| 00:21.33 | *** join/#asterisk mpe_ (~mpe@0xd99d3f8f.customer.cybercity.dk) |
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| 01:15.53 | oDesk | i would love to have callerID displayed into my x100p for DTMF sent before first ring on wcfxo |
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| 01:35.23 | oDesk | do you know of any linux Dtmf raw decoder to text ? anyone ? |
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| 01:52.19 | Montys | oDesk; LEt me understand your question, what you want to do, would be to take an audio file containing DTMF and then using an application export it into a text? |
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| 02:15.25 | *** join/#asterisk salviadud (~dick.gonz@189.156.175.229) |
| 02:15.39 | salviadud | guys, I'm not new to asterisk, but I am new to DID services |
| 02:16.04 | salviadud | I want my clients to call me at a mexican number (which I need to buy), so I can play with that call as I please |
| 02:16.52 | salviadud | I recently purchased an account at voipvoip.com, but they suck so bad... I can't get it to work. Asterisk keeps telling me about some invalid extension, I can't get calls to go out |
| 02:17.02 | salviadud | anybody know of a good DID service? |
| 02:17.03 | bougyman | out or in? |
| 02:17.13 | salviadud | well |
| 02:17.14 | bougyman | DID service has nothing to do with outbound calls |
| 02:17.20 | salviadud | both ways, they don't go out, don't go in |
| 02:17.29 | bougyman | they never hit your box at all? |
| 02:17.33 | bougyman | or asterisk doesn't route them? |
| 02:17.36 | salviadud | yes they do |
| 02:17.38 | bougyman | they're your outbound proxy, too? |
| 02:17.47 | salviadud | i got them as user and peer |
| 02:17.59 | salviadud | so, its 2 accounts on my sip.conf |
| 02:18.09 | salviadud | both fail |
| 02:18.20 | bougyman | ouch. |
| 02:18.24 | salviadud | what I really don't get is why the outbound fails |
| 02:18.28 | bougyman | i suck at sip.conf, hopefully someone here can help |
| 02:18.36 | salviadud | they just plain suck, I don't even want to deal with them |
| 02:18.49 | salviadud | so, I'm looking for someone in the know of those services |
| 02:18.50 | bougyman | doesn't sound like their issue if it's getting to asterisk |
| 02:18.52 | salviadud | by another company |
| 02:18.57 | bougyman | what does the log say when you get an outbound call? |
| 02:19.06 | bougyman | i use those services often, just not with asterisk |
| 02:19.08 | salviadud | busy/congested |
| 02:19.25 | salviadud | the real mistery, is with the inbound calls |
| 02:19.41 | salviadud | I get some gibberish about '' trying to get an unknown extension |
| 02:19.43 | bougyman | if they make it to your it's something in your dialplan or sip.conf that's wrong. |
| 02:19.52 | salviadud | no way man |
| 02:19.52 | bougyman | er to your asterisk box |
| 02:19.59 | bougyman | yes way man. |
| 02:20.00 | salviadud | i've used several sip services before |
| 02:20.07 | salviadud | these guys suck |
| 02:20.13 | *** join/#asterisk rolandbm (~brad@pool-108-0-6-208.lsanca.fios.verizon.net) |
| 02:20.32 | mmlj4 | teliax |
| 02:20.38 | salviadud | thank you! |
| 02:21.04 | rolandbm | hi guys. I haven't used IRC in quite a long time so tell me if I'm being a noob :) |
| 02:21.21 | bougyman | you were just there. |
| 02:21.34 | rolandbm | i've got a problem with some DID numbers and I'm hoping someone can assist? |
| 02:21.47 | bougyman | rolandbm: what's the problem? |
| 02:22.13 | ChannelZ | file not found |
| 02:22.36 | rolandbm | I have a sip provider that has given us 10 DID numbers. I want to setup different IVR's for the number called, but I can't tell if the sip provider if sending across the DID dialed |
| 02:22.53 | rolandbm | I have debug logging turned on, but not sure what exactly I'm looking for? |
| 02:22.59 | bougyman | rolandbm: capture the packet and look at it (with wireshark) |
| 02:23.11 | bougyman | that's the quickest road to this finish line |
| 02:23.15 | bougyman | it may be in a weird place. |
| 02:23.30 | bougyman | i've seen them send it in all sorts of places. |
| 02:23.48 | salviadud | rolandbm, who is that provider? |
| 02:23.48 | bougyman | or you could ask them, they may know exactly where it is. |
| 02:24.12 | rolandbm | thanks bougyman, I've tried asking but they are taking their time sending it up to the admins :) |
| 02:24.44 | rolandbm | so I figured I would check on my own what is being sent to it. so if I get the packets, that should tell me what is being sent to the system as far as numbers go? |
| 02:25.06 | rolandbm | salviadud, the provider is iiNet in australia |
| 02:25.18 | ChannelZ | Generally they should send to an extension of your DID |
| 02:26.23 | ChannelZ | exten => 5551112222,1,NoOp(Dude called 555-111-2222) |
| 02:26.37 | bougyman | i've seen some send it in P- headers, even. |
| 02:26.41 | bougyman | lame |
| 02:27.56 | rolandbm | ok cool. Thanks guys. I'm mostly familiar with tcpdump so will get it to dump the info I need. :) |
| 02:29.03 | rolandbm | I figured I could get asterisk to give me that info. didn't think about doing a raw packet dump to see what was going on. |
| 02:29.30 | ChannelZ | sip debug on |
| 02:29.37 | ChannelZ | call in. See what you're getting. |
| 02:29.43 | ChannelZ | No need to dig through raw transport dumps |
| 02:30.00 | ChannelZ | s/sip debug on/sip set debug on/ |
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| 02:32.19 | rolandbm | ChannelZ, I thought I tried that before, but going to give it a shot now |
| 02:33.07 | ChannelZ | So I'm not clear, have you just made an exten in your dialplan with your DID and it isn't working or you haven't tried anything? |
| 02:36.14 | rolandbm | What I've got is a trunk setup that has 10 DID's. I'm using FreePBX and have an inbound route setup for any number to go to our main IVR. What I want to do is use one of our other DID's to go directly to our sales. When I've tried to set up another inbound route for this, it still just goes to the main IVR. |
| 02:36.43 | rolandbm | So my guess is that either the SIP provider isn't sending across the DID that is being dialed, or I don't have it in there exactly as they are sending it |
| 02:37.37 | rolandbm | btw, the setup with the main IVR is working. It's just getting the other DID to go to the other route that doesn't seem to work |
| 02:37.59 | rolandbm | thats why I think it is the information being sent that is the problem |
| 02:38.56 | ChannelZ | or a freepbx problem. No help there. |
| 02:40.36 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 02:43.03 | rolandbm | yeah true. i've turned on the sip debug logging and that seems to be giving me tons of info. so hopefully it will be in there. thanks for the help |
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| 04:04.27 | oDesk | Montys: yes exactly, decoding dtmf into txt |
| 04:05.00 | *** join/#asterisk karthik (~karthik@122.183.17.90) |
| 04:09.03 | Montys | oDesk: multimon |
| 04:09.17 | Montys | here is the command that you need to use: multimon -t wav -a DTMF tx1.wav |
| 04:09.53 | Montys | You should be able to download the src @ http://www.baycom.org/~tom/ham/linux/multimon.html |
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| 04:34.39 | oDesk | Montys: the file extension is raw |
| 04:35.19 | oDesk | Montys: that i've built using dahdi_monitor |
| 04:37.29 | Montys | @Desk, I think dahdi_monitor can record wave file too |
| 04:38.22 | Montys | In the case that you already have the file in raw, you can use "sox" con convert it to wave ; sox -r 8000 -s -w -c 1 tx1.raw tx1.wav |
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| 04:39.40 | Montys | @oDesk ^^ |
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| 04:57.38 | ranjan | karthik, hi karthik |
| 04:58.07 | karthik | hi ranjan |
| 04:58.22 | ranjan | my virtual machine is ready. going to install asterisk |
| 04:58.31 | ranjan | i hope the ./configure will do |
| 04:58.48 | ranjan | is there any other parameters that has to be passed? |
| 05:06.14 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 05:14.13 | p3nguin_ | ranjan: Apparently the default prefix is wrong, so you might want to specify --prefix=/usr/local at least. |
| 05:17.26 | ChannelZ | hmm. haven't seen that one |
| 05:18.07 | ChannelZ | oops |
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| 05:31.28 | ranjan | karthik: hi karthik |
| 05:31.37 | ranjan | karthik: installed asterisk |
| 05:32.07 | ranjan | karthik: i think you are busy |
| 05:32.10 | ranjan | Hi all |
| 05:34.29 | ranjan | can any one give me the correct configuration for defining a sip peer in /etc/sip.conf ?? |
| 05:39.00 | ChannelZ | Not really |
| 05:39.16 | ChannelZ | Can you tell me how to fix my car? |
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| 05:58.25 | *** join/#asterisk seanjohn (~admin@gateways.sheltoncomputers.com) |
| 05:59.14 | seanjohn | if I put call-limit=4 for sip_general.conf will it affect my trunks, even with them having their own call-limit set? |
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| 06:08.37 | [TK]D-Fender | seanjohn: No |
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| 07:16.55 | ranjan | Hi all, just started with asterisk |
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| 07:30.05 | schmidts | good morning |
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| 07:53.16 | BenC[UK] | morning people |
| 07:53.40 | BenC[UK] | anyone used 7970 with asterisk before, really strugging to get mine to register |
| 07:53.48 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:20.12 | petern_ | I use one, but with chan_sccp. |
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| 09:20.24 | BenC[UK] | I got it working |
| 09:20.29 | BenC[UK] | But its using Nat |
| 09:20.35 | BenC[UK] | Imean, using NAT is the problem |
| 09:20.38 | BenC[UK] | its reigstered locally now |
| 09:21.05 | *** join/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 09:22.33 | nextime | hello all. Is there any browser based java applet sip softphone installable on own web server available somewhere for free ( to be used by a poor no-profit organization )? |
| 09:24.47 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
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| 09:43.36 | *** join/#asterisk jermey_g (~jeremey_g@gw-sthlm01.rebtel.com) |
| 09:45.06 | jermey_g | What does this mean, channel.c: No channel type registered for 'zap' |
| 09:45.06 | jermey_g | ? |
| 09:45.35 | *** join/#asterisk mpe_ (~mpe@178.72.27.239) |
| 09:45.55 | jermey_g | I got no tdm hardware in the box. I just use ztdummy. |
| 09:47.41 | kaldemar | jermey_g: it means that you don't have the zapata channel driver loaded. are you trying to call over zaptel or when do you get that? |
| 09:48.10 | jermey_g | I don't call over zaptel. I just use sip |
| 09:48.36 | *** join/#asterisk GhOnDiE (~GhOnDiE@92.29.179.207) |
| 09:48.38 | kaldemar | when does that message occur? |
| 09:49.17 | GhOnDiE | anybody good with the manager interface |
| 09:49.18 | GhOnDiE | ? |
| 09:51.48 | jermey_g | kaldemar:in the logs, i see it every 2nd minute WARNING: --message--. |
| 09:52.25 | kaldemar | jermey_g: what happens in the CLI when you get it? what is asterisk doing when it happens? |
| 09:55.24 | *** join/#asterisk darkskiez (~mhb@2002:5208:d7a8::1) |
| 09:58.34 | jermey_g | kaldemar:i have enabled verbose log to the file, so i ll see in few mins. |
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| 10:33.43 | *** join/#asterisk dandate2 (~gtejkgjke@58.69.25.48) |
| 10:38.48 | dandate2 | So i'm going to have a 1MB T1 line connected to a cisco ATA router with 6 iphones. the total bandwidth requirement is 1003.28kbps or .98 MB . will this be enough to support 6 ip phones using ulaw or should 1 phone use g722? |
| 10:39.47 | WIMPy | G.722 uses the same bandwith as G.711. |
| 10:40.32 | WIMPy | And 1 mbit schould be about twice as much as you need for 6 calls. |
| 10:40.57 | dandate2 | really? it says in the bandwidth calculator that total use is .98MB |
| 10:41.03 | dandate2 | http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 10:42.02 | dandate2 | and lists various rates for g722, one of them also being 64 kipbs, but also shows 56 and 48 |
| 10:42.30 | WIMPy | Hmm. Yes. Haven't seen them in use however. |
| 10:42.59 | WIMPy | The .98 must be the combined rate in both directions. |
| 10:43.14 | dandate2 | i'm told that the bandwidth of the T1 is pooled |
| 10:43.29 | dandate2 | so if i'm downing at .75MB i could only upload at .25MB |
| 10:43.39 | dandate2 | but this is just what the sales rep said =; |
| 10:44.49 | WIMPy | The line would do 1.5mbit full duplex, but I don't know what you pay for. |
| 10:45.39 | *** join/#asterisk metiu_ (~chatzilla@85-18-228-185.ip.fastwebnet.it) |
| 10:45.42 | dandate2 | they said 1.5mb would cost more |
| 10:46.11 | WIMPy | Check your contract. |
| 10:46.39 | *** join/#asterisk freckle (~viperdude@viperdudeuk.broker.freenet6.net) |
| 10:47.09 | metiu_ | hi, is it possible to have two UAs with the same extension on two servers call each other? it seems that the server sees the incoming call as if it's coming from its attached UA, instead of the UA attached to the remote server |
| 10:47.23 | metiu_ | e.g. different area codes, same telephone numbers |
| 10:47.37 | metiu_ | one asterisk for each area code (powerful machine indeed) |
| 10:48.07 | metiu_ | number 456-95616 calls 987-95616 |
| 10:48.10 | dandate2 | looking at the contract, really isnt very descriptive =/ |
| 10:48.33 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 10:48.35 | WIMPy | metiu_: It all depends on your dialplan. |
| 10:48.52 | dandate2 | Service Name: <I-Direct 1mb> Port Speed |
| 10:48.52 | dandate2 | (Kbps) : 1mb |
| 10:49.14 | dandate2 | or for $200 more per month i can get 1.5mb |
| 10:50.38 | metiu_ | WIMPy: I guess I'd have something like exten => _456XXXXX,n,Dial(SIP/server456/${EXTEN:1}-phone) |
| 10:50.46 | kaldemar | dandate2: find some technical person from the provider to clarify that. |
| 10:51.07 | metiu_ | and exten => _987,n,Dial(SIP/server987/${EXTEN:1}-phone) |
| 10:51.39 | metiu_ | what I see on the other end, however, is a call from 95616 which is seen as local |
| 10:52.14 | metiu_ | should I change the callerid to add the prefix? |
| 10:52.40 | WIMPy | metiu_: Either use the complete callerid or get the calls from the other server to a seperate context that adds th area code. |
| 10:54.36 | *** join/#asterisk cusco (~trilili@174.196.108.93.rev.vodafone.pt) |
| 10:54.36 | cusco | hi |
| 10:54.53 | cusco | why is this call from one * box to another * box forbidden? http://paste.debian.net/92704/ |
| 10:56.18 | cusco | peer 150 is registered in ast-box1, peer 202 in registered in ast-box2 |
| 10:56.20 | WIMPy | SIP/2.0 401 Unauthorized |
| 10:56.27 | cusco | yes, but why not? |
| 10:56.46 | cusco | 192.168.2.5:5060 N covilha 105 Registered Fri, 01 Oct 2010 11:55:56 |
| 10:56.53 | WIMPy | Wrong user and/or password. |
| 10:57.01 | WIMPy | Registration is a different matter. |
| 10:57.24 | cusco | err... |
| 10:57.53 | kaldemar | cusco: because you have a secret defined for the peer. it is supposed to send a new invite with a challenge response. |
| 10:57.55 | cusco | so do I need to specify dial SIP/202@lisboa with password? |
| 10:58.13 | cusco | well shouldn't it follow the registration settings? |
| 10:58.16 | cusco | :/ |
| 10:58.22 | kaldemar | cusco: no, it shouldn't. |
| 10:58.25 | cusco | err |
| 10:58.25 | WIMPy | no |
| 10:58.42 | cusco | so I must specify the apssword on the dial?! |
| 10:58.54 | kaldemar | cusco: or in a peer in sip.conf |
| 10:59.09 | dandate2 | i think i'm reading the bandwidth calculator incorrectly. when I state for incomming and outgoing bandwidth this is concerning the PBX corect? |
| 10:59.16 | cusco | kaldemar: box2 has a peer in sip.conf with a apssword |
| 10:59.27 | cusco | if box1 wants to dial.. also eneds the peer in sip.conf ? |
| 10:59.32 | cusco | needs |
| 10:59.44 | dandate2 | i'm calculating bandwidth for a remote location from the pbx where there will just be ip phones. will I need to factor incomming and outgoing bandwidth? |
| 10:59.49 | kaldemar | cusco: it needs credentials, whether they are in the dialplan or in sip.conf. |
| 11:00.01 | WIMPy | cusco: You need to put the password somewhere. |
| 11:00.21 | cusco | it is in a "register" line in sip.conf ... heh |
| 11:00.49 | kaldemar | cusco: the register line has nothing to do with calls that originate from the box. |
| 11:00.52 | cusco | ok so the dial can be SIP/202:secret@lisboa? |
| 11:01.38 | kaldemar | the sample sip.conf has a nice list of dial strings, that being one of them. |
| 11:02.13 | dandate2 | it just stumped me that the calculator looks at incomming and outgoing channels. this refers to the PBX being the centerpoint right |
| 11:03.13 | WIMPy | That would make sense. |
| 11:03.43 | dandate2 | ahahah need to change my contact to 512kb then |
| 11:04.02 | dandate2 | 512kb should support 6 ip phones at a remote location? |
| 11:04.07 | *** join/#asterisk kerframil (~kerframil@gentoo/user/kerframil) |
| 11:04.31 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 11:04.46 | cusco | kaldemar: what if the other box is registered here... |
| 11:04.52 | cusco | and the peer is set here |
| 11:04.56 | cusco | ok I got it reverse.. |
| 11:05.30 | WIMPy | Registratio is only to tell the other end, where to find you. |
| 11:05.53 | cusco | ok |
| 11:06.04 | cusco | well, but in sip.conf there is a peer with the secret on it |
| 11:06.07 | cusco | what is wrong then? |
| 11:06.40 | WIMPy | If you connect two servers, you probably don't need to register. |
| 11:06.44 | kaldemar | you're not using any information in sip.conf when you dial the way you do. |
| 11:06.54 | WIMPy | At least not if the have a static IP. |
| 11:07.29 | cusco | ok.. so I can use the same peer on both? |
| 11:07.59 | WIMPy | yes |
| 11:09.21 | dandate2 | will I need to include RTCP when deciphering bandwidth usage of ip phones at a remote location? (determining the bandwidth rquirement of the remote location) |
| 11:10.06 | kaldemar | dandate2: if you use RTP, there will be RTCP involved, just like last time you asked. |
| 11:11.12 | kaldemar | the bandwidth usage is minor though. |
| 11:11.31 | *** join/#asterisk Benwa (~Schnitzel@unaffiliated/benwa) |
| 11:13.54 | dandate2 | k |
| 11:13.59 | cusco | it is still replying with: Oct 1 12:13:26] NOTICE[27494]: chan_sip.c:17917 handle_response_invite: Failed to authenticate on INVITE to '"Tiago Geada" <sip:150@10.100.100.5>;tag=as22e3e4e4' |
| 11:14.26 | cusco | let me show you sip.conf bit on both boxes... hold |
| 11:15.03 | dandate2 | im just triyng to figure out what connection speed the remote locatino would require, ive been confusing myself this whole time thinking 1 call would need 159kpbs |
| 11:15.43 | dandate2 | but thats hwo much it requires at the pbx site right |
| 11:16.27 | cusco | is something wront here...? http://paste.debian.net/92707/ |
| 11:16.50 | metiu_ | WIMPy: where can I find a way of putting the incoming calls on different contexts based on the incoming server? right now I have a default context for incoming call and one for the local UAs, but as I said the UA "100" gets routed in the local UAs in any server where there is a UA "100" |
| 11:17.17 | metiu_ | I thought it would have been put in the default "incoming" context, since it's routed through another asterisk server |
| 11:17.33 | WIMPy | Put a context= into the peer. |
| 11:18.13 | metiu_ | you mean registering each server to each other as a peer, right? |
| 11:18.53 | metiu_ | so that I address e.g. SIP/server1/100 SIP/server2/100 |
| 11:19.04 | WIMPy | Also for you: If you know where to find the servers there's no need to register. |
| 11:19.32 | metiu_ | ok wrong wording, I was meaning adding an entry for the server in sip.conf or iax.conf |
| 11:19.54 | WIMPy | You don't have one? |
| 11:20.44 | metiu_ | still studying the issue, in the beginning I just added entries for the few users on each server, with the proper host= line |
| 11:21.00 | metiu_ | but I guess it should be the other way around |
| 11:21.23 | cusco | see... once again http://paste.debian.net/92713/ 401 Unauthorized |
| 11:21.37 | metiu_ | thanks |
| 11:21.55 | *** join/#asterisk Zeeek (~anonymous@pdpc/supporter/active/zeeek) |
| 11:22.17 | Zeeek | slept poorly last night |
| 11:23.26 | *** join/#asterisk Dovid (Dovid@213.8.121.90) |
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| 11:27.06 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:27.06 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 11:35.04 | Naikrovek | what is the newer technology: 3G or EDGE |
| 11:35.12 | Naikrovek | stupid question but i never followed the cell technologies |
| 11:35.29 | TobSnyder | 3g |
| 11:35.41 | Naikrovek | okay thanks |
| 11:36.01 | Naikrovek | got a kindle yesterday |
| 11:36.06 | Naikrovek | well, it arrived yesterday |
| 11:36.07 | Zeeek | 3g or edge? |
| 11:36.15 | Naikrovek | at home i get edge, at work it's 3g |
| 11:36.54 | Naikrovek | free wireless forever :) or until edge & 3g go away, anyway |
| 11:37.46 | Naikrovek | 3g definitely feels faster |
| 11:37.58 | Zeeek | it's way faster |
| 11:38.03 | Zeeek | when you have it |
| 11:38.28 | Zeeek | now you just need to hack the kindle to be able to use that 3g for something else |
| 11:38.50 | Naikrovek | can be done, kindle has been jailbroken for some time |
| 11:38.52 | Naikrovek | but, eh. |
| 11:39.21 | Naikrovek | not fussed on voiding the warranty atm, nor do i want to ruin it for everyone else |
| 11:40.00 | Naikrovek | i don't wanna be one of those guys that hacks all their stuff, giving device manufacturers reason to not include free lifetime 3G connectivity in future devices. |
| 11:40.02 | *** join/#asterisk rossand (~aross@173.243.47.194) |
| 11:41.09 | *** part/#asterisk rossand (~aross@173.243.47.194) |
| 11:42.30 | Naikrovek | some of the free ebooks are very poorly formatted, too |
| 11:42.38 | Naikrovek | which sucks, yo |
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| 11:43.55 | Zeeek | I think we should discuss leifmadsen 's upcoming wedding. It's more related to asterisk |
| 11:44.08 | Naikrovek | hey congrats to him! |
| 11:45.05 | leifmadsen | yay! |
| 11:46.11 | oDesk | how to decode the DTMF signal from wav file into text ? |
| 11:46.12 | Zeeek | so instead of a wedding planner, leifmadsen needs a dialplanner! |
| 11:46.13 | Naikrovek | dude marriage rules |
| 11:46.17 | Naikrovek | you're gonna like it hopefully |
| 11:46.27 | Naikrovek | marriage also sucks (at times) |
| 11:48.52 | Naikrovek | but if it's a good marriage, it's rewarding |
| 11:50.11 | *** join/#asterisk Nwab (~Benwa@unaffiliated/benwa) |
| 11:50.41 | leifmadsen | ya, we have already been living together for over a year and bought a new house (we moved in a month ago) and things have been pretty good so far :) |
| 11:51.03 | Zeeek | living in SIN? :-( |
| 11:51.29 | Zeeek | when she squeeze the toothpaste from the middle, it starts to go downhill fast |
| 11:51.54 | Zeeek | leaving the milk carton out of the fridge |
| 11:52.16 | Zeeek | pretty soon she finds it obnoxious that your shoes are there to be tripped over |
| 11:52.20 | drmessano | When she chases you with a knife, that's a red flag |
| 11:52.34 | Zeeek | and then the beginning of the end, when she realizes you own too many SIP phones :) |
| 11:52.56 | Zeeek | drmessano: no, when she chases you with a red flag, THAT'S a red flag |
| 11:53.02 | drmessano | lol |
| 11:53.22 | Zeeek | a red flag would be normal in Switzerland tho |
| 11:53.38 | Zeeek | so, VoIP Abuse - the ugly secrets |
| 11:53.49 | Naikrovek | flag football.. |
| 11:53.51 | Zeeek | today in about 4 hours on #vuc |
| 11:54.30 | drmessano | VoIP Abuse? Like "Show me on the doll where the Grandstream touched you" ? |
| 11:54.38 | *** join/#asterisk ukine_work (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 11:55.04 | Zeeek | drmessano: no, it's more show me on the VX-1500 where she put her .... |
| 11:55.11 | Zeeek | but we digress |
| 11:55.15 | Zeeek | http://www.voipusersconference.org/2010/voip-abuse-project/ |
| 11:55.25 | Zeeek | We love this idea |
| 11:56.11 | drmessano | Ah, Honeypots |
| 11:56.25 | Zeeek | the word is self-arousing |
| 11:56.32 | schmidts | to catch the big bad hacker bear ;) |
| 11:56.34 | Zeeek | but again, we digress |
| 11:58.00 | Zeeek | come on over to #vuc between now and 4 hours in the future |
| 11:58.05 | Zeeek | or come on over in the past |
| 11:58.10 | Zeeek | and stay for a while |
| 11:58.17 | drmessano | What about a simple Asterisk install that allows an easy registration, but only makes calls to the Chinese governments "Nark on your neighbor" line |
| 11:58.52 | Zeeek | I'd like to find a serious remedy for those robotic calls I get in the middle of the night |
| 12:01.55 | drmessano | Yeah, I had someone asking me for help on something yesterday in another channel that was omitting that he worked for a telemarking firm because he knew all help would stop if he had stated it up front.. but that there's "good telemarketers" and "bad telemarketers". I fail to see a difference, except for the ones exploiting other machines for making calls |
| 12:03.17 | Zeeek | It's really tough to moralize. When telemarketers call, I feel bad about being mean to them because they're struggling to make a living and exploited by the people who pay them |
| 12:03.43 | Zeeek | But the robot ones piss me off and I wish there was a way to infect them! |
| 12:03.52 | fauxalliance | drmessano, i like the 'if you generate revenue, you hire a consultant' rule of thumb |
| 12:04.23 | Zeeek | fauxalliance: totally agree |
| 12:04.37 | *** join/#asterisk viraptor (~viraptor@212.11.65.66) |
| 12:07.22 | drmessano | Well, what annoys me more than anything is that if you were spamming and had an issue with your postfix spooler, they would laugh in your face.. But since the open source telephony community is supposed to be such a friendly bunch, we should happily help them spam people over the phone with Asterisk. |
| 12:07.22 | *** part/#asterisk Montys (~dmartinez@nat/digium/x-clvbdysgzmrnlxxf) |
| 12:07.27 | drmessano | I call BS |
| 12:07.49 | TobSnyder | hello drmessano |
| 12:07.54 | TobSnyder | and all the others on here |
| 12:08.01 | Zeeek | I call Roto Rooter |
| 12:08.02 | drmessano | hi |
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| 12:09.47 | drmessano | Oh, and for what it's worth, Ubuntu 10.10 is pretty sweet |
| 12:09.54 | drmessano | That is all |
| 12:10.04 | viraptor | hi all, I was wondering if anyone has experience with database-based dialplans -vs- fastagi knowing all the logic -vs- very complicated static dialplan with hundreds of variables... any general opinions about this topic? let's say I have enough logic to generate 5M of rows in the db |
| 12:11.48 | TobSnyder | short question: I have "destination if no answer" set to ring groups, but even when someone hangs up his ringing phone, the call goes on to next ring group. Is there a way to avoid this, as I would say: if nobody is answering the phone its okay to go on to next ring group, but if someone hangs up the call without answering, it's clearly he doesn't want to accept that call |
| 12:12.44 | schmidts | viraptor what do you want to know? |
| 12:13.48 | drmessano | TobSnyder, FreePBX questions are more appropriate in #freepbx.. they are frowned upon in here |
| 12:15.20 | viraptor | schmidts: some general ideas about the performance of one -vs- the other... I'm running off of the db right now, but started to migrate some stuff to fastagi - I wonder if it's worth going all the way - if it takes many queries to route one call, maybe there's a better way |
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| 12:15.33 | Zeeek | drmessano: is also frowned upon here :) |
| 12:15.40 | drmessano | ~drmessano |
| 12:15.40 | infobot | [drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway, and is earning his reputation daily, or wearing oompa looma underwear |
| 12:15.46 | Zeeek | frowns upon drmessano |
| 12:16.02 | drmessano | I am the leading cause of censorship in here, afterall |
| 12:16.13 | Zeeek | ~Zeeek |
| 12:16.14 | infobot | rumour has it, zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 12:16.28 | Zeeek | wow that's a little out of date |
| 12:16.42 | schmidts | but still true :D |
| 12:16.42 | Zeeek | still true, though |
| 12:16.45 | Zeeek | yeah |
| 12:19.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:19.39 | Zeeek | oh oh. Better behave now |
| 12:20.04 | Zeeek | member:%5BTK%5DD-Fender is in the building |
| 12:20.38 | *** join/#asterisk mivaho (~mivaho@82-204-46-115.dsl.bbeyond.nl) |
| 12:20.53 | Zeeek | [TK]D-Fender doesn't work with my tab completion |
| 12:29.47 | TobSnyder | ~TobSnyder |
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| 12:47.05 | dandate2 | Does 2 people talking over eachover require 2x the bandwidth at the iphones end? |
| 12:49.57 | *** join/#asterisk willianmazzardo (~willianma@200-203-68-235.smace701.dsl.brasiltelecom.net.br) |
| 12:50.05 | willianmazzardo | tzafrir, hello |
| 12:51.06 | [TK]D-Fender | dandate2: HUH? |
| 12:52.45 | dandate2 | well if i am speaking and someone is trying to interrupt me, does this boost my bandwidth requirement for an ip phone remotely located from the pbx? |
| 12:53.06 | dandate2 | the bandwidth is pooled so uploading drains from the total downloading and vice versa |
| 12:53.44 | schmidts | dandate2 this depends on the signaling you use, if its sip you only have some sip packages incoming to your phone |
| 12:54.39 | willianmazzardo | [TK]D-Fender, hi |
| 12:54.58 | dandate2 | yes using SIP, the bandwidth calculator states the IP phone needs 79.63 kpbs, but if 2 people are trying to talk over eachother does this not double the bandwidth used? |
| 12:55.01 | willianmazzardo | do you have the patch for OSLEC in dahdi 2.4.0? |
| 12:55.03 | dandate2 | since im sending and receiving conversation at the same time |
| 12:55.16 | *** join/#asterisk coppice (~chatzilla@112.203.17.210.dyn.pacific.net.hk) |
| 12:55.53 | [TK]D-Fender | dandate2: RTP is CONTINUOUS in both directions. |
| 12:56.00 | dandate2 | gotcha |
| 12:56.01 | [TK]D-Fender | and SIP != voice. |
| 12:56.12 | Zeeek | SIP === voice |
| 12:56.42 | schmidts | SIP + RTP == Voice |
| 12:56.50 | dandate2 | so to have 6 ip phones at a remote location, they'll just need 477.75 kpbs of total bandwidth regardless of how much arguing is going on right |
| 12:56.56 | [TK]D-Fender | schmidts: No, RTP = voice. |
| 12:57.06 | [TK]D-Fender | schmidts: SIP = Call setup. |
| 12:57.10 | willianmazzardo | SIP = sinalization ! :D |
| 12:57.31 | schmidts | okay let me say this the right way SIP + RTP == CALL |
| 12:57.34 | schmidts | ;) |
| 12:57.49 | [TK]D-Fender | schmidts: SIP = call, RTP = media |
| 12:57.56 | dandate2 | because i'm ordering a T1 line tommarow morning, they gave me the option of 512kb, 768kb, and 1MB. dont wqnt to mess thsi up! |
| 12:59.00 | schmidts | with call i mean the whole call not only the signaling |
| 12:59.46 | dandate2 | so theres absolutely no extra bandwidth used when people are arguing over eachother? i worry because at 512kb i would only have 40kb difference from the total used by the ip phones according to the bandwidth calculato |
| 13:00.04 | dandate2 | so i imagine 1 person interrupting will cause all the lines to break heh |
| 13:01.56 | [TK]D-Fender | dandate2: You don't seem to get it. Voice is CONTINUOS. BOTH WAYS. Packets always flowing even if they are QUIET |
| 13:02.17 | [TK]D-Fender | dandate2: it is not variable |
| 13:02.30 | Zeeek | what about xmit silence? |
| 13:02.47 | [TK]D-Fender | Zeeek: You mean that concept that Asterisk does not support? |
| 13:02.49 | dandate2 | k thx |
| 13:03.00 | [TK]D-Fender | Zeeek: What about it? :p |
| 13:03.18 | Zeeek | yes, the one every SIP phone has a config param for |
| 13:04.02 | dandate2 | is .wav a supported audio codec for voip heh |
| 13:04.03 | [TK]D-Fender | Zeeek: Not every, but lots, sure |
| 13:04.10 | Zeeek | I was watching the bandwidth on a video ocnferencall today. The video bw was about 10k if my image was stable, 12 if I rolled my eyes and 25-50 if my body moved |
| 13:04.38 | Zeeek | Unfortunately, those tests shortened the call and she hung up on me before I could um, finish |
| 13:04.49 | [TK]D-Fender | .... |
| 13:04.49 | Zeeek | <ba da boom> |
| 13:05.17 | Zeeek | rim shot? |
| 13:05.22 | kaldemar | dandate2: .wav is not even a codec. |
| 13:05.31 | dandate2 | right its not compressed at all |
| 13:06.07 | *** join/#asterisk SuPrSluG (~SuPrSluG@host-64-179-106-158.ind.choiceone.net) |
| 13:06.16 | Zeeek | hence the name "lossless" |
| 13:06.25 | Zeeek | or "uncompressed" |
| 13:06.26 | [TK]D-Fender | dandate2: Waht does compression have to do with anything? |
| 13:06.32 | kaldemar | dandate2: it's a format, not a codec. |
| 13:07.35 | dandate2 | .wav and ulaw are the same file sizes tho right? |
| 13:09.01 | drmessano | Speaking of telemarketers |
| 13:09.09 | [TK]D-Fender | dandate2: wav does not imply a specific BITRATE or other properties |
| 13:09.10 | dandate2 | nm i figured it out, i was wondering cuz the pbx can play .wav files for the moh but i just realized it is only transcoding it to ulaw |
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| 13:33.38 | Katty | stretches |
| 13:33.42 | Katty | good mornings |
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| 13:34.09 | Zeeek | Katty: mine's long gone |
| 13:34.23 | Katty | well you can pretend. |
| 13:34.25 | Katty | hugs on Zeeek |
| 13:34.29 | anonymouz666 | why DTMF's coming from GSM network is so hard to work properly? |
| 13:35.14 | Zeeek | Katty: did you get adopted yet? |
| 13:35.23 | Katty | Zeeek: yes. |
| 13:35.39 | Zeeek | who are the lucky family? |
| 13:36.00 | Katty | Zeeek: -> |
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| 14:03.48 | *** join/#asterisk m_tadeu (~quassel@89-181-104-154.net.novis.pt) |
| 14:08.17 | m_tadeu | hi...I'm having some trouble understanding what externip represents in sip.conf. In my case, I have the asterisk server, a dsl router, and sip clients in and outside the local network. Which does the externip represents? |
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| 14:09.03 | tzafrir | willianmazzardo, pong |
| 14:10.13 | anonymouz666 | m_tadeu: você tem que definir o externip e também o localnet. todos os endereços que forem fora do intervalo do localnet serão substituidos pelo externip. |
| 14:10.51 | Zeeek | m_tadeu: that is your IP address that tje world sees |
| 14:11.22 | m_tadeu | so it would be the dsl router external ip? |
| 14:11.47 | [TK]D-Fender | m-It SHOULD be |
| 14:11.48 | anonymouz666 | could be in your case, don't forget to redirect the ports. |
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| 14:12.15 | Zeeek | m_tadeu: yes |
| 14:12.20 | Zeeek | rebooting |
| 14:12.29 | m_tadeu | thanx guys |
| 14:12.32 | *** part/#asterisk Zeeek (~anonymous@pdpc/supporter/active/zeeek) |
| 14:16.44 | m_tadeu | I'm only redirecting the port 5060...is this enough? |
| 14:16.50 | anonymouz666 | of course not |
| 14:17.09 | anonymouz666 | look at your rtp.conf and also redirect the range of rtp ports. |
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| 14:18.10 | m_tadeu | woww... got it ;) |
| 14:18.14 | m_tadeu | thanx again |
| 14:18.44 | anonymouz666 | m_tadeu: is it hard to trust in someone talking portuguese, right? |
| 14:19.27 | m_tadeu | anonymouz666: on the contrary ;) |
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| 14:23.59 | willianmazzardo | tzafrir, do you have a patch file for OSLEC in dahdi 2.4.0? |
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| 14:34.09 | tzafrir | willianmazzardo, try http://tzafrir.org.il/~tzafrir/dahdi_linux_extra.diff |
| 14:34.23 | willianmazzardo | tzafrir, thanks a lot |
| 14:35.27 | anonymouz666 | so a contribution from aligera? nice to see |
| 14:36.14 | anonymouz666 | the APC4XX driver is already in DAHDI 2.4? |
| 14:36.27 | anonymouz666 | AP4XX |
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| 14:43.43 | *** part/#asterisk izod (~support@ds0042.univhosting.com) |
| 14:43.43 | metiu_ | is it possible for 2 softphones on the same machine as the asterisk server to call each other? I have 2 linphones on one host both registered to localhost, if I call one from the other the asterisk log loops forever with "Native bridging SIP/..... and SIP/....." |
| 14:44.15 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 14:45.07 | m_tadeu | metiu_: I never tryed it, but ig you put them in a different por it should work |
| 14:46.03 | metiu_ | yes, they are on 5060 (asterisk), 5062 (linphone1) and 5064(linphone2) |
| 14:46.05 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 14:46.10 | metiu_ | very simple dialplan |
| 14:46.39 | m_tadeu | metiu_: what does asterisk console say? |
| 14:46.49 | leifmadsen | metiu_: ya, using x-lite all I ever had to do was make sure the clients were setup to be on different ports |
| 14:46.53 | leifmadsen | listening on different ports |
| 14:46.55 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:47.14 | metiu_ | e.g. exten => linphoneX,n,Dial(SIP/${EXTEN}-phone) |
| 14:47.20 | metiu_ | e.g. exten => _linphoneX,n,Dial(SIP/${EXTEN}-phone) |
| 14:47.30 | metiu_ | yes different ports |
| 14:47.35 | leifmadsen | exten => _linphoneX will not do as you expect |
| 14:47.45 | metiu_ | just guessing it, sorry |
| 14:47.46 | leifmadsen | exten => _li[n]pho[n]eX |
| 14:47.59 | leifmadsen | N == n == pattern match character |
| 14:48.00 | metiu_ | great that's a very good point |
| 14:48.08 | metiu_ | anyway |
| 14:48.18 | leifmadsen | SIP trace, console output, and RTP debug required |
| 14:48.22 | metiu_ | don't know why it keeps looping saying Native bridging |
| 14:48.26 | leifmadsen | either do we |
| 14:48.29 | leifmadsen | and we don't have the info |
| 14:48.35 | metiu_ | next thing I'll try is to reset all config to supersimple |
| 14:48.37 | leifmadsen | start with the debug |
| 14:48.46 | metiu_ | ok |
| 14:48.56 | metiu_ | back to you in some time thanks |
| 14:48.56 | leifmadsen | you should always start with super simple configs while developing until you get something working |
| 14:49.06 | leifmadsen | heads over to a meeting and some more work |
| 14:49.34 | m_tadeu | so, here's the deal. I have 2 agents waiting for calls in a queue. When I call from outside the network, it runs all the dial plan properly, enters the queue, get moh. When the agent picks up, no sound goes in or out. If I call from inside the local network everything works fine. |
| 14:50.00 | *** join/#asterisk bent_screwdriver (~socain00@173-9-133-102-Miami.FL.hfc.comcastbusiness.net) |
| 14:50.14 | p3nguin | ~sipnat |
| 14:50.14 | infobot | sipnat is, like, Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:51.06 | [TK]D-Fender | p3nguin: unfortunately my server is still down... |
| 14:51.34 | p3nguin | oops |
| 14:51.43 | m_tadeu | no prob...checking sip nat solutions wiki |
| 14:52.51 | *** join/#asterisk smooth_penguin (~root@59.95.24.120) |
| 14:53.28 | *** join/#asterisk bipolar (~bipolar@offsitesysadmin.com) |
| 14:53.42 | p3nguin | I should have stolen a copy of your sipnat stuff for a situation like this. |
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| 14:58.22 | *** join/#asterisk izod (~support@ds0042.univhosting.com) |
| 14:59.28 | bipolar | If asterisk loses it's pri timing source, does it switch back to internal timing or just drop the pri altogether? Our pri vendor called me today to tell me they are seeing some issues with our timing, and wants me to check to make sure our system is still using external timing. I'm not sure how to check. The PRI is set to use external timing, but I don't know if it has a fallback to internal or not. |
| 14:59.45 | *** part/#asterisk c0rnoTa (~c0rnoTa@178.177.142.121) |
| 14:59.57 | m_tadeu | p3nguin: would be nice :) [TK]D-Fender, is there a way to share that info? |
| 15:00.59 | *** part/#asterisk izod (~support@ds0042.univhosting.com) |
| 15:06.32 | smooth_penguin | hey p3nguin |
| 15:07.02 | smooth_penguin | still havnt got my CID working, even with the DTMF - FSK convertor :S |
| 15:08.10 | m_tadeu | everything in the sip nat solutions wiki checked...but still not working |
| 15:08.51 | p3nguin | smooth_penguin: That's a ditty sheal. |
| 15:09.01 | smooth_penguin | :s |
| 15:09.06 | smooth_penguin | any luck on the syslogs |
| 15:09.13 | smooth_penguin | Im still hoping to compare |
| 15:10.12 | p3nguin | Running tcpdump on the syslog port showed some packets, but syslogd never wrote anything to my file. |
| 15:10.19 | smooth_penguin | yeah |
| 15:12.25 | *** join/#asterisk jamicque (~jam@80.50.125.74) |
| 15:13.06 | jamicque | hi is there any way to use zaptel/dahdi service codes like dnd (*79) on sip friends? |
| 15:20.26 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
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| 15:27.48 | Katty | ugah. i'm not feelin so well. this cold weather is rough. |
| 15:28.44 | p3nguin | It has me messed up, as well. |
| 15:31.18 | *** join/#asterisk ManxPower (~manxpower@91.sub-75-204-61.myvzw.com) |
| 15:31.36 | ManxPower | The alabama department of health is AGRESSIVE. |
| 15:33.33 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 15:33.35 | wcselby | o/ |
| 15:34.22 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:34.22 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:43.04 | *** part/#asterisk ManxPower (~manxpower@91.sub-75-204-61.myvzw.com) |
| 15:43.57 | wcselby | in asterisk 1.4, is there some way for the queue / agent system to understand that an agent is on a call, that's not a queue call, and thus not send calls to that agent? agents are logging in with agentcallbacklogin() |
| 15:45.47 | leifmadsen | wcselby: yes, just enable the ability for end points to have device state working |
| 15:45.57 | leifmadsen | wcselby: I presume you mean the calls are delivered via Local channels? |
| 15:46.05 | leifmadsen | oh nevermind, AgentCallbackLogin() |
| 15:46.17 | wcselby | leifmadsen - well, kind of |
| 15:46.29 | leifmadsen | you just need to make sure device state is working (i.e. showing as InUse etc), and enable "ringinuse=no" in queues.conf |
| 15:46.30 | *** join/#asterisk ks3 (~ks3@74.203.195.1) |
| 15:46.55 | wcselby | okay |
| 15:47.02 | wcselby | see, I thought that's what I had done |
| 15:47.08 | wcselby | hmmm, let me look through al lthe conf files again |
| 15:47.18 | wcselby | this doesn't require any kind of patch for device state or anything, does it? |
| 15:47.36 | [TK]D-Fender | leifmadsen: IIRC 1.4 can't associate another device tot he agent channel as that points to indeterminate dialplan. |
| 15:47.37 | leifmadsen | no |
| 15:47.42 | leifmadsen | just make sure device state works correctly |
| 15:47.55 | leifmadsen | are you using Local channels to deliver to agents? |
| 15:47.58 | leifmadsen | and are all end points SIP ? |
| 15:48.01 | [TK]D-Fender | leifmadsen: IIRC that is only an option with AQM |
| 15:48.13 | wcselby | all endpoints are sip, yes |
| 15:48.37 | wcselby | this is my AgentCallbackLogin command -> AgentCallbackLogin(${CALLERID(number)},,${CALLERID(number)}@agent-calls) |
| 15:48.50 | wcselby | and then agent-calls is at this pastebin: .... |
| 15:49.25 | wcselby | http://pastebin.com/sFmKEBkR |
| 15:49.29 | leifmadsen | are your agents setup to be [4165551212] or something? |
| 15:49.46 | wcselby | 4 digit callerid's, when they dial in |
| 15:49.55 | wcselby | so they're Agent/2625, for example |
| 15:50.04 | wcselby | meh |
| 15:50.16 | leifmadsen | ok you're using GROUP() and GROUP_COUNT() -- so you could either enable that in other locations, or you need to make sure device state is working (which is unrelated to those functions) |
| 15:50.18 | [TK]D-Fender | leifmadsen: Just to be clear here, how is AgentCallBackLogin at all aware of the relationship between a SIP device and an Agetn channel Local Channel reference? |
| 15:50.48 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 15:50.59 | leifmadsen | ah ya, I don't use chan_agent. I use AddQueueMember() and RemoveQueueMember() so that the SIP peer is directly in the Queue() so device states work |
| 15:51.03 | [TK]D-Fender | ^^ |
| 15:51.08 | [TK]D-Fender | wcselby: AQM :) |
| 15:51.20 | wcselby | [TK]D-Fender - yeah.... |
| 15:51.22 | leifmadsen | I've personally never used AgentCallbackLogin() |
| 15:51.24 | ks3 | Is it expected that if a transfer context has an h extension, that gets called prior to a pattern-match extension that matches the transfer destination? http://www.fpaste.org/F5zP/ |
| 15:51.25 | wcselby | looks that way |
| 15:55.30 | wcselby | hmmmm |
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| 15:58.49 | ks3 | I seem to be having the same issues as https://issues.asterisk.org/view.php?id=14347#99115, but that was reported fixed over a year ago. |
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| 16:16.17 | Whitor | Howdy... I've got a polycom soundpoint ip 430 that I'm having trouble with... Actually its -all- of my ip-430's. When a user presses the message button, the phone dials the number in a certain field in its web config interface... the "Address" field. Problem is... that the Address is not the number called to access VM ... *98 is. |
| 16:16.27 | Naikrovek | i just answered you in #trixbox |
| 16:16.39 | Naikrovek | your phones are configured incorrectly |
| 16:16.56 | Whitor | woops, sorry. didn't say my nick so it didn't flag me |
| 16:17.01 | Whitor | I'l talk over there |
| 16:18.04 | *** join/#asterisk Letoric (~Letoric@tpi-dfw-uc-f1-69-94-238-22.totalprocess.net) |
| 16:31.17 | vader-- | hmmm cisco sent me a replacement 7940G phone and it's different then my current ones |
| 16:31.25 | vader-- | i can't seem to unlock the settings |
| 16:32.12 | wcselby | where can I see the sourcecode for AddQueueMember() ? |
| 16:34.29 | Kobaz | from app_queue.c |
| 16:34.39 | wcselby | yeah I just found it, thanks :) |
| 16:34.41 | Kobaz | wcselby: is your grep broken? |
| 16:34.42 | Kobaz | heh |
| 16:36.09 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 16:37.22 | ruyo | Can I Playback() a wav file? |
| 16:37.30 | ruyo | Or a mp3 for that matter. |
| 16:37.49 | wcselby | wav yes, mp3, not by defualt |
| 16:38.04 | wcselby | i think you can compile in mp3 support using asterisk-addons, though I've never done this |
| 16:39.30 | ruyo | Ok. About wav then, how can I specify the file type or can I transcode it? |
| 16:39.52 | ruyo | I'm getting [2010-10-01 17:36:02] WARNING[21566]: file.c:992 ast_streamfile: Unable to open usr/thankyou (format 0x1000 (g722)): No such file or directory |
| 16:41.11 | p3nguin | Playback(some-file) |
| 16:43.01 | ruyo | p3nguin, tried Playback(usr/ty) and Playback(usr/ty.wav) |
| 16:43.54 | p3nguin | Do you have a directory named /var/lib/asterisk/sounds/usr/ and a file in it called ty.wav? |
| 16:44.04 | ruyo | Yeah. |
| 16:44.11 | wcselby | it's looking for a g722 formatted thankyou file |
| 16:44.17 | ruyo | I have some .sln files there that are working. |
| 16:44.45 | ruyo | I'll try changing allow's. |
| 16:44.50 | p3nguin | Asterisk should be able to transcode a wav if it was looking for a g.722 file type. |
| 16:44.57 | wcselby | p3nguin -that's what I thought as well |
| 16:45.16 | wcselby | anyways, my brain is hurting, so I'm not much help in here. think i'll go get something to eat |
| 16:46.19 | ruyo | Ah. |
| 16:46.31 | ruyo | Removing g722 I get 2 more errors. |
| 16:46.48 | ruyo | [2010-10-01 17:45:56] WARNING[22686]: format_wav.c:156 check_header: Unexpected frequency 16000 |
| 16:46.48 | ruyo | [2010-10-01 17:45:56] WARNING[22686]: file.c:386 fn_wrapper: Unable to open format wav |
| 16:47.13 | p3nguin | The wav needs to be 8000 mono. |
| 16:47.18 | ruyo | Looks like it's an invalid/weird wav. |
| 16:47.30 | ruyo | Oh, it _must_ be 8000 mono? |
| 16:47.54 | p3nguin | I'd probably open it in audacity, change it to 8000, and export it. |
| 16:47.55 | ruyo | Thought it could transcode any wav. |
| 16:47.58 | p3nguin | Yes, it must be. |
| 16:48.05 | ruyo | Ok, then it's explained. |
| 16:48.09 | ruyo | Thanks. |
| 16:49.51 | *** join/#asterisk slidesinger (~slidesing@pool-98-110-13-28.cmdnnj.east.verizon.net) |
| 16:51.45 | ruyo | Neat, mp3 works as well. |
| 16:54.01 | *** join/#asterisk xpot-mobile (~james@70-91-210-237-BusName-Utah.hfc.comcastbusiness.net) |
| 16:55.43 | metiu_ | new problem: it seems that I cannot Page() a locally connected UA |
| 16:56.18 | metiu_ | it complains about not being able to translate to g723 to slin (I have disabled g723 everywhere, in sip.conf and even in phone.conf) |
| 16:56.37 | metiu_ | i.e. from g723 to slin |
| 16:58.22 | *** join/#asterisk murraytm (~murraytm@wsip-70-183-211-229.br.br.cox.net) |
| 17:00.11 | murraytm | i'm trying do use a callfile to dial out over a sip trunk and play a sound but the callfile just expires instantly. incoming calls work fine. can someone help me with that? |
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| 17:10.52 | bmoraca_work | offline defrag of a 22gb exchange database on a failing drive != fun |
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| 17:13.48 | Stultis | If I have a VPMADT032 configured, should I take out MG2 ec out of the .conf files? |
| 17:15.01 | [TK]D-Fender | Stultis: It won't use SWEC as long as you have HWEC installed. Its good to leave it |
| 17:15.17 | Stultis | thanks |
| 17:15.43 | Stultis | Couldn't find a definite answer online even though the question was asked multiple times |
| 17:17.33 | *** join/#asterisk Kevin` (~kevin@router.kwzs.be) |
| 17:20.05 | murraytm | when i try to dial out over SIP using a callfile, no SIP messages are generated. i think i must be doing something very basic wrong. |
| 17:24.30 | fifer | anyone have any experience with printing fax tiff files as they come in? Specifically with scalling issues? |
| 17:24.41 | [TK]D-Fender | murraytm: Maybe you should PASTEBIN your call file so we can SEE what you are doing. |
| 17:24.50 | fifer | I have a context that uses the system() command to print a fax after it comes in using cups |
| 17:25.11 | [TK]D-Fender | fifer: Perhaps you should ask in #printing |
| 17:25.16 | fifer | it works fine with one exception, the fax is not scalled properly to fit in the page. |
| 17:25.39 | fifer | I can but since this specifically has to do with fax printing there is a real chance someone here has done exactly what I'm trying to do |
| 17:26.16 | fifer | this is as much about the fax aspect of things as it is about cups |
| 17:27.21 | murraytm | [TK]D-Fender: http://pastebin.com/5387daNj going to pastebin the sip.conf also |
| 17:28.47 | [TK]D-Fender | fifer: It's a TIFF. It isn't a fax. It's just a file now. |
| 17:29.41 | fifer | Right but at a diferent resolution likely depending on the fax resolution sent, so a fixed scale will likely not work. |
| 17:30.13 | murraytm | my sip.conf: http://pastebin.com/qJrEXgaC |
| 17:31.28 | [TK]D-Fender | murraytm: You have global SIP debug enabled and get nothing? |
| 17:31.50 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 17:32.08 | [TK]D-Fender | murraytm: Also, don't archive the call-file, reduce your retry times, etc to 5 sec and a maxretry of 3 |
| 17:32.33 | [TK]D-Fender | murraytm: Also you you haven't described how you are preparing that file and passing it off to get called |
| 17:33.48 | murraytm | w/ sip debugging on, asterisk says it's attempting the call, says "using SIP RTP CoS mark 5" then "Really destroying SIP dialog Method: INVITE. But no INVITE was displayed in the debug info. |
| 17:34.36 | murraytm | this call file is just hand written in vim then moved to outgoing. i'll try your retry time suggestions now. |
| 17:36.29 | [TK]D-Fender | murraytm: that is NOT SIP DEBUG. |
| 17:36.58 | *** join/#asterisk dandate2 (~gtejkgjke@124.106.46.100) |
| 17:37.38 | murraytm | sip debug is on. plenty of other messages scrolling by but nothing is generated by this callfile. |
| 17:37.51 | dandate2 | what kind of specs on a machine do i need to run about 60 channels, queues with moh, no transcoding? |
| 17:38.28 | dandate2 | right now i got a dual core 2.8ghz 1mb ram, is that overkill? |
| 17:38.45 | [TK]D-Fender | My analog watch has more RAM than that... |
| 17:38.51 | dandate2 | im sorry |
| 17:38.52 | dandate2 | 1gig ram |
| 17:38.56 | dandate2 | lol |
| 17:39.05 | [TK]D-Fender | dandate2: Taht machine is plenty |
| 17:39.30 | dandate2 | is it overkill tho? reason is im overpaying for my server rental i'd like to just buy my own and put it in a datacenter for $250/mo |
| 17:40.02 | *** join/#asterisk jdoe_ (jdoe@falseprophet.ca) |
| 17:40.04 | dandate2 | but server machines are expensive =( |
| 17:40.11 | dandate2 | so trying to save every penny on a purchase heh |
| 17:40.50 | dandate2 | the weird stuff warehouse has single core machines flatbed u1 slot machines for like $100 |
| 17:41.13 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:42.17 | [TK]D-Fender | dandate2: What happened to your OMG itz teh SH1ZN1T Y0 Optiplex server? |
| 17:43.02 | *** join/#asterisk pyite_mac (~dschreibe@unaffiliated/pyite) |
| 17:43.07 | *** part/#asterisk pyite_mac (~dschreibe@unaffiliated/pyite) |
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| 17:45.47 | bmoraca_work | dandate2: you can get HP DL360 g3 dual processor with 3gb RAM and 72gb 7200RPM hard drives for <$200 shipped on ebay a lot of the time |
| 17:46.22 | murraytm | shortening the RetryTime and setting MaxRetries to 3 didn't change anything. could this have something to do with the host=dynamic in my sip.conf file? |
| 17:46.30 | dandate2 | fender: dont remember what u talking about heh |
| 17:46.54 | dandate2 | ok ill look for that dl360 |
| 17:47.54 | dandate2 | is it possible to backup freepbx and restore onto a new machine without having to do all the configurations again |
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| 17:52.38 | [TK]D-Fender | [13:46]<dandate2>fender: dont remember what u talking about heh <- You remember... the onle You "stole" and they went to complain that it wasn't a "server" like you claim they described it as... |
| 17:52.49 | [TK]D-Fender | dandate2: Your exploits are legendary.... |
| 17:53.36 | [TK]D-Fender | [13:47]<dandate2>is it possible to backup freepbx and restore onto a new machine without having to do all the configurations again <- no this info isn't clearly in FILES you could possibly just copy over raw...... |
| 17:53.55 | [TK]D-Fender | dandate2: And like I should ahve to remind you .... this isn't #freepbx |
| 17:57.00 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 18:14.23 | *** join/#asterisk nsgn (~nsgn@rrcs-24-227-246-117.sw.biz.rr.com) |
| 18:15.12 | nsgn | direct me to a more proper channel if it is not this one, but i'm having frustration with the flash operator panel. any time i drag/drop a call to a parking space both ends get hung up on. drag/drop transfers to actual phones work fine thoughts? |
| 18:17.36 | [TK]D-Fender | nsgn: There is no IRC support channel for that |
| 18:17.55 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 18:18.07 | [TK]D-Fender | nsgn: Perhaps you should look at the AMI request it generaets and debug that vs your dialplan, etc |
| 18:42.39 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 18:44.21 | DrDigi | dandate2, i dont know what your doing but i got 4 1U servers from weird stuff warehouse in person, both had dual opteron 3.0ghz processors and 1gb of ram (i upped them all to 4gb) I installed 300gb hds into each one and I have one running fine as an asterisk system |
| 18:45.29 | DrDigi | the other 3 have a 4U case i got from weirdstuff with a power supply for $45, brand new in the brand new box, i put aQ6600 processor in it and 3TB of disk space and 8gb of ram and put the 4 1u's and the 4u at he.net in fremont ca |
| 18:45.39 | DrDigi | cost me $200 a month for colocation |
| 18:47.10 | Katty | peeks in |
| 18:47.23 | DrDigi | i paid $140 per 1U and bought 2 and went back a few months later and they had them for $95 and i bought 2 more, i spent like $300 on ram, another $200 on hard drives for them and the 4U i had everything but the motherboard already, i bought a usb timer for the 1 asterisk system |
| 18:47.27 | carrar | DL360 G3 or higher aren't too bad |
| 18:47.41 | Katty | gives carrar a fry |
| 18:47.46 | carrar | G4 is 64bit, G3 is 32bit |
| 18:47.57 | beardy | peeks at Katty |
| 18:47.58 | carrar | runs with the FRY!! |
| 18:48.14 | carrar | WHAT DOES IT MEAN! |
| 18:48.17 | carrar | OMG |
| 18:48.23 | Katty | you...eat it? |
| 18:48.25 | Katty | beardy: ohai |
| 18:48.26 | carrar | Double Frys across the sky |
| 18:48.38 | beardy | Katty: allo |
| 18:48.52 | carrar | The Fry is so VIVIDE |
| 18:49.17 | Katty | idk about that. |
| 18:49.19 | Katty | but they are kinda salty |
| 18:49.34 | carrar | It's Double Complete Fry!! |
| 18:49.52 | Katty | you're a looney. |
| 18:50.16 | Katty | or possibly just caffinated. |
| 18:50.16 | Katty | that's another good explination |
| 18:50.28 | carrar | yes CAFFINATED |
| 18:51.02 | carrar | and approx 45 mins away from making another stbx size venti iced hazzlenut latte |
| 18:51.04 | Katty | i've not had a good blood pumping caffeine trip in awhile. |
| 18:51.16 | carrar | give or take a hour |
| 18:51.27 | carrar | I also have to go fix my lawn |
| 18:51.32 | Katty | what broke. |
| 18:51.36 | carrar | I missed a obvious spot yesterday while mowing |
| 18:51.42 | carrar | I can't believe I did that |
| 18:51.43 | Katty | oh i see. |
| 18:51.48 | Katty | how embarassing. |
| 18:51.50 | carrar | yeah |
| 18:51.56 | Katty | i won't tell, k? |
| 18:52.00 | carrar | OK!! |
| 18:52.05 | carrar | I TRUST YOU to tell NO ONE |
| 18:52.05 | Katty | <3 |
| 18:52.10 | wcselby | o/ |
| 18:52.18 | Katty | hi sweety |
| 18:52.28 | carrar | DO NOT TELL wcselby!!!! |
| 18:52.31 | beardy | hugs Katty |
| 18:52.40 | Katty | filters carrar's blood. |
| 18:52.44 | Katty | hugs beardy |
| 18:52.44 | carrar | heh |
| 18:52.53 | carrar | takes another sip |
| 18:53.11 | wcselby | heya Katty |
| 18:53.15 | Katty | dear universe, thank you for this amazing drug called caffeine. |
| 18:53.28 | wcselby | caffeine gives me migraines |
| 18:53.34 | wcselby | well, too much of it does |
| 18:53.43 | wcselby | so, decaf coffee for me all the way |
| 18:53.44 | Katty | nods. |
| 18:53.45 | wcselby | root beer, etc |
| 18:53.50 | Katty | root beer is yummy. |
| 18:53.54 | Katty | and cream soda too |
| 18:54.01 | wcselby | i can drink one or two dr. peppers during the day, but more than that and I'm regretting it |
| 18:54.28 | wcselby | i'm also supposed to avoid chocolate and alcohol....but c'mon, who avoids those? |
| 18:54.34 | wcselby | ;) |
| 18:54.43 | wcselby | i just try to avoid excess |
| 18:54.47 | Katty | i consume those in moderation |
| 18:55.00 | wcselby | and instead, fill myself up with fatty, unhealthy foods |
| 18:55.06 | wcselby | :) |
| 18:55.15 | Katty | i have no room to talk today. i'm eating chicken nuggets. |
| 18:55.16 | wcselby | but they sure taste good! |
| 18:55.28 | Katty | been eating well all week tho. |
| 18:56.09 | Katty | wcselby: that they do :> |
| 18:56.36 | wcselby | lol |
| 18:56.51 | beardy | had green tea |
| 18:56.57 | wcselby | ugh, trying to wrap my head around this queue / agent issue for my client |
| 18:57.08 | wcselby | they're using chan_agent, and it's just messing all kinds of things up |
| 18:57.10 | Katty | think about boston cream cupcakes instead. |
| 18:57.45 | wcselby | i'm afraid i may have to just rewrite their whole queue structure from the gound up, without chan_agent |
| 18:58.20 | wcselby | the problem then becomes, they've got a working, functioning queuemetrics system that, as far as I can tell, relies on chan_agent |
| 18:58.44 | Katty | :< |
| 18:59.16 | wcselby | guess I need to go spend some time on queuemetrics's website looking through manuals and faqs and forums and such |
| 19:03.27 | Naikrovek | just now watched the "double rainbow" video. far out. |
| 19:03.36 | Kobaz | all the way! |
| 19:03.43 | Naikrovek | 16million viewers before you means "you're late, sucka" |
| 19:03.48 | Kobaz | Naikrovek: now that you've watched that... watch this too |
| 19:04.05 | Naikrovek | double rainbow all the way oh man what does it mean |
| 19:04.08 | Naikrovek | that's so intense |
| 19:04.09 | Naikrovek | ... |
| 19:04.12 | Naikrovek | wtf is this dude smoking |
| 19:04.36 | wcselby | Naikrovek - peyote |
| 19:04.39 | Kobaz | Naikrovek: http://www.youtube.com/watch?v=031Dshcnso4 |
| 19:04.44 | Kobaz | watch that too |
| 19:04.44 | wcselby | according to a friend that has smoked much of it |
| 19:04.46 | Naikrovek | oh he's got long hair and a log beard |
| 19:04.49 | wcselby | i think i got the spelling wrong |
| 19:04.51 | Naikrovek | long* |
| 19:04.58 | Naikrovek | that clears a few things up |
| 19:05.45 | *** join/#asterisk martyndev (c815e702@gateway/web/freenode/ip.200.21.231.2) |
| 19:06.23 | *** part/#asterisk martyndev (c815e702@gateway/web/freenode/ip.200.21.231.2) |
| 19:08.03 | Kobaz | double rainbow... all the way |
| 19:08.12 | Naikrovek | all the way |
| 19:08.19 | Kobaz | heh, you watch it? |
| 19:08.38 | Naikrovek | not the psychedelic action scene, watching that now |
| 19:08.44 | Kobaz | k |
| 19:08.54 | Naikrovek | lol |
| 19:08.58 | Naikrovek | flower petals |
| 19:09.19 | Naikrovek | TF2 has a mode like that. |
| 19:09.31 | Naikrovek | you shoot someone and baloons and party streamers come out of them |
| 19:09.49 | Naikrovek | when you blow them to bits, instead of body parts, it's mechanical gears and wrapped presents with bows |
| 19:09.51 | Naikrovek | funny |
| 19:10.47 | Naikrovek | double rainbow ... all the way :) nice ending |
| 19:10.50 | Kobaz | hah yeah |
| 19:12.32 | jdoe | Naikrovek: birthday mode. |
| 19:12.36 | *** join/#asterisk guilhermebr (~Guilherme@189.63.49.247) |
| 19:12.37 | Naikrovek | that's it |
| 19:12.42 | Naikrovek | i love that mode |
| 19:12.44 | Naikrovek | not sure why |
| 19:12.44 | jdoe | Naikrovek: they turn it on for the tfc and tf2 birthdays |
| 19:12.56 | Naikrovek | you can turn it on whenever you want if you run a server |
| 19:12.58 | Naikrovek | birthdaymode=1 |
| 19:13.07 | Kobaz | heh |
| 19:13.10 | Naikrovek | as i recall |
| 19:13.15 | wcselby | Naikrovek - did you hear about the Mann-conomy update for tf2? |
| 19:13.20 | Kobaz | i played tf classic back in the day |
| 19:13.31 | Kobaz | waiting for the little red dot to glow for the sniper |
| 19:13.33 | Kobaz | and then BAM |
| 19:13.37 | Naikrovek | yes, i didn't like it at first, then i learned that the content creators get a cut of the money made from their stuff and then i liked it |
| 19:13.45 | wcselby | lol |
| 19:13.58 | Naikrovek | most of the stuff is craftable |
| 19:14.08 | Naikrovek | all of it is available via random drop so i won't be buying any most likely |
| 19:14.08 | wcselby | all of the new stuff is |
| 19:14.22 | Naikrovek | i crafted a wrapped fish a bit ago :) |
| 19:14.26 | wcselby | haha nice |
| 19:14.29 | jdoe | I don't like it, but it's mostly because I object to the stupid new stuff they added... on the blog in their wank design posts, they made a big deal out of how when they were designing the characters they wanted to make the silhouettes immediately identifiable so you'd know what a character was even if you couldn't see it. |
| 19:14.40 | Naikrovek | "fish kill!" in the combat log when you kill someone with it |
| 19:14.46 | jdoe | ... and then they killed that by making them have wildly different abilities depending on what they're holding ;) |
| 19:14.53 | jdoe | A+, Valve. |
| 19:15.04 | wcselby | i like some of the new toys |
| 19:15.12 | wcselby | scouts get a natasha handgun |
| 19:15.13 | wcselby | that's gonna suck |
| 19:15.24 | Naikrovek | yeah |
| 19:15.26 | wcselby | and they can cover you in mad milk, which the descriptions says "a non-dairy product"....... |
| 19:15.31 | Naikrovek | lol |
| 19:15.33 | Naikrovek | rofl |
| 19:15.35 | wcselby | yeah |
| 19:15.42 | wcselby | the more you think about it....lol |
| 19:15.49 | Naikrovek | all the TF2 stuff is like that |
| 19:15.52 | wcselby | yep |
| 19:15.56 | Naikrovek | the more you think about it, the funnier it all is |
| 19:16.00 | wcselby | the new spy knife is going to suck |
| 19:16.15 | Naikrovek | the spy with the beard with the SLR camera in it, all held on by a rubber band |
| 19:16.20 | wcselby | silent backstab and assume the new identity |
| 19:16.22 | Naikrovek | yes that new spy knife is ... mmm |
| 19:16.27 | Naikrovek | if only i played spy |
| 19:16.35 | wcselby | heh, yeah |
| 19:16.57 | wcselby | the spy video was awesome |
| 19:17.10 | Naikrovek | i tend to focus on one class for several days at a time |
| 19:17.14 | Naikrovek | right now: medic |
| 19:17.16 | Naikrovek | okay |
| 19:17.17 | Naikrovek | wcs |
| 19:17.27 | Naikrovek | i need to know your nick in tf2 so i can friend you |
| 19:17.31 | wcselby | lol, i've got over 100 hours played on soldier, everything else is 5 hours and under |
| 19:17.39 | Naikrovek | nice |
| 19:20.49 | Naikrovek | wcselby: well I play as PepsiMAX usually |
| 19:20.53 | Naikrovek | if you see me, say hi |
| 19:21.07 | wcselby | Naikrovek - lol, did you get those /msg I just sent you? |
| 19:21.18 | Naikrovek | oh .. um. yes. |
| 19:21.22 | wcselby | heh |
| 19:21.24 | Naikrovek | but i didnt' notice them until now |
| 19:21.28 | wcselby | :) |
| 19:32.22 | *** join/#asterisk drmessano-lt (~nonya@pdpc/supporter/active/drmessano) |
| 19:37.28 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 19:38.41 | *** join/#asterisk Alagar (~Administr@122.164.139.214) |
| 19:39.55 | joako | In sip.conf I set qualify=5000 for a sip peer, and afterwards every few hours asterisk stops communicating with the server. it won't communicate again until I stop asterisk and start it again. What causes this? |
| 19:41.14 | pabelanger | deadlock? |
| 19:47.00 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 19:47.29 | joako | pabelanger, No, everything is working except 1 SIP peer, and sip debug shows asterisk sends messages but never gets a response. If I place a call from peer to asterisk I see nothing in sip debug |
| 19:48.03 | [TK]D-Fender | joako: Clearly a networking issue |
| 19:48.12 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:48.38 | joako | [TK]D-Fender, If it's a networking issue, how can only restarting asterisk solve it? |
| 19:49.25 | [TK]D-Fender | joako: Because * got tired of sending and never getting a response and restarting * resets the "Am I tired of this non-responsive moron" counter |
| 19:51.06 | joako | The SIP peer is reachable, if i disable qualify it works fine seeminly forever, if I enable qualify for that peer it doesn't last more than 12-18 hours and then dies until I restart asterisk |
| 19:51.33 | joako | and since I am not registering, it makes no sense that sip debug does not show any messages from that peer when this happens |
| 19:51.52 | joako | If what you said were correct I would still see SIP messages from that peer |
| 19:52.14 | joako | And I see astersk sending SIP OPTIONS, so it isn't "tired of non-responsive moron" it keeps on trying |
| 19:53.30 | [TK]D-Fender | joako: Well you say * does send out requests that never get answered.. lack of answer... well... not an * problem |
| 19:54.27 | joako | No, it seems like a bug in *. Otherwise how can restarting * control the remote peer? |
| 19:54.57 | joako | qualify=5000 peer doesn't stay up *TO ASTERISK* more than 18 hours, when it dies restart asterisk and it works again |
| 19:55.33 | [TK]D-Fender | joako: I'd have to see a lot more comprehensive data to drill this much more.. |
| 19:55.46 | nextime | is there any browser oriented free sip client cross-platform ( a java applet maybe? ) deploiable on my own pbx? |
| 19:56.47 | joako | [TK]D-Fender, hence my question. Where do I look? |
| 19:58.14 | *** join/#asterisk zoid_ (~awainer@190.2.14.213) |
| 19:59.02 | zoid_ | Hi, I'm new to asterisk amn I'm having problems setting up extensions.conf for an ivr |
| 19:59.09 | zoid_ | so far, I have this http://pastebin.com/XLrkfh9Y |
| 19:59.27 | zoid_ | the problem is at lines 11 and 12 |
| 19:59.36 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:59.44 | zoid_ | I want to read a DTMF and act upon, send the user to a different menu |
| 20:00.10 | zoid_ | but both gotoifs return true, so the first one allways exectutes |
| 20:00.40 | wcselby | zoid_ - there's many ways to do that, but to address your specific situation, you need to add a $[] around ${opt}=2 |
| 20:01.31 | zoid_ | like this? exten => s,n,GotoIf($[${opt}=1]?mainmenu,1,1) |
| 20:01.55 | wcselby | yeah, that looks bettter |
| 20:01.57 | wcselby | better* |
| 20:02.13 | zoid_ | thank you wcselby, I'll try that |
| 20:02.14 | *** join/#asterisk TimeRider (~steve@5ac7b3c5.bb.sky.com) |
| 20:02.20 | wcselby | also, you could also look at WaitExten() instead of Read() |
| 20:02.23 | *** join/#asterisk ybit2 (~quassel@unaffiliated/ybit) |
| 20:04.17 | pabelanger | Looks like Aastra is hiring, if your based in Toronto |
| 20:05.21 | wcselby | hmmmm |
| 20:05.48 | wcselby | ringinuse=no, queue show support shows member is In Use, but still sends calls to it. |
| 20:06.23 | wcselby | using addqueuemember with a SIP interface listed in the last field |
| 20:09.04 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 20:09.40 | *** join/#asterisk fullstop (~fullstop@64-121-41-67.c3-0.tlg-ubr1.atw-tlg.pa.cable.rcn.com) |
| 20:09.44 | fullstop | happy friday, all! |
| 20:11.06 | fullstop | Let's say that I am interfacing with another system over SIP. Our system handles the IVR and other services and, if required, the caller is transferred to a remote system. |
| 20:11.40 | drmessano-lt | Lets not say that |
| 20:11.49 | fullstop | oh no! |
| 20:11.52 | fullstop | ;-) |
| 20:11.58 | drmessano-lt | Lets use IAX instead and add in some clowns |
| 20:12.23 | fullstop | I was hoping for some RFC 1149 |
| 20:13.37 | fullstop | Anyway, on the other end, calls may be placed into a queue based on the number of available agents. |
| 20:14.02 | fullstop | Their billing numbers include the call time from when the call leaves the queue until termination. |
| 20:14.20 | fullstop | There is absolutely no way of knowing how long the call sits in their queue, correct? |
| 20:18.03 | carrar | Write a API |
| 20:18.35 | carrar | Write some drivers |
| 20:18.41 | carrar | and write some programs |
| 20:18.54 | carrar | Then click GO |
| 20:19.12 | *** join/#asterisk ChUbB (~IceChat7@cpc2-aztw12-0-0-cust229.aztw.cable.virginmedia.com) |
| 20:19.35 | carrar | What kind of queuing phone system has no logs fullstop |
| 20:19.46 | carrar | Make it happen!! |
| 20:19.58 | carrar | Report back in a hour on how you made it work |
| 20:20.30 | carrar | I'm gonna go mow what I missed in my yard yesterday now |
| 20:20.47 | carrar | clock is ticking |
| 20:20.55 | beardy | Vroomvroom. |
| 20:21.24 | fullstop | It's not my queue -- it's some other black box in this situation |
| 20:25.17 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 20:34.55 | carrar | define black box |
| 20:35.04 | carrar | research it |
| 20:39.43 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 20:42.20 | *** join/#asterisk rossand (~aross@red-gw41.cs.toronto.edu) |
| 20:43.04 | *** part/#asterisk rossand (~aross@red-gw41.cs.toronto.edu) |
| 20:58.01 | *** join/#asterisk Nwab (~Benwa@unaffiliated/benwa) |
| 21:01.18 | carrar | makes obvious coffee sipping sounds in earshot of Katty |
| 21:03.54 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:07.56 | *** join/#asterisk xpot-mobile (~james@70-91-210-237-BusName-Utah.hfc.comcastbusiness.net) |
| 21:15.08 | *** join/#asterisk jsidhu (~js@173-8-149-45-SFBA.hfc.comcastbusiness.net) |
| 21:18.05 | *** join/#asterisk n3hxs (~HAMming@63.68.135.4) |
| 21:18.28 | *** join/#asterisk p3nguin_ (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 21:19.36 | jsidhu | hey guys, i just setup a bandwidth.com sip trunk and whne making a call to it, the calling party doesnt hear any ringing.. just silence.. would someone point me in the right direction on where to look? google wasnt much help, perhaps my keywords are not good enuf.. |
| 21:24.45 | joako | jsidhu, do you see 180 Ringing in your sip debugs? |
| 21:25.07 | wcselby | freaking commas |
| 21:29.00 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 21:31.20 | *** join/#asterisk Nwab (~Benwa@unaffiliated/benwa) |
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| 21:47.00 | *** part/#asterisk clintc (~clintc@n128-227-58-86.xlate.ufl.edu) |
| 21:51.06 | *** join/#asterisk astassistant (~skip@216.160.91.89) |
| 21:51.11 | *** part/#asterisk astassistant (~skip@216.160.91.89) |
| 21:54.42 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 21:58.39 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
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| 22:01.05 | *** part/#asterisk rossand (~aross@173.243.47.194) |
| 22:08.51 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 22:11.41 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 22:14.14 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 22:19.54 | *** join/#asterisk lesouvage (~lesouvage@524947AA.cm-4-2b.dynamic.ziggo.nl) |
| 22:23.07 | *** join/#asterisk oDesk (~f@188.52.111.11) |
| 22:23.26 | oDesk | can i join current active channel ? |
| 22:23.58 | WIMPy | oDesk: ??? |
| 22:24.05 | lesouvage | Does anybody of you knows about ESPA (European Selective Paging Manufacturing Association) support for Asterisk? It seems to be a regular interface to interface beepers with Asterisk, but I never heard about it. |
| 22:24.17 | oDesk | i mean how can i join current active call between two parties and speak to them ? |
| 22:25.15 | WIMPy | lesouvage: Never herad before. I only know UCP or TAP. |
| 22:25.36 | WIMPy | oDesk: Throw them and yourself into a meetme. |
| 22:27.53 | oDesk | WIMPy: if you can assistme to do so using asterisk CLI ? |
| 22:28.05 | *** join/#asterisk astassistant (~skip@216.160.91.89) |
| 22:28.08 | *** part/#asterisk astassistant (~skip@216.160.91.89) |
| 22:28.34 | lesouvage | There is a requirement in a paper for ESPA compliances but I actually don't know what it excactly is. What should be interfaced to a phonesystem when a beeper beeps? |
| 22:28.43 | WIMPy | oDesk: You have to use AMI. |
| 22:28.57 | oDesk | WIMPy: btw, why not to use something like Spycan context but this is different because i want to speak and they can hear me |
| 22:28.58 | WIMPy | AFAK you can't divert both call legs via CLI. |
| 22:30.07 | WIMPy | oDesk: IIRC you can "whisper" when spying, but that would only be to one of the other partivipants, as far as I understand it. |
| 22:31.47 | oDesk | WIMPy: sorry i meant ChanSpy() |
| 22:32.45 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 22:32.56 | WIMPy | Ok, it says you can actually talk to both. |
| 22:35.50 | oDesk | WIMPy: what parameter i need to pass to Chanspy inorder to do so ? |
| 22:36.12 | WIMPy | core show application chanspy |
| 22:39.52 | oDesk | WIMPy: i think it's mentioned that pressing 5 will change the mode to whisper |
| 22:40.36 | WIMPy | And 6 to barge. |
| 22:40.48 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 22:41.23 | oDesk | WIMPy: perfect , thank you for your help |
| 22:42.12 | *** join/#asterisk mducharme-laptop (cebc7904@gateway/web/freenode/ip.206.188.121.4) |
| 22:42.17 | mducharme-laptop | afternoon |
| 22:42.25 | mducharme-laptop | or evening rather |
| 22:42.48 | mducharme-laptop | I'm having an outbound caller ID issue.. it appears that my caller id on one server is passing out the extension number instead of the phone number |
| 22:46.38 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 22:57.56 | carrar | mducharme-laptop, set your caller id number to something |
| 22:58.15 | mducharme-laptop | it is set, that's what I'm so confused about |
| 22:58.35 | carrar | i don't see it |
| 22:58.50 | carrar | PROOVE IT |
| 23:00.15 | carrar | tick |
| 23:00.17 | carrar | tick |
| 23:00.20 | carrar | tock |
| 23:00.22 | carrar | tick |
| 23:00.23 | carrar | tock |
| 23:00.36 | carrar | drops a pin |
| 23:01.37 | mducharme-laptop | it looks like it's an issue in the database |
| 23:01.42 | mducharme-laptop | asterisk can't read the users caller id from the database |
| 23:01.43 | carrar | yeah |
| 23:01.44 | mducharme-laptop | so it's using the extension |
| 23:01.54 | carrar | awesome! |
| 23:02.11 | *** join/#asterisk [cannibalera] (~cannibale@201-41-239-27.fnsce703.dsl.brasiltelecom.net.br) |
| 23:02.14 | carrar | let me know if there is anything else I can help you fix |
| 23:02.30 | carrar | balance due: $250 |
| 23:05.21 | [TK]D-Fender | "Database" .... pardon? |
| 23:05.34 | carrar | yeah |
| 23:05.51 | carrar | You didn't see the paste? |
| 23:06.37 | [TK]D-Fender | [18:58]<carrar>i don't see it |
| 23:06.47 | carrar | mducharme-laptop wanted help so he deceded to show us abslutly nothing |
| 23:07.05 | mducharme-laptop | sorry, it's hard for me to paste from freepbx |
| 23:07.13 | carrar | thats good |
| 23:07.18 | carrar | since we don't support freebpx |
| 23:07.20 | WIMPy | offers brand new Net-Ray glasses for 99.95 each |
| 23:07.35 | carrar | join #freepbx |
| 23:07.42 | carrar | be FREE!! |
| 23:28.24 | joako | I enabled qualify=5000 for a SIP peer and ever since I loose connection to that peer ever few hours. Restarting asterisk resolves the issue, how can I fix this? |
| 23:30.32 | mducharme-laptop | carrar there is no help there |
| 23:30.38 | mducharme-laptop | I really need to get this working |
| 23:30.44 | [TK]D-Fender | joako: patebin peer dumpss, SIP debug, call attempts, AstDB dumps, etc. FFS SHOW US SOMETHING |
| 23:31.14 | mducharme-laptop | are there any debugging commands that will show me the callerid lookup process? |
| 23:31.30 | mducharme-laptop | when I run a database show ampuser |
| 23:31.53 | mducharme-laptop | the outboundcid for all users shows up just fine |
| 23:32.54 | joako | [TK]D-Fender, SIP debug shows only Asterisk end, nothing from the SIP peer, what else do you want me to say? |
| 23:32.56 | mducharme-laptop | when the pri card gets it, it is only getting the 4 digit extension instead of the direct line |
| 23:33.32 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 23:33.42 | [TK]D-Fender | mducharme-laptop: That is FreePBX dialplan busllshit and is NOT supported here |
| 23:37.21 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 23:39.45 | [TK]D-Fender | joako: If nothing comes back from the peer, there isn't much to debug is there? How is it *'s fault if they don't answer? |
| 23:41.52 | joako | [TK]D-Fender, when I restart * the peer comes back online. I understand what you are saying, but it is not that the peer has gone down. The moment I restart Asterisk -- be it 5 minutes or 5 hours after the peer becomes unreachable -- the peer becomes reachable again |
| 23:48.50 | [TK]D-Fender | joako: Do you see regular OPTIONS traffic on it all this time? Did you check logs for timeout warnings? |
| 23:55.32 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 23:59.03 | dandate2 | so im going to host my own pbx in a datacenter and need to know how much bandwidth to purchase. we see at the most 30 callers in at one time. 6-8 speaking with a live agent (remote from server) and 22-24 waiting in queue. is 5MB overkill for this? |