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| 01:35.59 | Kobaz | anyone know if it's possible to push a keypress to a polycom phone |
| 01:36.11 | Kobaz | not a link to a keypress, but an unattended keypress |
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| 02:33.08 | newzen | http://pastebin.centos.org/2320/ now i can go out side but in strange way. I putted a phone on fxs and if i call to his extension i get tone and could call outside phone |
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| 03:16.34 | igcewieling | newzen: please go to #FreePBX for FreePBX issues |
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| 03:19.47 | tuxbiker | Is there a way to see if jack support is enabled? Version is 1.6 if that makes a difference. |
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| 04:01.20 | igcewieling | tuxbiker: what is the asterisk module name which supports that? |
| 04:03.09 | igcewieling | "module show like X" where X is part of the module name, such as jack |
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| 04:15.41 | tuxbiker | awesome, thank you very much |
| 04:17.48 | tuxbiker | it's 'app_jack' but I couldn't find any references to it in the rpm file. |
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| 12:07.02 | emk | Hey anyone here know how to send a multiline SMS with DongleSendSMS()? |
| 12:07.39 | eirirs | emk: tried with /r/n ? |
| 12:07.49 | eirirs | eventually only /n |
| 12:08.10 | emk | eirirs: yep, the issue is that only the text beforethe newline gets sent. |
| 12:08.25 | eirirs | oh |
| 12:09.14 | eirirs | emk: you have whole message in a variable instead of putting it directly in the function parameter? |
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| 12:11.33 | emk | eirirs: I'm using call files. Generating them dynamically using a python script. I've tried both actual newlines and a string containg "slash+plus+n" etc |
| 12:12.09 | eirirs | emk: seen those? http://en.wikipedia.org/wiki/GSM_03.38 |
| 12:12.24 | eirirs | maybe these 0x0A and 0x0D |
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| 12:17.59 | tparcina | When I connect to asterisk (asterisk -r) I have high verbose (I see every call). How can I lover verbose from asterisk CLI? |
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| 12:22.14 | niohubala | eirirs => core set verbose 2 |
| 12:22.18 | niohubala | ( https://wiki.asterisk.org/wiki/display/AST/Changing+the+Verbose+and+Debug+Levels ) |
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| 12:27.18 | msaraiva | Does anyone know of a solution to monitor current calls/channels? Asterbilling seems to have a bit of a problem with calls terminated by Playback() or Confbridge(). |
| 12:27.33 | igcewieling | perhaps y'all should read The Asterisk Book |
| 12:27.35 | igcewieling | ~book |
| 12:27.35 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 12:28.10 | keiths_ | I read that book once....now i work even less hard |
| 12:28.14 | eirirs | niohubala: I think it was tparcina who asked for that. |
| 12:28.48 | tparcina | niohubala: Thank you. |
| 12:29.00 | tparcina | eirirs: Yes, I have asked. Thank you. |
| 12:29.13 | niohubala | i'm sorry eirirs, I must have been inattentive |
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| 12:34.42 | niohubala | does anyone know how to disable SRTP? My phones don't support it, and I can 't make calls because of this. The warning message is "We are requesting SRTP, but they responded without it!" |
| 12:35.38 | keiths_ | srtpcapable=no in sip.conf? |
| 12:36.29 | igcewieling | niohubala: all sip options are configured in sip.conf. See sip.conf.sample for a list of options. |
| 12:36.39 | igcewieling | niohubala: Maybe you should read the Asterisk Book? |
| 12:36.50 | msaraiva | For that type of questions, google really is your friend... |
| 12:37.03 | msaraiva | voip-info.org |
| 12:37.31 | msaraiva | And the Asterisk book others already told you about |
| 12:38.04 | niohubala | Thank you for your answers. I will try this. I did Google before, no offense |
| 12:39.32 | itgrl | http://www.voip-info.org/wiki/view/Asterisk+SRTP |
| 12:39.52 | itgrl | first result on google search |
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| 12:40.35 | keiths_ | indeed it is. msaraiva, here http://lmgtfy.com/?q=asterisk+srtp+disable |
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| 12:41.57 | keiths_ | and thanks, its been a while since I used lmgtfy.com :) |
| 12:42.09 | JOSHUAHh | hello everyone, i am just wondering what the best SIP phone you would recommend which will work the best with Asterisk out-of-the-box ? |
| 12:42.33 | JOSHUAHh | I prefer to spend under $200 if possible |
| 12:42.34 | msaraiva | Now THAT type of question os good for the channel...because it requires pratical experience. :) |
| 12:42.55 | keiths_ | I am a fan of the Cisco 50x series. others I think will disagree |
| 12:42.56 | msaraiva | Well, Linksys/Cisco is a no-brainer. |
| 12:43.13 | JOSHUAHh | I purchased a 7490G |
| 12:43.28 | JOSHUAHh | and spend hours trying to get it all up and running and could not get a "Register" command out of it for the life of me |
| 12:44.01 | keiths_ | Did you buy it used? |
| 12:44.19 | JOSHUAHh | yes :$ |
| 12:44.32 | keiths_ | wondering if its set for SCCP? |
| 12:44.44 | JOSHUAHh | It had the SIP firmware installed |
| 12:44.48 | keiths_ | k good. |
| 12:45.00 | JOSHUAHh | i saw the SIP menu under settings |
| 12:45.13 | JOSHUAHh | I tried connecting to my Pennytel SIP account and also my asterisk box |
| 12:45.20 | keiths_ | have you tried using your network tools? scope it out with ngrep? |
| 12:45.21 | JOSHUAHh | nothing.. :( |
| 12:45.40 | JOSHUAHh | no i did not .. hmm |
| 12:46.06 | JOSHUAHh | i ended up returning it, the seller was happy to take it back.. |
| 12:46.11 | JOSHUAHh | hence why iam get to drawing boards |
| 12:46.30 | keiths_ | check out the cisco 50x or linksys 94x |
| 12:46.51 | keiths_ | GUI is rather easy to learn and the provisioning is pretty nice |
| 12:47.05 | JOSHUAHh | Web GUI right ? |
| 12:47.21 | [TK]D-Fender | Cisco 7XXX series should all be provisioned for which there are dozens of guides out there |
| 12:48.01 | JOSHUAHh | [TK]D-Fender which phones do you use? |
| 12:48.04 | keiths_ | 7XXX are nice, but don't play as well behind NAT as the 504's do |
| 12:48.28 | [TK]D-Fender | JOSHUAHh: Mostly Polycom, a few Aastra |
| 12:48.34 | [TK]D-Fender | Polycom > All |
| 12:48.45 | itgrl | I've had that happen with a 7940G as well after switching it to SIP from SCCP. Now I can't remember what I did to resolve it. Think it may have become a paperweight |
| 12:49.01 | itgrl | I do like Polycom, except for their boot time. |
| 12:49.21 | JOSHUAHh | so something like this: http://www.ebay.com.au/itm/CISCO-SPA504G-4-Line-IP-Phone-with-2-Port-Switch-PoE-and-LCD-Display-1Yrs-Cisco-/280895634786?pt=LH_DefaultDomain_15&hash=item4166af3d62&_uhb=1 |
| 12:49.29 | keiths_ | the 7XXX series from my experiences DO NOT work when nat=yes is set for no reason |
| 12:49.39 | *** part/#asterisk mihamina (~mihamina@static-110-9.blueline.mg) |
| 12:49.56 | keiths_ | that would be her JOSHUAHh |
| 12:49.57 | JOSHUAHh | keiths_: i litrally tried just about every option i could think of |
| 12:50.01 | JOSHUAHh | never, no, yes, route etc |
| 12:50.08 | itgrl | we use the SPA504G here with good results. my main complaint with them is the headset jack. |
| 12:50.09 | JOSHUAHh | nothing worked :( |
| 12:50.21 | [TK]D-Fender | JOSHUAHh: Last I checks the Linksys/Cisco SPA series was the best quality/value choice for AU |
| 12:50.35 | niohubala | @igcewieling, msaraiva, keiths_: not sure if i'm hijacking the conversation now but I set encryption=no and srtpcapable=no in sip.conf under [general], restarted the server but still receive the same error message: We are requesting SRTP, but they responded without it! |
| 12:50.40 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 12:50.56 | [TK]D-Fender | niohubala: And what do you have under your peer? |
| 12:51.12 | JOSHUAHh | http://www.ebay.com.au/itm/33-x-Cisco-SPA502G-VOIP-Phone-ex-call-centre-/181134266803?pt=AU_Business_Industrial_Office_Products_Equipment_Business_Telephones&hash=item2a2c7191b3&_uhb=1 <----- 33 of them :) nice. |
| 12:51.54 | JOSHUAHh | what do you guys think of Digium branded phones? |
| 12:52.32 | [TK]D-Fender | JOSHUAHh: OK so far. Got their own plusses & minuses |
| 12:52.45 | [TK]D-Fender | JOSHUAHh: Depends what I'm comparing it to. |
| 12:52.52 | niohubala | @[TK]D-Fender: I'm not sure what you mean by "under your peer" |
| 12:52.54 | JOSHUAHh | someone was telling me out of the box, it has support from asterisk ? |
| 12:53.05 | itgrl | with the dpma module |
| 12:53.10 | [TK]D-Fender | niohubala: You know.. the entries you make for your device........... |
| 12:53.14 | niohubala | oh |
| 12:53.34 | [TK]D-Fender | JOSHUAHh: EVERY SIP phone has support for Asterisk out of the box |
| 12:53.38 | [TK]D-Fender | JOSHUAHh: SIP is SIP |
| 12:53.54 | keiths_ | anyone try and like the Digium phones |
| 12:53.55 | JOSHUAHh | Sorry i meant, like function wise |
| 12:54.09 | JOSHUAHh | like buttons etc |
| 12:54.12 | niohubala | @[TK]D-Fender: http://pastebin.com/37JQxpwK |
| 12:54.18 | [TK]D-Fender | JOSHUAHh: Your phone is not incompatible. You simply haven't set it, your network, or Asterisk up right |
| 12:54.26 | [TK]D-Fender | JOSHUAHh: Same answer |
| 12:54.49 | [TK]D-Fender | JOSHUAHh: The standard is the standard. Phone uses it, Asterisk uses it |
| 12:55.24 | [TK]D-Fender | niohubala: pastebin your sip.conf and the call attempt with SIP DEBUG enabled. |
| 12:55.25 | [TK]D-Fender | ~pbb |
| 12:55.28 | [TK]D-Fender | ~pb |
| 12:55.28 | 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. |
| 12:55.29 | [TK]D-Fender | ^^^ |
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| 13:04.14 | keiths_ | one option, disable the module. *might help* |
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| 13:07.23 | niohubala | @[TK]D-Fender. Thanks for the instructions. I pasted this at http://pastebin.com/Q3XGLiQ4 |
| 13:07.52 | niohubala | @keiths_ I tried disabling the SRTP module but then the call fails with a log message saying the SRTP module is not found |
| 13:08.02 | [TK]D-Fender | niohubala: you made TWO [general] tags This is bad. Remove the bottom one |
| 13:08.18 | niohubala | ouch |
| 13:08.46 | [TK]D-Fender | niohubala: allow=all <- you should DISALLOW=all, and then ALLOW= only the codec you intend for it to use |
| 13:09.10 | niohubala | okay, could this be causing the issue? |
| 13:09.26 | [TK]D-Fender | niohubala: Fix these up and retest. if it fails, pastebin your new config and the call debug I originally requested |
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| 13:10.07 | niohubala | by call debug, do you mean a log from the phone itself? |
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| 13:12.39 | [TK]D-Fender | niohubala: ASterisk CLI, verbose 10, "sip set debug" <----------- |
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| 13:16.27 | JOSHUAHh | how well does some of these cisco phones work over wifi ? |
| 13:18.59 | JOSHUAHh | also it looks like the SPA50x series have SIP preinstalled ? |
| 13:20.55 | [TK]D-Fender | JOSHUAHh: Only the 79XX are preferenctially Cisco SCCP by default |
| 13:23.40 | JOSHUAHh | ok great thanks |
| 13:25.29 | niohubala | @[TK]D-Fender: thanks for your help so far. I posted the config and debug at http://pastebin.com/CeZv9ADB . Line 203 contains the warning message metioned earlier |
| 13:29.23 | JOSHUAHh | now to make the decision, SPA501G, SPA509G or SPA504G :S |
| 13:29.48 | [TK]D-Fender | niohubala: Looks more like your device is demanding it... |
| 13:30.09 | [TK]D-Fender | JOSHUAHh: What do you really want out of your phone? How many are you looking for? |
| 13:30.19 | [TK]D-Fender | meeting... BBIAB |
| 13:30.39 | niohubala | oh, then my phone must have been configured in the wrong way |
| 13:31.02 | niohubala | looks like I should learn more about asterisk first |
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| 13:31.22 | JOSHUAHh | [TK]D-Fender - it will be used for business, taking up to 15-20 (~5-10 minute) calls per day. |
| 13:31.30 | JOSHUAHh | just the one for now |
| 13:31.50 | *** join/#asterisk mmlj4 (1000@ip68-11-55-215.no.no.cox.net) |
| 13:32.05 | JOSHUAHh | which that brings my next question, how many of these guys can i put behind the one network (just using standard billion adsl2 modem/router) |
| 13:36.51 | keiths_ | JOSHUAHh, you can get really fancy and go 525 |
| 13:37.18 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
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| 13:47.10 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 13:49.02 | JOSHUAHh | that is fancy keiths_ ! |
| 13:49.54 | keiths_ | I personally like the bluetooth functionality. |
| 14:00.00 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.125) |
| 14:08.31 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 14:08.41 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 14:09.19 | [TK]D-Fender | JOSHUAHh: Depends on your BW, what it will be communicating with, etc |
| 14:09.45 | [TK]D-Fender | JOSHUAHh: typically you have your server in your own LAN and can have as many phones as yoour subnet supports. |
| 14:09.57 | [TK]D-Fender | JOSHUAHh: How many calls depends on BW and your service |
| 14:10.11 | [TK]D-Fender | JOSHUAHh: Or whatever other tech you want them to go out over, etc |
| 14:16.49 | msaraiva | Anyone here using astercc? |
| 14:26.37 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:26.37 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:28.46 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 14:29.09 | msaraiva | Do ConfBridge() and Playback() generate "Link" events on AMI? |
| 14:33.20 | Greenlight | No, not as far as I recall |
| 14:33.53 | *** join/#asterisk newtonr (~newtonr@64.34.219.47) |
| 14:33.54 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:35.41 | msaraiva | Humm |
| 14:37.01 | msaraiva | That's probably why astercc is not working the way i was expecting. |
| 14:37.34 | msaraiva | Even though it detects the channel as up, the call disposition is still "RING". |
| 14:40.16 | *** join/#asterisk aberrios (~aberrios@mail.solutiontelecom.co.uk) |
| 14:41.19 | aberrios | anyone had experience with srtp with * <-> Polycom? |
| 14:41.58 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:42.12 | leifmadsen | aberrios: yes |
| 14:42.15 | leifmadsen | it works fine :) |
| 14:42.33 | *** join/#asterisk mihamina (~mihamina@static-110-9.blueline.mg) |
| 14:45.28 | aberrios | leifmadsen, I've been reading https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial shouldn't the polycom phone need a key from * uploaded? |
| 14:45.43 | leifmadsen | nope |
| 14:45.52 | leifmadsen | SRTP sets up the keys automatically |
| 14:45.56 | leifmadsen | you're thinking SIP-TLS |
| 14:45.57 | aberrios | hmm |
| 14:46.11 | leifmadsen | and that is via certificates |
| 14:46.18 | aberrios | well SIP signalling on TLS is working fine, phone registers on tls okay |
| 14:46.24 | leifmadsen | ok that's great |
| 14:46.28 | aberrios | just calls dont wanna work |
| 14:46.30 | leifmadsen | then the signalling is secured |
| 14:48.07 | leifmadsen | In the Polycom I just have settings like: |
| 14:48.08 | leifmadsen | <sec><sec.srtp sec.srtp.enable="1" sec.srtp.offer="1" sec.srtp.require="1"/></sec> |
| 14:48.16 | leifmadsen | and <reg reg.1.srtp.enable="0" reg.2.srtp.offer="0" reg.x.srtp.require="0"/> |
| 14:48.24 | leifmadsen | then encryption=yes for the peer in sip.conf |
| 14:48.26 | aberrios | i have sec.srtp.offer="1" |
| 14:48.26 | aberrios | <PROTECTED> |
| 14:48.45 | leifmadsen | otherwise, as long as res_srtp.so is loaded |
| 14:48.47 | aberrios | yup got all that |
| 14:48.54 | aberrios | lemmi check that |
| 14:48.55 | leifmadsen | nothing really magical is required to make it work |
| 14:49.23 | aberrios | well that would be that problem,, duh |
| 14:49.28 | leifmadsen | heh |
| 14:50.26 | aberrios | aw man not a re-compile... I want an rpm :( |
| 14:50.36 | leifmadsen | heh |
| 14:50.40 | leifmadsen | I build my own RPMs with mock |
| 14:51.01 | Qwell | aberrios: What distro? |
| 14:51.53 | *** join/#asterisk DennisG (~DennisG@82-170-131-186.ip.telfort.nl) |
| 14:51.55 | doctorray | Is there a way, either through meetme/confbridge/paging where user A can hear users B,C,D, and users B,C,D can hear user A, but where B,C,D cannot hear each other? |
| 14:52.06 | aberrios | Qwell, CentOS 5.9 |
| 14:52.26 | Qwell | aberrios: I'm pretty sure the packages on packages.asterisk.org for CentOS 6 have SRTP support. |
| 14:52.37 | Qwell | You'd think I'd know for sure... |
| 14:52.49 | leifmadsen | Qwell: meh :) |
| 14:52.54 | aberrios | well even more compelling reasons to go to CentOS 6 I suppose |
| 14:52.58 | Qwell | guess it doesn't. heh |
| 14:54.14 | Qwell | malcolmd: I should change that, now that we're using libsrtp for pjproject. Thoughts? |
| 14:59.04 | keiths_ | who uses cisco phones? 5xx series |
| 15:00.15 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.95) |
| 15:01.47 | [TK]D-Fender | ~poll |
| 15:01.47 | infobot | Script for automating Fidonet polls. URL: http://www.drmach.demon.co.uk/vashti/software/index.html |
| 15:01.51 | [TK]D-Fender | ~polls |
| 15:01.51 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 15:04.58 | doctorray | I remember fidonet |
| 15:06.29 | doctorray | [TK]D-Fender: Is what I was just asking possible or would it be extensive rewriting of a confbridge driver to configure audio routing as such? |
| 15:07.59 | [TK]D-Fender | doctorray: That .... wasn't for YOU |
| 15:08.08 | doctorray | I understand |
| 15:08.39 | doctorray | however you are an expert in the software and have helped me in the past with other questions. I figured you could tell me straight up if what I was trying to accomplish was impossible or not |
| 15:09.22 | doctorray | if not, I move on and deal with it |
| 15:09.24 | [TK]D-Fender | doctorray: Was thinking about it and not finding a solution yet myself... |
| 15:09.53 | aberrios | yay compiled res_srtp.c seperately |
| 15:11.03 | [TK]D-Fender | doctorray: Wait... think I've got something... |
| 15:12.29 | [TK]D-Fender | doctorray: Point A to some dead-ish end with no inbound audio (indefinite silence playback, etc), Bave B,C,D ChanSpy() with Whisper on each targeting A's channel. |
| 15:12.34 | [TK]D-Fender | doctorray: That might do it. |
| 15:12.52 | doctorray | [TK]D-Fender: Interesting idea... |
| 15:13.39 | *** join/#asterisk serafie (~erin@nat/digium/x-wnkdnmfibhvdjjwt) |
| 15:14.45 | doctorray | I'll give it a shot |
| 15:14.47 | aberrios | aw i was hoping the phone would change the line icon to a key like it said it should in the polycom manuals |
| 15:16.03 | doctorray | [TK]D-Fender: could I throw A into a MeetMe so that I could control their mute level and chanspy/whisper the established channel? |
| 15:18.17 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 15:21.42 | [TK]D-Fender | doctorray: No, Chanspy isn't really controllable that way. |
| 15:22.04 | [TK]D-Fender | doctorray: This isn't an "admin-able thing so much... but it does seem like it should acheive the audio routing goal |
| 15:25.34 | doctorray | k |
| 15:25.43 | doctorray | I'll give it a shot |
| 15:43.25 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:46.13 | *** join/#asterisk Tarso (~Tarso@186.215.70.48) |
| 15:54.38 | *** join/#asterisk navaismo (~navaismo@189.241.3.102) |
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| 16:05.11 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:05.12 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:05.53 | pabelanger | Q: How can I check _which_ echo canceller is being used in dahdi / zaptel. Assuming I did not build the box |
| 16:06.39 | *** join/#asterisk jblack (~jblack@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 16:08.10 | navaismo | lsdahdi maybe |
| 16:08.30 | navaismo | or opening the system.conf or you say via asterisk cli? |
| 16:08.40 | newtonr | pabelanger: cat /proc/dahdi/X "cat /proc/dahdi/1" i think |
| 16:08.54 | newtonr | pabelanger: thats if you want to see whats in use as well |
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| 16:31.09 | *** join/#asterisk aruntomar (~Thunderbi@49.248.154.154) |
| 16:31.15 | jsin | I have a number stored in speed dial memory on a plain old telephone. Is there anyway to convert the dtmf tones produced by the phone to the numbers they represent? |
| 16:31.58 | jsin | I've forgotten the numbers stored in the speed dial... |
| 16:34.16 | navaismo | that sound evil |
| 16:34.39 | drmessano | Change the dialplan on the phone to call a new extension. Have the extension read back the DTMF |
| 16:34.44 | drmessano | Autodial, I mean |
| 16:35.18 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:35.30 | drmessano | I guess if that works you could have just manually dialed the extension. Errr |
| 16:36.10 | drmessano | or modify the dialplan on the phone to append some digits that won't permit the phone to actually make the call |
| 16:36.17 | Qwell | record the tones, pass it through any DTMF detection utility |
| 16:36.20 | drmessano | Then use the speedial and check the CLI |
| 16:36.44 | drmessano | The CLI will show you whats being called. You just need to tweak the phone to make it dial garbage+number |
| 16:38.40 | drmessano | Oh.. You said a "plain old phone" |
| 16:39.59 | drmessano | Thats even easier, assuming its plugged into some FXS device. Have the FXS device dial some extension, on off-hook, that just reads back digits. |
| 16:40.04 | drmessano | Easy to do with an ATA |
| 16:57.36 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
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| 17:09.12 | *** join/#asterisk tangledhelix (~dan@99-59-105-172.lightspeed.bcvloh.sbcglobal.net) |
| 17:11.38 | Katty | So. |
| 17:11.45 | Katty | what tv series do i want to watch next. |
| 17:12.02 | tzanger | Better Off Ted is hilarious |
| 17:12.10 | tzanger | there's only two seasons though :-( |
| 17:17.34 | leifmadsen | ya, great show |
| 17:17.38 | leifmadsen | Katty: Sons of Anarchy |
| 17:17.44 | leifmadsen | my wife loves that show |
| 17:17.46 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.106) |
| 17:17.57 | Katty | the biker group?! |
| 17:17.59 | leifmadsen | alternatively, I like Suits and have really enjoyed The Newsroom so far |
| 17:18.01 | leifmadsen | Katty: yes |
| 17:18.10 | leifmadsen | girls seem to love the Jax character |
| 17:18.24 | Katty | interesting. ok |
| 17:18.26 | Katty | i'll watch one. |
| 17:19.06 | leifmadsen | Suits is filmed in Toronto |
| 17:23.49 | *** join/#asterisk aruntomar (~Thunderbi@49.248.157.170) |
| 17:29.08 | Penguin | Suits kicks ass. |
| 17:32.54 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.221.88) |
| 17:35.29 | *** join/#asterisk stones_ (~stones@static-70-107-216-88.nycmny.east.verizon.net) |
| 17:35.40 | stones_ | Hi guys. Anyone here use a Digium Iaxy S101i |
| 17:35.51 | stones_ | Im trying to configure it but cannot access the ip/web interface |
| 17:36.20 | coppice | oooh, a museum piece |
| 17:36.28 | stones_ | hah yeah ;) |
| 17:36.38 | stones_ | I had it stored away but want to ship it to my brother |
| 17:36.54 | stones_ | as far as I can remember i used to configure the device via a web interface but i can't even access the ip |
| 17:36.58 | stones_ | I just did a factory reset |
| 17:37.25 | *** join/#asterisk aruntomar (~Thunderbi@49.248.155.188) |
| 17:37.30 | chuckf | Katty: ever watch 'Coupling'? |
| 17:38.01 | [TK]D-Fender | stones_: Last I recall there was no web interface for it ever. Had to be provisioned off your server |
| 17:38.18 | stones_ | hrmf, yeah thats whats I'm reading |
| 17:38.30 | stones_ | but i never configured it via server and i definitely had it working before |
| 17:38.40 | stones_ | is there some sort of tool that would allow it to be provisioned? |
| 17:41.28 | *** join/#asterisk BoRiS (~raiden@S010660a44cdcb910.wp.shawcable.net) |
| 17:43.47 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.106) |
| 17:45.37 | [TK]D-Fender | iaxprov.conf IIRC |
| 17:50.17 | *** join/#asterisk slicknick5181 (~slicknick@204.195.131.94) |
| 17:50.47 | slicknick5181 | Hi all |
| 17:51.28 | leifmadsen | stones_: asterisk does the provisioning via iaxy.conf |
| 17:52.33 | slicknick5181 | In securing my asterisk 1.8 box more I decided to set allowguest=off and my service providers calls stopped coming through. so I got there ip addresses they send calls from and added them to my sip conf. even though the ip address the call is coming from matched my domain = setting the call is still denied |
| 17:52.40 | [TK]D-Fender | That's the one |
| 17:53.06 | [TK]D-Fender | slicknick5181: you match on host= , not domain= |
| 17:53.20 | slicknick5181 | thanks! |
| 17:53.34 | slicknick5181 | [TK]D-Fender, Thanks! |
| 17:54.36 | *** join/#asterisk Dovid (~Dovid@host-78-158-94-218.wlan-guest.nycmny02.us.sargasso.net) |
| 17:57.48 | slicknick5181 | [TK]D-Fender, Still sends Sending fake auth rejection for device |
| 17:58.54 | slicknick5181 | host = <ip-address> that calls come from |
| 17:59.07 | slicknick5181 | [TK]D-Fender, host = <ip-address> that calls come from |
| 18:04.01 | [TK]D-Fender | slicknick5181: That would be another matter... there is another parameter if you wish to allow calls from them to be UN-AUTHED -> insecure=port,invite |
| 18:05.02 | slicknick5181 | [TK]D-Fender, I already have this variable set |
| 18:05.29 | [TK]D-Fender | slicknick5181: then something else is wrong. PB your peer & the call with SIP DEBUG enabled |
| 18:06.58 | slicknick5181 | [TK]D-Fender, Thanks, I have to go but will try when I get back |
| 18:08.46 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
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| 18:26.21 | *** join/#asterisk chris_n (~Chris@koha/developer/chris-n) |
| 18:26.30 | *** join/#asterisk asilva (~asilva@gandalf.ai.unesp.br) |
| 18:28.21 | asilva | Hello All, can someone give me a help, I'm having trouble with IAX2, using Ubuntu 12.04 LTS and asterisk 11.0.1 with IAX2 user and peer, and the seconds server running Debian 6 Asterisk 1.8.22 same iax2 user and peer, can have communication with peers REACHBLE and such, but when using Asterisk 11.1 and UP on the Ubuntu server the IAX2 peers doesn't communicate at ALL, stays UNREACHABLE |
| 18:29.39 | *** join/#asterisk gnudna (~sklav@unaffiliated/sklav) |
| 18:29.43 | gnudna | hi guys |
| 18:30.16 | gnudna | anybody know a highly recommended asterisk consultant or firm in the Montreal area? |
| 18:30.57 | [TK]D-Fender | gnudna: What do you need? |
| 18:31.14 | gnudna | someone to come on site and fix our asterisk |
| 18:31.29 | [TK]D-Fender | Whats wrong, and what is your system set up like? |
| 18:32.31 | gnudna | [TK]D-Fender my system is acting all weird, it works but seems every 24 hours i need to restart asterisk for no reason i can see. no errors in the logs |
| 18:32.55 | gnudna | if you call an extension is rings but if you call back for the next 10 - 15 min it puts you to voicemail |
| 18:33.05 | gnudna | debug does not show this at all |
| 18:33.14 | gnudna | just weird stuff going on |
| 18:34.06 | gnudna | we have queue that when you call it you either get router properly or you end up on hold indefinatly |
| 18:34.37 | gnudna | when i do queue show queuename it says no calls in queue but when you dial in it says im caller #8 |
| 18:34.51 | gnudna | just to mention a few issues ;) |
| 18:35.40 | [TK]D-Fender | gnudna: Pretty messed up ... what versions are you running? hand-built config? What hardware? |
| 18:36.36 | gnudna | hardware is quad core Intel(R) Xeon(R) CPU E5410 @ 2.33GHz with gig ethernet and 12 gigs of ram |
| 18:36.46 | gnudna | 2 hdd's sata is raid 1 |
| 18:36.53 | gnudna | in ^ |
| 18:37.11 | gnudna | debian squeeze stable version of asterisk |
| 18:37.39 | gnudna | <PROTECTED> |
| 18:38.20 | gnudna | im starting to suspect the damn hardware at this point but i get no errors |
| 18:38.29 | asilva | Can someone help me out, heres an scenarion and problema - http://pastebin.com/ZZgDb0va |
| 18:42.11 | navaismo | asilva, have you tried adding the maxcallnumber and calltokenoptionall to both? What show iax2 show debug? |
| 18:43.09 | [TK]D-Fender | gnudna: No telecom cards? |
| 18:43.17 | asilva | navaismo: requirecalltoken=no on both servers |
| 18:43.17 | gnudna | no |
| 18:43.33 | gnudna | just iax2 definition and sip |
| 18:43.35 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 18:43.46 | asilva | navaismo: thru IAX DEBUG on the server with 1.8.21 i GET a lot of POKES and some PONGS, on the Asterisk 11 server POKES only |
| 18:44.03 | asilva | navaismo: thats the only things showing on debug |
| 18:44.43 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.130) |
| 18:44.56 | navaismo | hmm try adding the calltokenoptional=0.0.0.0/0.0.0.0 |
| 18:44.58 | [TK]D-Fender | asilva: #78 [dundi-rt] |
| 18:45.11 | [TK]D-Fender | asilva: That entry ... has not HOST specified... |
| 18:45.16 | [TK]D-Fender | ^ |
| 18:45.21 | [TK]D-Fender | Your entry is broken |
| 18:46.14 | asilva | been a user has to have a host ? using different versions( equal and below 11.0.2) on the ubuntu side works fine |
| 18:46.32 | [TK]D-Fender | gnudna: 1.6.2.9 is very old in its branch. can any calls work internally while this issue occurs? |
| 18:46.50 | gnudna | yes |
| 18:46.52 | [TK]D-Fender | asilva: Every entry has to have that line set |
| 18:46.55 | gnudna | internally it works |
| 18:47.08 | [TK]D-Fender | gnudna: Sounds like a routing / NAT issue |
| 18:47.42 | gnudna | [TK]D-Fender, am i better setting up asterisk on the public directly? |
| 18:47.53 | [TK]D-Fender | gnudna: Shouldn't be necessary. |
| 18:48.06 | [TK]D-Fender | gnudna: How long was it running fine for, and how long has it gone downhill? |
| 18:48.32 | gnudna | was running fine for a week and now it is just flakey as hell |
| 18:48.49 | [TK]D-Fender | ok, relatively new setup... |
| 18:48.57 | gnudna | this one yes |
| 18:49.03 | [TK]D-Fender | I'd start by checking your forwardings, firewalls, etc |
| 18:49.19 | gnudna | we forward everything to the internal asterisk |
| 18:49.39 | gnudna | and outgoing we make sure it routes based on it's public ip |
| 18:49.41 | [TK]D-Fender | If this work on reboot I might wonder about things like fail2ban, etc causing lock-outs |
| 18:49.57 | gnudna | no fail2ban yet installed |
| 18:50.07 | [TK]D-Fender | check firewalls just the same... |
| 18:50.08 | gnudna | i wanted to make sure it was stable before hand |
| 18:50.16 | gnudna | hold on |
| 18:50.21 | [TK]D-Fender | 1.6.2.9 is old as I said... |
| 18:50.25 | gnudna | i will show you the 3 lines in fw |
| 18:50.30 | gnudna | agreed |
| 18:50.34 | asilva | [TK]D-Fender: setting the host to the remote server IP or to dynamic didn't help :/ same problem! going to try setting calltokenoptionall |
| 18:50.40 | [TK]D-Fender | You might want to move that up.... and hopefully to a version of * that is still actually supported... |
| 18:50.45 | gnudna | i did not do this install last sys-admin did and then left 1 week later |
| 18:51.00 | [TK]D-Fender | asilva: that is normally something you look at when going between 1.4 and 1.6+ |
| 18:51.22 | *** join/#asterisk kuruption (kuruption@laffs.at.the.lol.lawlz.lulz.liberlawls.com) |
| 18:51.57 | gnudna | [TK]D-Fender, can i message you 3 lines privately? |
| 18:52.15 | asilva | [TK]D-Fender: I would though of something incompatibility between versions if the 11.0.2 or below wouldn't work too, but it stops working from 11.1.0 and UP |
| 18:55.24 | gnudna | here [TK]D-Fender the firewall rules in question http://pastebin.com/VBr2NdQz |
| 18:55.29 | *** join/#asterisk ageis (kevin@ageispolis.net) |
| 18:56.41 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:56.57 | ageis | got a quick and simple question. The way our dialplan was previously written, for each extension there are these lines that check the dial status and go to busy or unavailable. I'm interested in finding out if I can simplify this and include only Dial() and VoiceMail() and get the same effect. For one, we're not using seperate unavail/busy messages, only one of them. |
| 18:56.59 | ageis | http://pastebin.com/dW0ZMGRr |
| 18:57.33 | ageis | Should I change GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail) to GotoIf($["${DIALSTATUS}" = "BUSY"]?busy) and remove the two unavail lines, or just do a Dial followed by a VoiceMail and be all set? |
| 18:58.23 | asilva | you could use ${DIALSTATUS} != ANSWER go to voicemail |
| 18:58.43 | ageis | nice. |
| 19:00.11 | asilva | navaismo: nothing changed with calltokenoptionall=0.0.0.0/0.0.0.0 or maxcallnumber |
| 19:00.20 | asilva | [TK]D-Fender: any other thoughts ? |
| 19:01.40 | [TK]D-Fender | gnudna: Looks mostly sane.. a little backwards, but OK |
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| 19:09.57 | navaismo | asilva, pbx permit 4569? can ping each other? |
| 19:12.40 | asilva | navaismo: network is correct, they are on the same network , same switch, not firewall or acl between, and with version 11.0.2 works 100% perfect, when I upgrade to 11.1 or UP it stops working! |
| 19:12.45 | gnudna | [TK]D-Fender, i was hoping you would say there was a problem ;( |
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| 19:21.45 | [TK]D-Fender | gnudna: Are you getting timeouts on outside comms? |
| 19:21.53 | [TK]D-Fender | gnudna: Do you at least see attempts, etc? |
| 19:22.00 | [TK]D-Fender | verified internal routing... |
| 19:40.58 | gnudna | [TK]D-Fender, i rolled back to the virtualized container we were running |
| 19:41.19 | gnudna | im guessing some weird stuff with the physical box is the cause of the issues |
| 19:41.45 | gnudna | we had a nat issue before aka 2 weeks ago but resolved it with the rules i showed you |
| 19:42.35 | gnudna | i will look at re-doing the setup with asterisk 1.8.x |
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| 19:51.52 | gnudna | [TK]D-Fender, what scenario would the system not register it received a hangup |
| 19:52.17 | gnudna | aka clients calls in hangs up but the queue still says 4 callers in the queue but the queue shows nothing |
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| 19:56.51 | ageis | what's good, Zultys or Grandstream? |
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| 20:00.45 | [TK]D-Fender | gnudna: I'd have to see the speicifcs |
| 20:00.50 | [TK]D-Fender | ageis: neither |
| 20:02.12 | [TK]D-Fender | ageis: Polycom, Aastra, Digium, Cisco/Linksys SPA series, Yealink, Snom. That approximate order |
| 20:02.38 | [TK]D-Fender | ageis: Specific requirements will alter individual selections of course |
| 20:08.44 | ageis | [TK]D-Fender: cool. |
| 20:11.15 | ageis | [TK]D-Fender: wow, much more expensive |
| 20:13.05 | [TK]D-Fender | Depends which models, where you are located and where you're looking |
| 20:17.39 | gnudna | [TK]D-Fender, you in the montreal area by any chance? |
| 20:18.49 | [TK]D-Fender | gnudna: Yes |
| 20:20.03 | slicknick5181 | [TK]D-Fender, I'm back I turned my sip debug on and this is what i got http://pastebin.com/UddApe4r |
| 20:20.17 | ageis | Do I need my parking lot extensions to be defined as accounts in sip.conf ? |
| 20:21.58 | gnudna | [TK]D-Fender, msg me please if interested |
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| 20:25.32 | slicknick5181 | will asterisk understand 192.168.0.0/24? |
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| 20:26.13 | Qwell | slicknick5181: You'll need to be more specific. |
| 20:26.31 | [TK]D-Fender | slicknick5181: Not for a host= line |
| 20:26.48 | slicknick5181 | sip.conf I need to allow calls from ip addresses ranging from 192.168.0.0 to 192.168.0.255 |
| 20:26.55 | slicknick5181 | * 1.8 |
| 20:27.56 | [TK]D-Fender | slicknick5181: Clarify that... |
| 20:28.10 | [TK]D-Fender | slicknick5181: What exactly is that supposed to match peer-wise? |
| 20:29.22 | slicknick5181 | Qwell, [TK]D-Fender my VoIP provider sends me calls from two sets of addresses for example 192.168.0.0 - 192.168.0.255 and 192.168.1.0 - 192.168.1.255 |
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| 20:30.08 | WIMPy | slicknick5181: Why do you call that two sets of addresses? |
| 20:30.41 | slicknick5181 | WIMPy, it is 2 sets of 255 ip addresses |
| 20:31.01 | WIMPy | I'd call it one set of 512. |
| 20:31.12 | slicknick5181 | WIMPy, in all reality one starts with 22 and the other 205 |
| 20:31.13 | WIMPy | But you could also call it 4 sets of 128 or whatever. |
| 20:31.28 | Kobaz | i have 128 sets of one address |
| 20:31.39 | WIMPy | Then don't give us wrong data. |
| 20:32.12 | Kobaz | i have 27 sets of 40 subnets, each containing 48 sets of 83 addresses |
| 20:32.27 | [TK]D-Fender | slicknick5181: AND A PARTRIDGE IN A PEAR TREE! |
| 20:32.35 | Katty | pears? |
| 20:32.37 | Katty | perks up |
| 20:33.20 | WIMPy | thinks it all went pear shaped. |
| 20:33.35 | Kobaz | perky pears |
| 20:34.02 | Kobaz | Katty can see where this is going |
| 20:34.25 | slicknick5181 | [TK]D-Fender, I have my voip provider set up in sip.conf and working for about a month but when set allowguests=no to secure my system more I lost my incoming calls as my provider sends them from different addresses |
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| 20:35.55 | slicknick5181 | [TK]D-Fender, They provided me the 2 sets of addresses they send me calls from but even when I set those as host= or domain= my calls still get rejected |
| 20:36.11 | [TK]D-Fender | slicknick5181: What retard supplier would send you calls from any random IP from 2 entire subnets? |
| 20:36.17 | WIMPy | You cannot set more than one IP for host. |
| 20:36.38 | gnudna | [TK]D-Fender, you interested in some consulting work? |
| 20:36.39 | WIMPy | A good provider with high availability. |
| 20:37.01 | [TK]D-Fender | gnudna: perhaps. I'm heading home. PM |
| 20:37.23 | gnudna | care to msg me so i can send you my contact info? |
| 20:37.30 | slicknick5181 | [TK]D-Fender, they send me calls from 204.11.192.0 - 204.11.192.255 and 66.193.176.0 - 66.193.176.255 |
| 20:37.49 | [TK]D-Fender | sliHos are those anything like the previous 2 ranges you gave us? |
| 20:38.02 | gnudna | k just got it msg |
| 20:38.05 | [TK]D-Fender | slicknick5181: And why so large? Do they have thousands of servers? |
| 20:38.15 | [TK]D-Fender | gnudna: Will do |
| 20:38.54 | slicknick5181 | [TK]D-Fender, idk but they are for sure making this difficult to secure that way |
| 20:39.07 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
| 20:39.24 | WIMPy | Either allow guests again, create 508 peers or hope that oej solves the problem in Asterisk. |
| 20:39.43 | [TK]D-Fender | slicknick5181: Then firewall your server against everything else and restrict your internal peers to your local subnets |
| 20:40.11 | [TK]D-Fender | heading out... |
| 20:40.12 | [TK]D-Fender | BBL |
| 20:40.16 | slicknick5181 | [TK]D-Fender, I was thinking thats what I was going to have to do |
| 20:40.47 | slicknick5181 | Anyone have a good voip provider who sends calls from just a few ip address |
| 20:42.35 | ageis | question: Do I need my parking lot extensions to be defined as accounts in sip.conf ? |
| 20:43.24 | WIMPy | ageis: You don't define extensions in sip.conf. |
| 20:44.42 | slicknick5181 | WIMPy, Do you have a suggested voip provider that sends calls over a few ip addresses instead of over 510 |
| 20:45.03 | WIMPy | What country? |
| 20:45.11 | slicknick5181 | United States |
| 20:45.41 | WIMPy | ~itsplist-us |
| 20:45.41 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 20:46.13 | ageis | WIMPy: Hello, I meant accounts that correspond to my parking lot extensions, was that not clear? |
| 20:46.42 | Qwell | ageis: Your question doesn't make sense. |
| 20:46.43 | WIMPy | ageis: That doesn't make any sense, either. |
| 20:46.53 | WIMPy | Did you read the book? |
| 20:47.00 | ageis | Some tutorial somewhere must have recommended it that way. |
| 20:47.00 | WIMPy | ~book |
| 20:47.00 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:47.03 | ageis | Let's say I have 701-704 |
| 20:47.14 | ageis | then [701] through [704] are also defined in sip.conf |
| 20:47.16 | ageis | Not necessary? |
| 20:47.22 | Qwell | leifmadsen: Is there an update to ^^^ now that it's gone to print? |
| 20:47.27 | leifmadsen | no |
| 20:47.32 | Qwell | k |
| 20:47.34 | leifmadsen | not sure when I wil update the site to 4e |
| 20:47.57 | leifmadsen | there has to be SOME advantage to purchasing the book :) |
| 20:48.42 | Dovid | is there any way of getting the TO field from a sip invite into a variable? |
| 20:50.26 | Qwell | leifmadsen: You should charge nickels to view asteriskdocs.org. |
| 20:50.35 | ageis | WIMPy Qwell whoever originally wrote our configs set up our multiple parking lots 701-704, 801-804, 901-904 with corresponding sip accounts for each extension in sip.conf. These accounts are not used by any phone. I suspect it is not necessary and wrong and can be deleted, I just want confirmation. |
| 20:50.53 | Qwell | ageis: It's almost definitely wrong. |
| 20:50.58 | ageis | Great. |
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| 21:36.15 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 21:36.46 | saint_ | hi all |
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| 21:41.45 | rrittgarn | Anybody know offhand if there's a way to reboot a Cisco SPA series phone from the web interface? (it's not registered currently or i'd just do a sip notify) |
| 21:42.26 | WIMPy | Just make any senseless change and save it? |
| 21:42.41 | rrittgarn | not a terrible idea |
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| 21:49.25 | drmessano | http://ipaddress/admin/reboot should do it |
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| 22:06.59 | niohubala | I just tested what drmessano said, it works for me |
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| 22:13.48 | WIMPy | Let me guess: You don't even need a password to use it? |
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| 22:14.42 | niohubala | my admin password is blank |
| 22:14.52 | niohubala | i didn't need to enter any password |
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| 22:33.58 | nfenzan | Hello, I am having trouble configuring my firewall to route the UDP traffic correctly between endpoints and my SIP trunk. |
| 22:36.09 | nfenzan | Right now I have a mapped ip on the firewall and all of the SIP connections are operating correctly. I have found that the trunk is routing it's sdp traffic to the asterisk box, while the endpoints are attempting to send directly to the trunk. |
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| 22:36.52 | WIMPy | And where is the trouble? |
| 22:38.15 | nfenzan | I am not receiving incoming audio |
| 22:38.45 | nfenzan | I think the audio is making it to the asterisk box but may not being routed to the endpoint |
| 22:39.31 | nfenzan | the router log is showing incoming traffic being translated and pointed to the correct internal ip of the asterisk box |
| 22:39.52 | WIMPy | It probably shouldn't be. |
| 22:40.02 | WIMPy | But I shouldn't comment on these things. |
| 22:40.14 | niohubala | :p |
| 22:40.30 | nfenzan | haha |
| 22:46.41 | apb1963 | Is anyone using google voice to dial out? |
| 22:52.05 | drudge` | anyone use NExtiva? |
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| 23:33.04 | saint_ | apb1963: i use google voice for both dial out and in |
| 23:33.36 | saint_ | apb1963: i have 2 installations using google voice, voip.ms , and dpma provisioning on digium phones |
| 23:49.03 | niohubala | I'm still stuck with the "We are requesting SRTP, but they responded without it!" :-/ |
| 23:50.40 | niohubala | in sip.conf encryption=no as well as for each user i set encryption=no and srtpcapable=no |
| 23:50.49 | niohubala | still the error above is thrown :/ |
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| 23:51.57 | niohubala | is there any way to completely disable srtp? blacklisting it in modules.conf does not help |
| 23:55.35 | nfenzan | If anyone was interested, I fixed my issue by setting canreinvite=no in sip.conf This allowed the pbx to forward calls from the trunk to the endpoints. Thanks for your help. |
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