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| 00:32.04 | FuriousGeorge | hey all |
| 00:32.48 | FuriousGeorge | does anyone have any experience with polycom devices and provisioning? i got a soundstation ip 5000 that does not want to play nice. i can't get it to register with asterisk, despite following this guide: |
| 00:32.59 | FuriousGeorge | http://support.polycom.com/global/documents/support/technical/products/voice/Register_Standalone_SoundPointIP_SoundStationIP_QT44011.pdf |
| 00:33.29 | FuriousGeorge | i tried to set up a provisioning server to upgrade the bootROM and firmware, but the device just boots up and does not even try to log in. im kinda at a loss |
| 00:37.50 | snadge | lol |
| 00:38.21 | snadge | i just wrote a guide on how to set up soundstation 501 and 301 (out of support) for a customer recently |
| 00:38.59 | snadge | i used ftp.. went into setup manually, default password 456, set the boot server ip address and username/password manually |
| 00:40.49 | snadge | stick the split files (if your bootrom supports them, it should) into the home dir of the ftp login you just set up.. maybe reset the local settings on the phone first |
| 00:42.12 | snadge | the 301 wont register to asterisk at all.. unless you deploy its settings via the boot server.. there is a bug in older / legacy polycom phones that prevents successful registration when the phone is provisioned via the web interface |
| 00:43.05 | snadge | i cant help but feel that yealink and snom are the way to go these days.. with cisco a distant third ;) |
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| 00:44.00 | FuriousGeorge | i like snom stuff. i did everything you did, but the phone afaict does not even try to log in to the ftp server |
| 00:44.56 | WIMPy | had it fully automatic once. But that no longer works. |
| 00:44.58 | snadge | well.. not that it's relevant, I have used both proftpd and ftpd to do it.. in hindsight, proftpd is the better choice, because you can see login and file transfer activity in /var/log/proftpd/xferlog |
| 00:45.44 | snadge | wheres by default, ftpd only shows logins and not file transfer activity |
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| 03:12.28 | tulga | inbound call not registered in queue_log file. how to solve it? |
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| 03:25.03 | ageis | so I just flashed some Grandstream phones with an updated firmware via TFTP... and now I can't save configuration changes, a lot of fields are blank |
| 03:25.12 | ageis | anyone got ideas |
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| 03:58.43 | linocisco | why asterisk channel has only 176 online users while ubuntu has more than 1000. ? |
| 03:59.35 | linocisco | I would like to see asterisk has more users |
| 04:07.12 | ChannelZ | why |
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| 04:50.38 | linocisco | ChannelZ, the philosophy of asterisk has a lot of best objectives. but found less likes than others. |
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| 07:03.59 | FuriousGeorge | hey all |
| 07:04.07 | FuriousGeorge | what's wrong with this: |
| 07:04.35 | FuriousGeorge | exten => s,n,gotoif($[ ${ARG3} = "null" ]?s,exit) |
| 07:05.27 | FuriousGeorge | <PROTECTED> |
| 07:10.57 | D30 | is there a way to change the name of the "from" from email notification? |
| 07:11.40 | D30 | mine is sending from asterisk@mydomain.com, but i want to change the name when i received it like SIP.PBX but still it come from asterisk@mydomain.com |
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| 07:32.16 | ChannelZ | FuriousGeorge: Probably all the spaces. The expression parser can be fussy. |
| 07:34.09 | ChannelZ | D30: serveremail in voicemail.conf I assume you are talking of |
| 07:34.30 | D30 | hmmn checking on that ChannelZ , thanks.. |
| 07:40.39 | D30 | ChannelZ: ahmm actually that is the right config, emails are coming from asterisk@mydomain.com |
| 07:41.14 | D30 | im actually referring to its name? i dont know if "name" is the right term |
| 07:42.01 | D30 | so emails when send are from asterisk@mydomain.com, but when it arrived it will display as for example "Asterisk PBX" |
| 07:43.17 | FuriousGeorge | ChannelZ: i needed to put quotes around both varas |
| 07:43.20 | FuriousGeorge | vars |
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| 07:56.04 | D30 | like this http://i59.tinypic.com/dcehkk.jpg |
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| 08:23.07 | ChannelZ | D30: set fromstring |
| 08:23.28 | D30 | still in voicemai.conf ?? |
| 08:23.33 | ChannelZ | yes |
| 08:23.42 | D30 | hmmn |
| 08:24.35 | D30 | got it :) i didnt see this a while ago.. maybe because its commented |
| 08:25.13 | ChannelZ | Yeah, built-in default is "Asterisk PBX" |
| 08:25.59 | D30 | ChannelZ: would this take effect also on emails not related to voicemail? |
| 08:26.12 | D30 | like in trunk monitoring notification email |
| 08:26.29 | ChannelZ | I don't know what that is. |
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| 08:27.42 | D30 | i mean i made a custom scripts to monitor trunk activities, and will send mo notifications from time to time, |
| 08:27.47 | ChannelZ | which is to say I don't know of any other part of Asterisk that sends email. |
| 08:27.55 | D30 | is "frmstring" is also applicable to that? |
| 08:28.02 | D30 | hmmnn |
| 08:28.06 | D30 | okay.. |
| 08:28.16 | ChannelZ | Not unless your scripts are stealing random bits of config from voicemail.conf.. |
| 08:29.59 | D30 | ahh okay |
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| 11:55.07 | zapphir | Hi! I have a problem with device_status, cannot get it to work properly, it always states "not_in_use" even when the agent is in a call. I have put relevant parts of my configuration here: http://pastebin.com/f8Sut5bJ |
| 11:55.18 | zapphir | Would appreciate any tips! |
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| 12:08.07 | Tuju | how do i admin login to SPA301? |
| 12:09.14 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:13.32 | Tuju | can't even log in to cisco's web page, how did they make it so complex. |
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| 12:29.31 | zapphir | Hi! I have a problem with device_status, cannot get it to work properly, it always states "not_in_use" even when the agent is in a call. I have put relevant parts of my configuration here: http://pastebin.com/f8Sut5bJ, would appreciate any tips/ideas! |
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| 12:34.23 | [TK]D-Fender | paste is no good. |
| 12:34.27 | [TK]D-Fender | Show the code, and the call |
| 12:36.05 | zapphir | D-fender: which code do you mean? |
| 12:36.43 | [TK]D-Fender | zapphir: Your pastebin is empty |
| 12:37.00 | [TK]D-Fender | Show the actual dialplan and your attempt to use it including a channel dump prior |
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| 13:38.33 | zapphir | D-fender: can't get dumpchan to work, but the pastebin should now show some data at least :) http://pastebin.com/S5VDH3xw |
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| 13:43.06 | [TK]D-Fender | zapphir: we need to see actual calls. |
| 13:44.35 | [TK]D-Fender | we don't know what version your running, no dump of your queue, no queue config itself (FULL), no active channel dump. |
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| 14:05.24 | Entelin | what is the best means by which to use a physical fax machine through an asterisk pbx? |
| 14:05.46 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 14:07.00 | newtonr | Entelin, several people would probably give you a few different answers. Why do you need to run faxes through Asterisk? |
| 14:08.18 | file | fax is evil |
| 14:10.51 | SpeedEvil | Fax is also required by some retarded government or corporate bodies |
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| 14:15.32 | Entelin | because if we didnt it would be an extra 30*4$ /mo for pots phonelines |
| 14:15.40 | [TK]D-Fender | SpeedEvil: How is your PBX reaching the PSTN? |
| 14:15.56 | SpeedEvil | It's not. |
| 14:16.09 | Entelin | sip trunk |
| 14:17.03 | [TK]D-Fender | SpeedEvil: How can you send a fax to the government... without hitting the PSTN? |
| 14:17.23 | [TK]D-Fender | oops, bad aim I think... |
| 14:17.27 | SpeedEvil | yeah |
| 14:17.34 | [TK]D-Fender | inverted just the right way |
| 14:17.44 | [TK]D-Fender | Entelin: How is your PBX reaching the PSTN? |
| 14:17.55 | Entelin | a sip trunk via another provider |
| 14:19.18 | [TK]D-Fender | Entelin: Make very sure your provider supports T.38, then get an ATA that supports it and run your fax off that |
| 14:20.43 | Entelin | i take it things like sip "adapters" would be a bad idea? |
| 14:23.38 | [TK]D-Fender | No, that is exactly what I told you to get. |
| 14:26.16 | *** join/#asterisk disposable (disposable@shell.websupport.sk) |
| 14:26.53 | *** join/#asterisk mjordan (~matt@nat/digium/x-asbujstpqabgtnyw) |
| 14:26.54 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 14:32.04 | Entelin | ok thanks, I'm not entirely familiar with all the terminology yet. I will verify they support that |
| 14:35.47 | Groenleer | anyone some good resource about T.38 to PDF support with Asterisk 1.8 (yes i know it is ancient...) |
| 14:36.23 | [TK]D-Fender | PDF is an external step, nothing * is responsible for |
| 14:36.50 | [TK]D-Fender | And *'s standard application support it via SpanDSP / FFA |
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| 14:40.43 | tparcina | What do I have to take care of when I upgrade kernel on my Asterisk server? |
| 14:40.54 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 14:41.01 | jeffspeff | what OS is *now based on? |
| 14:41.05 | tparcina | Are there any modules that I have to reinstall? |
| 14:41.39 | [TK]D-Fender | jeffspeff: CentOS |
| 14:41.48 | jeffspeff | thanks |
| 14:42.10 | [TK]D-Fender | JeffExcept the truly ancient ones which were.. I forget... some realy 15th-tier distro |
| 14:42.55 | [TK]D-Fender | tparcina: DAHDI would require rebuilding. Generally * itself doesn't touch the kernel much except for that |
| 14:43.05 | jeffspeff | ok. My boss is wanting to me to use *now for a new office we're setting up. he's intimidated by CLI and loves GUI's. |
| 14:43.25 | jeffspeff | I haven't messed with it in a very long time |
| 14:44.00 | jeffspeff | i remember it used to be based on canary or something like that didn't it? |
| 14:44.15 | jeffspeff | i've killed a lot of brain cells since then |
| 14:44.24 | mjordan | jeffspeff: it's based on CentOS and uses FreePBX for its UI |
| 14:46.00 | tparcina | [TK]D-Fender: Thank you. Right now I have DAHDI installed, but I don't have any Digium hardware anymore. |
| 14:46.15 | tparcina | Do I still need DAHDI? |
| 14:46.58 | jeffspeff | tparcina, what version of * are you using? |
| 14:47.14 | [TK]D-Fender | Depends on your version |
| 14:47.18 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 14:47.28 | jeffspeff | if you don't have any * hardware then the only thing (that i'm aware of) that it's still used for is timing in meetme() |
| 14:47.36 | [TK]D-Fender | MeetMe requires it. Page changed in 11 IIRC to support confbridge for mixing |
| 14:47.38 | jeffspeff | but that was phased out in later versions |
| 14:47.57 | tparcina | jeffspeff: Asterisk 1.6.2.16.2 :D |
| 14:48.17 | tparcina | I have received it when I started to work here, and now I have to upgrade it. |
| 14:48.27 | jeffspeff | if you use meetme or page then you'll still need dahdi installed. |
| 14:48.49 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 14:48.51 | jeffspeff | if you upgrade to the latest version of * then you won't need dahdi for those apps anymore |
| 14:48.58 | Katty | OR WILL YOU |
| 14:49.03 | jeffspeff | LOL |
| 14:49.16 | Katty | morning, mister jeffspeff |
| 14:49.28 | jeffspeff | good morning Katty |
| 14:49.32 | Katty | how're you dear? |
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| 14:49.49 | jeffspeff | physically present and accounted for, need more coffee |
| 14:49.53 | jeffspeff | you? |
| 14:49.59 | Katty | same. |
| 14:50.02 | tparcina | jeffspeff: [TK]D-Fender: Thank you guys. |
| 14:50.08 | jeffspeff | np |
| 14:50.15 | Katty | jeffspeff: upgrading some client software, but otherwise...just kind of being a vegetable. |
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| 15:04.06 | *** join/#asterisk NiugeS (~NiugeS@217.20.27.187) |
| 15:04.19 | NiugeS | hi all.. a newb here.. just installed piaf green.. have configured everything but can't get the phones to pick up config files via tftp.. i even disabled firewall.. i've tried to locally tftp get to piaf server but still not getting anywhere.. having googled with no benefit, any suggesiton on where else i can get help? |
| 15:08.08 | newtonr | NiugeS, The PBIAF forums: http://pbxinaflash.com/community/index.php |
| 15:09.26 | newtonr | NiugeS, most in this room are not using a GUI or pre-packaged distros that do a lot of things for you, so their answers may or may not be confusing for you |
| 15:10.21 | NiugeS | okay |
| 15:10.22 | NiugeS | no prob |
| 15:10.41 | NiugeS | you have to start somewhere ;) |
| 15:14.24 | newtonr | NiugeS, oh I wasn't trying to be insulting at all, I hope you didn't take it that way. I use GUIs whenever I can :D |
| 15:14.32 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
| 15:14.44 | newtonr | NiugeS, just setting expectations for this channel for you |
| 15:16.21 | *** join/#asterisk Brixius (~kmurphy@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 15:18.21 | dan_j | Does anyone know of a SIP softphone that integrates with Salesforce? |
| 15:19.44 | *** join/#asterisk theron (~theron@69.63.185.56) |
| 15:19.56 | Brixius | Question on confrence calls in Asterisk: I have a requirement where Party A calls Party B, Party B Confrences/Begins Supervised Transfer to Party C. Parties A & C cannot be allowed to hear each other and Party B needs to hear both while being able to mute/unmute talking to A & C(Like PTT each Party). Is there a way to do this in Asterisk? |
| 15:20.40 | Brixius | This is an example of like a Collect Call or Operator Assisted Call. |
| 15:21.12 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 15:21.14 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
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| 15:26.58 | [TK]D-Fender | Brixius: This is a fair amount of scripting required with ChanSpy + whisper and features.conf to mute/unmute the other channels via func_volume |
| 15:33.07 | *** join/#asterisk upp (~ialami@146.140.16.62) |
| 15:33.11 | upp | does rasterisk cmd the same as asterisk -r? |
| 15:33.39 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 15:39.54 | Qwell | upp: yes |
| 15:40.08 | upp | ok thanks :) |
| 15:40.20 | *** join/#asterisk serafie_ (uid15202@gateway/web/irccloud.com/x-prsebgmzcbknafvo) |
| 15:46.36 | *** join/#asterisk adeeln (~adeeln@2607:fcf8:4120:face:55a3:d00b:9f0b:cc13) |
| 15:47.35 | adeeln | when using the mysql connector in an asterisk dialplan, does it create a new connection for each invocation of the dialplan? |
| 15:52.18 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 15:52.38 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 15:53.58 | upp | any one here use 3cx softphone ? |
| 15:57.43 | upp | i thing the last version work only with 3cx phone system and no more with asterisk |
| 15:58.31 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 16:01.19 | [TK]D-Fender | upp: Based on what? |
| 16:02.29 | upp | [TK]D-Fender: i try to connect it with my local asterisk server |
| 16:02.46 | upp | but i get always a message that there is no 3cx phone system installed |
| 16:03.07 | [TK]D-Fender | upp: What do you see in the actual debug? |
| 16:04.45 | [TK]D-Fender | upp: http://www.3cx.com/voip/voip-phone/ <-- this seems to say there is a version for generic use... |
| 16:04.46 | *** join/#asterisk mjordan (~matt@nat/digium/x-vzrgkpyiwvtigmdt) |
| 16:04.46 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 16:04.46 | upp | [TK]D-Fender: wich debug mode should i past? |
| 16:05.19 | [TK]D-Fender | [12:04]upp[TK]D-Fender: wich debug mode should i past? <- show us en error resulting for communication with Asterisk. |
| 16:08.18 | *** join/#asterisk neonerz (~g_bove@cpe-076-182-115-090.nc.res.rr.com) |
| 16:09.05 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:6935:9f1c:8bc0:c951) |
| 16:09.15 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 16:12.35 | upp | [TK]D-Fender: http://pastebin.com/YeFZZAs8 |
| 16:13.15 | [TK]D-Fender | Ok, well that's the veryEND on a comm with .. something... and it seems to be OK'd |
| 16:13.24 | [TK]D-Fender | Where do we see something fail? |
| 16:13.44 | upp | i sse at the 3CX screen |
| 16:13.48 | [TK]D-Fender | And a complete comm so we see where it's coming from? |
| 16:14.18 | upp | 3cx phone system is unreachable will try to reconnect automatically |
| 16:16.00 | upp | http://pastebin.com/G6224Cgx |
| 16:16.10 | upp | [TK]D-Fender: here it is |
| 16:16.16 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 16:16.34 | [TK]D-Fender | I see X-Lite there... |
| 16:16.38 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 16:16.45 | [TK]D-Fender | I do not see "3cx" anywhere |
| 16:18.08 | upp | i will then Xlite because it work just fein :) |
| 16:18.14 | upp | thanks |
| 16:18.37 | [TK]D-Fender | huh? |
| 16:21.02 | upp | [TK]D-Fender : like i say befor, no log at the asterisk |
| 16:21.03 | upp | side |
| 16:21.22 | upp | the only message is from 3CX no 3cx phone system is installed |
| 16:21.35 | [TK]D-Fender | So what is this debug from? |
| 16:21.47 | upp | 5 |
| 16:21.52 | upp | should i use 9? |
| 16:21.54 | [TK]D-Fender | ... |
| 16:21.58 | [TK]D-Fender | what DEVICE |
| 16:22.10 | upp | wich DEVICE? |
| 16:22.44 | [TK]D-Fender | You just PB'd debug for us. is that the device you installed 3CX on? |
| 16:23.47 | upp | yes |
| 16:24.02 | [TK]D-Fender | And what hardware is that on? |
| 16:25.03 | upp | Sorry [TK]D-Fender: i have to go out now, i'm so sorry i will come on later and go forward with this Problem |
| 16:25.14 | upp | thanks a lot for trying helping me |
| 16:25.22 | *** join/#asterisk navaismo (~navaismo@200-52-45-221.dynamic.axtel.net) |
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| 16:30.08 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-lcwyjqjgjwcnqfum) |
| 16:30.09 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 16:32.01 | *** join/#asterisk SuperBawlz (~IceChat@rrcs-71-43-76-226.se.biz.rr.com) |
| 16:34.43 | *** join/#asterisk imox (~imox@p4FC5C54B.dip0.t-ipconnect.de) |
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| 16:35.42 | SuperBawlz | I got a strange one, when I do a tcpdump there is traffic but when I go to Telephony -> VoIP Calls there is nothing displayed. There is clearing traffic on SIP and RTP and WireShark does recognize it as SIP and RTP but I still have nothing on the VoIP Call screen. |
| 16:37.09 | [TK]D-Fender | Asterisk has no such "VoIP Call screen" |
| 16:37.09 | SuperBawlz | Had this problem on two different computers, tried with a capture from another system and it works fine. |
| 16:37.22 | SuperBawlz | Yeah. I'm referring to WireShark. |
| 16:37.31 | SuperBawlz | I'm troubleshooting an issue with Asterisk |
| 16:37.57 | Greenlight | Did your call start *before* you started your capture, perhaps ? |
| 16:38.08 | [TK]D-Fender | What is the actual issue with Asterisk? |
| 16:38.30 | [TK]D-Fender | Since what you described is trouble with what wireshark is telling you |
| 16:39.46 | SuperBawlz | I will double check on the call thing. There are tons of calls on the system usually so I find it hard to believe that not a single one was started during a 30 minute peroid. I will do another capture and test though. |
| 16:40.09 | [TK]D-Fender | Yes, actual proof of a SIP invite would be good... |
| 16:40.12 | SuperBawlz | Call quality issues that, until 30 minutes ago I was sure was the carrier. |
| 16:41.42 | *** join/#asterisk SuperNull (~NigggaaaP@24-148-101-238.ip.mhcable.com) |
| 16:43.48 | SuperBawlz | OK. I know for a fact that there was a call that time because I placed it to one of their internal techs, had about a 45 second conversation, then hung up, waiting 5 seconds and then stopped the capture. |
| 16:43.58 | SuperBawlz | Still nothing in the VoIP Calls window. |
| 16:44.04 | Greenlight | Show us the SIP INVITE |
| 16:44.04 | SuperBawlz | So wierd. |
| 16:44.12 | SuperBawlz | standby |
| 16:45.13 | SuperBawlz | how do you want it? |
| 16:45.20 | SuperNull | hey guys, im seeing a lot of RTP Resyncs according this this adtran TA900 what causes that? |
| 16:45.23 | SuperBawlz | Or is there a specific line you want? |
| 16:45.40 | Greenlight | Well, if you can see the SIP INVITE then I'd say it's a problem more with WireSharks parsing |
| 16:46.18 | SuperBawlz | I would agree except that doesn't explain why captures from other PBX's are fine and the problem exists on two other known good copies of wireshark |
| 16:47.08 | Greenlight | I'm not sure what wireshark looks for when parsing calls, but suspect there is a subtle difference somewhere between your captures |
| 16:47.32 | [TK]D-Fender | This isn't #wireshark . If you have actual Asterisk debug for an issue you actually see in Asterisk then we can probably help you... |
| 16:47.35 | Greenlight | Wether that difference is related to your quality issue or not, I'm not sure |
| 16:48.25 | Greenlight | http://wiki.wireshark.org/SIP maybe is of use ? |
| 16:50.51 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 16:51.31 | *** join/#asterisk As001 (~uros@82.117.198.142) |
| 16:51.35 | SuperBawlz | I am waiting for a response in that room. |
| 16:53.47 | As001 | Hello did anyone have experience with running Asterisk over openVPN ? (asterisk listen on private ip like 10.8.0.1 and clients connect to it). Does it work good in terms on call quality,latency etc ? |
| 16:54.06 | SuperNull | As001 #cisco guys will tell you VPN + RTP = BAD TIMES |
| 16:54.31 | SuperNull | thats on hardware vpn which should be more ideal. |
| 16:56.27 | As001 | but if server where is asterisk is beast and routers where agents are are dedicated pcs with on example clearos. I would like to avoid expensive hard to configure cisco equipment. |
| 16:59.49 | [TK]D-Fender | When delivery goes bad you'll pay for using TCP. |
| 17:00.17 | [TK]D-Fender | As for latency, no system should get that taxed for it. |
| 17:00.23 | Greenlight | OpenVPN is usually over UDP |
| 17:00.49 | *** part/#asterisk Brixius (~kmurphy@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 17:00.54 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 17:01.15 | [TK]D-Fender | Which would be recommended. |
| 17:01.25 | [TK]D-Fender | as it supports both |
| 17:04.07 | As001 | yes openvpn is over udp i wonder if I put sip.conf to listen on 10.8.0.1 does it mean rtp will go over openvpn udp port 1194 ? |
| 17:08.24 | [TK]D-Fender | It will go through whatever your routing table has it go through. |
| 17:12.19 | As001 | ok I will first experiment with few calls to check call quality. Thanks |
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| 17:15.18 | *** join/#asterisk sumasuma (~kans@c-24-8-51-80.hsd1.co.comcast.net) |
| 17:15.35 | sumasuma | I need help in registering my polycom with asterisk |
| 17:15.49 | sumasuma | i used the template which came with asterisk, not seeing any registration |
| 17:15.57 | sumasuma | may someone please help ? |
| 17:16.50 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 17:17.15 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 17:29.07 | newtonr | sumasuma, pastebin a log https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information make sure the SIP packet trace is showing from "sip set debug on" |
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| 17:47.26 | *** join/#asterisk davlefouAMD (~david@41.225.169.51) |
| 17:47.52 | sumasuma | newtonr: got it fixed with the basic registration, it works fine now. Thank you. Seems I have something wrong in the full config |
| 17:48.18 | newtonr | sumasuma, sounds good |
| 17:48.33 | *** part/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
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| 18:34.03 | djahandarie | Hey folks. I'm approximately trying to 1) dial into a number, 2) send some DTMF tones, 3) interpret the machine-generated response into some text, 4) access that text in a normal programming language |
| 18:34.33 | djahandarie | I think I can do 1 & 2 with a normal call file, 3 seems to require some major pain in the ass with Sphinx, and I have no idea how to do 4. |
| 18:34.53 | djahandarie | Does anyone have any pointers for me? |
| 18:35.29 | djahandarie | ("Don't use Asterisk, use $X" is a totally valid one :p) |
| 18:38.40 | djahandarie | Ah, I see you can write to a file which accomplishes 4. Is there an easy way to do 3? |
| 18:45.51 | rrittgarn | i could see recording the call, but not sure how to effectively have asterisk act as a modem in that case |
| 18:47.02 | rrittgarn | is it all DTMFs coming back from the other side or is it text too? |
| 18:48.38 | djahandarie | It's computer-generated voice coming back from the other side. |
| 18:48.52 | rrittgarn | so something like cepstral then to parse it? |
| 18:51.07 | djahandarie | Not familiar with cepstral. What I'm seeing on Google is that it's text-to-speech, not speech-to-text. |
| 18:52.00 | rrittgarn | ah backwards my bad |
| 18:52.17 | rrittgarn | http://zaf.github.io/asterisk-speech-recog/ - never used it, but i like google's speech engine... |
| 18:53.46 | rrittgarn | i have used googlets (text to speech) for automated generation of voice, which also relies on google's speech engine, so if everybody is the same there it might work well |
| 18:56.58 | justdave | so when you use DUNDi to place an IAX call, it sets up the DUNDi connection on the fly... does it enable trunk mode on that connection when it places the call? Can't find anything in the docs that says so |
| 18:57.08 | justdave | er, sets up the IAX connection on the fly I mean |
| 19:01.58 | *** join/#asterisk bulkorok (~Adium@053d9234.dynamic.tele-ag.de) |
| 19:03.06 | djahandarie | rrittgarn, I actually have some secrecy goals here, so I don't want to fork out the speech recognition work to a 3rd party |
| 19:03.18 | djahandarie | (In fact, I'd just use twilio if I didn't have any secrecy goals) |
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| 19:50.27 | willwh | hi guys, I'm still cranking on trying to get my asterisk deploy working with sipml5 |
| 19:51.20 | willwh | I am running 11.8.1, compiled like so; make --with-crypto --with-ssl with-srtp=/usr/local/lib |
| 19:51.38 | willwh | I ran the install_prereq script before doing so |
| 19:52.10 | willwh | I still don't get ice replies; SetRemoteDescription failed: Called with an SDP without ice-ufrag and ice-pwd. |
| 19:52.40 | willwh | icesupport=true in rtp.conf, and enabled in sip.conf for the sip account |
| 19:52.55 | willwh | does anyone have any pointers or things to check? |
| 19:53.30 | willwh | i.e. my call dials and connects, and then fails with the "called with an SDP without ice-ufrag and ice-pwd" |
| 19:54.07 | willwh | navaismo: you around? :) |
| 19:55.25 | willwh | others seem to be having trouble too: http://forums.digium.com/viewtopic.php?p=198268 |
| 19:55.41 | willwh | I have a log from ./configure and make... I can pastebin if that would be helpful |
| 19:55.54 | newtonr | willwh, try uninstalling pjproject and reinstall following the instructions here https://wiki.asterisk.org/wiki/display/AST/Installing+pjproject |
| 19:56.23 | willwh | newtonr: hmm, it says it's not needed for 11? |
| 19:56.50 | newtonr | willwh, lol glazed over where you said you were using 11 |
| 19:57.03 | willwh | :) |
| 19:57.17 | willwh | yeah it's pretty odd, now I did compile srtp from source (1.4.2) |
| 19:57.31 | willwh | just frustrating the hell out of me, I am so close :P |
| 19:57.32 | newtonr | willwh, could be conflicts with the pjproject that you have installed? idk |
| 19:57.50 | willwh | yeah it could be, is there a way I could check? |
| 19:57.53 | newtonr | willwh, try uninstalling pjproject completely, and then reinstall asterisk 11 |
| 19:58.35 | willwh | so, newtonr I'm not that familiar with asterisk |
| 19:58.49 | willwh | uninstall pjproject? :) |
| 19:59.05 | willwh | is that like recompile and de-select it in make menuselect? |
| 19:59.51 | willwh | sorry the newb question |
| 20:00.37 | willwh | I definitely don't have it installed from elsewhere, it's in <asterisk src>/res/pjproject/third_party |
| 20:01.07 | willwh | newtonr: ^ :) |
| 20:01.51 | newtonr | i think i'm just going to quit helping today. I'm causing all the confusion. I thought up there you had said you installed pjproject :) |
| 20:01.54 | newtonr | I'm just causing problems now! |
| 20:02.57 | newtonr | I think you are probably fine on configuration. Did you already upload your debug and configs to the existing bugs on the tracker? |
| 20:03.02 | newtonr | I haven't had a chance to dig into those yet |
| 20:06.13 | willwh | I haven't yet |
| 20:06.26 | willwh | is there a doc that describes what is needed and how to get it? :) |
| 20:06.41 | willwh | presumably, sip.conf, rtp.conf, http.conf |
| 20:07.07 | willwh | I'll go and hunt down an existing bug to append to, I suppose :) |
| 20:07.31 | newtonr | willwh, https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 |
| 20:07.52 | willwh | ah |
| 20:08.29 | newtonr | willwh, other than that on the wiki, there is a variety of guides if you google. that one is probably the most recent and I wrote it, it worked fine for me at that time, however there has been a few ICE patches that went in since |
| 20:08.44 | newtonr | one of those ICE patches could have introduced an issue |
| 20:09.17 | willwh | I didn't run ./install_prereq install-unpackaged |
| 20:09.21 | willwh | I wonder if I am missing anything |
| 20:09.24 | willwh | make didn't complain |
| 20:09.51 | newtonr | did you get uuid-dev? |
| 20:09.58 | willwh | yessir |
| 20:10.10 | willwh | newtonr: http://forums.digium.com/viewtopic.php?p=198268 |
| 20:10.11 | newtonr | i missed that one when i first set things up |
| 20:10.15 | willwh | my post is the last one |
| 20:10.25 | willwh | i.e. libuuid version etc |
| 20:11.21 | willwh | I'm installing the unpackaged stuff now |
| 20:11.26 | willwh | I'll recompile after and see how I do |
| 20:12.05 | newtonr | when you recompile, try it from a fresh SVN checkout, or a fresh untarball, just to be sure |
| 20:12.24 | willwh | well, it's a prod system I'd like to run stable if I can |
| 20:12.57 | newtonr | yeah just unpackage the tarball again to a fresh directory |
| 20:13.11 | willwh | ah ok |
| 20:13.20 | newtonr | testing new stuff on a production system eh? :) |
| 20:13.30 | willwh | well, kinda ;] |
| 20:13.36 | newtonr | living on the edge! |
| 20:13.40 | willwh | I have a ubuntu deploy where it all works :P |
| 20:13.47 | willwh | but this is the debian box in our office |
| 20:17.14 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 20:17.26 | willwh | newtonr: there isn't some ice respecific thing I should see in resource modules us there? |
| 20:17.32 | willwh | it's just srtp / http_websockets |
| 20:17.36 | willwh | that's needed |
| 20:17.52 | newtonr | i think res_rtp_asterisk builds in the ICE stuff |
| 20:19.31 | newtonr | yeah, from the wiki "res_rtp_asterisk uses the pjproject library for ICE/STUN/TURN support and local address discovery" |
| 20:20.17 | newtonr | as long as you have those dependencies mentioned in the tutorial and asterisk detects them when configuring and building, res_rtp_asterisk and the others you need should build right |
| 20:20.43 | willwh | ok, I hope the fact I was missing a couple unpackaged deps was the problem |
| 20:20.50 | willwh | althgough none of them looked like much to do with ice/rtp |
| 20:21.54 | newtonr | seeing as several others are seeing the same issue suddenly, i'm betting there is a bug |
| 20:24.05 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 20:26.02 | willwh | you mean with 11.8.1? |
| 20:26.19 | willwh | my compile is *almost* done |
| 20:26.25 | willwh | I should be able to let you know in a couple mins |
| 20:26.31 | willwh | p.s. core restart when convenient ftw |
| 20:26.36 | navaismo | is testing now, installing 11.8.1 on fedora 20 |
| 20:26.44 | willwh | navaismo: :) |
| 20:27.14 | newtonr | willwh, yeah the bugs i linked you to yesterday, both of those guys see it in 11.8.1, https://issues.asterisk.org/jira/browse/ASTERISK-23515 |
| 20:28.25 | newtonr | willwh, one guy reported it happening as early as 11.7.0, i tested it there and could not reproduce |
| 20:28.30 | *** join/#asterisk pr0xie (~pr0xie@unaffiliated/pr0xie) |
| 20:29.37 | *** join/#asterisk JustinAiken (~Adium@74.120.39.46) |
| 20:30.11 | newtonr | willwh, one says that he could only reproduce it in a NAT scenario. That is where you uploading your configs will be helpful so that we compare to the other guys and see precisely how to reproduce it |
| 20:30.20 | *** join/#asterisk FuriousGeorge (182c2966@gateway/web/freenode/ip.24.44.41.102) |
| 20:30.21 | navaismo | in 11.7.0 its ok i have machines working with 11.7 |
| 20:30.30 | willwh | newtonr: ok |
| 20:30.38 | willwh | yeah - just deployed again and it's broken |
| 20:30.48 | willwh | 11.6 cert might be worth a shot? |
| 20:31.13 | willwh | going to give it a whirl anyway |
| 20:31.14 | willwh | :) |
| 20:31.19 | JustinAiken | I have two contexts set up in my sip.conf coming from the same host… When a call comes to the second one, I get: |
| 20:31.21 | willwh | I'll try 11.7.0 otherwise |
| 20:31.22 | JustinAiken | username mismatch, have <first-context>, digest has <context-i-actually-want> |
| 20:31.22 | JustinAiken | How do I get calls to the second context auth'd based on that user/pass instead of the first context matching the hostname? |
| 20:31.45 | rrittgarn | ~pb |
| 20:31.46 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 20:31.56 | *** join/#asterisk paulc (~root@unaffiliated/paulc) |
| 20:32.15 | newtonr | willwh, is your asterisk box behind NAT ? |
| 20:32.22 | willwh | nope, direct IP |
| 20:32.41 | willwh | well, office is like this: data network, and voice network |
| 20:32.49 | willwh | I am calling from a sipml5 client on the data network to the voice network |
| 20:33.02 | willwh | so it's over WAN, no NAT on the asterisk box - it's direct on to the net |
| 20:33.13 | willwh | but I am going through a NAT router out of our data network |
| 20:36.29 | newtonr | willwh, hrm. idk at the moment without having debug or configs in front of me. i'm working on something else, but this evening i might get a chance to help narrow it down. Remember to upload a reproduction guide and configs to that issue and I'll look at it later today. |
| 20:38.22 | *** join/#asterisk SuperBawlz (~IceChat@rrcs-71-43-76-226.se.biz.rr.com) |
| 20:40.53 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 20:41.48 | navaismo | willwh, yep confirmed, asterisk answer without the ice-ufrag and ice-pwd |
| 20:41.58 | navaismo | willwh, now let me downgrade to 11.7 |
| 20:43.11 | *** part/#asterisk JeffC_NN (~JeffC_NN@mail.nownerd.com) |
| 20:44.08 | navaismo | poke file ^ |
| 20:44.52 | newtonr | navaismo, was your setup the same as my tutorial and then you just called from the client to Asterisk? |
| 20:46.03 | *** join/#asterisk bulkorok (~Adium@053d9234.dynamic.tele-ag.de) |
| 20:46.18 | navaismo | the wiki post? if so, basically yes using jssip to call asterisk and receive that |
| 20:47.16 | newtonr | navaismo, can you try with the sipml5 demo as well and see if that results in the same thing? |
| 20:50.06 | novaflash | hello people. i have a question that i've been messing with for the past hour or so. i'd like to ring 2 phones at the same time - leave the first one ringing till it dies, and the second one should ring 20 seconds and then go to another phone. somehow it's doing quite funky things when i try stuff like Dial(SIP/1234@bla&5678@bla,20) as a first line. and idea how i should handle this properly? |
| 20:50.24 | navaismo | newtonr, yep same thing |
| 20:51.00 | rrittgarn | @novaflash: http://www.voip-info.org/wiki/view/Asterisk+local+channels |
| 20:51.03 | newtonr | navaismo, thanks a ton |
| 20:52.09 | navaismo | anytime |
| 20:54.21 | novaflash | rrittgarn: ah i see, so i can set up new stuff with its own dialplan to follow, and call 2 of these routines at the same time? |
| 20:54.52 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 21:07.24 | rrittgarn | novaflash: yes. You could dial one SIP channel and one Local that times out, only thing I'm not sure of without testing is if it will hangup the parent channel... if so you will have to do a wait() to wait out the initial dial... something to play with |
| 21:07.32 | *** join/#asterisk Hive (~Adam@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 21:08.01 | novaflash | i've elected to run 2 locals are the same time, each doing their thing. seems to work okay |
| 21:08.05 | novaflash | thanks :) |
| 21:08.35 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:08.45 | Hive | Is there a way to send a DTMF tone to the called party after a call is bridged? |
| 21:09.31 | [TK]D-Fender | Hive: You'll have to launch an external process that will bridge in for it |
| 21:09.49 | Hive | Ugh this is so weird |
| 21:09.58 | Hive | I had it working a week ago using the D option with dial |
| 21:10.30 | Hive | TK: like fire a script on answer? |
| 21:11.15 | justdave | I'm attempting to build Asterisk 1.8.26.1, and configure is failing on the DAHDI_RESET_COUNTERS in dahdi/user.h test, but /usr/include/dahdi/user.h exists and does include that variable |
| 21:12.09 | [TK]D-Fender | Hive: Yes. D() is pre-bridge and therefor not what you asked for |
| 21:12.13 | justdave | from config.log it looks like dahdi/user.h is complaining that dahdi/dahdi_config.h doesn't exist |
| 21:12.16 | justdave | which it doesn't. |
| 21:12.28 | *** part/#asterisk JustinAiken (~Adium@74.120.39.46) |
| 21:12.41 | justdave | what's supposed to generate that? I just built dahdi-tools, too, which is where the rest of those headers came from |
| 21:12.56 | Hive | TK: Could I just jump to a context on bridge then use SendDTMF? |
| 21:16.40 | Hive | Ok looks like you can't jump to a context after bridge. FML |
| 21:18.31 | willwh | navaismo: wow awesome, thx |
| 21:18.37 | willwh | newtonr: thx to you too |
| 21:18.50 | willwh | and both of you, I'll let you know how my 11.6-cert goes.. :) |
| 21:21.16 | Hive | [TK]D-Fender: Could I use the G( ) option of dial to go to a different priority which would use the SendDTMF app? |
| 21:22.35 | *** join/#asterisk xaristax (~xaristax@187-162-1-148.static.axtel.net) |
| 21:25.36 | xaristax | Hi someone know how can i show in the screen the callerid of the person that i call i mean if i digit in my phone 312 it only shows 312 but i wanna show a text is this posible? |
| 21:30.30 | *** join/#asterisk bulkorok (~Adium@053d9234.dynamic.tele-ag.de) |
| 21:39.16 | [TK]D-Fender | xaristax: Set the name/ |
| 21:40.01 | xaristax | Fender can you give me an example :D |
| 21:41.56 | [TK]D-Fender | "core show function CALLERID" |
| 21:42.38 | xaristax | Thaks fender im gonna read it |
| 21:43.27 | xaristax | Fender this only works for the peer im calling right? but what about my phone how can i show the info of the peer |
| 21:43.28 | xaristax | ??? |
| 21:52.44 | [TK]D-Fender | Your wording does not make sense |
| 21:53.25 | [TK]D-Fender | When you call for a device, it sends a callerID. then when * accepts that call, the peer that it matches against can override it. Then it goes to the dialplan where again you can specifically override it. |
| 21:53.45 | [TK]D-Fender | Then when you dial () whatever that last one did... is what they see |
| 21:56.06 | [TK]D-Fender | xaristax: Ok, I re-read your question and I think I missed what you wer loking for. |
| 21:56.44 | [TK]D-Fender | "core show function CONNECTEDLINE" |
| 21:56.47 | [TK]D-Fender | ^ |
| 21:56.59 | justdave | ok, found my dahdi_config.h, dahdi-tools was building it but epel's spec file wasn't including it in the package. fixed that up and everything builds like it's supposed to. |
| 21:57.43 | xaristax | i dont speak very well the english :P and maybe the idea that i have is wrong but i want to dial a number and get the info of this phone that im calling |
| 21:58.58 | xaristax | Fender is exactly what i was looking for THANKS :D |
| 22:00.20 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 22:01.24 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 22:01.41 | navaismo | OMG newtonr rolled back to 11.7 still the same error, was working :'( |
| 22:02.54 | *** part/#asterisk mjordan (~matt@nat/digium/x-vzrgkpyiwvtigmdt) |
| 22:04.07 | newtonr | navaismo, ewe, weird |
| 22:04.46 | newtonr | navaismo, be sure to upload your configs and stuff to one of those issues i linked so it'll be easier for us to examine it |
| 22:08.27 | willwh | hey guys |
| 22:08.33 | willwh | so interesting, 11.6-cert works, kinda |
| 22:08.43 | willwh | my calls conect and stay connected, people can hear me speak from sipml5 |
| 22:08.49 | willwh | but I don't get any sound back |
| 22:13.23 | willwh | > 0x244c1c0 -- Probation passed - setting RTP source address to IVR_LAN_IP:3000 |
| 22:13.29 | willwh | > 0x243f410 -- Probation passed - setting RTP source address to SIPML5_CLIENT_IP:50265 |
| 22:13.46 | willwh | I would've thought the first two should be IVR_WAN_IP:3000 |
| 22:13.57 | willwh | when I say IVR, I mean asterisk |
| 22:14.00 | navaismo | check your nat settings and the IPs used in the SDP transaction to be sure are the corrects |
| 22:15.26 | willwh | my nat settings? :) |
| 22:16.17 | willwh | nat=force_rport,comedia |
| 22:18.04 | willwh | ye what is interesting, in my sipml5 client... I see lines like so: a=rtcp:50264 IN IP4 192.168.0.220 |
| 22:18.06 | willwh | a=candidate:2203805960 1 udp 2113937151 192.168.0.220 50264 typ host generation 0 |
| 22:18.22 | willwh | lots of the a=candidates, but all using my LAN IP |
| 22:18.24 | navaismo | ice searching your IP |
| 22:18.35 | navaismo | the m= show the IP used in the sdp |
| 22:19.49 | pabelanger | willwh, pastebin please |
| 22:21.00 | willwh | http://pastebin.com/7QvA4Bga |
| 22:21.33 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 22:29.32 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 22:29.50 | willwh | pabelanger: anything obvious? :) |
| 22:46.54 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 22:49.59 | DiscusZ | anyone around to answer a question? |
| 22:50.32 | WIMPy | ~ask |
| 22:50.32 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 22:52.26 | DiscusZ | when I do a show sip peer <peername> the from domain has a port how do i change this? on my old trixbox doing the same command does not have the 5060 port my provider uses 5065 for trunks (using the exact same config on my trixbox as asteriskNOW and I can not call out) |
| 22:52.28 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 22:58.59 | [TK]D-Fender | fromdomain does not imply the target port |
| 22:59.08 | [TK]D-Fender | PORT implies the port. |
| 23:01.03 | willwh | pabelanger: I'm headed homeward, but I'll idle in here... :) |
| 23:02.44 | DiscusZ | what does this mean (output from sip show peer) FromDomain : sip.babytel.ca Port 5060 on another box I have it shows up as FromDomain : sip.babytel.ca |
| 23:02.50 | *** join/#asterisk bitvilag (~HUNbitvi@62-165-198-139.pool.digikabel.hu) |
| 23:04.23 | bitvilag | hey everyone i was wondering if anyone knows by hart why my cisco 7940 stopped registering both lines to the asterisk after switching to local network. one is fine and working perfectly but the other seems to timeout. It does not register however after about 20s call initiation ...it starts riningig |
| 23:04.27 | bitvilag | ringing |
| 23:04.47 | [TK]D-Fender | DiscusZ: Do you have an actual call failing? |
| 23:06.11 | DiscusZ | yes outbound calls come back and says the number is not answering (inbound works fine) |
| 23:07.43 | [TK]D-Fender | Show us the call. |
| 23:07.46 | [TK]D-Fender | ~pb |
| 23:07.47 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 23:07.48 | [TK]D-Fender | ^^^ |
| 23:08.13 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 23:11.15 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 23:12.00 | DiscusZ | I am getting outbound registration error -> [2014-03-27 17:10:07] NOTICE[25578]: chan_sip.c:23472 handle_response_register: Outbound Registration: Expiry for sip.babytel.ca is 55 sec |
| 23:14.22 | file | that's not an error. |
| 23:16.23 | bitvilag | hey d-fender:) |
| 23:21.13 | willwh | pabelanger: I figured it out :) |
| 23:22.48 | willwh | so interestingly - with a google stun server set in rtp.conf |
| 23:23.01 | willwh | I only got one way audio |
| 23:23.07 | willwh | yanked it, core restart when convenient |
| 23:23.10 | willwh | we're in business |
| 23:23.18 | willwh | :) |
| 23:23.28 | willwh | navaismo ^ weird :) |
| 23:32.38 | bitvilag | is there a way to check incoming registration requests? |
| 23:35.43 | [TK]D-Fender | "sip set debug on" <- |
| 23:40.31 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
| 23:41.04 | dan_j | Hi, is there a channel for general sip conversation (not asterisk specific) |
| 23:41.35 | dan_j | I'm trying to figure out how to get a polycom to only listen to sip packets from its registration server. |
| 23:44.20 | WIMPy | What about the manual? |
| 23:46.19 | jmmills | wishes for a join function right now |
| 23:46.37 | jmmills | Doing conditional expressions to build dial strings is no fun |
| 23:47.15 | bitvilag | there are too many messages on debug |
| 23:47.19 | bitvilag | how can I filter? |
| 23:47.25 | jmmills | change debug level |
| 23:47.29 | jmmills | core set debug $level |
| 23:47.37 | bitvilag | i just need one ext |
| 23:48.02 | bitvilag | by the way anyone used two lines with one ext? |
| 23:48.19 | [TK]D-Fender | Don't call a device an "extension" |
| 23:48.21 | [TK]D-Fender | And yes |
| 23:48.43 | bitvilag | i do not...I didnt mention device |
| 23:49.01 | bitvilag | extension is what I use in asterisk |
| 23:50.24 | bitvilag | in other words I have a 7940 which support two lines and for some reason if I nat and use the server from public both of them register just fine but if I use the local network with routed path only one of them register however ...the line that is not registered to the extention of asterisk ... is still working just with a 20s delay?! |
| 23:51.11 | bitvilag | i was trying to figure out what the crap is going on but too many messages going around |
| 23:51.16 | bitvilag | so whats the level I need? |
| 23:52.18 | *** join/#asterisk paulc (~root@unaffiliated/paulc) |
| 23:53.10 | bitvilag | i used 3 and what is odd is for a second I saw invalid sip messages caller id rejected...how is that possible if its identical...? or it might be a third issue that I didnt even notice not related to this... |
| 23:56.01 | [TK]D-Fender | first set your device as type=friend |
| 23:56.28 | [TK]D-Fender | because if you leave it as "peer" you'll get digest issues because it'll auth by IP and match the other entry it communicates as |
| 23:58.22 | bitvilag | I use at as friend |
| 23:59.30 | bitvilag | the thing I dont understand is how can one line register and not the other ...how can still work but with delay and I tried out...I can call at the same time |
