| 00:08.24 | Penguin | Not with SIP. |
| 00:12.55 | ChannelZ | Not sure what's up. I get a flashing orange light, which the admin manual says "denotes an undefined problem". Yet looking at the SIP debug the phone subscribes, Asterisk sends a dialog-info NOTIFY, and the phone says OK. |
| 00:14.32 | Penguin | I have old phones without lights. |
| 00:14.55 | Penguin | The BLF is on-screen rather than on lighted buttons. |
| 00:16.56 | ChannelZ | oo, I might have just fixed it. |
| 00:20.41 | ChannelZ | OK got it. Jesus Cisco's docs are horrible. |
| 00:21.01 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.123) |
| 00:24.45 | Freeaqingme_ | ChannelZ you should try their sales dept, they're usually eager to help ;) |
| 00:33.09 | ChannelZ | They have this syntax for the 'Extended Function' of a line key but the definition of the syntax makes no real sense. |
| 00:33.40 | Penguin | Don't act so surprised. |
| 00:37.28 | ChannelZ | Not, really. Linksys' and Sipura before them all sucked too |
| 00:38.31 | WIMPy | Show me a SIP phone that doesn't suck. |
| 00:39.45 | Penguin | Some are worse than others. |
| 00:40.01 | ChannelZ | Well I'm not unhappy with the phone(s), just the shit documentation |
| 00:40.19 | WIMPy | Penguin: Definitely |
| 00:46.18 | *** join/#asterisk andrewyager (~andrewyag@2-104-141-114.static-dsl.realworld.net.au) |
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| 01:00.27 | igcewieling | ChannelZ: When you said "Cisco Phone" I assume it was a Cisco phone, rather than a Linksys phone. |
| 01:01.33 | ChannelZ | well in this case it is |
| 01:01.47 | ChannelZ | 504G. I have some Linksys 942s as well |
| 01:03.59 | ChannelZ | New problem, trying to do BLF remotely. No idea how to make it identify its self as the correct account in the subscribes, it seems to use the first line key account. |
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| 02:32.18 | j4jackj | Dealing with an arsehat called kloeri is not fun. |
| 02:34.37 | phix | who? |
| 02:35.17 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 02:35.33 | Freeaqingme_ | j4jackj, imho he's a very valued freenode op |
| 02:36.19 | j4jackj | He's also a very intimidating one. I'm trying to get him off the list. |
| 02:36.40 | Freeaqingme_ | whatever |
| 02:36.41 | Freeaqingme_ | < afk |
| 02:39.30 | phix | What list? |
| 02:40.32 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 02:40.41 | igcewieling | Annoying an IRCop is like poking a bear. Best to avoid it. |
| 02:42.58 | j4jackj | Being an IRCop and annoying an Aspie is like poking a shark. Best stay away from me with the length of 10 5foot bargepoles. |
| 02:43.33 | igcewieling | When a bear and a shark tangle everyone loses. |
| 02:43.41 | igcewieling | bets to be avoided at all costs. |
| 02:43.42 | phix | Although being an IRCop, or a bear for that matter, doesn't make you more important than everyone else, the same level of courtesy is expected. |
| 02:44.28 | j4jackj | They should expect the same courtesy I expect from them. |
| 02:44.35 | j4jackj | And I expect rather highly of them. |
| 02:44.37 | igcewieling | phix: most any creature in the forest would disagree. |
| 02:45.03 | j4jackj | It's very unequal. I'll ban *!*@freenode/staff from my channel, ##wrpl as a protest. |
| 02:46.42 | phix | Yes and I don't agree with it |
| 02:47.19 | phix | igcewieling: then the creatures of the forest should unite and make the bears respect them :) |
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| 03:18.22 | ChannelZ | I love a good bitch fight |
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| 04:00.34 | j4jackj | ChannelZ: do you want to see my Haiku Review? |
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| 04:59.42 | hjf | argh |
| 04:59.49 | hjf | i tried using asterisk gui on freebsd |
| 04:59.55 | hjf | but it's complaining about DAHDI |
| 05:00.03 | hjf | i don't even have any dahdi devices in my system |
| 05:00.27 | hjf | i even installed dahdi as suggested by digium and there's no dahdi_genconf in freebsd at least |
| 05:02.54 | ChannelZ | well there's not much to gen if you have no hardware |
| 05:02.58 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.114) |
| 05:04.04 | Penguin | You won't be happy with asterisk gui. |
| 05:04.27 | [TK]D-Fender | It's been dead for years and is a waste of time |
| 05:07.30 | hjf | meh |
| 05:07.49 | hjf | i just wanted to see if there was a way to see peers or channels without having to ssh into the gui |
| 05:07.52 | hjf | into the cli* |
| 05:08.03 | Penguin | There is. |
| 05:08.10 | Penguin | AMI <--- |
| 05:11.14 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 05:23.04 | ChannelZ | ARGH now this shitting 504g won't even *try* to subscribe |
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| 06:25.58 | phix | ChannelZ: 504g? cisco? |
| 06:26.12 | phix | linksys? |
| 06:26.24 | phix | I prefer snoms btw |
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| 07:37.39 | ChannelZ | phix: Crisco |
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| 09:54.00 | kchehab | hi my Master.csv getting too big ,i there a way to make a log rotation or delete each line after inserting into mysql ? |
| 09:55.05 | j4jackj | Oh? |
| 09:56.29 | *** join/#asterisk kresp0 (~kresp0@109.Red-79-144-71.dynamicIP.rima-tde.net) |
| 10:07.39 | kaldemar | kchehab: use logrotate. you'll find a sample logrotate file in the source package under contrib/scripts/. |
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| 12:15.13 | cusco | hi |
| 12:15.25 | cusco | cdr variable peeraccount |
| 12:15.42 | cusco | https://wiki.asterisk.org/wiki/display/AST/CDR+Fields - states peeraccount: the account code of the bridged channel |
| 12:15.47 | cusco | supported in some backends |
| 12:15.59 | cusco | I'm using adaptative odbc |
| 12:16.06 | cusco | it doens't get written |
| 12:16.10 | cusco | comes always null |
| 12:21.22 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 12:23.13 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 12:23.27 | hrolf | Hi #asterisk, we are using Asterisk as a SIP client with Avaya IP office. |
| 12:23.47 | hrolf | They have provided us with six extensions, we are receiving calls on the extensions. |
| 12:24.01 | hrolf | Problem is that all the calls are received on one extension at Asterisk end. |
| 12:24.29 | hrolf | I put call-limit=1 on each extension, but now only 1 call can be received at at a time. |
| 12:24.46 | hrolf | The extensions are 302,303,304,305,306,307 |
| 12:24.54 | hrolf | but all calls are received on 302 |
| 12:25.35 | hrolf | While, on Avaya end it shows that the calls are going separately to each extension |
| 12:26.01 | hrolf | but on Asterisk we receive on SIP/302<something appended here> |
| 12:26.28 | hrolf | <PROTECTED> |
| 12:26.46 | hrolf | ^ in above line, call is received on extension 307, but Asterisk received it at SIP/302-00000015 |
| 12:27.03 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:27.06 | hrolf | I have registered all the 6 extensions |
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| 12:54.07 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 12:54.20 | hrolf | Hi #asterisk, we are using Asterisk as a SIP client with Avaya IP office. |
| 12:54.25 | hrolf | They have provided us with six extensions, we are receiving calls on the extensions. |
| 12:54.32 | hrolf | Problem is that all the calls are received on one extension at Asterisk end. |
| 12:54.38 | hrolf | I put call-limit=1 on each extension, but now only 1 call can be received at at a time. |
| 12:54.46 | hrolf | but all calls are received on 302 |
| 12:54.52 | hrolf | <PROTECTED> |
| 12:54.58 | hrolf | ^ in above line, call is received on extension 307, but Asterisk received it at SIP/302-00000015 |
| 12:56.21 | Greenlight | Lets see the dialplan which is getting executed |
| 12:56.37 | [TK]D-Fender | "type=friend", "insecure=no" <- |
| 12:56.59 | [TK]D-Fender | Greenlight: this has nothing to do with dialplan, and you see it right there already |
| 12:57.16 | [TK]D-Fender | Greenlight: it's not matching the sip.conf entry that they feel it should |
| 12:57.38 | hrolf | The dialplan is like exten => _30X,1,AGI(...) |
| 12:57.49 | hrolf | And we are dialing 300 hunt group |
| 12:57.51 | [TK]D-Fender | hrolf: Fix your entires as I advised |
| 12:58.02 | [TK]D-Fender | entries* |
| 12:58.13 | hrolf | [TK]D-Fender: Which entries? (I got disconnected, can you please repost?) |
| 12:58.26 | [TK]D-Fender | [08:56][TK]D-Fender"type=friend", "insecure=no" <- |
| 12:58.27 | Greenlight | [01:56pm] <[TK]D-Fender> "type=friend", "insecure=no" <- |
| 12:58.29 | [TK]D-Fender | two MINUTES ago. |
| 12:58.33 | [TK]D-Fender | You were not disconnected |
| 12:58.52 | hrolf | [TK]D-Fender: Okay, thanks. Let me try it out. |
| 13:00.55 | roderickm | hrolf: you've likely configured the sip friends to match based on ip address/port (which is the same every time coming from your avaya system). If you set "insecure" as Fender advised without explanation, chan_sip will match based on username/password, which are presumably different for each of your six incoming extensions. |
| 13:00.58 | Greenlight | <hrolf> -- Executing [307@access:1] AGI("SIP/302-00000015", <-- Although this does seem odd. It's hitting exten 307, but calling an AGI on 302 |
| 13:02.22 | roderickm | Greenlight: it may seem counterintuitive, but that's the behavior expected when matching peers based on ip address: the extension number need not match if the ip address does. |
| 13:03.00 | [TK]D-Fender | 307 is what was DIALED |
| 13:03.05 | [TK]D-Fender | 302 is the peer it MATCHED |
| 13:03.20 | [TK]D-Fender | Terms are getting mixed up here |
| 13:03.53 | Greenlight | Ahh yea |
| 13:04.15 | [TK]D-Fender | 302 = who, 307 = what. |
| 13:09.11 | hrolf | [TK]D-Fender: Now, it is sending SIP/2.0 401 Unauthorized |
| 13:09.55 | [TK]D-Fender | which "it"? |
| 13:10.03 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 13:10.04 | hrolf | [TK]D-Fender: Asterisk. |
| 13:10.20 | [TK]D-Fender | hrolf: And Avaya is not accepting the challenge? |
| 13:10.47 | Katty | good morning |
| 13:10.53 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:11.04 | hrolf | [TK]D-Fender: http://pastebin.com/j8pDCQXG |
| 13:11.36 | [TK]D-Fender | guess not |
| 13:11.43 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 13:11.59 | hrolf | [TK]D-Fender: sip.conf http://pastebin.com/WvPhCbPc |
| 13:12.13 | [TK]D-Fender | hrolf: Avaya doesn't like being challenged when sending out calls |
| 13:12.32 | hrolf | [TK]D-Fender: What do I do then? |
| 13:12.49 | cusco | hi ... I seem to remember to have installed oslec in the past but can't remember... found a topic on asterisk forum stating I need to copy oslec from the linux tree ?? to the dadhdi-linux tree ? |
| 13:13.04 | [TK]D-Fender | hrolf: a single * instance is not going to be able to make them amtch separate peers |
| 13:13.05 | cusco | so I need the linux kernel sources? |
| 13:13.27 | [TK]D-Fender | hrolf: So see if you can live with that and ID them by breking off the To: header in the dialplan |
| 13:14.40 | cusco | "Tuesday, June 16 2009: Oslec is now part of the Linux kernel (linux-2.6.28/drivers/staging/echo). Thanks to Tzafrir, and Greg Kroah-Hartman for making this happen. Wow, something" |
| 13:14.49 | hrolf | [TK]D-Fender: The To: field is '300' which is a hunt group, how do I identify then? |
| 13:16.13 | *** join/#asterisk Draecos (~Draecos@124.169.70.34) |
| 13:16.50 | [TK]D-Fender | hrolf: parse out the To: header in the dialplan. There is a function for this. go look for it. |
| 13:17.47 | hrolf | [TK]D-Fender: The To: header is always sip:300 (it is a hunt group) |
| 13:18.32 | hrolf | [TK]D-Fender: In the hunt group '300', we have added all the other extensions, 302, 303, 304, 305, 306, 307 at IP Office end. |
| 13:19.14 | *** join/#asterisk serafie (~erin@nat/digium/x-skfcsqenzdkvbnpz) |
| 13:20.40 | cusco | ok ... |
| 13:20.46 | [TK]D-Fender | INVITE sip:302@192.168.1.171 SIP/2.0 |
| 13:20.48 | [TK]D-Fender | hrolf: And you are mixing up the number the DIALED with the USER they are calling |
| 13:21.14 | kchehab | MAster.csv is getting too big any way to let mysql-cdr.php delete the line after it insert it in mysql ? |
| 13:22.16 | [TK]D-Fender | kchehab: whose PHP script is that? I'm not familiar with anything provided with Asterisk for this |
| 13:23.35 | kchehab | i have a script that reads Master.csv it upload it to mysql cdr ,i already get this file from internet |
| 13:23.49 | Greenlight | WHy would you do such a crazy thing |
| 13:23.59 | Greenlight | Have asterisk send the CDRs' directly |
| 13:24.02 | [TK]D-Fender | kchehab: If that script has a problem you should contact its author. |
| 13:24.02 | kaldemar | because internet writes good scripts. |
| 13:24.18 | kaldemar | kchehab: i told you to use logrotate. why are you not using it? |
| 13:24.32 | kchehab | [TK]D-Fender asterisk does it use a php script to parse Master.csv then intert it in database if res_mysql is not enabled ? |
| 13:24.53 | [TK]D-Fender | kchehab: Your wording is unclear please rephrase |
| 13:25.10 | hrolf | [TK]D-Fender: The number dialed is 300, IP Office gives call to 303, * always receives it in 302. |
| 13:25.10 | Greenlight | kchehab: No, that's a 3rd party tool or script you have doing that. |
| 13:25.12 | [TK]D-Fender | kchehab: Asterisk doesn't have scripts. They are all external |
| 13:25.26 | [TK]D-Fender | kchehab: Asterisk has MODULES to choose how things are stored from the start |
| 13:25.26 | kchehab | kaldemar sorry i didt recognize that ,as for log rotate can it cause a problem since asterisk is writing to the file and log rotate is renaming it ? |
| 13:25.28 | roderickm | kaldemar: are you @borat_voip? |
| 13:25.49 | kchehab | [TK]D-Fender thanks got it |
| 13:25.54 | [TK]D-Fender | kchehab: If you have logs rotating, look at that subsystem |
| 13:26.01 | kaldemar | roderickm: eh? no. |
| 13:26.10 | hrolf | [TK]D-Fender: I want it such that the extension, which is INVITE'd receives the call on Asterisk end. |
| 13:26.31 | roderickm | kaldemar: "because internet writes good scripts" made me chuckle much like that twitter persona does. |
| 13:26.43 | hrolf | [TK]D-Fender: So if INVITE sip:303@192.168.1.171 SIP/2.0, 303 should receive, but Asterisk recieves on 302. |
| 13:26.47 | [TK]D-Fender | hrolf: INVITE is the DIALED number, not the identity of the caller. |
| 13:26.59 | hrolf | [TK]D-Fender: Yes. |
| 13:27.07 | [TK]D-Fender | hrolf: You are mixing things up. |
| 13:27.08 | hrolf | [TK]D-Fender: Wait, 300 is the dialed number. |
| 13:27.10 | kaldemar | kchehab: what problem would it cause? |
| 13:27.15 | [TK]D-Fender | 303 is the dialed number |
| 13:27.20 | hrolf | [TK]D-Fender: Nope. |
| 13:27.21 | [TK]D-Fender | that is how SIP is built |
| 13:27.32 | hrolf | [TK]D-Fender: 300 is the dialed number. |
| 13:27.40 | [TK]D-Fender | hrolf: They are not following the spec then |
| 13:27.41 | hrolf | [TK]D-Fender: That's what we are dialing. |
| 13:27.55 | hrolf | [TK]D-Fender: 300 is a hunt group, 302, 303.. are all its members. |
| 13:28.07 | kaldemar | roderickm: that's a new one to me. :) |
| 13:28.10 | hrolf | We dial 300, and Avaya invites, 302, 303, etc. ... |
| 13:28.18 | [TK]D-Fender | hrolf: You'll have to parse it out in the dialplan yourself then like I said |
| 13:28.26 | roderickm | hrolf: can we step back for a bit? Is the goal to limit to 6 the simultaneous calls to 300? |
| 13:28.48 | [TK]D-Fender | hrolf: and 303 is the number being "invited", so that is what is "dialed" |
| 13:28.49 | Qwell | roderickm: There's only one person that could pull that account off. |
| 13:28.55 | hrolf | roderickm: Nope, the goal is that the INVITEd extension should receive call. |
| 13:29.27 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-eunybekdfixfuuqk) |
| 13:29.29 | [TK]D-Fender | hrolf: not going to happen on a single instance. |
| 13:29.34 | hrolf | [TK]D-Fender: But before parsing it, out Asterisk already accepts the call on the SIP/302. |
| 13:29.44 | [TK]D-Fender | hrolf: I know. DEAL WITH IT. |
| 13:29.52 | [TK]D-Fender | hrolf: There is no fix in a single instance of Asterisk |
| 13:30.01 | hrolf | [TK]D-Fender: Okay. |
| 13:30.12 | hrolf | [TK]D-Fender: One question, can there be any issues with such a setup? |
| 13:30.29 | [TK]D-Fender | hrolf: "What "issues"? |
| 13:30.49 | [TK]D-Fender | hrolf: We have no idea what you are expecting beyond at least being able to ID them during the call |
| 13:31.01 | kchehab | kaldemar while logrotate is renaming the file and asterisk is writing to the file |
| 13:31.01 | [TK]D-Fender | hrolf: HOW, and what you expect to DO with that information we don't know |
| 13:31.16 | [TK]D-Fender | kchehab: Go change your log rotator then |
| 13:31.19 | hrolf | [TK]D-Fender: Right now, the problem we are facing is that, the extension which is INVITEd by Avaya, remains in busy state. |
| 13:31.32 | hrolf | [TK]D-Fender: I assumed it is because of 302 always receiving the calls. |
| 13:31.56 | [TK]D-Fender | hrolf: No. What your avaya assumes is what your avaya assumes |
| 13:32.08 | Greenlight | Would it not be more wise to configure asterisk as a SIP trunk on the avaya side, rather than individual extensions which is what it current seems like |
| 13:32.13 | [TK]D-Fender | hrolf: It sent the call where it thought it did. Whatever "status" it is imagining is all in its head |
| 13:32.28 | roderickm | If the Avaya is keeping track of which extensions are in the hunt group by virtue of inviting a specific peer to answer for 300 each time, why not just treat asterisk as a sip trunk provider? |
| 13:33.49 | hrolf | Greenlight: We tried that option, we face issues with REFERing with that. |
| 13:34.00 | Greenlight | Others have it working.. |
| 13:34.00 | igcewieling | Welcome to the horror of integrating proprietary PBXs with...well...anything. |
| 13:34.15 | Greenlight | Or better yet, if you Avaya extensions are all SIP, but your Avaya in the bin, and use Asterisk throughout... |
| 13:38.33 | *** join/#asterisk serafie (~erin@nat/digium/x-dlrwmgcflqjnxkmd) |
| 13:44.34 | [TK]D-Fender | hrolf: "INVITE" -> exten, "From:" -> CallerID, "To:" <- parse this out with a dialplan function. |
| 13:44.41 | [TK]D-Fender | hrolf: That's what you've got. |
| 13:47.55 | kaldemar | kchehab: logrorate works just fine with asterisk. |
| 13:49.15 | modesto916 | Hi guys, I would like to know, is there any channel variable that tells if the call is being originated from a attended transfer? |
| 13:50.28 | [TK]D-Fender | modesto916: No. |
| 13:50.56 | [TK]D-Fender | modesto916: Because a call is never "originated" as such. The call is independent until requested to be reinvited |
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| 14:04.59 | igcewieling | SIP would have been so much better if it included an actual transfer feature. |
| 14:05.33 | igcewieling | well, actual attended transfer feature. |
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| 14:08.04 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:12.24 | jeffspeff | what are the general thoughts for using the newest version of asterisk, exposing 5060 and RTP ports and running fail2ban for security? |
| 14:13.37 | [TK]D-Fender | jeffspeff: "par for the course" |
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| 14:13.39 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:14.12 | jeffspeff | [TK]D-Fender, it's monday and i'm not getting the golf reference. |
| 14:14.29 | [TK]D-Fender | jeffspeff: "perfectly normal" |
| 14:14.52 | [TK]D-Fender | jeffspeff: The only factor you actually listed was "fail2ban" which is not a bad idea if you have open clients |
| 14:15.25 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 14:15.41 | [TK]D-Fender | Limit vectors, track attempts |
| 14:16.35 | jeffspeff | i have to have open clients, i've been reading on fail2ban with asterisk and most of the documentation is a little outdated and said it still left asterisk open to invite attacks and some other things. i didn't know if those concerns were still relevant with the latest versions. |
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| 14:43.50 | As001 | Hello I tried to do Originate command from Asterisk AMI. When I answer it call goes to dialplan where I call Dial command, but when dialed number cancel (reject) call first line does not hangup and it calls again number from dialplan until waittimeout from dial command pass. I have h and H for options in Dial command. |
| 14:46.53 | Greenlight | That's by design, why is it an issue? |
| 14:47.55 | [TK]D-Fender | As001: Your dialplan does what your dialplan does, and AMI can be set to retry or not. This is up to you. |
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| 15:07.01 | As001 | [TK]D-Fender thanks can you tell me how to disable that retry in originate command from AMI ? |
| 15:07.18 | [TK]D-Fender | As001: read the instructions for Originate. It's all right there |
| 15:07.50 | As001 | manager show command Originate ? |
| 15:08.37 | As001 | I can't find anything about retry there. |
| 15:09.31 | [TK]D-Fender | As001: Show us your actual debug for starters... |
| 15:09.37 | [TK]D-Fender | ~pb |
| 15:09.37 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:09.43 | [TK]D-Fender | ^^^ |
| 15:11.07 | [TK]D-Fender | As001: AMI doesn't appear to have the retry option that call-files have. This would mean that you are causing it in your dialplan |
| 15:11.50 | As001 | i Can pastebin you my originate command and then part of dialplan which executes when I answer on originate command. |
| 15:11.56 | Greenlight | I seem to recall the AMI Originate doing it by default -- i could be wrong though |
| 15:12.20 | Greenlight | I just have a vague recollection of doing some testing, and rejecting a call, and it calling back |
| 15:12.57 | [TK]D-Fender | As001: show us the calls |
| 15:13.09 | Greenlight | Then again, it could have been my dialplan :) |
| 15:14.49 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:15.50 | As001 | Here is my Originate call and part of dialplan: http://pastebin.com/cTPyhXWT |
| 15:15.51 | igcewieling | If you want to retry you should use a Local/ channel |
| 15:16.20 | As001 | I would like to avoid retry I would like to cancel originated call when called from dial number reject call. |
| 15:17.18 | [TK]D-Fender | As001: show us the calls |
| 15:17.48 | igcewieling | Use a Local/ channel |
| 15:19.23 | Greenlight | It's the other side of the call which he wants to avoid retrying to |
| 15:20.07 | As001 | yes when person I call with dial reject call I want that Originated channel hang up not to call that number again. |
| 15:21.02 | [TK]D-Fender | As001: show us the calls |
| 15:21.04 | As001 | Tk I am afraid I can't give you calls right now because I would interrupt agents. I will use debug and write it tomorrow if I don't find solution by then. |
| 15:22.29 | Greenlight | You can get a SIP trace to the destination though, surely ? |
| 15:22.47 | Greenlight | I would be interesting to see *how* it's rejecting it. |
| 15:23.20 | igcewieling | Greenlight: using Local/ channels for both legs lets you do whatever you want with the call |
| 15:24.38 | Greenlight | Yea, it's definetly worth a shot. But without seeing the actual call or some traces we can just speculate, I just seem to recall a similar situation with AMI Originate years ago |
| 15:25.51 | As001 | I will have more info tomorrow. I will be back then. Thanks for help. |
| 15:25.58 | Greenlight | One thing |
| 15:26.03 | Greenlight | WHy are you calling Answer |
| 15:26.09 | Greenlight | Before it's actually answered |
| 15:26.26 | As001 | I read somewhere that some phones can not hangup if I dont' answer first. |
| 15:26.57 | Greenlight | Dial will answer the channel, if the remote parties *answers* |
| 15:27.00 | As001 | I hoped it will fix the issue |
| 15:28.16 | igcewieling | Usually Answer()ing when you don't need to answer is not the answer. |
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| 15:29.25 | As001 | ok I will remove it. |
| 15:29.31 | As001 | see you tomorrow |
| 15:29.35 | *** part/#asterisk As001 (~uros@82.117.198.142) |
| 15:29.40 | hjf | no caller ID for em |
| 15:29.41 | hjf | for me |
| 15:29.50 | hjf | i called several ITSP providers here |
| 15:29.58 | hjf | none of them offer outbound caller ID |
| 15:30.27 | hjf | one of them told me it's because of their routing. they can't guarantee they will be able to send out caller ID |
| 15:30.43 | hjf | that sounds more like "caller ID is hard and i don't want to bother" |
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| 15:32.09 | Greenlight | hjf: Nearly all ITSPs will present a callerid when you make calls. What exactly are you trying to do ? |
| 15:32.28 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 15:32.30 | Greenlight | Are you wanting to be able to present a number of *your own* choosing instead ? |
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| 15:37.45 | igcewieling | Greenlight: I believe he is in Argentina |
| 15:38.06 | igcewieling | and his outbound CID is showing as private or something like that. |
| 15:38.09 | Greenlight | Still seems odd that his ITSPs all withhold callerid |
| 15:38.19 | Greenlight | Then again, I'm not in Argentina :) |
| 15:38.48 | igcewieling | Greenlight: I suspect all the ITSP are using grey routes |
| 15:39.15 | Greenlight | That's a possability |
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| 16:00.14 | hjf | Greenlight: yeah i'm in Argentina. most itsps offer DID numbers but when you place a call it shows as private number |
| 16:00.41 | hjf | i can see caller ID when someone calls me, but i can't send my own (all the ITSPs say they can't) |
| 16:03.28 | skrusty | Greenlight: did you get a chance to look at the agi gallery website? (http://www.theagigallery.co.uk) ? |
| 16:04.00 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 16:04.08 | igcewieling | skrusty: It is on my to do list. ETA third quarter 2015 |
| 16:04.51 | skrusty | ;) |
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| 16:05.52 | hjf | maybe they really can't set the caller ID |
| 16:05.57 | hjf | i was thinking about it |
| 16:06.08 | hjf | and i'm pretty sure if they could, they would always display your number |
| 16:06.14 | hjf | and charge extra if you want to block it |
| 16:07.03 | hjf | knowing the people in this country, that sounds a lot more likely |
| 16:07.48 | skrusty | going to drop an email on the users list tonight try and get some more feedback, but wanted to try and get a bit from here first :) |
| 16:08.07 | skrusty | igcewieling: but that's for at least adding it to your list ;) |
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| 16:10.53 | Dovid | is there anyone from India in here? |
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| 16:40.19 | epaphus | Hello. I have a Sipura device under a nat connected to an asterisk server (which btw is also under a nat). lately the registration state on the sipura is listed as Failed, and the only way to get it working is to switch port 5061 to 5060... but then again if i reset the sipura same thing happens and i have to switch bak the port.. this is happening to often |
| 16:40.23 | epaphus | how can i diganose this? |
| 16:40.29 | epaphus | make sure its not the asterisk server |
| 16:40.53 | epaphus | i did login to the asterisk with -vvvv and when the sipura cannot connect i dont see any attempts at it. |
| 16:41.10 | *** part/#asterisk epaphus (~user@201.199.62.74) |
| 16:41.24 | igcewieling | turn off SIP ALG and/or SPI on your NAT routers |
| 16:43.14 | *** join/#asterisk epaphus (~user@201.199.62.74) |
| 16:43.19 | epaphus | sorry. back. |
| 16:43.47 | jeffspeff | <igcewieling> turn off SIP ALG and/or SPI on your NAT routers |
| 16:44.01 | jeffspeff | epaphus, ^^^^^ that's what you missed |
| 16:44.46 | [TK]D-Fender | [12:40]epaphusi did login to the asterisk with -vvvv and when the sipura cannot connect i dont see any attempts at it. <- this is not SIP DEBUG and will not inherently show you anything |
| 16:44.54 | [TK]D-Fender | epaphus: "sip set debug on" <- |
| 16:45.08 | epaphus | ah, forgot about the sip debug. |
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| 16:52.17 | boom^time | [TK]D-Fender, when you're managing an asterisk server and you're using sip debug to stomp out an issue how do you isolate it down so you only see the channels you're looking for? |
| 16:52.49 | [TK]D-Fender | Ctrl-F |
| 16:53.47 | boom^time | Okay, I was hoping there was some cleaner method. |
| 16:55.32 | Katty | try using facescrub next time. |
| 16:55.33 | igcewieling | boom^time: you can limit the debug to specific peers. |
| 16:55.42 | Katty | smiles at igcewieling |
| 16:55.50 | Katty | igcewieling: how're you feeling today dear? |
| 16:56.07 | igcewieling | Katty: I'm super, thanks for asking! |
| 16:56.20 | Katty | lifts an eyebrow |
| 16:56.29 | Katty | igcewieling: feeling super? well that's something |
| 16:56.38 | Katty | igcewieling: i'm glad you're in good spirits! |
| 16:57.04 | igcewieling | Katty: I was recently reminded nobody wants an honest answer to that question so that is now my standard response. A quote from a Weird Al song, I think. |
| 16:57.15 | Katty | frowns. |
| 16:57.28 | Katty | igcewieling: i /always/ want an honest answer from you. |
| 16:57.38 | Katty | igcewieling: if i didn't want to know how you were, i wouldn't ask. |
| 16:57.47 | igcewieling | I need the practice at lying. |
| 16:58.08 | Katty | igcewieling: and despite the fact i already have a pretty good idea of how you're feeling, i want you to know that i'm still thinking of you. |
| 16:58.22 | igcewieling | I tried just looking at the person and not saying anything, but people get all weirded out by that. |
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| 16:58.31 | Katty | nods |
| 16:58.33 | Katty | i imagine so. |
| 16:58.34 | igcewieling | Katty: THAT I do appreciate. |
| 16:59.29 | Katty | igcewieling: well good (= even if you're not feeling too perky physically, it's still nice to not be depressed all the time. |
| 16:59.31 | igcewieling | Katty: http://www.songlyrics.com/big-gay-al/i-m-super-lyrics/ |
| 16:59.53 | Katty | hmm. that sounds familiar. |
| 17:01.52 | igcewieling | I must admit it is more fun to just look at the person like I wish I had an axe. |
| 17:02.11 | Katty | grins |
| 17:02.12 | Katty | i bet! |
| 17:02.30 | igcewieling | "How are you today?" "like Lizzie borden" |
| 17:03.31 | hjf | anyone know if there's a DECT base like the cisco SPA302D but universal? |
| 17:03.40 | hjf | something i can associate the phones i already have |
| 17:04.02 | hjf | cause i'm reading that cisco base is only for cisco phones |
| 17:06.15 | igcewieling | hjf: EU DECT (maybe world) or USA DECT? |
| 17:07.29 | hjf | igcewieling: i'm gonna say EU cause i'm in Argentina and we mostly adopt euro standards |
| 17:07.43 | igcewieling | USA DECT is called DECT 6.0. DECT 6.0 handsets do not interoperate. |
| 17:08.04 | igcewieling | Make sure you are not looking at DECT 6.0 Cisco stuff. |
| 17:08.14 | hjf | well all my phones say DECT 6.0 and they're a mix and match |
| 17:08.42 | hjf | all of the Philips but different models. only calls. i can't use the phone directory or call history in the phones |
| 17:08.47 | igcewieling | From the Wiki "DECT 6.0 products do not support "GAP", therefore phones of differing brands are not guaranteed to work together. E.g. A GE 28105EE1 DECT 6.0 headset will not receive calls if registered to a Uniden base unit." |
| 17:09.41 | hjf | the only phone that won't associate is a Siemens one i have |
| 17:09.56 | hjf | but the siemens doesn't say DECT 6.0. it says dect. made in germany fwiw |
| 17:11.05 | hjf | yes the manual says this phone supports GAP |
| 17:11.18 | hjf | and it says DECT 6.0 heh |
| 17:12.12 | hjf | i tried with some panasonic ones and they won't register with the base |
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| 18:12.25 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 18:22.04 | *** join/#asterisk mitchrodrigues (~mitchrodr@c-98-244-30-255.hsd1.ca.comcast.net) |
| 18:22.12 | mitchrodrigues | curious |
| 18:22.21 | Katty | george. |
| 18:22.23 | mitchrodrigues | what could cause 1 sided audio from an audiohook |
| 18:22.52 | mitchrodrigues | seems we just randomly loose read frame |
| 18:22.56 | Katty | a serious lack of oatmeal raisin cookies. |
| 18:24.36 | mitchrodrigues | i had a hunch that was the culprit |
| 18:25.22 | Katty | you should make more. |
| 18:28.14 | *** join/#asterisk pensmit (~pensmit@unaffiliated/pensmit) |
| 18:32.42 | hjf | how would you deal with 'robocalls' and spam calls ? |
| 18:32.50 | hjf | i get a lot of them on my land line |
| 18:32.58 | hjf | i was thinking of using asterisk to filter them out |
| 18:33.20 | hjf | if there's no caller ID info (all spammers do that). pick up and hang up |
| 18:33.40 | hjf | or pick up and play music on hold until they hang up |
| 18:34.30 | *** join/#asterisk Alex25 (~kvirc@112.134.110.217) |
| 18:34.47 | [TK]D-Fender | Go for it |
| 18:36.15 | hjf | heh or play a message saying "your number is not recognized. if you feel this is an error, please don't hang up and dial the number again" |
| 18:36.48 | hjf | the most annoying part is when they play a "PLEASE DON'T HANG UP - A REPRESENTATIVE WILL TALK TO YOU" |
| 18:36.51 | hjf | that is |
| 18:36.56 | hjf | they call you . and put you on hold |
| 18:37.54 | hjf | i'm tired of getting cold calls from all banks, from greenpeace, from my bank to sell me insurance, opinion polls, etc |
| 18:38.11 | hjf | when banks are spamming, they hide CID |
| 18:38.26 | hjf | when banks are calling for a debt, their CID is visible |
| 18:38.42 | hjf | guess they know people don't pick up hidden numbers anymore |
| 18:39.02 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.113) |
| 18:39.29 | *** join/#asterisk russum (~russum@ool-4574dc67.dyn.optonline.net) |
| 18:40.58 | Alex25 | Hi |
| 18:42.48 | Katty | hi there. |
| 18:43.42 | Alex25 | I tried to discuss my problem here before yesterday, but got disconnected |
| 18:43.46 | Alex25 | anyway |
| 18:43.52 | Katty | unfortunate. |
| 18:44.04 | Alex25 | Every time I build asterisk-1.8.3.3 on a Debian7, chan_sip and res_crypto are not available on menuselect (XXX) |
| 18:44.15 | Alex25 | I have openssl and libssl-dev installed, so there must be another reason why it doesn't compile |
| 18:44.29 | Alex25 | looking for any hint which can help solve this problem |
| 18:44.38 | Qwell | Did you re-run configure? |
| 18:44.46 | Katty | did you confirm it's listed in make menuselect? |
| 18:44.57 | Katty | and if so, did you load the module afterwards? |
| 18:45.07 | Katty | Qwell: hello dear. |
| 18:45.10 | Qwell | ohai |
| 18:45.19 | Katty | Qwell: how's you and the ladyfriend? |
| 18:45.25 | Alex25 | it's listed as XXX on menuselect |
| 18:45.30 | Qwell | Did you re-run configure? |
| 18:45.36 | Alex25 | and yes I tried reconfigure |
| 18:45.38 | Qwell | also, hold on, what? |
| 18:45.42 | Qwell | ~upgrade asterisk |
| 18:45.42 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 18:46.17 | Katty | Qwell: i'll take that as everyone is doing well. |
| 18:46.21 | Alex25 | I need that specific version 1.8 |
| 18:46.33 | Qwell | Alex25: No you don't. |
| 18:46.35 | Alex25 | I worked well on debian6' |
| 18:46.45 | Alex25 | I-it |
| 18:47.04 | Katty | i need more cookies. |
| 18:47.13 | Alex25 | what do you think could cause that problem? |
| 18:47.15 | Katty | this lack of cookies is clearly affecting my afternoon |
| 18:47.22 | Qwell | Alex25: Being 3 years behind. |
| 18:47.23 | Katty | what caused it? |
| 18:47.26 | Katty | i forgot to bring cookies! |
| 18:47.37 | Qwell | Katty: That must be reftified. |
| 18:47.40 | Qwell | rectified, too |
| 18:47.46 | Katty | nods |
| 18:48.01 | Katty | cept i'm not gonna buy those crap ones from the vending machine |
| 18:48.31 | Alex25 | and who said that being 3 years behnd means failure to compile chan_sip? |
| 18:48.38 | Qwell | I did. |
| 18:49.36 | Alex25 | really? so i'm speechless |
| 18:49.52 | Qwell | What happens when you compile 1.8.23.1? |
| 18:50.12 | Katty | Qwell: you don't have the energy for this |
| 18:50.15 | Alex25 | I never tried to |
| 18:50.22 | Qwell | Alex25: Then we can't help you. |
| 18:51.11 | Alex25 | I don't have any reason to believe I'd get better results with 1.8.23.1 |
| 18:51.20 | Qwell | Alex25: You're more than welcome to go back to the revision where 1.8.3.3 was tagged, and look at all of the changes since then, to see whether your issue has been fixed. |
| 18:51.26 | Qwell | We, however, aren't going to. |
| 18:52.43 | [TK]D-Fender | Alex25: If you're looking for commentary on your specific attempts you should trash that folder and pastebin the complete process including ./configure |
| 18:53.07 | Katty | hi fender. how's your clavical dear? |
| 18:56.38 | *** join/#asterisk Alex25 (~kvirc@112.134.211.176) |
| 18:56.41 | [TK]D-Fender | Katty: not too bad.. my BACk is going out on me something nasty now though.... |
| 18:57.24 | Katty | [TK]D-Fender: yuck! how much longer you have until physical therapy? |
| 18:58.52 | Alex25 | so beside SSL things - what could be blocking chan_sip from properly compiling? |
| 18:59.13 | [TK]D-Fender | Katty: I started working my range on my own this past weekend... and the muscles are voicing their extreme disagreement with their new configuration... |
| 18:59.23 | Katty | nods sadly |
| 18:59.33 | [TK]D-Fender | AlexShow us the fresh debug |
| 18:59.35 | [TK]D-Fender | Alex25: Show us the fresh debug |
| 18:59.36 | Katty | [TK]D-Fender: is the knot in your back at least going away some? |
| 18:59.47 | [TK]D-Fender | Katty: No, that's what's "newer". |
| 18:59.58 | Katty | [TK]D-Fender: hmm :< |
| 19:00.02 | [TK]D-Fender | Katty: I've got 2 different major "tringes" |
| 19:00.07 | [TK]D-Fender | twinges* |
| 19:01.05 | Katty | nods |
| 19:01.31 | Alex25 | [TK]D-Fender: chan_sip and res_crypto both not available at all |
| 19:01.46 | [TK]D-Fender | AlexI hear the supposed results... no do the rest |
| 19:01.51 | [TK]D-Fender | now* |
| 19:03.33 | Alex25 | maybe I should have run ./configure with some options? |
| 19:04.37 | cusco | or maybe you should paste the output of plain ./configure as advised before, or even better, read it and check what your system is missing ... ! |
| 19:04.53 | [TK]D-Fender | Alex25: You'd been requested to have re-run it 3 times today so far from a fresh extract... |
| 19:05.53 | Alex25 | I did it about 4 times man.. |
| 19:06.39 | Alex25 | but I'll try to get the debug |
| 19:07.54 | Alex25 | I believe it's something with debian7, since I had the same issue with another debian7 VPS before |
| 19:08.47 | cusco | I was reading up and not finding a paste from you |
| 19:09.00 | cusco | I use debian 7 |
| 19:09.04 | cusco | no problems with that |
| 19:09.46 | Alex25 | and again, I need that secific 1 8 version since due to difference in the way some applications work on latest versions |
| 19:10.00 | cusco | what application are you refering to? |
| 19:10.30 | Alex25 | mainly chanspy |
| 19:10.36 | cusco | and ... I'm sure you don't need that specific version but can't you go to the 'older releases' section and find it on the website? |
| 19:10.56 | cusco | can you specify the difference in chanspy?= |
| 19:12.35 | Alex25 | yes |
| 19:12.57 | Alex25 | when I installed asterisk from package manager |
| 19:13.11 | Alex25 | which i assume to be one of latest versions |
| 19:13.18 | Alex25 | the wishper (w) option of chanspy had different behavior |
| 19:13.24 | cusco | on debian 7? its 1.8 if i'm not mistaken |
| 19:13.30 | cusco | how is it different? |
| 19:14.04 | cusco | there is a difference between 'w' and 'W' |
| 19:14.21 | Alex25 | no, its 'w' |
| 19:14.23 | cusco | depending on what channel you're spying ... |
| 19:14.28 | cusco | ok so.. what is the difference? |
| 19:15.02 | Alex25 | when a channel had some music played on, and then being spied on |
| 19:15.18 | Alex25 | on asterisk 1.8 thw w option cut the music |
| 19:15.40 | cusco | the music on hold? |
| 19:15.45 | Alex25 | and on later version the music has not been cut |
| 19:16.26 | Alex25 | so when i talked to the spied channel I could hear the music in background |
| 19:16.37 | Alex25 | which is unwanted behavior for me |
| 19:16.55 | Alex25 | again |
| 19:17.24 | Qwell | and what is the issue # for the bug you reported? |
| 19:17.27 | Alex25 | i'm sure there could be some later asterisk versions which also have the right behavior i need |
| 19:18.20 | Alex25 | but I can test all the version it will take very long time, |
| 19:18.34 | cusco | can't you just test the latest 1.8 ? |
| 19:18.49 | Qwell | cusco: Don't be silly. |
| 19:18.54 | Alex25 | I can |
| 19:19.07 | Alex25 | but i'm sure it won't work |
| 19:19.16 | cusco | ok, which version did you test? |
| 19:19.37 | Alex25 | since it smells like the problem is some missing component on debian7 |
| 19:20.16 | Alex25 | I had the 1,8 version on debian6 until recently without problems |
| 19:20.29 | cusco | which one are you testing now? |
| 19:21.10 | Alex25 | but whenever I try to install it on debian7 (and i tried on 2 different VPS) cahn_sip fail to compile |
| 19:21.55 | Alex25 | I'm still with with 1.8.3.3 |
| 19:22.07 | Alex25 | trying to solve this problem |
| 19:22.13 | cusco | on debian squeeze Version: 1:1.8.13.1~dfsg-1~bpo60+1 |
| 19:22.45 | cusco | well... |
| 19:23.00 | cusco | the ./configure output - does it state about missing dependencys? |
| 19:23.01 | Qwell | ~upgrade asterisk |
| 19:23.01 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 19:23.04 | cusco | libs and so on? |
| 19:24.02 | Alex25 | I'm trying to get the log now |
| 19:24.08 | cusco | also .. the latest 1.8 release is 1.8.23.1 ... should have stuff from earlier versions fixed ... |
| 19:24.10 | Alex25 | into a file |
| 19:24.15 | cusco | err... |
| 19:24.37 | cusco | normally I always do ./configure |tee conf.txt so I can go and read what it was looking for whenever I miss something |
| 19:24.48 | cusco | (normally and always ... ) |
| 19:25.10 | cusco | I'm going to have dinner now.. be back later... good luck ! |
| 19:25.19 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 19:25.21 | Alex25 | thank you |
| 19:25.39 | cusco | also ... 1.8 is getting old, you should test 11 |
| 19:27.51 | igcewieling | cusco: 1.8 is LTS. There is nothing wrong with 1.8 |
| 19:28.06 | igcewieling | 1.8, is, in point of fact, still a supported version of Asterisk |
| 19:39.48 | *** join/#asterisk Blashyrkh (~chatzilla@88-117-33-192.adsl.highway.telekom.at) |
| 19:44.17 | Alex25 | [TK]D-Fender: here is the configure log: http://pastie.org/8311747 |
| 19:45.43 | [TK]D-Fender | Alex25: And what do you see in menuselect? |
| 19:46.35 | Alex25 | I see chan_sip is XXX, as well as res_crypto |
| 19:48.09 | [TK]D-Fender | And if you look it will tell you specifically what you're missing as dependencies goes... |
| 19:48.32 | Alex25 | ? what is it? |
| 19:48.38 | [TK]D-Fender | You tell us |
| 19:48.45 | [TK]D-Fender | the menuselect screen tells you |
| 19:49.51 | igcewieling | Alex25: go look at res_crypto! |
| 19:50.44 | Alex25 | not sure... |
| 19:51.08 | igcewieling | Alex25: what part of "depends on [something here]" is confusing? |
| 19:51.10 | Alex25 | the dependency for chan_sip is ssl which I have installed already |
| 19:51.30 | Alex25 | ah |
| 19:51.39 | igcewieling | Alex25: there that was not so hard. Did you re-run ./configure after installing openssl-devel? |
| 19:51.49 | Alex25 | I looked at it before |
| 19:52.00 | Alex25 | ^^ |
| 19:52.03 | Alex25 | I have ssl installed i said |
| 19:52.19 | Alex25 | that's the whole problem |
| 19:52.39 | Alex25 | although i have that dependency sip won't compile |
| 19:52.44 | igcewieling | you need the development headers in addition to the library |
| 19:53.16 | igcewieling | Alex25: did you re-reun ./configure after installing the development headers for ssl? |
| 19:53.48 | Alex25 | I had 'openssl' and 'libssl-dev' installed at all time |
| 19:54.22 | igcewieling | Alex25: how may files are in /usr/include/openssl/ (ls /usr/include/openssl/ | wc) |
| 19:54.25 | Alex25 | there is no package named 'openssl-devel' on debian7 |
| 19:55.51 | igcewieling | Alex25: apt-get install -y build-essential linux-headers-`uname -r` openssh-server apache2 mysql-server mysql-client libgnutls26 bison flex php5 php5-curl php5-cli php5-mysql php-pear php-db php5-gd curl sox libncurses5-dev libssl-dev libmysqlclient15-dev mpg123 libxml2-dev libnewt-dev sqlite3 libsqlite3-dev pkg-config automake libtool autoconf git subversion |
| 19:56.37 | igcewieling | once you've completed that, re-run ./configure to pick up any changes |
| 19:56.49 | [TK]D-Fender | libssl-dev <- |
| 19:56.50 | igcewieling | if that doesn't work then do a "make distclean" then re-run ./configure |
| 19:57.22 | Alex25 | [TK]D-Fender: I have it installed already |
| 19:57.39 | igcewieling | Alex25: have or have you not re-run ./configure |
| 19:58.28 | [TK]D-Fender | Yup, I've hit my "running in circles" quota for the day... |
| 19:58.31 | Alex25 | igcewieling: I did all that already, I;ve been struggling with ths problem several days |
| 19:59.13 | [TK]D-Fender | moves on to other matters |
| 19:59.48 | igcewieling | If it was a general problem with Debian 7 there would be lots of reports of issues with Debian 7 and Asterisk. |
| 20:00.15 | Alex25 | [TK]D-Fender: I already said above that I had openssl and libssl-dev installed at all time. That is NOT the problem |
| 20:01.07 | igcewieling | You'd think the person who wrote this http://blog.keer.info/install-asterisk-and-freepbx-on-debian-wheezy-7/ or this http://wiki.freepbx.org/display/HTGS/Installing+FreePBX+on+Debian+Wheezy or this http://thelamegerman.wordpress.com/2012/04/20/installation-of-freepbx-2-10-on-debian-wheezy-kernel-3-2/ would have had problems |
| 20:01.21 | tm1000 | sigh |
| 20:01.26 | tm1000 | igcewieling: no i didnt have problems |
| 20:01.27 | tm1000 | at all |
| 20:01.30 | tm1000 | I ran it myself |
| 20:01.33 | tm1000 | whats the error |
| 20:01.41 | igcewieling | tm1000: but you're smart! 8-) |
| 20:01.47 | tm1000 | igcewieling: tell me whats wrong |
| 20:02.06 | igcewieling | tm1000: not entirely sure. "(3:51:10 PM) Alex25: the dependency for chan_sip is ssl which I have installed already" |
| 20:02.26 | Alex25 | igcewieling: this is a problem with a secific version of * 1.8 |
| 20:03.01 | tm1000 | igcewieling: what? |
| 20:03.08 | tm1000 | read what Alex25 is saying above |
| 20:03.14 | tm1000 | my guide talks about Asterisk 11 |
| 20:03.20 | tm1000 | not astierk 1.8 |
| 20:03.34 | tm1000 | and not about tls or ssl |
| 20:04.00 | igcewieling | tm1000: are you saying 1.8 cannot be installed on Debian 7? |
| 20:04.25 | tm1000 | igcewieling: no I am saying dont reference my guide saying it's wrong when talking about asterisk 1.8 since I wrote it around 11 |
| 20:04.39 | tm1000 | or tell me why I would have had problems |
| 20:05.03 | igcewieling | Huh? I said your guide is wrong? |
| 20:05.17 | tm1000 | "You'd think the person who wrote this""would have had problems" |
| 20:05.21 | tm1000 | oh |
| 20:05.22 | tm1000 | god |
| 20:05.23 | igcewieling | What I'm sating is ALEX is wrong. |
| 20:05.24 | tm1000 | out of context |
| 20:05.28 | tm1000 | im an a**hat |
| 20:05.35 | tm1000 | is a**hole |
| 20:05.38 | tm1000 | sorry igcewieling <3 |
| 20:05.46 | [TK]D-Fender | Nope, just cross-eyed |
| 20:05.51 | tm1000 | yeah apparently |
| 20:05.52 | tm1000 | ugh |
| 20:06.01 | igcewieling | If Asterisk 1.8 didn't work on Debian 7 I'd expect the villiagers to storm huntsville. |
| 20:06.11 | [TK]D-Fender | quickly clarified, nothing to worry about |
| 20:06.17 | tm1000 | :-) |
| 20:06.17 | igcewieling | tm1000: you have very high karma, don't worry about it 8-0 |
| 20:06.22 | tm1000 | hahah |
| 20:06.23 | tm1000 | thanks |
| 20:06.28 | tm1000 | maybe -1 point now ;-) |
| 20:06.59 | tm1000 | if chan_sip is looking for ssl wouldn't that be someone trying to use TLS? |
| 20:10.00 | Alex25 | chan_sip " Depends on: chan_local(M), res_crypto(M) " |
| 20:10.09 | Qwell | ffs, where's the pastebin you've been asked for, for the last hour? |
| 20:10.14 | Alex25 | what is chan_local(m) |
| 20:10.16 | Alex25 | ? |
| 20:10.26 | tm1000 | the local chan |
| 20:10.54 | Alex25 | Qwell: already provided a paste |
| 20:11.07 | Alex25 | http://pastie.org/8311747 |
| 20:11.22 | Qwell | checking for ssl2_connect in -lssl... no |
| 20:11.32 | Qwell | Install openssl development libraries. Next? |
| 20:12.10 | Alex25 | I already have openssl + libssl-dev, anything else? |
| 20:12.12 | Qwell | If only somebody were running a newer version of Asterisk, which has this problem fixed. |
| 20:13.01 | Qwell | ~upgrade asterisk |
| 20:13.01 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 20:13.27 | Alex25 | infobot: I need that specific version |
| 20:13.27 | infobot | You need that specific version? |
| 20:14.02 | *** join/#asterisk Alex_Bkash (b4ea45c2@gateway/web/freenode/ip.180.234.69.194) |
| 20:14.13 | tm1000 | yes infobot. geesh get with the program (stupid bot) |
| 20:14.19 | Qwell | https://issues.asterisk.org/jira/browse/ASTERISK-17668 |
| 20:14.22 | Alex_Bkash | asterisk.h not found in /include /usr/include /usr/local/include /opt/local/include |
| 20:14.26 | *** kick/#asterisk [Alex25!~north@pdpc/sponsor/digium/Qwell] by Qwell (Come back later.) |
| 20:14.34 | Alex_Bkash | cannot run ./configure |
| 20:14.42 | Qwell | Alex_Bkash: context? |
| 20:14.57 | Alex_Bkash | installing chan_dongle |
| 20:15.47 | Alex_Bkash | trying to actually |
| 20:16.06 | Qwell | Alex_Bkash: Where did you install Asterisk? |
| 20:16.26 | Qwell | I suppose how is probably the better question. Via some package, probably? |
| 20:16.33 | Alex_Bkash | with apt-get install |
| 20:16.41 | Qwell | install the -dev package |
| 20:17.05 | Alex_Bkash | ? |
| 20:17.17 | Alex_Bkash | reinstall asterisk? |
| 20:17.21 | Qwell | apt-get install asterisk-dev |
| 20:18.11 | Alex_Bkash | will it remove previous configs? |
| 20:19.00 | Qwell | no |
| 20:19.32 | Alex_Bkash | i dont know about this actually |
| 20:19.40 | Alex_Bkash | whats this package? |
| 20:20.33 | igcewieling | Alex_Bkash: just run the command |
| 20:22.55 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 20:22.55 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:26.48 | Alex_Bkash | it worked |
| 20:26.54 | Alex_Bkash | thaks a lot |
| 20:28.06 | Alex_Bkash | @Qwell: Thanks |
| 20:28.33 | Alex_Bkash | igcewieling:Thanks |
| 20:34.42 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.113) |
| 20:35.57 | igcewieling | Why would someone forward their phone to 47*69#325802, what did they do have a seizure when forwarding their phone? |
| 20:38.08 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 20:44.19 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-68-160-43.home.otenet.gr) |
| 20:46.06 | *** part/#asterisk g_r_eek (~g_r_eek@ppp-94-68-160-43.home.otenet.gr) |
| 20:49.39 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 20:49.58 | Blashyrkh | if i dont define a username in a channel definition in the sip.conf will asterisk asssume the username is the channel name? |
| 20:50.56 | Qwell | s/channel/context/, maybe? |
| 20:51.47 | Blashyrkh | ? |
| 20:52.27 | Blashyrkh | i dont quite get what you are saying sry |
| 20:56.40 | igcewieling | maybe peer? |
| 20:56.54 | igcewieling | Blashyrkh: sip.conf has nothing to do with channels |
| 20:57.38 | Blashyrkh | the definite guide tells me something different :D |
| 20:58.11 | igcewieling | Blashyrkh: where? |
| 20:58.12 | Blashyrkh | it refers to definitions in sip.conf iax.conf etc. as channels |
| 20:58.22 | Blashyrkh | in my printed version on p.123 |
| 20:58.41 | igcewieling | paste the quote? |
| 20:58.47 | igcewieling | (no more than 3 lines) |
| 20:59.34 | Blashyrkh | i have the book liying in front of me but wait a moment |
| 21:00.31 | Blashyrkh | When you define a channel (which is not done in the extensions.conf file, but rather in files such as sip.conf, ia one of the required parameters in each channel definition is context. |
| 21:01.29 | igcewieling | heh, sorry, you did say printed. |
| 21:01.37 | igcewieling | The book is wrong. They are called peers, friends, or users. |
| 21:02.27 | Blashyrkh | okay thanks |
| 21:02.41 | Blashyrkh | but if i define a friend, say for a deskphone |
| 21:03.00 | Blashyrkh | if i dont define username, it will use the name as username right? |
| 21:03.36 | Blashyrkh | but if i define the username, can i then name the friend whatever i like? |
| 21:04.36 | igcewieling | Blashyrkh: username, fromuser, and the various other user related stuff changed between versions, check sip.conf. If you leave it out (at least for sip.conf) it should use the username defined in the [] |
| 21:04.42 | *** join/#asterisk Alex_Bkash (b4ea3c61@gateway/web/freenode/ip.180.234.60.97) |
| 21:05.38 | Alex_Bkash | hello @Qwell: cannot do "module load chan_dongle.so" |
| 21:05.55 | Qwell | Alex_Bkash: I can't help with third-party modules. Maybe try contacting the author? |
| 21:06.56 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:07.17 | Alex_Bkash | was not compiled with the same compile time option as the version of Asterisk |
| 21:07.22 | Alex_Bkash | :( |
| 21:10.31 | igcewieling | Alex_Bkash: contact the author of the module like Qwell said |
| 21:15.01 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:22.32 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 21:25.02 | Penguin | blashyrkh: The peer's name is what goes between the square brackets. It doesn't matter if the type is peer, friend, or user. |
| 21:25.27 | Penguin | The username value is used for something else. |
| 21:29.53 | Blashyrkh | fr what? |
| 21:30.41 | Penguin | For example, you need to register your asterisk to another server. You name the peer anything you want, but the username value is the username that you send for authentication to the other server. |
| 21:31.15 | Penguin | The username value is what they put into the square brackets on their machine. |
| 21:31.31 | Blashyrkh | ty |
| 21:34.23 | j4jackj | Alex_Bkash is trying to get me involved in a crime, and I'm diverting him. |
| 21:35.54 | Qwell | j4jackj: and what crime would that be? |
| 21:43.59 | Alex_Bkash | lol |
| 21:44.27 | Alex_Bkash | j4jackj what that? |
| 21:44.36 | WIMPy | Using VOIP. |
| 21:44.58 | Alex_Bkash | crime indeed |
| 21:44.59 | WIMPy | Everybody knows that only terrorists use VOIP. |
| 21:45.33 | Alex_Bkash | then its a terrorist room |
| 21:45.36 | [TK]D-Fender | I thought we left off with him trying to use GSM across the border to circumvent the VoIP restrictions... |
| 21:45.41 | Alex_Bkash | is not it? |
| 21:46.07 | Alex_Bkash | that was just and idea |
| 21:46.16 | Alex_Bkash | not every idea gets implemented |
| 21:46.24 | WIMPy | Oh, we had that some months back. |
| 21:46.25 | Alex_Bkash | even not possible |
| 21:46.37 | Alex_Bkash | not months |
| 21:46.40 | Penguin | In our country, it is the intent that makes it illegal. |
| 21:46.42 | Alex_Bkash | may be weeks |
| 21:46.47 | Penguin | Not the action, but the intention. |
| 21:47.08 | Penguin | Developing the plan to implement it proves intent. |
| 21:47.15 | WIMPy | No months. It was someone from Bangladesh. |
| 21:47.34 | Alex_Bkash | it was me WIMPy |
| 21:47.43 | Alex_Bkash | its weeks ago |
| 21:47.49 | *** join/#asterisk mitchrodrigues (~mitchrodr@c-98-244-30-255.hsd1.ca.comcast.net) |
| 21:47.57 | Alex_Bkash | i asked if its possible to use openBTS |
| 21:47.58 | WIMPy | Oh, ok. But it was a long time ago. |
| 21:48.11 | mitchrodrigues | hm i still wish i knew what would cause one side of frames to dissapear on an audiohook |
| 21:48.12 | mitchrodrigues | -.- |
| 21:48.20 | WIMPy | Nah, that's not what I was thinking of. |
| 21:48.21 | mitchrodrigues | this has been haunting me for weeks |
| 21:49.08 | WIMPy | It was someone who needed an extreme low bitrate codec so he vould hide a VOIP/GSM gateway in a remote village. |
| 21:54.14 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 21:55.54 | SeRi | Penguin: long time no see |
| 21:57.54 | Penguin | looks around curiously |
| 22:01.30 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 22:03.08 | *** join/#asterisk roderickm (~roderickm@67.63.143.254) |
| 22:03.56 | SeRi | waves @ Penguin |
| 22:04.11 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 22:04.44 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 22:07.30 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.242) |
| 22:12.29 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-scwjowdvnppfajfm) |
| 22:49.10 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 23:10.53 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 23:48.51 | hjf | how can i do sip show peers and sip show channels from the web? |
| 23:49.13 | hjf | i'm trying AMI |
| 23:49.26 | hjf | well, i think i am... from the /manager url in asterisk http |
| 23:49.54 | hjf | and i can login and do sip show peers and all but it's ugly |
| 23:49.55 | igcewieling | hjf: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Command |