| 00:00.00 | asteriskmonkey | i changed the account i was using, now i see xmpp error messages |
| 00:00.16 | asteriskmonkey | but they give the call description but no failure error |
| 00:00.21 | asteriskmonkey | so its really frustrating |
| 00:01.53 | ChannelZ | Is that through your Google Voice number? I don't even know how to call anyone now besides that.. hangouts on Android are video calls |
| 00:02.11 | asteriskmonkey | no, im just tring to get a call to a @google user address |
| 00:02.16 | asteriskmonkey | well @gmail :P |
| 00:02.20 | ChannelZ | ok |
| 00:03.22 | ChannelZ | Hmm I don't think mine is working either. |
| 00:03.42 | asteriskmonkey | ah well thats comforting, atleast i know ive not gone insane now |
| 00:04.06 | ChannelZ | LOLZ: "Sorry! The voice chat with Bob failed because of a problem with our servers at 6:03 PM. Please wait a bit and try again." |
| 00:04.27 | asteriskmonkey | well atleast you get that message |
| 00:04.30 | ChannelZ | though it did hit my dialplan.. |
| 00:04.43 | asteriskmonkey | yeah im goign sip phone->googletalk out |
| 00:04.50 | ChannelZ | might be confused because I'm on the same network my asterisk is on |
| 00:05.01 | asteriskmonkey | maybe, priority fixes that though |
| 00:07.26 | ChannelZ | hmm ok actually I just called myself from home and it appears to still work. |
| 00:07.50 | Mon|A|rch | this might be beyond the scope of this channel, let me know if it is: I'm having trouble reaching a server using func_curl, because I'm getting an HTTP 302, any ideas on how to get around that? |
| 00:08.04 | asteriskmonkey | http 302 error |
| 00:08.08 | Mon|A|rch | not sure how to include extra headers, since the docs aren't entirely explicit |
| 00:08.08 | asteriskmonkey | thats not curls fault |
| 00:08.17 | Mon|A|rch | it's not |
| 00:08.27 | Mon|A|rch | but normally you can just send a followredirect header |
| 00:08.33 | asteriskmonkey | thats a redirect |
| 00:08.45 | asteriskmonkey | ah right |
| 00:08.47 | Mon|A|rch | not sure how to do that with asterisk's flavor of curl though |
| 00:08.53 | asteriskmonkey | think it was CURLOPTS? |
| 00:09.00 | asteriskmonkey | in the cli |
| 00:09.15 | asteriskmonkey | voip wiki probably has it |
| 00:09.15 | Mon|A|rch | CURLOPT() is there, but it doesn't list an option you can use |
| 00:09.18 | ChannelZ | you'll probably have to do it externally |
| 00:09.29 | asteriskmonkey | yeah set it as a global |
| 00:10.12 | Mon|A|rch | ChannelZ, you mean do the redirect externally? |
| 00:10.19 | ChannelZ | no do the whole thing |
| 00:10.28 | Mon|A|rch | ah |
| 00:10.39 | ChannelZ | like write an AGI that hits your url however you need to/can (PHP, a shell script using curl on the commandline, etc.) |
| 00:11.09 | Mon|A|rch | yeah, i'd thought of that, was just hoping I didn't have to |
| 00:11.10 | Mon|A|rch | oh well |
| 00:11.13 | asteriskmonkey | I like to curl local php files that do my extended magic and populate arrays in asterisk |
| 00:11.49 | Mon|A|rch | I'm sure I've screwed something up on my server, since I can cURL to other scripts |
| 00:11.54 | Mon|A|rch | thanks for the help |
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| 00:14.44 | ChannelZ | heads out to dinner |
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| 01:02.04 | Katty | hi lads! |
| 01:03.26 | WIMPy | Hi Katty |
| 01:03.31 | Katty | waves to WIMPy |
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| 01:38.45 | Grady2000 | hey all been reading the docs for hours with no luck on this, i'm really hoping for some help here |
| 01:38.53 | Grady2000 | i migrated from asterisk 1.4 to 1.6 |
| 01:39.21 | WIMPy | 1.4 with a 1.2 config? |
| 01:39.45 | Grady2000 | and in my extensions.conf file i have macro(yes I know they are deprecated) that is called on the dialout command to a call center agent |
| 01:40.08 | Grady2000 | the working macro in 1.4 is: |
| 01:41.02 | Grady2000 | exten => s,1,Set(TIMEOUT(digit)=3) ; |
| 01:41.03 | Grady2000 | exten => s,n,Read(ACCEPT|/var/lib/prompts/ncall/eng/2321|1) |
| 01:41.28 | Grady2000 | and prompt 2321 is simply a prompt saying" to accept this call press 1, to deny press 3" |
| 01:41.45 | WIMPy | And there are pipes again. |
| 01:42.15 | Grady2000 | the call center rep then presse's 1 and the call connects(macro continues) but my problem is I can not replicate this in 1.6 |
| 01:42.38 | Grady2000 | anyone know the equivalent of my commands in 1.6? |
| 01:42.47 | WIMPy | What does (not) happen? |
| 01:43.18 | Grady2000 | the outbound call is made and then the person answering the phone does NOT hear prompt 2321 |
| 01:43.32 | Grady2000 | and therefore cannot press 1 to accept the call |
| 01:43.53 | WIMPy | Did you replace the pipes? |
| 01:44.06 | Grady2000 | with commas? |
| 01:44.11 | WIMPy | yes |
| 01:44.36 | Grady2000 | only the last pipe, i will try the first but then do I surround the directory location in quotes? |
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| 01:44.59 | WIMPy | no |
| 01:47.36 | Grady2000 | your a lifesaver wimpy, i can't belive it worked and that i missed that first pipe! thank you |
| 01:47.42 | Grady2000 | i really need to pay more attention |
| 01:48.07 | WIMPy | You should search for all pipes in your dialplan. |
| 01:48.23 | WIMPy | And you could have changed that in the 1.4 times. |
| 01:48.30 | Grady2000 | i did however get a red ERROR on the console: write() returned error: Broken pipe |
| 01:48.48 | Grady2000 | and in the docs it said I could add a [compat] section in the asterisk.conf file |
| 01:49.03 | Grady2000 | to support all my old pipe commands,, but that did not work eiter |
| 01:49.21 | Grady2000 | pbx_realtime=1.4 |
| 01:49.22 | WIMPy | That worked in 1.4, but I don't know for how long. |
| 01:49.30 | Grady2000 | is what the docs said,, and those are the 1.6 docs |
| 01:50.09 | WIMPy | Well, 1.6 has luckily long been erased from my memory. |
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| 01:57.15 | Weezey | I must say, I'm really enjoying asterisk 11 |
| 02:03.59 | Grady2000 | what do you like most about it? |
| 02:04.28 | WIMPy | You will find out if you upgrade in a few years :-) |
| 02:05.08 | WIMPy | It has quite some signalling advances, like CONNECTEDLINE and REDIRECTING and off corse ConfBridge. |
| 02:05.35 | WIMPy | And it's stable. |
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| 02:06.13 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 02:12.51 | Weezey | Yeah, most pleased with its stability and the ConfBridge is awesome. |
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| 02:31.10 | wchance | Hello, I have about 100 custom audio files I want to upload to new Asterisk system. But I do not want to do it via web interface. Any suggestions? |
| 02:31.34 | WIMPy | Asterisk doesn't have a web interface. |
| 02:32.08 | wchance | elastix |
| 02:33.15 | WIMPy | And apart from the fact that Asterisk is not involved in your file transfer in any way, you will have all the options you have on any Linux system. |
| 02:34.55 | wchance | I know how to upload to the correct directory. |
| 02:35.07 | wchance | Thanks WIMPy I will seek help with Elastix |
| 02:35.08 | wchance | cheers |
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| 02:42.40 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.11) |
| 02:43.50 | ruben231 | hi guys i have check an existing asterisk server when i do asterisk -rv --> Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) <---------maybe the user for asterisk is not root something is created how do i check which users are used for asterisk..? |
| 02:46.36 | ChannelZ | are you 1. sure it's running and/or 2. it's actually the one running as root yet you are not? |
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| 03:16.00 | ruben231 | ChannelZ: when i do asterisk -rvvv the result is this ---> Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) , even i run asterisk on this type command ---> asterisk |
| 03:16.20 | ruben231 | still Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
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| 03:31.26 | ruben231 | hi guys.? i have asterisk, 16 person dial is normal but when it reach to 18--we can evidently get voice quality issue -choopy lines <----------any idea guys..? |
| 03:31.52 | *** join/#asterisk AliRezaTaleghani (~Thunderbi@unaffiliated/AliRezaTaleghani) |
| 03:39.59 | igcewieling | ruben231: Asterisk is not running |
| 03:40.12 | igcewieling | "Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)" means "asterisk not running" |
| 03:41.09 | ruben231 | igcewieling: tried command --> asterisk <---still same error |
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| 03:50.33 | igcewieling | ruben231: how about asterisk -cvvv |
| 03:51.12 | igcewieling | asterisk and asterisk -r connect to an existing running instance of Astersik |
| 03:51.42 | igcewieling | asterisk -c starts asterisk in the foreground. If that works, ctrl-c out and run safe_asterisk |
| 03:53.40 | igcewieling | and for the love of all that is good and true in the world please read the asterisk book |
| 03:53.43 | igcewieling | ~book |
| 03:53.43 | 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 |
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| 04:15.08 | fling | I have an addpac ata. It is skipping dtmfs when I dial with a panasonic phone. |
| 04:15.16 | fling | 500 may become 00 or may become 5 |
| 04:15.34 | fling | It worked fine in the past with an old russian phone. |
| 04:15.40 | fling | How may I fix it/ |
| 04:15.48 | fling | ChannelZ: Hello. |
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| 04:25.57 | hebber | fling: I would check the dtmf parameters in SIP.conf on that account vs how you transfer dtmf from the ATA |
| 04:26.25 | fling | hebber: it is an analog problem. |
| 04:27.00 | fling | hebber: I still hear the tone when ata ignores phone's dtmf |
| 04:27.39 | hebber | fling: well how does ATA and asterisk handle dtmf? |
| 04:28.02 | fling | hebber: rfc, works fine |
| 04:28.07 | fling | I also tried different modes |
| 04:28.24 | fling | dtmf works fine between ata and asterisk |
| 04:28.39 | fling | it works bad between phone and ata :| |
| 04:29.39 | igcewieling | fling: sounds like you need to be looking at the phone and the ATA docs |
| 04:30.51 | fling | igcewieling: addpac ata does not have any setting on the phone port |
| 04:30.56 | fling | also I can't use pulse with it |
| 04:31.12 | hebber | fling: or relax dtmf settings on Asterisk with in-audio - which may create other problems with |
| 04:31.14 | fling | KX-TS2365RU phone does not have dtmf settings :| |
| 04:31.52 | *** join/#asterisk wrouesnel (~wrouesnel@203-206-134-36.perm.iinet.net.au) |
| 04:32.43 | fling | hebber: but ata is not sending anything to asterisk (including audio) until I finish dialing |
| 04:32.47 | wrouesnel | does anyone know why this piece of code doesn't evaluate properly: ${DB_EXISTS(DoctorExtensions/${EXTEN:-1:1}) |
| 04:33.02 | fling | wrouesnel: -1 |
| 04:33.26 | wrouesnel | fling: Its to grab the last digit of the extension |
| 04:33.34 | fling | wrouesnel: and you are missing '}' |
| 04:33.56 | wrouesnel | and the } is there i just screwed up copying it out of my dialplan... |
| 04:34.03 | wrouesnel | and actually i may have just figured it out - go figure. |
| 04:34.10 | wrouesnel | happens everytime i bring a problem to irc |
| 04:35.23 | wrouesnel | urgh! spent an hour on this the other day, just realized that my extensions don't start at 1. |
| 04:35.33 | wrouesnel | and the phone I was using is still programmed to that. |
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| 04:38.14 | Lepon | hey all |
| 04:38.31 | hebber | fling: then I'm not sure on how to help you |
| 04:40.02 | fling | hebber: ok, looks like I need another phone or ata |
| 04:41.10 | Lepon | Question for all you helpful people: I get an incoming call and then connect it through by dailing out to an external number. The incoming call is recorded but can be turned on/off by a macro that the external callee can trigger. (this all works great) What I am trying to do is play a sound file to the callee (external number that triggers the macro) when the macro is triggered. I tried |
| 04:41.10 | Lepon | putting Playback(file) in my macro but that only plays the sound the original incoming caller not the callee who triggered the macro. Any thoughts? |
| 04:41.25 | Lepon | If that makes any sense |
| 04:42.43 | ChannelZ | well there's a Dial option for playing sounds to one channel or the other |
| 04:43.45 | Lepon | unfortunately I need to play it at what ever point durring the call they trigger the macro. I do use the Anncoumenet option in dial to play an original greeting when calling though |
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| 04:45.29 | Grady2000 | i'm running 1.6 meetme and I notice that parties can press star to access a menu during the conf,, is there anyway to disable that? |
| 04:46.50 | igcewieling | Lepon: You are using features.conf to handle the in-call recording toggle? |
| 04:46.57 | Lepon | I am indeed |
| 04:47.31 | igcewieling | Lepon: if there is a fix, it would likely be in features.conf |
| 04:47.59 | Lepon | ok, I might investigate that a bit futher then, thanks. |
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| 04:54.56 | igcewieling | called .vs. calling channel can be confusing on the universe of features.conf |
| 05:00.38 | fling | Friend in my lan behind nat can't connect to some sip server because of my nat. |
| 05:01.01 | fling | I have a separate wan port on _my_ pbx. |
| 05:01.29 | fling | May I proxy friend's sip somehow? stun? |
| 05:06.47 | Lepon | Thanks igcewieling, had a look at my features setup etc. I'm a bit stuck now though. My feature is setup to run the macro on the caller (because that is where the recording is taking place so needs to run there) which is why they are hearing the sound and not the callee who triggered the macro. But I can't change the macro to run on the callee channel because then the recording won't be |
| 05:06.47 | Lepon | start/stopped on the right channel. seems like I need a way to either 1) execute two macros on different channels at the same time or 2) run the macro as is but somehow get playback to execute on the other channel |
| 05:06.57 | Lepon | Doesn't seem like this is going to work easy for me |
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| 05:18.34 | wrouesnel | has anyone had any experience with Grandstream phones and asterisk suddenly providing no audio? The call picks up fine, plays a fractional second of audio and then just cuts out completely, even though asterisk still thinks the call is alive. |
| 05:24.17 | [TK]D-Fender | Allowing a reinvite Where the networking can't accomodate it |
| 05:25.54 | wrouesnel | hm...the usual scenario is between a Sipura acting as a PSTN ingress and a Grandstream, but these are all on the same network. |
| 05:27.41 | wrouesnel | and it doesn't always happen either - but surely a packet drop/delay somewhere wouldn't cause it? |
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| 05:30.02 | ChannelZ | I blame squirrels |
| 05:38.09 | Itsm | any non US folks in here? |
| 05:38.32 | [TK]D-Fender | plenty |
| 05:41.03 | Itsm | so give me your non US voip provider :) |
| 05:48.41 | [TK]D-Fender | Is there a problem with the SUA specifically.. or should you maybe tell us where YOU want to be calling... |
| 05:48.54 | [TK]D-Fender | USA* |
| 05:49.49 | Itsm | i want a server which is located out side of the US, and im calling worldwide |
| 05:50.32 | [TK]D-Fender | les.net |
| 05:51.10 | Itsm | do you have any experience with them? |
| 05:52.04 | [TK]D-Fender | yes |
| 05:52.11 | [TK]D-Fender | They've worked fine. |
| 05:53.39 | Itsm | do they offer international DID's |
| 05:53.43 | Itsm | ? |
| 05:57.25 | [TK]D-Fender | go look.... |
| 05:57.31 | [TK]D-Fender | bed time... |
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| 08:22.16 | tzafrir_laptop | Fix for the host appearing as "(none)" at register time when using TLS in sip: |
| 08:22.29 | tzafrir_laptop | explicitly write tlsbindaddress=0.0.0.0 |
| 08:22.52 | tzafrir_laptop | (no idea what happens when you have to use ipv6. don't have it here) |
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| 09:54.46 | wrouesnel | is it possible to pull in a keytree using the asterisk manager api? |
| 09:55.02 | wrouesnel | i.e. like what you get with database show |
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| 10:00.17 | vlt | Hello. I installed asterisk 1.8 on Ubuntu 12.04 LTS. I tried to get MeetMe to work but I got error messages: dahdi not found. I installed the pkg asterisk-dahdi, now I have got the MeetMe app but still no dahdi. There’s no "*dahdi*" file in /lib/modules/. Any idea what to try next? |
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| 10:01.35 | WIMPy | Do you have old scripts tailored for MeetMe? |
| 10:01.47 | bombev | guys from sip show channelstats I got jitter 0.0022 |
| 10:01.52 | bombev | is it good enough? |
| 10:02.08 | kaldemar | vlt: install dahdi. asterisk-dahdi only includes an asterisk channel driver for dahdi. |
| 10:03.38 | kaldemar | vlt: asterisk-dahdi should have dahdi in its dependencies though. |
| 10:04.20 | vlt | kaldemar: I think I found something: |
| 10:04.49 | vlt | kaldemar: I use an ubuntu-virtual kernel but dahdi installed the -headers pkg for -generic. |
| 10:05.03 | vlt | Trying to install linux-headers-...-virtuel manually |
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| 10:18.11 | vlt | kaldemar: That worked. |
| 10:18.15 | vlt | Thank you. |
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| 11:04.51 | Ice_Strike | Is possible to change the caller ID from a mobile sim card? |
| 11:05.20 | WIMPy | Yes, your provider can do it. |
| 11:05.45 | Ice_Strike | Ah provider |
| 11:05.58 | Ice_Strike | I was thinking something liek this: Set(CALLERID(num)= |
| 11:06.02 | Ice_Strike | to any number. |
| 11:07.01 | WIMPy | Not even the mobile knows it's number. It's never transmitted. |
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| 11:07.40 | Ice_Strike | Hmm |
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| 11:36.29 | Itsm | Hi, anyone up to assist with mixmonitor? |
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| 11:38.06 | Greenlight | What issue are you having? |
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| 11:38.39 | Itsm | No issues really, I just want it to save the file under a certain name |
| 11:38.56 | Itsm | the defualt is something like this: OUT536-20130627-103333-1372318413.2946.WAV |
| 11:39.16 | Greenlight | Indeed - you can specify the save name when you call it |
| 11:39.36 | Itsm | now I've added in run after record this: mv ^{MIXMON_DIR}/^{CALLFILENAME}.^{MIXMON_FORMAT} ^{MIXMON_DIR}/`/usr/bin/mysql -u root -pasteriskpass-N -B -D asteriskcdrdb -e "SELECT calldate,'From:',src,'To:',dst,'' FROM cdr WHERE uniqueid = ^{UNIQUEID}"|sed -s s'/\s/_/ g'`.^{MIXMON_FORMAT} |
| 11:39.50 | Itsm | my bad.. pastebin one sec |
| 11:40.16 | Greenlight | Remember that those get evaluated WHEN RECORDING STARTS |
| 11:40.18 | Greenlight | Not when it ends |
| 11:40.19 | Itsm | http://pastebin.com/5XL80tia |
| 11:40.35 | Itsm | Yes sure, so after that, the file is being saved to something like this: |
| 11:40.51 | Itsm | 2013-06-27_10:38:19_From:_535_To:_phonenumber_.WAV |
| 11:41.22 | Itsm | now I want it to still keep the original begining of call direction, if it's out,in,or group |
| 11:41.28 | Itsm | Any idea how can I do that? |
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| 11:42.45 | Greenlight | Less of an asterisk/mixmonitor question, more bash/string manipulation |
| 11:43.33 | Itsm | I thought maybe there is some cdr variable that can achieve this |
| 11:44.06 | Greenlight | The concept of OUT and IN or GROUP is not an asteris concept, that's FreePBX |
| 11:44.40 | Itsm | I see, thank you very much. |
| 11:44.48 | Greenlight | To asterisk a call has no "direction"... |
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| 11:45.09 | Greenlight | However, to achieve your goal you're better just editing the FreePBX script that makes the filename |
| 11:45.34 | Itsm | Problem is, I don't know which file that is.. and I tried hard to search for it. lol |
| 11:45.35 | Greenlight | You can overrite in by copying it out to extension_override_freepbx.conf if memory serves |
| 11:45.51 | Greenlight | Which FreePBX version you on ? |
| 11:46.06 | ni10381 | my disa is sarting dtmf recognition only after 10 seconds after recebe the call, anyone have idea to resolve this? |
| 11:46.06 | Itsm | 2.8.1 |
| 11:46.27 | Greenlight | Ok, that's the older version. Umm two secs |
| 11:47.19 | Greenlight | Okay, you want to overwrite the macro "[macro-record-enable]" |
| 11:47.44 | Greenlight | On a sidenote, freepbx 2.10+ has much better naming of recordings |
| 11:47.52 | Greenlight | And likely puts in the info you want anyway |
| 11:48.45 | Itsm | I'm afraid that upgrading would take me into a much more then I can spare the time for in case it would break things |
| 11:49.06 | Greenlight | Yup - so you overwrite that macro inside extension_override_freepbx.conf |
| 11:49.37 | Greenlight | Can't recall which file you'll get it from, but there's only a couple to search through - start looking in extensions_additional.conf |
| 11:49.51 | Greenlight | Right - I've gotta head off now - lunch - good luck! |
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| 11:50.07 | Itsm | Thanks! |
| 11:50.11 | ni10381 | my disa is sarting dtmf recognition only after 10 seconds after recebe the call, anyone have idea to resolve this? |
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| 12:13.25 | Frontside180 | Hello, I have a weird issue I'd need help with ... We have 2 phones / extensions and they are both able to send and receive calls, however, if phone1 (just as an example) is on a call and phone2 dials or answer a call, the sound get mixed up and phone1 can no longer hear the person he was speaking with as phone2 is now receiving that audio |
| 12:13.36 | Frontside180 | anybody has seen that before? |
| 12:15.58 | geeksteve | Is there NAT involved? |
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| 12:16.34 | Frontside180 | Yes, they are remote extentions |
| 12:16.41 | geeksteve | Any sort of ALG on the remote router? |
| 12:17.09 | geeksteve | If so I'd turn that off |
| 12:17.23 | Frontside180 | what do you mean by ALG? just to be sure |
| 12:17.55 | geeksteve | Application Layer Gateway - it's a thing that sits on the router to make SIP 'work better' with NAT - but they almost always cock it up.. |
| 12:18.32 | Frontside180 | Nah there's none, it's basically a AsteriskNow server with 2 remote phones (from the same location) |
| 12:18.48 | WIMPy | ... at that remote location. |
| 12:19.21 | Frontside180 | It's a basic linksys router, could that be something within the configurations of that router? |
| 12:19.47 | WIMPy | That's what geeksteve is talking about. |
| 12:20.11 | Frontside180 | aaaah |
| 12:20.16 | WIMPy | It need not be an ALG, just any kind of SIP support on the router. |
| 12:20.45 | Frontside180 | That would make sense |
| 12:21.44 | geeksteve | Cisco/Linksys are notoriously bad at SIP stuff on cheap routers.. |
| 12:22.03 | *** join/#asterisk davlefouAMD (~david@41.225.104.31) |
| 12:22.04 | [TK]D-Fender | The RV series anyway... |
| 12:22.10 | [TK]D-Fender | the older stuff worked just fine... |
| 12:22.50 | Frontside180 | So Any Linksys router could interfere with other devices SIP even though it's not supposed to take care of that part? |
| 12:23.49 | WIMPy | No, the question is if it is supposed to take care of sip. |
| 12:24.16 | WIMPy | Many vendors think it's a good idea, but in practice it usually backfires quite badly. |
| 12:25.44 | Frontside180 | Wouldn't be the first time vendors "good ideas" backfires on them |
| 12:26.34 | geeksteve | On linksys stuff its: Administration, Management, under side heading 'Advanced Features' SIP ALG |
| 12:26.37 | geeksteve | says google |
| 12:27.07 | geeksteve | I know on their Cisco SRP52x series stuff it's terrible too - only seems to behave if you have one device on UDP one on TCP. More than two you have to turn it off!.. |
| 12:28.05 | Frontside180 | I'd make sense though, I originally had them on a freeswitch server and they had similar issues |
| 12:28.55 | geeksteve | We've had all sorts of 'odd' things happen when an ALG is involved. Registrations flipping between phones, calls only working in one direction or the other, phones going unreachable during calls. They're usually quite badly implemented. |
| 12:29.22 | Frontside180 | Good to know, I'll definately disable that by now |
| 12:29.54 | geeksteve | http://www.voip-info.org/wiki/view/Routers+SIP+ALG that lists ways for some routers, but they all vary a bit |
| 12:31.51 | Frontside180 | Thanks guys! |
| 12:32.02 | Frontside180 | (or girls, who knows) |
| 12:33.14 | *** join/#asterisk pontis (~pontis@83.137.2.144) |
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| 12:38.19 | msaraiva | Hey guys |
| 12:39.08 | msaraiva | Should the "port" parameter of a peer matter when Asterisks try to match one from a realtime source? |
| 12:39.20 | msaraiva | For a incoming connection. |
| 12:39.46 | [TK]D-Fender | no. |
| 12:39.50 | msaraiva | I have a problem that a known peer is going to default context because of this. |
| 12:40.00 | *** join/#asterisk andrewyager (~andrewyag@CPE-144-132-193-27.nsw.bigpond.net.au) |
| 12:40.38 | msaraiva | No matching peer for 'EXTEN' from 'X.X.X.X:60904' |
| 12:40.46 | msaraiva | 60904 is the source port of the peer... |
| 12:40.59 | msaraiva | And that, of course, is usually dynamic. |
| 12:42.41 | [TK]D-Fender | northing more to say with that small portion you've shown. |
| 12:42.44 | [TK]D-Fender | -r |
| 12:43.27 | msaraiva | I have a bunch of peers on a database, and all of them are working just fine. |
| 12:43.32 | msaraiva | This is a new one... |
| 12:43.55 | carrar | Any peers in your office? |
| 12:44.16 | [TK]D-Fender | Can't do an autopsy when you don't have a body... |
| 12:44.26 | msaraiva | I can select it just fine with "realtime load sippeers host X.X.X.X". |
| 12:44.28 | [TK]D-Fender | moves on to more productive matters |
| 12:45.07 | *** part/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
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| 12:48.46 | msaraiva | [TK]D-Fender: Ok, then: http://pastebin.com/q1qrv5sL |
| 12:49.13 | msaraiva | Calls from peer BLAH, which has the ip 1.2.3.4, go to default context, instead of going to "noway" context. |
| 12:49.56 | msaraiva | And the message i get when debugging sip is: "No matching peer for '123456789' from '1.2.3.4:60904' |
| 12:50.19 | msaraiva | Now you have a body... |
| 12:51.23 | [TK]D-Fender | no I don't. |
| 12:51.39 | [TK]D-Fender | I have a tiny heavily redacted snippit and not full call debug |
| 12:52.05 | [TK]D-Fender | And there is no way I'll trust that something redacted like that isn't masking a typoe |
| 12:52.34 | [TK]D-Fender | Debugging means NOT trusting things someone would like to assume are right. |
| 12:52.47 | [TK]D-Fender | And I'm not playing that game |
| 12:53.04 | msaraiva | I've changed the IP and context name, for obvious reason. |
| 12:53.19 | carrar | haha |
| 12:53.32 | msaraiva | And something else is obvious too, but nevermind... |
| 12:53.34 | [TK]D-Fender | Also don't ask for an autopsy while you're screwing with the evidence. |
| 12:53.42 | [TK]D-Fender | And that one-line isn;t full debug |
| 12:53.46 | carrar | WHy not just rot13 the whole output! |
| 12:53.50 | WIMPy | msaraiva: You can assume, that everyone here thinks that the obvious reason is that you don't want help. |
| 12:53.59 | [TK]D-Fender | we can't see what kind of actual negotiation is made |
| 12:54.06 | [TK]D-Fender | and don't know WHAT that peer even is |
| 12:54.13 | [TK]D-Fender | because those factors affect AUTH <- |
| 12:54.52 | msaraiva | The ONLY difference from that peer to the others that ARE working is the source UDP port. All the others use 5060 as the source port. |
| 12:54.58 | msaraiva | That's the reason for my first question. |
| 12:55.15 | msaraiva | And i know now that i should not trust the answer you gave to that question. |
| 12:55.44 | msaraiva | But thanks, anyway. |
| 12:55.53 | [TK]D-Fender | You are the one making claims and not showing things. |
| 12:56.05 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 12:56.09 | [TK]D-Fender | I have no reason to trust that the auth part is passing |
| 12:56.15 | [TK]D-Fender | so who cares about IP |
| 12:56.47 | [TK]D-Fender | I'm not playing this game. When you want actual help, show something we can actually help with. Everything else is a delusional waste of time. |
| 12:57.22 | msaraiva | I showed you the FULL peer configuration and the ONLY error message i get on the Asterisk CLI. |
| 12:57.36 | [TK]D-Fender | that isn't SIP DEBUG |
| 12:58.12 | [TK]D-Fender | And if you aren't looking with that enabled then you are even trying to debug anything. |
| 12:58.20 | msaraiva | I can't give you the full trace, simple as that. If you can't work with what i gave you, that's not a problem, i'm fine with it. Just don't say that it's impossible... |
| 12:58.57 | [TK]D-Fender | Don't show me a picture of a car after a crash and make me guess HOW. You are not showing the evidence.. you are showing the RESULT |
| 12:59.12 | [TK]D-Fender | the VIDEO shows the crash. |
| 12:59.16 | [TK]D-Fender | Not the "after photo" |
| 12:59.31 | [TK]D-Fender | This is just pathetic. |
| 12:59.37 | msaraiva | That's not a good analogy... |
| 12:59.42 | [TK]D-Fender | yes it is |
| 12:59.51 | [TK]D-Fender | You show a SUMMARY statement and notht eh conversation that LEADS to it |
| 13:00.09 | Greenlight | msaraiva: Why are you being so awkward when people are trying to help. |
| 13:00.10 | [TK]D-Fender | it says "bad", WHY!?!?!?!??!?! |
| 13:01.21 | msaraiva | Greenlight: I said i can't show the full trace, and it seems one's word is not trusted anymore. |
| 13:01.31 | [TK]D-Fender | msaraiva: youa re showing nothing |
| 13:01.37 | [TK]D-Fender | I don't see what auth that call comes in with |
| 13:01.39 | msaraiva | And i said if it's not possible to help with that, it's fine. |
| 13:01.43 | [TK]D-Fender | I don't see if you missed a typo. |
| 13:01.48 | Greenlight | It's not that we suspect you're lying, just that you may have made a mistake |
| 13:01.51 | [TK]D-Fender | You haven't even said WHAT is sending that call |
| 13:01.56 | [TK]D-Fender | You are messed up the head. |
| 13:02.26 | Greenlight | Probably 70o% of these types of things are simple typos or other mistakes, which are most easily solved by the steps [TK]D-Fender is tryhing to guide you through |
| 13:02.29 | [TK]D-Fender | And I'm not playing "secret squirrel" on this |
| 13:02.50 | WIMPy | msaraiva: Your IPs are not interesting to anyone. They will be scanned regularly anyway. |
| 13:02.57 | Greenlight | Indeed |
| 13:03.05 | msaraiva | It's not my ip that's being redacted, anyway. |
| 13:03.10 | msaraiva | It's the other party's. |
| 13:03.15 | Greenlight | Same thing |
| 13:03.44 | [TK]D-Fender | Whatever... I'm not wasting my time on redacted partial crap. |
| 13:03.50 | [TK]D-Fender | moves on |
| 13:03.55 | Greenlight | Don't blame you... |
| 13:03.55 | msaraiva | As they are, anyway. I get hit thousands of times per day by assholes trying to freeload... |
| 13:04.03 | msaraiva | Ok. And again, thanks. |
| 13:04.10 | *** join/#asterisk Cuzner (~ccuzner@198.41.29.45) |
| 13:04.30 | WIMPy | msaraiva: See what I mean? |
| 13:05.03 | Greenlight | msaraiva: I'm sure you'll find some paid support somewhere who'll happily believe what you say is gospel and spend hours and hours trying to work out whats going wrong based on your patial detail |
| 13:06.19 | msaraiva | Nah, i'll keep doing what i always did: use Google. |
| 13:06.28 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:06.33 | *** join/#asterisk pontis (~pontis@83.137.2.144) |
| 13:06.38 | Greenlight | Okay. Good luck. |
| 13:06.43 | msaraiva | Thanks. |
| 13:06.57 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 13:07.08 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 13:07.08 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 13:18.09 | msaraiva | Thanks God i didn't trust the inital answer. For anyone that bumps into the same problem: port parameter on sip peer DOES matter when doing IP AUTH if "insecure" parameter is not set. Set "insecure=port" and problem solved. |
| 13:19.48 | wrouesnel | is it possible to have a hint-extension monitor several devices, and blink if any one of them is in use? |
| 13:20.20 | [TK]D-Fender | wrouesnel: Yes |
| 13:20.30 | [TK]D-Fender | wrouesnel: "&" <- |
| 13:20.41 | wrouesnel | a follow up then: is it possible to have the devices it monitors be dynamically changed? |
| 13:20.57 | [TK]D-Fender | wrouesnel: like? |
| 13:21.16 | [TK]D-Fender | wrouesnel: Ah... hold on that |
| 13:21.35 | [TK]D-Fender | wrouesnel: I don't see a direct way of doing that... |
| 13:21.44 | [TK]D-Fender | You'd have to reload config changes... |
| 13:21.58 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 13:22.01 | wrouesnel | hm... |
| 13:22.10 | [TK]D-Fender | wrouesnel: What are you actually trying to accomplish? |
| 13:22.29 | wrouesnel | i've got a system which connects the phone in a room to a user's personal call queue when they login. |
| 13:22.54 | wrouesnel | since they can login in several places at once, they may have two or more devices connected at once. |
| 13:23.11 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 13:23.29 | wrouesnel | what i'd like to do is detect if they're actually talking on any of the devices in their queue. though i'd settle for knowing if their queue has calls waiting. |
| 13:24.13 | [TK]D-Fender | there is a presence state device to check for that |
| 13:24.58 | [TK]D-Fender | You could also add your memeber to the queue as a local channel instead of a direct interface and make your hint refer to that... |
| 13:25.05 | [TK]D-Fender | Should work |
| 13:25.58 | *** join/#asterisk ChadAragorn (~ChadArago@206.251.40.221) |
| 13:27.14 | wrouesnel | so how would the presence state for queues work ? |
| 13:28.19 | *** join/#asterisk [404] (~404]@12.179.117.114) |
| 13:32.08 | [TK]D-Fender | exten => 8501,hint,Queue:itg_queue ;Provide a hint for the queue |
| 13:33.31 | wrouesnel | thanks I'll give it a try. |
| 13:34.38 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 13:36.04 | Katty | GOOD MORNING CUPCAKES |
| 13:36.28 | [TK]D-Fender | Katty: Mornin' Muffin :) |
| 13:36.43 | carrar | mmm cupcakes |
| 13:36.47 | carrar | for breakfast |
| 13:36.58 | Katty | squees! |
| 13:37.01 | Katty | hugs carrar |
| 13:37.08 | carrar | woo woo |
| 13:37.19 | vlt | Hello. On Asterisk 1.8 I get many broken pipe errors when I try to originate. But NOT everytime. If I hit the originate button a few times it eventually works. Any idea what that could be? |
| 13:37.35 | wrouesnel | sounds like a bad network connection. |
| 13:41.15 | vlt | wrouesnel: Really? All the phone calls are fine. No dropouts. The manager client and the asterisk server are guests on the same physical host machine. Hmmm … |
| 13:41.24 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
| 13:41.27 | [TK]D-Fender | Originate ....button? What "button"? |
| 13:41.47 | Katty | [TK]D-Fender: the one with construction paper and tape. with originate scribbled in crayon on it. |
| 13:41.48 | vlt | [TK]D-Fender: I meant the button in my CRM app |
| 13:41.50 | Katty | [TK]D-Fender: obviously. |
| 13:42.02 | vlt | :-D |
| 13:42.13 | Katty | gosh. fender is so slow sometimes. |
| 13:42.20 | Katty | jabs [TK]D-Fender in the rib with an elbow (gently) |
| 13:42.25 | [TK]D-Fender | vlt: We have no idea how its coded... perhaps there is something off there... |
| 13:43.43 | vlt | But I have. It does the usual Action: Login … Action: Originate … sequence that worked exactly this way for more than 6 years now on asterisk 1.2 |
| 13:44.29 | Katty | rereads last sentance. |
| 13:44.46 | Katty | so, 3 major revisions later |
| 13:44.50 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 13:44.54 | vlt | Katty: Yes. |
| 13:44.55 | Katty | and you're wondering why it's acting quirky |
| 13:45.11 | vlt | Katty: Actually I do, yes. |
| 13:45.17 | Katty | okay. |
| 13:45.34 | Katty | that's /kind of/ like wondering why windows 95 apps aren't running on windows 7 correctly |
| 13:45.39 | Katty | but hey, i'm not a programmer. |
| 13:45.44 | Katty | and i don't get how all that works in asterisk. |
| 13:46.04 | vlt | Katty: It’s not NOT working because some syntac change. It’s working SOMETIMES and often not. |
| 13:46.12 | kaldemar | vlt: the manager interface has been updated more than once after asterisk 1.2. |
| 13:46.19 | Katty | so i'mma just sit right here on my tail since i'm unqualified to answer that question. |
| 13:46.28 | vlt | kaldemar: Aaah, thanks. I’ll to find more about that. |
| 13:46.37 | vlt | +try |
| 13:47.17 | vlt | looks for the new "Action: JustWorkAlways" command ;-) |
| 13:47.34 | [TK]D-Fender | vlt: And what are you running now? |
| 13:47.56 | kaldemar | vlt: Login and Originate are still valid actions. |
| 13:48.54 | kaldemar | vlt: the responses that AMI sends have changed, iirc. |
| 13:49.49 | kaldemar | vlt: even the first input when you open a connection. |
| 13:50.03 | wrouesnel | oh wait - I actually encountered this I think |
| 13:50.20 | wrouesnel | asterisk can get tetchy about originate if you don't clear the socket buffer before you log off |
| 13:50.36 | vlt | kaldemar: I tried the exact sequence that is used in the app manually via telnet. Works. Everytime I wrote it. |
| 13:51.10 | kaldemar | vlt: guess your code has some expectations that your eyes don't. |
| 13:51.33 | vlt | kaldemar: ;-) I’ll pastebin the code … |
| 13:51.42 | vlt | wrouesnel: Do you know how to? |
| 13:51.57 | wrouesnel | vlt: one sec i'll send you the php script i have which works |
| 13:51.58 | kaldemar | vlt: for example the AMI client might be waiting for "Asterisk Call Manager 1.0" after connect but your asterisk sends "Asterisk Call Manager 1.3". |
| 13:52.07 | *** join/#asterisk skirge (~skirge@196.15.233.254) |
| 13:52.36 | wrouesnel | vlt: http://pastebin.com/kUih2V9a |
| 13:53.47 | wrouesnel | I remember specifically having trouble with originate sometimes not working, and it seems to be related to some race condition between the AMI receiving a request and trying to write to the outbound socket. So to make it work I had to explicitely wait for feof on the socket before closing it. |
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| 13:55.28 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:55.38 | vlt | kaldemar: That would not explain why it worls sometimes. |
| 13:56.00 | vlt | My current code: http://pastebin.com/nihWusBz |
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| 13:56.06 | kaldemar | vlt: you might want to make sure you read all that asterisk sends your app, such as "Authentication accepted" etc. don't just blindly read until newline, EOF or 127 bytes. |
| 13:56.35 | wrouesnel | vlt: yeah your code is closing the socket right after putting the event on |
| 13:57.21 | wrouesnel | you need to read the socket till EOF (or a confirmation message that the originate is done) before you close it |
| 13:57.28 | msaraiva | vlt: You will get broken pipe if you don't read everything. Learned that the hard way. |
| 13:57.39 | *** join/#asterisk skirge (~skirge@196.15.233.254) |
| 13:57.59 | kaldemar | vlt: you know there are even PHP libraries for AMI, do you? those are supposed to taked care of this kind of issues already. |
| 13:58.19 | vlt | wrouesnel, msaraiva: Thank you both! |
| 13:58.45 | vlt | kaldemar: Thanks, maybe I should use a PHP lib. |
| 13:59.06 | msaraiva | vlt: Np! But kaldemar suggestion is a good one...you don't need to reinvent the wheel. ;) |
| 13:59.34 | wrouesnel | i think you usually end up reinventing the wheel when you "just need this one thing..." |
| 13:59.46 | msaraiva | True... |
| 14:00.49 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:00.49 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:24.05 | vlt | wrouesnel: I chose the quick solution. Again. I just put your fgets() lines into my code. Do you have any idea why I can’t use a while loop the first time between Login and Originate? |
| 14:24.52 | wrouesnel | vlt: there's no need? You only need to prevent the broken pipe - you can keep pushing data to the socket for short interactions like this. |
| 14:25.37 | vlt | wrouesnel: Damn, I didn’t understand that :-( |
| 14:26.03 | vlt | wrouesnel: How does reading 128 bytes prevent the broken pipe the first time? |
| 14:26.27 | wrouesnel | oh that's probably unnecessary |
| 14:26.32 | wrouesnel | I don't use it in my other scripts |
| 14:26.41 | wrouesnel | you're looking at *very* old code. |
| 14:26.48 | vlt | :-D |
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| 14:30.24 | igcewieling | Most people use an agi / manager library to handle all the ugly protocol stuff |
| 14:32.21 | Katty | ICE WEASEL! |
| 14:32.24 | Katty | hugs igcewieling |
| 14:34.20 | igcewieling | squirrel grrl! |
| 14:34.47 | vlt | igcewieling: Yes, I’ll have a look at them. I just needed to fix it quickly. And dirty ;-) |
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| 14:42.28 | WIMPy | Does anyone here know of a DSL modem that supports multiple VCs? |
| 14:46.19 | leifmadsen | does the sangoma adsl card do that? |
| 14:46.27 | leifmadsen | might not be what you want to deal with though |
| 14:46.57 | WIMPy | hadn't thought of using a card. |
| 14:47.30 | WIMPy | I have an ADSL card somewhere, but I'm pretty sure I'd need a really old Linux to make that work :-( |
| 14:47.40 | *** join/#asterisk itgrl (~itgrl@12.25.151.58) |
| 14:48.27 | carrar | WIMPy, I think the Cisco ADSL WIC card does |
| 14:49.07 | leifmadsen | WIMPy: ah gotcha, I have a Sangoma ADSL card still in the box in my closet |
| 14:49.22 | [TK]D-Fender | same here |
| 14:49.33 | [TK]D-Fender | Had min since ~ 2007 |
| 14:49.57 | WIMPy | And I need two VCs on the ethernet, no routing. |
| 14:50.13 | carrar | yes |
| 14:50.14 | carrar | it does |
| 14:50.21 | carrar | <PROTECTED> |
| 14:50.21 | carrar | <PROTECTED> |
| 14:50.21 | carrar | <PROTECTED> |
| 14:50.21 | carrar | <PROTECTED> |
| 14:50.21 | carrar | <PROTECTED> |
| 14:50.24 | carrar | <PROTECTED> |
| 14:50.30 | WIMPy | takes a look at what Sangoma has to offer, but has a fear for dirvers already. |
| 14:52.05 | [TK]D-Fender | Wanpipe is solid.... |
| 14:52.17 | [TK]D-Fender | There were in this game LONG before that.. |
| 14:52.45 | carrar | WIMPy: Curious why you want multiple VC's? |
| 14:53.13 | WIMPy | Using Asterisk has reminded me, not to buy hardware that isn't supported by Linux again. |
| 14:53.25 | WIMPy | carrar: For Telephony. |
| 14:53.39 | carrar | Why not use QoS instead? |
| 14:53.44 | carrar | mix em both |
| 14:53.57 | WIMPy | Not my choice. |
| 14:54.07 | carrar | that pretty much ends it right there then :) |
| 14:54.11 | WIMPy | Many Telcos deliver that way. |
| 14:54.18 | carrar | not us |
| 14:54.21 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.235) |
| 14:54.45 | WIMPy | But O2 for example. |
| 14:54.54 | WIMPy | And yes, it's annoying. |
| 14:55.13 | WIMPy | But I want to replace their hardware, because it's shit. |
| 14:55.27 | carrar | That might be nice if you have a ATM OC3 :) |
| 14:55.31 | carrar | but not DSL :) |
| 14:55.37 | WIMPy | They try to make that as hard as possible, off course. |
| 14:55.49 | carrar | yeah |
| 14:56.42 | WIMPy | Sangoma is out. Not compatible with our lines. |
| 14:57.01 | carrar | Just get a ebay 1700 with a ADSL WIC |
| 14:57.03 | carrar | cisco |
| 14:57.33 | carrar | maybe even a old 262X |
| 14:57.57 | WIMPy | Someone offered an 876. |
| 14:58.54 | carrar | We run v4/v6 with customer on the 1700 |
| 14:59.01 | carrar | do dsl bonding |
| 14:59.23 | carrar | works nicely |
| 14:59.54 | WIMPy | Will look in to that. |
| 14:59.59 | WIMPy | Or actually already do :-) |
| 15:01.16 | WIMPy | Cisco is probably extremly costly to run. That way the card idea wasn't bad. |
| 15:01.53 | carrar | cisco is one time purchase off ebay |
| 15:02.06 | carrar | unless you want to buy new |
| 15:02.09 | carrar | *SILLY* |
| 15:02.28 | WIMPy | No, but it will want lots of electricity. |
| 15:02.47 | carrar | a SUN E450 wants lots of electricity |
| 15:02.55 | carrar | cisco 1700 just a trickle |
| 15:03.17 | WIMPy | And if I don't have to pay a tenner exta per month, I don't want to. |
| 15:03.17 | WIMPy | Yes, you need an aoutomatic bak note feeder for it :-) |
| 15:03.24 | igcewieling | Cisco 175x are inexpensive |
| 15:03.46 | carrar | and they are quite, no noisy fan |
| 15:04.26 | WIMPy | At least they are compatible to out lines. |
| 15:04.31 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.235) |
| 15:04.33 | igcewieling | They are my favorite affordable router. |
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| 15:36.19 | WIMPy | carrar: There's one thing I don't see, yet: How do I get the two VSs out of the Cisco? I need the pppoe, no IP. |
| 15:38.21 | zoid_ | Do I need to do something special to dial a sip extension on the internet? I have the following defined in my pbx: exten => 804,1,Dial(SIP/1234@192.168.200.1) |
| 15:38.37 | zoid_ | That works, but if I switch to a public IP it doesn't |
| 15:39.01 | zoid_ | it fails with this error: chan_sip.c:5441 create_addr: Purely numeric hostname (805), and not a peer--rejecting! |
| 15:39.31 | Cuzner | could be NAT |
| 15:40.25 | carrar | WIMPy, you mean a layer 2 bridge across the ATM interface PVC and the ethernet interface? |
| 15:40.41 | zoid_ | using an alias from /etc/hosts doesn't work either |
| 15:41.21 | WIMPy | carrar: Yes, where the only way I see to export multiple VSc would be using VLANs. |
| 15:41.48 | zoid_ | Cuzner: I don't think so, I see no traffic out of the pbx |
| 15:41.52 | WIMPy | VSc |
| 15:41.57 | WIMPy | Damn |
| 15:42.10 | carrar | or vrf's |
| 15:42.14 | WIMPy | VCs VCs VCs VCs VCs |
| 15:43.28 | WIMPy | Without IPs? |
| 15:44.43 | Grady2000 | anyone know where the system prompts for asterisk are stored and/or if they are changeable? |
| 15:44.49 | Grady2000 | ex: the meetme menu prompt |
| 15:44.54 | *** join/#asterisk kresp0 (~kresp0@89.Red-88-15-137.dynamicIP.rima-tde.net) |
| 15:46.20 | pabelanger | Grady2000: what does asterisk.conf say |
| 15:46.54 | [TK]D-Fender | Grady2000: And yes, you can checge the files. |
| 15:46.57 | [TK]D-Fender | change* |
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| 15:48.42 | Weezey | Is it possible to use Realtime extensions table to supply hints to dialplan? priority is an integer in my table. |
| 15:49.07 | WIMPy | hint is -1 |
| 15:49.19 | WIMPy | But I don't know it will be accepted that way. |
| 15:49.34 | Weezey | WIMPy: I tried that, doesn't seem to be adding to the hints |
| 15:49.48 | Weezey | but maybe I need to specify something else? |
| 15:50.02 | Weezey | or maybe core show hints doesn't see it because it's realtime. |
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| 15:59.52 | Weezey | maybe I'm hinting in the wrong place. |
| 16:01.08 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
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| 16:05.03 | igcewieling | heh, apparently we sell "Astro" phones. 8-| |
| 16:05.19 | Weezey | far out |
| 16:07.32 | igcewieling | Weezey: realtime is like slow internet: http://theoatmeal.com/comics/no_internet Realtime has various limitations which make is too much of a hassle for many people. An alternative is to write a small script which reads the database of hints and generates dialplan. The script would be called with #exec /path/to/script.php in extensions.conf |
| 16:07.59 | igcewieling | We use this method for sip.conf peers |
| 16:10.21 | Weezey | gotcha. If I'm trying to call SIP/exten@peername from queue on boxA. I need the hint on peername, right? |
| 16:10.53 | Weezey | in whatever context boxA lands on on peername, right? |
| 16:11.45 | WIMPy | The context you use for subscriptions. |
| 16:12.08 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 16:12.10 | Weezey | hmm, maybe that's the problem, I don't have. |
| 16:12.31 | Weezey | where's that the Book at. |
| 16:12.52 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 16:13.40 | igcewieling | Weezey: hints are mainly useful for monitoring the state of devices. I don't believe they are in an required when dialing or queues |
| 16:13.55 | *** join/#asterisk djbelieny (~djbelieny@cpe-76-183-251-196.tx.res.rr.com) |
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| 16:14.51 | [TK]D-Fender | [12:10]Weezeygotcha. If I'm trying to call SIP/exten@peername from queue on boxA. I need the hint on peername, right? <- use XMPP for distributed presence for this and use that as the state device. |
| 16:15.07 | djbelieny | hey all... quick question: I have an autodialer based on asterisk and in Texas we are being required to provide an FCC reg istration for software and software for voip. anyone been through that ? |
| 16:15.47 | Weezey | [TK]D-Fender: where so I set that state device? I have XMPP talking between both boxes and it's updating states for locally available things. |
| 16:16.15 | Weezey | [TK]D-Fender: It's seeing a state for SIP/exten not SIP/exten@peername |
| 16:23.42 | [TK]D-Fender | if it's distributed... is SHOULD be to the local mapping XMPP sets up |
| 16:23.53 | [TK]D-Fender | I haven't actually done this myself... but it should NOT include the peer name... |
| 16:28.08 | *** join/#asterisk vfabi (~fabi@host-static-37-75-102-243.moldtelecom.md) |
| 16:31.46 | djbelieny | Let me change the question: Does Asterisk have an FCC/ACTA number ? |
| 16:35.59 | igcewieling | djbelieny: not that I'm aware of, but you are welcome to contact Digium. |
| 16:36.10 | djbelieny | Thanks |
| 16:36.24 | Qwell | You aren't connecting to the PSTN. Your provider is. |
| 16:36.40 | djbelieny | Thank you very much. |
| 16:36.54 | Qwell | TA == Terminal Attachments |
| 16:39.45 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 16:41.48 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 16:42.33 | gavimobile | if I was to use res_caldev to connect my pbx to my google calendar, whould I have to forward any ports to make things work? |
| 16:43.21 | *** join/#asterisk m0spf (~steve@2001:ba8:1f1:f12e::2) |
| 16:57.58 | *** join/#asterisk KenBeannet (~KenBeanne@static-96-245-214-202.phlapa.fios.verizon.net) |
| 16:58.54 | KenBeannet | Hello Gents! Got an issue with Asterisk + Skype (Sip) intergration. I am having a 1-2 second delay on calls going in or out as audio doesn't start on time (1-2 second delay). I think it has to do with my external traffic going out through one IP and coming back on another. |
| 16:59.18 | KenBeannet | And I also NAT my traffic once internal. Once the audio starts, its great quality and is perfect. |
| 17:02.06 | KenBeannet | It def has to do with Qualify as well, as if I turn that to no, no audio goes through. |
| 17:02.28 | KenBeannet | Epp, cancel that. |
| 17:04.27 | *** join/#asterisk peetaur2 (~peter@x2f08fb8.dyn.telefonica.de) |
| 17:07.43 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 17:10.22 | igcewieling | What is it with all the weird stuff today? autodialers, caldev, skype, etc. Must be Thursday, I never could get the hang of Thursdays. |
| 17:10.58 | KenBeannet | Skype is no good? :( |
| 17:11.23 | [TK]D-Fender | Of course not... |
| 17:11.34 | KenBeannet | What SIP provider do you suggest then? |
| 17:11.43 | [TK]D-Fender | And you are using an extra layer in between... you are introducing delays and points of failure |
| 17:11.48 | KenBeannet | I am trying to connect Lync to a SIP (Using Asterisk as the middle man) |
| 17:12.03 | [TK]D-Fender | And how is Lync talking to *? |
| 17:12.14 | KenBeannet | TCP Locally to the Asterisk |
| 17:12.19 | KenBeannet | Asterisk calls out to SIP |
| 17:12.20 | *** join/#asterisk c|oneman (cloneman@2605:6400:2:fed5:22:0:3b06:3913) |
| 17:12.23 | [TK]D-Fender | TCP with what? |
| 17:12.36 | KenBeannet | 5060? I dont understand the "with what"? |
| 17:12.40 | [TK]D-Fender | lAYER 7 <- |
| 17:12.45 | [TK]D-Fender | what PROTOCOL over TCP? |
| 17:12.49 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 17:13.22 | ChannelZ | Telnet! |
| 17:13.32 | *** join/#asterisk bandroidx (~bandroidx@unaffiliated/bandroid) |
| 17:13.50 | Greenlight | But Telnet is eviiiilll |
| 17:14.00 | KenBeannet | I dunno the correct answer. Sorry. |
| 17:14.12 | ChannelZ | pats backup-infobot Greenlight on the head |
| 17:14.21 | [TK]D-Fender | You are the pone how connected Asterisk to Lync... you don't even know what protocol you're using? |
| 17:14.26 | [TK]D-Fender | one* |
| 17:14.45 | KenBeannet | host=10.1.10.60 |
| 17:14.45 | KenBeannet | transport=tcp |
| 17:14.45 | KenBeannet | port=5060 |
| 17:14.56 | ChannelZ | It's speaking SIP over TCP? |
| 17:14.58 | [TK]D-Fender | Sure looks like SIP |
| 17:15.10 | Greenlight | SIP over TCP is also evil |
| 17:15.10 | KenBeannet | Yes, SIP |
| 17:15.19 | KenBeannet | Well, limited budget. |
| 17:15.27 | ChannelZ | Microsoft implementing SIP.. I can only imagine |
| 17:15.27 | [TK]D-Fender | So you need to talk SIp to Asterisk ... so Lync can talk SIP a an ITSP? |
| 17:15.27 | Greenlight | UDP costs more? |
| 17:15.41 | KenBeannet | UDP not option with Lync |
| 17:15.53 | [TK]D-Fender | And is TCP an option with the providers you have looked at? |
| 17:15.55 | KenBeannet | Yes, Lync to Asterisk, Asterisk to Skype (SIP). |
| 17:16.04 | Greenlight | Skype isn't SIP |
| 17:16.09 | ChannelZ | It is if you pay |
| 17:16.12 | Greenlight | Oh |
| 17:16.15 | KenBeannet | We pay Skype |
| 17:16.17 | KenBeannet | for SIP. |
| 17:16.18 | [TK]D-Fender | Greenlight: But his hackish gateway to it is... |
| 17:16.19 | ChannelZ | Or was, not sure if MS killed that too |
| 17:16.40 | KenBeannet | MS is coming out with Skype intergration next month in the CU of Lync |
| 17:16.41 | Greenlight | This setup sounds a recipie for disaster |
| 17:16.41 | ChannelZ | So why are you involving lync? |
| 17:16.58 | ChannelZ | <-- just wandered in |
| 17:17.04 | KenBeannet | Because I want its features (mobile phone, exchange, meetings, PC apps) |
| 17:17.17 | KenBeannet | We are all exchange/windows company. |
| 17:17.27 | KenBeannet | I use snom 760 phones as well. |
| 17:17.34 | KenBeannet | But the lag happens with Phone or PC app. |
| 17:17.35 | ChannelZ | Hmm. Well in theory it should kind of work. |
| 17:17.42 | KenBeannet | Yet, Lync to Lync works great. |
| 17:17.49 | ChannelZ | but yes the more people you put in the middle, the longer the delay will be |
| 17:18.03 | KenBeannet | But why once the call is connected, there is no lag. |
| 17:18.15 | KenBeannet | I feel like I had this problem before, we have 4 ISPs in 1 Firewall. |
| 17:18.16 | Greenlight | So, Asterisk here is a gateway so you can do SIP(TCP) <--> SIP(UDP) |
| 17:18.21 | KenBeannet | And the traffic outbound is not the same inbound. |
| 17:18.36 | KenBeannet | No, PSTN Gateway to SIP (Asterisk) |
| 17:18.40 | ChannelZ | oh so it's not a latency issue? I guess I should go back and read what your original question is |
| 17:18.44 | KenBeannet | Lync only accpets a PSTN |
| 17:18.54 | KenBeannet | No, once the call is created, the quality is amazing. |
| 17:19.09 | KenBeannet | But when I call, for example, and automated service. I miss the first 3 words. |
| 17:19.13 | [TK]D-Fender | [13:18]KenBeannetBut why once the call is connected, there is no lag. ,- because it's SETUP & REINVITES that are taking time. Once the path is negotiated then the packets just flow. |
| 17:19.51 | KenBeannet | Fender, is that statement to say that its not possible? Or that there is a misconfiguration with reinvites? |
| 17:20.05 | ChannelZ | so you have some routing issues or something where packets are getting dumped on the floor or something until they find their way. Lots of NAT involved? |
| 17:20.09 | [TK]D-Fender | You have 4 pieces all confirming their end of this chain. So when the far end accepts it takes 3 comms of chains to ack it |
| 17:20.13 | Greenlight | I wouldn't expect things to take 3 seconds to setup |
| 17:20.27 | [TK]D-Fender | Skype + SIP = stupid variable |
| 17:20.37 | KenBeannet | There is NAT. It goes out with one IP address. When coming back it comes in through another. I think that is the cause of the delay, but not certain. |
| 17:20.49 | Greenlight | Lync to Asterisk is local ? |
| 17:20.57 | Greenlight | If so, disable reinvites for that leg |
| 17:21.08 | KenBeannet | Correct Green. |
| 17:21.44 | KenBeannet | Green, I disabled for Lync ReInvites. Should NAT and Qualify be on? |
| 17:21.59 | Greenlight | It may be worth grabbing a SIP trace on the asterisk box to see what's being sent, and more importantly *when* |
| 17:22.26 | Greenlight | nat=yes will no present any issues. Best to leave qualify on for endpoints |
| 17:22.28 | KenBeannet | Green, is there a way to see the tcp traffic (Ips) in the log with miliseconds timed? |
| 17:22.45 | Greenlight | The logs have timestamps |
| 17:22.53 | KenBeannet | Only with seconds though. |
| 17:23.01 | Greenlight | Oh, you can set that in logger.conf iirc |
| 17:23.07 | Greenlight | (Someone else can confirm ... ) |
| 17:23.15 | ChannelZ | tcpdump and wireshark or something |
| 17:23.20 | Greenlight | Or that. |
| 17:23.38 | KenBeannet | for Skype, I should leave reinvite on? |
| 17:24.11 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 17:24.13 | Greenlight | Are they reinviting ? |
| 17:24.52 | Greenlight | Also, in logger.conf: dateformat=%F %T.%3q ; with milliseconds |
| 17:24.58 | KenBeannet | Ty |
| 17:26.00 | Greenlight | Damn is that the time. Gotta go. |
| 17:27.21 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 17:27.52 | ChannelZ | Time keeps on slippin, slippin, slippin.. |
| 17:28.50 | KenBeannet | [2013-06-27 13:30:09.105] VERBOSE[21698][C-0000008f] app_dial.c: -- SIP/skype-0000012b answered SIP/lync2013-0000012a |
| 17:28.51 | KenBeannet | [2013-06-27 13:30:14.155] VERBOSE[21698][C-0000008f] pbx.c: -- Executing [h@from-internal:1] Hangup("SIP/lync2013-0000012a", "") in new stack |
| 17:30.02 | [TK]D-Fender | KenBeannet: And you have novisibilty on the connection latency from SIP>Skype |
| 17:30.19 | KenBeannet | From asterisk to Skype? |
| 17:30.48 | [TK]D-Fender | No, from that SIP SERVICE you use to GET to Skype |
| 17:31.08 | KenBeannet | Skype is the SIP service. |
| 17:31.17 | KenBeannet | http://www.skype.com/en/features/skype-connect/ |
| 17:32.51 | KenBeannet | Does that answer your question? |
| 17:33.26 | [TK]D-Fender | KenBeannet: It doesn't matter that they offer it direct.. it is still an intermediary. |
| 17:33.36 | [TK]D-Fender | they still have to converse and convert |
| 17:33.43 | [TK]D-Fender | And we hav no idea of the delay that causes |
| 17:33.54 | [TK]D-Fender | The whole point is that is out of your hands |
| 17:33.59 | KenBeannet | Is there a better service for Asterisk I can use to dial through? |
| 17:34.06 | [TK]D-Fender | Yes. Any. |
| 17:34.14 | KenBeannet | Any personal suggestion? |
| 17:34.19 | [TK]D-Fender | voip.ms |
| 17:34.22 | [TK]D-Fender | vitelity.net |
| 17:34.25 | [TK]D-Fender | les.net |
| 17:34.40 | [TK]D-Fender | flowroute.com |
| 17:34.45 | [TK]D-Fender | tons more |
| 17:35.31 | KenBeannet | But it is abnormal to have that two second delay right? |
| 17:36.48 | [TK]D-Fender | after that chain and a giant unknown to a foreign protocol? I wouldn't be surprised |
| 17:37.05 | [TK]D-Fender | perhaps you should just test that theory.. |
| 17:37.49 | KenBeannet | Ill try flow right now |
| 17:38.44 | ChannelZ | Are you literally only using Skype for PSTN calls? |
| 17:39.08 | KenBeannet | Well, using Asterisk for PSTN |
| 17:39.19 | KenBeannet | For Lync connectivity. |
| 17:40.55 | *** join/#asterisk tilt_ (~tilt@173-13-180-97-sfba.hfc.comcastbusiness.net) |
| 17:41.19 | KenBeannet | Even with Flowroute. I have a 2-3 second delay after the call is picked up until audio is connected. |
| 17:41.26 | [TK]D-Fender | No, You're using Skype for PSTN. |
| 17:41.31 | [TK]D-Fender | Asterisk talks to Skype for that |
| 17:41.51 | [TK]D-Fender | Asterisk is SOFTWARE.. it does not get you to the PSTN. The side that has actual HARDWARE does that. |
| 17:42.02 | KenBeannet | I had something very similar issue before, where if I muted the phone, I could only hear the one way, yet once the phone was unmuted, then the 2 way connection was made. |
| 17:42.39 | [TK]D-Fender | Again... reinvites... |
| 17:42.43 | [TK]D-Fender | The bane of SIP |
| 17:42.44 | KenBeannet | If that makes sense, the phone had to have audio going out before the 2 way connection made. |
| 17:42.59 | KenBeannet | I solved this because the packets coming inbound were not being routed by the right ISP |
| 17:43.17 | KenBeannet | Once I changed the policy to go out through the correct ISp, that was resolved. |
| 17:46.25 | *** join/#asterisk serafie (~erin@50.58.247.162) |
| 18:00.11 | Hive | Hey all, I have an asterisk server running which people from many different places in the US connect to. Sometimes, many peers (5 to 15 or so) will go unreachable or lagged, then become reachable about 5 - 10 seconds later. It is not always people from the same location, and it is not always the same SIP peer that drops/reconnects. I tried increasing the qualify time to 10,000 on a few peers but they have still dropped |
| 18:00.11 | Hive | like the others. |
| 18:00.52 | Hive | It must be something with my server and I really have no idea where to look. If anyone has somewhere to point me I would greatly appreciate it! ( Asterisk 10.7.0) |
| 18:03.48 | ChannelZ | well increasing the qualify time makes it check less, so if you've got NAT issues or something you've gone the wrong direction. |
| 18:04.54 | ChannelZ | You need to see if it's connection-related.. traceroute to those IPs and again when they go offline and see if the traffic is dying somewhere upstream of you |
| 18:07.08 | Hive | Okay I'll try giving that a shot and seeing if I can find something. Thanks for the suggestion. |
| 18:07.17 | Hive | I'm at a loss with this issue haha |
| 18:07.31 | Hive | If it were somewhat consistant in some way it would be a lot easier to troubleshoot -_- |
| 18:08.48 | ChannelZ | aren't they all |
| 18:09.25 | *** join/#asterisk imox (~imox@24-134-17-195-dynip.superkabel.de) |
| 18:09.33 | Weezey | AHA! I've got it! |
| 18:09.42 | ChannelZ | Give it back! |
| 18:09.58 | Weezey | stabs brain with pen |
| 18:10.03 | Weezey | there it goes |
| 18:12.09 | Weezey | AddQueueMember(queuename,SIP/exten@peername,,,exten,SIP/exten) |
| 18:12.34 | Weezey | without specifying the stateinterface (last param) the queue doesn't know where to look to find the state. |
| 18:13.17 | Qwell | Weezey: That's not entirely true. If it's left off, then it'll just use the member interface |
| 18:13.19 | Weezey | combined with setting callcounter=yes in sip.conf and having any call-limit=>0 in the sip peer. |
| 18:13.45 | Weezey | Qwell: but for XMPP distributed queue that's how to make the state shine through. |
| 18:14.02 | Qwell | Weezey: sure, I'm not suggesting it isn't useful in a lot of cases |
| 18:14.19 | Weezey | It was the missing lettuce on my sandwich |
| 18:15.29 | Weezey | Now, asterisk 11 peers are working great but asterisk 1.8 peers aren't updating state, but I don't think that's a big deal, just means I need to upgrade 20 servers to 11. |
| 18:15.51 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 18:31.05 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 18:37.03 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 18:43.26 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 18:57.16 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 18:58.47 | *** join/#asterisk FonJockey (~FonJockey@23.31.156.133) |
| 19:01.25 | Katty | REESES PEANUTBUTTER CUPCAKES |
| 19:01.26 | Katty | that is all. |
| 19:03.14 | [TK]D-Fender | http://ih2.redbubble.net/image.5581031.2176/fc,550x550,white.jpg |
| 19:04.47 | *** join/#asterisk FonJockey (~FonJockey@23.31.156.133) |
| 19:10.18 | KenBeannet | Channel and Fender, I think I got it fairly close. |
| 19:10.20 | leifmadsen | Katty: fuck you |
| 19:10.23 | leifmadsen | Katty: I mean, I want |
| 19:10.37 | KenBeannet | By setting the external IP address, also I set the internal Ip specifically |
| 19:11.01 | KenBeannet | That seemed to take away about 1 full second, I think the last portion is the external IP on the 1st packet is different then the sip.address.com. |
| 19:11.26 | KenBeannet | So I am speaking with our firewall dept about getting any internal traffic that goes out by the asterisk server to go out as the sip.address.com. |
| 19:12.07 | Katty | pats leifmadsen |
| 19:12.12 | *** join/#asterisk kresp0 (~kresp0@89.Red-88-15-137.dynamicIP.rima-tde.net) |
| 19:12.12 | leifmadsen | :) |
| 19:12.20 | leifmadsen | <3's reeses peanut butter cups so hard |
| 19:12.27 | leifmadsen | Katty: reeses peices are also fantastic |
| 19:12.30 | leifmadsen | pieces* |
| 19:13.35 | Katty | http://42ndrecipestreet.blogspot.com/2011/10/reeses-peanut-butter-cup-cupcakes.html <- hop to it, buttercup! |
| 19:13.48 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 19:14.33 | apb1963_ | i'm more of a brownie with or w/out nuts or oatmeal cookie with raisins fan. |
| 19:15.00 | apb1963_ | Hmmm...oatmeal brownies.... <patent pending> |
| 19:15.07 | Katty | http://42ndrecipestreet.blogspot.com/2011/02/best-brownies.html <- i got that too |
| 19:15.20 | apb1963_ | I just do the box thing |
| 19:15.24 | Katty | whattttttttttttttttttt |
| 19:15.39 | Katty | i guess that works if you don't cook much (= |
| 19:16.06 | sruffell | 'firewall dept' sounds ominous |
| 19:16.17 | apb1963_ | I only cook when I have to |
| 19:16.37 | apb1963_ | t is teaspoon and T would be tablespoon then... yes? |
| 19:16.50 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 19:17.12 | apb1963_ | Damn I forgot my real question |
| 19:18.11 | apb1963_ | oh I remember now |
| 19:18.50 | apb1963_ | Is there a way to make meetme "boink" only to the existing particpants? So new entrants don't hear any entry sounds? |
| 19:19.35 | Katty | yes t is teaspoon and T is tablespoon |
| 19:19.44 | igcewieling | apb1963_: if there is it would be documented in "core show application meetme" and the meetme.conf file |
| 19:19.54 | Katty | hi mister ice weasel |
| 19:20.06 | apb1963_ | wonders where Mr. Tablespoon buys his jewelry. |
| 19:20.20 | apb1963_ | igcewieling: cool. TY |
| 19:20.52 | WIMPy | Katty: You have to be nice to igcewieling today. We agreed on one topic yesterday. |
| 19:21.04 | Katty | oh? |
| 19:21.08 | Katty | what did you agree on? food is tastey? |
| 19:21.24 | Katty | that it's always Qwell's fault? |
| 19:21.31 | WIMPy | No, about the usibility of phones. |
| 19:21.44 | Katty | go on. |
| 19:21.46 | igcewieling | Katty: everyone knows it is always Qwell's fault. |
| 19:21.56 | Katty | mnmm |
| 19:21.57 | Katty | mhmm |
| 19:22.09 | WIMPy | Is that his job title? |
| 19:22.18 | igcewieling | hands Katty some fresh baked sourdough from about 8 hours in the future |
| 19:22.42 | Katty | yay! |
| 19:22.56 | Katty | i have a loaf of amish white bread in the freezer, if you want it (= |
| 19:23.42 | apb1963_ | blah... it's backwards... you can |
| 19:24.02 | apb1963_ | set it for all participants... or just the first one... but not subsequent |
| 19:24.16 | igcewieling | Katty: no thanks 8-| |
| 19:24.34 | Katty | well how about some shepherd's pie instead? |
| 19:24.53 | Katty | the real stuff. with lamb and rosemary and cinnamon! |
| 19:26.04 | apb1963_ | wonders why there are strippers in a lamb pie |
| 19:26.11 | *** join/#asterisk Cuzner (~ccuzner@198.41.29.45) |
| 19:26.25 | apb1963_ | wonders why there's lamb in a pie. |
| 19:26.48 | Katty | i've never heard of a stripper named rosemary |
| 19:27.07 | apb1963_ | When's the last time you were at a strip club? |
| 19:27.15 | KenBeannet | is there a way to upgrade easy from 11.3 to 11.4? |
| 19:27.56 | Katty | uhhh |
| 19:27.58 | Katty | ponders |
| 19:28.02 | Katty | 2 or 3 years i think |
| 19:28.13 | Katty | something like that. maybe 2ish |
| 19:29.19 | apb1963_ | Well Rosemary got pregnant and decided to stop stripping. |
| 19:29.30 | Katty | >.< |
| 19:29.32 | Katty | i loled at that. |
| 19:29.36 | Katty | that's awful |
| 19:29.51 | apb1963_ | Yeah, the patrons weren't too happy either. |
| 19:30.30 | igcewieling | for the stripper conversation http://www.tshirthell.com/funny-shirts/i-support-single-moms |
| 19:31.39 | apb1963_ | i can't afford to support single moms |
| 19:31.52 | apb1963_ | (that's why they're single) |
| 19:32.49 | apb1963_ | So do you have a good recipe for broccoli pie? |
| 19:35.03 | ChannelZ | no such thing |
| 19:35.41 | apb1963_ | Well I figure if she knows how to make sheep into pie, she can cook anything. |
| 19:36.06 | ChannelZ | Broccoli smells like feet |
| 19:36.18 | apb1963_ | yes but it tastes like chicken |
| 19:36.33 | WIMPy | You can't have broccoli pie before you finish your PASCAL coding. |
| 19:36.45 | apb1963_ | I also have to wonder where your feet have been. |
| 19:37.11 | apb1963_ | Does anyone still use pascal? |
| 19:37.14 | WIMPy | In Edam. |
| 19:37.18 | ChannelZ | on my feet |
| 19:37.33 | apb1963_ | you use pascal on your feet? |
| 19:37.34 | ChannelZ | Steamed broccoli smells like feet anyway |
| 19:37.55 | apb1963_ | Not if you mix it with bok choy, spinach and butter. |
| 19:38.04 | WIMPy | Does it als taste the same? |
| 19:38.14 | apb1963_ | just don't forget the onions, garlic cloves and mushrooms |
| 19:38.14 | WIMPy | also |
| 19:38.47 | ChannelZ | If you need all that other stuff to make it taste good, it's easier just not to eat broccoli |
| 19:39.05 | apb1963_ | nobody ever said it was going to be easy |
| 19:40.42 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 19:41.16 | apb1963_ | they're talking about toast in #freepbx. Very strange. |
| 19:41.23 | WIMPy | It's more than 25 years since I last looked at PASCAL, but I like broccoli anyway. |
| 19:41.45 | apb1963_ | I never liked pascal much, but broccoli grew on me. |
| 19:41.56 | ChannelZ | <insert fungus joke> |
| 19:42.02 | WIMPy | You're a zombie? |
| 19:42.03 | atan | Any chance the Cisco supports the sidecar w/SIP firmware? |
| 19:42.13 | *** join/#asterisk jeev (~j@unaffiliated/jeev) |
| 19:42.14 | ChannelZ | The Cisco? |
| 19:42.25 | apb1963_ | Cisco Kid was a friend of mine |
| 19:42.44 | atan | Yes the finicky Ciscos :-( if it's anything like the 79xx I doubt the cars work with SIP, but figured I might poke in and ask :D |
| 19:43.03 | apb1963_ | Cisco builds cars now? |
| 19:44.09 | atan | apb1963_, they run on ios |
| 19:44.11 | ChannelZ | god help you to get in one |
| 19:44.34 | apb1963_ | I gotta get back to work. |
| 19:44.40 | Katty | two words. BOSE suspension. |
| 19:44.45 | apb1963_ | gets into his Cisco and takes off |
| 19:44.52 | ChannelZ | Are the sidecars a separate device? I figured they just talked to the phone and it did things. |
| 19:44.59 | WIMPy | Yes, they built cars. On one series the brakes didn;t work, but they refused to repair them, because some users might have gotten used to crashing in to everything. |
| 19:45.06 | Katty | oh. wait. wrong cars. |
| 19:45.10 | Katty | should've known better. |
| 19:45.22 | Qwell | WIMPy: braking is supported on the MGCP version |
| 19:45.32 | Katty | HAI QWELLERY |
| 19:46.14 | WIMPy | doesn;t know much more about MGCP other than it exists. |
| 19:47.57 | ChannelZ | So, how about that WebRTC!? |
| 19:48.14 | WIMPy | Gesundheit! |
| 19:50.50 | KenBeannet | Channelz |
| 19:50.54 | KenBeannet | Also< i switch to flowroute |
| 19:50.59 | KenBeannet | And its much better than Skype |
| 19:51.42 | ChannelZ | Not shocked |
| 20:09.39 | tilt_ | hello all, when using realtime can you also add parameters to a peer with the [XXX](+) in sip.conf? Doesnt seem to be working for me. |
| 20:10.15 | *** join/#asterisk Free99 (~Free99@50.12.22.59) |
| 20:11.44 | Free99 | hello everyone. what does this mean? chan_sip.c:14398 check_auth: username mismatch, have <4441012001>, digest has <> |
| 20:27.55 | WIMPy | So, do we get 11.5 for a nice upgrade friday tomorrow? |
| 20:32.30 | Free99 | wow man. Asterisk is upgrading seemingly as quickly as firefox haha |
| 20:33.06 | igcewieling | upgrading withing a major version is generally trivial |
| 20:34.23 | Free99 | hey igcewieling, any idea what this is about? chan_sip.c:14398 check_auth: username mismatch, have <4441012001>, digest has <> |
| 20:34.38 | Free99 | this is pertaining to a device (a linksys pap2t) with two lines on it, one anonymous and one with registration info |
| 20:34.43 | igcewieling | Free99: likely harmess |
| 20:35.06 | Free99 | my guess is that the * box is having trouble figuring out which is which |
| 20:35.36 | Free99 | igcewieling, issue is that they cannot make any calls with the anonymous line... its supposed to trigger my AGI script to ask them to authenticate |
| 20:36.17 | Free99 | the registered line works fine, but... not the anonymous line |
| 20:36.17 | Free99 | weird thing is, anonymous line on our other devices does in fact work |
| 20:37.28 | Free99 | so just to be sure, odds are low that its the asterisk server right? since the others work just fine |
| 20:39.00 | igcewieling | Free99: if others work, then it is likely an endpoint config issue |
| 20:39.33 | igcewieling | I've seen similar issues if I do a sip reload when a device is registering, just have to wait for the device to re-register and everything is OK |
| 20:39.40 | igcewieling | s/issues/messages/ |
| 20:41.16 | Free99 | oh dang, that was pretty cool lol |
| 20:41.21 | Free99 | s#dang#cool |
| 20:41.36 | Free99 | :-/ |
| 20:43.04 | igcewieling | #asterisk is steeped in coolness. |
| 20:46.38 | WIMPy | So cool we even have ice waesels :-) |
| 20:48.13 | igcewieling | The question is: Is #asterisk cool because we have ice weasels or do we have ice weasels because #asterisk is cool? |
| 20:49.01 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 20:49.04 | WIMPy | I want to call my lawyer. |
| 20:53.25 | ChannelZ | get your checkbook |
| 20:54.29 | Free99 | LOL |
| 20:54.41 | Free99 | in which order remains to be determined |
| 20:55.00 | ChannelZ | "Billing starts when phone rings" |
| 20:55.17 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 20:58.12 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 21:03.13 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:07.37 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 21:09.10 | atan | Trying to find the cause of this warning: db.c: Couldn't execute statment: SQL logic error or missing database. Is there a way to get something a little more verbose? I want to know what database it's trying to work with, and the query |
| 21:29.35 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 21:38.27 | Mon|A|rch | hey guys, do you know if there's an answering machine detection system that you can buy? |
| 21:38.49 | Mon|A|rch | asterisk seems to have at best a 15-30% fail rate |
| 21:39.03 | Mon|A|rch | or 90% if set up badly |
| 21:40.13 | *** join/#asterisk bandroidx (~bandroidx@unaffiliated/bandroid) |
| 21:40.47 | *** join/#asterisk c|oneman (cloneman@2605:6400:2:fed5:22:0:3b06:3913) |
| 21:41.04 | WIMPy | That's why the right configuration is to secret and sold at open ended rates. |
| 21:41.30 | Mon|A|rch | :( |
| 21:41.40 | Mon|A|rch | :((( |
| 21:41.47 | WIMPy | +p |
| 21:43.12 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.162) |
| 21:45.40 | Mon|A|rch | i guess I'll just try and get as low a fail rate as possible |
| 21:45.42 | Mon|A|rch | poop |
| 21:47.50 | WIMPy | Are yoy checking REDIRECTING information as well? |
| 21:49.07 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 21:50.29 | *** part/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 21:55.49 | Mon|A|rch | no |
| 21:55.56 | Mon|A|rch | should I be? |
| 21:56.26 | WIMPy | Network hosted VM usually sends diversion information. |
| 21:57.29 | Mon|A|rch | that would be helpful to know i suppose |
| 22:02.34 | Mon|A|rch | how do I get that information? |
| 22:03.38 | WIMPy | core show function REDIRECTING |
| 22:06.08 | Mon|A|rch | thanks |
| 22:13.45 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.182) |
| 22:14.45 | Mon|A|rch | WIMPy, which data type implies that there's been redirection to a VM? |
| 22:15.02 | Mon|A|rch | the to-name, to-tag etc? |
| 22:16.09 | WIMPy | The to-num, if you know it. But just checking it for non-empty might be enough. |
| 22:16.30 | Mon|A|rch | alright |
| 22:17.24 | Mon|A|rch | well, I should know the to-num shouldn't I? if I dial out to some number, if to-num isn't that number, has it probably redirected? |
| 22:18.04 | WIMPy | If it wasn't redirected it should be empty. |
| 22:18.14 | Mon|A|rch | gotcha |
| 22:19.21 | WIMPy | BTW: Does someone know if CONNECTEDLINE is r/w or write-only? |
| 22:20.13 | Mon|A|rch | so, would a dialplan excerpt look like: |
| 22:20.14 | Mon|A|rch | exten => blah,1,Dial(SIP/somestuff) |
| 22:20.15 | Mon|A|rch | <PROTECTED> |
| 22:20.43 | Mon|A|rch | do i need to originate to use that? |
| 22:20.54 | WIMPy | No, that would be executed after the call ends. |
| 22:21.00 | WIMPy | Yes. |
| 22:21.23 | WIMPy | You need to have one leg call out and the other connected to your dialplan. |
| 22:22.06 | WIMPy | I guess that means using MASTER_CHANNEL. Never tried that. |
| 22:28.09 | Mon|A|rch | WIMPy, would this work: |
| 22:28.20 | Mon|A|rch | I've got a system that sends call files to asterisk |
| 22:28.33 | Mon|A|rch | the channel is local/ext@context |
| 22:28.37 | Mon|A|rch | and it dials out there |
| 22:28.50 | Mon|A|rch | then it hits another context after that's picked up |
| 22:29.07 | Mon|A|rch | and dials an internal number in our network |
| 22:29.27 | Mon|A|rch | if I check the to-num in between those two dial()s |
| 22:29.36 | Mon|A|rch | would that do what I'm hoping for? |
| 22:30.07 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 22:30.07 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 22:30.49 | WIMPy | Sounds about right. |
| 22:31.07 | Mon|A|rch | good deal |
| 22:31.14 | Mon|A|rch | I guess I'll test it out |
| 22:31.23 | WIMPy | Not sure if you have a chance to detect diversions before the call is answered, wich might be nice. |
| 22:34.30 | Mon|A|rch | well |
| 22:34.39 | Mon|A|rch | I tried not doing Local/ |
| 22:34.56 | Mon|A|rch | but I was having huge problems getting SIP/whatiwant to work |
| 22:35.07 | Mon|A|rch | was just hanging up |
| 22:36.05 | WIMPy | There shouldn't be a need for a local channel on the first leg, unless you want to do more than just placing a single call. |
| 22:37.02 | Mon|A|rch | other than some logging and AMD, there isn't really |
| 22:37.12 | Mon|A|rch | I'll pastebin the call file |
| 22:39.19 | Mon|A|rch | http://pastebin.com/9sibpVax |
| 22:39.25 | Mon|A|rch | that's pretty much it |
| 22:41.44 | WIMPy | You hangup() if there's no redirecting. |
| 22:42.18 | Mon|A|rch | er, i just coded that in |
| 22:42.21 | Mon|A|rch | haven't tested that yet |
| 22:42.24 | Mon|A|rch | obviously a problem |
| 22:42.57 | Mon|A|rch | but, in concept, that should detect redirection |
| 22:44.08 | WIMPy | I think you need to explicitely need to check the oter channel using MASTER_CHANNEL. |
| 22:44.39 | Mon|A|rch | why is that? |
| 22:45.25 | WIMPy | You're on the 2nd channel there. |
| 22:45.28 | Mon|A|rch | will the redirecting info go out of scope? |
| 22:52.03 | *** join/#asterisk dfighter (~lka@arcemu/staff/dfighter) |
| 22:55.45 | *** join/#asterisk dwayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 22:56.06 | *** join/#asterisk newtonr (~newtonr@64.34.219.47) |
| 22:56.06 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 23:00.22 | *** join/#asterisk sawgood (~sawgood@23-24-214-61-static.hfc.comcastbusiness.net) |
| 23:04.52 | Mon|A|rch | so, WIMPy, if i enclose REDIRECTING() like ${MASTER_CHANNEL(${REDIRECTING(to-num)})}, will that grab the master channel's version of the redirection info? |
| 23:05.41 | Katty | looks at WIMPy |
| 23:06.59 | Mon|A|rch | er, without that extra set of curlies |
| 23:07.18 | Mon|A|rch | ${MASTER_CHANNEL(REDIRECTING(to-num))} |
| 23:08.58 | Katty | stares at WIMPy |
| 23:27.14 | WIMPy | Sorry, trying to build some statistics. |
| 23:27.31 | WIMPy | Yes, I think that's the way is has to be. |
| 23:27.32 | ChannelZ | 3 out of 4 people are bored. |
| 23:28.00 | WIMPy | Katty: You only stare because you can't see me. |
| 23:28.21 | Katty | aww |
| 23:28.23 | Katty | pats WIMPy |
| 23:29.24 | WIMPy | miaouws |
