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| 01:00.58 | Valduare | hi guys |
| 01:01.14 | Valduare | can I use asterisk with a google voice number |
| 01:27.10 | Penguin | Yes. |
| 01:30.05 | Valduare | very interesting. |
| 01:30.33 | Penguin | https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 01:30.56 | Valduare | I still dont know too much about what asterisk is exactly, but it sounds like i could have google voice for my buisness number and with asterisk collecting and handling all the call routing - I could route calls to sales repâs cell phones? |
| 01:31.18 | Penguin | Potentially. |
| 01:31.41 | Valduare | isnt gv dropping xmpp er did already? |
| 01:31.46 | Valduare | just reading through the link you gave me |
| 01:31.54 | Penguin | They never said they were. |
| 01:31.58 | Penguin | They said they were dropping "support" for it. |
| 01:32.12 | Penguin | It's currently still working. |
| 01:32.27 | Valduare | well thats ambiguous lol |
| 01:32.42 | Valduare | didnt know I could ever call up google and have them give me support for anything :P |
| 01:32.56 | Penguin | Why? They stopped giving support for their xmpp channel. It seems clear to me. |
| 01:33.15 | Penguin | Well, commercial support isn't the same as end user support. |
| 01:33.19 | Valduare | what are my options if I get this all setup and next week the connections stop working |
| 01:33.32 | Penguin | Change to a real ITSP. |
| 01:33.54 | Valduare | I beleive I am able to port gv number out to another carrier |
| 01:34.03 | Penguin | You can. |
| 01:34.17 | Valduare | very interesting |
| 01:34.29 | Penguin | I haven't done it personally, so I only know the procedure from the documentation. |
| 01:34.35 | Valduare | ok |
| 01:34.47 | Valduare | and then about routing calls to cell phones etc |
| 01:34.52 | Valduare | is it complicated? |
| 01:34.58 | Penguin | Time to learn asterisk. |
| 01:35.00 | Penguin | ~book |
| 01:35.00 | 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 |
| 01:35.36 | Penguin | It's not really that complicated, but if all you know how to do is plug a phone wire into a jack and dial numbers, it might be a little intimidating. |
| 01:35.39 | Valduare | poking around learning the capabilities before im nose in a book about the technical details . |
| 01:36.28 | Valduare | I havnt played with pbx or come across it ever - but I run servers for everything else i need. |
| 01:36.45 | Penguin | Do you use vim to configure services? |
| 01:37.01 | Valduare | I have but perfer nano |
| 01:37.39 | Penguin | Does nano provide coloring for syntax? |
| 01:37.56 | Valduare | mine dosnt but I imagine it could. |
| 01:38.27 | Penguin | I recommend vim with syntax on. It makes configuring the asterisk files pretty simple. |
| 01:40.28 | Penguin | You can do it all in black and white, but the coloring makes things more clear to me. |
| 01:42.16 | Valduare | iâll check it out some time. |
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| 09:04.54 | Penguin | This seems new: http://pastebin.centos.org/14241/59742114/ |
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| 09:12.02 | ChannelZ | Hmm. I seem to remember having to install some xml lib when I compiled * 13, but I've never seen that stuff as runtime warnings |
| 09:12.36 | Penguin | It showed up when restarting (core restart now). |
| 09:12.45 | Penguin | I never saw it before... or at least didn't notice it. |
| 09:12.52 | WIMPy | That looks like after the upgrade that changed from one file to a dir full of files. |
| 09:13.14 | WIMPy | But that must have been 1.4>1.6 or something. |
| 09:15.15 | Penguin | I'm wondering if I haven't seen it before because I never was on the console when asterisk started. Typically I would start asterisk first and then connect. I may not have restarted it from the CLI before now. |
| 09:17.07 | Penguin | I'll want to put in whatever it's complaining about, though, because if I see those warnings too often, they will bother me. |
| 09:18.24 | ChannelZ | not really seeing it here |
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| 09:44.53 | Geek-Linux | Hi All: I am using progress() application in my dialplan. from the telco side call hangs up. but when i use Answer() it works fine. Is there any parameter in sip.conf that i set and the call continues. I have achieved the goal in SS7 when i set can_connect=no and it sends ACM packet where the call continues. Any help would be highly appriciated. |
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| 09:49.39 | WIMPy | We need more input. How are you connected to that telco? |
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| 10:10.26 | Geek-Linux | WIMPy: I m connected to the telco over the sip. |
| 10:13.12 | _omer | Hi, I have a SIP PEER that allows RTP on 3000 to 5000 port range. CAn I specify RTP port range to a specific SIP Peer? |
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| 10:16.42 | Geek-Linux | WIMPy: Any luck ? |
| 10:17.45 | WIMPy | Geek-Linux: Doesn't look like you got the lucky version of SIP there. |
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| 10:18.04 | WIMPy | Isn't there a progressinband or something? |
| 10:18.25 | WIMPy | _omer: No. And you can only specify the local ports anyway. |
| 10:18.43 | Geek-Linux | I have tried progressinband but still getting issue :( |
| 10:22.17 | _omer | WIMPy: then what is the solution when other SIP Server is strictly listening on 3000 to 5000 RTP port range? Changing port range from RTP.Conf will affect on all the peers. |
| 10:23.35 | WIMPy | It won't have any effect. |
| 10:24.01 | WIMPy | You can't change the remote port. Everyone can only change their own ports. |
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| 10:29.52 | _omer | WIMPy: what is the solution then? |
| 10:30.03 | WIMPy | To what? |
| 10:32.48 | _omer | a SIP Server is strictly listening on 3000 to 5000 RTP port range. I cant change RTP port range from RTP.conf because it will affect on all the peers so what should I do then? |
| 10:34.41 | WIMPy | Read my answer. |
| 10:35.14 | WIMPy | You don't change their ports. Never. |
| 10:35.25 | WIMPy | Asterisk sends to where they tell it to send to. |
| 10:35.32 | WIMPy | There is nothing to be done. |
| 10:37.38 | _omer | so you mean to say, if I open asterisk port range from 3000 to 20000 then it will automatically select any port from 3000 to 5000 for this specific peer or what? |
| 10:38.18 | Chainsaw | _omer: As long as the remote end does not lie about its capabilities... the correct thing will happen. |
| 10:39.04 | Chainsaw | _omer: The remote end dictates the port for the outwards RTP stream. |
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| 12:55.47 | aster1sk | Good morning, boos is asking for an 800 toll-free; I am looking for recommendations of suppliers. |
| 12:56.02 | aster1sk | *for suppliers |
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| 13:12.50 | wdoekes | aster1sk: perhaps you should state your region first. (not everyone lives in NA) |
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| 13:51.32 | Mango45 | aster1sk: http://www.anveodirect.com/prices/tollfree |
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| 14:17.43 | aster1sk | Thank you, and yes Canada |
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| 14:25.06 | hammam | Hello all |
| 14:26.09 | hammam | we have a an Asterisk based PBX, we would like to integrate it with our CRM system. how can this be accomplished? |
| 14:27.34 | [TK]D-Fender | hammam: "integrate" can mean anything including affixing them together via duct tape. |
| 14:27.46 | [TK]D-Fender | hammam: You'll have to be more specific about what you actually want to accomplish |
| 14:27.53 | aster1sk | I found the easiest solution is to cURL the CRM API any time there is an event on the PBX |
| 14:28.19 | aster1sk | And for c2c use the Asterisk manager interface. |
| 14:29.38 | hammam | okay here's the requirement, Once a call is recieved the system will look up the telephone number from the database, if found the operator would be presented with he sored details of the caller |
| 14:30.18 | aster1sk | Is it a propreitary CRM or something like Sugar / Tiger? |
| 14:31.04 | hammam | it's a custom made CRM |
| 14:31.36 | hammam | built using .net framework |
| 14:31.49 | aster1sk | Ahh so not web based then. |
| 14:31.53 | hammam | no |
| 14:31.54 | [TK]D-Fender | hammam: It's your dialplan do whatever you want. Go call a script when the call comes in and do your lookup and push the info to your user however you want |
| 14:32.46 | aster1sk | The reason I suggested cURL was I immediately assumed it was a web-based CRM. |
| 14:32.48 | hammam | any web resources covering the technicality of this? |
| 14:32.54 | [TK]D-Fender | ~book |
| 14:32.55 | 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 |
| 14:32.56 | [TK]D-Fender | ^^^ |
| 14:33.12 | [TK]D-Fender | AGI is a common choice |
| 14:33.24 | WIMPy | hammam: It's your system. If it's custom I doubt you find anything you don't know, yet. |
| 14:33.40 | [TK]D-Fender | Or if you just need to do it based on the CallerID number, then just use SHELL so System and pass it as a parameter |
| 14:34.03 | WIMPy | The Asterisk side is easy. Call whatever you want from your dialplan or listen on AMI for the wanted information. |
| 14:34.58 | hammam | callerID will be sufficient |
| 14:35.33 | [TK]D-Fender | hammam: So do a System() application call or SHELL() function call and pass the callerid to it |
| 14:36.39 | hammam | I'm quite new to Asterisk, I need go through the mentioned book above |
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| 14:39.43 | hammam | while searching the web, i came accross something called AMIConnector which is a .net library, but couldn't find the download link. can this be of any help? |
| 14:40.17 | WIMPy | Probably. |
| 14:40.23 | [TK]D-Fender | possibly |
| 14:40.34 | [TK]D-Fender | but I don't see a need for AMI here yet |
| 14:40.48 | [TK]D-Fender | and that would be far more complex if it isn't necessary |
| 14:41.24 | [TK]D-Fender | hammam: You need to learn how your system processes calls right now to see where you could just mod the dialplan to add your system call |
| 14:41.25 | hammam | I think a solution based on callerID would be suffecient |
| 14:41.48 | [TK]D-Fender | hammam: We're discussing WHERE to hook in to do that |
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| 14:42.13 | [TK]D-Fender | hammam: Yes you obviously need the callerid. |
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| 14:42.16 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:43.16 | tomodachi | should i use the ilbc codec or g711 over the iax trunk between two pbxes?1 (phone on each side just support g711) |
| 14:43.40 | tomodachi | they usually call in to a conference room though so its not direct calls |
| 14:44.00 | [TK]D-Fender | tomodachi: Direct and conference don't mean anything here |
| 14:44.23 | [TK]D-Fender | tomodachi: If the phones are talking G.711 the translating to anything else only degrades your quality |
| 14:44.29 | tomodachi | what im thinking is that if its a can reinvite call ,then the phones will talk between themselves over our vpn hence not going over the trunk?! |
| 14:45.03 | [TK]D-Fender | tomodachi: Are your phones IAX as well? |
| 14:45.05 | tomodachi | nope |
| 14:45.12 | tomodachi | sip |
| 14:45.13 | [TK]D-Fender | tomodachi: There there is no reinvite |
| 14:46.02 | [TK]D-Fender | tomodachi: You can have an IAX call know there are 2 SIP ends and then somehow tell those 2 to connect |
| 14:46.07 | [TK]D-Fender | tomodachi: NOT happening. |
| 14:46.23 | tomodachi | ok |
| 14:46.41 | Penguin | can have? |
| 14:46.52 | [TK]D-Fender | can't* |
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| 14:47.22 | tomodachi | whatabout if i have a conf room in one end where both phones on each pbx call in to , no point in using g722 on the trunk itself?! |
| 14:47.39 | tomodachi | each phone again using g711 |
| 14:47.50 | [TK]D-Fender | [09:44][TK]D-Fendertomodachi: If the phones are talking G.711 the translating to anything else only degrades your quality |
| 14:48.06 | tomodachi | OK, that makes it clear |
| 14:48.07 | tomodachi | thxn |
| 14:54.53 | hammam | thank you all |
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| 14:56.31 | alami | hello, i have a new istall of asterisk 13 on centos 7, but i can't register with sip wsing softphone |
| 14:56.50 | alami | do i have to change firewall rules? |
| 14:57.00 | tenspeed705 | Good morning folks. I am trying to enforce a password policy on an existing server. Our default mailboxes are 10 digit, and passwords are 4 digit...This is going to cause issues with using forcename and forcegreet options (probably wrong, and to lazy to look it up - you guys get the point). Any ideas as to how I can do this? |
| 14:58.52 | [TK]D-Fender | [09:56]alamido i have to change firewall rules? <- maybe. Then again there could dozens of different reasons why it's not working. Do you SEE packets arriving? |
| 15:03.48 | WIMPy | Hmmm. |
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| 15:11.11 | alami | [TK]D-Fender: i run now tcpdump and i can see packet between the asterisk and the host with the softphone |
| 15:12.03 | [TK]D-Fender | alami: "sip set debug on" <- what you should be doing from * CLI |
| 15:12.17 | [TK]D-Fender | alami: Go see how the conversation is actually going |
| 15:13.56 | alami | [TK]D-Fender: that's the Problem, i have set debug on, but i asterisk CLI don't output anything |
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| 15:14.30 | [TK]D-Fender | alami: Well if you see it in tcpdump, but not CLI, either you're debugging the wrong channel driver (chan_sip vs pjsip), or your firewall is blocking it |
| 15:14.43 | [TK]D-Fender | alami: So the easy first step is to trash your firewall |
| 15:18.56 | *** join/#asterisk smellis_werk (~smellis@67.20.30.132) |
| 15:19.38 | alami | [TK]D-Fender:so the firewall is now inactive, but still have the same issue, any idea where to continue? |
| 15:20.19 | alami | module show like chan_sip.so return 1 modules loaded |
| 15:20.39 | tenspeed705 | alami: Just out of curiosity, as I have seen other users do this, is the softphone running on the same host as the PBX? |
| 15:21.03 | alami | tenspeed705: different subnet |
| 15:22.06 | tenspeed705 | is the softphone is being a NAT firewall? |
| 15:22.09 | *** join/#asterisk pplatek (~pplatek@74-216-249-25.dedicated.allstream.net) |
| 15:22.11 | tenspeed705 | behind* |
| 15:22.58 | alami | tenspeed705: actually i have a more then asterisk, and the softphone can register with other * instance, only with the new one not |
| 15:23.12 | alami | so the Problem is with the installations :-) |
| 15:23.21 | alami | perhaps i have miss something |
| 15:24.40 | tenspeed705 | alami: possible, but doubtful. If the sip drivers are running, and asterisk is running, I wouldn't say its install errors, sounds more config/network |
| 15:24.40 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 15:26.42 | tenspeed705 | do you have/can install ngrep on your Asterisk system? |
| 15:27.14 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ndmfxdyixnistgkr) |
| 15:27.14 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:29.58 | alami | tenspeed705: tcpdump do the samething or not? |
| 15:30.35 | tenspeed705 | yeah. that would work. Can you put the output you got in a pastebin? |
| 15:30.36 | [TK]D-Fender | alami: Go prove you've flushed the rules |
| 15:31.06 | alami | tenspeed705: that was my next idea :-) |
| 15:31.27 | tenspeed705 | alami: Yeah, do what [TK]D-Fender says also. is this a hardware firewall, or just using iptables? |
| 15:31.51 | *** join/#asterisk e4voip (uid13742@gateway/web/irccloud.com/x-crvfimemwdkvrklx) |
| 15:32.33 | alami | ofcourse, i have stop the firewall |
| 15:33.19 | alami | that's only the centos default iptables, i don't have firewall between softphone and asterisk |
| 15:34.07 | tenspeed705 | can you get us the output of the TCPDUMP and of iptables -nL |
| 15:34.52 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 15:37.44 | alami | http://pastie.org/9758364 iptables -nL |
| 15:39.27 | [TK]D-Fender | [10:30][TK]D-Fenderalami: Go prove you've flushed the rules |
| 15:39.42 | [TK]D-Fender | iptables --flush |
| 15:39.43 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 15:41.34 | *** part/#asterisk aster1sk (~vivi@135-23-109-73.cpe.pppoe.ca) |
| 15:41.55 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 15:44.11 | alami | [TK]D-Fender: so thanks it's was iptables, i tought with service iptables stop will do it |
| 15:47.01 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 15:53.23 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 15:54.06 | alami | [TK]D-Fender: thansk a lot now works, do i have to iptables --flush each time the server restart? |
| 15:54.17 | *** join/#asterisk Ibrahim22 (57d5e366@gateway/web/cgi-irc/kiwiirc.com/ip.87.213.227.102) |
| 15:54.50 | Ibrahim22 | I have a question regarding ARI via websockets. Is it possible to connect to multiple applications with one websocket? |
| 15:56.19 | [TK]D-Fender | alami: Chances are it loads that from somewhere... reconsider how you're running your firewall and learn what you have. |
| 15:58.52 | file | Ibrahim22, yes - separate the names with "," when connecting |
| 15:59.26 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 15:59.42 | Ibrahim22 | Ah, okay, thank you! |
| 16:07.45 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 16:08.27 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 16:14.32 | *** join/#asterisk stefan27 (~stefan27@212.247.4.149) |
| 16:15.05 | stefan27 | there's no chan_local.so anymore? |
| 16:15.28 | file | in 12+ it's part of the core |
| 16:15.59 | stefan27 | queues.conf.sample in ast 13 says if you use Local channels as queue members, you must also preload pbx_config.so and chan_local.so |
| 16:16.07 | stefan27 | What does this mean for me using asterisk 13? |
| 16:16.25 | [TK]D-Fender | It means the sample config needs to be updated |
| 16:16.25 | file | you don't need to preload chan_local |
| 16:16.36 | file | yeah |
| 16:17.18 | stefan27 | Can preloading pbx_config.so ever be bad? |
| 16:20.29 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-kwqatavtlqbhorvb) |
| 16:20.30 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 16:24.18 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 16:26.47 | ChannelZ | Sure, when you've jacked up your dialplan.. |
| 16:27.29 | ChannelZ | oh heh I read that as REloading not PRE |
| 16:34.48 | alami | on ubuntu after service asterisk start i can asterisk -r, on centos i get the Unable to connect to remote asterisk.. |
| 16:35.26 | [TK]D-Fender | have you proven that * is running? |
| 16:36.24 | alami | <PROTECTED> |
| 16:36.46 | [TK]D-Fender | I mean REAL proof |
| 16:37.14 | alami | lsof -i ? |
| 16:37.17 | [TK]D-Fender | ps -A |
| 16:37.25 | [TK]D-Fender | your asterisk.conf also needs to be pointing to the right place for the PID file <---- |
| 16:37.28 | [TK]D-Fender | Go verify this |
| 16:40.28 | alami | [TK]D-Fender: ps -A | grep asterisk return 18581 ? 00:00:00 safe_asterisk |
| 16:40.51 | *** join/#asterisk dwayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 16:43.08 | alami | wich line in asterisk.conf do you mean? |
| 16:43.42 | WIMPy | First you need to get Asterisk started. |
| 16:43.56 | alami | WIMPy: asterisk is started now |
| 16:44.01 | alami | 12226 ? 00:00:01 asterisk |
| 16:45.32 | WIMPy | Check permissions |
| 16:46.47 | alami | WIMPy: can you specify |
| 16:47.00 | alami | asterisk user have permissions |
| 16:47.58 | WIMPy | The permissions of the socket and your user. |
| 16:48.30 | alami | WIMPy: not help thanks |
| 16:48.32 | alami | have a nice day |
| 16:49.34 | dwayne | Is their a relationship between Asterisk's qualifyfreq and a phone's register interval or does the register interval just need to fall between the minexpiry and maxexpiry? |
| 16:49.56 | WIMPy | dwayne: The later. |
| 16:50.22 | dwayne | WIMPy, cool, thanks |
| 16:52.11 | dwayne | I am using 11.13.0 for SIP TLS/SRTP with Blink phones and I noticed that every other Blink registration attempt fails when the Blink registrar interval was equal to or less than the qualifyfreq |
| 16:52.43 | dwayne | s/registrar interval/register interval |
| 16:53.06 | WIMPy | If you use tcp, there shouldn't be a need for re-registering, but I have no idea, what Asterisk thinks of that. |
| 16:55.36 | dwayne | OK. One more question, if you don't mind. I'm using the default minexpiry and maxexpiry and caching friends but I notice that a peer will sometimes drop out of the 'sip show peers' output for a few seconds, then re-appear; but calling the peer during the time that it drops out will still succeed. Any idea what might cause that? |
| 16:56.10 | dwayne | (using real-time) |
| 16:56.54 | WIMPy | No idea. I've got enough issues to battle without using realtime. |
| 16:57.02 | dwayne | I was thinking that maybe it was not in the 'sip show peers' output maybe during the time between my qualifyfreq (120) and register interval (180) |
| 16:57.35 | WIMPy | Qualify is independant of anything. |
| 16:57.37 | Qwell | woah, it's a dwayne |
| 16:57.45 | dwayne | Qwell, hello sir :-) |
| 16:57.55 | Qwell | dwayne: we should lunch some time |
| 16:57.58 | *** join/#asterisk jiuweigui (~jiuweigui@unaffiliated/jiuweigui) |
| 16:58.09 | dwayne | Yeah we should it would be cool to see you again |
| 16:58.13 | WIMPy | You can even go and qualify some random IP you don't have an account with. |
| 16:59.12 | dwayne | from my observations, having the peer not displaying in the 'sip show peers' output is far different than having it show up as UNREACHABLE |
| 16:59.21 | dwayne | but that may be an obvious statement :-/ |
| 17:00.00 | dwayne | Qwell, maybe towards the end of December when things quiet down a bit? |
| 17:00.25 | Qwell | dwayne: sure |
| 17:00.28 | file | mmm lunch |
| 17:00.45 | dwayne | offers file some beans |
| 17:00.55 | file | dwayne, I have pinto bean chips |
| 17:01.33 | Qwell | file: chips with pinto bean flavoring? |
| 17:01.39 | file | Qwell, no - they are made out of pinto beans |
| 17:01.45 | Qwell | Canada, you're weird. |
| 17:01.48 | file | http://beanitos.com/#productPintoBean |
| 17:02.12 | file | "Product of the USA" |
| 17:02.26 | dwayne | maybe we only sell them to Canadians |
| 17:02.30 | coppice | pinto beans? do they turn into a fireball when you hit them? |
| 17:02.30 | *** join/#asterisk SuperBawlz (~SuperBawl@66.87.148.238) |
| 17:03.03 | WIMPy | coppice: No, after you're done digesting them :-) |
| 17:03.16 | file | I want to try the Nacho Cheese White Bean ones but I don't know if they are carried here... will have to see |
| 17:03.28 | SuperBawlz | Quick question. Can I compile a module without recompiling all of asterisk? |
| 17:03.54 | SuperBawlz | My cdr/mysql module didnt compile right. |
| 17:03.56 | Penguin | yes |
| 17:03.56 | WIMPy | SuperBawlz: If you have the source of YOUR Asterisk, yes. |
| 17:04.12 | Penguin | Did you compile asterisk yourself? Do you still have the build tree? |
| 17:04.23 | SuperBawlz | I started with a clean os install and did it manually. No distro. |
| 17:04.30 | SuperBawlz | Yes |
| 17:04.32 | WIMPy | But recompiling it all would probably have taken less time than asking :-) |
| 17:04.37 | Penguin | If you still have the build tree, you can. |
| 17:05.00 | Penguin | That's a valid point, wimpy. |
| 17:05.42 | Penguin | Don't clean, just make again. |
| 17:06.03 | Penguin | If you clean, you'll recompile all of it. |
| 17:07.04 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:da4:8219:34ff:fecf:17f0) |
| 17:07.49 | WIMPy | *Create ram:NotePad*SetType ram:NotePad Text*Build ram:NotePad |
| 17:07.54 | WIMPy | *Create ram:NotePad*SetType ram:NotePad Text*Build ram:NotePad |
| 17:07.59 | Penguin | NOOOOOO!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 17:08.01 | WIMPy | *Create ram:NotePad*SetType ram:NotePad Text*Build ram:NotePad |
| 17:08.03 | WIMPy | *Create ram:NotePad*SetType ram:NotePad Text*Build ram:NotePad |
| 17:08.05 | alami | WIMPy: stop spam here |
| 17:08.06 | WIMPy | *Create ram:NotePad*SetType ram:NotePad Text*Build ram:NotePad |
| 17:08.11 | WIMPy | *Create ram:NotePad*SetType ram:NotePad Text*Build ram:NotePad |
| 17:08.11 | Penguin | That's not spam. |
| 17:08.13 | WIMPy | *Create ram:NotePad*SetType ram:NotePad Text*Build ram:NotePad |
| 17:08.22 | Penguin | Spam would be if he sent this to you in an email. |
| 17:08.23 | WIMPy | *Create ram:NotePad*SetType ram:NotePad Text*Build ram:NotePad |
| 17:08.35 | WIMPy | *Create ram:NotePad*SetType ram:NotePad Text*Build ram:NotePad |
| 17:08.37 | WIMPy | *Create ram:NotePad*SetType ram:NotePad Text*Build ram:NotePad |
| 17:08.44 | WIMPy | *Create ram:NotePad*SetType ram:NotePad Text*Build ram:NotePad |
| 17:08.54 | WIMPy | *Create ram:NotePad*SetType ram:NotePad Text*Build ram:NotePad |
| 17:08.56 | WIMPy | *Create ram:NotePad*SetType ram:NotePad Text*Build ram:NotePad |
| 17:08.58 | Penguin | file, qwell, mjordan, malcolmd |
| 17:08.59 | WIMPy | *Create ram:NotePad*SetType ram:NotePad Text*Build ram:NotePad |
| 17:09.01 | WIMPy | *Create ram:NotePad*SetType ram:NotePad Text*Build ram:NotePad |
| 17:09.03 | *** kick/#asterisk [WIMPy!north@asterisk/developer/Qwell] by Qwell (WIMPy) |
| 17:09.18 | alami | Penguin: what was that then? |
| 17:09.23 | Penguin | a flood |
| 17:09.31 | Qwell | a misclick >.> |
| 17:09.47 | Penguin | mis- something |
| 17:09.53 | Penguin | I've never seen that before. |
| 17:09.54 | alami | lol .. ahh good so he will be back :-) |
| 17:10.05 | stefan27 | if Set(CHANNEL(language)=foo) was called earlier in the dialplan, how would i query the value of the item CHANNEL(language) later in the dialplan? |
| 17:10.29 | Penguin | ${CHANNEL(language)} I'd imagine. |
| 17:11.22 | stefan27 | ah that worked |
| 17:11.36 | stefan27 | thanks |
| 17:11.54 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 17:12.37 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 17:12.42 | WIMPy | Sorry but that. |
| 17:12.54 | WIMPy | Did I mention that I need more space? |
| 17:12.55 | SuperBawlz | The rest of the system is working fine. Plus i was just curious. |
| 17:13.01 | Penguin | What happened? |
| 17:13.18 | SuperBawlz | And whats with the channel spamming dawg? |
| 17:13.21 | WIMPy | Just some stuff ending up on the keyboard once again. |
| 17:13.33 | Penguin | As we already covered, that wasn't spam. |
| 17:13.40 | Penguin | Spam would have been if he sent that to you in an email. |
| 17:13.46 | WIMPy | But I ffel old now. That keybinding surely hasn't made sense dor much more than a decade. |
| 17:13.52 | SuperBawlz | Floodinh |
| 17:14.00 | SuperBawlz | Flooding. |
| 17:14.18 | Qwell | It's done. Get over it. |
| 17:14.29 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-53-253.dynamic.qsc.de) |
| 17:14.35 | stefan27 | when i set a normal variable on a asterisk-chan i can use underscores as _happyVariable=boo to make it survive a dial, but how can i do that to make CHANNEL(language) survive a dial |
| 17:14.38 | WIMPy | We should talk about spit here, not spam :-) |
| 17:15.50 | stefan27 | so that if SIP\A executes Set(CHANNEL(language)=foo) then Dial(LocalX... LocalX would still have ${CHANNEL(language)} returning true |
| 17:15.59 | stefan27 | i meant boo |
| 17:16.02 | stefan27 | not true |
| 17:16.03 | Penguin | stefan27: Survive a dial? What does that mean? Preceeding variables with (an) underscore(s) is for variable inheritance to child channels. |
| 17:16.31 | stefan27 | yeah that's what i meant -- but CHANNEL(item) is a function not a variable? |
| 17:16.36 | *** part/#asterisk alami (~ialami@unaffiliated/alami) |
| 17:16.57 | WIMPy | stefan27: Use pre-dial handlers? |
| 17:17.07 | Penguin | If you set the value to a variable, then you can look it up as a variable. Channel values often change per channel. |
| 17:17.26 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 17:18.38 | Penguin | If you need a newly created channel to have the CHANNEL(language) value set, set it in the target extension. |
| 17:19.29 | Penguin | If it's for a SIP device, set it on the peer definition. |
| 17:21.51 | *** join/#asterisk jhlavacek (~jirka@84.19.95.180) |
| 17:22.48 | file | https://wiki.asterisk.org/wiki/display/AST/Pre-dial+handlers are also useful for outgoing |
| 17:23.47 | Penguin | I've got to remember to implement some of the new handlers that weren't available in previous branches. |
| 17:28.53 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 17:50.47 | rrittgarn | Caller A calls in and is blind transferred to a specific parking spot by setting PARKINGEXTEN=71 (for example), this lights up the BLF on the phones and all is happy until caller B calls in, and less qualified receptionist blind transfers Caller B to 71 as well... Now Caller A and Caller B are talking without an easy way to retrieve them... |
| 17:50.59 | rrittgarn | Is there any way to prevent this (aside from more qualified end users) while maintaining the pick your parking spot functionality? |
| 17:51.09 | rrittgarn | PB of the parking code: http://pastebin.com/P4Tp87j3 |
| 17:54.11 | Penguin | The concept would be to return the call back to the lqr who sent it there. |
| 17:54.30 | Penguin | I'm not sure how to write that. |
| 17:55.00 | [TK]D-Fender | rrittgarn: You already have code there to act differently if its inuse.... |
| 17:58.20 | rrittgarn | yes except that if you blind transfer to it, the transfer still completes |
| 17:58.23 | rrittgarn | and joins the calls |
| 17:58.57 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 17:59.24 | [TK]D-Fender | It will always complete |
| 17:59.29 | [TK]D-Fender | You can't stop that |
| 17:59.37 | [TK]D-Fender | Dialplan has to decide what to do. |
| 18:00.22 | [TK]D-Fender | There is no "lets do some stuff before actually accepting this blind transfer" |
| 18:00.33 | Penguin | I have an idea. |
| 18:00.56 | [TK]D-Fender | Actually, so do I... |
| 18:01.04 | [TK]D-Fender | (with caveats) |
| 18:01.09 | Penguin | Set some arbitrary variable on the initial inbound call. Check for it before allowing the Dial(). |
| 18:01.19 | [TK]D-Fender | Check if it IS a blind transfer and not a new call for pickup purposes there. |
| 18:01.31 | Penguin | If the variable exists AND the parkl channel is in use, reject it. |
| 18:01.40 | [TK]D-Fender | Penguin: Shouldn't need an arbitrary one.. there already is one for blind transfers |
| 18:01.44 | Penguin | Oh? |
| 18:01.49 | Penguin | That's even easier. |
| 18:01.51 | [TK]D-Fender | Penguin: Yup... |
| 18:02.09 | Penguin | But wait... |
| 18:02.21 | [TK]D-Fender | Now either way the transfer will still be complete, but at least you'll know WHY it was called and act accordingly. |
| 18:02.24 | Penguin | All of the transfers into that park will be blind. |
| 18:02.34 | Penguin | So that's not going to be a good check. |
| 18:02.36 | [TK]D-Fender | pehe also uses that exten for pickup |
| 18:02.39 | [TK]D-Fender | Penguin: ^ |
| 18:02.44 | [TK]D-Fender | Wchi would be a new call |
| 18:03.01 | [TK]D-Fender | he just wants to save his ass on a blind to an occupied spot |
| 18:03.16 | [TK]D-Fender | So it'll still have to do some logic to call him back, etc... but hey, that's on him |
| 18:03.23 | Penguin | My idea will have a value set on any call that comes from outside. If that value exists on the channel that would allow it to be parked when the park is not in use. |
| 18:03.50 | Penguin | If the park is in use AND if the value exists, reject it. |
| 18:04.21 | Penguin | If the park in in use and the value does not exist, retrieve the parked call. |
| 18:04.48 | Penguin | If the value exists but the park is not in use, allow parking of the call. |
| 18:04.54 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 18:04.58 | rrittgarn | i think i understand what both of you lovely people are describing, and it sounds almost like the same thing, except fender is relying on built in ${BLINDXFER} variable? |
| 18:05.09 | [TK]D-Fender | yup |
| 18:05.23 | Penguin | But I'm saying every attempt at parking will have that set on it. |
| 18:05.25 | Penguin | So that's out. |
| 18:05.38 | rrittgarn | thing is if all calls being transfered to 71 (example) are blind, and I'm already checking if its in use and its failing, what could i check for beyond that |
| 18:05.44 | rrittgarn | pretty much what penguin said |
| 18:05.55 | rrittgarn | only difference would be if i were having issues picking up (which i'm not unless they get joined) |
| 18:06.29 | Penguin | If every transfer to park is blind, every transfer will have that value. |
| 18:07.05 | Penguin | Wait, I'm thinking of it from the wrong angle. |
| 18:07.29 | [TK]D-Fender | PICKUP usage won't be a transfer |
| 18:07.30 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 18:07.36 | Penguin | ...right |
| 18:07.42 | Penguin | That's what I was about to modify. |
| 18:07.48 | [TK]D-Fender | it's only double-blind we need to watch for |
| 18:07.54 | Penguin | Check for the absence of the blind transfer value. |
| 18:08.04 | [TK]D-Fender | or presence thereof |
| 18:08.10 | Penguin | If the blind transfer value does not exist, it must be a pickup. |
| 18:08.26 | [TK]D-Fender | that's the theory |
| 18:09.02 | Penguin | That should be easy to add to what is already in place. |
| 18:09.29 | [TK]D-Fender | yup |
| 18:09.34 | rrittgarn | i guess i'm unclear on how knowing its a blind helps me or not. If i see its blind and in use, bounce back to the original caller via either another dial or a redirect? |
| 18:09.47 | [TK]D-Fender | With consideration to what he actually intends to do with that information. |
| 18:09.53 | rrittgarn | just making sure i have my head around your discussion (and extreme helpfulness) |
| 18:09.55 | [TK]D-Fender | rrittgarn: yes |
| 18:09.58 | Penguin | A call comes in, gets transferred to park71. |
| 18:10.33 | Penguin | A second call comes in, gets transferred to park71, but this time it is inuse so you jump to a different part of dialplan. |
| 18:10.56 | Penguin | Check for the presence and/or absence of that blind transfer value. |
| 18:11.23 | Penguin | If inuse=1 AND blindxfer=1, reject the park. |
| 18:11.36 | Penguin | If inuse=1 AND blindxfer=0, pickup the park. |
| 18:12.04 | rrittgarn | k pretty sure that makes sense. I'll test and report back. Thanks for the help |
| 18:12.35 | Penguin | You'll still have to figure out what you're going to do with the call that is being rejected. |
| 18:12.55 | Penguin | I would want it to return to the goofball who sent it there when the park was already in use. |
| 18:13.11 | rrittgarn | yeah as i said... end users... bane of my existence... |
| 18:16.53 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 18:18.47 | *** join/#asterisk mbowie (~mbowie@162.212.36.9) |
| 18:21.13 | Penguin | Maybe a hangup handler in conjunction with those tests would be able to return the call to the person who sent it. |
| 18:21.22 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 18:24.06 | *** join/#asterisk o_be_one (~o_be_one@92.39.243.154) |
| 18:24.14 | o_be_one | hi community :) |
| 18:24.15 | [TK]D-Fender | nope. |
| 18:24.28 | rrittgarn | thing is there isn't a hangup, the calls get bridged by the parking application |
| 18:24.34 | [TK]D-Fender | the channel is actually gone at that point |
| 18:24.37 | rrittgarn | then people just get confused |
| 18:24.47 | o_be_one | i'm looking for an awesome voice recognition wich is free, i've already heard something about sphinx but someone knows that and can suggest something ? |
| 18:24.51 | Penguin | When I make a call which goes out over my google voice account, my phone shows a call to +1NXXNXXXXXX@voice.google.com instead of just NXXNXXXXXX. Is that an operation of the channel driver for the phone? |
| 18:25.16 | Qwell | o_be_one: even the non-free ones aren't something I would call "awesome" |
| 18:25.50 | [TK]D-Fender | awesome voice recognition <- contradiction in terms |
| 18:25.52 | Penguin | The transferrer hangs up. That doesn't trigger a hangup handler in any way? |
| 18:26.13 | [TK]D-Fender | Penguin: There is nothing to save there. |
| 18:26.32 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 18:27.00 | o_be_one | Qwell, yeah sure, i just need to recognize some things like "yes" "no" "one" "two" ... |
| 18:28.02 | [TK]D-Fender | o_be_one: sphinx can do that. Also there is direct Google ASR support out there. |
| 18:28.14 | [TK]D-Fender | o_be_one: Which you can google up a guide for |
| 18:28.40 | o_be_one | Google offer only 50 use by day, i need many more :( |
| 18:34.41 | o_be_one | [TK]D-Fender, you talked about that right ? |
| 18:35.35 | [TK]D-Fender | yes |
| 18:37.34 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:39.26 | Penguin | I added the I option to the Dial() and now it doesn't do that. |
| 18:40.40 | zaf | having difficulties getting grandstream device to autoconfigure with an xml file. Anyone have a working example? |
| 18:42.53 | o_be_one | Qwell, and nuisance (dragon translate) or humanvox are awesome no ? but not free :( |
| 18:43.49 | Penguin | Something called a nuisance doesn't sound very favorable. |
| 18:43.54 | Qwell | Lumenvox, you mean? And Dragon takes a *HUGE* amount of time to train, onto a specific voice. |
| 18:49.34 | o_be_one | Lumenvox yes |
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| 18:50.01 | o_be_one | Penguin, lol sure about nuisance x) |
| 18:50.05 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
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| 18:51.22 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:01.47 | mjordan | with speech recognition, you get what you pay for - both in terms of accuracy and in terms of ease of use. Plus, with the silly number of patents surrounding the technology, the free solutions are somewhat encumbered. |
| 19:04.06 | o_be_one | i think that the free solution could help for small recognitions like "yes" "no" ... |
| 19:08.03 | [TK]D-Fender | Go for it |
| 19:08.21 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 19:17.56 | *** join/#asterisk EC_Joe (~EC_Joe@204.11.61.56) |
| 19:19.05 | EC_Joe | hey guys we're running a copy of freepbx, and i'm trying to use Yii2 with my asterisk database. Came across the devices table and noticed it has no primary key, is this something that would be considered being changed? |
| 19:21.10 | Qwell | ~freepbx |
| 19:21.10 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:21.54 | EC_Joe | Qwell: is the "devices" table not an asterisk table? |
| 19:22.07 | EC_Joe | (just asking because they told me it wasn't theirs) |
| 19:22.29 | Qwell | I don't know how you got that from what you were told. |
| 19:23.11 | Chainsaw | "I don't know what to do" "Say it wasn't you" *guitar riff starts* |
| 19:23.54 | [TK]D-Fender | [14:22]EC_Joe(just asking because they told me it wasn't theirs) <- not true |
| 19:24.33 | Chainsaw | gives [TK]D-Fender a large, quiz show style "wrong" buzzer |
| 19:24.55 | EC_Joe | I don't have the exact message but thinking it was something along the lines of only "freepbx" tables are freepbx |
| 19:25.04 | [TK]D-Fender | Chainsaw: I want a giant gong, and vaudeville-style cane.... |
| 19:25.14 | Penguin | (1238.01) <[TK]D-Fender> all of the tables from FreePBX ... are FreePBX |
| 19:25.24 | [TK]D-Fender | eccALL of the tabels maintained by FreePBX |
| 19:25.42 | [TK]D-Fender | EC_Joe: ALL of those tables are maintained by FreePBX |
| 19:25.50 | [TK]D-Fender | can't type at all today |
| 19:25.53 | *** join/#asterisk SuperBawlz (~IceChat@mail.gocomputech.com) |
| 19:25.53 | rrittgarn | asterisk realtime database structures != FreePBX |
| 19:26.00 | SuperBawlz | [2014-12-03 14:23:59] ERROR[4934] cdr_mysql.c: Failed to connect to mysql database asteriskcdrdb on localhost. |
| 19:26.04 | [TK]D-Fender | rrittgarn: He's using FreePBX |
| 19:26.13 | EC_Joe | then I shall return to the freepbx channel to continue the conversation |
| 19:26.13 | SuperBawlz | Is there any way to get more information from the system on that error? |
| 19:26.23 | Penguin | If FreePBX created them, they are part of FreePBX. |
| 19:26.25 | rrittgarn | Fender: yes, i was agreeing with you |
| 19:26.27 | SuperBawlz | I can access the database just fine. |
| 19:26.28 | [TK]D-Fender | SuperBawlz: have you proved that it does exist? |
| 19:26.37 | SuperBawlz | Does what exist? |
| 19:26.43 | [TK]D-Fender | Because next would be that the user has no rights to it |
| 19:26.49 | SuperBawlz | Hmmmm |
| 19:26.51 | [TK]D-Fender | that database & table |
| 19:26.56 | SuperBawlz | Hang on, let me test something |
| 19:27.04 | SuperBawlz | the database and table does |
| 19:27.08 | SuperBawlz | That much I know. |
| 19:27.17 | SuperBawlz | Let me double check user rights. |
| 19:27.18 | [TK]D-Fender | time to quintuple-check your configs too... |
| 19:27.26 | Chainsaw | EC_Joe: You should! You received incorrect information. |
| 19:27.41 | EC_Joe | Chansaw I think I misinterpreted information given to me is all |
| 19:29.10 | Chainsaw | EC_Joe: It appears #freepbx is mounting a "It wasn't me" defence. You should dig deeper. |
| 19:29.55 | Chainsaw | EC_Joe: Particularly as Fender's politeness is a shared resource. It is finite and you are reaching the end. |
| 19:31.11 | SuperBawlz | Ok, I used the console and tested the user and pass and rights. No problem logging in and now problem inserting a record and displaying records. |
| 19:31.30 | SuperBawlz | I'm going to triple check the configs. |
| 19:32.01 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 19:32.05 | [TK]D-Fender | [14:29]ChainsawEC_Joe: Particularly as Fender's politeness is a shared resource. It is finite and you are reaching the end. <- not yet |
| 19:32.35 | Chainsaw | [TK]D-Fender: But the end is abrupt. EC_Joe needs to be prepared. |
| 19:32.49 | *** join/#asterisk jhlavacek (~jirka@84.19.95.180) |
| 19:33.32 | [TK]D-Fender | Chainsaw: Don't go playing psychic and getting people worried for what may come at some undetermined point. |
| 19:34.29 | Chainsaw | [TK]D-Fender: Do I have a no capslock guarantee? |
| 19:35.22 | [TK]D-Fender | Chainsaw: Nope. And it's not just used for yelling you know. |
| 19:35.52 | Chainsaw | [TK]D-Fender: That is what I thought. It is fair warning. |
| 19:41.12 | Chainsaw | I have been asked to leave. |
| 19:44.06 | [TK]D-Fender | ? |
| 19:49.12 | hardwire | are we talking about relationships to software? |
| 19:49.14 | hardwire | is intrigued. |
| 19:51.01 | [TK]D-Fender | hardwire: My software fucks me all the time ... but it's more like "rape" than "relationship". |
| 19:51.25 | [TK]D-Fender | dies a little inside |
| 19:51.56 | hardwire | so you're just overly crass then. |
| 19:52.03 | *** join/#asterisk antiochIst (~taylorhaw@168.244.49.230) |
| 19:52.44 | antiochIst | anyone else noticed setting CDR variable does not work in h extension on 12+? |
| 19:53.19 | [TK]D-Fender | Last I heard that was read-only once you reach that point regardless of version |
| 19:53.31 | antiochIst | really? |
| 19:54.29 | antiochIst | https://issues.asterisk.org/jira/browse/ASTERISK-16874 |
| 19:55.50 | antiochIst | also why would âendbeforehextenâ be an option in cdr.conf? |
| 19:55.53 | WIMPy | It depends on your cdr configuration. |
| 19:56.04 | mjordan | CDRs in 12 != CDRs in prior versions. |
| 19:56.09 | mjordan | or rather, 12+. |
| 19:56.13 | WIMPy | Either it's read-only or the callstats are available. |
| 19:56.20 | WIMPy | AFAIR that is. |
| 19:57.01 | mjordan | endbeforehexten does still do stuff. You can modify a CDR in the 'h' extension. However, the CDR that represented the path of communication between party A and party B is already done: the parties hung up. As such, you can't modify it. Modifying things in the 'h' extension modifies a new CDR for party A. |
| 19:57.21 | mjordan | Once a path of communication between two parties is closed, that CDR is done. We don't go around mucking with it still. |
| 19:58.51 | hardwire | I used a 'final' variable for CDR modification in h |
| 19:59.45 | mjordan | Generally, you'll get two CDRs for a channel if you want to modify things in the 'h' extension. If you don't want that, then you should use one of the Dial interception routines (pre-dial, pre-bridge) to modify the party A's records. |
| 19:59.57 | mjordan | You can certainly look at the last CDR for a channel in the 'h' extension |
| 20:00.35 | antiochIst | ok, ill look into it more |
| 20:00.45 | antiochIst | Im also using cdr_mysql |
| 20:01.06 | mjordan | which is deprecated. |
| 20:01.11 | mjordan | so your mileage may vary. |
| 20:01.16 | antiochIst | yea |
| 20:01.32 | mjordan | Granted, the backends weren't touched in 12+, but using anything that is deprecated is generally a "good luck" |
| 20:01.41 | antiochIst | yea i feel you |
| 20:12.33 | *** part/#asterisk EC_Joe (~EC_Joe@204.11.61.56) |
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| 20:44.58 | mdhas | Does anyone know why a registration for all my sip devices timeout since I upgraded my modules? |
| 20:47.30 | [TK]D-Fender | What modules? |
| 20:48.20 | mdhas | Core and a few others...I don't remember off hand but can look it up |
| 20:48.59 | [TK]D-Fender | Not sure what you're referring to so please look it up and be more elaborate |
| 21:13.54 | *** join/#asterisk saint_ (~saint@66.85.174.98) |
| 21:14.48 | saint_ | hi all - If I have a digium phone D70 , how can I program it so depending on which LINE button is pressed on the main screen, the outgoing call will use X or Y did that I own ? |
| 21:15.40 | [TK]D-Fender | saint_: have them as separate reg's |
| 21:17.50 | *** join/#asterisk dxd828 (~dxd828@195.191.107.205) |
| 21:24.47 | saint_ | [TK]D-Fender so i just looked into it, and i am not sure to follow what you are suggesting. |
| 21:25.13 | [TK]D-Fender | Make each it's own SIP account |
| 21:25.13 | Penguin | [TK]D-Fender: its |
| 21:25.24 | saint_ | i have a registration (?) within the sip.conf , but everything else is in res_digium_phone.conf |
| 21:26.06 | saint_ | [TK]D-Fender so make 2 registrations in the sip.conf ? |
| 21:26.54 | [TK]D-Fender | Clearly |
| 21:28.12 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 21:36.56 | saint_ | [TK]D-Fender so i was able to add another line to the phone, with the new registration. now.. in the dialplan, how do i figure out which button line is pressed so I can pick what DID to send out ? |
| 21:37.24 | [TK]D-Fender | because you'll set each peer to its own context |
| 21:38.30 | [TK]D-Fender | ok, checkout time, heading home |
| 21:41.39 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 21:46.27 | mdhas | Interesting turns out it was my sip provider completely co-incidental regarding the fact that I upgraded |
| 21:46.55 | *** join/#asterisk tzica (~newirc@unaffiliated/tzica) |
| 21:47.17 | tzica | who can advise on the following setup: |
| 21:47.28 | saint_ | if I have a context=xxx in a template , in sip.conf , and i create a new reg from this template but I add context=yyy in it, will this one (yyy) overwrite the context from the template ? |
| 21:48.33 | tzica | [pbx 1 + phones] - trunk - [asterisk] - trunk - [SIP provider] |
| 21:49.02 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 21:49.16 | tzica | what to do to make outbound calls through SIP provider from phones connected to PBX 1 ? |
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| 22:07.09 | *** join/#asterisk saint_ (~saint@66.85.174.98) |
| 22:07.26 | saint_ | Is the digit_map taken in account with Digium phones, in res_digium_phones.conf ? |
| 22:07.52 | saint_ | Because I have a very simple one: [2-9]xxxxxxxxx , and it will let me dial numbers starting with 1 |
| 22:12.39 | *** join/#asterisk tzica (~newirc@unaffiliated/tzica) |
| 22:15.27 | malcolmd | digit_map is for off-hook dialing. for on-hook dialing, you dial whatever you like. if you'd like it to complete the dial at any point during your on-hook dialing, take the phone off-hook. |
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| 22:28.48 | saint_ | malcolmd hhhaaaa, okay, thanks. |
| 22:54.42 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:59.48 | saint_ | So in the PrivacyManager . once she says press 1 for xx , press 2 for xx , press 3 for xx , press 4 for xx , right after she is done she says "I am sorry I did not understand your response" , and go again. Is this a bug ? or miss configuation ? |
| 23:10.17 | [TK]D-Fender | saint_, Show us.... |
| 23:10.35 | saint_ | well... you would have to listen to it ... |
| 23:11.04 | saint_ | all I have in my dialplan is same => n(accept),PrivacyManager() |
| 23:11.27 | saint_ | beside the fact that I test ${DIALSTATUS} just after, but that is irrelevant. |
| 23:11.54 | saint_ | When the caller is in the privacymanager, and the privacymanager call the number I have in Dial, that's when it goes wrong |
| 23:12.59 | saint_ | [TK]D-Fender but if you really want to see it, here it is: http://pastebin.com/PwhAHxB6 |
| 23:20.55 | [TK]D-Fender | Show the CALL |
| 23:38.02 | saint_ | [TK]D-Fender debub, verbose, or sip ? |
| 23:38.35 | [TK]D-Fender | debug & verbose |
| 23:41.25 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ndmfxdyixnistgkr) |
| 23:45.52 | saint_ | [TK]D-Fender here you go: http://pastebin.com/eJXDHnnb |
| 23:48.56 | [TK]D-Fender | What are you pressing during that? |
| 23:49.24 | saint_ | nothing |
| 23:49.30 | saint_ | it just goes on a loop |
| 23:49.34 | saint_ | i can press a key anytime |
| 23:49.43 | saint_ | it just bother me that it goes in a loop without waiting for my input |
| 23:49.48 | saint_ | and there is no option for that |
| 23:50.56 | [TK]D-Fender | Well that's just the way it works. If you don't like it then you can write your own confirmation processing |
| 23:57.46 | saint_ | well.. that is bad user experience |