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| 07:20.13 | gdeeble | Is there way to have a read command(exten=>inbound,n,Read(Menu|||1|1)) wait 1-3 seconds before it moves to the next step? I've tried the command above, which goes from it to a GotoIf statement which looks for input, and continues if nothing/wrong info in there, but it seems to delay for at minimum 10 seconds and I want it to be a max 3, giving just enough time to press something like * to |
| 07:20.13 | gdeeble | change call flow without anyone knowing. |
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| 07:42.33 | mirela666 | gdeeble: Read(variable[|filename][|maxdigits][|option][|attempts][|timeout]) |
| 07:42.33 | mirela666 | timeout is not helping? |
| 07:42.33 | mirela666 | <PROTECTED> |
| 07:43.43 | mirela666 | gdeeble: Try out : exten=>inbound,n,Read(Menu|||1|1|3) |
| 07:46.44 | gdeeble | Let me try, as I noticed that I was using | rather than , but didn't fix. so let me try that. |
| 07:47.43 | gdeeble | Same thing, except the CLI shows now it's only accepting 1 digit |
| 07:48.07 | gdeeble | maybe i missed something? I'm running 1.8.5.0 |
| 07:48.29 | gdeeble | eww i think i see my problem. |
| 07:50.35 | gdeeble | mirela666: you rock, apparently because way i was putting it in, I apparently missed 1 more , and didn't even realize it. |
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| 08:05.59 | mirela666 | gdeeble::) glad to help |
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| 08:37.12 | *** join/#asterisk rfrail3 (~rfrail3_@213.229.163.5) |
| 08:37.39 | rfrail3 | hello |
| 08:37.54 | rfrail3 | what is the last dots and one number in this line |
| 08:37.55 | rfrail3 | EXTEN:1 |
| 08:37.58 | ChannelZ | holla |
| 08:38.21 | ChannelZ | ${EXTEN:1} means everything of ${EXTEN} minus the first character |
| 08:38.48 | rfrail3 | ok |
| 08:38.50 | rfrail3 | thanks! |
| 08:39.30 | ChannelZ | yup |
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| 08:40.42 | kaldemar | rfrail3: https://wiki.asterisk.org/wiki/display/AST/Manipulating+Variables+Basics |
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| 11:13.47 | hwt | hi |
| 11:13.47 | hwt | i have a bit of a weird problem. when setting up a call (bridging two legs), the calling party sends an SDP in the 183. the port is changed in the 200 OK, but asterisk still attempts to send the media to the port from the SDP in the 183. known bug? |
| 11:15.14 | WIMPy | What's your nat configuration? |
| 11:15.36 | hwt | nat=no |
| 11:15.37 | _omer | Is freepbx created by Digium ? |
| 11:15.42 | hwt | WIMPy: |
| 11:15.48 | WIMPy | _omer: no |
| 11:16.01 | _omer | that's why, it's a piece of SHIT ! |
| 11:16.07 | hwt | it seems to be doing comied/symmrtp on one of the legs, but not the other |
| 11:17.07 | _omer | queue context is not there in freepbx with asterisk 1.8 :-/ I am really stucked in freepbx |
| 11:17.47 | WIMPy | ~freepbx |
| 11:17.47 | 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 |
| 11:20.35 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.164.116) |
| 11:20.51 | FireAndIce | Hi everyone!! |
| 11:20.59 | *** join/#asterisk bulkorok (~chatzilla@gw1.pinguin.ag) |
| 11:21.20 | FireAndIce | Is there a way to create custom sip header with To: and From: fields so that I can test from asterisk CLI? |
| 11:25.36 | FireAndIce | Like this one, http://pastebin.centos.org/2500/ ? |
| 11:26.14 | kaldemar | test what? |
| 11:26.40 | bulkorok | wdoekes: ping ?! |
| 11:27.42 | FireAndIce | I'm unable to make a call from CLI to softphone. On sip show peers, the status is UNREACHABLE. |
| 11:28.12 | FireAndIce | The asterisk server is behind NAT and the phone uses a data carrier plan. |
| 11:28.51 | FireAndIce | Moreover, the asterisk server is bound to 192.168.0.102, but why does it show 192.168.0.105 in the sip message? |
| 11:29.41 | kaldemar | FireAndIce: because you haven't told it otherwise. you need nat=yes, exteraddr and localnet configured under [general] in sip.conf. |
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| 11:29.54 | FireAndIce | I've done that.. |
| 11:30.34 | kaldemar | FireAndIce: maybe you've misconfigured it somehow. |
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| 11:31.12 | FireAndIce | kaldemar, I'll share the file with you, just a second. |
| 11:31.50 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:33.53 | FireAndIce | kaldemar, http://pastebin.centos.org/2503/ |
| 11:35.32 | kaldemar | your permit lines are pointless, btw. |
| 11:36.09 | FireAndIce | ok, I'll comment them. |
| 11:39.33 | kaldemar | what does your "sip show settings" say? |
| 11:40.46 | FireAndIce | Ohh, it has Externaddr: 192.168.0.105 |
| 11:41.59 | FireAndIce | kaldemar, http://pastebin.centos.org/2506/ |
| 11:43.04 | FireAndIce | ifconfig shows me 192.168.0.102 |
| 11:46.03 | kaldemar | your externhost resolves to 192.168.0.105. |
| 11:46.45 | kaldemar | the point of that setting is to use an address that is publicly available. |
| 11:47.41 | FireAndIce | yes, that's right and on nslookup it gives me my public address. |
| 11:48.37 | FireAndIce | I'm using dns. |
| 11:51.17 | FireAndIce | And I've port forwarded to 192.168.0.102:5059~5061 on my router. |
| 11:55.19 | FireAndIce | FireAndIce, I want my asterisk server to be accessible over the internet. |
| 11:55.42 | FireAndIce | kaldemar, |
| 11:57.14 | kaldemar | then use a public address for it |
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| 11:58.21 | FireAndIce | that's why I'm using fqdn of the server in Externhost. Please correct me if I'm wrong. |
| 11:58.51 | FireAndIce | My public ip is dynamic that's why using a dns service. |
| 12:01.51 | kaldemar | the name on your asterisk server resolves to a private address. that is your issue. |
| 12:04.12 | FireAndIce | ok, I'll check it out. thanks for helping me with patience. :) |
| 12:10.06 | FireAndIce | kaldemar, you are absolutely right. my asterisk server resolves to 192.168.0.105. It showed me this address on ping. |
| 12:10.57 | wdoekes | bulkorok: ~ask |
| 12:11.10 | wdoekes | ~ask |
| 12:11.10 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:11.26 | bulkorok | it's about jira 21041 |
| 12:11.39 | wdoekes | ASTERISK-21041 |
| 12:11.40 | LieutPants | [ASTERISK-21041] [Status: Open] Asterisk crashes during a frame copy while receiving a fax - https://issues.asterisk.org/jira/browse/ASTERISK-21041 |
| 12:12.32 | *** join/#asterisk seik0 (b244f143@gateway/web/freenode/ip.178.68.241.67) |
| 12:13.31 | bulkorok | I'm sure that I have a pcap... anything special you are looking for!? |
| 12:14.43 | wdoekes | including the SIP setup? |
| 12:14.54 | wdoekes | and the rtp and udptl?\ |
| 12:15.21 | bulkorok | sure |
| 12:15.37 | bulkorok | some huge files... ~ 1G |
| 12:16.17 | seik0 | Hi, everyone. Today my questions are easy. I have to asterisk (let call them 1.8 and 1.4). I make a call from 1.8 to 1.4 via IAX2, where 1.4 executes plan, where calls some extension SIP/123. sometimes (now i'm not sure everytime or from time to time) SIP/123 is "busy here", but 1.8 got "answered", but because it's actually "busy" we get hangup |
| 12:16.33 | wdoekes | then one should be able to replay the pcap, similar to https://code.osso.nl/projects/sipp/browser/scenario/sendfax.xml |
| 12:16.56 | wdoekes | and that should hopefully cause the same crash |
| 12:17.08 | FireAndIce | kaldemar, got it, my /etc/hosts file had 192.168.0.105. A week back I had assigned this address. Forgot to change it. Thank you so much for your help. :) |
| 12:17.20 | *** join/#asterisk Draecos (~Draecos@124-168-234-57.dyn.iinet.net.au) |
| 12:18.22 | bulkorok | wdoekes: so I have to put the SIP data and execute it with sipp...?! |
| 12:19.02 | bulkorok | ah I see... pcap file... |
| 12:19.36 | bulkorok | wdoekes: I'll check if I can get it run this week... thanks! |
| 12:19.38 | wdoekes | if you browse the pcap you should be able to filter away all unnecessary data from it, and create a new smaller one |
| 12:19.50 | wdoekes | it'll take a bit of work, I know |
| 12:20.18 | wdoekes | let me know if you need assistance |
| 12:20.27 | bulkorok | I'll do |
| 12:20.30 | kaldemar | FireAndIce: np |
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| 12:29.38 | seik0 | nobody has a tip? |
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| 12:38.05 | seik0 | [TK]D-Fender, you always have some advice =) |
| 12:38.14 | seik0 | I have to asterisk (let call them 1.8 and 1.4). I make a call from 1.8 to 1.4 via IAX2, where 1.4 executes plan, where calls some extension SIP/123. sometimes (now i'm not sure everytime or from time to time) SIP/123 is "busy here", but 1.8 got "answered", but because it's actually "busy" we get hangup |
| 12:38.54 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-iftxxbgqajktnikz) |
| 12:39.09 | WIMPy | What do your dialplans do? |
| 12:40.16 | [TK]D-Fender | It got "answered" because you answered |
| 12:40.25 | [TK]D-Fender | something in the remote end's dialplan caused it to |
| 12:40.47 | seik0 | don't have Answer() explicitly (at least) |
| 12:41.18 | seik0 | dialplay sets some variables, dials target, makes some work depending on Dial result |
| 12:41.56 | seik0 | i can print dialplan as it is |
| 12:42.04 | seik0 | if it's interesting |
| 12:42.10 | [TK]D-Fender | seik0: if a bridged call doesn't actually answer, then it is something else in your dialplan. Get digging. |
| 12:47.33 | seik0 | ok, i've got something. does Voicemail(...) answer the call? |
| 12:48.25 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 12:48.55 | [TK]D-Fender | Clearly. |
| 12:48.58 | *** part/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 12:49.31 | seik0 | even if vm-box does not exist? |
| 12:49.34 | [TK]D-Fender | Server A calls server B. Not "server B's phone". Voicemail gives Server A audio. that is answering the call. |
| 12:49.36 | [TK]D-Fender | ^^^ |
| 12:49.55 | seik0 | it's clear |
| 12:50.41 | seik0 | thanks |
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| 13:53.05 | BorjaGVO | Hi everyone. Anyone can tell me how to use regex when using eventfilters in manager.conf? |
| 13:57.23 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.164.36) |
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| 14:08.19 | *** join/#asterisk cousin_luigi (~luigi@unaffiliated/cousinluigi/x-395723) |
| 14:08.22 | cousin_luigi | Greetings. |
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| 14:11.45 | cousin_luigi | I need to set up an answering machine that would play a different message depending on the CLI (or lack thereof). As bonus I could use recording some of those calls. Would Asterisk be a good match for this? What are the computing power requirements for such a purpose? |
| 14:12.21 | WIMPy | yes, next to none. |
| 14:12.52 | cousin_luigi | I should specify that my server is a Geode-based SBC, so very weak in the CPU department. |
| 14:13.38 | WIMPy | You're not the first to run Asterisk on some router or the like. |
| 14:14.07 | cousin_luigi | Good to hear that. |
| 14:14.47 | cousin_luigi | Is Asterisk the best PBX for this task? |
| 14:15.25 | WIMPy | Asterisk is not a PBX. |
| 14:15.45 | WIMPy | Some say you can use it to bilid one. |
| 14:15.54 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.138) |
| 14:16.51 | cousin_luigi | WIMPy: Ok, is it the software to build PBXes that you would recommend for this? |
| 14:17.25 | WIMPy | I'm pretty sure all others will do it just as well. |
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| 14:38.42 | linocisco | hi all |
| 14:39.06 | linocisco | is there anyone who is using small form factor PC hardware for 24/7 asterisk running? |
| 14:39.18 | linocisco | lowest Watt powersupply |
| 14:39.53 | WIMPy | just mentioned that people install Asterisk on their routers. |
| 14:40.15 | WIMPy | And there's even a distro for the Raspberry PI. |
| 14:40.40 | *** join/#asterisk malmut (~malmut@176.104.132.24) |
| 14:43.35 | navaismo | linocisco, raspberry pi |
| 14:43.57 | linocisco | WIMPy, which distro? |
| 14:44.04 | linocisco | WIMPy, raspberry is found as motherboard. not as a whole PC |
| 14:44.37 | WIMPy | Why isn't it a whole PC? |
| 14:45.03 | WIMPy | And PC is rather big compared to what I said. |
| 14:45.40 | linocisco | WIMPy, so... |
| 14:46.08 | WIMPy | AFAIK there are binary packages available for ddwrt/openwrt as well as Freetz as well. |
| 14:47.00 | WIMPy | So people are obviousely using much smaller hardware than what you're asking for. |
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| 14:56.22 | leifmadsen | linocisco: astlinux for net4801 and others |
| 14:56.31 | leifmadsen | runs as my primary router |
| 14:56.35 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:56.35 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:59.18 | coppice | leifmadsen: how fast can that thing route? |
| 14:59.31 | leifmadsen | not sure |
| 14:59.36 | leifmadsen | it routes fast enough for my network |
| 15:00.24 | coppice | well, we have 1G service, and few things can keep the internet link busy. I have a small i3 box doing my routing |
| 15:00.37 | leifmadsen | I'm not that lucky in Canada |
| 15:00.49 | leifmadsen | I have like... 22/1 service or something |
| 15:01.23 | coppice | we have 1G/1G |
| 15:01.29 | leifmadsen | that's nice |
| 15:10.40 | linocisco | leifmadsen, i m looking for what brand is net4801 |
| 15:12.53 | leifmadsen | linocisco: I typed "net4801" into google and the maker came up as the first link |
| 15:13.07 | linocisco | do u also try solar system for asterisk box for 24/7 power |
| 15:13.58 | coppice | there are solar powered asterisk boxes |
| 15:15.42 | linocisco | coppice, namely pls? |
| 15:16.10 | [TK]D-Fender | NET4801 is their older model... |
| 15:16.20 | [TK]D-Fender | Soekris = pricy & weak. |
| 15:16.32 | linocisco | leifmadsen, hi bro, i have found net4801 from soekris. how can we put digium cards or PSTN or GSM lines on it? |
| 15:16.47 | [TK]D-Fender | Atom box is actually cheaper, and considerably more powerful |
| 15:16.48 | *** join/#asterisk threesome (~threesome@2E6B6602.dsl.pool.telekom.hu) |
| 15:16.57 | linocisco | [TK]D-Fender, don't u recommend. ? ok. this will be good debate. go ahead |
| 15:17.08 | linocisco | [TK]D-Fender, atom box? |
| 15:17.30 | [TK]D-Fender | linocisco: Debate ended. Atom = more powerful. Look at what boxes you can find to fit your profile. |
| 15:17.42 | WIMPy | Use an old Laptop with a broken screen :-) Even comes with built-in UPS :-) |
| 15:18.14 | linocisco | WIMPy, but no cards or no PSTN or GSM slots inside as built in |
| 15:18.36 | WIMPy | Xorcom will help you there. |
| 15:19.43 | WIMPy | And even laptops come with PCIe, even if you need an adaptor. |
| 15:20.46 | coppice | linocisco: these things are solar powered http://www.villagetelco.org |
| 15:24.53 | linocisco | coppice, tango yankie |
| 15:25.11 | linocisco | out |
| 15:25.47 | linocisco | WIMPy, I hate pricy isareli brand xorcom |
| 15:27.33 | coppice | a large xorcom box attached to a raspberry pi has a pleasing whimsy about it |
| 15:28.02 | WIMPy | You can put the RPI in to the Astribank case. |
| 15:28.41 | WIMPy | Et voila, you've got a appliance :-) |
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| 15:30.08 | leifmadsen | WIMPy: patent pending! |
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| 15:35.57 | tzafrir | [TK]D-Fender, "Atom = more powerful": consumes more power? |
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| 15:40.44 | [TK]D-Fender | tzafrir: Probably a nominal amount, yes. |
| 16:05.53 | cousin_luigi | bbl |
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| 17:00.30 | navaismo | Hi im trying to add new fields to the CDR, so I edited the cdr.h and so far the cdr_mysql & cdr_psgql are working fine inserting the new value, but the csv show the filed empty. I edited the cdr_csv.c to add the new value with append_string(buf, cdr->NEWVAR, bufsize) but still empty that field. |
| 17:00.43 | navaismo | What I'm missing? |
| 17:05.21 | leifmadsen | navaismo: it would be much easier to do it if you just used cdr_adaptive_odbc |
| 17:05.52 | leifmadsen | navaismo: otherwise, your question is more appropriate for #asterisk-dev, although a good chunk of the US is on holidays today |
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| 17:07.39 | navaismo | Thanks |
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| 19:13.20 | WIMPy | Oh, niche. You can "make isntall". |
| 19:13.26 | WIMPy | -h |
| 19:14.07 | navaismo | ¿?¿? |
| 19:14.42 | WIMPy | made that typo, but Asterisk installed anyway. |
| 19:23.02 | *** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 19:23.34 | hexanol | is there a way in the asterisk dialplan to get the original context of a line |
| 19:23.43 | hexanol | for example, i have a simple line with context=a |
| 19:23.58 | hexanol | and my context "a" include a context "b" |
| 19:24.29 | WIMPy | ${CONTEXT} should tell you where you are. |
| 19:24.30 | hexanol | I'd like to get the original context of the channel, i.e. "a", from an extension defined in context "b" |
| 19:24.47 | hexanol | well, I don't want to know that I'm in "b" |
| 19:24.56 | WIMPy | You aren't. |
| 19:24.56 | hexanol | I'd like to know that the original lookup was in context "a" |
| 19:25.03 | hexanol | ph |
| 19:25.07 | hexanol | oh |
| 19:25.14 | WIMPy | So I'd expect CONTEXT to tell you a. |
| 19:26.24 | igcewieling | hexanol: have you confirmed when you are in context b via an include => ${CONTEXT} does indeed contain "b" |
| 19:26.25 | hexanol | indeed, my mistake |
| 19:26.41 | hexanol | still, is there another way |
| 19:26.51 | hexanol | for example, if I do a goto to some other context |
| 19:27.01 | igcewieling | hexanol: A goto or gosub would change the context |
| 19:27.01 | hexanol | now, $CONTEXT will really tell me i'm in the new context, right ? |
| 19:27.07 | WIMPy | Then you have to save CONTEXT somwhere. |
| 19:27.21 | hexanol | ok, and there is no way to know the "original" context ? (except from setting in a separate variable) |
| 19:27.30 | hexanol | k |
| 19:27.31 | WIMPy | no |
| 19:28.22 | hexanol | in my case, that would sometime be useful |
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| 19:45.20 | hexanol | in fact, I found my solution, i.e. to add a "setvar" line for my devices |
| 19:45.28 | hexanol | thank you |
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| 20:04.07 | danfromuk | Hi. With regards to ReadExten, the timeouts seem to start counting as soon as the message begins playing. Is there any way to set the timeout to be 'message length' + 2 seconds? |
| 20:06.15 | WIMPy | User Background/WaitExten instead? |
| 20:06.22 | WIMPy | -r |
| 20:09.07 | danfromuk | Whats the -r use for? |
| 20:09.30 | WIMPy | there was an r too much at the end of "Use". |
| 20:09.50 | danfromuk | Ah. Ok. Looks good. Thanks |
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| 20:26.32 | danfromuk | WIMPy: With Background/WaitExten, what happens after an extension is dialled? Does it jump straight there? |
| 20:26.53 | WIMPy | As soon as it has a match. |
| 20:27.09 | danfromuk | Ok. Thanks |
| 20:27.53 | WIMPy | Or after digittimeout if you have overlapping extensions. |
| 20:31.36 | danfromuk | Thanks again. |
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| 20:37.18 | danfromuk | WIMPy: Sorry, final question, in WaitExten, what context does it use? Does it simply take the context specified in the Background command? |
| 20:38.07 | WIMPy | The current one. |
| 20:38.27 | WIMPy | I have no idea what happens if you specify another one in background. |
| 20:39.22 | WIMPy | Maybe you end upt in different context depending on when you dial? |
| 20:39.47 | danfromuk | Ok, Looks like I have to do a bit more work. Thanks again. |
| 20:45.03 | navaismo | Which is the maximum length of the UNIQUEID? |
| 20:45.48 | navaismo | not the size of the varchar field, but the maximum numeric length |
| 20:47.36 | igcewieling | navaismo: that info should be in the cdr and cel schemas included in Asterisk's source, under contrib/ I think. |
| 20:48.14 | igcewieling | navaismo: I believe uniqueid is current unix time the channel started + the microtime the channel started |
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| 20:54.38 | navaismo | igcewieling, now im very confused cdr.h says uniqueid is a char[150] * 150 = 127 (max systemname) + "-" + 10 (epoch timestamp) + "." + 10 (monotonically incrementing integer) + NULL */ |
| 20:54.55 | navaismo | but sql field is varchar(32) |
| 20:55.11 | igcewieling | navaismo: Welcome to Asterisk! |
| 20:56.20 | navaismo | weeeeeird |
| 20:56.28 | navaismo | im going to 32 max size |
| 20:56.30 | navaismo | LOL |
| 20:56.36 | igcewieling | navaismo: Code has been slowly made consistent, but I guess they have not gotten to that code yet |
| 21:03.21 | navaismo | Db designer are so annoying about the varchar |
| 21:03.36 | navaismo | and the DB strcu use it alot XD |
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| 21:55.51 | [TK]D-Fender | ~savemoney |
| 21:55.51 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
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| 21:59.49 | WIMPy | Unfortunatly some professional hardware is built like that. |
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| 22:15.34 | stephen86 | hi guys, anyone have some spare time to help me resolve nat problem? :) |
| 22:15.56 | WIMPy | ~sipnat |
| 22:15.56 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 22:15.59 | WIMPy | ~ask |
| 22:15.59 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 22:16.54 | stephen86 | i have devices connecting through vpn, range 172.16.0.0 eg..... on another side, i have skype trunk |
| 22:17.19 | stephen86 | the problem is, that all packets sent towards sip.skype.com are originated from 172.16.0.9 |
| 22:17.23 | stephen86 | instead of public ip |
| 22:17.35 | stephen86 | any clue |
| 22:18.01 | WIMPy | That's a routing / network configuration thing. |
| 22:18.12 | stephen86 | actually, routing is correct |
| 22:18.21 | stephen86 | but for some reason, |
| 22:18.26 | WIMPy | Sounds like you need NAT. |
| 22:18.34 | jayk | titz |
| 22:18.49 | WIMPy | jayk: Show us |
| 22:19.21 | stephen86 | nat or not nat, source ip in udp packet is always the same :) |
| 22:19.52 | stephen86 | equal to udpbindaddr |
| 22:19.56 | WIMPy | Nt after leaving your tunnel. |
| 22:20.13 | WIMPy | Sure. That's the idea. |
| 22:20.20 | stephen86 | hmm |
| 22:20.32 | stephen86 | i need all my internal devices connected to udpbindaddr |
| 22:20.41 | stephen86 | but still have outgoing skype trunk |
| 22:21.08 | WIMPy | That's just like any LAN/NAT setup. |
| 22:21.36 | stephen86 | so just use iptables and rewrite the source ip if destination is sip.skype.com? |
| 22:22.23 | WIMPy | No, SIP doesn't like being NATted, so you need to configure it. |
| 22:23.01 | stephen86 | how on earth i do that when it always use udpbindaddr, which should be used only for devices to connect to asterisk |
| 22:23.13 | stephen86 | not in outgoing packets towards trunk which are routed through public ip |
| 22:23.16 | stephen86 | :) |
| 22:23.48 | stephen86 | to be clear, asterisk route packet correctly through eth0, but using the source of tun0 |
| 22:23.51 | stephen86 | :D |
| 22:24.31 | WIMPy | If you bind to an address, that's the one that will be used. |
| 22:24.42 | WIMPy | If you need more than one, don;t specify it. |
| 22:25.27 | WIMPy | Assuming that the box in question is both on the tuinnel ald also on a public IP. |
| 22:25.52 | stephen86 | right |
| 22:26.30 | stephen86 | idea was not to allow incoming connection outside the tunnel, |
| 22:26.44 | stephen86 | but allow asterisk to connect to trunk |
| 22:27.12 | stephen86 | so i have device1, device2, device3 -> tunnel -> asterisk -> publicip -> sip.skype.com |
| 22:27.29 | WIMPy | Use iptables. Either to NAT the IP or to block unwanted connections or both. |
| 22:27.46 | WIMPy | Not in any way diffent from any other LAN. |
| 22:27.59 | stephen86 | all clear |
| 22:28.10 | stephen86 | just wanted to ask if there is more elegant solution |
| 22:29.00 | stephen86 | thanks WIMPy |
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| 22:50.24 | orion_noobula | Just made a booboo entering this line inthe terminal (I changed the name in quotes) then realised I was not supposed to. How do I delete the user to start again? adduser asterisk --disabled-password --no-create-home --gecos "Asterisk User" |
| 22:51.56 | WIMPy | userdel or deluser, but that's not really Asterisk related, is it? |
| 22:53.48 | orion_noobula | I was following an asterisk how to . . .first time install sorry, thinking now it is SQL right? |
| 22:54.05 | WIMPy | No. That is unix. |
| 22:54.30 | orion_noobula | :( |
| 22:55.32 | orion_noobula | I can only get better I guess. Just registered this name so you can track my progress if you care. . .thanks for answering either way |
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| 23:42.22 | apb1963 | How can I debug an AGI script? It works ok (prints output) if executed from the command line, but no output when called from *. The * log file shows it is in fact being executed. |
| 23:48.43 | sawgood | well, if it is in the log files ... but not in console output (what) is your verbose level? |
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| 23:51.46 | apb1963 | no i'm saying that * logs the fact that it's executing the script |
| 23:52.05 | apb1963 | the output from the script doesn't appear anywhere |
| 23:52.12 | apb1963 | unless I execute it from the command line |
| 23:52.18 | apb1963 | to verify it's a working script. |
| 23:53.10 | [TK]D-Fender | Show us the script |
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| 23:58.08 | igcewieling | apb1963: the easiest thing to do is run asterisk in the foreground "asterisk -cvvvv" and you'll see the stderr output from your script. Don't do this in production, if you kill the process you've killed Asterisk |