get good call quality when dialed through the trunk.. |
| 03:43.37 | ectospasm | ...though it probably won't work. |
| 03:44.11 | ectospasm | Yourname`: what's the bandwidth on the connection to the SIP trunk? |
| 03:44.28 | Yourname` | It's at a rackspace facility |
| 03:44.39 | *** join/#asterisk dualcore (~simon@c-68-61-204-7.hsd1.mi.comcast.net) |
| 03:44.41 | ectospasm | that doesnt' answer the question. |
| 03:45.04 | ectospasm | ...and I'm sure Rackspace can limit your bandwidth like any other ISP. |
| 03:45.46 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-blgjbvdolajremrg) |
| 03:46.16 | Yourname` | True, but others who are connected to that same box are able to make great calls |
| 03:46.31 | ectospasm | ... |
| 03:46.59 | Yourname` | Meaning its when the officepbx is sending calls, its choppy. When other pbxs are connected to it, those calls are fine, or other sip trunks, etc |
| 03:47.00 | ectospasm | you STILL haven't answered what the bandwidth is between Asterisk and the SIP trunk. |
| 03:47.16 | ectospasm | Yourname`: are those other SIP trunks at your location? |
| 03:47.20 | ectospasm | er, PBXes |
| 03:47.25 | Yourname` | Ohhhh BETWEEN asterisk and the sip trunk. Asterisk is on a 6mb dsl line. And the SIP trunk is at rackspace with gigE backbones |
| 03:47.34 | Yourname` | ectospasm: Nope |
| 03:47.56 | ectospasm | Yourname`: I bet your upload capacity on that DSL line is <<6mb. What's your upload capacity? |
| 03:48.09 | Yourname` | 1mb |
| 03:48.23 | carrar | get a second DSL for voice traffic |
| 03:48.31 | Yourname` | ectospasm: Oh also, I have a GSM gateway connecting to the same sip trunk (gateway is in the same location as the pbx) -> no problemos. |
| 03:48.46 | Yourname` | The gsm gateway is also using the same dsl |
| 03:49.12 | ectospasm | ...at the same time as Asterisk? What happens if you disable the GSM gateway? |
| 03:52.45 | Yourname` | Not same time |
| 03:52.52 | Yourname` | OHHH!~ |
| 03:52.58 | Yourname` | Just saw this during a call through the gsm gatway: |
| 03:52.59 | Yourname` | <PROTECTED> |
| 03:52.59 | Yourname` | [2012-05-14 22:52:03] NOTICE[31404]: rtp.c:849 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 208.64.6.2 |
| 03:53.26 | ectospasm | turn off comfort noise in the client, just as it says. |
| 03:53.55 | Yourname` | Yeah makes sense, but I can't find that on eyebeam anywhere |
| 03:54.20 | Yourname` | Also, asked one of the users who are in the office, connected to the pbx directly to call me over that sip trunk .. and they experienced choppiness as well |
| 03:54.21 | ectospasm | ...but that probably isn't a clue to your choppy audio. Your limited bandwidth probably is. Remember that GSM using 13kpbs/channel, whereas G.711 uses 64kbps/channel |
| 03:54.39 | Yourname` | ectospasm: Currently on GSM as you requested |
| 03:55.57 | ectospasm | what does speedtest.net say about your bandwidth? Have you done a speed test? I bet it's less than 6Mbps/1Mbps... |
| 03:57.49 | Yourname` | Yeah, did that yesterday when I was experiencing the same problems.. |
| 03:58.11 | Yourname` | And it was 6.1mbps down and 1.5mbps up |
| 03:59.00 | ectospasm | A DSL connection in general is not enough. I have a cable connection, getting 10.84Mbps/2.36Mbps, and it has choppy audio at times. |
| 04:00.02 | dualcore | and dsl is very inconsistent. plus the speedtests are with extremely fast servers. |
| 04:01.34 | greenwolf | dualcore: wrong!! DSL is very consistant as it is dedicated |
| 04:01.41 | greenwolf | sorry your knowledge is totally off |
| 04:01.47 | greenwolf | its cable thats inconsistant |
| 04:02.49 | Yourname` | Guys, I'm positive it's not bandwidth issues at this point. No one's using the connection other than this test, so 2 employees in the office. And me and another employee at the remote site. |
| 04:03.23 | dualcore | i could certainly be wrong. i supposed when i had dsl it was consistently slow. sorry about that! |
| 04:03.32 | dualcore | suppose* |
| 04:03.32 | F2Knight | Yourname`, just poking in... what is your issue? (the short simple to the point version) |
| 04:04.35 | Yourname` | F2Knight: Hi. Call going out from Remote agent over NAT, out the firewall, on the internet.. to another router, then to port-forwarded asterisk, then out again over the internet through another sip trunk.. and choppy audio. |
| 04:09.09 | F2Knight | Yourname`, the asterisk server, is it doing any transcoding? |
| 04:09.54 | Yourname` | F2Knight: Nope. Ulaw all the way. But if transcoding was the issue. The local extensions of the pbx would have quality issues too, right? |
| 04:10.00 | F2Knight | Yourname`, and is it possible to put it on a public IP? at least to test. |
| 04:10.25 | Yourname` | F2Knight: Nope :S I did however put it on dmz and it was the same |
| 04:11.04 | greenwolf | which codecs do transcoding and which dont> |
| 04:11.29 | greenwolf | like which codec is easier on asterisk where it needs no transcoding? |
| 04:11.46 | F2Knight | Choppy calls are hardly nat issues... normaly they are bandwidth issues... poor upload from one side, or high latency , |
| 04:11.46 | Yourname` | Good question greenwolf |
| 04:11.53 | _370z-hsv | Diablo3 - 2 hours |
| 04:11.58 | jpsharp | Or a server that can't keep up. |
| 04:12.22 | F2Knight | Yourname`, what is your result from this http://voiptest.8x8.com/ |
| 04:12.23 | Yourname` | F2Knight: I understand and ectospasm and JamKo have been trying to tell me that as well. |
| 04:12.46 | Yourname` | F2Knight: I'm at the remote site right now, where would you want that test from? |
| 04:13.03 | Yourname` | Is anyone good at SIP packets in this situation btw? |
| 04:13.12 | greenwolf | off topic, but why is everyone in the freeswitch channel such assholes |
| 04:13.28 | greenwolf | its like they think they are so great and holy over there...sorry this was off topic |
| 04:13.37 | Yourname` | greenwolf: Cuz they know they got it good lol |
| 04:14.08 | greenwolf | lol my god i asked one question of there and they are just complete asshole losers if you ask me |
| 04:14.41 | greenwolf | first they just completely ignore me then yell at me like they are freakin god and i should be a freeswitch guru...so irritating those ppl..airheads |
| 04:15.06 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 04:15.14 | F2Knight | Yourname`, this is not a sip packet issue.. the call is completing no need to look at that. |
| 04:15.42 | Yourname` | F2Knight: Ah. I was thinking maybe an internal IP is being listed somewhere causing packets to not know where to go.. |
| 04:16.28 | F2Knight | greenwolf, because they think freeswitch is better then asterisk ... and they have a point it works better in lots of places.. but to me its a lot harder to configure. so thats why not a lot of people use it and those that do ..(*big names* have spent thousands to get it working |
| 04:16.46 | jpsharp | Set everything to "canreinvite=no" or "directmedia=yes" and that will make sure asterisk stays in the middle of the call and there's no issues of RTP packets not going where they need to go. |
| 04:16.48 | F2Knight | Yourname`, run the test |
| 04:17.19 | F2Knight | jpsharp, you could also call Dial with the Tt options too |
| 04:17.55 | greenwolf | f2knight: totally agree..i am an asterisk guru and love it all the way |
| 04:18.00 | Yourname` | jpsharp: canreinvite is set to no everywhere.. just the directmedia part, where should I set it? trunk settings? and maybe the extension for now? |
| 04:18.18 | greenwolf | asterisk is the most straight forward way to program phone systems and i wouldnt have it any other way |
| 04:18.38 | jpsharp | Its directmedia or canreinvite depending on your version of * |
| 04:18.39 | greenwolf | asterisk just makes sense...freeswitch is stupid XML which makes no sense if you ask me |
| 04:18.42 | F2Knight | greenwolf, asterisk is wonderful as a PBX its not so good as a telephony platform. |
| 04:19.21 | F2Knight | Yourname`, what is the result of the test i pointed you to |
| 04:20.07 | greenwolf | f2knight: thats why you put opensips in front of asterisk and now you got a full robost enterprise voip platform |
| 04:20.09 | greenwolf | :) |
| 04:20.15 | greenwolf | better than freeswitch |
| 04:20.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 04:20.29 | greenwolf | plus openSIPS has true load balancing |
| 04:20.37 | F2Knight | Kamailio |
| 04:20.44 | greenwolf | which i like and if you integrate that with asterisk switch....GAME OVER |
| 04:21.04 | greenwolf | opensips, kamailio same thing right? |
| 04:21.04 | F2Knight | but yes.. again though Kamailio/OpenSIP/SER has a very steep learning curve.. |
| 04:21.26 | Yourname` | F2Knight: upstream jitter: 1.5ms, downstream jitter: 3.5ms, packetloss up/down: 0.0% |
| 04:21.27 | greenwolf | yea i just bought the openSIPS book online last night off amazon |
| 04:21.39 | Yourname` | F2Knight: MOS 4.0 |
| 04:21.44 | F2Knight | actually have a Kamailio / Yate setup. works nice.. Yate handles the interface to the PRI's |
| 04:21.57 | Yourname` | jpsharp: I'd think canreinvite=no.. ALL over the place lol |
| 04:22.27 | F2Knight | Yourname`, after the test is done the advance tab has a linke in the bottom right that says view results. Pastebin that |
| 04:22.43 | Yourname` | F2Knight: http://voiptest.packet8.net:82/myspeed/db/report?id=1667225 |
| 04:24.12 | Yourname` | jpsharp: This might interest you, all public IPs in sip packets.. thanks to canreinvite=no |
| 04:25.30 | F2Knight | Yourname`, your ip is 202.78.81.148? |
| 04:25.41 | Yourname` | F2Knight: Yup |
| 04:25.48 | F2Knight | you are in Japan? |
| 04:25.49 | Yourname` | (remote site) |
| 04:26.12 | jpsharp | Yourname`: When asterisk is in NAT mode, it ignores what's in the SIP packet and looks at the IP source fields. |
| 04:26.14 | Yourname` | F2Knight: Philippines |
| 04:26.27 | F2Knight | and where are you connecting too? |
| 04:26.37 | Yourname` | jpsharp: Even ip source fields are public ips |
| 04:26.51 | F2Knight | because I am in the NW UsA and am seeing HIGH lag times starting around kddnet |
| 04:27.04 | Yourname` | F2Knight: Locally, here. PBX is 500meters away in another office. The sip trunk rackspace server is in SG. |
| 04:27.43 | F2Knight | what do you see from a linux console when you run mtr and the IP of your server |
| 04:28.57 | jpsharp | Er, wait. If the SIP device is behing NAT, the SIP packet should have an internal IP but the IP header will have the public. Something is rewriting your SIP packets and probably jacking with your RTP stream. |
| 04:29.57 | greenwolf | make sure u put in your sip.conf externip and local fields |
| 04:30.13 | greenwolf | that will fix your nat so it can be found outside the firewall |
| 04:30.30 | greenwolf | thats how i got it working on mine when i had that problem |
| 04:31.11 | Yourname` | F2Knight: MTR all over the place: http://pastebin.ca/2148486 |
| 04:31.47 | Yourname` | greenwolf: Good one. Here's the issue with that though. I did that yesterday, and I had absolutely no audio. :S |
| 04:32.32 | greenwolf | make sure you add the subnet mask |
| 04:32.41 | greenwolf | and dmz the box |
| 04:32.47 | greenwolf | should b good then |
| 04:32.49 | F2Knight | Yourname`, good response times for sure... no real lag even over a time period.. and all calls on remote are choppy.. |
| 04:32.52 | Yourname` | jpsharp: http://pastebin.ca/2148489 errr really? :S |
| 04:33.13 | Yourname` | F2Knight: Now you see what I mean, bandwidth nowhere is an issue as of now. Even during heavy times.. :S |
| 04:33.28 | Yourname` | greenwolf: I'm gonna try this now.. |
| 04:33.59 | Yourname` | greenwolf: Really scared to tho, lol.. gonna see what jpsharp says about the packets.. |
| 04:34.23 | greenwolf | why if it does not work then revert..no biggie |
| 04:34.36 | greenwolf | u dont have to dmz just port forward then |
| 04:34.45 | Yourname` | jpsharp: I initiated the call to 63927... from local IP: 172.18.111.5 fyi |
| 04:34.50 | greenwolf | i say dmz so atleast u can see if thats the prob |
| 04:35.08 | Yourname` | greenwolf: portforwards are done, that's the thing yesterday when i did that, it gave me no audio |
| 04:35.29 | greenwolf | its your router |
| 04:35.40 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 04:35.43 | greenwolf | you must add those fields to sip.conf then for it to work correctly |
| 04:35.56 | greenwolf | ill send you an example |
| 04:37.52 | Yourname` | k |
| 04:38.59 | *** join/#asterisk roham (~ali@94.101.189.4) |
| 04:40.43 | F2Knight | Good night all. Off to bed for me. |
| 04:40.54 | Yourname` | Thanks for your help f2 |
| 04:43.06 | Yourname` | Just tested a call guys, didn't change anything and it worked :S totally werid |
| 04:43.08 | Yourname` | weird* |
| 04:43.57 | Yourname` | The only thing that was happening is I was pinging the office pbx public IP from the computer I called from |
| 04:44.05 | *** join/#asterisk TehRabbitt (~User@unaffiliated/tehrabbitt) |
| 04:50.39 | jpsharp | Yourname`: You've got something rewriting your SIP packets. An Application Layer Gateway somewhere. |
| 04:51.09 | Yourname` | jpsharp: I'm at the remote site. Firewall is a cisco. You think that's doing it? |
| 04:51.36 | jpsharp | What kind of Cisco box is it? An ASA/PIX or a router? |
| 04:52.09 | jpsharp | I think Cisco gear defaults to having the ALG to "on". At least the more recent stuff. |
| 04:52.48 | Yourname` | jpsharp: Let me find out |
| 04:52.56 | Yourname` | jpsharp: I think it's AS something |
| 04:53.48 | jpsharp | I've had nothing but heartburn with Cisco's ALG. I always turn it off. |
| 04:56.50 | Yourname` | jpsharp: Getting the exact model now. |
| 04:56.56 | Yourname` | jpsharp: Catalyst 2600 he says |
| 04:58.31 | jpsharp | That's a switch, not a router/firewall. |
| 04:59.07 | jpsharp | Unless its a 2650 series router. |
| 05:00.01 | jpsharp | But it won't say Catalyst. |
| 05:02.16 | Yourname` | jpsharp: Ah, ok. Apparently their "firewall" is a mandrake server |
| 05:04.00 | jpsharp | I don't know if Linux iptables can rewrite SIP/RTP packets. |
| 05:05.46 | Yourname` | ok |
| 05:06.08 | Yourname` | jpsharp: I asked him to check what sip alg is like on any of their ciscos, he said he'll find out |
| 05:06.33 | Yourname` | jpsharp: But emphasized he doesnt wanna turn it off till sunday :S i wonder if there's anyother thing on their network that uses sip alg |
| 05:07.43 | jpsharp | You won't find ALG on any of the Cisco switches. If they're running a Mandrake box as a firewall, that's what's playing games with your SIP traffic. |
| 05:08.06 | jpsharp | But I don't know how or even if you can turn off SIP rewriting. |
| 05:08.14 | jpsharp | Of course, that may not even be the problem at all. |
| 05:08.51 | jpsharp | export DISPLAY=belchfire.fivecats.org:0 |
| 05:09.01 | jpsharp | Duh. Wrong window. |
| 05:10.33 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 05:11.29 | Yourname` | True, but let's hope it's turned on SOMEWHERE lol |
| 05:13.38 | jpsharp | True. Otherwise your SIP calls are working by smoke and mirrors. |
| 05:17.29 | Yourname` | lol |
| 05:17.29 | Yourname` | While that's going on, totally weird but all calls are now being clear :S |
| 19:01.05 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 19:01.05 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.4.0 (2012/05/02), 1.8.12.0 (2012/05/02), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 19:01.33 | GeoGeek | leifmadsen: Looking forward to the new edition. I have just about worn my ebook version of the current one out. Have a hard copy too. |
| 19:02.22 | *** join/#asterisk \khan (~khan@S0106d49a205b440c.ok.shawcable.net) |
| 19:02.31 | \khan | hi, tangentally related, any softphone recommendations for windows? |
| 19:02.39 | \khan | pulling my hair out trying to find one that isn't terrible |
| 19:02.54 | sarsaeol | tried yate? |
| 19:03.06 | \khan | negative |
| 19:03.07 | leifmadsen | jitsi & zoiper |
| 19:03.15 | wtf911 | p3nguin: i don't need the exten => 808,1,Answer? |
| 19:03.33 | sarsaeol | \khan: im a fan of yate's softphone… as many accounts as you want |
| 19:03.34 | p3nguin | You almost never need an Answer() before a Dial(). |
| 19:03.39 | \khan | thanks for the recommendations, sarsaeol and leifmadsen |
| 19:04.10 | sarsaeol | \khan: np |
| 19:04.24 | tzafrir | ~pb |
| 19:04.25 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:04.38 | tzafrir | (the bot has to do some work) |
| 19:05.01 | tzafrir | sarsaeol, I like twinkle |
| 19:05.16 | GeoGeek | Is twinkle on windoze? |
| 19:05.50 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 19:06.40 | p3nguin | I prefer zoiper classic on Windows. |
| 19:11.47 | wtf911 | p3nguin: i get connected, 30 seconds or so of silence, a busy signal...other phone never rings |
| 19:11.54 | *** join/#asterisk teknoprep (~chrisrawl@unaffiliated/teknoprep) |
| 19:11.55 | teknoprep | hi all |
| 19:12.22 | teknoprep | still having some issues? is there any way to make sure the to header is written with phone#@external.ip |
| 19:12.29 | teknoprep | instead of phone#@internal.ip |
| 19:12.33 | wtf911 | p3nguin: this error with verbose 10 debugging on... |
| 19:12.47 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 19:12.49 | wtf911 | http://pastebin.com/pCBD6KeR |
| 19:13.11 | teknoprep | also the contact header needs to display my external ip address |
| 19:13.13 | teknoprep | not the internal |
| 19:13.15 | Katty | so. i was thinking about going back to being a vegetarian. |
| 19:13.21 | kaldemar | ~sipnat |
| 19:13.21 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 19:13.24 | teknoprep | my from also |
| 19:13.33 | kaldemar | teknoprep: ^ |
| 19:13.37 | teknoprep | ? |
| 19:13.37 | sarsaeol | Katty: yes do |
| 19:13.58 | teknoprep | i already did externaddr |
| 19:14.00 | teknoprep | localnet |
| 19:14.07 | teknoprep | nat=yes |
| 19:14.13 | Katty | sarsaeol: elaborate on your thoughts. |
| 19:14.23 | sarsaeol | Katty: not the appropriate channel |
| 19:14.35 | Katty | not the appopriate channel my rear end. |
| 19:14.48 | Katty | we talk about crazier things than /that/ here. |
| 19:15.39 | p3nguin | wtf911: Perhaps you didn't correctly configure asterisk. I assure you asterisk works with ipkall -- I use it every day. |
| 19:15.51 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v017-224.mobile.uci.edu) |
| 19:16.30 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 19:16.34 | sarsaeol | Katty: pm? |
| 19:16.55 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 19:17.11 | teknoprep | http://pastebin.com/Hb14NK09 <---- no nat being done to the to / from / contact headers |
| 19:17.12 | teknoprep | wtf |
| 19:19.10 | Katty | sure! |
| 19:19.31 | *** join/#asterisk garymc (~chatzilla@host109-148-254-122.range109-148.btcentralplus.com) |
| 19:20.09 | [TK]D-Fender | teknoprep, Where is the outbound call in there? |
| 19:20.39 | [TK]D-Fender | nvm, I see it |
| 19:20.59 | [TK]D-Fender | Contact: <sip:6102860788@10.41.71.8:5060> |
| 19:21.04 | [TK]D-Fender | clearly bad NAT settings |
| 19:21.43 | [TK]D-Fender | Reliably Transmitting (NAT) to 206.80.67.68:5060: <- and ITSPs should always be "nat=no" |
| 19:21.54 | [TK]D-Fender | Looks pretty much entirely wrong |
| 19:27.20 | teknoprep | ok give me one sec |
| 19:27.24 | teknoprep | i will look at this in a minute |
| 19:27.33 | teknoprep | i have another lurking issue that i will get a sip debug on |
| 19:27.44 | teknoprep | its an inbound issue? i just don't know how to route inbound calls |
| 19:28.21 | *** join/#asterisk drhonk (~drhonk@static.57.112.4.46.clients.your-server.de) |
| 19:28.25 | drhonk | good afternoon |
| 19:28.55 | [TK]D-Fender | teknoYour outbound settings are all wrong. |
| 19:29.06 | [TK]D-Fender | I did not say inbound. |
| 19:29.14 | teknoprep | http://pastebin.com/PQ7MnF9j |
| 19:29.24 | teknoprep | yeah this is a different server |
| 19:29.34 | teknoprep | i was just working on another server.. this servers outbound is working perfectly |
| 19:29.40 | teknoprep | as i have all the nat settings correct |
| 19:32.15 | teknoprep | here is my configs -> http://pastebin.com/zugrUQ9q |
| 19:36.35 | wtf911 | i just can't connect ipkall :( ....Really destroying SIP dialog '7afeb8562831b0d0545f3a175d637ef7@192.168.1.1' Method: REGISTER |
| 19:37.38 | teknoprep | [TK]D-Fender, any ideas ? |
| 19:42.00 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:44.54 | teknoprep | inbound-ezuce] |
| 19:44.54 | teknoprep | exten => _X.,1,Dial(SIP/${EXTEN}@bluecloud) |
| 19:45.25 | teknoprep | that was the issue? i had exten => _X.,1,Dial(SIP/bluecloud/${EXTEN}) |
| 19:46.26 | teknoprep | lol too bad the call hangs up after 7 seconds |
| 19:46.31 | teknoprep | lol wtf? always something |
| 19:49.50 | [TK]D-Fender | wtf911, You don't register to ipkall. |
| 19:50.01 | [TK]D-Fender | wtf911, You need to give them a fixed URI to call |
| 19:50.19 | wtf911 | i did? |
| 19:52.00 | wtf911 | http://pastebin.com/BsfdaMku |
| 19:55.24 | wtf911 | http://i.imgur.com/vZCwK.png |
| 19:57.29 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 20:01.11 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 20:01.52 | wtf911 | are there any good alternatives to ipkall? |
| 20:01.56 | wtf911 | free of course |
| 20:02.19 | [TK]D-Fender | wtf911, I don't see an attempt to register to ipkall... but you showed something you claim was an attempt |
| 20:02.30 | [TK]D-Fender | wtf911, the pieces don't add up. show complete debug next time |
| 20:02.46 | [TK]D-Fender | <wtf911> i just can't connect ipkall :( ....Really destroying SIP dialog '7afeb8562831b0d0545f3a175d637ef7@192.168.1.1' Method: REGISTER |
| 20:03.29 | wtf911 | you're welcome to VNC :D |
| 20:05.00 | wtf911 | Connected to Asterisk 1.6.2.22 currently running on unknown (pid = 3280) |
| 20:05.01 | wtf911 | unknown*CLI> core set verbose 10 |
| 20:05.01 | wtf911 | Verbosity was 0 and is now 10 |
| 20:05.01 | wtf911 | unknown*CLI> sip set debug on |
| 20:05.01 | wtf911 | SIP Debugging enabled |
| 20:05.01 | wtf911 | Really destroying SIP dialog '3b3a4fec45fe92334951bbde3d698375@192.168.1.1' Method: REGISTER |
| 20:05.02 | wtf911 | unknown*CLI> |
| 20:05.04 | teknoprep | [TK]D-Fender, could you look at thie please -> http://pastebin.com/GxpaVs2P |
| 20:05.06 | \khan | dear god |
| 20:05.11 | \khan | pastebin you suck |
| 20:05.13 | \khan | pastebin you sucka |
| 20:05.35 | wtf911 | with the exact settings i put in pastebin thats the entire messages i get from calling, waiting 30 seconds, busy tone |
| 20:05.56 | teknoprep | [TK]D-Fender, -> i have an issue with line 57 ? the contact header is a local ip address |
| 20:06.01 | wtf911 | vnc : wtf911.hopto.org pass : help :P |
| 20:06.23 | teknoprep | does anyone here know which setting allows me to change my external communications in SIP contact header to an external IP address ? |
| 20:06.56 | teknoprep | becuase the contact header has a local ip address.. my ITSP does not know whom to respond to |
| 20:07.09 | \khan | teknomega, http://www.voip-info.org/wiki/view/Asterisk+SIP+externip |
| 20:07.10 | \khan | err |
| 20:07.12 | \khan | teknoprep, http://www.voip-info.org/wiki/view/Asterisk+SIP+externip |
| 20:08.48 | teknoprep | i already have those options set |
| 20:09.47 | teknoprep | yet the contact header is still wrong |
| 20:09.55 | \khan | teknoprep, and you have nat=yes in your peer config? |
| 20:10.14 | teknoprep | no i have nat=no in my [appia] config |
| 20:10.20 | teknoprep | they are not behind a nat |
| 20:16.43 | [TK]D-Fender | teknoprep, Retransmitting #2 (no NAT) to 69.54.92.156:5060: |
| 20:17.15 | [TK]D-Fender | teknoprep, Contact: <sip:7175463004@192.168.1.246:5060> |
| 20:17.18 | [TK]D-Fender | still bad. |
| 20:19.04 | teknoprep | agreed |
| 20:19.13 | teknoprep | i don't know how to change it |
| 20:19.41 | teknoprep | i have all of my NAT settings set for externip= localnet= |
| 20:19.45 | teknoprep | nat=no on the peer settings |
| 20:23.10 | *** join/#asterisk acedia (~garland@unaffiliated/ffs) |
| 20:24.00 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 20:24.24 | teknoprep | i found it |
| 20:24.51 | teknoprep | there was an uncommented [pingtel] from the default config below [default].. and my nat settings were below that |
| 20:24.52 | teknoprep | lol |
| 20:25.58 | [TK]D-Fender | checkout time, BBIAB |
| 20:27.26 | wtf911 | so no ipkall alternatives? |
| 20:29.32 | teknoprep | wtf911? try nexvortex |
| 20:29.36 | teknoprep | i have had alot of success with them |
| 20:29.42 | teknoprep | plus they are cheap |
| 20:29.48 | teknoprep | if its not for business its the way to go |
| 20:29.58 | teknoprep | if you want a business solution i would recommend bandwidth.com or Appia |
| 20:30.16 | wtf911 | i wanted a free solution :D |
| 20:30.20 | teknoprep | appiaservices.com? bandwidth.com? they both have great tech support |
| 20:30.21 | teknoprep | oh |
| 20:30.24 | teknoprep | well sorry man |
| 20:31.36 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 20:35.46 | teknoprep | ok next question |
| 20:35.52 | teknoprep | i want reinvites to happen on my LAN |
| 20:36.05 | teknoprep | so audio works directly from endpoint to endpoint |
| 20:37.43 | *** join/#asterisk sp00kz (~ilubj00@unaffiliated/sp00kz) |
| 20:39.10 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 20:39.31 | *** join/#asterisk thehar (thehar@diddlebox.thehar.com) |
| 20:43.04 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 20:47.34 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 20:54.22 | p3nguin | wtf911: I gave you the EXACT DATA to put into your sip.conf for ipkall. What happened there? |
| 20:55.09 | p3nguin | There's very little wrong with ipkall. |
| 20:55.34 | p3nguin | Do you need me to hold you hand a little further? |
| 20:56.40 | wtf911 | it didn't work :( |
| 20:56.49 | p3nguin | You didn't follow my instructions. |
| 20:56.53 | p3nguin | So it's no wonder. |
| 20:57.04 | wtf911 | :o i used the pastebin you provided but no luck |
| 20:57.39 | wtf911 | would you mind VNC viewing me? |
| 20:57.40 | p3nguin | http://pastebin.com/SEqA1mip <---------- |
| 20:57.56 | p3nguin | Just follow my instructions. I don't need to watch you NOT following them. |
| 20:58.43 | p3nguin | Maybe I have to do the configuration for you? |
| 20:59.10 | wtf911 | the configuration for me? |
| 20:59.23 | p3nguin | Look at the pastebin I just gave you (again). |
| 20:59.31 | bdfoster | p3nguin, i have issues with ipkall all the time so yes, indeed there are some things wrong with their stuff |
| 20:59.52 | p3nguin | Copy the two sections. Paste them in place of your existing ipkall section. |
| 21:00.05 | wtf911 | check...will test (again) now |
| 21:00.17 | p3nguin | bdfoster: The only thing wrong with it is their failure to correctly send caller ID sometimes. |
| 21:00.34 | bdfoster | uh... no lol |
| 21:00.40 | p3nguin | Other than that, there isn't much that can go wrong with it. It is either working or not working. |
| 21:00.55 | bdfoster | i miss calls on a continual basis, and nothing hits the server so it's not my stuff |
| 21:01.25 | bdfoster | (when it does fail) |
| 21:01.25 | p3nguin | All service providers have a tendency to have failures. They just happen to have less redundancy and failover protection. |
| 21:01.46 | *** join/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 21:01.54 | p3nguin | s/nd fa/nd less fa/ |
| 21:01.56 | bdfoster | uh huh... so like i said, stuff goes wrong fairly often in my case, and im sure others feel the same |
| 21:02.18 | wtf911 | well i tried it again |
| 21:02.23 | p3nguin | I certain do not use ipkall for mission critical applications. |
| 21:02.26 | wtf911 | literally no messages when i call |
| 21:02.37 | bdfoster | neither do i, it's for conferencing mainly |
| 21:02.46 | bdfoster | (b.s.'ing conferences) |
| 21:03.05 | p3nguin | wtf911: Show me that you've done what I asked you to do. Pastebin "sip show peer ipkall" for me. |
| 21:05.36 | wtf911 | http://pastebin.com/1Fw43j9q |
| 21:06.28 | p3nguin | Okay, good. Now verify your DNS name that you configured in ipkall's interface. |
| 21:07.28 | wtf911 | http://baremetal.com/cgi-bin/dnsip?target=wtf911.hopto.org = wtf911.hopto.org -> 67.251.103.137 |
| 21:07.39 | wtf911 | Your public IP address is 67.251.103.137 |
| 21:08.06 | wtf911 | and i can make and recieve calls as is with voipms |
| 21:08.23 | wtf911 | when i connect xlite i connect through wtf911.hopto.org |
| 21:08.47 | p3nguin | And you've verified the name in ipkall? |
| 21:09.07 | p3nguin | voipms doesn't rely on your entry of the dns name. |
| 21:09.17 | p3nguin | It relies on your registration from asterisk. |
| 21:09.44 | p3nguin | You can't register to ipkall, though, so you have to be sure you put in the correct dns name or IP address for "SIP Proxy." |
| 21:10.17 | p3nguin | The SIP Phone Number can be arbitrary, but I recommend using the DID number. |
| 21:11.48 | *** join/#asterisk talntid (~talntid@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 21:11.56 | p3nguin | IPkall will deliver the call to you by URI. <SIP Phone Number>@<SIP Proxy> is the format. |
| 21:12.08 | p3nguin | So if your dns name is wrong, you'll never get the call. |
| 21:12.10 | wtf911 | http://i.imgur.com/XX2vZ.png |
| 21:13.35 | p3nguin | Seems right. |
| 21:13.53 | wtf911 | i'll ask to ask ....could port forwarding be a problem? it hasn't been before? |
| 21:14.07 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 21:14.19 | talntid | whats that teardrop icon in the taskbar? |
| 21:14.22 | p3nguin | Yes. You have to forward port 5060 UDP to asterisk. |
| 21:14.29 | wtf911 | rainmeter |
| 21:14.48 | p3nguin | You also have to forward your RTP range. The range is in rtp.conf, also UDP. |
| 21:15.05 | wtf911 | http://i.imgur.com/d9oCJ.png |
| 21:15.34 | wtf911 | what's a good way to check if the port is open? asterisk is running literally on my router lol |
| 21:15.44 | p3nguin | Check your port forwarding to enure it has been configured. |
| 21:15.53 | p3nguin | Oh, then it's not forwarding. |
| 21:15.59 | p3nguin | You forward to another host. |
| 21:16.05 | p3nguin | On that router, you just allow it. |
| 21:16.30 | p3nguin | If the port is allowed to reach asterisk, the call should arrive. |
| 21:16.41 | wtf911 | how do i know if it is? :/ |
| 21:16.58 | wtf911 | (i've port forwarded before to computers) i just am confused it being the router itself |
| 21:17.30 | p3nguin | If you are using iptables, check the INPUT chain. iptables -L INPUT -nv |
| 21:17.46 | wtf911 | as per a website... Port 5060 is closed on 67.251.103.137. |
| 21:17.58 | p3nguin | That tested TCP. |
| 21:18.03 | *** part/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 21:18.06 | p3nguin | You'll be using UDP. |
| 21:18.28 | p3nguin | You also have to make sure you aren't forwarding the ports. If you are forwarding the ports, that preempts acceptance on the router. |
| 21:18.52 | *** join/#asterisk acedia (~garland@unaffiliated/ffs) |
| 21:21.34 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 21:26.30 | wtf911 | well would it be possible on my routers port forwarding page to forward port 5060 to 192.168.1.1 |
| 21:26.34 | wtf911 | to open it up? |
| 21:28.21 | talntid | no |
| 21:28.26 | talntid | you don't need to do that |
| 21:28.43 | talntid | and IF it would work like that, you'd likely need to send it to 127.0.0.1 |
| 21:29.33 | *** join/#asterisk bdfoster_ (~bdfoster@unaffiliated/bdfoster) |
| 21:31.43 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:32.24 | *** join/#asterisk GameGamer43 (u5533@gateway/web/irccloud.com/x-wkyfusqnbbqqfpom) |
| 21:33.09 | wtf911 | http://pastebin.com/HGMURGVs - from iptables |
| 21:37.37 | p3nguin | wtf911: You are not allowing UDP 5060 (SIP port) into the router. |
| 21:39.15 | *** join/#asterisk Ice_Strike (Ice_Strike@94-192-112-241.zone6.bethere.co.uk) |
| 21:39.33 | Ice_Strike | on VPS, it is doing 10 concurrents call all the time |
| 21:39.49 | Ice_Strike | How do I monitor if VPS is handling well |
| 21:39.58 | Ice_Strike | eg: no packet lost |
| 21:40.10 | Ice_Strike | enough UDP allowance? |
| 21:44.00 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 21:46.05 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a05e1.pool.mediaWays.net) |
| 21:52.00 | *** join/#asterisk wwalker (~wwalker@208.92.232.27) |
| 21:53.16 | wwalker | Anyone know what SA stands for in the following SIP status: "503 Inbound SA Constraints Exceeded" Just trying to figure out if they are complaining about total ports or about call initiation rate |
| 21:54.51 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 21:57.13 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 22:02.35 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 22:03.00 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 22:03.33 | Dovid | what is the payload of a RTP packet packet? Meaning if say I would want to test something by sending an rtp packet say via php fsock what would the content of the packet be? |
| 22:10.02 | mpe | hi do any of you know if there is a asterisk 10 .deb packet available |
| 22:10.02 | mpe | I can only find 1.8.11.1 |
| 22:10.40 | mjordan | mpe: there are no debian packages of Asterisk 10 yet, we are working on correcting that. |
| 22:12.37 | mpe | ok thanks |
| 22:13.54 | [TK]D-Fender | Dovid: Google-able |
| 22:13.55 | mpe | do you know if the comunity are working on addind CentOS 6 |
| 22:15.32 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@190.166.21.0) |
| 22:16.15 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 22:17.48 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 22:20.22 | Ice_Strike | Would large fragmented UDP packets improve performance of voip calls? |
| 22:23.14 | [TK]D-Fender | large + broken = same size? |
| 22:23.29 | [TK]D-Fender | I'll take the jumbo-shrimp to go please.... |
| 22:25.19 | Ice_Strike | [TK]D-Fender Should I ask VPS provider to increase UDP allowance on my VPS account? |
| 22:25.42 | Ice_Strike | which may improve performance of voip call? |
| 22:26.20 | *** join/#asterisk Merlin (visi@evendata.net) |
| 22:27.01 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:27.03 | Merlin | is there a good source for centos6 rpms for asterisk 1.6? or should i compile from source? |
| 22:27.19 | Dovid | TK: May I PM? |
| 22:32.34 | Ice_Strike | Merlin centos6-mininal |
| 22:32.49 | Ice_Strike | CentOS 6 minimal install CD |
| 22:33.40 | Dovid | [TK]D-Fender: Don't you know that i am horrible @ googling? |
| 22:35.07 | *** part/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 22:35.36 | [TK]D-Fender | Dovid: No-one is going to hand-retype the RFC for you to read. the IETF is already handing that out. |
| 22:36.03 | [TK]D-Fender | iceyour terminology is vague and you're assuming bandwidth is the problem. |
| 22:36.32 | *** join/#asterisk Naikrovek (~sbrule@unaffiliated/naikrovek) |
| 22:41.06 | *** join/#asterisk mazpe (~lesterm@99-140-76-247.lightspeed.miamfl.sbcglobal.net) |
| 22:43.12 | Dovid | [TK]D-Fender: Ok. I guess I was looking for a sample packet that I could just use as opose to going through thousands of lines for a few k of payload |
| 22:46.36 | *** join/#asterisk Naikrovek (~sbrule@unaffiliated/naikrovek) |
| 22:52.16 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 22:52.50 | *** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net) |
| 22:56.20 | michael-i | I'm trying to find the right combo of Background() WaitExten() and GoTo() so incoming calls are dropped into an extension where the caller can enter their destination extension, that extension is validated as legit, then passed on |
| 22:57.09 | michael-i | Background doesn't "loop back" when the extension is invalid and I cannot grab the return data from Background() |
| 22:57.33 | WIMPy | That's what the i extension is there for. |
| 22:58.03 | michael-i | oh boy, let me peek at that…thanks |
| 22:58.34 | michael-i | I know this is basic stuff, just not finding my way for some reason today. |
| 23:10.12 | *** join/#asterisk acedia (~garland@unaffiliated/ffs) |
| 23:22.42 | *** join/#asterisk Janos (~cramos@186.15.71.141) |
| 23:23.19 | Janos | hey there, got a question, is there any way to print the data in the asterisk database outside of asterisk ? |
| 23:24.00 | WIMPy | asterisk -rx |
| 23:24.07 | WIMPy | or via ami |
| 23:24.16 | *** join/#asterisk acedia_ (~garland@unaffiliated/ffs) |
| 23:26.32 | Janos | WIMPy, that's an excellent answer for my question, but my question was not very valid :P. I have a backup of an asterisk server that no longer exists, i was wondering if there was a way to print this data without having to install asterisk |
| 23:27.31 | WIMPy | Oh, that might be tricky. |
| 23:28.09 | Janos | is this a berkely db ? gdbm ? or something asterisk cooked up ? |
| 23:29.50 | [TK]D-Fender | BDB <- |
| 23:29.58 | *** join/#asterisk TehRabbitt (~User@unaffiliated/tehrabbitt) |
| 23:30.02 | Janos | if this is a BDB i should be able to use the BDB client to read it, but i don't know which format this db is |
| 23:30.08 | [TK]D-Fender | PHP has libs for this as do many others |
| 23:37.09 | Janos | kk thanks a lot, later |
| 23:38.27 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 23:41.27 | *** join/#asterisk cyborg-one (1000@212-178-2-196.broadband.tenet.odessa.ua) |
| 23:48.49 | *** join/#asterisk s[x] (~sx]@eth589.qld.adsl.internode.on.net) |
| 23:49.03 | *** join/#asterisk acedia (~garland@unaffiliated/ffs) |
