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| 01:25.39 | p3nguin | seri: This is a very interesting piece of equipment. I can't believe this was trash! |
| 01:25.41 | golgi | hello everyone! |
| 01:26.33 | golgi | hmmm |
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| 01:27.32 | navaismo | someone needs feedback |
| 01:28.11 | golgi | confused about my nick |
| 01:29.34 | golgi | i just set up my first asterisk box and have one or two questions. Is it still possible to use google talk/voice for outgoing calls? |
| 01:30.16 | golgi | I'm using asterisk 11 with requisite modules installed. at least, the ones I've found listed on the 2 or 3 tutorials out there online |
| 01:30.39 | p3nguin | I use it on 1.8, so I have to assume you can use it on 11. |
| 01:31.12 | golgi | so from a google standpoint its still possible- like, there haven't been any policy changes since those tutorials were written... |
| 01:32.30 | golgi | so far at this point i have a basic setup- two users with soft phones that can call each other. if i can get google outgoing/incoming working then I'll have all i need to uild from there |
| 01:32.41 | golgi | *build |
| 01:34.45 | *** join/#asterisk Russ_ (~russ@pool-74-100-57-85.lsanca.fios.verizon.net) |
| 01:35.32 | navaismo | golgi, https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
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| 01:47.58 | golgi | ill try that one again. i think i get confused with contexts |
| 01:48.05 | golgi | ill let you know what i run in to. |
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| 01:53.11 | golgi | ok, so based solely off the example, the extensions.conf file would look like this...? http://pastebin.com/Y7609qze |
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| 02:03.07 | golgi | ok finally getting somewhere now |
| 02:10.54 | navaismo | seems ok |
| 02:17.17 | golgi | success! |
| 02:17.32 | golgi | see, i just needed the company of smart people |
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| 02:25.42 | golgi | but, over vpn on 3G doesnt work too well. |
| 02:26.02 | p3nguin | No surprise there. |
| 02:26.13 | golgi | i can talk, but i can't hear anything. |
| 02:26.54 | golgi | the callee can hear me but i can't hear anything they say |
| 02:27.01 | golgi | what does this mean |
| 02:28.34 | golgi | hmmm. my vpn ip is waaaaay off the subnet than what is permitted in sip.conf. but that would be an all or nothing proposition, right? |
| 02:32.48 | slav3_kitten | golgi, i had a SIP ALG on the ISP side that caused that problem for me |
| 02:34.07 | golgi | the isp can identify SIP even if the traffic is encrypted? |
| 02:34.20 | p3nguin | I'm still trying to figure out what SIP has to do with Google Voice. |
| 02:35.13 | slav3_kitten | p3nguin, me not paying attention at ALL |
| 02:35.30 | slav3_kitten | and the sip.conf bit |
| 02:35.53 | golgi | wait |
| 02:36.08 | slav3_kitten | golgi, you could have an ALG in your vpn route |
| 02:36.38 | slav3_kitten | i'd assume it's traversing nat to get to your vpn but i don't really know how you have things setup |
| 02:36.54 | golgi | yeah |
| 02:37.17 | golgi | im about to either learn something new or violently snap something in my brain |
| 02:37.30 | slav3_kitten | i'm hoping for a snap |
| 02:37.47 | golgi | if i go silent for a long time you'll know |
| 02:37.55 | golgi | my address is.... |
| 02:38.19 | slav3_kitten | stream it live so we can watch the snap |
| 02:38.40 | golgi | blood dripping from the ear |
| 02:39.52 | slav3_kitten | so you have a router with vpn, tethered to your 3g phone for internet, which has the vpn endpoint where? |
| 02:39.54 | golgi | ok, when I'm on vpn the iphone's soft phone does register with the asterisk server. but, my ip address on the iphone is not the ip address listed in sip.conf |
| 02:40.15 | golgi | it's only that ip address when i'm on wifi |
| 02:40.37 | slav3_kitten | i'm confused and thus going to just be quiet |
| 02:40.43 | p3nguin | What do you mean by the "address listed in sip.conf"? |
| 02:40.55 | golgi | in sip.conf's permit |
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| 02:41.15 | golgi | nevermind |
| 02:41.23 | golgi | the blood is starting to stream |
| 02:41.30 | golgi | confused myself there |
| 02:41.53 | p3nguin | If you have created an ACL on a peer, you'll either have the ability to register and make calls or you won't. |
| 02:41.54 | WIMPy | Is there any known cure for using Asterisk, BTW? |
| 02:42.17 | golgi | p3nguin: that's what i thought- it would be all or nothing |
| 02:42.27 | golgi | so i need to look elsewhere |
| 02:43.43 | p3nguin | Now, I'm a little concerned that you're saying you have created an ACL on the peer and you can still use it from another address. That is a big problem. |
| 02:43.49 | golgi | ok, astrisk registerd my softphone with an ip of the VPN server |
| 02:44.33 | golgi | so the issue os probably going to be on the VPN server. agreed? |
| 02:44.36 | golgi | *is |
| 02:44.55 | p3nguin | I wouldn't have any way to know. |
| 02:45.07 | golgi | it's forwarding traffic such that my phone rings, but that's about it |
| 02:45.18 | p3nguin | I would start looking at the sip debug. |
| 02:45.35 | p3nguin | It sounds like you didn't deal with RTP. |
| 02:45.55 | p3nguin | SIP starts/stops the calls, but RTP is for the media stream. No RTP stream, no audio. |
| 02:46.12 | golgi | my voice gets out but nothing comes in |
| 02:46.25 | p3nguin | It's still RTP. |
| 02:46.31 | golgi | ok |
| 02:46.37 | p3nguin | Does it work without the VPN? |
| 02:46.42 | golgi | yes |
| 02:46.44 | golgi | quite well |
| 02:47.01 | golgi | have placed several calls now at this point |
| 02:47.05 | golgi | no issues |
| 02:47.11 | p3nguin | So then you better start looking at the sip debug to see where your audio went. |
| 02:47.31 | golgi | probably to the curiosity rover |
| 02:47.35 | slav3_kitten | what's your vpn server |
| 02:47.42 | golgi | openvpn |
| 02:47.59 | golgi | just stood it up today actually as a replacement- its the standlone VM |
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| 02:48.16 | golgi | it also otherwise works flawlessly |
| 02:48.59 | p3nguin | Why would the end point be using an address other than its own VPN-assigned address? You said it is using the server's address, but that seems wrong to me. |
| 02:49.33 | slav3_kitten | http://lists.digium.com/pipermail/asterisk-users/2011-January/257911.html not sure if this helps you golgi |
| 02:49.45 | golgi | sorry, quick paste.... -- Registered SIP 'zac' at 10.100.0.4:5060 |
| 02:49.45 | golgi | that's the VPN server, not the phone's VPN IP |
| 02:50.11 | p3nguin | I don't understand why it would do that. |
| 02:51.05 | golgi | ah |
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| 02:51.52 | golgi | second paragraph in item 1)... i am using tcp and udp adaptive setup in openvpn |
| 02:52.23 | golgi | this post advises using only the UDP port. |
| 02:52.58 | golgi | that seems like low hanging fruit but its worth a try |
| 02:54.10 | p3nguin | Is your asterisk configured to use tcp? Does your phone also use tcp? I suspect no to both of those questions, making that difference irrelevant in my opinion. |
| 02:54.52 | golgi | i did not explicitly set that one way or another. which config keeps that option? |
| 02:55.19 | p3nguin | the tcpbindaddr parameter in sip.conf |
| 02:55.36 | p3nguin | By default, sip doesn't use tcp. |
| 02:55.58 | golgi | so if not set then UDP? it's not set |
| 02:56.44 | p3nguin | By default, it is UDP only. If you want SIP to use TCP, you have to configure a tcpbindaddr value. |
| 02:59.10 | golgi | ok, switching VPN to only use UDP |
| 03:07.54 | golgi | here goes nothing |
| 03:09.27 | golgi | no that didnt do it |
| 03:09.45 | p3nguin | Surprise! |
| 03:09.49 | golgi | hehe |
| 03:09.50 | golgi | i know |
| 03:09.58 | p3nguin | So... |
| 03:10.03 | golgi | the caller hears is own voice back in the earpiece |
| 03:10.07 | p3nguin | How about that sip debug I told you to look at? |
| 03:10.12 | golgi | right |
| 03:10.24 | golgi | *his own |
| 03:10.42 | golgi | that's where the audio is going ;) |
| 03:11.32 | p3nguin | I do not understand your remark. |
| 03:12.31 | golgi | the caller, when trying to call me when I'm on vpn, simply hears his own voice in his earpiece when he talks, i hear nothing. he can hear me when I speak though |
| 03:12.45 | p3nguin | So how about that sip debug? |
| 03:13.01 | carrar | NO DEBUG 4U!! |
| 03:13.08 | p3nguin | Feel free to pastebin it soon. |
| 03:13.38 | p3nguin | ~pb |
| 03:13.38 | 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. |
| 03:18.09 | golgi | yeah sorry i had to check the vpn for something |
| 03:18.21 | golgi | sip debug on ....how to redirect it to a file?\ |
| 03:18.39 | golgi | or should i redirect the asterisk -rvvvv itself to a file? |
| 03:19.01 | p3nguin | Are you using putty, by chance? |
| 03:19.04 | golgi | yes |
| 03:19.13 | golgi | set the scrollback to 1 gazillion? |
| 03:19.14 | p3nguin | Just set the log file in putty. |
| 03:22.47 | golgi | wait for it....wait for it.... |
| 03:24.38 | p3nguin | *Yawn* I waited long enough. Bed time! |
| 03:25.55 | p3nguin | Okay, but on a serious note, if you take too long, I actually will go to bed. |
| 03:26.18 | golgi | http://pastebin.com/XM2MfAk1 |
| 03:26.23 | golgi | sorry |
| 03:26.46 | golgi | i had to get it nice and clean as to only zero in on a vpn session |
| 03:27.06 | p3nguin | gsm codec? Barf! |
| 03:27.31 | golgi | n00b |
| 03:30.17 | golgi | so asterisk is aware of my VPN ip it seems |
| 03:30.23 | golgi | Contact: <sip:zac@5.5.12.2:5060> |
| 03:33.12 | p3nguin | What is this 5.5.12.2 address? |
| 03:33.21 | golgi | that's the iphone on VPN |
| 03:33.46 | golgi | thank openvpn for that goofiness. I'm too lazy to change it. |
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| 03:41.05 | kannan | hello, in a context (where background is being used) , is there any way to reduce the dialplan overlap time , which is currently around 4 seconds |
| 03:45.17 | golgi | anything jump out at anyone? |
| 03:50.30 | p3nguin | I'm interested in the routing table on the vpn box. |
| 03:50.53 | golgi | i'll paste those for you in a sec |
| 03:51.06 | p3nguin | I once had a similar situation and it was caused by no return route. |
| 03:51.16 | golgi | you want the iptabls entries? |
| 03:51.23 | p3nguin | Nope, just the routing table. |
| 03:51.30 | golgi | okie |
| 03:51.34 | golgi | dokie |
| 03:51.38 | p3nguin | (for now, anyway) |
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| 03:51.59 | p3nguin | For testing like this, any firewall rules should have been turned off. |
| 03:52.34 | golgi | http://pastebin.com/MpxxCUyG |
| 03:54.26 | p3nguin | This is rather confusing for me. |
| 03:55.47 | p3nguin | I don't know what those interfaces are, and I don't know how those routes get traffic back to your phone. |
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| 03:58.41 | golgi | its an openvpn appliance i'm using, just seeing these interfaces for the first time too |
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| 04:18.01 | apb1963 | So I have a few problems... network changed over the weekend... I added a router where before I had none. My router NATs, however my asterisk server is in the DMZ - I assume that makes no difference because my * server is getting a private address. So with that said, the problem is that even though asterisk _appears_ to be doing the right thing in the logs by playing voicemail files, I get a busy signal on the POTS phone. Any clues as to what I shoul |
| 04:19.45 | jpsharp | POTS + network = doesn't add up. How is the phone connected to asterisk? And did you teach asterisk about its external IPs via a STUN server or the like? |
| 04:20.20 | apb1963 | I have a softphone extension. |
| 04:20.30 | apb1963 | Which rings |
| 04:20.34 | apb1963 | But I don't answer it |
| 04:21.38 | apb1963 | I'm not sure about the external IPs and STUN server... so, I'd have to guess no? |
| 04:23.56 | jpsharp | Since you put Asterisk behind NAT, you'll need to tell it. There's an entry in sip.conf called externip that tells Asterisk what its external IP is or how to find it. |
| 04:24.41 | p3nguin | externaddr or externhost |
| 04:25.00 | jpsharp | Ohyes. You're right. |
| 04:25.14 | jpsharp | I don't use it, so I never remember it. |
| 04:25.51 | apb1963 | I'm using freepbx... there's a file called sip_general_additional.conf that has an externip and a localnet |
| 04:26.21 | apb1963 | the externip is what is now the router's public IP |
| 04:27.11 | apb1963 | oh wait... that's not even right... it's the old host address before I had a router when my ISP allowed me more than one IP addr. |
| 04:27.29 | apb1963 | and localnet is wrong now |
| 04:27.37 | p3nguin | ~freepbx |
| 04:27.37 | 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 |
| 04:27.53 | apb1963 | kk thanks |
| 04:28.11 | jpsharp | That would explain it. You've got packets going where they don't need to go. |
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| 04:28.56 | apb1963 | well the question at this point is, what are the right places? I assume externip is my router's public address and localnet is... my private network yes? |
| 04:29.13 | LemensTS | Im repackaging up some 3rd party software for asterisk into a debian iso, was on asterisk 1.8, should I go to asterisk 10 or 11? I know 11 is still fairly new.... |
| 04:29.51 | jpsharp | I'd go with 10. |
| 04:30.00 | p3nguin | You should stay on 1.8 unless you need something in 11. |
| 04:30.18 | p3nguin | 1.8 and 11 are LTS. |
| 04:30.49 | g_r_eek | anyone knows of any good call control/call manager for asterisk on windows? FREE and not? |
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| 04:37.01 | apb1963 | Hmmm... I changed those addresses and reloaded..... same results |
| 04:37.34 | LemensTS | Ok cool thanks guys, I'm not needing anything new. I looked at the changelog of 11, is there a doc that says what new features were added to 10 and 11? |
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| 04:58.31 | epaphus | Hello iam using MusicOnHold() but all i hear is silence... how can i debug this |
| 04:58.38 | golgi | interesting development |
| 04:58.40 | epaphus | i placed my audio file in mod dirctory |
| 04:58.42 | epaphus | moh |
| 04:59.55 | golgi | with my openvpn issue, i can call/hear and speak both ways with phones on the LAN from my VPN'd phone. If i try to call a number outside of the network from my VPN'd phone I connot hear anything. |
| 05:00.42 | epaphus | nevermind, all i had to do was reload |
| 05:00.45 | epaphus | bah |
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| 05:11.20 | jpsharp | golgi: NAT issues? |
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| 05:29.57 | slav3_kitten | jpsharp, i suggested that hours ago |
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| 06:23.04 | apb1963 | I think I'm having NAT issues.... I'm confused... should I be forwarding RTP packets on my firewall somewhere? |
| 06:23.39 | ectospasm | apb1963: typically you will forward ports 10000 through 20000 to your Asterisk system |
| 06:24.02 | apb1963 | that's what's confusing... my firewall and asterisk are on the same server |
| 06:24.29 | apb1963 | i'm using iptables |
| 06:25.18 | apb1963 | do i still need to forward? if so, where? to the private ip? or to the public ip on the router? |
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| 06:26.07 | ectospasm | is Asterisk listening on all interfaces (0.0.0.0), or just the local subnet? |
| 06:26.44 | apb1963 | hmmm... not sure... where can i check that? |
| 06:27.11 | ectospasm | typically that will be in sip.conf, in the general section |
| 06:30.34 | apb1963 | appears to be in a trunk? but it simply says port=5060, the only reference to a network is permit & deny |
| 06:31.03 | kaldemar | apb1963: you just neep to ACCEPT UDP to the ports, not forward it anywhere. |
| 06:31.49 | apb1963 | I do accept udp... but I will doublecheck just to be sure |
| 06:31.53 | kaldemar | apb1963: port=... does not configure what asterisk listens on, it is for configuring a port on a remote device. |
| 06:33.21 | apb1963 | well.. I grepped for 5060 and that's all that came up |
| 06:34.32 | kaldemar | 5060 is not used for rtp. what is the issue you are experiencing exactly? |
| 06:36.38 | apb1963 | yes I accept udp to ports from 10000-20000 |
| 06:38.03 | apb1963 | wait... if my router is NATing.... |
| 06:39.02 | apb1963 | the issue is twofold... I never get my voicemail message even though asterisk hits all the right routines.. and I can't hear anything if I actually pickup the phone.... so to me that says rtp |
| 06:39.58 | apb1963 | i'm just not sure what to do about it... |
| 06:40.06 | apb1963 | I think my router needs to forward those ports then |
| 06:40.10 | apb1963 | yes? no? |
| 06:40.17 | apb1963 | it's not the firewall |
| 06:40.26 | apb1963 | right? |
| 06:40.42 | apb1963 | waits for confirmation of his suspicions |
| 06:41.24 | kaldemar | yes, your NAT router needs to forward the ports to your asterisk. |
| 06:42.40 | apb1963 | including 5060 |
| 06:43.03 | apb1963 | damn... I was originally thinking if I put the server in the DMZ I wouldn't have to do that. |
| 06:43.28 | apb1963 | but... I guess if the router is NATting regardless... |
| 06:44.49 | apb1963 | this is what I've been dreading... the router appears to take one port at a time |
| 06:45.06 | apb1963 | so unless something like 10000:20000 works... |
| 06:45.16 | apb1963 | I think I have to return this puppy |
| 06:48.13 | ectospasm | there's no way to set a range? Lame. Return it post haste. |
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| 07:33.13 | apb1963_ | am I back? |
| 07:33.32 | apb1963_ | client is a nightmare sometimes |
| 07:34.15 | apb1963_ | so that doesn't seem to have worked |
| 07:34.51 | apb1963_ | my router seems to have accepted my changes afaik... but i'm still stuck with the same results |
| 07:41.52 | kaldemar | apb1963_: enable sip debug and make a call. pastebin what you get in CLI. |
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| 08:02.39 | apb1963_ | ok |
| 08:11.41 | apb1963_ | kinda having trouble with it but... see if this is good? http://pastebin.com/TbEAA05Y |
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| 08:15.31 | kaldemar | apb1963_: you paste does not have a full call, but it does show retransmits. which suggests that asterisk is not getting SIP responses back from iptel. |
| 08:20.26 | apb1963_ | let me try to get you a full call |
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| 08:31.03 | apb1963_ | this should be better http://pastebin.com/K0p2TUKY |
| 08:31.26 | apb1963_ | keep in mind, the call goes through... I simply can't hear anything from either side. |
| 08:31.40 | apb1963_ | the extension is 101 |
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| 08:32.25 | Guest90715 | hi |
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| 08:35.53 | kaldemar | apb1963_: if you have nat=yes under [general] and don't have nat=no under [from-IPTEL], add the latter. |
| 08:36.27 | apb1963_ | k |
| 08:36.31 | apb1963_ | please wait |
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| 08:47.29 | kaldemar | your asterisk keeps retransmitting OK to 217.9.36.145 because it is not getting an answer to it. the call setup is not complete. |
| 08:48.21 | *** join/#asterisk Guest90715 (~IceChat9@80.214.8.237) |
| 08:48.23 | Guest90715 | hi |
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| 08:49.21 | fukuda76140 | I have a problem for send fax (command sendfax) with hylafax |
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| 08:49.36 | fukuda76140 | here my configuration : http://pastebin.com/ZDjL3hZ8 |
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| 08:56.34 | pbxman | morning |
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| 09:01.11 | fukuda7766 | Sorry for my déconnection |
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| 09:06.27 | apb1963_ | sorry this is taking so long... my system is moving like a snail. I have way too many windows open and it takes forever even just to close them. |
| 09:06.52 | apb1963_ | on the pastebin step though :) |
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| 09:07.54 | apb1963_ | http://pastebin.com/0YgW6QWX |
| 09:08.25 | apb1963_ | as far as I know, I made the change you said to... but I'm not positive I did it right. |
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| 09:16.44 | apb1963_ | sorry about that... don't know what happened.... got d/c'ed |
| 09:18.37 | apb1963_ | <knock knock> is this thing on? |
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| 09:32.01 | apb1963_ | Non-authoritative answer: 145.36.9.217.in-addr.arpa canonical name = 145.128-25.36.9.217.in-addr.arpa. 145.128-25.36.9.217.in-addr.arpa name = sip.iptel.org. |
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| 09:56.18 | tsipizic | hello |
| 09:56.48 | tsipizic | I am trying to activate radius cdr but asterisk send no data |
| 09:57.22 | tsipizic | I am able to send radius data from command line using radiusclient and radclient but asterisk does not send anything |
| 09:57.42 | tsipizic | although it produces that error in syslog |
| 09:57.49 | tsipizic | <PROTECTED> |
| 09:57.54 | tsipizic | any help? |
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| 10:34.06 | Diffen | Hello. I just want to clearify something here. When there is an invite incoming to Asterisk, the Asterisk will check the header that looks like this "INVITE sip:number@asterisk ip-address SIP/2.0" for the number in the Asterisk right? |
| 10:36.33 | kaldemar | eventually, among other headers. |
| 10:40.23 | Diffen | what header does it look at first? |
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| 10:46.41 | kaldemar | Diffen: why are you asking? |
| 10:47.48 | kaldemar | see handle_incoming() in chan_sip.c |
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| 10:57.51 | Diffen | kaldemar im having a debate with a friend about this |
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| 11:05.24 | kaldemar | Diffen: well, the first header is Cseq. were you both wrong? |
| 11:06.30 | Diffen | kaldemar: in that case yes but we were discussing what header to where the call should end up by using the B-number |
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| 11:18.58 | kaldemar | Diffen: destination is parsed from the request-line. |
| 11:20.02 | kaldemar | Diffen: i.e. the first line of the message (INVITE sip:...) |
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| 11:27.45 | Diffen | kaldemar ahh ok :) |
| 11:27.51 | Diffen | Thanks for the information |
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| 11:31.48 | gavimobile | I have an application with a time condition which goes to a queue if within business hours. http://pastebin.com/8DaffD1i in my outgoing call cdr fields when I dial the queue from inside it shows "start". can I change this that it shows extention 300 for example? cause that's the extention I use for the application. |
| 11:32.57 | kaldemar | gavimobile: no. |
| 11:33.23 | gavimobile | kaldemar: ok! thanks |
| 11:33.51 | kaldemar | it shows the extension. your is "start" so the CDR gets "start". make it use the real extension instead of start. |
| 11:35.02 | gavimobile | kaldemar: not sure how to do that. my guess is to use goto |
| 11:35.14 | gavimobile | goto 300 |
| 11:35.23 | gavimobile | and change start with 300 |
| 11:35.26 | kaldemar | replace "start" with something else. |
| 11:35.39 | gavimobile | I tried replacing it with 300, but it didn't work |
| 11:35.54 | kaldemar | what "it"? |
| 11:36.05 | gavimobile | when I tried dialing 300 |
| 11:36.12 | gavimobile | after changing start to 300 |
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| 11:36.40 | kaldemar | your extension is 0773354673. it does not match 300 now does it? |
| 11:37.01 | gavimobile | well that's how I intercept the inbound route |
| 11:37.10 | gavimobile | the inbound call* |
| 11:37.12 | gavimobile | sorry |
| 11:37.38 | kaldemar | where did you dial 300? |
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| 12:07.50 | gavimobile | from one of my peers kaldemar |
| 12:11.45 | kaldemar | that has nothing to do with anything you showed. |
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| 13:29.02 | rolandow | hi guys.. what is your favorite phone brand if you had to choose from Aastra, Cisco, Siemens, Grandstream, Panasonic, Polycom, SNOP, Tiptel/Yealink .. ? |
| 13:29.26 | rolandow | i will be implementing a new (small) asterisk setup soon (only 10 phones or so ) .. i was thinking about yealinks |
| 13:29.59 | rolandow | i'd like a remote phonebook to be able to integrate our CRM tool |
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| 13:33.32 | leifmadsen | likes Digium and Polycom devices |
| 13:34.27 | rolandow | ok ... digium isn't available at our supplier.. polycom is .. |
| 13:34.31 | rolandow | what's the advantage ? |
| 13:34.49 | rolandow | because they don't look too nice imho :) |
| 13:34.59 | leifmadsen | I just prefer the plastic and look plus the software is rock solid |
| 13:35.12 | leifmadsen | tons of features too |
| 13:35.19 | leifmadsen | pick what you want, I'm just one person :) |
| 13:35.57 | rolandow | of course.. i respect everybody's opinion .. but if 99% says i should go for polycom, i'm curious about the reason :) |
| 13:36.11 | leifmadsen | you should really test your devices before doing a deployment regardless |
| 13:36.26 | leifmadsen | which means you should order a few devices and test them |
| 13:36.31 | rolandow | yes.. of course.. |
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| 13:36.47 | rolandow | i'm going jobhopping .. i already used many tiptel's at my current job |
| 13:37.22 | rolandow | but well .. trying all differents brands would be an expensive hobby :) |
| 13:39.02 | WIMPy | And probably rather frustrating. |
| 13:39.18 | rolandow | that too |
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| 13:47.52 | [TK]D-Fender | rolandow, Polycom has superior build & sound quality, featureset, software stability, and is decently priced. Other may be cheaper or have more features but they sacrifice on other bits. |
| 13:48.26 | [TK]D-Fender | rolandow, Incertain applications I recommend specific model from other makers as I do for the Aastra 6739i for instance. |
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| 13:57.48 | rolandow | [TK]D-Fender: ok .. that's what i was curious about .. |
| 13:57.54 | rolandow | i'll check polycoms again |
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| 14:14.50 | [sr] | its cold!! |
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| 14:31.27 | WIMPy | It is. And we had a little snow the last two days. |
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| 14:36.07 | jmetro | we went from 18 degrees to 60 degrees to 25 degrees in the past 4 days. |
| 14:36.38 | coppice | 60 is bloody hot |
| 14:37.11 | jmetro | it was actually 68 at one point, hot for the midwest yeah. |
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| 14:37.40 | coppice | oh, you mean F |
| 14:37.56 | rrittgarn | haha yeah he means F |
| 14:38.23 | coppice | how quaint |
| 14:39.02 | santa0536 | guys, does asterisk support ITU reccomendations for T.38? http://www.itu.int/ITU-T/recommendations/rec.aspx?rec=T.38 |
| 14:39.24 | santa0536 | or in other words what version of those reccoendation does asterisk support? |
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| 14:39.43 | jmetro | I didnt know anywhere habitable actually got to 140 F [60C] |
| 14:40.07 | coppice | I think death valley gets to something like that |
| 14:41.28 | rrittgarn | anyone play with externnotify and voicemail? I added the line externnotify=/scripts/script.sh to a mailbox but am not seeing it call the script. I haven't restarted asterisk. but i've done a voicemail reload, and if i show that mailbox i see the externnotify= info in there |
| 14:41.50 | coppice | santa0536: the meaning of the T.38 versions is unclear. * supports 14400bps |
| 14:43.02 | jacekowski | anybody familiar with digium D40 and other phones, i've got problem with users complaining about echo |
| 14:43.44 | jacekowski | but the problem is, i can't hear anything wrong myself, and from their description it sounds more like too loud sidetone rather than anything else |
| 14:43.56 | jacekowski | and i'm just wondering if there is any way of controlling that |
| 14:44.05 | santa0536 | coppice: ok, let me periphrase: there is a list of ITU specifications for T.38. does asterisk implements the most recent one? |
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| 14:50.32 | coppice | santa0536: as I said before, yes and no. if you think the latest one requires V.34 support, then no. if you think it doesn't, then yes |
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| 14:59.24 | apb1963_ | am I here? |
| 14:59.26 | apb1963_ | yay |
| 14:59.35 | apb1963_ | So.. I'm still having problems with asterisk. I dial my DID, my softphone rings, I pick it up... no sound from either side. If I let it ring through to voicemail it plays the voicemail routines but I can't hear them and I just get a busy signal. I recently bought a router that is NATing. My centOS server has iptables running. i was able to capture an RTP packet with wireshark and I could listen to it. It was me! from when I answered the softphone |
| 15:02.23 | [TK]D-Fender | apb1963_, show us the call attempt with SIP DEBUG enabled |
| 15:02.24 | [TK]D-Fender | ~pb |
| 15:02.24 | 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. |
| 15:02.26 | [TK]D-Fender | ^^^ |
| 15:04.46 | apb1963_ | http://pastebin.com/VQgbE2ny |
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| 15:08.12 | [TK]D-Fender | apb1963_, <--- Transmitting (NAT) to 217.9.36.145:5060 ---> <--- your provider is NOT behind NAT. fix this first |
| 15:09.06 | jacekowski | i get that on my asterisk |
| 15:09.10 | jacekowski | when talking to sip phones |
| 15:09.10 | apb1963_ | Yah... but I'm behind NAT so... ?? |
| 15:09.14 | [TK]D-Fender | Indeed the IP's between their signalling and media server's don't match. This should be an immediate fail for RTP |
| 15:09.23 | jacekowski | even though there is no NAT between me and the phone |
| 15:09.34 | [TK]D-Fender | apb1963_, **THEY** are not. |
| 15:09.46 | apb1963_ | ok |
| 15:10.09 | [TK]D-Fender | apb1963_, When they say their audio is at IP X, Port Y, when you say they are behind NAT, you don't TRUST that and insist on the signalling IP. |
| 15:10.14 | jacekowski | [TK]D-Fender: i get exactly same thing when my asterisk server is talking to my phones |
| 15:10.21 | jacekowski | [TK]D-Fender: on sam subnet and same switch |
| 15:11.06 | [TK]D-Fender | jacekowski, Generally won't matter with phones since they use the same IP for both. ITSP's might not. |
| 15:11.25 | [TK]D-Fender | jacekowski, And it only says it because you told * they were nat'd |
| 15:11.35 | [TK]D-Fender | jacekowski, So if they aren't .... stop doing that then. |
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| 15:12.19 | jacekowski | i didn't |
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| 15:17.35 | ghost75 | if i do a originated call, the caller will up show wrong in cdr, can this be changed? |
| 15:18.31 | ghost75 | erm not caller, the called number doesnt show up, it displays my own number |
| 15:19.18 | apb1963_ | ok, no NAT for provider http://pastebin.com/R2Tc9iic |
| 15:22.09 | [TK]D-Fender | apb1963_, No more 5000 line pastebin's. That is an insane amount of garbage we don't need to see |
| 15:22.16 | apb1963_ | ok |
| 15:22.19 | ghost75 | the cdr dst field is read only? |
| 15:22.32 | apb1963_ | just wanted to make sure I didn't leave anything out |
| 15:22.49 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 15:23.11 | WIMPy | ghost75: You can change it by using Goto(). |
| 15:23.26 | ghost75 | why goto? |
| 15:24.05 | rolandow | is that NAT article mirrored somewhere? i saved this link in my bookmarks: http://www.aocomputing.net/?p=3 |
| 15:24.10 | rolandow | but now that i need it, it's gone. |
| 15:24.55 | WIMPy | ghost75: That changes the destination. |
| 15:25.19 | *** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 15:26.02 | ghost75 | goto changes context i thought |
| 15:26.15 | WIMPy | It can do that as well. |
| 15:26.58 | ghost75 | how |
| 15:28.41 | [TK]D-Fender | rolandow, I haven't gotten to restoring the server in a while. |
| 15:28.50 | [TK]D-Fender | rolandow, I'll look at it this weekend |
| 15:29.24 | rolandow | ah ok .. it's your server? :) |
| 15:30.00 | [TK]D-Fender | rolandow, Yes |
| 15:30.20 | [TK]D-Fender | ghost75, It Goto's wherever you tell it to. |
| 15:30.50 | *** join/#asterisk jrgill2 (~jrgill@unaffiliated/jrgill) |
| 15:30.54 | [TK]D-Fender | ghost75, You can use Goto the just jump 1 priority higher in the same exten for all it matters for the purpose of updating DST |
| 15:33.07 | p3nguin | Goto() changes either the priority, the priority and extension, or the priority and extension and context. |
| 15:34.04 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
| 15:34.13 | ghost75 | http://pastebin.com/acB3y100 <- like this? |
| 15:34.33 | ghost75 | loophole |
| 15:34.37 | p3nguin | nope |
| 15:34.45 | p3nguin | See above. |
| 15:34.56 | *** part/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 15:35.32 | ghost75 | i dont understand how |
| 15:35.40 | apb1963_ | [TK]D-Fender any further thoughts? |
| 15:36.09 | p3nguin | It cannot change ONLY the context. It cannot change ONLY the extension. It cannot change ONLY the context AND extension. |
| 15:37.11 | p3nguin | Priority only; extension and priority; or context, extension, and priority. |
| 15:37.53 | [TK]D-Fender | apb1963_, I'm not seeing a SANE output for a call attempt and I have not seen configs for the 2 ends involved. |
| 15:38.03 | ghost75 | you mean i should jump to another context with different exten ? |
| 15:38.11 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 15:38.16 | p3nguin | Nope. |
| 15:38.43 | *** join/#asterisk k611 (~K610@cred.epid.ucl.ac.be) |
| 15:38.52 | p3nguin | I'm just telling you what Goto() does. |
| 15:38.54 | *** join/#asterisk bpriddy (~bpriddy@ipv4.host.stabbyspazzout.net) |
| 15:39.40 | apb1963_ | Sorry I'm not sure what you're saying |
| 15:40.04 | p3nguin | <p3nguin> Goto() changes either the priority, the priority and extension, or the priority and extension and context. |
| 15:40.15 | p3nguin | <p3nguin> It cannot change ONLY the context. It cannot change ONLY the extension. It cannot change ONLY the context AND extension. |
| 15:40.19 | p3nguin | <p3nguin> Priority only; extension and priority; or context, extension, and priority. |
| 15:40.23 | *** join/#asterisk jrgill3 (~jrgill@unaffiliated/jrgill) |
| 15:40.23 | WIMPy | ghost75: The Destination is the extension. |
| 15:40.25 | p3nguin | That. Is. All. |
| 15:40.53 | jeffspeff | ...and now you know the rest of the story. |
| 15:41.10 | p3nguin | Thank you, Paul Harvey. |
| 15:41.32 | jeffspeff | :) |
| 15:41.48 | jeffspeff | it just felt right |
| 15:41.54 | ghost75 | can i use variable as exten ? |
| 15:42.01 | p3nguin | Perhaps. |
| 15:42.34 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 15:42.39 | [TK]D-Fender | apb1963_, I don't see a PB with a PROPER amount of conect so I don't spend an hour scrolling through crap, and I don't see your CONFIGS. What is not clear about this? |
| 15:43.11 | ghost75 | its a global var |
| 15:43.25 | jeffspeff | ghost75, global vars will work as extens |
| 15:44.39 | p3nguin | jeffspeff: Give me a quick example of what you're saying. |
| 15:46.11 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:46.32 | apb1963_ | This better? http://pastebin.com/9A9qXt1z The extension is 101 |
| 15:46.47 | *** join/#asterisk navaismo (~navaismo@189.144.120.135) |
| 15:47.32 | jeffspeff | if you set global var foo=123 then later in a context you can say exten=foo,1,blah() |
| 15:47.39 | jeffspeff | IIRC |
| 15:47.58 | p3nguin | That's what I thought you meant, but I needed to make sure I understood you. |
| 15:48.01 | ghost75 | exten => s,1,Goto(10) |
| 15:48.01 | ghost75 | exten => ${EXTENORIGINATE},10,Dial(SIP/arcor_out998780/${EXTENORIGINATE}) |
| 15:48.21 | cusco | hey folks |
| 15:48.42 | p3nguin | I can't say that idea will or will not work, but I have a suspicion that it will not. |
| 15:48.43 | cusco | queue has members like Local/100@context right? |
| 15:48.52 | p3nguin | That's one choice, yes. |
| 15:49.04 | cusco | and dialplan does several things and finally a Dial(SIP/100) or so |
| 15:49.07 | cusco | ok my question now is: |
| 15:49.29 | cusco | how can I make SIP/100 hear a 'beep' if the calle hangs up? |
| 15:49.33 | p3nguin | 100 is a terrible name for a SIP phone. |
| 15:50.01 | cusco | some signal... |
| 15:50.25 | jeffspeff | sip devices != extensions |
| 15:50.26 | p3nguin | The phone disconnecting should be a clue. |
| 15:51.51 | p3nguin | ghost75: If you ${EXTENORIGINATE} = s, then what you have might work... if variables can be used as extensions. |
| 15:51.58 | rolandow | what would you guys do? i created an image with my options: http://i.troll.ws/c63159f3.gif |
| 15:52.16 | rolandow | or well, at least I think those are my options :) |
| 15:52.22 | cusco | p3nguin: the phone disconecting does play a beep... |
| 15:52.29 | cusco | can I somehow play that beep? |
| 15:52.43 | ghost75 | doesnt work, log is saying: Goto (originatecall,s,10) |
| 15:52.45 | [TK]D-Fender | apb1963_, Do you think the call requires 300 lines of debug. You aren't even trying here. I still also don't see your CONFIGS. Last chance before I move on to something else... |
| 15:52.50 | ghost75 | its still using s |
| 15:53.03 | p3nguin | If the phone makes the sound when the call ends, that solves your problem. If the caller hangs up, your phone will disconnect. |
| 15:53.04 | *** join/#asterisk rowshi (~mroszkows@089-101-219195.ntlworld.ie) |
| 15:53.14 | WIMPy | I somehow got lost. Maybe ghost75 should elaborate on what exactely he needs. |
| 15:53.26 | p3nguin | ghost75: That's what Goto() does. I explained it two or three times already. |
| 15:53.51 | apb1963_ | working on the configs |
| 15:53.52 | p3nguin | If extensioin s runs Goto(10), you will go to extension s priority 10. |
| 15:54.06 | jeffspeff | ghost75, http://www.voip-info.org/wiki/view/Asterisk+cmd+Goto |
| 15:54.08 | apb1963_ | how many lines of debug would you like to see? |
| 15:54.09 | cusco | p3nguin: it doesnot produce. thats the thing... |
| 15:54.15 | p3nguin | If extensioin s runs Goto(bar,10), you will go to extension bar priority 10. |
| 15:54.16 | [TK]D-Fender | <ghost75> exten => s,1,Goto(10) <-- DST is set to the EXTEN you are on. I said you could jump withing the exten you're on ... but you don't even LIKE where you are. Do you think staying on that same exten is going to transfor the number? No, jump to the NUMBERED exten you want. |
| 15:54.17 | WIMPy | rolandow: Asterisk doesn;t really like to be multi-homed. |
| 15:54.26 | ghost75 | ok i missed something |
| 15:54.36 | rolandow | WIMPy: multi-homed, as being behind the multi wan nat you mean? |
| 15:54.52 | p3nguin | If extensioin s runs Goto(foo,bar,10), you will go to context foo extension bar priority 10. |
| 15:54.53 | *** join/#asterisk asr33 (~asr33@unaffiliated/asr33) |
| 15:55.13 | WIMPy | rolandow: As in having multiple public IPs. |
| 15:55.27 | rolandow | hm... yes.. now that you mention it, i read about that.. |
| 15:55.31 | ghost75 | Goto(extension,priority) <- this is also valid or? |
| 15:55.39 | p3nguin | Yes. |
| 15:55.41 | rolandow | although i can set externalip to a hostname, and restart asterisk, right? |
| 15:55.55 | *** join/#asterisk morfin (~morfin@morfin.telenet.ru) |
| 15:55.56 | p3nguin | <PROTECTED> |
| 15:55.59 | p3nguin | <PROTECTED> |
| 15:56.03 | *** join/#asterisk Yxa (~Yxa@bb119-74-74-189.singnet.com.sg) |
| 15:56.05 | jeffspeff | rolandow, dont' even have to restart asterisk, just sip |
| 15:56.06 | rolandow | i could even write a script that restarts asterisk when that happens. |
| 15:56.09 | p3nguin | <p3nguin> Priority only; extension and priority; or context, extension, and priority. |
| 15:56.16 | rolandow | jeffspeff: even better :) |
| 15:56.27 | WIMPy | rolandow: The point is that you will get the usual NAT issues in extreme. |
| 15:56.30 | Yxa | hi for a new production deployment, should i go with 11 or stick with 10? |
| 15:56.32 | rolandow | WIMPy: how could i create a failsafe asterisk otherwise? :) |
| 15:56.38 | jeffspeff | from *nix cli asterisk -rx "sip reload" |
| 15:56.41 | rolandow | yes.. i was afraid of that |
| 15:56.42 | jeffspeff | or something like that |
| 15:56.46 | rolandow | WIMPy: so you would choose option 2? |
| 15:56.58 | p3nguin | yxa: Do you need the features 11 offers that 1.8 doesn't have? |
| 15:57.51 | WIMPy | rolandow: If you want fail over, just make sure that a) you have only one default route and b) your Asterisk knows the public IP it's using any time. |
| 15:58.08 | Yxa | p3nguin nope. stability is my primary criteria |
| 15:58.28 | p3nguin | yxa: Better stick to 1.8, then. |
| 15:58.35 | rolandow | WIMPy: what do you mean with one default route?? if it's behind the vigor, it'll have one default route, right? |
| 15:58.39 | p3nguin | yxa: Not 10, for sure. 1.8 and 11 are LTS, 10 is not. |
| 15:58.47 | [TK]D-Fender | p3nguin, Is 11 currently too unstable for you? |
| 15:58.48 | morfin | hello |
| 15:58.56 | rolandow | WIMPy: or do you mean that it shouldn't use both wan's?? |
| 15:58.59 | morfin | about asterisk 11 |
| 15:59.08 | Yxa | can i get a 2nd opinion? |
| 15:59.11 | WIMPy | rolandow: Make sure it only ever uses one of the external lines. |
| 15:59.22 | rolandow | WIMPy: right.. that shouldn't be too hard. |
| 15:59.23 | morfin | i heard it should have websocket transport support is that true? |
| 15:59.33 | p3nguin | [tk]d-fender: It's too new for a production system. |
| 15:59.37 | WIMPy | rolandow: Yes. That's probably going to hurt your brain otherwise. |
| 16:00.09 | p3nguin | Of course the term "production" has various meanings for different people, so it could be just fine for some. Not for me, though. |
| 16:00.14 | [TK]D-Fender | p3nguin, Is that based on a gheneric theory of how long it should be out before you should consider it "stable", or due to actual reported issues? |
| 16:00.24 | [TK]D-Fender | p3nguin, because for the latter .... I really haven't heard much |
| 16:00.36 | rolandow | WIMPy: i think i could tell ddclient to restart sip when the ip changes .. something like that.. |
| 16:00.36 | [TK]D-Fender | p3nguin, No more than any other release |
| 16:00.41 | ghost75 | thx guys it works |
| 16:00.47 | rolandow | or even write my own shell script that checks the ip of the dynamic host |
| 16:01.14 | rolandow | when the internet goes down, a restart of asterisk shouldn't be too much of a problem, since the connection is gone anyways :) |
| 16:01.14 | WIMPy | rolandow: Best to get information from the router(s). If you can. |
| 16:01.45 | morfin | can anyone tell me: how can i check on my own asterisk if account exists from remote machine |
| 16:01.59 | rolandow | ok .. so then my remaining question is how can i prevent the nat issues.. i mean .. i have seen/read so many problems about it .. is it even possible to have asterisk behind a nat properly? |
| 16:03.02 | p3nguin | With much software, there are often more bugs in earlier versions. As those bugs get worked out, less obvious bugs will remain. The less obvious bugs don't get found quickly because they show up in less common usages. If more bugs show up in more common usages and less common usages yield less obvious bugs, it makes sense to me that the majority will not see those less obvious bugs. |
| 16:03.06 | WIMPy | rolandow: That should work if you configure it for that situation. |
| 16:04.16 | morfin | p3nguin non obvious bugs are usually being used by hackers |
| 16:04.30 | rolandow | ok .. i'll read some more about it tomorrow |
| 16:04.39 | rolandow | thanks for your input! |
| 16:05.00 | morfin | i mean they can be abused |
| 16:05.01 | Yxa | how about multicore cpu support? does 11 support them much better than 10 or 1.8? |
| 16:05.03 | apb1963_ | Here's the configs. I deleted everything that i believe isn't directly related, including other providers and other extensions so you could focus on just the one provider and one extension. . http://pastebin.com/v2zFpgbg How many lines would you like to see from the call? 50? 100? 150? |
| 16:05.15 | p3nguin | yxa: No. |
| 16:08.26 | morfin | i can use AMI to send command "sip show users" from script right? |
| 16:08.56 | [TK]D-Fender | p3nguin, That's like waiting for problems you don't know exist without a expiring on how long you wait... |
| 16:09.51 | p3nguin | morfin: Maybe SIPpeers, but not sip show users. |
| 16:10.02 | morfin | i need all users |
| 16:10.16 | morfin | not only registered |
| 16:10.17 | p3nguin | Are you talking about devices that are set to type=user? |
| 16:11.03 | [TK]D-Fender | morfin, AMI <- |
| 16:11.04 | [TK]D-Fender | ~book |
| 16:11.04 | 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 |
| 16:11.05 | [TK]D-Fender | ^^^^ |
| 16:11.45 | morfin | i did use it before to originate |
| 16:11.56 | p3nguin | Then sip show users isn't the correct CLI command, anyway. AMI's "SIPpeers" is the equivalent to CLI's "sip show peers" |
| 16:12.08 | morfin | hmmm |
| 16:12.52 | p3nguin | CLI's "sip show users" only lists devices configured as type=user. |
| 16:12.59 | p3nguin | If it is type=peer, you won't see it there. |
| 16:13.14 | p3nguin | And chances are that you don't have very many devices set up only as a user. |
| 16:13.52 | *** join/#asterisk mihamina (~mihamina@100.155.159.197-ip-dyn.orange.mg) |
| 16:14.25 | morfin | users i need are listed in that list |
| 16:15.34 | ghost75 | because u use users.conf ? |
| 16:16.14 | p3nguin | Fuck. I just explained that type=user is what shows up in sip show users. It has little to do with users.conf. |
| 16:16.59 | apb1963_ | [TK]D-Fender Here's the configs. I deleted everything that i believe isn't directly related, including other providers and other extensions so you could |
| 16:17.13 | ghost75 | i was only curious why he has users |
| 16:17.16 | apb1963_ | focus on just the one provider and one extension. . http://pastebin.com/v2zFpgbg How many lines would you like to see from the call? 50? 100? 150? |
| 16:17.29 | morfin | but i don't use users.conf |
| 16:17.31 | morfin | :O |
| 16:17.47 | morfin | actually that was configured not by me |
| 16:18.17 | [TK]D-Fender | apb1963_, s=Asterisk PBX 1.8.11.0 |
| 16:18.42 | morfin | there only fields: username, secret and accountcode |
| 16:19.03 | [TK]D-Fender | apb1963_, You have "canreinvite" being set there but that parameter does not even exist in that version. It was long ago replaced with "directmedia=no" which you should have in your trunk definition AND in your extensions. |
| 16:19.41 | [TK]D-Fender | apb1963_, Your FreePBX versions is also quite out of date. I recommend catching up to something currently supported |
| 16:20.32 | apb1963_ | OK, I will change that.. but for the record this was working until I added my new router into the mix and started NATing. |
| 16:21.09 | [TK]D-Fender | apb1963_, Also make sure your route has NO SIP ALG or other proxying between your server and the internet |
| 16:21.49 | apb1963_ | ok I don't know what that means.. please don't get mad. |
| 16:25.25 | [TK]D-Fender | apb1963_, Make sure your router isn't messing with SIP. |
| 16:26.23 | morfin | is Websockets support fully implemented in Asterisk 11? |
| 16:26.25 | apb1963_ | well... I set it to pass the packets to the server... should i remove that? |
| 16:27.34 | *** join/#asterisk cyborg-one (~cyborg-on@130-0-32-145.broadband.tenet.odessa.ua) |
| 16:27.45 | morfin | just asking because i was thinking about browser<=>asterisk<=>providers |
| 16:28.12 | [TK]D-Fender | morfin, Where does a browser have anything to do with Asterisk? |
| 16:28.23 | apb1963_ | Firewall > Virtual Servers This function will allow you to route external (Internet) calls for services such as a web server (port 80), FTP server (Port 21), or other applications through your Router to your internal network. More Info |
| 16:29.34 | morfin | what about voice |
| 16:31.59 | *** join/#asterisk rowshi (~mroszkows@089-101-219195.ntlworld.ie) |
| 16:32.21 | apb1963_ | Hmm... if it's NATing, doesn't that mean by default it's going to mess with ALL packets including SIP and RTP? |
| 16:32.40 | morfin | did you use nat=yes option? |
| 16:32.46 | apb1963_ | where? |
| 16:33.15 | apb1963_ | On the router? Yes. |
| 16:33.32 | morfin | If a peer is configured with nat=yes, it causes Asterisk to ignore the address information in the SIP and SDP headers from this peer, and reply to the sender's IP address and port. nat=yes enables a form of Symmetric RTP and SIP Comedia mode in Asterisk. |
| 16:34.36 | *** join/#asterisk labdi (~labdi@mail.woodbinemedical.com) |
| 16:34.42 | apb1963_ | in the trunk's incoming context? |
| 16:34.54 | morfin | http://www.voip-info.org/wiki/view/Asterisk+sip+nat |
| 16:35.01 | ghost75 | sip.conf |
| 16:35.25 | morfin | i think it can be set globally |
| 16:36.28 | morfin | so what about websockets && asterisk? |
| 16:36.52 | morfin | how stable is support if it's already implemented |
| 16:37.21 | labdi | hello, is there a way for all users to have their own passwords when dialing long distance |
| 16:38.40 | [TK]D-Fender | labdi, It's your dialplan, do it however you'd like. |
| 16:39.11 | labdi | ive managed to set each dept but havent figured a way to do each user. |
| 16:39.18 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.143) |
| 16:39.23 | apb1963_ | so I think I want qualify=yes in addition to nat=yes.... sound reasonable? That page is a bit over my head... at the moment. |
| 16:39.58 | ghost75 | qualiy is only checking if alive i think |
| 16:40.38 | p3nguin | labdi: You can use the password configuration for voice mail. I would guess each person has his own voice mail. |
| 16:41.25 | *** join/#asterisk felipealmeida (~user@querubim.tecgraf.puc-rio.br) |
| 16:41.49 | [TK]D-Fender | labdi, Look at the SIP peer that is calling ... or its callerid, or some other var you set in the peer, etc. Process based on that |
| 16:41.51 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 16:42.39 | [TK]D-Fender | apb1963_, You are not using DIRECTMEDIA=NO like I told you. Your verson of FreePBX does not seem to be configured to set this right. |
| 16:42.56 | p3nguin | Authenticate(), VMAuthenticate() |
| 16:42.59 | [TK]D-Fender | apb1963_, Perhaps it is not aware of the version of Asterisk it is set to configure |
| 16:44.09 | apb1963_ | ok qualify=yes and nat=yes seems to be an improvement |
| 16:44.20 | labdi | maybe i amnot asking properly, users sometimes move from phone to phone |
| 16:44.21 | *** join/#asterisk elico (~Thunderbi@109.64.229.90) |
| 16:44.49 | apb1963_ | now I get voicemail when I don't answer |
| 16:44.49 | p3nguin | So then using the callerid or phone info is out. Use the voice mail configuration like I mentioned. |
| 16:44.58 | [TK]D-Fender | labdi, Then you also need to make some dialplan to log what "user" a DEVICE is associated with |
| 16:44.59 | labdi | ahhh |
| 16:45.10 | labdi | *light goes off* |
| 16:45.11 | labdi | i see |
| 16:45.14 | p3nguin | Off? |
| 16:45.14 | labdi | thank you :) |
| 16:45.17 | labdi | on |
| 16:45.18 | labdi | lol |
| 16:45.25 | apb1963_ | jury is still out on whether I can hear myself... I seem to get breathing but not sure about voice :) |
| 16:45.26 | p3nguin | Usually lights go on when a person gets an idea. |
| 16:45.30 | labdi | hahah |
| 16:45.47 | ghost75 | apb1963: directmedia=no |
| 16:45.50 | labdi | thank you nonetheless |
| 16:47.50 | apb1963_ | what you're seeing for canreinvite... isn't that just in the outgoing context? |
| 16:47.58 | apb1963_ | currently? |
| 16:48.12 | apb1963_ | my incoming doesn't have anything about canreinvite |
| 16:49.22 | p3nguin | labdi: You can use Read() and Authenticate() to auth against a file that you populate with user IDs and md5 passwords, or you can use VMAuthenticate() to compare to the voicemail.conf values. |
| 16:51.03 | [TK]D-Fender | apb1963_, "directmedia=no" <- put it in your peer. I've told you repeatedly. |
| 16:51.31 | [TK]D-Fender | Nobody with a nickname of "apb" should FAIL TO GET THE MEMO |
| 16:51.37 | p3nguin | Only 17 or so more times to go before it happens. |
| 16:52.25 | [TK]D-Fender | apb1963_, canreinvite=no <- this is not appropriate for your ver of *. FreePBX either doesn't know what you're running or has a bug. UPGRADE IT. |
| 16:53.28 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 16:56.39 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 16:57.43 | *** join/#asterisk PipBoy (~PipBoy@66.212.187.33.tor.pathcom.com) |
| 16:58.03 | apb1963_ | ok the extension has canreinvite HARDCODED onto the menu. You can fill in that field with a word, but you cannot change the field name which is "canreinvite". |
| 16:58.10 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:58.10 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:58.39 | apb1963_ | On the peer, yes you can type anything you want. |
| 16:59.28 | apb1963_ | And, I have changed it to DIRECTMEDIA=NO as you suggested. There was little change, if anything it got worse as I seem to have lost my breathing and definitely don't have voice. |
| 16:59.52 | [TK]D-Fender | apb1963_, What have you forwarded to your server? |
| 16:59.54 | apb1963_ | However, the voicemail seems to be the same... functioning having put the qualify=yes and nat=yes int. |
| 16:59.56 | apb1963_ | int |
| 16:59.57 | apb1963_ | in |
| 17:00.06 | Yxa | how do i get adaptive odbc setup? I have already installed mysql and created the relevant table. |
| 17:00.12 | p3nguin | Are the parameters in sip.conf case sensitive? |
| 17:00.12 | apb1963_ | 5060 and 10000-20000 |
| 17:00.18 | [TK]D-Fender | apb1963_, Show us |
| 17:00.28 | [TK]D-Fender | apb1963_, tinypic.com / pastebin.com |
| 17:00.54 | apb1963_ | you're saying you want to see a snapshot of my router? |
| 17:01.41 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 17:02.16 | [TK]D-Fender | yes |
| 17:02.22 | apb1963_ | ok, np |
| 17:04.09 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:04.49 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:04.49 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:10.25 | apb1963_ | two choices.... does this one work? http://tinypic.com/r/35n2em8/6 |
| 17:11.44 | apb1963_ | or this one? http://i46.tinypic.com/35n2em8.jpg |
| 17:12.00 | ghost75 | is this same rtp as in rtp.conf? |
| 17:12.04 | apb1963_ | yes |
| 17:12.36 | ghost75 | my asterisk behind nat works even without port forwarding and nat=no |
| 17:13.37 | apb1963_ | maybe you're not behind nat? go to amibehindnat.com |
| 17:13.49 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.139) |
| 17:15.13 | ghost75 | tells me yes |
| 17:16.35 | apb1963_ | dunno |
| 17:18.33 | ghost75 | nat=no is still using this i think: http://www.iptel.org/wiki/ser/2.1/ref/std/rfc/3581?do=show |
| 17:20.02 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 17:21.31 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:23.25 | navaismo | Hi, my asterisk its crashing and isnt generating a core file, i have enabled the DEBUG_THREAD option when i compiled it. Im using 1.8.11-cert8 with static realtime, in the full log the last thing i can see is the queries to the QUEUE_MEMBERS |
| 17:23.54 | ghost75 | apb1963_ did u try to turn off firewall |
| 17:24.51 | *** join/#asterisk Greenlight (~email@cpc1-dund9-0-0-cust142.16-4.cable.virginmedia.com) |
| 17:25.45 | apb1963_ | you mean on the router? |
| 17:26.11 | ghost75 | yes, maybe its a filter blocking rtp |
| 17:26.13 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 17:28.02 | *** join/#asterisk apb1963_ (~apb1963@174.134.102.14) |
| 17:28.18 | apb1963_ | it's a good idea |
| 17:29.08 | apb1963_ | but it doesn't seem to change anything. Also, the server is in the DMZ so I don't think the firewall affects it?? |
| 17:29.24 | Qwell | if it's DMZed, why are you forwarding ports? |
| 17:29.35 | leifmadsen | heh |
| 17:29.46 | Qwell | You are no longer behind NAT once you DMZ. |
| 17:32.34 | apb1963_ | That's what I had thought at first, but I don't think that's true. My server has a private IP. Server--->Router---> ISP My router gets the public IP and dhcp assigns an IP to the server... How can I not be behind NAT? |
| 17:33.35 | apb1963_ | Plus... amibehindnat.com says I am. So I'd say that was pretty definitive. |
| 17:35.12 | *** join/#asterisk Galen (~Galen@rrcs-24-43-17-237.west.biz.rr.com) |
| 17:35.24 | apb1963_ | I can certainly take out the port forwarding and see if that makes a diff.... |
| 17:35.52 | rowshi | @Qwell: DMZ and NAT relationships depend on the router/firewall you're using. |
| 17:37.25 | *** join/#asterisk apb1963_ (~apb1963@174.134.102.14) |
| 17:37.37 | ghost75 | sip debug should show problems i think |
| 17:38.57 | apb1963_ | now it doesn't ring, but I get voicemail |
| 17:39.28 | apb1963_ | firewall off, forwarding off. |
| 17:40.12 | apb1963_ | keep in mind, iptables is still in effect. |
| 17:40.48 | apb1963_ | it does NOT do any forwarding whatsoever. |
| 17:40.57 | apb1963_ | and I have a feeling that's part of the problem |
| 17:41.24 | apb1963_ | if not the whole problem. |
| 17:42.10 | p3nguin | Most routers don't know how to DMZ things anyway; they still NAT to the DMZ, but they simply forward everything to the DMZ that isn't sent somewhere else. It's pretty useless and should never be used for Asterisk. |
| 17:42.21 | p3nguin | ~dmz |
| 17:42.21 | infobot | [~dmz] De-Militarized Zone, or usually a separate physical or logical network that has limited access to your internal systems and is accessible in limited ways from untrusted networks such as the Internet. Putting Asterisk in the DMZ is not an acceptable alternative to properly forwarding the appropriate ports, so don't do it. Plastic router appliances generally do not implement DMZ well. |
| 17:42.58 | navaismo | Is there a maximun timeout for asterisk when is making queries? seems like the remote DB is taking too long and asterisk died |
| 17:43.58 | *** join/#asterisk apb1963_ (~apb1963@174.134.102.14) |
| 17:45.43 | apb1963_ | Yes... I put it in the DMZ before I understood how NAT affected it and just didn't other taking it back out. |
| 17:45.50 | apb1963_ | +b |
| 17:46.14 | apb1963_ | That's why I forwarded the ports anyway |
| 17:46.42 | apb1963_ | I can take it back out of the DMZ... shouldn't make a difference either way. |
| 17:47.02 | p3nguin | Never put asterisk in the DMZ. Just forward the ports and that is all. |
| 17:47.47 | [TK]D-Fender | apb1963_, Are you on the same network as your server? |
| 17:48.27 | [TK]D-Fender | RUN FORREST RUN!!!\ |
| 17:48.47 | *** join/#asterisk felipealmeida (~user@querubim.tecgraf.puc-rio.br) |
| 17:48.47 | *** join/#asterisk apb1963_ (~apb1963@174.134.102.14) |
| 17:49.04 | apb1963_ | . |
| 17:49.38 | apb1963_ | Yes |
| 17:49.51 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:49.51 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:50.03 | apb1963_ | its a private network |
| 17:50.10 | apb1963_ | there are two machines on it |
| 17:50.16 | apb1963_ | one real, one virtual |
| 17:50.35 | apb1963_ | the virtual one runs centOS and asterisk |
| 17:50.52 | ghost75 | (18:39:43) apb1963_: now it doesn't ring, but I get voicemail <- from where? |
| 17:51.00 | apb1963_ | from asterisk |
| 17:51.21 | [TK]D-Fender | apb1963_, Are you on the same network as your server? |
| 17:51.25 | apb1963_ | yes |
| 17:51.37 | [TK]D-Fender | * [apb1963_] (~apb1963@174.134.102.14): |
| 17:51.46 | [TK]D-Fender | [2012-12-05 07:12:17] VERBOSE[4420] chan_sip.c: Reliably Transmitting (NAT) to 174.134.102.14:1726: |
| 17:51.50 | apb1963_ | That's my public IP... of the router |
| 17:51.53 | [TK]D-Fender | Contact: <sip:Unknown@50.23.197.95:5060> |
| 17:52.06 | [TK]D-Fender | And what your told ASTERISk the IP was does not look at all the same |
| 17:52.26 | apb1963_ | 50.* is a different provider |
| 17:52.52 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 17:53.07 | apb1963_ | but let me doublecheck on that |
| 17:54.35 | apb1963_ | ok that's the nameserver |
| 17:54.46 | apb1963_ | freeDNS |
| 17:55.33 | [TK]D-Fender | You did NOT tell Asterisk your WAN IP. |
| 17:55.39 | [TK]D-Fender | ~soso |
| 17:55.39 | infobot | [~soso] Shoot-On-Sight Offense |
| 17:55.41 | [TK]D-Fender | ^ |
| 17:56.14 | [TK]D-Fender | Fix your WAN IP settings in * |
| 17:56.17 | p3nguin | Oh boy. Bad DNS is just as bad as statically programming the wrong IP address. |
| 17:57.15 | p3nguin | If your public address is assigned dynamically, be sure your dynamic DNS host name is updated! |
| 17:57.42 | apb1963_ | I think I know which setting that is, and if so yeah.. I was wondering about that. I was told to use the DNS server for that setting. I thought it was odd but.... |
| 17:57.51 | apb1963_ | what do I know? |
| 17:59.35 | apb1963_ | jeez... now I can't connect to the server |
| 18:00.13 | p3nguin | I'm sure the person didn't mean the DNS server's hostname/address. |
| 18:01.14 | PipBoy | When directing a call over a PBX A to PBX B . Does PBX b check its inbound routes to see if it can handle the call? or does it just check its extension list? |
| 18:01.18 | *** join/#asterisk mogra (477b818a@gateway/web/freenode/ip.71.123.129.138) |
| 18:01.30 | p3nguin | "routes" are something in your mind. |
| 18:01.43 | p3nguin | A SIP call will be matched against the peers that are configured in sip.conf. |
| 18:01.48 | carrar | s/mind/router/ |
| 18:01.57 | PipBoy | the routrix has you neo? |
| 18:02.09 | *** join/#asterisk apb1963_ (~apb1963@174.134.102.14) |
| 18:02.11 | PipBoy | but thanks p3nguin that awnsered my question |
| 18:02.22 | p3nguin | If there is no match, and, if you allow anonymous calls, the call will be sent to the context configured in the general section. |
| 18:02.35 | PipBoy | Ah ok |
| 18:02.42 | p3nguin | If a peer is matched, the context configured for that peer will be used. |
| 18:02.53 | p3nguin | If you have two PBXs, you most certainly have a peer configured for it. |
| 18:03.00 | PipBoy | of course |
| 18:03.13 | p3nguin | http://pastebin.com/Ag7tknm2 |
| 18:03.41 | PipBoy | I am just missing something small. Its funny what things I can get stuck on |
| 18:03.44 | PipBoy | thanks again p3nguin |
| 18:04.13 | *** join/#asterisk apb1963_ (~apb1963@174.134.102.14) |
| 18:06.23 | apb1963_ | unreal... i can't connect to the server anymore |
| 18:06.56 | apb1963_ | time for a reboot I think |
| 18:07.45 | *** join/#asterisk DarthExpeditor (~IceChat9@rrcs-71-43-76-226.se.biz.rr.com) |
| 18:08.18 | apb1963_ | too many changes to the router I think |
| 18:09.43 | *** join/#asterisk j4m3s_ (~j4m3s_@pdpc/supporter/active/j4m3s) |
| 18:10.07 | [TK]D-Fender | apb1963_, You're on the same LAN... why is a ROUTER even in play? |
| 18:11.08 | apb1963_ | somebody told me to go buy one |
| 18:11.51 | PipBoy | geeksquad? |
| 18:12.01 | jmetro | ^ |
| 18:12.06 | apb1963_ | someone in ... I believe it was #networking |
| 18:12.41 | apb1963_ | I was having a problem with not being able to traceroute to a device on my provider's network. |
| 18:12.59 | apb1963_ | turned out to be something to do with UDP vs ICMP |
| 18:13.02 | navaismo | alo alo ¬¬ |
| 18:13.07 | apb1963_ | so no... I don't think I need the router |
| 18:13.11 | p3nguin | You obviously had a router before. |
| 18:13.17 | apb1963_ | before... what? |
| 18:13.27 | PipBoy | before you bought a new router |
| 18:13.28 | p3nguin | Before you bought the router that you were told to get. |
| 18:13.29 | apb1963_ | No |
| 18:13.46 | apb1963_ | I was just fine and dandy w/out one. |
| 18:13.58 | apb1963_ | Life was good |
| 18:14.03 | apb1963_ | well.. mostly |
| 18:14.16 | PipBoy | mama always said the worlds your lan |
| 18:14.20 | p3nguin | If you had two computers with private addresses and had internet service from a provider, how did you not have a router? |
| 18:14.56 | apb1963_ | Originally my ISP was giving out IPs freely. So, I had two. One for my real machine, and one for the virtual guest OS. |
| 18:15.23 | apb1963_ | Then along came the big bad wolf technician who said "Gee, that's weird... you're only supposed to have one IP" and that's when my world collapsed. |
| 18:15.45 | apb1963_ | So, I turned on NAT on my virtual machine. |
| 18:15.58 | apb1963_ | And life was mostly good again. |
| 18:16.12 | apb1963_ | Except... one of my user/extensions overseas couldn't register with asterisk. |
| 18:16.13 | p3nguin | So then that vm became your router. |
| 18:16.28 | *** join/#asterisk k610 (~K610@host-78-129-3-116.brutele.be) |
| 18:16.30 | p3nguin | Extensions and users don't register. Devices register. |
| 18:16.52 | apb1963_ | So, I ran a traceroute and it stopped on a particular IP address before reaching my machine. |
| 18:17.02 | apb1963_ | That address was owned by my provider. |
| 18:17.06 | p3nguin | Is asterisk on that virtual machine? |
| 18:17.13 | apb1963_ | Yes. CentOS |
| 18:17.37 | apb1963_ | Long story short, by telling traceroute to use ICMP instead of UDP the trace was completed. |
| 18:17.48 | p3nguin | CentOS can act as a fine router as long as you know how to configure a handful or less of iptables rules. |
| 18:17.48 | PipBoy | also note. that a LOT of networks dont allow ICMP traffic through them |
| 18:17.53 | apb1963_ | But I didn't figure that out until after I bought the router this past weekend. |
| 18:18.26 | p3nguin | What kind of router is it? |
| 18:18.36 | PipBoy | waaaaait.. someone from #networking told you to buy a new router because you couldnt ping something? maaaaaan thats rough lol |
| 18:18.40 | apb1963_ | So one guy thought I needed a router... another guy knew that UDP was the culprit and suggested I force traceroute to use ICMP and that worked. |
| 18:19.27 | apb1963_ | People make mistakes. |
| 18:19.59 | apb1963_ | But I figured as long as I had it, I might as well give it a tryout. |
| 18:20.11 | p3nguin | What kind of router is it? |
| 18:20.13 | *** join/#asterisk j4m3s_ (~j4m3s_@pdpc/supporter/active/j4m3s) |
| 18:20.19 | apb1963_ | Belkin N150 |
| 18:20.24 | p3nguin | O M G |
| 18:20.41 | apb1963_ | I don't like their software... or at least certain parts of it. |
| 18:20.42 | p3nguin | Never ever ever use Belkin if asterisk is involved. |
| 18:21.06 | p3nguin | I've never seen a Belkin that works with SIP. Ever. |
| 18:21.07 | apb1963_ | Whys that? |
| 18:21.09 | apb1963_ | oh |
| 18:21.12 | apb1963_ | ok |
| 18:21.13 | PipBoy | p3nguin I sence some pent up fustration there :P |
| 18:21.13 | apb1963_ | well |
| 18:21.40 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:22.22 | p3nguin | Honestly, I would take it back and get something else that is known to work fine with SIP, or I would take it back and get money for it. Then I would reconfigure a computer that I have to be my router. |
| 18:22.59 | p3nguin | It's only a few commands to make a router with NAT. |
| 18:26.19 | jmetro | i never use belkin period. or netgear. |
| 18:27.06 | apb1963_ | so what's the command to tell asterisk my wan? |
| 18:27.41 | apb1963_ | externhost? |
| 18:27.42 | p3nguin | In sip.conf, use externaddr or externhost. |
| 18:28.00 | p3nguin | externaddr if you have a static address, externhost if you use dynamic DNS. |
| 18:28.22 | PipBoy | p3nguin if you got a minute I was wonderin about something else. Lets say I have a trunk between pbx A and pbx b. In freePBX i have an inbound route that directs an external call to go over that Cross site trunk. Once on the other side it checks if it has a peer that matches the inbound call.. So in that case, where in asterisk would I find a list of peers? should there not be a list on |
| 18:28.23 | PipBoy | phone number that I would of placed into "inbound routes" |
| 18:28.53 | p3nguin | Asterisk has sip peers in sip.conf. |
| 18:29.27 | p3nguin | And I don't do FreePBX. |
| 18:29.43 | PipBoy | Ok I understand that. I am asking more so about the asterisk side |
| 18:30.03 | p3nguin | sip.conf is where you configure sip peers. |
| 18:30.37 | PipBoy | So lets say the call comes over the trunk and matches one of my peers in sip.conf. But you wouldnt find a DID in sip.conf would you? |
| 18:30.43 | p3nguin | Any phone/device/ITSP, or even another of your own asterisk systems... all are peers and get configured in sip.conf. |
| 18:30.47 | p3nguin | If you're talking trunk, sip does not do trunking. |
| 18:31.00 | p3nguin | If you want trunks, you'll use IAX2 and iax.conf. |
| 18:31.08 | PipBoy | Ah I just worded it wrong |
| 18:31.43 | p3nguin | And an IAX2 trunk between two asterisk systems is a great idea in many cases. |
| 18:32.10 | PipBoy | Yea I have used IAX at a previous job. Did the trick |
| 18:32.27 | PipBoy | But yea, lets say it does match a Peer in sip.conf. Again, I dont believe I have ever seen a DID in sip.conf . Where would the DID's be referenced? |
| 18:32.43 | p3nguin | DIDs are just phone numbers. The calls are destined to extensions. The call matches a peer, which has a context assigned to it, which tells the call where to look for that extension. |
| 18:32.57 | p3nguin | Extensions obviously go in extensions.conf. |
| 18:33.08 | *** join/#asterisk j4m3s_ (~j4m3s_@pdpc/supporter/active/j4m3s) |
| 18:34.10 | PipBoy | Ok I understand the relation between a peer and what context is applied to the peer. But the DID has to point to a destination at some point, and I cant quiet understand where that happens |
| 18:34.23 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-yeyktkhwfnwiumqx) |
| 18:34.38 | [TK]D-Fender | PipBoy, A DID *is* a destination |
| 18:34.44 | p3nguin | So if your peer sends a call to 3145551212, the call first matches the peer entry, which has a context of, let's say, inbound. Under the inbound context, you need to have extension 3145551212 or a pattern that matches 3145551212 for it to work. |
| 18:34.50 | [TK]D-Fender | pigpen, it is the very number they dialed to reach you |
| 18:34.55 | [TK]D-Fender | PipBoy, ^ |
| 18:35.17 | p3nguin | DIDs are just phone numbers. Extensions match those phone numbers when calls are sent in to asterisk. |
| 18:35.17 | [TK]D-Fender | PipBoy, Typically it should be passed as such and you process it in the dialplan (extensions.conf) |
| 18:35.41 | PipBoy | Well, not always Fender. In my case, I am redirecting an inbound call to a second PBX.. and the second PBX could not match it to anything |
| 18:35.42 | p3nguin | And, as mentioned, those extensions to match your phone numbers are in extensions.conf. |
| 18:35.53 | PipBoy | thats exactly what i wanted to know thanks |
| 18:35.59 | p3nguin | You didn't configure a peer, a context, or an extension correctly. |
| 18:36.04 | p3nguin | The call has to match in that order. |
| 18:36.10 | p3nguin | First a peer must match. |
| 18:36.25 | PipBoy | It matches the peer. The issue is after that, and I wasnt certain where to look |
| 18:36.38 | p3nguin | Then the peer sends to call to an extension within a context. If the context isn't there, it dies. |
| 18:36.47 | p3nguin | If the extension isn't in the context where the call went, it dies. |
| 18:36.50 | p3nguin | EXTENSIONS |
| 18:36.54 | p3nguin | You need to configure an extension. |
| 18:37.11 | PipBoy | I understand, I was just clarifying my quesiton. I appreciate the help |
| 18:37.19 | [TK]D-Fender | <PipBoy> Well, not always Fender. In my case, I am redirecting an inbound call to a second PBX.. and the second PBX could not match it to anything <- this doesn't tell us how you passed this call off. |
| 18:37.42 | [TK]D-Fender | PipBoy, There are a LOT of different common ways you MIGHT have done this and the means by which it'd be matched would vary |
| 18:37.43 | p3nguin | I'm guessing there was no extension to match the call when it goes there. |
| 18:37.48 | PipBoy | exactly |
| 18:38.01 | p3nguin | s/goes/got/ |
| 18:38.42 | PipBoy | I agree, I am going to check my extensions now. I just was not aware that DID's were listed that way in Asterisk |
| 18:39.12 | p3nguin | The problem is that FreePBX suggests that phones are extensions. |
| 18:39.41 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 18:39.48 | [TK]D-Fender | PipBoy, there is no "list" A DID is just a number dialed. How it gets passed may vary. As you are going from one server to another this would depend entirely on how you set those 2 servers up with each other |
| 18:39.51 | p3nguin | If it said that DIDs are phone numbers and phone numbers match extensions, it would save me a lot of grief. |
| 18:40.02 | p3nguin | ~did |
| 18:40.02 | infobot | i heard did is Direct Inward Dialing, or just a phone number |
| 18:40.14 | PipBoy | thats the word on the street yo |
| 18:41.19 | WIMPy | Didn't we find out a long time ago that a simple directory number is actualy non-DID? |
| 18:41.45 | p3nguin | What do you mean by simple directory number? |
| 18:42.03 | PipBoy | Unfortunately I started my Career working with a lot of FreePBX. And although I do work in regular asterisk now, some things are a bit cloudy |
| 18:42.17 | WIMPy | Just a plain tlephone numbner. |
| 18:42.40 | [TK]D-Fender | PipBoy, This is still just one server of yours connecting to another. This isn't "really" about DID's at all in the raw-er sense |
| 18:42.57 | [TK]D-Fender | PipBoy, And you don't seem to ahve told us how you set those two system up with each other at all |
| 18:43.28 | PipBoy | Ah sorry, just a "sip trunk" |
| 18:43.45 | PipBoy | or at least thats what freepbx calls it |
| 18:43.46 | p3nguin | SIP doesn't trunk. |
| 18:43.53 | PipBoy | hehe I knew you were going to say that |
| 18:43.59 | p3nguin | So that's pretty silly to call it that. |
| 18:44.19 | PipBoy | I am a victim in this! lol . I wish I would of learned correctly the first time |
| 18:44.27 | p3nguin | would have? |
| 18:44.31 | *** join/#asterisk fakhir_ (~fakhir@unaffiliated/fakhir) |
| 18:45.02 | WIMPy | Sip may be pants, but it doesn't trunk. LOL. |
| 18:45.31 | PipBoy | Would you define "trunking" by how it deals with signaling ? |
| 18:45.43 | p3nguin | ~trunk |
| 18:45.44 | infobot | well, trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
| 18:45.59 | PipBoy | haha excellent |
| 18:46.18 | apb1963_ | I packed my trunk with camping equipment. |
| 18:46.44 | apb1963_ | anyway... externhost didn't help any |
| 18:47.06 | apb1963_ | so, I'm going to disconnect my router and return it. |
| 18:47.43 | p3nguin | What value did you configure for the externhost parameter? |
| 18:47.46 | apb1963_ | don't anyone try to stop me |
| 18:48.03 | apb1963_ | ummm... asterisk.saveabunny.com |
| 18:48.22 | *** join/#asterisk ayrjola (~androirc@77-105-67-106.lpok.fi) |
| 18:48.31 | p3nguin | 174.134.102.14? |
| 18:48.35 | apb1963_ | Yes |
| 18:48.43 | p3nguin | Perfect. |
| 18:48.45 | apb1963_ | that's how freedns knows it |
| 18:48.47 | apb1963_ | me |
| 18:48.50 | apb1963_ | it |
| 18:48.55 | p3nguin | Did you remember to run sip reload after making the change to sip.conf and saving it? |
| 18:49.01 | apb1963_ | yes |
| 18:49.18 | apb1963_ | I have a memory like an elephant |
| 18:49.23 | apb1963_ | that's getting old and dying |
| 18:49.37 | apb1963_ | with a trunk |
| 18:49.40 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 18:49.45 | apb1963_ | filled with camping equpment. |
| 18:49.56 | apb1963_ | :D |
| 18:50.05 | apb1963_ | and a router |
| 18:50.28 | [TK]D-Fender | So basically you're a ton-ton. |
| 18:50.35 | apb1963_ | more like a won-ton |
| 18:50.39 | [TK]D-Fender | And I thought they smelled bad on the OUTSIDE.... |
| 18:50.39 | PipBoy | Do you have any sip in your trunk? lol |
| 18:50.50 | apb1963_ | no but I took a sip with my trunk |
| 18:50.58 | PipBoy | haha |
| 18:51.23 | apb1963_ | have you any eggrolls? |
| 18:51.26 | PipBoy | the funny thing is... that will help me remember that sip does not trunk lol |
| 18:51.37 | apb1963_ | i'm big on mnemonics |
| 18:51.40 | [TK]D-Fender | But trunk sips? |
| 18:51.51 | apb1963_ | You can't sip from your trunk |
| 18:52.18 | PipBoy | I do not have any sip in my trunk, but I have taken a sip with my trunk.. Should be in a Kids book for IT people |
| 18:52.21 | p3nguin | I certainly cannot sip from YOUR trunk. |
| 18:52.30 | apb1963_ | leave my trunk alone |
| 18:52.40 | apb1963_ | it's very sensitive |
| 18:52.48 | apb1963_ | if you tickle it, I'll sneeze |
| 18:53.00 | apb1963_ | so anyway, back in the real world. |
| 18:53.17 | *** join/#asterisk wonderworld (~w@dsdf-4d0a0d39.pool.mediaWays.net) |
| 18:54.02 | PipBoy | real world?! now im sad :( |
| 18:54.47 | apb1963_ | it happens |
| 18:54.51 | apb1963_ | I gotta pay some bills |
| 18:54.53 | apb1963_ | like my rent |
| 18:55.13 | PipBoy | Me too, and fix some PBX's lol |
| 18:55.20 | apb1963_ | and since I have to go to the bank, I'll be right next door to radio shack, so it would be a good idea to return the router at this time. |
| 18:55.30 | apb1963_ | otherwise I'll end up stuck with it |
| 18:55.37 | apb1963_ | because I'm terrible at returning things |
| 18:55.43 | Qwell | You bought something from Radio Shack? |
| 18:55.49 | p3nguin | I didn't even know they sold routers. |
| 18:55.54 | apb1963_ | That's why I still have an elephant in the back yard. |
| 18:56.05 | p3nguin | I thought they only had shitty RC cars and lots of batteries. |
| 18:56.14 | apb1963_ | Yeah... they sell cell phones too |
| 18:56.18 | apb1963_ | by the seashore. |
| 18:56.38 | apb1963_ | no they have RC helicoptors too |
| 18:56.43 | apb1963_ | and nanobots |
| 18:56.59 | apb1963_ | and occasionally a woman or two |
| 18:57.04 | apb1963_ | but not for sale usually |
| 18:57.26 | jmetro | only as sales people |
| 18:57.31 | jmetro | just like gamestop. |
| 18:57.33 | apb1963_ | mostly |
| 18:58.10 | apb1963_ | so... time to disconnect the old....new router. |
| 18:58.40 | apb1963_ | if I'm not back soon.... then it's a routing problem :) |
| 19:00.22 | PipBoy | I will avenge you at the nearest Radio Shack |
| 19:00.48 | PipBoy | which is pretty far away considering im in Canada |
| 19:01.01 | *** join/#asterisk apb1963__ (~apb1963@174.134.102.14) |
| 19:01.13 | apb1963__ | . |
| 19:01.17 | apb1963__ | I'm back?? |
| 19:01.21 | apb1963__ | wow |
| 19:01.24 | apb1963__ | that was easy |
| 19:01.26 | Qwell | PipBoy: The Source? |
| 19:01.45 | PipBoy | Yea.. the source is friggin dumb :S The "IT GUY" didnt even know what thermal paste was |
| 19:01.46 | apb1963__ | of course now I get a busy signal |
| 19:01.47 | PipBoy | like wth |
| 19:01.47 | apb1963__ | sigh |
| 19:02.54 | apb1963__ | we shall overcome |
| 19:04.12 | apb1963__ | wow... 2700 ms to get from virtual host to real host |
| 19:04.50 | *** join/#asterisk cervajs2 (~cervenka@gatekeeper.bm.ipex.cz) |
| 19:05.09 | apb1963__ | oh much better now |
| 19:05.16 | SuperNull | what are you using for virtualization ? (i missed some of this) |
| 19:05.31 | apb1963__ | 31 ms |
| 19:05.34 | apb1963__ | on the high side |
| 19:05.42 | apb1963__ | vmware |
| 19:05.46 | cervajs2 | hello, which srtp library i must use for DTLS-SRTP? (i'm trying webrtc) |
| 19:05.52 | SuperNull | esx ? |
| 19:06.01 | apb1963__ | Player |
| 19:06.05 | SuperNull | ahhh ic. |
| 19:06.15 | apb1963__ | it's actually pretty nice |
| 19:06.16 | SuperNull | esxi is def miles away from player and vmware server. |
| 19:06.21 | SuperNull | i like it for dev .. |
| 19:06.28 | apb1963__ | It does what I need for the most part |
| 19:06.41 | apb1963__ | though the jury is still out on the networking :) |
| 19:07.01 | apb1963__ | busy signal |
| 19:07.06 | SuperNull | doesnt have the crazy features esxi does but ..it also doesnt have the price tag for a full version (esxi without vSphere is free with limitation) |
| 19:07.24 | apb1963__ | Player is free w/out limitation |
| 19:07.56 | apb1963__ | so now I new IP addresses |
| 19:08.01 | apb1963__ | +have |
| 19:08.16 | apb1963__ | and a busy signal |
| 19:08.19 | p3nguin | Let us hope that your dns updater is working! |
| 19:08.35 | apb1963__ | well lets think about that |
| 19:09.12 | apb1963__ | the dns points to the address my ISP assigns through DHCP. That hasn't changed. |
| 19:09.31 | apb1963__ | Only the address of my virtual machine has changed. |
| 19:09.37 | apb1963__ | Which is what runs asterisk. |
| 19:09.53 | cervajs2 | when i try dtlsenable = yes in sip.conf then i get this message chan_sip.c: No DTLS-SRTP support present on engine for RTP instance '0xb7520c74', was it compiled with support for it? |
| 19:10.53 | p3nguin | Oh, I thought you meant that your public address changed. |
| 19:10.55 | apb1963__ | so before, my router had that public address... now my real machine/host has it. |
| 19:11.13 | file | cervajs2, DTLS-SRTP isn't used yet for WebRTC just SDES negotiated SRTP |
| 19:11.16 | p3nguin | Since any servers inside the LAN would be assigned with static addresses, that was the only part that COULD have changed. |
| 19:11.18 | apb1963__ | and the host is now set to do NAT |
| 19:11.58 | apb1963__ | Nope. vmware changed the address of the virtual machine and interfaces. |
| 19:12.28 | apb1963__ | It's still a private address.... but it's a different private address. |
| 19:13.18 | cervajs2 | file: can you explain it more? asterisk-asterisk works with DTLS-SRTP but not with websockets? or what is mean by "DTLS-SRTP support" new in asterisk 11 |
| 19:13.44 | apb1963__ | yeah so now the host is the public ip |
| 19:13.46 | file | no WebRTC capable browsers currently implement DTLS-SRTP |
| 19:13.59 | apb1963__ | hmmmmm |
| 19:16.49 | cervajs2 | file: tnx. i'll check chromium changelog |
| 19:17.15 | apb1963__ | look ma, no router! |
| 19:17.37 | cervajs2 | file: i'm testing sipml5 + asterisk 11. everything looks ok, but no audio :( |
| 19:18.00 | file | Google made the SDP stuff more strict and they haven't finished writing it |
| 19:18.14 | file | so it may or may not be broken, depending on the version of Chrome |
| 19:19.53 | cervajs2 | file: its shame because i have tomorrow presentation about asterisk history for 50 people from telco industry :( |
| 19:20.30 | file | WebRTC: The definition of a moving target. |
| 19:20.37 | cervajs2 | :) |
| 19:20.59 | cervajs2 | i'll try speak something good about it :) |
| 19:22.14 | *** join/#asterisk mbrit (~mbrit@186.120.97.194) |
| 19:22.29 | cervajs2 | i tried chrome 23 and canary 25 on win7 |
| 19:23.16 | p3nguin | Maybe firefox 61 has something useful in it. |
| 19:23.31 | p3nguin | They are up to 61 already, aren't they? |
| 19:23.55 | *** join/#asterisk Dovid (~Dovid@host-78-158-94-201.wlan-guest.nycmny02.us.sargasso.net) |
| 19:28.44 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 19:48.49 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 19:51.33 | *** join/#asterisk areski (~areski@80.174.255.87.dyn.user.ono.com) |
| 19:53.39 | *** join/#asterisk ageis (kevin@ageispolis.net) |
| 19:54.03 | ageis | I got phones that like to disconnect themselves and we constantly have to reboot them... Any one know how to implement QoS so SIP packets are given priority or something? |
| 19:54.29 | Qwell | if QoS solves that for you, you've got much larger issues. |
| 19:55.07 | ageis | i'm just brainstorming, I want to increase the reliability of the ethernet connectivity |
| 19:56.19 | PipBoy | run a sepperate network :P thats the best qos :P |
| 19:56.27 | jmetro | a VPN |
| 19:57.45 | *** join/#asterisk wonderworld (~w@dsdf-4db556a8.pool.mediaWays.net) |
| 20:03.24 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 20:04.38 | [TK]D-Fender | VPN doesn't magically make your packets any more reliable. |
| 20:06.23 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.184) |
| 20:07.41 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 20:08.20 | file | drmessano, hi |
| 20:08.50 | file | drmessano, I approve of this stuff and things |
| 20:09.21 | *** join/#asterisk corretico (~luis@190.211.93.38) |
| 20:10.02 | Qwell | drmessano: ^^ what he said |
| 20:11.30 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 20:15.42 | *** join/#asterisk solmsted (~solmsted@pool-71-251-234-174.rcmdva.fios.verizon.net) |
| 20:16.10 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 20:18.27 | *** join/#asterisk planezit (~gaston@ool-45787011.dyn.optonline.net) |
| 20:19.16 | PipBoy | I am trying to figure out if a call is going through my PBX to PBX trunk or going back through the pstn.. If core show channels shows nothing, dosnt that mean it just sent it back out through the pstn? |
| 20:20.32 | planezit | You should see the sip traffic in the asterisk console |
| 20:20.55 | planezit | Sip connects and disconnects |
| 20:21.26 | [TK]D-Fender | pigpen, if you see no channels then there is no channel going on at your server at all. |
| 20:21.49 | [TK]D-Fender | pigpen, Maybe it died completely. |
| 20:22.22 | [TK]D-Fender | pigpen, "a call is going through my PBX to PBX trunk or going back through the pstn" <- doesn't offer us any details for even aneducated guess (at best)\ |
| 20:22.30 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 20:22.40 | PipBoy | thanks I appreciate the nicknames |
| 20:22.49 | PipBoy | thats not childish at all |
| 20:23.14 | jmetro | i was gonna say, i'm pretty sure its pipboy. Like from fallout. |
| 20:23.24 | PipBoy | correct |
| 20:23.55 | jmetro | i think he just transposed..there's literally someone named pigpen here right above you on the list. |
| 20:24.26 | [TK]D-Fender | auto-correct SNAFU |
| 20:24.26 | PipBoy | ah I see.. I figured he was going after the fact that my question was so messy lol |
| 20:25.01 | [TK]D-Fender | autocomplete even.... |
| 20:25.28 | PipBoy | *sigh* I figured out my question either way. Honestly, I think I am going to give up on FreePBX. It just convolutes things if you are trying to do something custom |
| 20:25.30 | ghost75 | apb1963_ so its working now? |
| 20:26.23 | [TK]D-Fender | PipBoy, what kind of "custom" anyway? You have giving virtually nothing so far and all of your questions seem to emanating from "Generic Land" |
| 20:27.42 | PipBoy | Ok I will be specific then. I have a phone system in production and I was to add improved CDR tracking (pretty graphs and all that jazz). So what I would like to do is create a second PBX to sit in the middle of all calls, gathering call information, writing to mysql and all that.. |
| 20:28.23 | PipBoy | So my middleman PBX is configured to recieve all the calls, and the thought is that it just passes all the calls along to the production system VIA a sip Peer. |
| 20:29.17 | [TK]D-Fender | PipBoy, How is this intermediary's CDR any better than what's on the main? |
| 20:29.19 | PipBoy | All I am trying to do is simply take an inbound call on one system, and pass it along through a local SIP peer to a production system |
| 20:29.34 | PipBoy | CDR-stats . Its a nice program, check it out |
| 20:29.41 | [TK]D-Fender | PipBoy, And I take it that FreePBX is what you've set up on your intermediary, correct? |
| 20:29.53 | PipBoy | Yes, and its a pain in the ass lol |
| 20:30.10 | [TK]D-Fender | This is a petty job and doesn't require anything "custom" |
| 20:30.22 | PipBoy | Id agree, I think I was over thinking this |
| 20:31.10 | jpsharp | You need a base install of Asterisk, cdr_mysql (or cdr_odbc), and about 4 lines in extensions.conf |
| 20:31.21 | [TK]D-Fender | trunk in -> point to from-internal. Set up outbound route using trunk to other PBX. The end |
| 20:31.26 | [TK]D-Fender | Should have taken all of 5 minutes tops |
| 20:31.45 | [TK]D-Fender | That's being generous.... |
| 20:31.47 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 20:32.01 | PipBoy | I know >.< I set it up like that in 5 minutes.. But something went wrong and I started exploring a zillion options :/ |
| 20:32.32 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-vuhrbiratulcqusp) |
| 20:32.45 | jpsharp | FreePBX will just get in your way here. |
| 20:34.01 | PipBoy | Id agree.. But I had a lot of issues setting up asterisk writting to Mysql. And didnt want to really take the time to figure it out. Where as the FreePBX distro (although bulky). Should of taken me about an hour to do the whole install with setup |
| 20:34.15 | p3nguin | should have |
| 20:34.24 | PipBoy | I am the weakest link here.. Not freepbx lol |
| 20:34.40 | [TK]D-Fender | PipBoy, 11 steps to go.... |
| 20:34.48 | PipBoy | Ha! jerk |
| 20:35.13 | [TK]D-Fender | PipBoy, I've acknowledged your PROGRESS. Don't balk at that.... |
| 20:35.49 | [TK]D-Fender | PipBoy, So, got something "actual" you'd like some real help with? |
| 20:36.13 | PipBoy | Well, I guess I will set up everything the simple way again.. And go from there |
| 20:36.40 | [TK]D-Fender | PipBoy, We'll be waiting... |
| 20:36.47 | PipBoy | thanks :) |
| 20:37.56 | wonderworld | [TK]D-Fender: what would be the correct way to set up this routing through a second pbx? just asking out of interest |
| 20:38.49 | [TK]D-Fender | wonderworld, I just described it... |
| 20:39.11 | wonderworld | yeah, i didn't get it |
| 20:39.22 | [TK]D-Fender | <[TK]D-Fender> trunk in -> point to from-internal. Set up outbound route using trunk to other PBX. The end |
| 20:39.27 | [TK]D-Fender | wonderworld, What part? |
| 20:40.02 | [TK]D-Fender | Only 17 words there. at least 3 non-essential |
| 20:40.09 | wonderworld | hehe |
| 20:40.30 | wonderworld | how would it look in the dialplan? |
| 20:40.44 | wonderworld | is there no rewriting of IPs etc needed? |
| 20:40.55 | [TK]D-Fender | wonderworld, rewrint IP's? Pardon? |
| 20:40.58 | jpsharp | No. Asterisk is a middleman, not a proxy. |
| 20:41.02 | [TK]D-Fender | wonderworld, * is not a PROXY. |
| 20:41.28 | wonderworld | ok, lets clarify my confusion |
| 20:41.39 | [TK]D-Fender | wonderworld, And no "dialplan" involved. (manually speaking). This is a conversation concerning FREEPBX is you somehow missed that aspect. |
| 20:41.44 | [TK]D-Fender | if* |
| 20:42.33 | wonderworld | a call is coming in through the trunk, wanting to reach an extension on the local * but it needs to go to the other * |
| 20:42.39 | [TK]D-Fender | wonderworld, If you're wondering what a "vanilla config" would look like, the dialplan is just about as dumb..... only a tiny handful of lines at most. |
| 20:43.18 | [TK]D-Fender | wonderworld, There is no extension on local. It's only purpose is to be an intermediary between his actual provider & his main server |
| 20:43.28 | wonderworld | ok, if there is no dialplan involved, how would that routing be setup? |
| 20:43.40 | [TK]D-Fender | wonderworld, Immediate passthrough. no other functionality. Call in. Call out. record CDR for reporting. |
| 20:43.40 | wonderworld | thats waht i don't understand |
| 20:43.49 | [TK]D-Fender | wonderworld, FREEPBX <------- |
| 20:44.05 | [TK]D-Fender | <[TK]D-Fender> <[TK]D-Fender> trunk in -> point to from-internal. Set up outbound route using trunk to other PBX. The end <--- OUTBOUND ROUTE |
| 20:44.26 | wonderworld | ok, this is freepbx specific, i see |
| 20:44.33 | wonderworld | how would it be done with two * boxes? |
| 20:44.59 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:45.26 | [TK]D-Fender | wonderworld, Since it's call in, call out, it really only needs 1 line of dialplan anyway. |
| 20:45.46 | [TK]D-Fender | For a single direction. The reverse for the other. |
| 20:45.51 | jpsharp | Wouldn't you need...yes. |
| 20:45.57 | [TK]D-Fender | So wat ... 4 lines including context headers? |
| 20:47.22 | jpsharp | Hence my comment about freepbx just getting in the way. |
| 20:48.42 | [TK]D-Fender | jpsharp, Side benefit for it potentially having been installed in a way that pre-configures CDR storage to DB as he likes and maybe includes if not simplifies the installation of the reporting he's looking for. |
| 20:48.58 | PipBoy | both true |
| 20:49.08 | [TK]D-Fender | jpsharp, "depends". From what *I* would do... yes inserting a whole box SOUNDS remarkably stupid. |
| 20:49.18 | wonderworld | well thanks for explaining... |
| 20:49.47 | jpsharp | Eh, I've never had problems with cdr_mysql. :) |
| 20:49.59 | [TK]D-Fender | wonderworld, exten => _X.,1,Dial(SIP/otherserver/${EXTEN}) |
| 20:50.07 | [TK]D-Fender | wonderworld, There's HALF the code right there.... |
| 20:50.29 | PipBoy | Well inserting the whole box isnt that dumb of an idea. Testing out a product without messing around too much with an in production system? |
| 20:52.37 | jacekowski | i've got a problem with users complaining about echo |
| 20:52.43 | [TK]D-Fender | PipBoy, You know how hard it is to take a CSV file on that same server and just dump it into a SQL table on that box and drop the webscripts in to "see" data processing? Petty |
| 20:52.51 | jacekowski | on any call, including pure sip calls |
| 20:53.03 | jacekowski | and i can't hear anything wrong with it myself |
| 20:53.19 | wonderworld | [TK]D-Fender: thanks "SIP/otherserver" was what i falsely described with "ip rewriting" |
| 20:53.31 | [TK]D-Fender | wonderworld, If you say so... |
| 20:54.03 | wonderworld | hearing your discussion i thought there was something like a routing.conf ..... |
| 20:54.48 | jpsharp | jacekowski: All users or just one or two? |
| 20:55.43 | PipBoy | D-Fender . I appreciate your feedback. But I am neck deep in projects I have to complete. Its very tempting to take the path of least resistance when available |
| 20:56.57 | *** join/#asterisk Alex25 (~kvirc@bzq-79-176-209-240.red.bezeqint.net) |
| 20:58.08 | Alex25 | Do you knwo a way to hangup a specific extension using bash CLI? |
| 20:59.17 | wonderworld | Alex25: you can use asterisk -x "command" from BASH |
| 20:59.21 | [TK]D-Fender | Alex25, "channel request hangup [thechannel]" |
| 20:59.41 | Alex25 | yea but this is for a channel |
| 20:59.46 | [TK]D-Fender | Alex25, "channel request hangup [thechannel]" <--------- |
| 20:59.54 | Alex25 | I need it for an extension |
| 21:00.00 | [TK]D-Fender | no such thing |
| 21:00.01 | [TK]D-Fender | no such thing |
| 21:00.03 | [TK]D-Fender | Alex25, "channel request hangup [thechannel]" <--------- |
| 21:00.21 | [TK]D-Fender | Devices place calls creating channels. |
| 21:00.31 | [TK]D-Fender | You end channels. Not "extensions". |
| 21:00.31 | Alex25 | there must be a way to so this in bash |
| 21:00.37 | PipBoy | he just told you lol |
| 21:00.57 | PipBoy | asterisk -rx "channel request hangup [the channel]" |
| 21:00.57 | [TK]D-Fender | Alex25, You been told. Another 5 times?out 5 times |
| 21:01.32 | Alex25 | I already know the 'channel request hangup ' command |
| 21:01.37 | PipBoy | haha |
| 21:01.46 | PipBoy | -rx flag likes you pipe in commmands from bash |
| 21:01.47 | Alex25 | but I must do this for a channel |
| 21:01.54 | [TK]D-Fender | Alex25, <PipBoy> asterisk -rx "channel request hangup [the channel]" |
| 21:01.56 | [TK]D-Fender | ^ |
| 21:01.58 | [TK]D-Fender | that is how |
| 21:03.08 | p3nguin | You can't hang up an extension in that manner, but you can hang up the channel for sure. |
| 21:03.21 | Alex25 | moment |
| 21:03.33 | p3nguin | To hang up the extension, you'll need to use a much fancier command. See channel redirect. |
| 21:03.40 | wonderworld | Alex25: you can get the channel, an exension is currently using with "sip show channels" grep the channel from there and hang it up. probably there is a better way, but it should work |
| 21:03.50 | PipBoy | if you wanted to check the channel of a specific persons call you could always asterisk -rx "core show channels" | grep EXT |
| 21:03.54 | Alex25 | i found something here http://lists.digium.com/pipermail/asterisk-users/2009-February/226712.html |
| 21:04.07 | Alex25 | > Exten => _86XXXX,1,system('/usr/sbin/asterisk -rx "soft hangup |
| 21:04.08 | Alex25 | > $(/usr/sbin/asterisk -rx 'core show channels' | grep SIP/${EXTEN(2)| awk '{ |
| 21:04.08 | Alex25 | > print $1 '} )") |
| 21:04.16 | [TK]D-Fender | Alex25, We have just handed you the EXACT command repeatedly. What is there to find? |
| 21:04.23 | PipBoy | soft hangup is just the old cannel request hangup |
| 21:05.23 | Alex25 | how can I use the above workaround in new asterisk system? |
| 21:05.34 | Alex25 | i need only the asterisk -rx part |
| 21:05.39 | p3nguin | s/soft/channel request/ |
| 21:05.40 | [TK]D-Fender | Alex25, CHANGE THE COMMAND IT CALLS |
| 21:06.10 | [TK]D-Fender | <[TK]D-Fender> Alex25, <PipBoy> asterisk -rx "channel request hangup [the channel]" |
| 21:06.30 | PipBoy | I'm not the greatest at this stuff... But If channel request hangup replace softhangup.... wouldnt you just Find/replace and see what happens? |
| 21:07.13 | PipBoy | Aggh.. Im having too much fun with this.. I gotta get back to work |
| 21:07.18 | *** join/#asterisk mogra (477b818a@gateway/web/freenode/ip.71.123.129.138) |
| 21:07.38 | Alex25 | I don't want to hangup a channel, since it's default on my dialplan |
| 21:07.52 | p3nguin | You don't make a lick of sense. |
| 21:07.55 | Alex25 | it doesn't have a real device |
| 21:08.05 | Alex25 | I call it using a phone |
| 21:08.10 | p3nguin | Channels are channels. What more do you want? |
| 21:08.45 | Alex25 | i'll try to explain |
| 21:08.53 | [TK]D-Fender | <Alex25> I don't want to hangup a channel, since it's default on my dialplan <-- the only thing that CAN be hung up is a channel. |
| 21:08.54 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 21:09.00 | [TK]D-Fender | Alex25, There is no such thing as "default" |
| 21:09.08 | [TK]D-Fender | Alex25, None of your words are appropriate. |
| 21:09.12 | p3nguin | You can also hang up your hat, of course. |
| 21:09.58 | jmetro | coats , its december. |
| 21:10.03 | [TK]D-Fender | Alex25, CHANNEL = CALL. There is nothing else to "hangup" |
| 21:10.33 | Alex25 | I want to create a new extension, and in its first priority i want to comman asterisk to hangup all OTHER calls - but not the current one of course |
| 21:10.49 | Alex25 | and then move on with my plan.,, |
| 21:11.12 | p3nguin | Yep. To hang up "calls," you will hang up channels. |
| 21:11.14 | Alex25 | so if i hangup the channel i hangup current extension as well |
| 21:11.26 | [TK]D-Fender | Alex25, So don't hang up the CURRENT CHANNEL |
| 21:11.27 | PipBoy | the extension is in a call |
| 21:11.29 | PipBoy | lol |
| 21:11.32 | p3nguin | You don't hang up extensions, anyway. |
| 21:11.38 | [TK]D-Fender | Alex25, Nobody was telling WHICH ones to hang up |
| 21:11.40 | p3nguin | You hang up channels. |
| 21:11.52 | p3nguin | Pick one. Hang up it. |
| 21:12.04 | p3nguin | s/Hang up/Hangup/ |
| 21:12.13 | jmetro | he wants to hang up his conf files but not his dialplan |
| 21:12.14 | PipBoy | you know what.. just start a call.. hang up the channel.. and see what happens |
| 21:12.44 | p3nguin | And since [tk]d-fender made it abundantly clear how to hangup a channel, you already know how to do that. |
| 21:12.49 | [TK]D-Fender | jmetro, On a scale of 1 to 10 what's your favourite colour of the alphabet? |
| 21:13.00 | jmetro | sour. |
| 21:13.04 | [TK]D-Fender | YES |
| 21:13.08 | p3nguin | Round. |
| 21:13.13 | [TK]D-Fender | SOMETIMES |
| 21:13.40 | p3nguin | "My name is Arthur, and I can count to potato." |
| 21:14.49 | Alex25 | so if i want the first priority of extension number 400 to start by hanging up extension 200 - how to do that when both extensions are in [default] , and not bing to any SIP device? |
| 21:15.02 | p3nguin | (1511.32) <p3nguin> You don't hang up extensions, anyway. |
| 21:15.10 | [TK]D-Fender | ^^^ |
| 21:15.11 | p3nguin | You don't hang up extensions. |
| 21:15.16 | p3nguin | You hang up CHANNELS. |
| 21:15.35 | p3nguin | Usage: channel request hangup <channel>|<all> |
| 21:15.38 | p3nguin | Note: channel |
| 21:15.48 | PipBoy | If we are taking... We are in a conversation.. You dont end a person, you end a conversation |
| 21:15.50 | Alex25 | ok you call it channels. whatever |
| 21:15.52 | p3nguin | Request that a channel be hung up. |
| 21:15.57 | PipBoy | unless your a murdered... and then your ending a person :P |
| 21:15.59 | Alex25 | how to achive the desired effect |
| 21:16.07 | Alex25 | ? |
| 21:16.14 | p3nguin | <PROTECTED> |
| 21:16.24 | [TK]D-Fender | Alex25, LOOK at the list of all channel. One by one hang up on all of them |
| 21:16.43 | p3nguin | or pick one randomly, if you prefer. |
| 21:16.46 | [TK]D-Fender | EXCEPT whichever one(s) you feel like excluding. |
| 21:17.03 | Alex25 | but both extension belond to the same default channel |
| 21:17.08 | [TK]D-Fender | NO |
| 21:17.11 | p3nguin | No |
| 21:17.15 | parasitodelsur | NO |
| 21:17.17 | p3nguin | That statement doesn't even make sense. |
| 21:17.23 | Alex25 | why? |
| 21:17.31 | p3nguin | There is no default channel, for one. |
| 21:17.43 | *** join/#asterisk asr33 (~asr33@unaffiliated/asr33) |
| 21:17.45 | p3nguin | And B, extensions don't belong to channels. |
| 21:17.49 | [TK]D-Fender | Alex25, each channel has its own name. It is different. |
| 21:18.14 | p3nguin | A channel can execute extensions, but for this exercise, that is irrelevant. |
| 21:18.26 | Alex25 | so maybe i got it wrong |
| 21:18.31 | p3nguin | Channels are active. Destroy one or more and be happy. |
| 21:18.32 | Alex25 | moment |
| 21:18.38 | PipBoy | I am telling you... This analogy is perfect... If two people are talking.. They are in a conversation.. You dont end people, you end a conversation.. Just like you dont hangup an extension, you hangup a channel |
| 21:18.48 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 21:18.57 | wonderworld | Alex25: start a call and try "core show channels". this might clarify things |
| 21:19.05 | Alex25 | good explanation :) |
| 21:19.08 | [TK]D-Fender | Alex25, Show us you even have a clue by SHOWING us a channel. You do not seem to be demonstrating even a basic understanding of what a channel is. |
| 21:19.42 | Alex25 | thanks for the compliment.. |
| 21:20.02 | wonderworld | mafia does end conversations by ending people. |
| 21:20.03 | [TK]D-Fender | PipBoy, You are confusing a call with a "channel". This is also inappropriate. |
| 21:20.17 | PipBoy | I think it gets the point accrossed :P |
| 21:20.24 | [TK]D-Fender | PipBoy, a call Is simply a term for 2 channels that happen to be BRIDGED |
| 21:20.56 | p3nguin | purely coincidental |
| 21:20.57 | [TK]D-Fender | PipBoy, Using inappropriate terms in correcting someone else using inappropriate terms gets the WRONG IDEA across |
| 21:21.46 | PipBoy | ok ok .. Was just trying to help... Continue hitting your head against the wall :P |
| 21:22.09 | [TK]D-Fender | PipBoy, We are (unfortunately) all too used to it. |
| 21:22.32 | PipBoy | Hmmm shame :S |
| 21:23.00 | p3nguin | Did you know that if you hangup one channel which happens to be bridged with another channel in a call, the call will end and the other channel will also die? |
| 21:23.53 | p3nguin | Now if that overflows through the phone and the person on the phone also ends... there is little I can do about that. |
| 21:23.56 | p3nguin | Sounds like a bug. |
| 21:23.59 | [TK]D-Fender | p3nguin, 7-10 split! |
| 21:24.04 | Alex25 | ok i just tried |
| 21:24.08 | PipBoy | does the channel go to channel heaven? :( |
| 21:24.18 | Alex25 | i'm getting SIP/1001-000000b5 |
| 21:24.20 | [TK]D-Fender | goes to get another bowling-ball "blessed" |
| 21:24.30 | p3nguin | If by heaven, you mean /dev/null, maybe. |
| 21:24.32 | [TK]D-Fender | Alex25, Good. That is ONE channel |
| 21:24.40 | Alex25 | so i guess the 000000b5 is the identifier |
| 21:24.52 | [TK]D-Fender | Alex25, No, the WHOLE THING is the identifier |
| 21:24.52 | p3nguin | Qiuck! Kill it! |
| 21:25.01 | p3nguin | The unique ID is part of the channel's name. |
| 21:25.09 | Alex25 | is this a random identifier? |
| 21:25.15 | p3nguin | Not really random, no. |
| 21:25.20 | [TK]D-Fender | Alex25, practically speaking, yes |
| 21:26.00 | Alex25 | so every call to the that extension receive the same identifier? or not? |
| 21:26.15 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 21:26.20 | p3nguin | Random would indicate that one time it might be 000000b5 and the next it might be 4a643e100. |
| 21:26.39 | p3nguin | So no, not really random. |
| 21:27.00 | p3nguin | That channel name has NOTHING to do with extensions whatsoever. |
| 21:27.04 | p3nguin | Not at all. |
| 21:27.34 | [TK]D-Fender | <Alex25> so every call to the that extension receive the same identifier? or not? <- NO, it is UNIQUE |
| 21:27.41 | [TK]D-Fender | <p3nguin> The unique ID is part of the channel's name. |
| 21:28.08 | [TK]D-Fender | <Alex25> is this a random identifier? <[TK]D-Fender> Alex25, practically speaking, yes |
| 21:28.27 | [TK]D-Fender | Nowhere does that sound remotely like "<Alex25> so every call to the that extension receive the same identifier? or not?" |
| 21:28.29 | p3nguin | That channel name means that a SIP phone by the ridiculous name of 1001 tried to start a "call." |
| 21:28.42 | wonderworld | Alex25: just grep SIP/yourextension out of the list, cut out the channel, hang it up. |
| 21:29.09 | p3nguin | SIP/yourextension <------- this doesn't make sense. |
| 21:29.21 | wonderworld | *sigh |
| 21:29.27 | [TK]D-Fender | And that wouold exclude ALL channels by a DEVICE (more miused words) |
| 21:29.29 | p3nguin | <p3nguin> That channel name means that a SIP phone by the ridiculous name of 1001 tried to start a "call." <------------- |
| 21:29.42 | [TK]D-Fender | "All but this call" = something ELSE |
| 21:29.43 | Alex25 | ok I've just tested |
| 21:29.51 | Alex25 | it's really random |
| 21:29.53 | p3nguin | or perhaps received one. |
| 21:29.56 | Alex25 | b5 b6 b7 b8 |
| 21:30.06 | [TK]D-Fender | that looks SEQUENTIAL to me... |
| 21:30.10 | p3nguin | That't not random by any definition. |
| 21:30.15 | p3nguin | That's not random by any definition. |
| 21:30.19 | Alex25 | i know |
| 21:30.20 | [TK]D-Fender | Someon just failed at math. HARD |
| 21:30.54 | Alex25 | but for my issue - how can I hangup it programatically if it's random? |
| 21:31.00 | p3nguin | It isn't random. |
| 21:31.42 | p3nguin | Do you know on which device is the call that you want to end? |
| 21:31.44 | Alex25 | i mean each call it gets another value |
| 21:31.52 | [TK]D-Fender | <[TK]D-Fender> Alex25, So don't hang up the CURRENT CHANNEL |
| 21:31.54 | Alex25 | yes |
| 21:32.09 | p3nguin | For example, a SIP phone named 000011112222. |
| 21:32.17 | [TK]D-Fender | <[TK]D-Fender> Alex25, LOOK at the list of all channel. One by one hang up on all of them |
| 21:32.21 | [TK]D-Fender | ^^^^ |
| 21:32.32 | Alex25 | ok |
| 21:32.39 | [TK]D-Fender | core show channels^^ LIST |
| 21:32.40 | p3nguin | channel request hangup SIP/000011112222<PRESS TAB KEY> |
| 21:32.50 | [TK]D-Fender | ok, I'm off |
| 21:35.47 | *** join/#asterisk mogra (477b818a@gateway/web/freenode/ip.71.123.129.138) |
| 21:38.35 | *** part/#asterisk cervajs2 (~cervenka@gatekeeper.bm.ipex.cz) |
| 21:39.05 | PipBoy | lol this was fun |
| 21:49.34 | jeffspeff | is away: Please leave a message after the tone... |
| 21:49.48 | p3nguin | I hope that wasn't an automatic away message. |
| 21:51.49 | PipBoy | bhaha heres a good story |
| 21:52.03 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:52.09 | *** join/#asterisk svm_invictvs (c7e77ca0@gateway/web/freenode/ip.199.231.124.160) |
| 21:52.11 | svm_invictvs | Heya |
| 21:52.19 | svm_invictvs | How do I delete all messages in an inbox? |
| 21:52.52 | [TK]D-Fender | svm_invictvs: 7. 6. Was. Rinse. Repeat. |
| 21:52.56 | [TK]D-Fender | Wash* |
| 21:53.09 | *** part/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:53.14 | PipBoy | Lady is working at a shrinks office. Dials her queue code to get into queue, it asks her for her Ext number to log her into queue. She forgets she dialed the queue code and then enters it again. Crazy person calls in to see their shrink, Guess who is the first person in queue to awnser? |
| 21:53.14 | svm_invictvs | can I just go on the box and delete the files? |
| 21:53.18 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:53.26 | pigpen | [TK]D-Fender, you sure are talking to me today. |
| 21:53.35 | svm_invictvs | [TK]D-Fender: Can i Just delete the files on the server? |
| 21:53.56 | PipBoy | the queue code awnsers :D and asks the customer what their extension is... Customer hangs up and the queue program logs his phone number into the queue |
| 21:54.52 | [TK]D-Fender | pigpen: Yeah, my bad aim and new IRC client are a boatload of FAIL today... |
| 21:54.58 | [TK]D-Fender | svm_invictvs: Sure |
| 21:55.09 | PipBoy | So then someone else calls in for their shrink.. Dials the other crazy persons number (because he is now a queue member) and now two crazy people are talking out their issues with each other.. the best part is... the person who was signed into the queue, thinks their shrink just called him back but is now asking for mental help :D |
| 21:55.43 | navaismo | Any one can help with this--->utils.c:565 lock_info_destroy: Thread 'pbx_thread started at [ 5631] pbx.c ast_pbx_start()' still has a lock! - 'q' (0x2aaaac6dcaa0) from 'update_realtime_members' in app_queue.c:2492! |
| 21:55.55 | pigpen | [TK]D-Fender, I saw the conversation, gotta love the tab button. Hope all is well, just wanted to give you a hard time |
| 21:59.01 | svm_invictvs | [TK]D-Fender: Ideally, I'd like to delete the message after it's been emailed |
| 21:59.42 | asr33 | svm_invictvs: they are in /var/spool/asterisk/voicemail/default/"your voice account"/INBOX |
| 22:00.03 | svm_invictvs | I could just set a cron job to periodically clean out old messages |
| 22:00.14 | asr33 | sure |
| 22:02.59 | [TK]D-Fender | svm_invictvs: delete=yes <-------- |
| 22:04.35 | svm_invictvs | But, my big question is, does it need to be done in such a way that Asterisk needs to "know" about the delete? |
| 22:04.54 | svm_invictvs | Like, do I have to invoke a command to have them removed, or is it just looking for the presence of the recordings? |
| 22:05.23 | [TK]D-Fender | svm_invictvs: You want it deleted as e-mailed. there is a box option for this already |
| 22:05.44 | svm_invictvs | Ah okay |
| 22:05.47 | svm_invictvs | I'll look into it |
| 22:05.59 | svm_invictvs | I'ma ctaully movin my asterisk server to a new machine, probalby will roll that change in there |
| 22:06.30 | [TK]D-Fender | It's totally worth the 5 seconds it should take. |
| 22:16.18 | jacekowski | jpsharp: 3 users |
| 22:16.20 | jacekowski | jpsharp: out of 30 |
| 22:19.42 | *** join/#asterisk corretico (~luis@190.211.93.38) |
| 22:21.37 | *** join/#asterisk lvlinux (~n1gg@c-50-147-64-9.hsd1.tn.comcast.net) |
| 22:22.13 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 22:23.40 | *** join/#asterisk ulogic (421e6b4f@gateway/web/freenode/ip.66.30.107.79) |
| 22:34.10 | *** join/#asterisk cyborg-one (~cyborg-on@130-0-32-145.broadband.tenet.odessa.ua) |
| 22:41.47 | *** join/#asterisk Alex25 (~kvirc@109.64.206.159) |
| 22:44.13 | Alex25 | I'm trying to a channel from CLI, but asterisk outputs "SIP/xxxzz is not a known channel" - how to hangup such channels? |
| 22:44.54 | Alex25 | or the extension they are using? |
| 22:49.27 | jpsharp | jacekowski: Check those phones. Make sure they've got good handsets. And make sure your users don't have cranial rectalitis. |
| 22:49.42 | jpsharp | Alex25: You need the full channel identifier. |
| 22:49.52 | jpsharp | it should be something like SIP/xxxzz-yyyy |
| 22:51.21 | Alex25 | I have it - but cannot hangup |
| 22:51.38 | Alex25 | "SIP/xxxzz is not a known channel" |
| 22:51.58 | Alex25 | that's an example |
| 22:53.07 | jpsharp | You're typing it exactly as it shows up in 'core show channels'? |
| 22:53.17 | Alex25 | yes sure |
| 22:53.39 | jacekowski | jpsharp: i did check the phones, and i can't hear anything |
| 22:53.48 | jacekowski | jpsharp: and those are brand new phones |
| 22:55.03 | Alex25 | asterisk -rx "channel request hangup SIP/sipsorcery.com-0" |
| 22:55.09 | Alex25 | outputs |
| 22:55.13 | Alex25 | SIP/sipsorcery.com-0 is not a known channel |
| 22:55.36 | jacekowski | try doing asterisk -r and using tab autocompletion |
| 22:55.55 | jacekowski | jpsharp: i'm just wondering if it isn't a sidetone issue |
| 22:56.04 | jpsharp | jacekowski: I'd blame the users then. Probably cranking up the volume all the way or something. |
| 22:56.22 | jpsharp | Or yelling into the phone. |
| 23:02.38 | Alex25 | ok i found the problem |
| 23:03.01 | *** join/#asterisk elico (~Thunderbi@109.64.229.90) |
| 23:03.05 | jpsharp | What was it? |
| 23:03.09 | Alex25 | when I do |
| 23:03.15 | Alex25 | CHANNEL=`asterisk -rx "core show channels" | grep SIP/ | grep 200 | cut -f1 -d" "`; asterisk -rx "channel request hangup $CHANNEL" |
| 23:03.27 | Alex25 | I get only partial SIP id |
| 23:03.51 | Alex25 | how to get it in full? |
| 23:04.46 | [TK]D-Fender | Alex25You should probably be looking at each step in that yourself |
| 23:04.47 | Alex25 | I'm getting SIP/sipsorcery.com-0 when it's actually SIP/sipsorcery.com-000000e1 |
| 23:04.56 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-uiezpnjzaraxwjpr) |
| 23:05.07 | Alex25 | what do you think? |
| 23:05.46 | [TK]D-Fender | Alex25: I think you should be LOOKING at each step of this... |
| 23:06.02 | [TK]D-Fender | Alex25: core show channels <-- starting with |
| 23:07.41 | Alex25 | so you mean "core show channels" only provide limited output lenght? |
| 23:08.51 | [TK]D-Fender | Alex25: I mean GET OFF YOUR ASS AND LOOK |
| 23:09.04 | Alex25 | hey |
| 23:09.15 | Alex25 | that was not my code |
| 23:09.24 | Alex25 | and i'm not a bash expert |
| 23:10.16 | [TK]D-Fender | That isn't even BASH |
| 23:10.22 | Alex25 | so if u see something i shd know just say it |
| 23:10.24 | [TK]D-Fender | that is an ASTERISK CLI command |
| 23:10.35 | *** join/#asterisk fritz09 (~Adium@pop1-765.catv.wtnet.de) |
| 23:10.36 | [TK]D-Fender | You aren't looking |
| 23:11.20 | Alex25 | asterisk -rx "core show channels" |
| 23:11.21 | *** join/#asterisk lvlinux (~n1gg@c-50-147-64-9.hsd1.tn.comcast.net) |
| 23:11.27 | Alex25 | what's wrong? |
| 23:11.58 | [TK]D-Fender | Alex25: What do YOU see in it? |
| 23:12.42 | Alex25 | I see a command to * to output active channels |
| 23:12.54 | Alex25 | i dont knwo why string lenght is limited |
| 23:13.13 | [TK]D-Fender | IS it limited? |
| 23:13.35 | [TK]D-Fender | I'm not seeing you SHOWING me a channel dump and maybe some kind of qualified comparison to base a conclusion on... |
| 23:13.58 | Alex25 | I'm getting SIP/sipsorcery.com-0 when it's actually SIP/sipsorcery.com-000000e1 |
| 23:14.12 | [TK]D-Fender | Then I guess that command isn't good enough |
| 23:14.21 | Alex25 | when i try on bash cli its: SIP/sipsorcery.com-0 |
| 23:14.41 | Alex25 | when on asterisk cli itself its: SIP/sipsorcery.com-000000e1 |
| 23:15.30 | Alex25 | I need to get it in full with bash, that's all |
| 23:16.02 | Alex25 | if u see something wrong with my comman which causes that. pls tell me |
| 23:16.27 | [TK]D-Fender | Of course something is wrong with the command. |
| 23:16.33 | [TK]D-Fender | It doesn't output the WHOLE CHANNEL NAME |
| 23:16.40 | [TK]D-Fender | How is this even a question? |
| 23:16.46 | [TK]D-Fender | It's not there |
| 23:16.54 | [TK]D-Fender | this isn't a mystery |
| 23:17.06 | [TK]D-Fender | It has a limited length |
| 23:17.10 | [TK]D-Fender | It is not reliable. |
| 23:17.19 | [TK]D-Fender | You should be looking at something else then. |
| 23:17.25 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 23:17.41 | [TK]D-Fender | go enter the command at * cli and hit <tab> and LOOK |
| 23:18.31 | Alex25 | i did it, and it worked fine on * cli |
| 23:18.43 | Alex25 | the only problem was with bash |
| 23:18.44 | svm_invictvs | Hm |
| 23:18.56 | svm_invictvs | Next dumb question, is it possible to resolve extensions using LDAP? |
| 23:19.25 | svm_invictvs | So, if the extension starts with say 1, then tell Asterisk to use an LDAP query to find where to forward? |
| 23:20.02 | Alex25 | so how to get output the WHOLE CHANNEL NAME in bash CLI? |
| 23:21.21 | jpsharp | svm_invictvs: Use the LDAP realtime driver. |
| 23:23.01 | [TK]D-Fender | Alex25: It isn't going to be different one way from the other. And I am not SEEING output from both. |
| 23:23.30 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 23:26.13 | p3nguin | core show channels concise |
| 23:26.39 | p3nguin | That will show the whole channel name, and it even has nice separated values for easier parsing. |
| 23:28.17 | Alex25 | thanks it's working |
| 23:28.32 | Alex25 | but it provide extra data in a long string |
| 23:28.54 | Alex25 | I need only the sip id for hanging it up |
| 23:29.16 | *** join/#asterisk lvlinux (~n1gg@c-50-147-64-9.hsd1.tn.comcast.net) |
| 23:29.46 | [TK]D-Fender | Not going to look or show anything |
| 23:29.52 | [TK]D-Fender | I've wasted enough time on this. |
| 23:30.33 | p3nguin | 2.5 hours! |
| 23:31.10 | Alex25 | sure. you go to sleep a little bit |
| 23:31.19 | Alex25 | you sound tired |
| 23:33.16 | p3nguin | Get your awk on, and parse that damned channel name before someone shows up at your house and not only hangs up your channel, but hangs you as well. |
| 23:34.39 | [TK]D-Fender | p3nguin: Best part is if that command ends up killing the channel ISSUING it first... |
| 23:34.50 | Alex25 | sorry i'm not a genius like you and your friend |
| 23:35.04 | Alex25 | I'm a simple person |
| 23:35.16 | Alex25 | who try to setup somethinh |
| 23:35.24 | [TK]D-Fender | Alex25: You aren't looking and you aren't showing. |
| 23:36.02 | [TK]D-Fender | You show NO initiative. You seem to expect someone else to rewrite your broken 3rd party scripts for you and not ddo a single thing to help the process or learn anything |
| 23:36.17 | Alex25 | what do I need to "show"? |
| 23:36.23 | Alex25 | if i came here |
| 23:36.39 | Alex25 | that means i only need a quick solution for something |
| 23:36.44 | *** join/#asterisk navaismo (~navaismo@189.144.120.135) |
| 23:36.53 | p3nguin | What is your definition of quick? |
| 23:36.59 | Alex25 | I didn't ask you to code a complex script for me |
| 23:37.17 | Alex25 | just a small tip |
| 23:38.38 | Alex25 | but you like to play arrogance on newbies |
| 23:39.16 | Alex25 | maybe it's your way to feel special |
| 23:39.34 | Alex25 | but I'm not in this game |
| 23:39.47 | [TK]D-Fender | You are running the game. |
| 23:39.53 | [TK]D-Fender | I asked you to show the output of MULTUIPLE commands. |
| 23:40.03 | [TK]D-Fender | CLI straight VS bash which you claim is different |
| 23:40.04 | Alex25 | I'm just trying to setup something for my parents |
| 23:40.19 | [TK]D-Fender | 'I asked for something comparative to show that what you see isn't in fact good. |
| 23:40.44 | [TK]D-Fender | And all you have is a STORY about the reasons you arent even hitting TAB in CLI where I tell you to to see the OPTIONS |
| 23:40.48 | [TK]D-Fender | You are doing NOTHING for yourself |
| 23:40.53 | [TK]D-Fender | You INSIST on doing nothing |
| 23:40.57 | [TK]D-Fender | Noone owes you anything |
| 23:41.06 | Alex25 | ypu see? |
| 23:41.10 | p3nguin | [tk]d-fender: How dare you try to help someone do exactly what they were looking to do. You're such an asshole! |
| 23:41.24 | Alex25 | here u play the game |
| 23:41.44 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:41.45 | [TK]D-Fender | Alex25I'm just trying to setup something for my parents <- do0esn't amtter if you want it for a school project. Or your job. Or your parents. Or as an exercise in masochism |
| 23:42.06 | Alex25 | I didn't come here to undergo quizes, just looking for an answer, and make it as simple as possible |
| 23:42.40 | Alex25 | so stop your arrogant game please |
| 23:42.59 | Alex25 | didn't you say u were going to sleep? |
| 23:43.04 | Alex25 | anyway |
| 23:43.12 | [TK]D-Fender | Alex25: Answer : Since you aren't even going to look at what you're doing piece by piece, or show anything, and you clearly don't want to learn anything, then your laziness doesn't deserve a hand-out |
| 23:43.16 | Alex25 | u should learn to respect people |
| 23:43.32 | [TK]D-Fender | Alex25: Respect is cooperating with those whose help you want |
| 23:43.43 | [TK]D-Fender | Alex25: You seem to feel you're entitled to us doing everything for you |
| 23:43.52 | Alex25 | you are not a teacher or a mentor here |
| 23:44.00 | Alex25 | you shd know your place |
| 23:44.01 | [TK]D-Fender | I am. You are no STUDENT |
| 23:44.11 | [TK]D-Fender | you don't want to know anything and give nothing but excuses. |
| 23:44.28 | [TK]D-Fender | Don't feel surprised at the result |
| 23:44.38 | Alex25 | if someone comes here and ask a question - he onle looking for an answer |
| 23:44.57 | Alex25 | don't play quizes with people here |
| 23:45.07 | Alex25 | you're not better than anyone |
| 23:45.10 | [TK]D-Fender | Alex25: It isn't a quiz. You're too lazy to do anything at all. |
| 23:45.17 | [TK]D-Fender | You've provien it time and again |
| 23:46.43 | *** join/#asterisk elico (~Thunderbi@109.64.229.90) |
