| 06:45.58 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 06:45.58 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.2.1 (2013/01/22), 10.12.1 (2013/01/22), 1.8.20.1 (2013/01/22), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 08:22.50 | Coffeecoco | hi guys |
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| 08:39.48 | WIMPy | When exactely does chan_sip allocate RTP ports? Or the other way round: If you immediately hangup a call would it be possible to skip that stage? |
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| 10:41.34 | wdoekes | WIMPy: immediately hangup after 200? or hangup before the dialog is established? |
| 10:42.40 | WIMPy | That's not a point where I have any influence. |
| 10:42.54 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 10:44.06 | wdoekes | the rtp ports need to be allocated when the sdp data is created.. which is in the initial invite (normally), or in the final ack (if you skip the sdp in the invite) |
| 10:44.07 | *** join/#asterisk ra21vi (~ravi@122.177.147.80) |
| 10:44.56 | ra21vi | if I have sip id, can I know anout the call connected to that SIP client and what IVR options were selected by caller? I hope it is something in AMI. |
| 10:45.04 | WIMPy | That kind of defeats the idea of rejectiong call, doesn;t it? |
| 10:45.18 | wdoekes | asterisk doesn't know which ports it'll get until it gets them, so the allocation happens before the ports are published in the sdp |
| 10:46.12 | ra21vi | hi WIMPy |
| 10:47.51 | wdoekes | I'm not sure where you're going, WIMPy |
| 10:48.55 | WIMPy | I'd like to minimise the impact of calls that won't be accepted anyway. |
| 10:49.39 | wdoekes | ok.. the calls that get rejected, i.e. the ones without any 200 |
| 10:49.40 | WIMPy | The current shitstorms of malicious calls eats up lots of RTP ports that will never be used. That's kind of bad. |
| 10:49.54 | wdoekes | oh.. incoming calls |
| 10:50.24 | WIMPy | Err, should have mentioned that. Yes, incomming. |
| 10:50.26 | wdoekes | that's.. well.. harder |
| 10:50.34 | wdoekes | or.. |
| 10:50.44 | wdoekes | it shouldn't be.. |
| 10:53.55 | wdoekes | with allowguests? |
| 10:54.04 | WIMPy | yes |
| 10:54.13 | wdoekes | why? |
| 10:54.48 | WIMPy | What's the point of using voip without allowing guests? |
| 10:56.10 | wdoekes | my customers just use it as replacement for pstn |
| 10:56.25 | wdoekes | but ok.. what's the rejection criterion, no valid destination? |
| 10:58.21 | WIMPy | yes |
| 11:00.31 | wdoekes | ok.. check_user initialized the rtp. handle_request_invite later checks the destination (for which it needs to know the right context) |
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| 11:04.49 | wdoekes | you could certainly move the rtp_initialize and find_sdp to after the gotdest = get_destination() |
| 11:05.15 | wdoekes | but you might need to add extra is-null checks to portions of code that assume you have initialized rtp |
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| 11:24.26 | *** join/#asterisk thecoda (~kevin@b0fc0bad.bb.sky.com) |
| 11:25.22 | thecoda | So if I call out over DADHI (to my mobile) and my mobile's ringing, but the phone from which I'm calling isn't. What have I overlooked? |
| 11:26.37 | thecoda | Didn't get this dialling out via my SIP provider, so the only thing changed is the argument to Dial() |
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| 11:35.43 | thecoda | anyone? |
| 11:38.26 | jzaw | thecoda: you may have to wait a few mins or even an hour or so |
| 11:42.15 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 11:43.26 | kaldemar | thecoda: start with showing a CLI output for the call. |
| 11:44.17 | kaldemar | thecoda: what kind of a phone are you dialing with? |
| 11:44.36 | thecoda | Gigaset, going via SIP |
| 11:45.04 | thecoda | dialplan is trivial: |
| 11:45.06 | thecoda | exten => _X.,1,Log(NOTICE,making external call to ${EXTEN}) |
| 11:45.07 | thecoda | <PROTECTED> |
| 11:45.27 | thecoda | it works if I use same => n,Dial(SIP/${EXTEN}@directvoip) |
| 11:46.14 | kaldemar | pastebin CLI output and sip debug for the call. |
| 11:46.40 | WIMPy | What's that "#" doing there after the number? |
| 11:47.47 | thecoda | something I found via google :) |
| 11:47.57 | thecoda | I had the same issue without it though |
| 11:48.50 | thecoda | oh! This is new |
| 11:49.19 | thecoda | Drop the # but keep the r, I can hear ringing now - but it keeps on going even after I answer |
| 11:49.54 | WIMPy | I take it it's a POTS line? |
| 11:50.10 | thecoda | yup |
| 11:50.57 | WIMPy | Then the hack that makes the channel answered immediately is probably the only option. |
| 11:51.39 | thecoda | <PROTECTED> |
| 11:51.39 | thecoda | <PROTECTED> |
| 11:51.40 | thecoda | <PROTECTED> |
| 11:51.51 | thecoda | That's all I get, until I hang up |
| 11:52.11 | thecoda | (numbers changed to protect the innocent) |
| 11:52.40 | thecoda | Which hack is this? |
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| 11:53.57 | WIMPy | I'm not familiar with analog stuff, but I'm sure someone else can tell you what options to set. |
| 12:01.49 | jzaw | WIMPy: do you know anything about device status via xmpp ? |
| 12:02.06 | WIMPy | nope |
| 12:02.12 | jzaw | ie when i pick up my phone to dial / its ringing/ or im on a call ... my xmpp status is updated as such |
| 12:02.23 | jzaw | np :) |
| 12:02.44 | jzaw | i used to use mod_client_asterisk.beam from ejabberd |
| 12:02.54 | jzaw | but that doesnt seem to work any more |
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| 13:02.58 | _zoom_ | I'm trying to create the config files for sonagoma a102, I have no wancfg_dahdi my wanpipe-utils version is -3.5.20-0 |
| 13:03.01 | _zoom_ | help plz |
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| 13:18.41 | ra21vi | is it possible to know about the call info to a destination sip id.? |
| 13:18.48 | ra21vi | using any AMI commands> |
| 13:21.14 | WIMPy | ra21vi: What call info? |
| 13:23.24 | ra21vi | if there is inbound call to SIP, is there any AMI command which can give the details, like source no, dialled param etc |
| 13:24.07 | ra21vi | WIMPy: there is one command I am looking - sip show peer <sip_id> .. but the output is so long, it just prints last 25 lines in my elastix tty |
| 13:24.09 | WIMPy | You don't need an action for that. All information is supplied via events. |
| 13:24.48 | ra21vi | WIMPy: I am trying to get the info of connection from A->B in another desktop app. |
| 13:24.50 | WIMPy | how are thos two related? |
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| 13:25.28 | ra21vi | WIMPy: someone from out calls DID no, get transferred to Queue, at last some sip registered user picks call.. |
| 13:26.01 | ra21vi | so I want to see what IVR options were punched by caller. |
| 13:26.04 | WIMPy | What exactely do you want to know? And when/where? |
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| 13:27.14 | ra21vi | I want to know the caller info (number/ or caller ID), and the IVR options he selected to connect to a sip user. |
| 13:28.11 | ra21vi | and these info will be queried by another app that is outside of Asterisk server, so it will connect using AMI |
| 13:28.28 | WIMPy | The caller ID is supplied in an event when the call starts. What your IVR does and how that informations is stored or might be accessible is up to your dialplan. |
| 13:29.52 | ra21vi | ok |
| 13:30.35 | ra21vi | one more question. Do you know how I can pipe output of command in asterisk -r (cli) to vim or some other pager app |
| 13:31.16 | WIMPy | Just like with any other command. |
| 13:31.19 | ra21vi | due to small tty (80x25), I am unable to see the full output of commands in cli, everything scrolls up |
| 13:31.38 | ra21vi | WIMPy: I tried command | vim - , but it didn't work |
| 13:31.43 | WIMPy | And your terminal doesn't allow you to scroll back up? |
| 13:31.52 | ra21vi | WIMPy: no |
| 13:32.00 | WIMPy | Did you use -rx? |
| 13:32.07 | WIMPy | Get a decent terminal then. |
| 13:32.33 | ra21vi | I did only asterisk -r |
| 13:33.16 | WIMPy | That keeps running. So you can't sensibly pipe that. |
| 13:34.15 | ra21vi | oh Now i got you. You meant i should do - asterisk -rx <command> . Is it right> |
| 13:34.28 | WIMPy | exactely |
| 13:34.39 | ra21vi | ok, let me try. Thank you WIMPy |
| 13:34.50 | WIMPy | But changing to a terminal that allows you to scroll back up might be the better idea. |
| 13:36.29 | ra21vi | WIMPy: yes. Since I am using Elastix distro, I am getting default stone-age tty, I don't know how to change that. There is no x-server. |
| 13:37.16 | WIMPy | Oh, the console? You can sroll up there with Shift+PgUp. |
| 13:38.00 | ra21vi | haha. I didn't know that. That works :) Thank you. |
| 13:38.36 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 13:41.15 | gavimobile | folks, im trying to use googletts here is my debug can someone give me a hand? http://pastebin.com/EaMqUT1v |
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| 13:56.46 | gavimobile | updated http://pastebin.com/nwJq8f4c |
| 14:01.46 | thecoda | Any hints for DAHDI not detecting a caller answer in the uk? |
| 14:09.31 | ra21vi | WIMPy: I got the term for what I was doing. Its CTI in telephony world. Computer Telephony Integration. |
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| 14:40.44 | danfromuk | Hi, can GotoIfTime contain an Execif as one of the applications? |
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| 15:31.44 | ghost75 | if i use gosubif and jump to other context, the exten and prio has also to be specified, right? |
| 15:33.07 | [TK]D-Fender | ghost75: Context is the LEAST important of those parameters |
| 15:33.37 | ghost75 | yes but context requires exten? |
| 15:34.41 | [TK]D-Fender | ghost75: clearly. You should see that right in the instructions |
| 15:36.23 | leifmadsen | ghost75: you can either specify priority, exten and priority, or context, exten and priority |
| 15:36.55 | leifmadsen | context -> exten -> priority (requires) |
| 15:41.27 | ghost75 | my dialplan gets so long on the incoming part oO |
| 15:41.46 | leifmadsen | then break it into subroutines |
| 15:42.12 | ghost75 | different contexts? |
| 15:42.17 | leifmadsen | GoSub() |
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| 15:43.02 | *** join/#asterisk astra05 (~astra05@94.23.143.195) |
| 15:43.06 | astra05 | greetings |
| 15:45.00 | ghost75 | and use there _X. ? |
| 15:45.18 | leifmadsen | ghost75: see asteriskdocs.org in the chapters on dialplan for information on using GoSub() |
| 15:46.57 | [TK]D-Fender | ghost75: When do you not want to go somer specific? Why shuould you be needing a pattern match? |
| 15:47.17 | [TK]D-Fender | somewhere* |
| 15:47.38 | *** part/#asterisk sparrowjs (~Jon@c-76-25-87-128.hsd1.co.comcast.net) |
| 15:47.43 | ghost75 | the other context could be used for multiple exten then |
| 15:48.21 | [TK]D-Fender | ghost75: That's as good as only having ONE, not "multiple" |
| 15:50.46 | *** join/#asterisk thecoda (~kevin@b0fc0bad.bb.sky.com) |
| 15:51.02 | leifmadsen | that's the gist yes; reusable dialplan |
| 15:52.20 | thecoda | anybody got any experience with DAHDI over POTS in the uk? |
| 15:52.38 | thecoda | I'm calling out, but the answer isn't being detected |
| 15:53.13 | thecoda | dialling out over a sip trunk works |
| 15:53.24 | thecoda | but not dahdi :( |
| 15:55.55 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:55.55 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:59.10 | gavimobile | I am trying to get the googletts.agi to work, what can I provide to help debug why this agi doesn't work? this is what I have http://pastebin.com/AMKN8UdY |
| 15:59.36 | gavimobile | here is my dialplan http://pastebin.com/r6huKETU |
| 16:03.48 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.181.243) |
| 16:05.40 | [TK]D-Fender | Check that your AGI is actually in the right place with the Asterisk user being the owner, and the contents are right,e tc |
| 16:06.54 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@190.166.31.8) |
| 16:07.08 | din3sh | @leifmadsen: sorry to bother, I am using Gareth's Presence patch for Cisco Phones [https://issues.asterisk.org/jira/browse/ASTERISK-13145], I am getting loads of Publish authentication requests. Is there a fix for this? |
| 16:07.29 | leifmadsen | din3sh: no idea, check the issues tracker. I don't know anything about that issue. |
| 16:09.15 | din3sh | Ok thnks |
| 16:09.49 | gavimobile | [TK]D-Fender: I believe the agi is in the correct place /var/lib/asterisk/agi-bin/, I don't have an asterisk.conf file. I've never played with agi scripts yet. as for my ownership -rwxrwxrwx 1 root root 8468 Jan 17 23:34 googletts.agi |
| 16:10.40 | [TK]D-Fender | gavimobile: If you aren't running your * as root then that is wrong |
| 16:10.46 | leifmadsen | I suggest starting with a simpler agi to make sure your agi stuff is working fine |
| 16:10.55 | gavimobile | [TK]D-Fender: ok ill change it to my asteriskuser |
| 16:10.56 | leifmadsen | otherwise, there is nothing in your output that shows anything of use for debugging |
| 16:10.58 | [TK]D-Fender | gavimobile: If you are running * as root, everything is wrong |
| 16:11.20 | gavimobile | [TK]D-Fender: and what about chmod |
| 16:11.28 | [TK]D-Fender | gavimobile: chown <- |
| 16:11.35 | gavimobile | [TK]D-Fender: I used chown already |
| 16:11.43 | gavimobile | -rwxrwxrwx 1 asteriskpbx asteriskpbx 8468 Jan 17 23:34 googletts.agi |
| 16:12.04 | [TK]D-Fender | I'm pretty sure you don't need EVERYONE to have access to it |
| 16:12.15 | gavimobile | so 755? |
| 16:12.27 | *** join/#asterisk srp_ (~sandeep@223.238.152.55) |
| 16:12.39 | gavimobile | I mean 775 |
| 16:13.03 | gavimobile | no 755 |
| 16:13.15 | [TK]D-Fender | better |
| 16:13.32 | gavimobile | regardless it still doesn't work even with 777 |
| 16:14.13 | [TK]D-Fender | And that was only a small part of what I said |
| 16:14.28 | gavimobile | [TK]D-Fender: ill go back and have another look at what you said |
| 16:15.04 | astra05 | man, asterisk is uber complex |
| 16:15.19 | gavimobile | Check that your AGI is actually in the right place with the Asterisk user being the owner, and the contents are right,e tc so the agi is in the right place. asterisk user is now the owner of the files and what do you mean about content being right? |
| 16:15.24 | leifmadsen | with power comes complexity :) |
| 16:15.58 | [TK]D-Fender | gavimobile: How do i know what is in that file is a proper AGI? |
| 16:16.32 | gavimobile | [TK]D-Fender: I didn't expect the channel to support it. its from http://zaf.github.com/asterisk-googletts/ |
| 16:16.59 | [TK]D-Fender | gavimobile: I'm not seeing YOUR file |
| 16:17.08 | gavimobile | [TK]D-Fender: just a sec |
| 16:19.30 | gavimobile | [TK]D-Fender: here is my googletts.agi file. is this what you wanted or did I misunderstand you http://pastebin.com/Dza0ptPZ? |
| 16:19.36 | gavimobile | http://pastebin.com/Dza0ptPZ |
| 16:19.55 | gavimobile | the second link please |
| 16:20.50 | [TK]D-Fender | gavimobile: #!/usr/bin/env perl <--- and is that right? |
| 16:21.04 | [TK]D-Fender | gavimobile: Is your perl executable there? |
| 16:21.38 | gavimobile | [TK]D-Fender: unfortunetly, I have no idea |
| 16:21.42 | [TK]D-Fender | gavimobile: Have you tried executing it from CLI direct? |
| 16:21.55 | gavimobile | [TK]D-Fender: nope |
| 16:21.56 | [TK]D-Fender | gavimobile: Have you considered looking? That usually helps the "knowing" part |
| 16:25.20 | astra05 | so i am fairly newb and i intend not to waste anyones time, but what configuration would be the issue if asterisk is registering the sip service, but there no dial tone and calls are not atoned for |
| 16:25.29 | gavimobile | [TK]D-Fender: thanks for pointing out these questions. I now have a direction |
| 16:25.30 | astra05 | maybe tcp/ip config issue me thinks |
| 16:26.12 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 16:26.13 | [TK]D-Fender | astra05: What "sip service" are you talking about with regards to this registration? |
| 16:26.37 | [TK]D-Fender | astra05: And what is this that you are listening for "dial tone" on exactly? |
| 16:26.42 | astra05 | registering with the voip/sip provider, the company that is handling the tcp/ip to regular phone convesion, in this case, callcentric |
| 16:27.28 | astra05 | [TK]D-Fender: as in calling the number i've paid for; though, i suppose with asterisk you would define you own tone |
| 16:27.42 | [TK]D-Fender | [11:26][TK]D-Fenderastra05: And what is this that you are listening for "dial tone" on exactly? |
| 16:27.51 | astra05 | calling from my cell phone |
| 16:28.13 | [TK]D-Fender | astra05: As in ? |
| 16:28.27 | astra05 | [TK]D-Fender: excuse my daftness, but i fail to understand you last question |
| 16:31.10 | ghost75 | next release will be called uber * ? |
| 16:31.27 | astra05 | [TK]D-Fender: scenario is this: pay for a sip service with a registered 10 digit phone number. have asterisks handle the SIP and connections from the softphones, use an extension per softphone |
| 16:32.42 | [TK]D-Fender | astra05: You said you didn't head dial tone. That is something you hear on a phone. As in something physical you hold to your head. I asked precisely what phone you were using and expected to hear dial tone relating to Asterisk on. |
| 16:33.00 | [TK]D-Fender | ghost75: Feel free to start your petition now. |
| 16:33.24 | astra05 | [TK]D-Fender: my wirelss phone dialing the SIP i've obtained; i simply dial it as you would any other phone |
| 16:34.09 | [TK]D-Fender | astra05: What do you mean "dialing the SIP"? This term does not make sense. And "wireless" is too vague. |
| 16:34.55 | WIMPy | ghost75: Isn't if fynny how '*' becomes less when you increase it by one? |
| 16:34.58 | astra05 | [TK]D-Fender: my wireless, cellular phone from a wireless courier, like ATT or TMobile. I am using a samsung device to dial the phone number i've purchased |
| 16:36.18 | [TK]D-Fender | astra05: Why is Asterisk supposed to give you dialtone then? If I have a PBX with ana analog phone directly connected to it and pick it up, THEN I expect to hear dialtone... BEFORE I DIAL. That is what dialtone is. BEFORE you dial. |
| 16:36.22 | ghost75 | WIMPy: then its in a cloud? |
| 16:36.28 | [TK]D-Fender | astra05: I suspect you have a lot of terms mixed up. |
| 16:36.42 | [TK]D-Fender | astra05: So please rephrase your question |
| 16:36.45 | astra05 | [TK]D-Fender: probably, i've done a voice project before |
| 16:36.58 | astra05 | *never |
| 16:37.16 | astra05 | [TK]D-Fender: i really appreciate you trying to help |
| 16:37.19 | [TK]D-Fender | astra05: If you've ever owned or used a phone before you should know what "dialtone" is |
| 16:37.37 | [TK]D-Fender | Hear tone. dial. "dial tone" |
| 16:37.46 | astra05 | [TK]D-Fender: true, i supposed it would be artificially generated by my SIP provider |
| 16:38.02 | [TK]D-Fender | astra05: So please rephrase the chain of what you're doing, and what is happening, and what is not happening that should |
| 16:38.46 | [TK]D-Fender | astra05: No, your cell doesn't have "dialtone" in the first place. It si a digital device that sends the complete # you dial directly to your cell phone company |
| 16:39.08 | [TK]D-Fender | astra05: You are still using the wrong term for the SOUND you are tying to describe |
| 16:39.53 | [TK]D-Fender | astra05: Dialtone is BEFORE you dial. If you haven't dialed anything on your cell then where does Asterisk ever come in? Your cell network hasn't even recieved the call from you. |
| 16:40.00 | astra05 | [TK]D-Fender: from scratch. I bought a SIP service that is bound to a 10 digit +1 number. As such, I've configured sip.conf to register and "bind" to the providers servers. I have confirmed via the asterisk console and with my provider than the asterisk box is connecting on standard SIP TCP. |
| 16:40.24 | astra05 | [TK]D-Fender: my exterme confusion; the RINGING sound that typically accompanies a succesful dial of a real number |
| 16:40.31 | [TK]D-Fender | astra05: BETTER |
| 16:40.44 | [TK]D-Fender | astra05: So is the call actually arriving to your * server? |
| 16:41.26 | astra05 | facepalms |
| 16:41.28 | astra05 | nothing in the log |
| 16:41.44 | astra05 | yet, it registerd on their servers, so obviously a communication issue |
| 16:42.52 | [TK]D-Fender | "sip set debug on" , "sip reload" <- pastebin both where we can see your actual registration atempt |
| 16:48.19 | astra05 | [TK]D-Fender: http://pastie.org/private/2rxwzic4be8mjffjvpfia |
| 16:48.31 | [TK]D-Fender | astra05: New pastebin WITHOUT masking |
| 16:49.19 | astra05 | [TK]D-Fender: do you really need the CCID ? |
| 16:49.19 | [TK]D-Fender | astra05: And try an incoming call there too |
| 16:50.27 | *** join/#asterisk TimeRider (~steve@host217-36-208-184.in-addr.btopenworld.com) |
| 16:50.36 | *** join/#asterisk KNERD (~KNERD@cpe-68-201-123-108.gt.res.rr.com) |
| 16:52.10 | ghost75 | is there a way to automatically start mixmonitor in dialplan only when the call was picked up? |
| 16:52.58 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 16:53.18 | [TK]D-Fender | ghost75: "core show application mixmonitor" |
| 16:54.08 | astra05 | [TK]D-Fender: http://pastie.org/private/htvij5r2mkavpai1zaao8q de-sensitized |
| 16:54.46 | ghost75 | <PROTECTED> |
| 16:58.40 | [TK]D-Fender | astra05: Where are you relative to your * server? |
| 16:59.00 | [TK]D-Fender | ghost75: Does that not sound like exactly what you asked for? |
| 16:59.41 | ghost75 | but it will still create a new file as soon its started right? |
| 16:59.44 | [TK]D-Fender | astra05: Did you try a call before posting that pastebin? |
| 16:59.58 | [TK]D-Fender | ghost75: Did you try it? |
| 17:00.05 | ghost75 | no :p |
| 17:02.15 | astra05 | neg |
| 17:03.50 | thecoda | Anyone here a guru on answer detection over dahdi/pots? |
| 17:07.20 | astra05 | [TK]D-Fender: are you looking for data from the call log as well? |
| 17:08.34 | [TK]D-Fender | astra05: * CLI with SIP debug. Place a call. |
| 17:11.30 | astra05 | [TK]D-Fender: i have, it is going to the provder vm directly, not touching the server as far as i can tell from the logs |
| 17:11.59 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 17:16.29 | [TK]D-Fender | astra05: [11:58][TK]D-Fenderastra05: Where are you relative to your * server? |
| 17:17.04 | [TK]D-Fender | astra05: <--- SIP read from UDP:204.11.192.159:5060 ---> <---- this also doesn't look like TCP to me. |
| 17:20.32 | astra05 | [TK]D-Fender: or UDP, IP datagrams in general. this is a home setup, running on my firewall device in order to remove any additional machine dependencies |
| 17:21.07 | astra05 | [TK]D-Fender: this is one of the few projects i've done where i have zero idea of most what needs to be configured. i reading through documention, I think i just need to a cheive a higher understanding |
| 17:21.11 | [TK]D-Fender | asSo your * server is also your router? |
| 17:21.20 | astra05 | [TK]D-Fender: aye |
| 17:21.51 | *** join/#asterisk phunguy (santas@will.one.day.hack-the-pla.net) |
| 17:21.55 | [TK]D-Fender | "iptables --list' , "ifconfig" |
| 17:21.57 | [TK]D-Fender | ^ |
| 17:22.08 | astra05 | [TK]D-Fender: you are going to hate me, but this is running on FBSD |
| 17:23.37 | [TK]D-Fender | equivalents..... |
| 17:27.25 | astra05 | <PROTECTED> |
| 17:39.04 | *** join/#asterisk mihamina (~mihamina@252.217.74.41-ip-dyn.orange.mg) |
| 17:39.37 | *** join/#asterisk davlefou (~david@unaffiliated/davlefou) |
| 17:40.59 | astra05 | [TK]D-Fender: i really appreciate all of your help |
| 17:41.28 | astra05 | [TK]D-Fender: i am going to start from scratch again, i am thinking the problem now is connecting to the SIP provider, thank you so much |
| 17:42.13 | [TK]D-Fender | It changed somewhere along the way? |
| 17:45.11 | astra05 | i just have zero idea of what i am doing |
| 17:45.13 | astra05 | tbh |
| 17:45.27 | astra05 | i thought it would be simpler, though i was wrong |
| 17:45.56 | dpilon | if it was easy..everyone would do it |
| 17:46.04 | astra05 | true enough |
| 17:46.06 | astra05 | just a last question |
| 17:46.27 | astra05 | so i bought a sip service for POTS 10 digit phone number number |
| 17:46.46 | *** join/#asterisk TimeRider (~steve@151.227.126.40) |
| 17:47.07 | astra05 | so you will simply need to configure sip.conf and extensions.conf to A) Have sip.conf connect to the sip provider and define extensions and B) have extensions.conf define how to hanlde VM, and softphone/IP clients? |
| 17:47.42 | [TK]D-Fender | softphone/IP clients <- also sip.conf |
| 17:47.53 | [TK]D-Fender | extensions.conf is for how to process CALLS from all of these |
| 17:48.10 | [TK]D-Fender | You define the devices & services in sip.conf and process the actual calls in extensions.conf |
| 17:48.12 | astra05 | so extensions would handle voicemail routing and busy signals? |
| 17:48.32 | [TK]D-Fender | "what to do when the call comes in" <- all of that |
| 17:48.41 | astra05 | gotcha |
| 17:48.51 | astra05 | so if i wanted to test outgoing calls, that would be sip.conf, no? |
| 17:51.16 | [TK]D-Fender | only the part of telling * how to call your provider |
| 17:51.50 | [TK]D-Fender | The part where * even gets the request from your phone is extensions.conf processing to choose what to do with the number you dialed which in your case is expected to dial out. |
| 17:51.53 | astra05 | [TK]D-Fender: i almost tempted to hire a freenlancer to help with initial configuration |
| 17:52.08 | astra05 | that explains a lot, thanks [TK]D-Fender |
| 17:52.19 | [TK]D-Fender | ~book |
| 17:52.19 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:52.21 | [TK]D-Fender | ^^^ |
| 17:52.23 | dpilon | Fender is available :) |
| 17:57.51 | jpsharp | I think most people here are available. |
| 17:58.10 | dpilon | true |
| 18:05.04 | *** part/#asterisk mihamina (~mihamina@252.217.74.41-ip-dyn.orange.mg) |
| 18:07.37 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 18:23.24 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:23.24 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:30.35 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 18:53.35 | astra05 | hip hip hooray for [TK]D-Fender |
| 18:56.40 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 19:11.19 | [TK]D-Fender | I didn't really do much, but you're welcome |
| 19:16.22 | *** join/#asterisk k611 (~K610@cable-78.29.241.186.coditel.net) |
| 19:23.15 | *** join/#asterisk mjordan (~mjordan@216.186.152.216) |
| 19:23.15 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 19:34.26 | astra05 | throws book at * |
| 19:34.37 | astra05 | back to step 1 yet again |
| 20:54.16 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 21:14.40 | ChannelZ | yawns |
| 21:20.47 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 21:29.10 | *** join/#asterisk mmmamiens (6dbe125a@gateway/web/freenode/ip.109.190.18.90) |
| 21:29.23 | mmmamiens | Bonjour/hello |
| 21:30.02 | ChannelZ | yes both |
| 21:31.40 | mmmamiens | I search to use say.conf for Voicemailmain, is possible? for grouping number to two in saying 03 22 22 22 22 in place of 0 3 2 2 2 2 2 2 2 2 |
| 21:41.47 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 21:43.00 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 21:43.37 | danfromuk | In call queues, is there an option for the caller to be able to press 1 if they want to exit the queue and leave a message? |
| 21:44.26 | danfromuk | Currently, i time out of the queue and go to an ivr saying press 1 to leave a message or nothing to continue to hold. But the caller loses their position. |
| 21:45.11 | [TK]D-Fender | danfromuk: Yes |
| 21:47.01 | mmmamiens | I search to use say.conf for Voicemailmain, is possible? for grouping number to two in saying 03 22 22 22 22 in place of 0 3 2 2 2 2 2 2 2 2 |
| 21:47.16 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 21:47.56 | [TK]D-Fender | mmmamiens: No, because it is saying the "digits" vs the "number" |
| 21:48.21 | danfromuk | [TK]D-Fender: do you recall what the option is called? I can't seem to find it. |
| 21:48.28 | *** join/#asterisk Wildblue (~guy@c-69-140-154-17.hsd1.md.comcast.net) |
| 21:48.30 | [TK]D-Fender | mmmamiens: "core show application saydigits" vs "core show application saynumber". |
| 21:48.36 | *** part/#asterisk Wildblue (~guy@c-69-140-154-17.hsd1.md.comcast.net) |
| 21:48.40 | mmmamiens | I have to change app_voicemail.c? |
| 21:48.41 | [TK]D-Fender | danfromuk: It's in the sample config.... |
| 21:48.45 | [TK]D-Fender | mmmamiens: Yes |
| 21:48.49 | mmmamiens | :( |
| 21:48.52 | danfromuk | I thought so but i can't locate it. |
| 21:50.16 | [TK]D-Fender | danfromuk: http://svn.digium.com/svn/asterisk/branches/11/configs/queues.conf.sample |
| 21:51.50 | danfromuk | I have that file but can't see an option to allow the caller to leave the queue |
| 21:52.20 | mmmamiens | You do not have an example, I'm not very undersheet C |
| 21:55.55 | [TK]D-Fender | danfromuk: It is right there... |
| 21:56.00 | [TK]D-Fender | danfromuk: Top 1/3rd .... |
| 21:56.43 | [TK]D-Fender | danfromuk: The wording is very obvious |
| 21:58.53 | danfromuk | I'm sorry. I really dont see it. |
| 22:00.40 | [TK]D-Fender | "if the user types a SINGLE digit extension while they are in the queue, they will be taken out of the queue and sent to that extension in this context" |
| 22:02.10 | danfromuk | Ah. Thank you. I must be tired |
| 22:02.14 | danfromuk | Sorry about that. |
| 22:16.17 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 22:24.56 | *** join/#asterisk KNERD (~KNERD@cpe-68-201-123-108.gt.res.rr.com) |
| 22:47.25 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-aibnccaczdnpirfh) |
| 22:47.25 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 22:47.37 | leifmadsen | mjordan: phai |
| 22:47.42 | leifmadsen | errr... ohai |
| 22:48.04 | leifmadsen | and thanks for the answer on the named groups; was pretty sure that was the solution in later versions |
| 22:48.15 | leifmadsen | thought it had been bumped to 128 for some reason, but appears as though it was not the case |
| 22:50.04 | file | don't stop believin' |
| 22:51.27 | [TK]D-Fender | hang on to that feelin' |
| 22:55.23 | leifmadsen | :) |
| 22:55.26 | leifmadsen | runs away |
| 23:01.21 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:10.54 | *** join/#asterisk k1ng (~k1ng@unaffiliated/k1ng) |
| 23:15.59 | *** join/#asterisk |KNERD| (~KNERD@cpe-68-201-123-108.gt.res.rr.com) |
| 23:23.44 | *** join/#asterisk zhando (~user@c-67-161-122-77.hsd1.wa.comcast.net) |
| 23:26.51 | zhando | can anyone comment on asterisk internals these days? I've been using 1.4.17 forever as a household pbx and i'd like to upgrade to asterisk 11. just curious if 11 is that much better internally.. |
| 23:27.36 | [TK]D-Fender | Yes |
| 23:27.53 | zhando | [TK]D-Fender: in what ways? |
| 23:28.39 | [TK]D-Fender | Tons of performance fixes, preventing deadlocks, the massive number of critical security fixes you're missing , timing issues, etc. |
| 23:29.31 | [TK]D-Fender | Your version is over 5 YEARS old |
| 23:29.51 | [TK]D-Fender | And has a giant target on its ass so to speak |
| 23:33.12 | zhando | [TK]D-Fender: oh yeah my system is a dino a 2.4ghz Pentium 4 that I want to upgrade to an athlon 64 x2.. my server is behind the typical home firewall but I suppose the SIP ports are exposed in some way to the outside world.. Are they that vulnerable to certain attacks? |
| 23:33.34 | [TK]D-Fender | Only a ton of them |
| 23:33.45 | [TK]D-Fender | And well documented |
| 23:34.13 | zhando | And 11 shuts all that down? |
| 23:34.21 | [TK]D-Fender | All the ones they found so far |
| 23:35.41 | zhando | interesting.. thx.. |
| 23:35.56 | *** join/#asterisk tamiel (~tamiel@c-67-169-76-114.hsd1.ca.comcast.net) |
| 23:37.43 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 23:44.33 | zhando | [TK]D-Fender: Is this a fairly complete listing of * vulnerabilities over the years: http://www.cvedetails.com/product/3085/Digium-Asterisk.html?vendor_id=1802 |
| 23:46.13 | *** part/#asterisk tompaw (~tompaw@tompaw.xxx) |
| 23:52.40 | [TK]D-Fender | no idea |
| 23:53.52 | [TK]D-Fender | But taken at face value, that's a lot since 2007 |
| 23:57.49 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-231-146.ph.ph.cox.net) |