| 00:00.04 | mroe | assuming I can setup an asterisk server outside my nat |
| 00:00.14 | mroe | which I can't |
| 00:07.35 | [TK]D-Fender | You clearly need more friends... |
| 00:09.26 | mroe | so true |
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| 00:24.36 | mroe | while I'm waiting for a 'friend' to call me, can someone help me figure out why I cannot connect to jfk-primary.voicepulse.com but I can connect to jfk-backup.voicepulse.com |
| 00:25.53 | mroe | all I get is jfk-primary.voicepulse.com' timed out, trying again (Attempt #224) |
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| 00:36.25 | mroe | figured ito ut |
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| 01:10.22 | wrouesnel | has anyone encountered an issue, where using Sipura line adapters with grandstream phones via asterisk, occasionally when picking up a call the audio will go dead (but the call can still be picked up directly on the line with a regular phone). But only occasionally - I've been able to establish no real pattern and can't find any errors that look like the problem in the logs |
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| 01:25.09 | elcashout | I have Asterisk 10.9.0 running on AskoziaPBX and i can't login to the webgui |
| 01:25.56 | WIMPy | You need to ask on whatever support channels askozia offers. |
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| 02:01.10 | mroe | I'm trying to get all calls that enter a context to ring the sip user 6004 |
| 02:01.36 | mroe | this is what I have in the context in extesions.conf |
| 02:01.38 | mroe | exten => _XX.,n,Dial(SIP/6004) |
| 02:02.06 | mroe | it doesn't ring, and I admit that I don't totally understand the syntax |
| 02:07.18 | WIMPy | That would match all extensions starting with two digits. |
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| 02:08.58 | mroe | and 6004 starts with two digits |
| 02:10.20 | mroe | Call from 'xxxxxxxxxxx' (64.xx.xx.190:5060) to extension '1nnnnnnnnnn' rejected because extension not found in context 'in' is the error I get on the console |
| 02:10.47 | WIMPy | So did you put it in "[in]"? |
| 02:11.02 | mroe | yes |
| 02:11.11 | mroe | it's the only thing in [in] |
| 02:11.23 | WIMPy | Did you reload your config? |
| 02:11.27 | mroe | yes |
| 02:11.49 | WIMPy | Ok, so if that's the first line of the extension, use priority 1. |
| 02:12.08 | WIMPy | n doesn't know about extensioins. It just increments regardless. |
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| 03:43.31 | aster1sk | Regarding this article : http://www.voip-info.org/wiki/view/GXP-2000+Disaster+Recovery |
| 03:43.49 | aster1sk | is in need of the old 1.1.2.27 FW if anyone has. |
| 03:44.28 | aster1sk | The phone's not worth a squirt of piss but I cannot be defeated by legacy. |
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| 05:16.57 | aster1sk | NVM was able to set provisioners addr - everythings good. |
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| 06:56.30 | m0sphere | Can anyone help me out with this? http://pastebin.com/EeLGJfFC - I am trying to clear, or set manually, the value of the variable EXTEN using PHPAGI |
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| 09:09.04 | ChannelZ | m0sphere: I don't really understand your question. |
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| 09:10.46 | ChannelZ | If you want to jump extensions in the dialplan from your AGI, use SET EXTENSION / SET PRIORITY |
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| 09:31.17 | jaflong | In the ami api does it allow to register specific events. For instance can I set a filter to only send me hangup events |
| 09:32.50 | Stefan27 | Does anyone recognize ERROR[22462]: stasis.c:997 stasis_log_bad_type_access: Use of queue_member_added_type() before init/after destruction |
| 09:33.40 | Stefan27 | is it most likely due to bad queues.conf? |
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| 12:48.48 | Teduardo | Are there any decent open source or commercial sip proxies that can deflect brute force attacks? |
| 12:49.32 | Teduardo | meaning if $ip address makes X failed attempts in Y seconds/minutes block it for Z seconds/minutes |
| 12:49.50 | Tim_Toady | Teduardo: kamailio should work for you |
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| 12:51.01 | Tim_Toady | what you described is also possible by combining asterisk and something like fal2ban |
| 12:51.22 | Tim_Toady | but kamailio offers far more options |
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| 12:59.42 | Teduardo | ah okay, I am just asking for someone who is running something called vos3000 and something else called voipswitch and i guess they are getting brute forced constantly and i'm trying to give them decent advice. I will tell them to check out kamailio. |
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| 13:06.35 | Tim_Toady | sipscanner attacks are very annoying, but also easy to deal with, with something like kamailio and just by blocking certain user agent strings you get rid of 99% of them |
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| 13:07.52 | Teduardo | is there a document anywhere that explains how to setup kamailio as a front end proxy to secure back end servers? I've seen references to it being possible to do in google but so far i've struck out actually finding it |
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| 13:08.36 | Katty | hi. |
| 13:09.12 | file | hi |
| 13:09.14 | Tim_Toady | Teduardo: the documendation on kamailio site is where you should start from |
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| 13:32.50 | nny | Is there any tricks/way to forward voicemails between two pbxs or a setup that would allow two pbxs to share voicemails? |
| 13:35.57 | nny | I assume I could just have all phones/voicemail on one server if needed... or let them forward via email |
| 13:36.36 | nny | I figured it out mostly nm |
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| 13:41.21 | udp` | hi, how do i send SMS with .call files? |
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| 13:50.09 | udp` | hi, how do i send SMS with .call files? |
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| 14:16.01 | gavimobile | my pbx started to take its time before making a call today. its taking about 30 seconds for it to make the call. its wasn't like this yesterday. I did make some minor configuration changes yesterday but not anything I can think of which could have caused the issue. here is a call I made with sip debug for the peer making the call http://pastebin.com/F5Bk7TVv the problem is intermittant, it's a 50/50 chance that it will go through immedit |
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| 14:20.06 | gavimobile | the changes I remember making yesterday was installing logrotate, I also created an alias email within mysql (postfix w/ mysql) for 1 email to forward to 3 emails. i changed a voicemail greeting using mv, and I edited voicemail.conf |
| 14:21.40 | gavimobile | im recompiling now just incase |
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| 14:32.17 | wasanzy | hello, I want to use controlplayback to determine where user stopped listening to a music or sound, how do I achieve this? |
| 14:34.55 | [TK]D-Fender | wasanzy, "core show application controlplayback" |
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| 14:40.37 | wasanzy | How do I use the ${CPLAYBACKOFFSET} to get that? |
| 14:40.48 | wasanzy | something like this? exten => 1234,n,ControlPlayback(/var/lib/asterisk/sounds/main_menu_en,${CPLAYBACKOFFSET}) |
| 14:40.49 | fireman_biff | Does asterisk require moh WAVs to be 8kHz or is 44kHz fine? |
| 14:42.15 | [TK]D-Fender | wasanzy, when it exits it SETS that variable. |
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| 14:42.43 | [TK]D-Fender | fireman_biff, WAV is a fixed format. it is not different because you want to use it for MoH |
| 14:43.05 | wasanzy | oh ok, so I can just go ahead and print the result of that variable? |
| 14:43.28 | [TK]D-Fender | wasanzy, "It sets the following channel variables upon completion" |
| 14:44.47 | fireman_biff | [TK]D-Fender: Are you sure? I'm seeing instructions on voip-info for resampling it to 8k, but when I do that it the quality isn't very good |
| 14:45.00 | wasanzy | exten => 1234,n,Verbose(Stopped at: ${CPLAYBACKOFFSET}) ? |
| 14:45.22 | fireman_biff | it's not a bitrate |
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| 14:46.10 | [TK]D-Fender | fireman_biff, that is what the PSTN uses. PSTN != Audiophile HappyLand |
| 14:46.42 | [TK]D-Fender | wasanzy, Are yo asking how to rget the value out of a variable? |
| 14:46.49 | fireman_biff | [TK]D-Fender: but asterisk should be fine with 44k then? |
| 14:46.55 | [TK]D-Fender | fireman_biff, No. |
| 14:47.05 | [TK]D-Fender | fireman_biff, WAV has ONE permitted setting. |
| 14:47.20 | fireman_biff | [TK]D-Fender: ok, thanks |
| 14:47.42 | [TK]D-Fender | fireman_biff, Each format is explicit in its settings |
| 14:48.00 | fireman_biff | [TK]D-Fender: so its still 8k mono then, right? |
| 14:48.12 | wasanzy | [TK]D-Fender: yes am asking how to output the value stored in the variable |
| 14:51.13 | [TK]D-Fender | wasanzy, the same as any variable |
| 14:51.15 | [TK]D-Fender | ~book |
| 14:51.19 | 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 |
| 14:51.20 | [TK]D-Fender | ^^^ |
| 14:51.29 | [TK]D-Fender | fireman_biff, Yes. |
| 14:51.49 | fireman_biff | [TK]D-Fender: cool, thanks a lot |
| 14:52.33 | wasanzy | [TK]D-Fender: what is wrong with the exten => 1234,n,Verbose(Stopped at: ${CPLAYBACKOFFSET}) |
| 14:53.52 | [TK]D-Fender | wasanzy, who says there is something wrong with it? |
| 14:54.06 | [TK]D-Fender | wasanzy, I don't see a call where things are not working as they should. |
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| 14:59.06 | wasanzy | [TK]D-Fender: No one says so but I tested but didn't get a result. I don't understand your second point. |
| 14:59.29 | [TK]D-Fender | wasanzy, I don't see your call doing something it shouldn't |
| 14:59.46 | wasanzy | oh ok |
| 15:00.04 | wasanzy | but am not getting an output in the log |
| 15:02.20 | [TK]D-Fender | wasanzy, I'm getting nothing. I don't see your complete code nor the call itself. |
| 15:04.31 | wasanzy | http://paste.scsys.co.uk/407433 when I dial 1234, I hear the menu, and when I press hungup on the softphone, I didn't see the verbose output |
| 15:08.18 | [TK]D-Fender | wasanzy, I don't see the call |
| 15:08.31 | wasanzy | you want the logs? |
| 15:08.51 | [TK]D-Fender | wasanzy, Do you want an autopsy? If so, provide a body... |
| 15:09.01 | WIMPy | Well, if you hangup, the dialplan doesn't continue for obvious reasons. |
| 15:09.11 | WIMPy | See the h extension. |
| 15:09.47 | *** join/#asterisk aross42 (~aross@72.38.109.67) |
| 15:10.38 | wasanzy | http://paste.scsys.co.uk/407435 |
| 15:10.53 | wasanzy | WIMPy: h extension? |
| 15:11.06 | [TK]D-Fender | <PROTECTED> |
| 15:11.07 | [TK]D-Fender | ^^^^ |
| 15:11.20 | [TK]D-Fender | you do not seem to have learned your dialplan basics.... |
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| 15:11.59 | wasanzy | [TK]D-Fender: I don't understand |
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| 15:15.28 | [TK]D-Fender | wasanzy, You don't uderstand the basics of the dialplan. Normal execution stops when you hangup. You don't jsut go to the next pririoty. Go read the book. |
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| 15:15.37 | [TK]D-Fender | priority |
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| 15:22.13 | gavimobile | my pbx started to take its time before making a call today. its taking about 30 seconds for it to make the call. its wasn't like this yesterday. I did make some minor configuration changes yesterday but not anything I can think of which could have caused the issue. here is a call I made with sip debug for the peer making the call http://pastebin.com/F5Bk7TVv the problem is intermittant, it's a 50/50 chance that it will go through immedit |
| 15:22.13 | gavimobile | <PROTECTED> |
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| 15:51.46 | gavimobile | if my router supports network address translation, so modem to wan and lan 1 to pbx and lan 2 to sip device, would I or would I NOT specifiy nat=force_rport,comedia for peer who uses lan 2 in asterisk sip.conf file? |
| 15:59.10 | jaflong | In the ami api does it allow to register specific events. For instance can I set a filter to only send me hangup events |
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| 16:12.55 | mjordan | jaflong: yes. There are whitelist and blacklist filters you can set up for an account in manager.conf |
| 16:13.27 | Penguin | gavimobile: If both asterisk and the sip device are on the same LAN (without any NAT between them), you can safely set nat=no for that device. |
| 16:13.34 | _grim_ | hey all.... got one for you.... need to initiate a call with a call file, however I need to update a database record with the call uniqueid when the initial call from the call file is placed.... ideas? |
| 16:14.51 | mjordan | _grim_: create a Local channel from the call file that goes to the dialplan. Update the database from there using func_odbc. |
| 16:16.14 | _grim_ | I thought about that, but then how do I send the call to the queue app once the dial is completed? wait, nevermind... dial flag that will send the answered call to a macro which dumps the call to the queue app.... |
| 16:16.26 | jaflong | <@mjordan> : is there way to do it on the api level. like how phpagi does it : $ami->add_event_handler('Dial', ... |
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| 16:26.11 | gavimobile | Penguin: so in my situation, only remote peers should have nat option enabled |
| 16:26.35 | Penguin | gavimobile: Only if those remote peers are behind a NAT. |
| 16:27.07 | gavimobile | meaning, only if thse remote peers are behind a nat suported device? |
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| 16:27.22 | Penguin | If they sit on the internet with a public address, they are not behind NAT, so your configuration would be nat=no for those. |
| 16:27.23 | gavimobile | cause if I understand correctly, not all routers use nat |
| 16:27.38 | gavimobile | Penguin: thanks for that clarification |
| 16:34.27 | mjordan | jaflong: I'm not sure what you mean. That library is still receiving all the events, it just drops the ones it doesn't want on the floor. That's not a function of Asterisk; that's how they wrote the library. AMI itself is not pub/sub: you get the events unless you filter them out. |
| 16:35.39 | mjordan | _grim_: A local channel doesn't change that scenario. If what you were doing was creating a SIP channel directly and placing it into a Queue app, you can do the same thing with a Local channel. One half goes and does the dial, the other half gets put directly in the Queue. |
| 16:36.08 | mjordan | _grim_: the power of the Local channel is that it is in the dialplan before doing a Dial. That means you can make changes (or update databases, in your case) before you dial out to the SIP peer. |
| 16:38.40 | _grim_ | so something like this for the local context - exten s,1,{change database} - same n,dial(sip/carrier/18005551212) - same n,queue(callqueue) |
| 16:40.16 | _grim_ | oh.... duh.... sorry, a little slow this morning.... |
| 16:40.40 | _grim_ | the call file will connect the local channel (which is where the dial happens) to the queue ..... |
| 16:40.48 | _grim_ | ok, thanks for the help!!!! |
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| 17:57.20 | TazzNZ | morning all |
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| 18:07.28 | mbowie | Moin |
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| 18:13.00 | saint_ | hi all - is there any channel where people talk about pure sip development by any chance ? |
| 18:15.02 | TazzNZ | saint_: *just* SIP dev or asterisk dev ? |
| 18:15.15 | TazzNZ | I think "SIP" is kinda done "dev'ing" anyways |
| 18:15.42 | saint_ | sip. I have an issue with a third party phone, and I'm talking to their r&d. we are arguing about the RFC ... So I m looking for a channel where it's pure technical sip. |
| 18:15.55 | saint_ | ouch. i guess i m stuck then. |
| 18:16.13 | TazzNZ | what are you guys stuck on ? |
| 18:16.31 | saint_ | DTMF & RFC2833 |
| 18:16.38 | saint_ | The E bit, and its duration |
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| 18:17.29 | TazzNZ | I would ask in asterisk-dev |
| 18:17.29 | saint_ | rfc2833 is not really clear about the duration field of an rtpevent with the bit E used. |
| 18:17.39 | saint_ | ha, sweet. thanks. |
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| 18:20.10 | coppice | saint_: is RFC4733 any clearer? |
| 18:20.21 | saint_ | nah. |
| 18:22.04 | TazzNZ | saint_: so what part of the RFC are they saying is not clear ? |
| 18:22.28 | saint_ | i m looking for the number. we've only copy / past part of the rfc about it. stand by. |
| 18:22.41 | TazzNZ | if you going to paste more than 3 lines |
| 18:22.43 | TazzNZ | ~pb it |
| 18:22.46 | TazzNZ | :D |
| 18:22.48 | TazzNZ | lol |
| 18:22.50 | TazzNZ | ~pb |
| 18:22.50 | 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. |
| 18:23.42 | saint_ | got it. i'll talk about it in asterisk-dev |
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| 19:48.56 | eppigy | hello |
| 19:48.58 | eppigy | i am dave |
| 19:49.33 | newtonr | hi dave! |
| 19:49.35 | eppigy | I heard you were looking for me, kayatwork |
| 19:49.40 | eppigy | I heard you were looking for me, Katty |
| 19:49.44 | eppigy | hello newtonr |
| 19:49.47 | kayatwork | um, no |
| 19:49.58 | eppigy | sorry kayatwork tab completioin |
| 19:50.04 | eppigy | strikes again |
| 19:51.52 | TazzNZ | did you bring Katty's hair brush ? |
| 19:51.59 | eppigy | I did not |
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| 20:12.10 | qakhan | hi all, how to array value coming from .php in dial plan |
| 20:13.40 | qakhan | get* |
| 20:14.13 | TazzNZ | qakhan: are you using an AGI ? |
| 20:16.45 | qakhan | yes |
| 20:18.04 | Katty | :> |
| 20:18.09 | Katty | glomps eppigy |
| 20:18.21 | Katty | eppigy: i am always looking for you. |
| 20:20.31 | TazzNZ | qakhan: http://www.voip-info.org/wiki/view/set+variable |
| 20:21.56 | Katty | TazzNZ: did you find my hairbrush? |
| 20:22.37 | TazzNZ | I have *a* hair brush.....doubt it's yours thou |
| 20:22.46 | TazzNZ | and it's a tad far to fetch I guess :) |
| 20:22.48 | qakhan | TazzNZ i alreay have done that. when i call variable in dialplan it reture "Array" |
| 20:23.17 | TazzNZ | qakhan: that is a PHP issue |
| 20:23.22 | qakhan | execute_agi("SET VARIABLE result $pieces"); |
| 20:23.46 | TazzNZ | you need to refer to an elemet in the array |
| 20:24.05 | qakhan | NoOp("SIP/3288-0000003d", "Array") in new stack |
| 20:24.06 | eppigy | Katty: awww :> |
| 20:24.07 | TazzNZ | execute_agi("SET VARIABLE result ".$pieces[1]); |
| 20:24.11 | TazzNZ | as example |
| 20:24.24 | mbowie | (Or convert the array into a string which is consumable.) |
| 20:24.43 | qakhan | but i dont know how many elements will be there in an array |
| 20:24.47 | mbowie | Ala implode(',', $foo) or similar. |
| 20:24.52 | qakhan | i am getting data from DB |
| 20:25.00 | mbowie | Are you setting them into one value or multiple values? |
| 20:25.02 | qakhan | it can be from 1 to 5 |
| 20:25.04 | TazzNZ | what mbowie said :) |
| 20:25.19 | Katty | eppigy: how'rechu? |
| 20:26.28 | qakhan | http://pastebin.com/AQHXineY |
| 20:26.49 | qakhan | mbowie see my pastebin |
| 20:26.52 | mbowie | I see that. |
| 20:26.52 | eppigy | Katty: I am well. A little sleepy |
| 20:27.19 | mbowie | So you're taking all the strings returned in $records and setting them into one AGI value? |
| 20:27.40 | qakhan | yes |
| 20:28.03 | mbowie | So why explode the original string? |
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| 20:28.46 | qakhan | to get values in separate rows |
| 20:29.27 | mbowie | That doesn't really make sense. $records is a string... not rows. |
| 20:30.06 | mbowie | Regardless, to set VARIABLE to the contents of $pieces, you're going to have to implode it again with whatever delimiter you desire. |
| 20:30.08 | qakhan | yes $records is string but $pieces are rows |
| 20:30.37 | mbowie | To be specific, $pieces aren't rows, it's an array of strings. |
| 20:30.47 | qakhan | ok |
| 20:31.26 | mbowie | You either need to assign each string to it's own AMI variable... or squeeze all the strings into one string to assign to a single AMI variable. |
| 20:33.07 | qakhan | ok i got you. |
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| 20:36.53 | mbowie | Excellent. |
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| 21:11.50 | brad[] | Hi all, I'm connecting to a SIP trunking service which specifies differing media and signal servers but I've not encountered this before with Asterisk. I don't see anything potentially relevant in the list of configuration directives available to sip.conf - are there any pointers? |
| 21:15.09 | Qwell | brad[]: the signaling server will tell Asterisk where media needs to go. |
| 21:21.12 | brad[] | Qwell: hrm. |
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| 22:35.19 | pjensen00 | I've been doing some bridging of PJSIP channels through the AMI manually. It works just fine, but the bridge that is created does not destroy itself once all members leave. I am unable to destroy any of these bridges manually either |
| 22:35.20 | pjensen00 | http://pastebin.com/UXruRnMe |
| 22:36.24 | pjensen00 | I'm curious if there is a way to destroy these bridges |
| 22:37.57 | pjensen00 | I also checked JIRA for open issues pertaining to bridges in Asterisk 12 and didn't see anything |
| 22:39.33 | mjordan | pjensen00: what AMI command are you using? |
| 22:40.50 | pjensen00 | mjordan: "Bridge" |
| 22:42.03 | mjordan | looks at something |
| 22:43.05 | mjordan | nuts. Yeah, there's a bug someplace; ref leak on the bridge. |
| 22:43.33 | mjordan | Do you mind filing a bug? Please include a DEBUG log with pjsip debugging on showing a single instance of a bridge getting orphaned |
| 22:43.42 | mjordan | that should let us figure out who is holding onto the thing |
| 22:43.45 | pjensen00 | ok |
| 22:43.48 | mjordan | thanks! |
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| 23:26.57 | pjensen00 | what level of verbosity and what debug level should I have? |
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