| 00:00.42 | epaphus | Penguin, thank you./ |
| 00:11.41 | dlynes | great...the package maintainer decided it wouldn't be a wise idea to include that function |
| 00:20.01 | *** join/#asterisk aross42 (~aross@192-0-133-151.cpe.teksavvy.com) |
| 00:23.25 | *** join/#asterisk raspberrypifan (~raspberry@186.42.242.158) |
| 00:29.59 | dlynes | I really don't see how eval() is any different than not using eval() |
| 00:32.28 | paulc | I did something like that using cut and a loop counter. It wasn't the prettiest, but it worked just fine. |
| 00:39.24 | dlynes | paulc: looping through ${ARG1} ... ${ARGN} in a macro? |
| 00:42.06 | paulc | dlynes: it wasn't specifically in a macro, and I knew N, but it was variable |
| 00:53.05 | *** join/#asterisk zem_ (~krikkit@dh207-31-28.xnet.hr) |
| 00:54.48 | *** join/#asterisk sgriepentrog (~sgriepent@107-220-72-223.lightspeed.brhmal.sbcglobal.net) |
| 01:00.48 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 01:18.02 | *** join/#asterisk oli-werk (~oli-work@230.129.233.220.static.exetel.com.au) |
| 01:21.36 | *** join/#asterisk sgriepentrog (~sgriepent@107-220-72-223.lightspeed.brhmal.sbcglobal.net) |
| 01:22.43 | *** join/#asterisk sgriepentrog (~sgriepent@107-220-72-223.lightspeed.brhmal.sbcglobal.net) |
| 01:32.01 | *** join/#asterisk cyberfab007 (~cyberfab0@203.128.94.34) |
| 01:32.51 | cyberfab007 | hey anyone around? |
| 01:33.08 | cyberfab007 | someone have the trunk settings for connecting to a skype sip profile service |
| 01:36.12 | [TK]D-Fender | they do |
| 01:37.58 | [TK]D-Fender | its the same basic stuff as any other SIP account you'll ever setup. |
| 01:38.20 | cyberfab007 | I just chatted with skype tech support 5 times , and they keep telling me they dont provide I must go to the manufature of my pbx |
| 01:38.42 | Penguin | Don't ask them how to configure asterisk. |
| 01:38.43 | cyberfab007 | its funny I have install 100's of these systems and I never had a issue with interconnect like I am having with them |
| 01:38.46 | Penguin | They don't know. |
| 01:38.58 | cyberfab007 | I just ask the for the trunk settings |
| 01:39.00 | Penguin | But you do need to get your credentials and proxy information from them. |
| 01:39.03 | [TK]D-Fender | what do you see actually happening? |
| 01:39.07 | Penguin | You asked the wrong question. |
| 01:39.21 | [TK]D-Fender | probably several |
| 01:39.22 | cyberfab007 | anyways , if I put the sip info directly into the softphone it works fine |
| 01:39.36 | [TK]D-Fender | <[TK]D-Fender> what do you see actually happening? <- |
| 01:39.58 | cyberfab007 | well the trunk is registered , but I am get this during the call |
| 01:40.26 | [TK]D-Fender | ~pb |
| 01:40.26 | 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. |
| 01:40.27 | [TK]D-Fender | ^^^^ |
| 01:41.23 | cyberfab007 | http://pastebin.com/1Aw3R4Qc |
| 01:41.41 | cyberfab007 | come on I am not that new lol |
| 01:42.07 | [TK]D-Fender | You aren't looking at SIP debug .... so I remain unconvinced.... |
| 01:42.31 | [TK]D-Fender | try again.... |
| 01:42.37 | cyberfab007 | I have looked at sip debug |
| 01:42.39 | cyberfab007 | I will send |
| 01:46.34 | cyberfab007 | almost there |
| 01:47.09 | Penguin | I get a feeling we're going to see a paste with a bunch of info hidden or changed. |
| 01:48.13 | [TK]D-Fender | Step 1: Collect evidence. Step 2: screw with evidence. Step 3: begin investigation. Any questions? |
| 01:48.49 | cyberfab007 | no |
| 01:48.54 | cyberfab007 | dont be an ass please |
| 01:49.25 | cyberfab007 | ok here |
| 01:49.27 | cyberfab007 | http://pastebin.com/HxUEJz9j |
| 01:49.56 | cyberfab007 | line 701 you see the sip trace from the call being made |
| 01:50.10 | [TK]D-Fender | SIP/2.0 404 Not Found |
| 01:50.16 | [TK]D-Fender | they don't like what you dialed |
| 01:50.47 | [TK]D-Fender | From: "6001" <sip:6001@104.207.139.187>;tag=as7080bcae |
| 01:50.55 | [TK]D-Fender | or quite likely who you are claiming to BE |
| 01:51.00 | cyberfab007 | yegh see this is where it is werid cause when I put the sip info in the softphone it goes through fine |
| 01:51.05 | [TK]D-Fender | fromuser=YOURUSERNAME |
| 01:51.07 | [TK]D-Fender | ^ |
| 01:51.42 | [TK]D-Fender | Fix your peer |
| 01:51.46 | cyberfab007 | humm |
| 01:52.37 | phix | [TK]D-Fender: I am still having issues with incoming SIP calls. I need to reload asterisk every 5 minutes or any calls I make to the number ring out and dont hit the asterisk server |
| 01:53.06 | cyberfab007 | [TK]D-Fender, wait so your saying I have to add fromuser=SKYPEUSERNAME ? |
| 01:53.11 | phix | [TK]D-Fender: I messed with a few settings to try and fix it but all failed :( |
| 01:53.58 | Penguin | phix: Check your registration to your remote peer. |
| 01:56.16 | phix | sip show register? |
| 01:56.27 | Penguin | registry |
| 01:56.32 | Penguin | sip show registry |
| 01:57.24 | phix | yup, it is registered but I can't call it |
| 01:57.41 | Penguin | Time to look at sip debug when you try calling it. |
| 01:57.42 | phix | when I ring the number it dials out and asterisk doesn't see the call |
| 01:57.49 | phix | nothing is there |
| 01:57.59 | [TK]D-Fender | lets see it register |
| 01:57.59 | phix | It doesn't hit asterisk |
| 01:58.05 | pjensen00 | Hey, what's the size limit on a channel variable? |
| 01:58.14 | phix | [TK]D-Fender: ok |
| 01:58.22 | phix | can I force a reregister? |
| 01:58.27 | phix | sip reload? |
| 01:58.39 | Penguin | Just wait. It'll do it soon. |
| 02:02.01 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 02:03.24 | Penguin | If it hasn't given a REGISTER by now, that is likely the cause of your problem. |
| 02:07.02 | *** join/#asterisk cyberfab007 (~cyberfab0@203.128.94.34) |
| 02:07.15 | cyberfab007 | [TK]D-Fender, hey sorry got dissconnected |
| 02:07.21 | cyberfab007 | thank , that fixed things |
| 02:07.35 | cyberfab007 | I cant beleive skype tech support could not tell me that |
| 02:08.09 | Penguin | (2038.42) <Penguin> Don't ask them how to configure asterisk. |
| 02:08.11 | Penguin | (2038.46) <Penguin> They don't know. |
| 02:08.21 | Penguin | ^^^^^^^^^^^^^^^^^^^^ |
| 02:10.25 | *** join/#asterisk sgriepentrog (~sgriepent@107-220-72-223.lightspeed.brhmal.sbcglobal.net) |
| 02:13.44 | phix | Penguin, [TK]D-Fender: http://pastebin.com/13j7Xbz7 |
| 02:14.30 | [TK]D-Fender | [Mar 26 12:58:47] Contact: <sip:AAAAAAAAAA@192.168.2.99:5060> |
| 02:14.32 | [TK]D-Fender | ^ |
| 02:14.44 | [TK]D-Fender | Sending them a PRIVATE IP as your contact? |
| 02:14.45 | [TK]D-Fender | FAIL |
| 02:14.49 | Penguin | If Microsoft would have known how to configure asterisk, that would have totally blown my mind. |
| 02:14.55 | [TK]D-Fender | Another screwed up NAT setup |
| 02:15.05 | Penguin | ~sipnat |
| 02:15.06 | 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. |
| 02:15.21 | Penguin | nat, directmedia, externhost or externaddr, and localnet. |
| 02:15.23 | phix | [TK]D-Fender: I have set externalip |
| 02:15.30 | [TK]D-Fender | <[TK]D-Fender> Step 1: Collect evidence. Step 2: screw with evidence. Step 3: begin investigation. Any questions? <--- this was sarcasm ... not a guide |
| 02:15.33 | Penguin | externalip not valid. |
| 02:15.37 | [TK]D-Fender | phix, You did the rest wrong |
| 02:15.41 | Penguin | externhost or externaddr |
| 02:15.47 | *** join/#asterisk CeBe (~CeBe@xd9baf947.dyn.telefonica.de) |
| 02:15.48 | phix | I have those set too |
| 02:15.53 | Penguin | one or the other |
| 02:15.54 | [TK]D-Fender | You've done it wrong. |
| 02:15.54 | Penguin | not both |
| 02:16.42 | phix | ah crap, I know what is wrong |
| 02:16.58 | Penguin | You didn't set the NAT setting on the remote side? |
| 02:17.02 | phix | the asterisk server is doing a DNS lookup :/ |
| 02:17.13 | [TK]D-Fender | HOSTS file .. right? |
| 02:17.39 | Penguin | split DNS, maybe |
| 02:17.40 | phix | No, the hostname I have set for the external address resolves to a local address within the network (because cisco is shit) |
| 02:18.01 | phix | if you look it up externally it resolves to the correct address |
| 02:18.19 | phix | I guess I can set the static IP there |
| 02:18.27 | [TK]D-Fender | http://tinyurl.com/nmkykt3 |
| 02:18.39 | Penguin | If you have a static address, use externaddr. |
| 02:18.48 | phix | thnx [TK]D-Fender |
| 02:19.13 | [TK]D-Fender | NEXT @!!!@@! (c) bkw |
| 02:19.22 | phix | I didn't know the asterisk server did a lookup, I thought it just sent the hostname as is to the sip provider |
| 02:19.51 | Penguin | If you can enable the NAT setting on the remote side, it wouldn't matter what you send. |
| 02:20.02 | phix | outside_addr? |
| 02:20.13 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 02:20.23 | Penguin | I'll rephrase. |
| 02:20.37 | phix | I am using version 11 |
| 02:20.39 | Penguin | If your provider offers a NAT setting that you can toggle, you could fix the problem. |
| 02:21.27 | phix | hmmm I am not sure |
| 02:21.57 | Penguin | If your ITSP uses asterisk and if they give you control over your peer settings, you could enable NAT on their side and then it won't matter what is inside the packets you send them... because they'll ignore it and use the public address that your packets come from. |
| 02:22.14 | Penguin | That's how our nat settings in asterisk work with phones behind a remote NAT. |
| 02:22.34 | Penguin | To your provider, your asterisk may as well be a phone. |
| 02:23.01 | [TK]D-Fender | users are typically respnsible for their own systems. |
| 02:23.17 | phix | no I think my provider is using ciscos |
| 02:23.26 | phix | *shrugs* |
| 02:23.39 | phix | It should say in that paste :) |
| 02:23.44 | Penguin | Try to fix the problem on your end and forget about whatever they're using. |
| 02:24.20 | phix | so is outside_addr deprecated too? |
| 02:24.44 | Penguin | Never heard of it. |
| 02:25.04 | phix | and do I set externaddr in global or in the peer setting? |
| 02:25.12 | Penguin | general |
| 02:25.25 | phix | and i won't interfere with internal SIP phones?? |
| 02:25.28 | phix | io = it |
| 02:25.31 | Penguin | localnet |
| 02:25.53 | Penguin | I'll re-post... |
| 02:25.55 | phix | I have localnet set |
| 02:25.57 | Penguin | ~sipnat |
| 02:25.57 | 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. |
| 02:26.13 | *** join/#asterisk pEYEd (~kvirc@ool-43561899.dyn.optonline.net) |
| 02:26.16 | phix | Penguin: 404 ? Fancy meeting you here! |
| 02:26.33 | phix | The 2nd link works though |
| 02:26.38 | Penguin | Ignore the URL and read what is in the paragraph. |
| 02:27.08 | phix | which para |
| 02:27.12 | Penguin | Starts with, "Typically, you MUST configure..." |
| 02:27.17 | phix | ok |
| 02:27.22 | pEYEd | how do you setup a forward from the trixbox cli? (I know, I know, this is a rescue while one is being built) |
| 02:27.37 | Penguin | ~trixbox |
| 02:27.38 | infobot | Delving into Trixbox is like exploring a pyramid; it's ancient, forgotten, dark, and dangerous. Trixbox was one of the earliest complete PBX distros and a relic of a bygone era. While it was a great idea, it was implemented by a horrible group of Wizards from an evil, barren wasteland that stuffed it full of black magic and FUD. Also, an example of how not to run a business. |
| 02:28.09 | Penguin | ~freepbx |
| 02:28.09 | 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 |
| 02:31.24 | phix | Penguin: ok so I have set localnet=10.0.0.0/8, externaddr=XXX.XXX.XXX.XXX, nat=yes and directmedia=no under [general] is that all? |
| 02:31.46 | pEYEd | Penguin: I am trying to bypass trixbox everything and just hit it with pure asterik. |
| 02:31.48 | Penguin | Should be good for the general part. |
| 02:32.08 | Penguin | Give it a sip reload and watch the debug for REGISTER. |
| 02:32.20 | phix | nat=force_rport,comedia |
| 02:32.32 | Penguin | You said you set nat=yes. |
| 02:32.46 | Penguin | yes means force_rport,comedia |
| 02:33.09 | Penguin | Now wait just a minute... |
| 02:33.41 | Penguin | You set localnet to 10.0.0.0/8, but [tk]d-fender spotted you sending 192.168.2.99 |
| 02:33.49 | Penguin | Something is amiss. |
| 02:35.35 | Penguin | peyed: "setup a forward" <------ explain this to me and I'll see if I can help you do it with pure asterisk. |
| 02:39.32 | phix | Penguin: yes, I have multiple localnet lines, there is one for 192.168.0.0/16 |
| 02:40.05 | phix | Penguin: ok I have re-registered and it is using the external IP |
| 02:40.14 | phix | now to test it out :S |
| 02:42.10 | pEYEd | Penguin: this is the inbound trunk info. I am trying to bypass trixbox everything and forward calls to a given number https://bpaste.net/show/5fdc67538db7 |
| 02:42.26 | *** join/#asterisk Zer0legend (~ZerOlegen@72-160-218-223.dyn.centurytel.net) |
| 02:42.27 | phix | I have stopped crontab jobs which reloads asterisk, the issue usually presents itself after 5 minutes, so I will try dialing it in 10 |
| 02:42.28 | *** join/#asterisk mjordan (mjordan@nat/digium/x-yslzngjrfeprqyxe) |
| 02:42.28 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:43.22 | Penguin | peyed: Direct your inbound calls to an extension which dials that given number. |
| 02:43.42 | pEYEd | Penguin: yes |
| 02:44.05 | Penguin | exten => _X.,1,Dial(SIP/otherhost/55512345) |
| 02:44.44 | Penguin | What you showed me in that paste isn't enough for me to do anything with it. |
| 02:46.30 | Penguin | How do you make contact to the "number" that you're trying to send all calls to? |
| 02:46.40 | pEYEd | Penguin: but how would I catch just that one DID? I apologize, I am trying to get more trunk and context information for you now. |
| 02:47.15 | Penguin | Define that as the extension where I used a pattern. |
| 02:47.24 | pEYEd | nm, what post posted just clicked. |
| 02:47.25 | Penguin | Instead of _X. change it to whatever DID number. |
| 02:48.16 | pEYEd | Penguin: do I need to put that in a particular context in extensions.conf? |
| 02:48.30 | Penguin | Yes |
| 02:48.43 | Penguin | Your peer has a context assigned to it already... |
| 02:48.59 | Penguin | Your DID is probably already an extension in that context. |
| 02:49.26 | Penguin | You can either change the context to a new one for your purposes, or you can change the existing extension. |
| 02:49.48 | pEYEd | Penguin: is the context pbx_config? and I reading that correctly? |
| 02:50.11 | *** join/#asterisk raspberrypifan (~raspberry@186.42.242.158) |
| 02:51.09 | Penguin | No. In your peer definition in sip.conf, there is a parameter "context" which is where a call goes when it matches that peer entry. |
| 02:51.21 | pEYEd | am I reading it correctly? o.0 all thumbs |
| 02:51.51 | *** join/#asterisk cyford_q (~cyford@c-73-207-183-115.hsd1.ga.comcast.net) |
| 02:52.52 | pEYEd | Penguin: the dialplan is in mysql, which right now I do not have access to (about to hack a lost mysql root pass). Is there a way to dump the context definitions via the cli? |
| 02:53.41 | Penguin | You want to see the contexts in dialplan or you want to see what context is assigned to your peer? |
| 02:54.26 | pEYEd | Penguin: context assigned to a peer. and then filter the dialplan for just those peer definitions. |
| 02:54.51 | Penguin | sip show peer <the name of the peer as configured in sip.conf> |
| 02:57.44 | pEYEd | Penguin: no peers under sip show peers 0.o |
| 02:58.02 | Penguin | Then you have no peers configured. |
| 02:58.12 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 02:58.26 | pEYEd | nm, I forgot the s |
| 02:58.51 | Penguin | tab key |
| 02:59.17 | Penguin | sip show peerSPACETABKEY |
| 03:03.13 | pEYEd | Penguin: looks like from-sip-external https://bpaste.net/show/9ad1064f2c72 |
| 03:03.49 | Penguin | So then you can try dialplan showSPACEBARTABKEY |
| 03:03.59 | Penguin | or dialplan show fromTABKEY |
| 03:04.24 | Penguin | or type it all out if you feel that your "all thumbs" problem has dissipated. |
| 03:05.25 | Penguin | I'm tabkey crazy. I do diTAB shTAB blahTAB |
| 03:06.17 | phix | <3 |
| 03:06.23 | Penguin | But... I'm a penguin and I have lived in a Linux shell for many years. |
| 03:08.05 | phix | hmmm, time for a new keyboard?? --> sup shos registry |
| 03:08.13 | Penguin | haha |
| 03:08.23 | Penguin | siTAB shTAB reTAB |
| 03:08.49 | Penguin | actually rTAB is enough there. |
| 03:09.01 | phix | <3 CTRL+R |
| 03:09.05 | phix | (in bash) |
| 03:09.25 | phix | not sure what the cycle command is though |
| 03:18.02 | pEYEd | Penguin: is this what i need? https://bpaste.net/show/73ab79c9c968 |
| 03:19.59 | Penguin | no |
| 03:20.01 | Penguin | all wrong |
| 03:20.15 | [TK]D-Fender | the entire idea is all wrong |
| 03:22.09 | Penguin | If I had someone/something sending a call to my asterisk that I wanted to in turn send some other place, I would make an extension for the number I wanted to send elsewhere. exten => DIDNUMBER,priority,Dial(SIP/differentPeer/targetnumber) |
| 03:22.47 | Penguin | exten => 8667013113,1,Dial(SIP/asterisk2/3149691077) |
| 03:23.14 | phix | Penguin: if I set externaddr in general my internal SIP phones fail to register |
| 03:23.28 | Penguin | This will send a call which is going to DID number 8667013113 to 3149691077 via peer named asterisk2 |
| 03:24.40 | *** join/#asterisk pEYEd (~kvirc@ool-43561899.dyn.optonline.net) |
| 03:24.43 | phix | Penguin: how can I set the externaddr for register only and not my internal sip phones? |
| 03:24.51 | pEYEd | oops, my inet borked. |
| 03:24.52 | [TK]D-Fender | phix, And are you doing something suicidal like pointing your internal phones that EXTERNAL address? |
| 03:25.03 | Penguin | externaddr has nothing to do with your internal phones. |
| 03:25.12 | phix | [TK]D-Fender: I am setting it in general so it is applying to everything |
| 03:25.18 | [TK]D-Fender | no... |
| 03:25.23 | Penguin | That's a negative. |
| 03:25.24 | [TK]D-Fender | I asked where you pointed your PHONES At |
| 03:25.29 | phix | well I set it and I cant dial in |
| 03:25.52 | phix | If I set externaddr I get an engaged signal |
| 03:25.55 | [TK]D-Fender | ..... |
| 03:25.59 | Penguin | What is the proxy address that you set on your phones? |
| 03:26.09 | phix | 192.168.2.99 |
| 03:26.12 | Penguin | I don't even know what an engaged signal is. |
| 03:26.20 | phix | Penguin: hangup() |
| 03:27.59 | [TK]D-Fender | phix, Where is the debug? |
| 03:28.00 | pEYEd | Penguin: how would I forward the calls from that did? |
| 03:29.45 | Penguin | I just told you. |
| 03:31.12 | pEYEd | Penguin: My inet borked, I don't think I got that post my friend. |
| 03:31.24 | [TK]D-Fender | <Penguin> exten => 8667013113,1,Dial(SIP/asterisk2/3149691077) |
| 03:31.38 | Penguin | Oh, okay... |
| 03:31.41 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 03:32.06 | Penguin | I wanted to ask you how you contact that target number. I don't know if I asked or not. |
| 03:32.18 | Penguin | How do you reach it? Do you have another peer to dial out of? |
| 03:33.46 | oli-werk | [TK]D-Fender, if you dont mind me asking... how do you have so much time to help out in here? |
| 03:33.48 | pEYEd | Penguin: there are probably 30 dids that come in from that 1 peers. |
| 03:33.49 | oli-werk | are you an asterisk dev? |
| 03:34.06 | [TK]D-Fender | oli-werk, nope |
| 03:34.06 | Penguin | And that's why you specify only the single number you care about. |
| 03:34.22 | pEYEd | Penguin: yes |
| 03:34.27 | Penguin | yes |
| 03:34.36 | pEYEd | all others I need to leave alone. |
| 03:34.43 | Penguin | yes |
| 03:35.37 | pEYEd | [TK]D-Fender: thank you! |
| 03:35.47 | *** join/#asterisk king1337-2 (~king1337@S0106c8fb2641d848.vs.shawcable.net) |
| 03:36.57 | Penguin | I get irritated when I ask questions trying to reach a solution and my questions go unanswered. |
| 03:39.16 | pEYEd | Penguin: my wireless is running like butt crack. big huge time out delays. I sincerely apologize. |
| 03:48.57 | phix | [TK]D-Fender: there is none, it doesn't hit the asterisk server |
| 03:49.24 | [TK]D-Fender | phix, No setting in Asterisk will magically make packets stop REACHING your server |
| 03:49.38 | [TK]D-Fender | phix, You have screwed something else up in that case |
| 03:49.50 | phix | [TK]D-Fender: then the remote party is sending packet to the wrong port or ignoring my setting |
| 03:49.52 | Penguin | Oh come on... we all know networking is smoke and mirrors. |
| 03:49.57 | [TK]D-Fender | phix, Rejecting the request is one thing. Seeing nothing is another |
| 03:50.07 | phix | It is accepting the register |
| 03:50.17 | phix | I just cant make calls to the number |
| 03:50.18 | [TK]D-Fender | <[TK]D-Fender> phix, Where is the debug? |
| 03:50.33 | [TK]D-Fender | I don't see a register |
| 03:50.45 | [TK]D-Fender | or a call attempt immediately afterwards |
| 03:50.57 | [TK]D-Fender | even better : AND |
| 03:51.04 | phix | exactly, it isnt hitting the server |
| 03:51.49 | [TK]D-Fender | Not your server's issue |
| 03:51.57 | [TK]D-Fender | (asterisk itself that is) |
| 03:52.14 | [TK]D-Fender | You're either pointing the phone wrong, or it's getting filtered along the way |
| 03:52.31 | [TK]D-Fender | You said it's registering, so where is that to look at? |
| 03:52.32 | Penguin | I'm getting the feeling the big picture wasn't painted accurately. |
| 03:53.06 | ChannelZ | "I want to make calls. NSA says otherwise." |
| 04:14.41 | phix | [TK]D-Fender: I pasted it above, just assume 192.168.2.99 is an external address :) because everything else is the same (I diffed it) |
| 04:14.54 | phix | (oh and the times are different too obviously) |
| 04:15.01 | [TK]D-Fender | I assume nothing |
| 04:15.07 | [TK]D-Fender | Which is why I find mistakes |
| 04:15.08 | Penguin | You didn't paint the picture very well. |
| 04:15.22 | [TK]D-Fender | The less you show, the less I trust |
| 04:15.31 | Penguin | Your complaint was that your asterisk wasn't receiving calls from your ITSP. |
| 04:15.48 | Penguin | Now your complaint is that your phones don't talk to your asterisk. |
| 04:16.31 | Penguin | My assumption is that you're using a DNS name in the phones rather than the IP address you claimed to be using. |
| 04:16.38 | [TK]D-Fender | Asterisk never tells a phone where to call it. |
| 04:16.47 | [TK]D-Fender | This whole line of thinking is defective. |
| 04:16.53 | Penguin | And then you changed the DNS name to be the external address instead of the internal address that it used to be. |
| 04:17.06 | Penguin | But this is pure speculation since you showed nothing. |
| 04:28.39 | phix | [TK]D-Fender: ok |
| 04:28.51 | phix | Penguin: no no it is ITSP still |
| 04:29.26 | phix | ok I will get you some debugs |
| 04:56.26 | Penguin | falls asleep waiting |
| 04:58.59 | *** join/#asterisk [[thufir]] (~thufir@96.48.128.162) |
| 05:02.26 | *** join/#asterisk timahvo1 (~rogue@196.216.71.122) |
| 05:03.49 | [[thufir]] | [TK]D-Fender: I got extension 201 to call 202 successfully :) |
| 05:04.44 | [TK]D-Fender | [[thufir]], And at this point I don't remember your situation at all... |
| 05:05.17 | [[thufir]] | [TK]D-Fender: that's ok, you don't have to. I just want to say thanks for all your help and patience. |
| 05:06.30 | phix | Feel the <3 |
| 05:16.21 | *** join/#asterisk zapata (~zapata@2001:470:1f0b:11bc:c5f5:cb7e:970e:f0a4) |
| 05:44.01 | oli-werk | when a caller is waiting to be connected to the called party |
| 05:44.10 | oli-werk | during a Dial() function |
| 05:44.34 | oli-werk | is there a way to play music on hold or ringing to the caller |
| 05:44.58 | [TK]D-Fender | it should ring be default. |
| 05:45.18 | [TK]D-Fender | If it doesn't then something is wrong... for the rest "cosre show application dial" <--- |
| 05:45.45 | oli-werk | it's silent :( |
| 05:46.37 | oli-werk | exten => _555.,1,NoOp() |
| 05:46.37 | oli-werk | <PROTECTED> |
| 05:47.56 | [TK]D-Fender | Hrm.. not sure if that's because a local channel can't report rining back as a state directly like that... |
| 05:48.08 | [TK]D-Fender | you COULD Answer first.. but that might make CDR look confusing, etc. |
| 05:48.11 | [TK]D-Fender | Your call... |
| 06:06.35 | *** join/#asterisk keiserr (~selven@unaffiliated/keiserr) |
| 06:14.50 | *** join/#asterisk miha- (~uporabnik@tonttu.softnet.si) |
| 06:15.20 | miha- | hi is it possible to use to external ip addresses on asterisk if you have two network providers for two sip trunks? |
| 06:16.00 | *** join/#asterisk muanang (~muanang@203.115.183.18) |
| 06:16.09 | muanang | finally, im in! |
| 06:17.51 | muanang | Hi! Can I post questions here? |
| 06:18.29 | muanang | Just having problem with our asterisk setup. |
| 06:18.56 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
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| 06:19.52 | muanang | I can hear myself talking when calling outside. |
| 06:20.10 | muanang | can someone lighten me up where to start checking? |
| 06:20.14 | muanang | thank you! |
| 06:20.34 | muanang | btw, im using digium tdm800p |
| 06:22.30 | *** join/#asterisk carrar (~tim@osburn.jp) |
| 06:22.47 | carrar | pokes Katty |
| 06:22.53 | carrar | probably sleepin |
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| 07:05.23 | muanang | hi, anyone with the echo prob? |
| 07:10.53 | [sID] | hi, |
| 07:17.01 | muanang | hey |
| 07:27.59 | ChannelZ | teh echo |
| 07:28.00 | ChannelZ | I hears it |
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| 07:28.31 | ChannelZ | You should run fxotune |
| 07:28.54 | ChannelZ | I have a link to an old but good writeup about it somewhere, let me see if I can find it |
| 07:31.34 | ChannelZ | well, this isn't it, but it will get you the basic procedure: http://kb.digium.com/articles/Troubleshooting/how-to-basic-fxotune |
| 07:32.14 | ChannelZ | The first important part is making sure your gains are set properly, which is also the hardest part |
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| 07:50.53 | ChannelZ | damnit I wish I could find the writeup I'm thinking of. It was quite instructive. |
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| 09:01.25 | muanang | Thanks ChannelZ! |
| 09:01.52 | muanang | i'll check it now. |
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| 09:14.15 | jzu_ | uhh |
| 09:14.15 | jzu_ | freepbx*CLI> sip show registry |
| 09:14.15 | jzu_ | Host dnsmgr Username Refresh State Reg.Time |
| 09:14.18 | jzu_ | sip.sonetel.com:5060 N jussi@jus.si 120 Unregistered |
| 09:14.21 | jzu_ | how the hell I can debug that |
| 09:14.44 | jzu_ | the Asterisk that is trying to make the conenction to sip.sonetel.com is behind NAT |
| 09:15.00 | jzu_ | freepbx*CLI> sip show peers |
| 09:15.00 | jzu_ | Name/username Host Dyn Forcerport Comedia ACL Port Status Description |
| 09:15.03 | jzu_ | 100/100 192.168.1.8 D No No A 5060 OK (104 ms) |
| 09:15.06 | jzu_ | Sonetel/jussi 212.72.62.115 Yes Yes 5060 Unmonitored |
| 09:15.09 | jzu_ | peers are OK |
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| 09:17.09 | jzu_ | this is bizzare |
| 09:18.47 | jzu_ | this is really strange, after reboot it's again connected |
| 09:18.48 | jzu_ | Host dnsmgr Username Refresh State Reg.Time |
| 09:18.51 | jzu_ | sip.sonetel.com:5060 N jussi@jus.si 105 Registered Thu, 26 Mar 2015 11:18:06 |
| 09:18.54 | jzu_ | 1 SIP registrations. |
| 09:18.54 | jzu_ | had to reboot the whole Linux running Asterisk |
| 09:20.39 | jzu_ | https://issues.asterisk.org/jira/browse/ASTERISK-17880 - seems to be this but it should already be fixed |
| 09:42.32 | *** join/#asterisk yun1989 (5f5c784c@gateway/web/freenode/ip.95.92.120.76) |
| 09:46.19 | pEYEd | jzu_: ever heard of https://bpaste.net/ ? |
| 09:46.25 | pEYEd | :) |
| 09:54.45 | jzu_ | pEYEd: nah, I'm oldskool :) |
| 09:55.00 | jzu_ | but yeah uh - funny thing, if I reload Astreisk configuration the peer dies |
| 09:55.07 | jzu_ | and I need to reboot the whole host ' |
| 09:58.21 | pEYEd | jzu_: which os? |
| 10:00.23 | pEYEd | pastings which span more than one line are considered bad. it reaps havoc on text based irc clients such as epic5 which I use. use bpaste for configs and outputs. |
| 10:01.38 | jzu_ | pEYEd: FreePBX distro, some form oh CentOS I guess :) |
| 10:01.42 | jzu_ | pEYEd: so, Linux |
| 10:02.02 | jzu_ | yeah I've been using Irssi since 1999 or so |
| 10:05.29 | pEYEd | jzu_: post the output of (on bpaste) -> rpm -qa --last |
| 10:08.32 | *** join/#asterisk danjenkins_ (anonymous@nat/digium/x-zczxrwnyifiyjwip) |
| 10:09.36 | jzu_ | pEYEd: http://jus.si/rpm.txt |
| 10:11.49 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 10:11.50 | pEYEd | run tcpdump -w mycap.pcap , then restart asterisk, making the connection die and then post the file |
| 10:14.53 | jzu_ | well harhar, now it just doesn't want to die when I run tcpdump :-D |
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| 11:19.14 | snadge | im gonna ask in here because you guys are the most knowledgeable guys in the universe ;) |
| 11:20.07 | snadge | i stuffed around for ages getting a cisco 7941G to work.. and finally after the right magic incantations.. it registers with tcp transport |
| 11:21.14 | snadge | UDP should work.. but doesnt seem to.. it seems to use a REFER packet (whatever that means) which gets truncated.. and the registration process doesn't finish |
| 11:23.52 | snadge | the obvious solution is to a) not use EOL cisco phones.. b) use tcp.. except a good percentage of our legacy servers don't support tcp |
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| 11:30.06 | Katty | carrar: yes, sleep is good |
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| 11:40.21 | Katty | carrar: I should be around in about 2 hours |
| 11:41.16 | jzu_ | I had funny problem with Cisco SIP firmware P0S3-08-3-00 yesterday - 7960G would reload itself within 10-15 minutes |
| 11:41.49 | jzu_ | I upgraded to P0S3-8-12-00 and the problem was magically gone |
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| 12:29.48 | snadge | hmm.. im using 9.4 |
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| 12:44.04 | jzu_ | what exactly does erase protflash do on Cisco 79xx phones? |
| 12:44.25 | jzu_ | I did erase protflash remotely and now the shows as unprovisioned :O |
| 12:44.29 | jzu_ | +phone |
| 12:46.21 | carrar | *Y*A*W*N* must get espresso |
| 12:46.52 | jzu_ | Also, erase protflash will remove all the setting stored in flash memory, forcing the phone to reget the settings from TFTP. Much better than doing a reboot to reload the config files. |
| 12:47.02 | jzu_ | so yes - it doesn't know the IP of TFTP server anymore etc :) |
| 12:47.12 | jzu_ | no wonder it wasn't able to come up online |
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| 13:02.20 | Katty | ASPARTAME. |
| 13:04.19 | [TK]D-Fender | asperWILD! |
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| 13:06.51 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:40.23 | carrar | !! |
| 13:40.34 | carrar | getting espresso'ed up |
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| 14:37.22 | mikealeonetti | What's the best way to set the file permissions on an recording that Asterisk makes to ensure that it's 0660? Or is there an option that automatically sets Asterisk's output files to 0660? |
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| 14:46.04 | yun1989 | hello all |
| 14:46.19 | yun1989 | i have one problem to configure one gsm gateway in my asterisk |
| 14:46.47 | yun1989 | i can't make the outgoing call but i don't receive call's from gsm |
| 14:47.17 | yun1989 | how can show me one example of this ? |
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| 15:02.05 | [TK]D-Fender | yun1989: You ask for an example and you aren't even telling us what you are USING. You also haven't show us a working call going out, or an attempt to debug an incoming call. |
| 15:07.39 | marceloamorim | guys, could I have both cdr and cel working at the same time on asterisk 13.2.0? |
| 15:08.09 | [TK]D-Fender | Yes. It isn't an either/or |
| 15:11.01 | marceloamorim | ok, thx, I'll figured out why isn't work at both |
| 15:12.19 | yun1989 | @[TK]D-Fender http://www.voip-info.org/wiki/view/How+to+connect+VoIP+GSM+gateway+to+Asterisk+PBX |
| 15:12.23 | yun1989 | i see this link |
| 15:12.33 | [TK]D-Fender | yun1989: Don't point to a guide. Show us what YOU did. |
| 15:15.04 | yun1989 | in my dialplan for outgoings call I have one rule |
| 15:15.57 | yun1989 | exten => _9.,1, Dial(SIP/${EXTEN}@10001) |
| 15:16.13 | yun1989 | and 10001 is the extension for Gsm Gateway |
| 15:16.40 | yun1989 | and in sip.conf |
| 15:17.00 | yun1989 | i configure the extension 10001 |
| 15:17.11 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-hxxtwrxsjzlnlioz) |
| 15:18.06 | [TK]D-Fender | ~pb |
| 15:18.06 | 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. |
| 15:18.08 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 15:18.14 | [TK]D-Fender | Show us the ACTUAL CALLS |
| 15:18.21 | Katty | i'll show you a call |
| 15:18.24 | Katty | RIGHT HERE, RIGHT NOW |
| 15:18.29 | yun1989 | http://pastebin.com/bvPJPCmL |
| 15:19.18 | [TK]D-Fender | Katty: https://www.youtube.com/watch?v=2rlQqWbp7rY |
| 15:19.29 | Katty | this better not be banana phone |
| 15:19.36 | Katty | oh good ^_^ |
| 15:19.57 | [TK]D-Fender | yun1989: THE CALL. That is a piece of dialplan. That tells us nothing. Show us A CALL. |
| 15:21.05 | yun1989 | one moment please |
| 15:24.09 | *** join/#asterisk krefik (krefik@arm.generacja.pl) |
| 15:24.13 | krefik | hello |
| 15:27.17 | mikealeonetti | Still love this channel. |
| 15:28.26 | file | falls over |
| 15:38.19 | *** join/#asterisk raspberrypifan (~raspberry@186.42.216.233) |
| 15:45.23 | mikealeonetti | So, Asterisk saves files as permission 0644 when they are recorded, is there an option to set the default on that or change afterwards? |
| 15:48.17 | mikealeonetti | I guess I can try using the system option. |
| 15:48.38 | mikealeonetti | (That's a good place to start.) |
| 15:48.44 | mikealeonetti | Awesome. Love you guys! |
| 15:51.38 | wdoekes | umask? |
| 15:53.05 | pjensen00 | I love you too Mike. |
| 15:55.09 | yun1989 | @[TK]D-Fender http://tinypic.com/view.php?pic=9k70c3&s=8#.VRQrwfmsVYw |
| 15:55.35 | yun1989 | I see it when I try to make a call from gsm to sip extension |
| 15:57.20 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 16:01.06 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 16:02.28 | [TK]D-Fender | yun1989: It is telling you exactly what it is looking for in the dialplan to process taht call ... and it is not there |
| 16:02.57 | [TK]D-Fender | yun1989: Go makee that extension |
| 16:04.43 | yun1989 | but in 10001 i configure the context = default ? |
| 16:06.02 | *** join/#asterisk aui_ (591a2a31@gateway/web/freenode/ip.89.26.42.49) |
| 16:06.29 | [TK]D-Fender | the EXTENSION it is looking for is not there |
| 16:06.34 | aui_ | Hi there! I have one question: |
| 16:07.02 | [TK]D-Fender | yun1989: to 'SOMETHING' rejected <-- you don't have an extension for 'SOMETHING' |
| 16:08.44 | aui_ | I'm playing with a grandstream phone which supports "preview", which is working with directmedia=yes; but not with directmedia=no; does someone know if this is really possible without directmedia? |
| 16:08.52 | yun1989 | but i need to establish a call first |
| 16:09.02 | yun1989 | ? |
| 16:10.49 | [TK]D-Fender | yun1989: that IS a call coming in. And it is getting rejected because you didn't make an extension to PROCESS it. |
| 16:11.31 | yun1989 | I configure the "public" section in dialplan with the exten => s,1, Dial(SIP/10001) |
| 16:11.51 | [TK]D-Fender | yun1989: You are sending the call to [default]. It is TELLING you that is where it is going. |
| 16:12.10 | [TK]D-Fender | that is a call FROM SIP/10001 |
| 16:12.14 | [TK]D-Fender | context = default <------------------- |
| 16:12.19 | [TK]D-Fender | that is where calls FROM it go |
| 16:12.32 | *** join/#asterisk raymondhim (~raymondhi@c-66-41-216-194.hsd1.mn.comcast.net) |
| 16:20.19 | yun1989 | http://tinypic.com/r/21bqipu/8 |
| 16:20.35 | yun1989 | @[TK]D-Fender in other execution of the call I see it |
| 16:20.52 | [TK]D-Fender | yun1989: PASTEBIN another call. do not do a screen-cap. I want this in a pastebin as straight full text with nothing cut off. |
| 16:20.56 | [TK]D-Fender | No more screen pics |
| 16:20.59 | [TK]D-Fender | PASTEBIBN <----- |
| 16:21.01 | [TK]D-Fender | ~pb |
| 16:21.01 | 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. |
| 16:21.03 | [TK]D-Fender | ^^^^ |
| 16:21.21 | yun1989 | I can't copy this log |
| 16:21.42 | yun1989 | ::( |
| 16:22.03 | [TK]D-Fender | Get a better client |
| 16:22.13 | [TK]D-Fender | You are cutting off important information |
| 16:25.55 | *** join/#asterisk jkroon (~jkroon@dustpuppy.is.co.za) |
| 16:26.00 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 16:26.05 | jkroon | !seen sruffel |
| 16:26.11 | jkroon | ~seen sruffel |
| 16:26.18 | infobot | jkroon: i haven't seen 'sruffel' |
| 16:26.55 | jkroon | ~seen rusself |
| 16:26.55 | infobot | jkroon: i haven't seen 'rusself' |
| 16:34.27 | jzu_ | well this is weird, Asterisk is not showing my Cisco phone in Peers but I'm able to call VoiceMail etc |
| 16:34.31 | jzu_ | any clue how to debug it? |
| 16:36.16 | [TK]D-Fender | "sip set debug on" |
| 16:37.58 | aui_ | <PROTECTED> |
| 16:38.01 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 16:38.50 | mikealeonetti | Now here is an interesting one. When I'm using STREAM FILE or RECORD FILE in AGI script, and I press #, the next two commands return the DTMF tonevalue. |
| 16:39.32 | jzu_ | aui_: well I'm able to call to VoiceMail and other Asterisk behind IAX2 Trunk |
| 16:39.43 | jzu_ | aui_: so yes, phone is working but it doesn't register to Asterisk :O |
| 16:39.46 | mikealeonetti | Or really the character value. |
| 16:40.29 | mikealeonetti | Maybe I need a pause. |
| 16:40.54 | jzu_ | [TK]D-Fender: http://pastebin.com/CuvG48BC |
| 16:40.56 | jzu_ | there's sip debug |
| 16:41.44 | aui_ | I had a similar issue today and last week: some phones were able to register, some not. One solution I had -> restart server completely, ^^ |
| 16:41.56 | jzu_ | done it already, rebooted whole Linux |
| 16:41.59 | jzu_ | and the phone |
| 16:42.11 | [TK]D-Fender | jzu_: So it hasn't registered |
| 16:42.38 | aui_ | the second time I was running a soft phone on the same machine which was using "the same/at least one" port from asterisk |
| 16:42.41 | jzu_ | [TK]D-Fender: yes, wonder why - password etc is correct |
| 16:42.47 | jzu_ | [TK]D-Fender: it's not like it's rocket science |
| 16:43.57 | *** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de) |
| 16:44.10 | [TK]D-Fender | jzu_: Maybe it isn't trying at all |
| 16:44.45 | jzu_ | yeah |
| 16:46.23 | paulc | c |
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| 17:03.42 | *** join/#asterisk kb3ien (~kb3ien@gazelle.ppm-vpn.databit7.com) |
| 17:04.56 | kb3ien | Why is asterisk returning LoopDetected on an INVITE ? (the invite has 3 ip addresses) Contact: <sip:gazelle@192.168.47.144:5060>;q=1, <sip:gazelle@192.168.6.47:5060>;q=0.667, <sip:gazelle@192.168.31.204:5060>;q=0.334 (ekiga) ? |
| 17:17.08 | *** join/#asterisk _-Jon-_ (~jon@windows8.net-xero.com) |
| 17:25.51 | *** join/#asterisk areski (~areski@80.174.128.96.dyn.user.ono.com) |
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| 17:35.52 | *** join/#asterisk Iruta-san (~Iruta-san@n136-138.tmp.edunet.ru) |
| 17:35.58 | Iruta-san | Hello |
| 17:36.11 | carrar | Hello |
| 17:36.33 | carrar | Ohayoo |
| 17:38.04 | Iruta-san | could somebody give me any advise on installing asterisk from packages on centos 6.6? here is a trouble with $releasever... |
| 17:39.06 | Iruta-san | I've added repos automatically with rpm following https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages |
| 17:40.27 | carrar | What sort of problems are you having? |
| 17:40.28 | Iruta-san | any asterisk repo file substitutes $releasever in URL which for me is 6.6 |
| 17:41.05 | Iruta-san | but there is only directory named '6' for centos, no subversions. |
| 17:41.47 | Iruta-san | I am newe to centos and want to know - is it okay to manually replace $releasever with '6' or there is any better way? |
| 17:42.08 | Iruta-san | i mean changing baseurl=http://packages.asterisk.org/centos/6/asterisk-1.8/$basearch/ in the .repo file |
| 17:43.29 | carrar | cept asterisk-1.8 doesn't exist |
| 17:43.55 | carrar | point your web browser at your url http://packages.asterisk.org/centos/6/ |
| 17:44.31 | Iruta-san | really. |
| 17:44.41 | *** join/#asterisk gusto (~gusto@2a02:810d:a840:3c20:8219:34ff:fecf:17f0) |
| 17:46.06 | Iruta-san | okay, let it be asterisk-11 or whatever. I do not know any changes but that i will google :) |
| 17:46.22 | carrar | anyreason you want 1.8 and not 13? |
| 17:47.06 | carrar | (13 being the most latest) |
| 17:52.07 | Iruta-san | What I want is actually the simpliest asterisk server without any web-face. Whe have small sip network, using only sip provider and now we have a task: connect to another similiar sip network and secure calls in-between. I assume that SRTP+TLS is the thing we need, but our current asterisk is built on red hat 4.1.2. I've decided to make a new similiar server and rewrite those old and dirty |
| 17:52.07 | Iruta-san | configs we have now :) |
| 17:57.08 | Iruta-san | Okay, i've replaced that var in repo-file, installing asterisk-11 now |
| 17:57.13 | Iruta-san | thanks :) |
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| 18:14.24 | ChannelZ-Wk | Hey if a some of you are bored, please go to https://www.google.com/safebrowsing/report_phish/ and report the URL http://realhighteaathome.com/dev/backup/docin/view.html as someone trying to phish people's Google account info |
| 18:14.29 | ChannelZ-Wk | Hopefully get it shut down faster |
| 18:14.39 | ChannelZ-Wk | OT I know but too bad :) |
| 18:16.35 | jzu_ | argh :-D |
| 18:16.40 | jzu_ | proxy_register : was 0 |
| 18:16.51 | jzu_ | no wonder it didn't try to register .... |
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| 18:36.31 | jzu_ | http://pastebin.com/cAgCqNS3 - any clue why outgoing calls die right away? |
| 18:36.34 | jzu_ | Trunk is Sonetel |
| 18:36.47 | *** part/#asterisk mikealeonetti (~mike@ool-2f141af4.dyn.optonline.net) |
| 18:39.29 | jzu_ | incoming calls through that same trunk work OK |
| 18:40.26 | cryptic | I have to admit, asterisk is pretty cool |
| 18:40.35 | cryptic | I went from knowing nothing three days ago to having a functional system |
| 18:42.02 | jzu_ | the same, almost functioning system |
| 18:42.16 | jzu_ | internal calls work ok, and so does incoming through a trunk but outgoing don't yet |
| 18:43.36 | cryptic | my next goal is a bit more difficult: to implement 2FA by having a user read the code their mobile device/computer/security token displays and verify that the voice reciting the code matches the voice associated with that account |
| 18:43.42 | [TK]D-Fender | jzu_: SIP/2.0 404 Not Found |
| 18:43.45 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 18:43.51 | [TK]D-Fender | jzu_: INVITE sip:00407007600@212.72.62.115 SIP/2.0 |
| 18:44.06 | [TK]D-Fender | jzu_: # is no good for them |
| 18:44.29 | cryptic | jzu_: for outgoing, I had to make sure the context associated with my voip provider matched the context of my internal SIP users |
| 18:45.00 | [TK]D-Fender | Context isn't relevant |
| 18:45.27 | cryptic | [TK]D-Fender: in his particular situation or in general? |
| 18:45.59 | [TK]D-Fender | Mostly his, in a way both |
| 18:46.38 | [TK]D-Fender | His call is GOING out, and getting rejected. That means it hit the dialplan to actually Dial. Therefor it certainly isn't a case of "wrong context" |
| 18:49.05 | jzu_ | [TK]D-Fender: hmm, so Sonetel thinks the phone number is NOT valid? |
| 18:49.15 | jzu_ | [TK]D-Fender: that's my mobile phone in Finland |
| 18:49.52 | jzu_ | it doesn't say anything, just immediately hangs up without tone |
| 18:51.05 | cryptic | ah ok [TK]D-Fender |
| 18:51.22 | cryptic | that's an issue with the VoIP provider, then |
| 18:52.20 | [TK]D-Fender | jzu_: Either they can't match your identity (From: "Jussi" <sip:Sonetel@jus.si>;tag=as7b1c49c2) |
| 18:52.50 | jzu_ | hmm so I have something fishy there |
| 18:52.51 | [TK]D-Fender | I'd make sure you have "fromuser=YOURUSERNAME". I'm pretty sure "Sonetel" is not it |
| 18:52.58 | jzu_ | yes :-) |
| 18:53.00 | cryptic | jzu_: check out http://wiki.voip.ms/article/Asterisk_(SIP) |
| 18:53.26 | cryptic | that's a decent configuration that works perfectly for me (obviously replace the host/etc. with your information |
| 18:55.22 | jzu_ | sip show registry shows sip.sonetel.com as registered, hmmm |
| 18:55.31 | jzu_ | and all peers are online |
| 18:55.38 | jzu_ | that is, all three Sonerel IP's |
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| 18:58.52 | [TK]D-Fender | Doesn't actually mean anything |
| 18:59.37 | [TK]D-Fender | Your issue is either bad username, or invalid # format being passe |
| 19:00.57 | jzu_ | ah, could it be outbound caller ID that is wrong? |
| 19:01.17 | jzu_ | should it match the external number or internal Sonetel number? |
| 19:02.01 | [TK]D-Fender | jzu_: PM your peer config |
| 19:02.16 | [TK]D-Fender | jzu_: So I can see in private if something looks off... |
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| 19:08.11 | IRCJOE | hey guys -newbie which book i could i start with |
| 19:09.31 | [TK]D-Fender | ~book |
| 19:09.31 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:09.32 | [TK]D-Fender | ^^^^^^^^^^^ |
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| 21:03.41 | jzu_ | so |
| 21:03.51 | jzu_ | http://pastebin.com/vbStDLpY - any clue why I'm getting that error? |
| 21:04.34 | jzu_ | context for that SIP Trunk is from-trunk |
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| 21:25.24 | Kobaz | anyone use panasonic dect/ip phones? |
| 21:25.30 | Kobaz | trying to get them to auto answer |
| 21:26.55 | jzu_ | well that one was easy, changed from-trunk to from-pstn-toheader |
| 21:27.14 | jzu_ | now I just need to figure out why the hell caller ID is hidden on outgoing calls |
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| 21:56.56 | snadge | jzu_, how did you do the protflash ? |
| 21:57.21 | snadge | to reboot my 7941 just settings **#** |
| 21:57.50 | snadge | i just cant get the sucker to register if i use UDP.. which is annoying.. have to default to prefer tcp |
| 21:57.58 | snadge | i might try a firmware downgrade |
| 22:00.29 | snadge | also have a 942 from a customer thats admin locked.. lol.. people just buy these things off ebay for like $20 and go.. "can you get it to work.. k thx" |
| 22:00.44 | snadge | they rang up cisco and apparently cisco told them.. if you dont know the admin password, forget it |
| 22:01.02 | snadge | but you can just tftp a new admin password :| |
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| 22:41.56 | jeev | does anyone use hOMER? |
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