| 00:00.49 | ChannelZ-Wk | There was a security thing in October with SRTP so who knows what their package is (is lenny LTS of some sort?) |
| 00:01.54 | *** join/#asterisk RobertLaptop (~rmiddle@74.112.203.154) |
| 00:02.17 | jamicque | all debian version are LTS, however lenny is outdated :( |
| 00:03.35 | jamicque | lenny was released in 2009, support edned in 2012. I have one system where I would like to lunch webrtc and don;t like to update os |
| 00:03.36 | jamicque | :) |
| 00:05.58 | ChannelZ-Wk | resistence is futile |
| 00:08.16 | jamicque | I'll first try installing openssl from source... maybe this would help... |
| 00:13.11 | MaliutaLap | jamicque: so don't re-install the OS, migrate the services off to something newer |
| 00:14.43 | jamicque | MaliutaLap: I don't want to reinstall OS, the problem is that on this server there is notonly asterisk and I want to neable webrtc there. All my newer instances are running on 14.04 - on Docker! :) |
| 00:15.32 | jamicque | However, I have one old lenny in my backpack... And I do not want to touch it harder than I should :) |
| 00:34.49 | RobertLaptop | I am looking to control asterisk using a web app. Asterisk gui would have fit the bill but it delevelopment was dropped. Is there anything that I can use instead? And I don't think freepbx is an option as I need something that stays out of the way. Freepbx does anything but that. |
| 00:35.25 | ChannelZ-Wk | Asterisk + GUI + Stays Out Of The Way = Does not compute |
| 00:40.18 | robmal | RobertLaptop: There's XiVu and SAIL. |
| 00:40.31 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 00:41.07 | robmal | We have multiple sites running SAIL, some quite large so it might be worth taking a look for you. |
| 00:42.38 | RobertLaptop | ChannelZ-Wk, The Asterisk GUI project was designed to stay out of your way. All says were made direct to the dialplan files no DB or other middle layers in between. |
| 00:42.51 | RobertLaptop | robmal, I will have to look at those. |
| 00:43.57 | robmal | They both have their downsides, if you need any help /msg me |
| 00:45.43 | RobertLaptop | robmal, OK. SAIL looks like a PBX project not a library that allows the processing of dialplans using AJAX type calls. |
| 00:48.25 | *** part/#asterisk kharwell (kharwell@nat/digium/x-jkqrrftkczyxujhy) |
| 00:50.05 | robmal | Yup, but they're starting to catch up with web2.0 as we speak ;-) If you need any features prepared, aside custom apps, they can put them in place quite fast. |
| 00:55.26 | mbowie | RobertLaptop: Let me know if you find anything promising... I took a quick glance last week (briefly) without success. (vi education for admins FTW!) |
| 01:00.48 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 01:04.26 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |
| 01:08.18 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |
| 01:10.51 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |
| 01:12.17 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |
| 01:32.58 | RobertLaptop | mbowie, I understand. The Asterisk QUI filled a nich that doesn't seem to be covered anymore. |
| 01:35.00 | mbowie | RobertLaptop: Agreed. |
| 02:05.17 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 02:06.56 | Borg | anyone about? I need help with 'No SMDI interfaces are available to listen on, not starting SMDI listener.' |
| 02:10.18 | jab416171 | hmm... I'm trying to set it up so if I call from my windows PC (extension 130) it uses a different route than if I call from anything else. |
| 02:10.29 | jab416171 | I thought it was just a matter of setting "route CID" but that didn't do it |
| 02:10.42 | jab416171 | oh duh, I'm looking at outbound, should be looking at inbound |
| 02:10.43 | *** join/#asterisk calum_ (~calum_@cpc70817-harg4-2-0-cust754.7-1.cable.virginm.net) |
| 02:18.55 | *** join/#asterisk zerick_ (~zerick@179.7.77.196) |
| 02:19.01 | jab416171 | dang, it's still not working |
| 02:22.33 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 02:22.33 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:26.21 | *** join/#asterisk sgriepentrog (~sgriepent@2602:30a:2ef2:840:20c:29ff:fe4f:cac8) |
| 02:27.21 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 02:27.23 | RobertLaptop | mbowie, https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Asterisk+REST+API |
| 02:27.25 | *** join/#asterisk Ta^3 (~tacvbo@187.188.107.19) |
| 02:39.22 | *** join/#asterisk ulogic (421e6b4f@gateway/web/freenode/ip.66.30.107.79) |
| 02:39.34 | Borg | anyone? |
| 02:43.36 | *** join/#asterisk DrKK` (~DrKK@unaffiliated/drkk/x-3450090) |
| 02:43.42 | *** part/#asterisk DrKK` (~DrKK@unaffiliated/drkk/x-3450090) |
| 02:48.04 | jab416171 | doesn't look like anyone is here Borg |
| 02:50.07 | RobertLaptop | I am sure there are people around but without knowning the question kinda hard to answer. |
| 02:51.47 | WIMPy | Yes. might be hard to find anyone using SMDI. I can't remember it ever being mentioned here before. |
| 02:51.48 | Borg | RobertLaptop, this is the problem : I need help with 'No SMDI interfaces are available to listen on, not starting SMDI listener.' |
| 02:52.28 | RobertLaptop | Yes not even sure what SMDI even is. |
| 02:52.33 | Borg | hmm... well the problem is I'm trying to get Asterisk to start on FreeBSD, but i'm not sure it's actually running, hung or what. |
| 02:53.02 | jab416171 | how do I configure asterisk so, if I'm calling from extension 130 to any number, use Trunk X, but for all other calls, use Trunk Y? |
| 02:53.09 | Borg | Simplified Message Desk Interface (SMDI) is a protocol that defines the interface between a voice mail system and a phone system such as a PBX or public telephone switch. It was developed by Bell Labs. It is used to provide the voice mail system the information it needs to process the call. |
| 02:53.33 | WIMPy | Borg: So so where do you get so far? |
| 02:54.18 | WIMPy | jab416171: Use contexts. And Don't call devices "extensions". Extensions don't make calls. |
| 02:55.05 | Borg | WIMPy, well I compiled it from ports, did an initial configuration of /usr/local/etc/asterisk/asterisk.conf and tried to fire up the daemon. I've also been looking for a handy GUI to manage it via a web interface, but no success on that yet either. |
| 02:55.40 | jab416171 | I guess a better way to put it: I want to use Trunk X for outgoing calls that originate from me (I call asterisk, and then call out to someone, or make a call from a softphone), and Trunk Y for all incoming calls from other people that then call my phones. |
| 02:55.46 | jab416171 | WIMPy, what are contexts? |
| 02:56.26 | WIMPy | Borg: But you already got some output. So it at least starts (to start). |
| 02:56.45 | WIMPy | ~book |
| 02:56.45 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 02:56.49 | WIMPy | ~primer |
| 02:56.50 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
| 02:57.09 | jab416171 | if it makes a difference, I'm using asterisk 11 and PIAF |
| 02:57.11 | WIMPy | jab416171: there you have two starting points on how to use Asterisk. |
| 02:58.16 | WIMPy | Yes. We can't support any configuration utilities here. |
| 02:58.59 | WIMPy | They tend to use very different terminology from Asterisk and are hard to understand from teh Asterisk point of view. |
| 02:59.07 | Borg | WIMPy, yeah. I'm talking with the Webmin devs to see if they might consider such a module. would be quite handy. |
| 02:59.09 | jab416171 | understood |
| 03:00.09 | WIMPy | jab416171: So PIAF uses FreePBX, which is the topic of #freepbx. |
| 03:00.24 | jab416171 | WIMPy, I tried asking in #freepbx but nobody's around |
| 03:01.49 | WIMPy | That happens frequently, but asking here about FreePBX configuration won't work. |
| 03:03.01 | jab416171 | got it |
| 03:06.45 | WIMPy | Borg: About that SMDI message: You can safely ignore that. But you still haven't told us how far you got otherwise. |
| 03:07.02 | jab416171 | ah, I figured it out. I just need to change my dial pattern so it matches on caller id for the outbound route |
| 03:07.27 | Borg | WIMPy, I haven't gotten much further. the daemon just crashes with no other error. :-s |
| 03:07.52 | WIMPy | Borg: Then start it with -cvvvddd. |
| 03:10.26 | Borg | WIMPy, aha! now that was useful. '/usr/local/lib/asterisk/modules/pbx_lua.so: Undefined symbol "luaL_loadbuffer" |
| 03:10.26 | Borg | ' |
| 03:10.40 | Borg | so LUA is a problem I see. interesting. |
| 03:11.06 | WIMPy | Pretty interesting if you compiled yourself. |
| 03:11.45 | WIMPy | But that should probably only keep the module from loading. |
| 03:18.52 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 03:25.24 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 03:27.13 | linocisco | what is asterisk SCF? |
| 03:29.07 | jab416171 | anyone have a recommendation for a decent windows SIP softphone? |
| 03:29.28 | jab416171 | I have zoiper right now and it keeps having problems with routing audio to the right source |
| 03:30.04 | linocisco | jab416171, what about x-lite? |
| 03:30.20 | jab416171 | haven't tried it |
| 03:37.54 | jab416171 | thanks linocisco, it works a lot better than zoiper |
| 03:38.09 | jab416171 | zoiper wouldn't even ring when I called it, and for some reason, it was registering itself with my public IP |
| 03:38.41 | linocisco | jab416171, i was using 3CX before, but now it is only compatible with 3CX PBX system |
| 03:40.21 | jab416171 | well that's silly |
| 03:44.16 | jab416171 | yeah this client's a lot better, thanks again |
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| 04:03.58 | jab416171 | can asterisk do SMS? |
| 04:04.12 | volga629 | Hello Everyone, how to troubleshoot this issue |
| 04:04.43 | volga629 | [2015-02-17 23:02:04] WARNING[1951]: chan_sip.c:3751 __sip_xmit: sip_xmit of 0x7ff014028960 (len 593) to clienpubip:1025 returned -2: No such file or directory |
| 04:04.43 | volga629 | [2015-02-17 23:03:14] WARNING[1951]: chan_sip.c:3751 __sip_xmit: sip_xmit of 0x7ff014028960 (len 593) to cline pub ip:1025 returned -2: Success |
| 04:04.47 | WIMPy | jab416171: If you're talking real SMS, yes. |
| 04:05.07 | jab416171 | WIMPy, "real" SMS? |
| 04:05.17 | volga629 | this I only see with TLS clients |
| 04:05.23 | WIMPy | Like what's used in the PSTN. |
| 04:08.39 | jab416171 | so, what clients support receiving SMS in that way? |
| 04:08.44 | jab416171 | can I send an SMS to a cell phone? |
| 04:08.59 | [TK]D-Fender | It has nothing to do with where it goes |
| 04:09.03 | [TK]D-Fender | it's abot how it gets there |
| 04:09.07 | [TK]D-Fender | about* |
| 04:09.16 | WIMPy | If you have access to an SMSc that supports it, yes. |
| 04:09.41 | jab416171 | SMSc? |
| 04:09.53 | jab416171 | does google voice count? |
| 04:10.19 | WIMPy | The Short Message Service Center. |
| 04:10.40 | WIMPy | The switch for SMs. |
| 04:10.58 | [TK]D-Fender | jab416171, Forget GV |
| 04:11.30 | jab416171 | why do you say that? |
| 04:11.42 | [TK]D-Fender | Because they are pulling XMPP and are effectively dead |
| 04:11.45 | [TK]D-Fender | Forget them |
| 04:12.08 | jab416171 | I can send an SMS without relying on XMPP |
| 04:12.19 | [TK]D-Fender | using what to get to GV? |
| 04:12.28 | jab416171 | http? |
| 04:12.29 | [TK]D-Fender | What would Asterisk have to speak to do this? |
| 04:12.35 | WIMPy | Real SMS involves a modem connection. |
| 04:13.09 | WIMPy | Off course there are http services and the like. But that has nothing to do with anything telephony related any more. |
| 04:13.23 | jab416171 | that's true |
| 04:13.35 | jab416171 | okay, how would I send an SMS with the PSTN? |
| 04:14.08 | RobertLaptop | WIMPy, smpp is much more used method for sending SMS. I have no idea if Asterisk supports SMPP. |
| 04:14.09 | [TK]D-Fender | <jab416171> can asterisk do SMS? |
| 04:14.09 | [TK]D-Fender | <[TK]D-Fender> Sortof |
| 04:14.09 | [TK]D-Fender | <[TK]D-Fender> It can do SMS over E1, or via SIP MESSAGE if you have an ITSP that supports it |
| 04:14.09 | [TK]D-Fender | <[TK]D-Fender> Or via chan_dongle, etc |
| 04:14.26 | jab416171 | right |
| 04:14.35 | WIMPy | What is SMPP? |
| 04:14.37 | jab416171 | I didn't understand most of what you said |
| 04:14.45 | WIMPy | core show application SMS. |
| 04:15.10 | RobertLaptop | WIMPy, http://en.wikipedia.org/wiki/Short_Message_Peer-to-Peer |
| 04:15.17 | jab416171 | compatible with BT SMS PSTN service in UK and Telecom Italia in |
| 04:15.17 | jab416171 | Italy. |
| 04:15.40 | WIMPy | RobertLaptop: Is that a new thing? |
| 04:15.42 | jab416171 | is it saying that's the only PSTN it's compatible with? |
| 04:15.59 | WIMPy | jab416171: ETSI |
| 04:16.29 | jab416171 | ah |
| 04:16.38 | WIMPy | Ugh. That looks horrible. |
| 04:17.58 | RobertLaptop | WIMPy, Not sure if Asterisk supports it. That is the way all the carriers pass messages around. |
| 04:18.28 | WIMPy | Not something I came across so far. |
| 04:20.31 | coppice | SMPP is one of several protocols uses to exchange SMS between carriers, and between large customers and carriers. its pretty much unmaintained now |
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| 04:38.11 | *** join/#asterisk mattsl (~user@68.169.165.200) |
| 04:38.24 | mattsl | Anyone have any suggestions on what cause stuck calls? |
| 04:42.26 | WIMPy | That's not enough information to make a comment. |
| 04:42.41 | WIMPy | And define "stuck". |
| 04:45.58 | mattsl | Specifically, I've got a server that becomes unusable after 1-3 days. I can't find anything consistent other than I have noticed a relatively high occurrence of SIP channels remaining open after the remote end has release the call. I'm trying to determine if that's a symptom or a cause. |
| 04:47.31 | WIMPy | Asterisk channels or sip channels? |
| 04:48.19 | mattsl | SIP |
| 04:48.36 | WIMPy | How many? |
| 04:48.47 | mattsl | not enough that I would think they would matter, 3 or 4... |
| 04:49.02 | mattsl | And I see no significant change in CPU usage |
| 04:49.10 | WIMPy | That sounds perfectely normal. |
| 04:50.57 | mattsl | Yeah. I figured that they weren't really an issue. But the server becomes so unresponsive that I can't even get it to let me ssh to it without several tries. |
| 04:51.15 | mattsl | That was the only thing different on it than I could see from any other server I have running smoothly |
| 04:51.36 | WIMPy | So that might be completely unrelated to Asterisk. |
| 04:52.01 | mattsl | quite possibly |
| 04:52.08 | *** join/#asterisk Lexus45 (~lexus@89.237.49.94) |
| 04:53.17 | mattsl | Seeing as how it's the only thing running on the box and I saw at least one thing that was out of the ordinary with it, I started there. I'm thinking the stuck channels though are just a symptom of it dropping packets |
| 04:53.40 | mattsl | But I'm left wondering what else I could possibly troubleshoot... |
| 04:53.51 | WIMPy | Likely, yes. |
| 04:54.12 | mattsl | (It does have Zabbix, but that was added after the problem to try to get some sort of info about machine state when it occurred) |
| 04:54.42 | *** join/#asterisk timahvo1 (~rogue@196.216.71.122) |
| 04:54.48 | WIMPy | Login in and stay connected until it happens. |
| 04:55.08 | mattsl | like I said, that can be days... |
| 04:55.30 | mattsl | And I don't know what I'd be looking for |
| 04:56.17 | WIMPy | You can try to dump to process list to files every few minutes in the hope to find something. |
| 04:58.23 | mattsl | Yeah. I've been able to get into it when it was in the degraded state a couple times and saw nothing in top |
| 04:58.34 | mattsl | I mean the CPU usage never exceeded 10% |
| 04:58.50 | mattsl | It's clearly something blocking, but I have no idea what |
| 04:59.22 | WIMPy | Maybe the network interface itself? |
| 05:00.25 | mattsl | That's what I was thinking. Could there be a problem with the interface on the software side? It seems like I'm able to hit it over the VPN tunnel for several minutes longer than through it's actual IP |
| 05:00.47 | mattsl | But that tunnel is on the same physical interface to which I can't connect |
| 05:01.29 | WIMPy | That definitely looks like the way to investigate. |
| 05:03.04 | mattsl | I've got it in production currently, so I'm about to just pull it, replace it entirely, and then let it run for days until it dies again. I'm just concerned that it might be something crazy unrelated to the box itself. |
| 05:04.32 | mattsl | Or that without active calls going through it it won't ever actually lock up. |
| 05:04.48 | WIMPy | Try to do more tests on the network connection, like e.g. different packet sizes. |
| 05:06.15 | WIMPy | But it's not a case of shooting yourself in the foot with something like fail2ban or so? |
| 05:07.23 | mattsl | how so? |
| 05:07.34 | mattsl | You mean with it actually blocking? |
| 05:07.43 | mattsl | Or with it just screwing up? |
| 05:07.53 | WIMPy | Blocking |
| 05:07.57 | mattsl | Nope |
| 05:08.32 | mattsl | It's currently configured to look backwards pretty far, so a reboot wouldn't help if that were it. |
| 05:08.41 | mattsl | (like a week) |
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| 05:37.40 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
| 05:39.16 | hebber | Using negative eventfilters in manager (AMI) doesn't seem to work - does someone know how to do the trick? |
| 05:40.13 | hebber | [TK]D-Fender: thanks for the tip using an Answer in Originate to avoid the unwanted CDR |
| 05:42.55 | [TK]D-Fender | hebber, You're welcome |
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| 06:45.24 | *** join/#asterisk RaptorJesus (~RaptorJes@unaffiliated/raptorjesus) |
| 06:45.42 | RaptorJesus | can anyone tell me why my asterisk is using 1.5gb of virtual mem? |
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| 06:49.17 | hebber | thats a lot |
| 06:51.21 | hebber | do you refer to VIRT or virtual memory in virtualization? |
| 06:54.10 | RaptorJesus | uhm |
| 06:54.11 | RaptorJesus | virt |
| 06:54.28 | RaptorJesus | and it keeps freezing |
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| 07:07.45 | RaptorJesus | ok it freezes when i turn off a phone |
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| 07:32.04 | RaptorJesus | ok yea wtf |
| 07:32.09 | RaptorJesus | i unplug a pjsip phone |
| 07:32.11 | RaptorJesus | server crashes |
| 07:32.40 | Chainsaw | It isn't "surprise removal" compliant. It's not a USB port... |
| 07:33.34 | RaptorJesus | older versions don't do this |
| 07:33.55 | Chainsaw | If you can make it occur at will, perhaps it's time to make a debug version explode so you can get a backtrace. |
| 07:34.34 | *** join/#asterisk bulkorok (~Benjamin@p5099b1e7.dip0.t-ipconnect.de) |
| 07:34.46 | Chainsaw | Backtrace turns into bug report, and then either you fix it or someone else does. And all will be well with the world. |
| 07:35.04 | RaptorJesus | how does I do such? |
| 07:35.30 | Chainsaw | Did you build your Asterisk from source or is it a distro package? |
| 07:35.57 | RaptorJesus | from source |
| 07:36.10 | RaptorJesus | with freepbx |
| 07:36.51 | Chainsaw | In menuselect, you need these options: DONT_OPTIMIZE DEBUG_THREADS BETTER_BACKTRACES |
| 07:37.26 | Chainsaw | Run ulimit -c unlimited before you start the Asterisk binary. You can process the resulting .core file with gdb. |
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| 08:54.25 | *** join/#asterisk areski (~areski@80.174.128.2.dyn.user.ono.com) |
| 08:55.23 | Tuju | nope, nope - fromdomain issue remains. |
| 08:56.26 | Tuju | http://www.google.fi/search?hl=fi&source=hp&q=asterisk+outbound+invite+fromdomain |
| 08:57.00 | Tuju | https://issues.asterisk.org/jira/browse/ASTERISK-20841 |
| 08:57.33 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 09:01.48 | *** part/#asterisk mattsl (~user@68.169.165.200) |
| 09:07.29 | *** join/#asterisk ganbold_ (~Ganbold@173.244.215.173) |
| 09:41.19 | stefan27 | Getting 19823 errors last day res_rtp_asterisk.c: PJ ICE Rx error status code: 370401 'Unauthorized'. |
| 09:41.58 | stefan27 | as well as 3 segmentations faults, many clients are using webrtc client with dtls encryption |
| 09:42.28 | stefan27 | Im trying to start the debugging process, I'll post a core-dump: |
| 09:42.29 | *** join/#asterisk zblk (~andrey@92.53.115.234) |
| 09:43.53 | stefan27 | Crash 1: http://pastebin.com/Z96fpCNJ looks dtls related |
| 09:45.58 | stefan27 | using chan_sip, not pjsip |
| 09:53.29 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 10:00.40 | h8 | hi everyone |
| 10:01.16 | h8 | what problema am I looking at if I'm trying to connect to an asterisk server behind nat, through a public ip but the registration doesn't get to asterisk yet I can make calls |
| 10:01.31 | h8 | sip show peers doesn't show the peer as being logged in, yet I See the outgoing call |
| 10:02.41 | rox | h8: you can make calls TO it, or FROM it |
| 10:02.52 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 10:03.00 | h8 | FROM it |
| 10:03.19 | h8 | so it's like the peer is connected, yet it doesn't show up and I can't call the peer |
| 10:07.54 | rox | h8: if you are behind a NAT, caling FROM it is not a problem, you need to configure your NAT to route SIP packets TO it |
| 10:08.59 | rox | h8: if this is the only asterisk behin this particular NAT, then you can probably just route all SIP traffic to it |
| 10:09.25 | rox | h8: is your router SIP aware? |
| 10:10.26 | h8 | the firewall is a pfsense, firewall of the asterisk server that is |
| 10:10.34 | h8 | it is the only asterisk, yes |
| 10:14.13 | h8 | I'm connected through a vpn to that network, through there, the registration works, it just doesn't work through the public ip |
| 10:21.15 | Tuju | h8: you might need to figure out how your fw handles port 5060 |
| 10:21.27 | Tuju | and how your phones behaves. |
| 10:21.48 | Tuju | some cisco's listen return packets in 5060 too, some models in high ports. |
| 10:22.07 | Tuju | former are very hard to get to work. |
| 10:22.51 | Tuju | some phones allow you to set in config that it's behind nat. |
| 10:23.00 | Tuju | some allow setting public ip too. |
| 10:41.31 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
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| 10:56.49 | Tuju | i'm trying to rewrite To header in dialplan but have not been able to. |
| 10:57.27 | Tuju | exten => 2226,2,SipAddHeader(To: "${EXTEN}@example.com") should work, but it doesn't. |
| 10:57.47 | Tuju | that 'Add' part smells suspicious to me. |
| 10:58.03 | Tuju | i need to overwrite, not add anything. |
| 11:06.53 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
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| 11:27.27 | *** part/#asterisk rox (~rox@BSN-142-168-6.static.siol.net) |
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| 12:49.34 | tparcina | Today users are having small periods of time (second, maybe two) when they don't hear the other person (or the other person doesn't hear them). |
| 12:50.10 | tparcina | We record all phone calls, and when I listen for the phone call, I can hear both speakers fine. |
| 12:50.57 | tparcina | I'm checking the network usage, and it's under 100 Mbit (tipicly 20-30 Mbit) on 1 Gbit network. |
| 12:51.10 | tparcina | What could be the problem? |
| 12:57.53 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 13:00.56 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 13:01.21 | Tuju | they hesitate to talk knowing that everything is recorded? :) |
| 13:01.50 | Tuju | i least it would make me stutter a bit. |
| 13:07.54 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:08.15 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |
| 13:08.41 | *** join/#asterisk roramirez (~rodrigo@186.35.206.182) |
| 13:09.26 | roramirez | Hello, exist any way to compiling a specific module in Asterisk source?, i have source 1.8.25 |
| 13:16.26 | *** join/#asterisk [[thufir]] (~thufir@96.48.128.162) |
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| 13:36.52 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:45.02 | RobertLaptop | Anyone know when RHEL 7 RPM's are planed for Asterisk 13? |
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| 14:16.54 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:17.51 | Tuju | [TK]D-Fender: i had to hack today to make fromdomain to work. |
| 14:18.45 | Tuju | i found quite many blogs like this http://mikepultz.com/2009/04/handling-sip-uri-dialing-in-asterisk/ someone else have fought the same issue. |
| 14:20.16 | Tuju | nah, it broke again. it doesn't work. |
| 14:20.54 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 14:21.36 | *** join/#asterisk crised (~crised@186.67.181.203) |
| 14:22.33 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:23.12 | Tuju | if i do sip reload, it works for a while and after half an hour, it fails again. |
| 14:23.37 | crised | [TK]D-Fender: Hello there, I'm having problems with the SIP phone. Outgoing calls goes ok, but DID incoming not. I changed a firewall setting (I'm using freebsd, similar to pfsense): (not to change the UDP source port), and it could receive DID inbound calls, but problem is that started to ring in the middle of the night, just because the telephone wanted to. |
| 14:23.46 | crised | no one was ringing it. |
| 14:24.12 | [TK]D-Fender | You're probably being scanned |
| 14:24.40 | crised | [TK]D-Fender: how's that |
| 14:25.04 | [TK]D-Fender | some other hacker is looking for server's to dial out and your phone is accepting their call |
| 14:25.24 | [TK]D-Fender | You're just reg'ing a Polycom to some outside provier right? |
| 14:25.30 | [TK]D-Fender | provider* |
| 14:25.34 | crised | [TK]D-Fender: to voip.ms only |
| 14:26.03 | [TK]D-Fender | crised: There is a provisioning option to lock out calls from sourcees you have not registered to. |
| 14:26.29 | [TK]D-Fender | crised: But IIRC you stopped doing it the way I told you and you started doing it directly on the phone itself |
| 14:26.40 | crised | [TK]D-Fender: so to only accept phone calls coming from voip.ms? |
| 14:26.48 | [TK]D-Fender | yes |
| 14:26.49 | crised | [TK]D-Fender: yes, lol |
| 14:27.03 | crised | [TK]D-Fender: that would make lot of sense |
| 14:27.11 | [TK]D-Fender | Or you could try to just lock down SIP at your firewall itself |
| 14:27.35 | crised | [TK]D-Fender: what if I change the rule of static nat port? |
| 14:27.59 | [TK]D-Fender | that's what I said. restrict the forwarding range |
| 14:28.09 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-cyjvzaysjitxoiug) |
| 14:28.30 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 14:28.52 | crised | I modified this rule: nat on $ext_if from $localnet to any -> ($ext_if) static-port and added these two: rdr pass on $ext_if proto udp from any to any port 5060 -> 192.168.1.69 |
| 14:28.52 | crised | rdr pass on $ext_if proto udp from any to any port 5080 -> 192.168.1.69 |
| 14:28.52 | crised | <PROTECTED> |
| 14:29.05 | crised | I think the first rule, made all the difference |
| 14:29.17 | crised | I added static-port at the end |
| 14:29.47 | Tuju | crised: you need to make another context where your phones are. |
| 14:29.48 | crised | "static-port Withnat rules, the static-port option prevents pf(4) from modify- ing the source port on TCP and UDP packets." |
| 14:30.27 | crised | Tuju: I only have one phone... |
| 14:30.35 | *** part/#asterisk mjordan (~mjordan@75.76.55.191) |
| 14:30.47 | Tuju | okay, well that's probably then okay to listen it ringing over night. |
| 14:31.10 | crised | Tuju: how do I do that? |
| 14:31.40 | Tuju | you need to have more than one [name] sections in extensions and context=name in peer settings. |
| 14:31.56 | crised | Tuju: I don't follow at all |
| 14:32.02 | crised | Tuju: are you talking on the phone? |
| 14:32.12 | Tuju | ? |
| 14:32.29 | crised | Tuju: are you talking about firewall config or phone config? |
| 14:32.45 | Tuju | this is asterisk channel, i'm talking about asterisk configuration. |
| 14:32.57 | crised | Tuju: I don't have asterisk |
| 14:33.08 | Tuju | well, my bad then. |
| 14:33.08 | crised | Tuju: asterisk is inside my voip provider |
| 14:33.12 | crised | Tuju: :) |
| 14:33.38 | *** join/#asterisk Cust0sL1men (~CustosLim@unaffiliated/cust0slim3n) |
| 14:33.49 | Tuju | one moment, we hack into their systems and fix it. |
| 14:34.07 | crised | [TK]D-Fender: so what do you suggest? |
| 14:39.30 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 14:40.13 | [TK]D-Fender | crised: I already gave you 2 ways. take your pick. |
| 14:41.05 | crised | [TK]D-Fender: Can I do the config of lock out calls directly from the phone? |
| 14:41.51 | [TK]D-Fender | [09:26][TK]D-Fendercrised: There is a provisioning option to lock out calls from sourcees you have not registered to. <- |
| 14:41.55 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:42.04 | [TK]D-Fender | This is not the sort of thing I think they give you direct on the interface |
| 14:42.35 | crised | [TK]D-Fender: ok, so you advice to restric this rule? Konsole output rdr pass on $ext_if proto udp from any to any port 5060 -> 192.168.1.69 |
| 14:43.34 | *** join/#asterisk sgriepentrog (~sgriepent@2602:30a:2ef2:840:20c:29ff:fe4f:cac8) |
| 14:43.54 | [TK]D-Fender | Firewall them out or tell your phone to ignore them |
| 14:43.57 | [TK]D-Fender | Take your pick. |
| 14:44.53 | *** join/#asterisk robscow (~robsco@wimbledon.brainboxdigital.com) |
| 14:45.10 | crised | [TK]D-Fender: I prefer firewall, how would the rule look like? I'm onnecting to server tampa.voip.ms, this means only THIS server can access my phone? |
| 14:46.07 | robscow | I have transfers enabled in features.conf, when people press *2. however, when we make calls to outside numbers, like a conference, and they ask for a pin, the moment we enter the code and press #, I heard "Transfer... sorry, that's not a valid extension" from our own Asterisk system. is there something common/stupid Ive done to the configs? |
| 14:46.32 | robscow | for external calls I simply have this... |
| 14:46.47 | dan_j | Hi. As anyone seen a cheap SIP based loud ringer which is compatible with asterisk? Maybe a raspberry PI or arduino project if that works out cheaper. |
| 14:46.54 | dan_j | Has* |
| 14:46.55 | robscow | exten => _0[12378]XXXXXXXXX,1,Set(CALLERID(num)=20310) same => n,Dial(SIP/${EXTEN}@spitfire,0,Tg) same => n,Hangup |
| 14:49.14 | [TK]D-Fender | crised: That's what I said. |
| 14:49.39 | [TK]D-Fender | robscow: What kind of phones are you using? |
| 14:49.47 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 14:51.04 | robscow | panasonic sip phones, KX-TGP500 to be precise |
| 14:53.07 | [TK]D-Fender | Those should support native transfers on their own without *-based DTMF features for this |
| 14:55.29 | robscow | I'm not seeing anything in the configs for the phones |
| 14:57.16 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 14:58.07 | crised | [TK]D-Fender: How does this looks? rdr pass on $ext_if proto udp from tampa.voip.ms to any port 5060 -> 192.168.1.69 |
| 14:59.27 | [TK]D-Fender | crised: I have no idea how your firewall works. You'll have to figure that out for yourself |
| 14:59.57 | [TK]D-Fender | RobWon't be in configs. Typically it's fixed functionaly. For ATA like devices it's usually hook-flash based |
| 15:00.59 | robscow | ahh ok, so maybe i should remove the asterisk capability? and allow the handsets to do it themselves? |
| 15:02.30 | [TK]D-Fender | yes |
| 15:02.35 | *** join/#asterisk ghostlines (~ghostline@82.201.93.50) |
| 15:11.34 | crised | [TK]D-Fender: how can I make an anonymous call to see if my firewall settings are OK, now? |
| 15:13.16 | crised | [TK]D-Fender: how these hackers scanned my ip? |
| 15:13.29 | [TK]D-Fender | crised: They threw a call at you. Your mphone accepted |
| 15:13.46 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:14.06 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 15:14.24 | crised | [TK]D-Fender: Why me? |
| 15:14.34 | crised | They are sniffing my ISP traffic or something? |
| 15:15.02 | crised | [TK]D-Fender: How can I make an anonymous call to see if my firewall blocks these? |
| 15:15.08 | [TK]D-Fender | they are scanning THE WHOLE INTERNET |
| 15:15.23 | [TK]D-Fender | Get some machine to throw a call at you. |
| 15:16.05 | crised | wow |
| 15:16.10 | Tuju | [TK]D-Fender: what was the logic that is used to map incoming calls to defined trunks? |
| 15:16.24 | Tuju | you mentioned something about the order how they're defined. |
| 15:16.31 | [TK]D-Fender | Tuju: type= <--- |
| 15:16.55 | [TK]D-Fender | peer = by IP, user = username, friend = both |
| 15:16.55 | Tuju | [TK]D-Fender: and if i've got more than one 'friend' ? |
| 15:17.11 | crised | How can I make anonymos call? |
| 15:17.12 | [TK]D-Fender | first one it finds that matches gets it |
| 15:17.23 | Tuju | hmmm....maybe i should put user then. |
| 15:17.28 | [TK]D-Fender | crised: Pick a SIP client and have it call out to your IP |
| 15:19.08 | crised | [TK]D-Fender: sighs... how can I do this online easily? |
| 15:19.37 | [TK]D-Fender | crised: Go find some service or somebody to do the test. Or remote control some other system on the outside |
| 15:20.41 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 15:22.25 | crised | [TK]D-Fender: can you do this test? |
| 15:22.54 | [TK]D-Fender | Not currently |
| 15:23.09 | crised | [TK]D-Fender: ok, I'll give it sometime |
| 15:23.11 | crised | thanks |
| 15:23.12 | *** join/#asterisk Xaviertoor (~jerson.ju@177.99.205.154) |
| 15:23.15 | *** join/#asterisk mjordan (mjordan@nat/digium/x-rwcvvkofjwysvkaf) |
| 15:23.15 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:24.55 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 15:25.41 | Tuju | [TK]D-Fender: I've two asterisk hosts, there are two trunks in between. So i cannot use ip, right? |
| 15:26.04 | Tuju | [TK]D-Fender: so either friend or user? |
| 15:26.05 | [TK]D-Fender | Why 2? |
| 15:26.05 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 15:26.15 | Tuju | [TK]D-Fender: two different context |
| 15:26.23 | Tuju | two organizations. |
| 15:26.37 | [TK]D-Fender | then make user sections to match |
| 15:26.59 | Tuju | if i put type=user, sip show peers doesn't list them anymore and registration fails. |
| 15:27.02 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 15:27.42 | [TK]D-Fender | sip show users <-------- |
| 15:27.44 | Tuju | if i have type=firend, those trunks get mixed. |
| 15:28.07 | [TK]D-Fender | And if you know the IP of the opther box you shouldn't be registering anyway |
| 15:28.37 | Tuju | okay, they are listed in users, right. |
| 15:29.14 | Tuju | can i still have two trunks if i don't register? |
| 15:29.49 | yun1989 | hello all |
| 15:30.12 | [TK]D-Fender | Tuju: You can have as many as you want |
| 15:30.13 | Tuju | hi yun1989 |
| 15:30.25 | Tuju | [TK]D-Fender: ack, good. |
| 15:30.44 | yun1989 | someone know if is possible to connect CP 352 to asterisk ? |
| 15:30.47 | Tuju | so i should drop register => lines ? |
| 15:30.52 | yun1989 | http://en.intelbras.com.br/business/intercoms/condominiums/stations/cp-352 |
| 15:31.46 | *** join/#asterisk kharwell (kharwell@nat/digium/x-qbfcuhelomibfmrl) |
| 15:32.03 | [TK]D-Fender | yun1989: What does it speak? |
| 15:33.44 | yun1989 | @[TK]D-Fender hello |
| 15:34.16 | yun1989 | it's possible to connect the condominums analog system to asterisk ? |
| 15:34.24 | yun1989 | in this case cp352 |
| 15:35.08 | [TK]D-Fender | yun1989: I just asked you a very direct question. What does it SPEAK? |
| 15:35.45 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 15:35.50 | Tuju | hmmm... it appears to work, this is scary. |
| 15:36.15 | Tuju | [TK]D-Fender: maybe you two don't speak common protocol :) |
| 15:36.29 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 15:38.03 | [TK]D-Fender | yun1989: That thing appears to be analog FXS/FXO so you could get an interface for * to speak to it |
| 15:38.10 | yun1989 | http://en.wikipedia.org/wiki/Analog_telephone_adapter |
| 15:38.41 | Tuju | yun1989: cisco has ATA in SPA series. |
| 15:38.42 | yun1989 | yes i need the gateway beteween the IP network to the condominuims network |
| 15:38.55 | [TK]D-Fender | So go get an ATA |
| 15:39.29 | yun1989 | but i think the problem is in cp352 don't support ATA |
| 15:39.40 | [TK]D-Fender | if you need to use a PSTN port on the CP for connectivity, or a SIP>FXO gateway if you need * to act like a phone on it |
| 15:39.50 | yun1989 | i need understand more this case |
| 15:39.56 | [TK]D-Fender | Go read its manual |
| 15:40.00 | [TK]D-Fender | We don't use that here |
| 15:40.28 | Tuju | yun1989: ATA has plain analong telephone port in it, it does the ip.network <-----> analog conversion. |
| 15:40.50 | Tuju | those cost few tens dollars. |
| 15:41.03 | Tuju | i've got one here. |
| 15:41.43 | Tuju | http://www.cisco.com/c/en/us/products/unified-communications/spa122-ata-router/index.html |
| 15:42.17 | Tuju | "2 voice channel ATA with router" |
| 15:43.28 | [TK]D-Fender | That is if you need to take a PSTN port on the CP352 for this |
| 15:43.40 | [TK]D-Fender | You'll need a different device if you need * to act like a phone to it |
| 15:46.18 | yun1989 | thanks |
| 15:47.57 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-104-77.dynamic.qsc.de) |
| 15:52.14 | Tuju | [TK]D-Fender: you're probably only here who understands * in deep guts. |
| 15:53.10 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:248:82fa:5bff:fe0a:dfef) |
| 15:53.23 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-wolwuqbfejxwxzts) |
| 15:53.34 | [TK]D-Fender | No, there are many here who understand far more than me. |
| 15:53.41 | [TK]D-Fender | I don't get into the actual code. |
| 15:53.59 | [TK]D-Fender | I do get the premise it's built around and how pieces interact in order to determine a path |
| 15:54.05 | [TK]D-Fender | That's just linear thinking. |
| 16:00.40 | coppice | That's better that thinking like a log |
| 16:01.16 | [TK]D-Fender | mmmm log |
| 16:01.42 | coppice | or thinking like a square |
| 16:10.53 | file | moo |
| 16:15.15 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 16:16.11 | pjensen00 | thinking^2 |
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| 16:34.32 | *** join/#asterisk human0x27 (~smk@pit11.pair.com) |
| 16:37.55 | human0x27 | Hello all, any jabber users? We use jabber to send call information to call center employees. I recently upgraded from 1.8.8.8 to 1.8.9.3. I'm getting the following errors around every 2 minutes |
| 16:37.59 | human0x27 | http://pastebin.com/raw.php?i=Fzng48Lg |
| 16:38.36 | human0x27 | (debugging was enabled when I captured that output) |
| 16:39.27 | Qwell | Is there a newline after the <? That's crazy. It should totally still work, but wat? |
| 16:39.55 | human0x27 | yeah.. that confused me as well.. but it was happening elsewhere without causing an error. |
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| 16:47.43 | mjordan | the fact that it is complaining about invalid XML is a sign that something isn't right with the world |
| 16:49.10 | Qwell | mjordan: libiksemel is pretty simple. I could easily see a random newline breaking things. |
| 16:58.49 | mjordan | yup, but it isn't wrong at that point. That is invalid XML :-) |
| 17:03.01 | Qwell | huh, xmlvalidation.com agrees. I assumed spacing didn't matter at all. |
| 17:03.24 | *** part/#asterisk robscow (~robsco@wimbledon.brainboxdigital.com) |
| 17:06.19 | Qwell | '<' name (whitespace attribute)* whitespace? '>' |
| 17:07.41 | human0x27 | so, the newline is the issue? |
| 17:08.50 | Qwell | assuming it's actually there, and something else isn't happening to make it bail early. |
| 17:09.00 | human0x27 | hm.. interesting. |
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| 17:24.43 | *** join/#asterisk RPerre (~rperre@alpes.nortenet.pt) |
| 17:26.35 | RPerre | Hello there. I'm using ARI and I'm trying to record a bridge. I make a POST /bridges/{bridgeId}/record and get a positive result. But when i try to stop and save (using POST /recordings/live/{recordingName}/stop) I get 404: Recording not found. Any help? thanks! |
| 17:28.03 | *** join/#asterisk jamicque (~jamicque@89-71-40-164.dynamic.chello.pl) |
| 17:28.57 | jamicque | Hi, does anyone know if asterisk java supports AMI v2 ins Asterisk 12? Or is there any other java lib that works with AMIv2? |
| 17:36.10 | mjordan | RPerre: what are you using for the recordingName? And if you list all of the live recordings, does it list the one that you started on the bridge? |
| 17:37.49 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 17:38.24 | RPerre | mjordan, I'm using random strings for the name. I think there is no method to list recordings taking place, only stored ones. |
| 17:42.33 | RPerre | mjordan, results: http://pastebin.com/JFhWFdWT |
| 17:42.58 | RPerre | i'm using curl for debug |
| 17:43.47 | *** join/#asterisk riess82 (~riessma@188-22-43-160.adsl.highway.telekom.at) |
| 17:44.16 | RPerre | on asterisk CLI: |
| 17:44.17 | RPerre | <PROTECTED> |
| 17:44.17 | RPerre | <PROTECTED> |
| 17:44.17 | RPerre | <PROTECTED> |
| 17:44.17 | RPerre | <PROTECTED> |
| 17:44.17 | RPerre | <PROTECTED> |
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| 17:59.12 | RPerre | I'm stuck on this recording stuff :/ |
| 18:19.43 | pabelanger | might be the first person to do something with it |
| 18:24.05 | RPerre | pabelanger really? :o |
| 18:24.25 | RPerre | ARI seems so elegant |
| 18:24.32 | pabelanger | new hotness |
| 18:24.33 | RPerre | well...it IS elegant |
| 18:24.46 | pabelanger | only know of a handful of people using it ATM |
| 18:24.48 | pabelanger | might you |
| 18:24.57 | pabelanger | more could be using, but not saying anything |
| 18:25.08 | RPerre | pabelanger, so you think this might me a bug? |
| 18:25.15 | pabelanger | *shurgs* |
| 18:25.28 | pabelanger | I'd check the testsuite to see if there is coverage for what you are doing |
| 18:25.44 | pabelanger | then compare |
| 18:25.50 | RPerre | pabelanger, testsuite? i dont know what ur talking about :/ |
| 18:27.00 | pabelanger | http://svn.asterisk.org/svn/testsuite/asterisk/trunk/tests/rest_api/ |
| 18:27.15 | pabelanger | there are some recording tests |
| 18:27.21 | pabelanger | so, you could see what they are doing |
| 18:30.36 | RPerre | thanks pabelanger! |
| 18:36.15 | RPerre | pabelanger, well it seems i'm doing things the right way. but getting same problem *facepalm* |
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| 19:12.02 | matthew-moretalk | anyone any ideas how I can set the Content-Type header for CURL Function Post Requests? |
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| 19:17.28 | mbowie | RobertLaptop: My scroll-back was too short (now fixed) but I see I had a mention... did you have any luck on the UI front? |
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| 19:28.29 | KNERD | is there a way to activate a feature code from the * CLI? |
| 19:39.53 | ndb | hi, is it possible to launch a thread with a socket binded without blocking from an asterisk module? |
| 19:40.13 | ndb | my plan is to connect on this socket and send audio |
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| 20:05.49 | *** join/#asterisk killfill (~killfill@190.107.178.50) |
| 20:05.54 | killfill | hi guys |
| 20:06.24 | killfill | im trying to connect a fanvil phone to my asterisk , using IAX. this is what i see in the logs: http://pastebin.com/raw.php?i=W27mdYFF |
| 20:06.35 | killfill | there is no movment after that. |
| 20:07.47 | killfill | this is the config: http://pastebin.com/raw.php?i=YCSwzx9w |
| 20:07.59 | killfill | is there any obvious thing im missing?.. |
| 20:12.37 | doop | why are you denying then permitting the same thing |
| 20:13.48 | killfill | yeah, i copy pasted it from an already created extension.. the permitting seems to win, as i can connect just fine with zoiper tho |
| 20:14.15 | doop | it's silly. take out the deny. |
| 20:14.21 | killfill | ok |
| 20:15.46 | killfill | doop: doesnt resolve the problem tho |
| 20:16.37 | doop | have you tried some other piece of hardware or softphone with the same credentials |
| 20:17.08 | killfill | yup, with zoiper works great |
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| 20:17.19 | doop | then the problem is your fanvil phone |
| 20:17.55 | killfill | yah, i would thing so too, but this is the third one i test, and this product says it supports iax.. |
| 20:18.16 | killfill | i know this is a chinees phone, so i dont expect the bet quality product.. but at least it should connect |
| 20:18.18 | doop | it may be a problem with the design of the phone |
| 20:18.38 | doop | expect it to not work until it proves otherwise |
| 20:18.40 | killfill | tested 2 diff models |
| 20:18.54 | doop | fwiw i've never even heard of fanvil |
| 20:19.00 | doop | until now |
| 20:19.10 | mbowie | It's only one "f" away from being an anvil |
| 20:19.13 | killfill | me neither.. its the only iax phone i could buy in here.. :P |
| 20:19.18 | killfill | heh |
| 20:19.26 | doop | why did you need it to be iax |
| 20:20.31 | killfill | because i got multiple vpns connecting to the same asterisk |
| 20:20.46 | killfill | and want to bypass the horror nat story of sip.. :) |
| 20:21.05 | doop | explain |
| 20:21.52 | Qwell | killfill: IAX, or IAX2? |
| 20:21.52 | killfill | as far as i know, to enable SIP on different networks (or behind a nat) i would need to setup a sip-proxy, right? |
| 20:22.08 | killfill | Qwell: IAX2 |
| 20:22.21 | doop | killfill which part of the system is behind NAT exactly |
| 20:22.28 | Qwell | Don't most of those phones also support SIP? |
| 20:23.02 | killfill | doop: actually none. but the traffic flows thourhgt different networks. (different vpns) |
| 20:23.38 | killfill | doop: when i connect them throught sip, only 1 party hears, not the other one. i think its the same problem as with nat, when your not passing thorght a sipproxy (?) |
| 20:23.45 | killfill | Qwell: i wanted to avoid SIP |
| 20:24.00 | Qwell | Why? |
| 20:26.03 | killfill | Qwell: could not make sip work. when establishing a call, one end doesnt hear the other one. |
| 20:26.12 | Qwell | So configure things properly. |
| 20:26.51 | killfill | trying to. i thought iax would be another way to do it too. |
| 20:28.20 | killfill | you guys think iax2 is no way to go? |
| 20:30.22 | doop | i think designing your network appropriately is the way to go |
| 20:32.06 | killfill | oh i would not expect that answear |
| 20:34.56 | doop | is any of your traffic running through a firewall? |
| 20:35.03 | doop | that could also be the problem |
| 20:36.34 | doop | which party can hear and which party cannot? |
| 20:36.36 | killfill | doop: yup, its running throught several firewalls, but the udp port is open , i have connectivity |
| 20:36.52 | Qwell | The UDP port? |
| 20:36.52 | doop | more than one udp port |
| 20:38.06 | killfill | yup the udp port |
| 20:38.16 | doop | which udp port did you open |
| 20:40.44 | killfill | 4569 |
| 20:40.52 | doop | no, for SIP |
| 20:42.20 | killfill | aah |
| 20:42.32 | killfill | 5060 tcp+udp, and 10.000 - 20.000 |
| 20:42.55 | doop | i hope you didn't specify those with the decimal point |
| 20:43.43 | killfill | nope.. :P |
| 20:43.50 | doop | which party can hear and which party cannot? |
| 20:44.00 | killfill | i added them for knowing how much 0 i needed to write in here.. :P |
| 20:45.49 | killfill | i just re-test the setup, none of the parties hear the other one |
| 20:46.12 | doop | well that's not exactly the problem you described earlier now is it |
| 20:48.44 | killfill | yeah i know. im sorry. i probably got confused, did much tests |
| 20:51.07 | doop | what happens when the box and the phone are on the same network without any firewall between them |
| 20:53.23 | killfill | i didnt test it, but it should work.. i mean, i got cisco phones working here just fine. |
| 20:53.37 | doop | let's test it to make sure your phone works |
| 20:54.06 | doop | but i'm seriously thinking your danfil phones are crap |
| 20:54.13 | doop | or whatever name |
| 20:54.20 | doop | fanvil |
| 20:57.58 | killfill | yeah, damn phone.. i would need to get it tomorrow at the other office |
| 20:58.17 | doop | you're asking for help but you don't even have the phone available for testing? |
| 20:58.44 | killfill | yes i have it for testing, just remotely |
| 20:58.52 | doop | blinking |
| 20:58.55 | doop | ok |
| 20:58.56 | doop | so |
| 20:59.22 | doop | with all your VPNs and firewalls, presumably you can put the phone and the box on the same virtual network |
| 20:59.51 | killfill | i guess i was naive to think iax would just work from a different network as the asterisk is in... |
| 21:00.02 | doop | iax works fine |
| 21:00.11 | doop | it's the phone that appears to be cocked up |
| 21:01.17 | doop | your naïveté perhaps comes more from expecting a crap phone to implement the protocol correctly |
| 21:01.18 | killfill | damn phone.. i dont find similar problems on the internet either |
| 21:01.34 | killfill | heh |
| 21:03.06 | killfill | iax2 is just 1 port right? |
| 21:03.13 | doop | correct |
| 21:03.34 | doop | and your iax2 softphone works fine from the same network, yes? |
| 21:03.49 | doop | (the same network as the fanvil phone) |
| 21:04.08 | killfill | yup, your comming aobut my naive-ness is correct. its not phone not the protocol. sorry. |
| 21:04.15 | killfill | zoiper works fine |
| 21:04.28 | killfill | s/comming/comment |
| 21:04.54 | killfill | aah.. 'its the phone not the protocol' sorry for the typos. |
| 21:04.59 | doop | so if zoiper works fine from that network then in all likelihood your hardphone is not working properly |
| 21:05.30 | killfill | yeah, i thought maybe there were additional iax config trick i could set in asterisk to make this work |
| 21:05.32 | doop | test it on the same network and if you still have the same trouble contact the manufacturer or send it back |
| 21:06.42 | killfill | yeah. |
| 21:15.20 | killfill | so when peoapl talks aobut "iax2" there is only 1 version of the protocol right? |
| 21:15.25 | killfill | no iax2.1 or such |
| 21:15.42 | doop | afaik |
| 21:30.55 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 21:40.46 | WIMPy | Oh. You have an IAX hardware phone? |
| 21:46.43 | *** join/#asterisk igeek_88 (~iweinberg@26-139-89-200.fibertel.com.ar) |
| 21:47.23 | igeek_88 | Hello everyone. I need a program that will allow me to convert a wma file to WAV PCM Uncompressed' 8000hz 16bit mono in order to use it as an announcement on Asterisk freepbx. |
| 21:48.10 | WIMPy | sox |
| 21:48.32 | WIMPy | and asking the same question to google will give you a lot of examples on how to use it. |
| 21:49.06 | malcolmd | i like how that same question was asked in #freepbx also |
| 21:49.38 | igeek_88 | Yup. I've asked it myself hehe. Sorry. |
| 21:49.53 | igeek_88 | Thanks. |
| 21:49.57 | malcolmd | np |
| 21:55.09 | *** join/#asterisk [[thufir]] (~thufir@96.48.128.162) |
| 21:58.47 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:24.12 | *** join/#asterisk mbowie (~mbowie@162.212.36.9) |
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| 22:27.24 | [[thufir]] | when you "include" for a dialplan: https://wiki.asterisk.org/wiki/display/AST/Include+Statements+Basics what's getting included? is it a file which gets appended during run time? |
| 22:28.49 | [TK]D-Fender | No, CONTEXTS do |
| 22:41.49 | [TK]D-Fender | [[thufir]], Calls landing in [users] can also dial the things contained in that included context |
| 22:42.41 | [[thufir]] | [TK]D-Fender: what I mean, pardon, is what's a context? physically. it's a script file? |
| 22:42.59 | [TK]D-Fender | [users] <- THAT is a context |
| 22:43.07 | [TK]D-Fender | ist is a logical separation in your dialplan |
| 22:43.12 | [[thufir]] | ok. |
| 22:43.33 | [TK]D-Fender | I want Joh to be able to dial certain things, but I don't want Mark to be able to dial those |
| 22:43.37 | [TK]D-Fender | John* |
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| 23:09.48 | RobertLaptop | <mbowie> https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Asterisk+REST+API |
| 23:10.03 | RobertLaptop | mbowie ^^^^ |
| 23:12.04 | mbowie | RobertLaptop: Thanks for the replay... I did see that. We're on v11 for now, but we'll factor this into our upgrade motivation. ;-) |
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| 23:22.44 | *** join/#asterisk fornax (~fornax@85.183.43.221) |
| 23:25.12 | roler | I have an Elastix NLX4000 box with a Sangoma T1 card. It also has 4 fxo and 1 fxs port. Right now the system is working great with the pstn lines. We just had a T1 PRI installed and the cutover date is Monday. I can't get the system to receive or make calls, and our ISP says there are errors on the T1. How can I debug these errors on the command line to see what is going on? |
| 23:27.40 | *** part/#asterisk mjordan (mjordan@nat/digium/x-rwcvvkofjwysvkaf) |
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| 23:30.43 | *** join/#asterisk Trevizoli (bd2fd731@gateway/web/freenode/ip.189.47.215.49) |
| 23:32.29 | moy | @roler you might want to do 'wanpipemon -i w1g1 -c Ta', assuming w1g1 is your T1 interface. You can check what other interfaces you have using 'ifconfig' |
| 23:32.33 | moy | then read this: http://wiki.sangoma.com/wanpipemon-T1-E1-line-alarms-0 |
| 23:33.22 | Trevizoli | Hi friends! Could someone explain me what means a message channel.c "Prodding Channel 'SIP/xxxxxx-00000000' failed". I searched in source code and find only that is something to make thinks moving...but that have failed... |
| 23:34.04 | roler | moy; thanks. I'll read that URL. I have 2215 line code violations :) |
| 23:34.40 | moy | @roler I think some are normal on startup, as long as they are not constantly increasing |
| 23:36.47 | roler | ok.. going up stairs to try once more. |
| 23:40.21 | *** join/#asterisk roler (~roler@unaffiliated/roler) |
| 23:40.53 | roler | moy; all of my alarms are off. Rx level is >-2.5 |
| 23:42.16 | WIMPy | roler: 'dahdi show status' |
| 23:42.46 | roler | WIMPy; that looks okay too |
| 23:43.04 | WIMPy | And what doesn't ? |
| 23:43.35 | roler | WIMPy; alarms OK, IRQ 0, bpviol 0, CRC 0, Fra ESF, Codi B8ZS, LBO 0 db (CSU)/0-133 feet (DSX-1) |
| 23:43.45 | roler | thats the wanpipe1 card |
| 23:44.29 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 23:45.37 | roler | can I check anything else? :) |
| 23:45.55 | WIMPy | Make a call |
| 23:46.24 | roler | Ok. that sounds like a plan |
| 23:46.27 | roler | thanks guys |
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| 23:50.32 | [[thufir]] | does "telnet 192.168.1.2 5060" show me whether the asterisk server at 192.168.1.2 is accepting connections on port 5060? or that's done with netcat? |
| 23:50.34 | *** part/#asterisk Trevizoli (bd2fd731@gateway/web/freenode/ip.189.47.215.49) |
| 23:51.00 | WIMPy | SIP usually uses UDP |
| 23:51.07 | moy | [[thufir]]: I guess that depends on whether you have configured your sip driver for tcp |
| 23:51.21 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 23:51.29 | [[thufir]] | ok. |
| 23:51.30 | moy | [[thufir]]: you can use sipsak for a quick 'sip' ping over udp |
| 23:51.41 | moy | err 'ping' |
| 23:52.09 | [[thufir]] | ahhhh, thank you. wow, I've been trying to do that for a bit! |