| 00:06.11 | *** join/#asterisk PepOSX (n=pepOSX@201.248.215.16) |
| 00:12.03 | teknoprep | hey all |
| 00:12.21 | timeshell | hi ho |
| 00:15.06 | teknoprep | hey Bananaskin |
| 00:16.14 | Bananaskin | lo |
| 00:16.26 | *** join/#asterisk adam1 (n=adam1@d150-220-108.home.cgocable.net) |
| 00:16.31 | *** part/#asterisk adam1 (n=adam1@d150-220-108.home.cgocable.net) |
| 00:17.10 | WilliamK | can anyone verify if it's just me - on the latest SVN - if you try and do sip debug peer 134 the command works, however the CLI help menu doesn't finish out |
| 00:17.10 | Bananaskin | teknoprep u have mail |
| 00:17.37 | teknoprep | ty |
| 00:18.26 | Bananaskin | create the sccp.conf from the example on the forum link |
| 00:18.59 | teknoprep | ok |
| 00:19.03 | teknoprep | give me a minute |
| 00:19.04 | teknoprep | lol |
| 00:19.19 | teknoprep | i have to move upstairs i don't have a cisco phone here next to me... you know what |
| 00:19.22 | teknoprep | i am going to move it down here brb |
| 00:19.30 | Bananaskin | k |
| 00:21.34 | teknoprep | ok back |
| 00:22.25 | Bananaskin | k |
| 00:23.49 | teknoprep | 10 carmina dr 19608 |
| 00:23.56 | teknoprep | Dec 22 19:23:40 WARNING[27933] loader.c: Loading module chan_sccp.so failed! |
| 00:24.27 | teknoprep | Dec 22 19:23:40 VERBOSE[27933] logger.c: [chan_sccp.so]Dec 22 19:23:40 WARNING[27933] loader.c: /usr/lib/asterisk/modules/chan_sccp.so: undefined symbol: __ast_string_field_empty |
| 00:24.53 | Bananaskin | hmmmm, you may have to compile your own version |
| 00:25.11 | Bananaskin | I know the 1.2 version works across who 1.2.x relese |
| 00:25.59 | teknoprep | i can get it off the repository i hav einstalled |
| 00:26.30 | Bananaskin | k |
| 00:26.55 | *** join/#asterisk droemel (n=droemel@p548EB24C.dip0.t-ipconnect.de) |
| 00:27.20 | teknoprep | to do this tho i have to upgrade some stuff.. will take a bit |
| 00:27.25 | Bananaskin | nps |
| 00:28.20 | teknoprep | if sccp works |
| 00:28.22 | teknoprep | in asteris |
| 00:28.32 | teknoprep | there are alot of benifits with sccp on cisco devices |
| 00:28.39 | Bananaskin | not a case of if... it does |
| 00:28.48 | teknoprep | sorry for using enter as punctuation... i usually do not do that |
| 00:28.55 | Bananaskin | yep, you loose some of the features using sip |
| 00:29.29 | Bananaskin | ie the ability to show the called or received number on a handset on the line appearances of a different phone etc |
| 00:29.31 | teknoprep | doing a yum -y update |
| 00:29.34 | teknoprep | going to take a bit |
| 00:29.38 | Bananaskin | Using the Hints facility |
| 00:29.56 | teknoprep | tweaking of input gain |
| 00:29.59 | teknoprep | seems to have been lost |
| 00:30.48 | teknoprep | only case from www.cisco.com i found on changing input gain was if you were using cisco call manager on a cisco switch/router and you set the router/switch port to gain or lower the input gain |
| 00:31.31 | Bananaskin | hmmm, must look into that cos I run a few bits of cisco kit here |
| 00:31.36 | Bananaskin | routers and switches |
| 00:32.05 | *** join/#asterisk matsk (n=mk@83.233.97.210) |
| 00:32.29 | teknoprep | yeah 3 years ago i ended up having to get my ccna ccnp for a contracted long-term posistion |
| 00:32.33 | teknoprep | havn't needed it since |
| 00:33.18 | Bananaskin | u got a smartnet account or do u want me to grab firmware for ya |
| 00:33.25 | teknoprep | yes please grab it |
| 00:33.28 | teknoprep | i never renewed it |
| 00:33.37 | teknoprep | i am thinking about having a client of mine do that fo rme |
| 00:33.40 | teknoprep | and just use thers |
| 00:34.03 | teknoprep | i have 7940 phones |
| 00:34.15 | teknoprep | but i think the firmware is the same for 7960 |
| 00:34.26 | Hadi- | anyone here using the cisco phone |
| 00:34.28 | Hadi- | with 729 |
| 00:34.31 | teknoprep | 110 updates to go |
| 00:34.32 | Hadi- | codec with asterisk? |
| 00:34.39 | Bananaskin | Hadi- yep |
| 00:34.44 | teknoprep | Hadi-, we are setting up sccp with * |
| 00:34.55 | teknoprep | Hadi-, Bananaskin tells me it works better with these phones |
| 00:35.46 | teknoprep | hey Bananaskin have you ever worked with cisco sip ? |
| 00:36.05 | teknoprep | do you notice a difference using SCCP with * and cisco phones over * / sip / cisco |
| 00:36.45 | teknoprep | i mean this is what this is really all about lol |
| 00:36.47 | Bananaskin | yeah, I used to have 7940's and 60's on sip |
| 00:37.14 | teknoprep | from what [TK]D-Fender tells me... cisco sip implementation blows |
| 00:37.36 | teknoprep | and from what i have found from use... is pretty much the same |
| 00:37.53 | *** join/#asterisk Deeewayne (n=Deeewayn@ool-43522b13.dyn.optonline.net) |
| 00:37.53 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 00:37.56 | Hadi- | well |
| 00:38.01 | Hadi- | im having nothing but issues |
| 00:38.06 | Hadi- | with the Cisco 7960 |
| 00:38.09 | Bananaskin | IMHO the cisco performs better using sccp |
| 00:38.14 | Hadi- | codec g729a |
| 00:38.17 | *** join/#asterisk xim010 (n=xim010@75-120-205-174.dyn.centurytel.net) |
| 00:38.17 | Hadi- | on asterisk 1.2 |
| 00:38.22 | teknoprep | Hadi-, what is the problem? |
| 00:38.25 | teknoprep | Hadi-, echo ? |
| 00:38.27 | Bananaskin | Hadi- what issues ? |
| 00:38.27 | Hadi- | even when I disable VAD |
| 00:38.32 | Hadi- | I lose audio |
| 00:38.38 | teknoprep | Hadi-, are you using trixbox ? |
| 00:38.39 | Hadi- | every once in a while |
| 00:38.54 | teknoprep | Hadi-, i had that problem ALOT when using trixbox with cisco phones |
| 00:39.04 | Bananaskin | what g729 are u using |
| 00:39.04 | Hadi- | well |
| 00:39.15 | Hadi- | the one from digium |
| 00:39.30 | teknoprep | Hadi-, the free one or the one that you pay for ? |
| 00:39.31 | Bananaskin | hmmm |
| 00:39.38 | Hadi- | pay for it |
| 00:39.53 | Hadi- | there is no issues with any of the ATA's |
| 00:39.54 | Bananaskin | I use the free one with no bother, I bought 1 licence from Digium and the fecker didn't work |
| 00:39.59 | Hadi- | only issue is with Cisco phones |
| 00:40.15 | teknoprep | Hadi-, are you using trixbox ? |
| 00:40.22 | Hadi- | yes |
| 00:40.23 | Bananaskin | how many licences did you buy ? 1 |
| 00:40.27 | Hadi- | 10 |
| 00:40.27 | teknoprep | Hadi-, that is your problem |
| 00:40.32 | xim010 | Could someone assist me with attempting to get my X100P working? |
| 00:40.41 | Hadi- | I think the issue is not trixbox |
| 00:40.46 | Hadi- | its the version of asterisk |
| 00:40.48 | Hadi- | 1.2 |
| 00:40.50 | teknoprep | Hadi-, i had the same problem as you |
| 00:41.02 | teknoprep | Hadi-, i would suggest using elastix if you must have a GUI |
| 00:41.10 | teknoprep | Hadi-, or just install asterisk / freepbx by hand |
| 00:41.11 | Bananaskin | Well TBH guys, I am running 1.2 here and G729 is fine |
| 00:41.15 | teknoprep | Hadi-, freepbx is REALLY easy to install |
| 00:41.31 | Hadi- | well |
| 00:41.35 | Hadi- | the issue is with Silence Suppression |
| 00:41.36 | Deeewayne | Bananaskin: if you bought a license from Digium and it 'doesn't work' then contact their technical support |
| 00:41.39 | Hadi- | and asterisk 1.2 |
| 00:42.06 | teknoprep | Hadi-, if you turn off VAD and you loose voice ... its not silence supression |
| 00:42.20 | teknoprep | Hadi-, are you using SIP channels ? |
| 00:42.21 | Bananaskin | Deeewayne the "doesn't work" bit is that it didn't create the appropriate file when the licence server ran |
| 00:42.24 | teknoprep | Hadi-, to your provider? |
| 00:42.40 | Hadi- | well every time |
| 00:42.42 | Hadi- | i lose audio |
| 00:42.44 | Hadi- | I get: |
| 00:42.45 | Hadi- | 2007-12-19 21:12:57 NOTICE[6760]: rtp.c:415 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389) |
| 00:42.48 | Hadi- | in asterisk CLI |
| 00:43.02 | *** join/#asterisk eluizbr (n=eluizbr@201.78.140.135) |
| 00:43.04 | Bananaskin | hmmm |
| 00:43.15 | Hadi- | teknoprep: yes |
| 00:43.20 | Hadi- | SIP is what we are using |
| 00:43.24 | teknoprep | Hadi-, try enabling the jitterbuffer |
| 00:43.35 | teknoprep | Hadi-, in your sip.conf do this |
| 00:43.42 | eluizbr | hello |
| 00:44.14 | Bananaskin | Hadi- a wee google of the error would have given you - 1. Explanation: |
| 00:44.15 | Bananaskin | Asterisk does not (yet) support voice activity detection (and comfort noise generation). |
| 00:44.15 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 00:44.18 | eluizbr | Someone can help me with an error on the asterisk CLI 1.4.16.2 |
| 00:44.38 | eluizbr | my error: |
| 00:44.39 | eluizbr | asterisk -rcvvvvvvvv |
| 00:44.39 | eluizbr | Asterisk 1.4.16.2, Copyright (C) 1999 - 2007 Digium, Inc. and others. |
| 00:44.39 | eluizbr | Created by Mark Spencer <markster@digium.com> |
| 00:44.39 | eluizbr | Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. |
| 00:44.39 | eluizbr | This is free software, with components licensed under the GNU General Public |
| 00:44.41 | eluizbr | License version 2 and other licenses; you are welcome to redistribute it under |
| 00:44.43 | eluizbr | certain conditions. Type 'core show license' for details. |
| 00:44.45 | eluizbr | ========================================================================= |
| 00:44.47 | Deeewayne | eluizbr: what is the error message ? |
| 00:44.47 | eluizbr | <PROTECTED> |
| 00:44.49 | eluizbr | <PROTECTED> |
| 00:44.51 | eluizbr | Segmentation fault |
| 00:45.06 | eluizbr | deeewayne: asterisk -rcvvvvvvvv |
| 00:45.06 | eluizbr | Asterisk 1.4.16.2, Copyright (C) 1999 - 2007 Digium, Inc. and others. |
| 00:45.06 | eluizbr | Created by Mark Spencer <markster@digium.com> |
| 00:45.06 | eluizbr | Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. |
| 00:45.06 | eluizbr | This is free software, with components licensed under the GNU General Public |
| 00:45.07 | eluizbr | License version 2 and other licenses; you are welcome to redistribute it under |
| 00:45.07 | teknoprep | in your sip.conf add this |
| 00:45.09 | eluizbr | certain conditions. Type 'core show license' for details. |
| 00:45.11 | eluizbr | ========================================================================= |
| 00:45.13 | eluizbr | <PROTECTED> |
| 00:45.14 | teknoprep | jbenable =yes |
| 00:45.14 | Hadi- | eluizbr |
| 00:45.15 | eluizbr | <PROTECTED> |
| 00:45.15 | Bananaskin | ffs |
| 00:45.17 | eluizbr | Segmentation fault |
| 00:45.18 | Hadi- | please |
| 00:45.18 | Hadi- | use |
| 00:45.19 | teknoprep | jbforce = yes |
| 00:45.20 | jwh | pastebin? |
| 00:45.21 | Hadi- | pastebin |
| 00:45.28 | jwh | ~pb |
| 00:45.28 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste |
| 00:45.29 | teknoprep | Hadi-, jbenable = yes |
| 00:45.32 | *** join/#asterisk RoyK (n=roy@ti211210a080-0234.bb.online.no) |
| 00:45.34 | teknoprep | Hadi-, jbforce = yes |
| 00:45.37 | Hadi- | ook |
| 00:45.39 | teknoprep | Hadi-, in your sip.conf |
| 00:45.40 | Hadi- | let me try |
| 00:45.44 | Bananaskin | Hadi- did u see the post ? |
| 00:45.56 | Deeewayne | eluizbr: pastebin the entire output and I'll look at it |
| 00:45.58 | Bananaskin | look at - http://www.asteriskguru.com/tutorials/comfort_noise_support_incomplete.html |
| 00:46.13 | eluizbr | Deeewayne: pastebin is out |
| 00:46.20 | xim010 | is there one for setting up FXO with X110P |
| 00:46.30 | xim010 | X100P sory |
| 00:46.39 | Deeewayne | eluizbr: where ? |
| 00:46.40 | Hadi- | teknoprep: under [general] ? |
| 00:46.57 | eluizbr | Deeewayne: http://pastebin.org/12575 |
| 00:47.19 | teknoprep | Hadi-, yeah |
| 00:47.34 | Bananaskin | xim010 - http://users.pandora.be/Asterisk-PBX/InstallWildcard.htm |
| 00:47.54 | Hadi- | what is that for anyways? ;P |
| 00:48.28 | eluizbr | Deewayne: http://pastebin.org/12577 my CPU |
| 00:48.29 | xim010 | Thanks Bananaskin |
| 00:48.36 | Bananaskin | nps |
| 00:48.39 | xim010 | I will give it a shot |
| 00:49.09 | teknoprep | wow so much crap to update wtf |
| 00:49.13 | teknoprep | Hadi-, jitterbuffer |
| 00:49.34 | Hadi- | okay lets try this and see if it made any changes |
| 00:49.39 | eluizbr | Deeewayne: can you help me? |
| 00:49.51 | teknoprep | Hadi-, make sure it parses that properly i don't know if that does anything inside of * 1.2 |
| 00:50.12 | timeshell | Yah, on the subject of the X100P... mine seems stuck on IRQ7 |
| 00:50.13 | Hadi- | teknoprep: it looks like its only for 1.4 |
| 00:50.21 | Hadi- | http://www.voip-info.org/tiki-index.php?page=Asterisk+new+jitterbuffer |
| 00:50.24 | teknoprep | Hadi-, i would suggest updating |
| 00:50.27 | timeshell | But I need that IRQ for my LPT port |
| 00:50.33 | *** join/#asterisk windsor510 (n=win@adsl-75-24-215-230.dsl.pltn13.sbcglobal.net) |
| 00:50.35 | teknoprep | Hadi-, try using elastix if you need to have the GUI |
| 00:50.48 | teknoprep | Hadi-, elastix keeps up2date on new releases of asterisk |
| 00:51.06 | teknoprep | i would honestly suggest what i am doing |
| 00:51.06 | Bananaskin | timeshell u can assign irq for printer in BIOS |
| 00:51.17 | teknoprep | and that is read and learn asterisk |
| 00:51.18 | timeshell | I tried...didn't work |
| 00:51.33 | teknoprep | Bananaskin, only 160 updates to go |
| 00:51.38 | Bananaskin | is that all |
| 00:51.40 | Bananaskin | :) |
| 00:51.56 | timeshell | At any rate, I'd prefer to change the IRQ of the X100P. For a modern card, it seems unusual that you can't change the IRQ |
| 00:52.01 | timeshell | What if you wanted to use 2 of them? |
| 00:52.15 | eluizbr | Someone can help me with an error on the asterisk CLI 1.4.16.2 |
| 00:52.30 | eluizbr | my error http://pastebin.org/12575 |
| 00:52.43 | eluizbr | my CPU info http://pastebin.org/12577 |
| 00:52.54 | teknoprep | hey Bananaskin wtf is up with the .exe you sent me? |
| 00:52.58 | windsor510 | Hi, I've been reading how-to's on setting up Asterisk behind a DSL router / firewall. If I am able to port forward to my internal Asterisk server, do I stand any chance of establishing voice calls from outside my firewall (Internet)? |
| 00:53.17 | Bananaskin | its a rar |
| 00:53.44 | Bananaskin | ohh, I see what u mean |
| 00:53.49 | teknoprep | inside the rar is an exe file |
| 00:53.51 | teknoprep | i run ubuntu man |
| 00:53.56 | Bananaskin | just ran it herte |
| 00:53.59 | teknoprep | i have a window smachine upstairs lol |
| 00:54.03 | Deeewayne | eluizbr: can you post a backtrace? |
| 00:54.12 | teknoprep | i have crossoffice which ran the rar file |
| 00:54.16 | Bananaskin | 2 secs teknoprep |
| 00:54.17 | teknoprep | but didn't do much other then error out |
| 00:54.59 | eluizbr | Deeewayne: What it is? |
| 00:55.28 | eluizbr | Deeewayne: Better .. How so? |
| 00:55.36 | teknoprep | windsor510, use IAX2 provider |
| 00:55.41 | Deeewayne | ~bt |
| 00:55.41 | jbot | bt sux0rs. Bhutan |
| 00:55.51 | teknoprep | windsor510, you only have to forward port 4569 |
| 00:57.18 | Deeewayne | eluizbr: http://forums.digium.com/viewtopic.php?p=59438&sid=ae696e057d9b3617739d3444ee11d422 |
| 00:57.42 | Bananaskin | teknoprep email sent |
| 00:57.52 | *** part/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 00:57.52 | teknoprep | ty and the updates are done |
| 00:58.25 | Deeewayne | eluizbr: check russell's nov 2,2007 8:41 am post |
| 00:59.24 | teknoprep | i just install chan_sccp from the repo |
| 01:00.02 | teknoprep | [root@asterisk1 modules]# asterisk -r |
| 01:00.02 | teknoprep | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 01:00.11 | teknoprep | asterisk is running tho... |
| 01:00.23 | teknoprep | is there a way to fix this problem without rebooting the server |
| 01:00.26 | Bananaskin | did it install ok, make sure that you have in modules.conf noload statement for the skinny.so |
| 01:00.31 | teknoprep | yeah it installed fine |
| 01:00.34 | teknoprep | let me reboot the server |
| 01:00.52 | eluizbr | Deeewayne: http://pastebin.org/12581 |
| 01:01.05 | jwh | teknoprep: do you really need to reboot? why not just kill asterisk.. |
| 01:01.13 | teknoprep | killing asterisk |
| 01:01.17 | teknoprep | is what makes that happen |
| 01:01.20 | jwh | oh |
| 01:01.30 | jwh | does it get stuck? |
| 01:02.00 | jwh | ie; is the process in a STOP state? |
| 01:02.10 | teknoprep | no |
| 01:02.13 | teknoprep | asterisk is working fine |
| 01:02.21 | jwh | oh right |
| 01:02.21 | teknoprep | it just won't connect to the asterisk -r cli |
| 01:02.34 | jwh | and the path is correct? |
| 01:02.43 | teknoprep | well think of it this way |
| 01:02.44 | teknoprep | if i reboot |
| 01:02.47 | teknoprep | everything works fine |
| 01:02.53 | jwh | bizarre |
| 01:02.58 | teknoprep | yup |
| 01:03.21 | xim010 | OK I am not sure where to go from here ... all this thing will show me is that I have pseudo channel |
| 01:03.46 | xim010 | lspci shows me the x100p but I can't seem to do anything with it |
| 01:04.04 | windsor510 | teknoprep, sorry I got distracted. Can you recommend any IAX2 soft-phone's that will work in Linux and/or Windows? |
| 01:04.52 | windsor510 | I have calls going in/out, but the RTP hand-off is failing when going through my firewall, so I just don't hear any audio. |
| 01:05.25 | teknoprep | idefisk |
| 01:05.28 | Bananaskin | windsor510 idefisk |
| 01:05.30 | teknoprep | Bananaskin, you still there |
| 01:05.33 | teknoprep | guess so |
| 01:05.38 | Bananaskin | :) |
| 01:05.48 | teknoprep | which file do i update for sccp to tell the phone to use the sccp file you gave me |
| 01:06.12 | teknoprep | is it still the SIPDefault.cnf |
| 01:06.17 | Bananaskin | in the tftp dir ? |
| 01:06.29 | teknoprep | yeah |
| 01:06.42 | Bananaskin | 2 secs |
| 01:07.14 | windsor510 | awesome, I will give this a shot. looks like idefisk is now called zoiper. |
| 01:07.15 | squigly | so im trying to debug why i cant register with my provider |
| 01:07.20 | squigly | any ideas where to start? |
| 01:07.28 | teknoprep | yeah thats it |
| 01:07.29 | teknoprep | zoiper |
| 01:08.29 | windsor510 | lol, lynx web browser doesn't like my firewall's javascript configuration menu. I may not be able to forward iax2 remotely. |
| 01:08.44 | xim010 | Where can I go to find out what I need to do to determine if this card is no good or what I need to do to make it work? Please help if you can or direct me to where I can get help with this thing. |
| 01:10.20 | Bananaskin | teknoprep |
| 01:10.28 | teknoprep | yo |
| 01:10.58 | Bananaskin | ok, in the tftpboot dir there is a file OS79XX.TXT ? |
| 01:11.06 | teknoprep | yup |
| 01:11.11 | teknoprep | what do i need to put in there? |
| 01:11.17 | *** join/#asterisk eluizbr (n=eluizbr@201.78.140.135) |
| 01:11.18 | Bananaskin | 2 secs, will paste |
| 01:11.35 | Bananaskin | P00308000700 |
| 01:11.38 | eluizbr | Deeewayne: http://pastebin.org/12581 |
| 01:12.06 | Bananaskin | make sure that you unzip the contents of the zip to the tftpboot dir as well of course |
| 01:12.16 | teknoprep | rebooting the phone |
| 01:12.18 | teknoprep | well yeah of course |
| 01:12.20 | Bananaskin | you will also require a config file for the phone |
| 01:12.24 | teknoprep | i apreciate the obvious |
| 01:12.29 | eluizbr | Deeewayne: My internet fell .. You entered something? |
| 01:12.29 | Bananaskin | so don't reboot the phone yet |
| 01:12.39 | teknoprep | well i want to see if the updates go through |
| 01:12.45 | teknoprep | hmm |
| 01:12.46 | Bananaskin | k |
| 01:12.54 | *** join/#asterisk Unst4ble (n=Unstable@87-194-206-186.bethere.co.uk) |
| 01:13.14 | Unst4ble | Hey all, anyone got experience setting up a VoIPStunt SIP trunk? |
| 01:14.07 | teknoprep | its not taking the updates |
| 01:14.15 | teknoprep | Bananaskin, can you send me a sample SEP*.cnf file |
| 01:14.16 | *** join/#asterisk ariel_ (n=ariel_@dsl-20-177.cofs.net) |
| 01:14.21 | Bananaskin | 2 secs |
| 01:14.23 | teknoprep | ty |
| 01:16.06 | eluizbr | Deeewayne: ?? |
| 01:17.06 | Bananaskin | teknoprep check mail |
| 01:18.06 | Bananaskin | example config - make sure you save it as SEP<MACADDY>.cnf.xml |
| 01:18.33 | Bananaskin | make changes as necessary IP wise to suit network |
| 01:18.58 | teknoprep | yup |
| 01:19.02 | teknoprep | already done |
| 01:19.06 | teknoprep | and loading up the new firmware |
| 01:19.19 | teknoprep | i am pretty good with stuff.. this is just really new configurations |
| 01:20.29 | xim010 | Does Wildcard x100p carry such a stigma that no one anywhere will even talk about it? |
| 01:20.34 | teknoprep | Bananaskin, missing some file United_states |
| 01:20.36 | Deeewayne | eluizbr: sorry, I'm in and out... |
| 01:20.45 | Bananaskin | teknoprep yes thats fine |
| 01:21.00 | Deeewayne | <PROTECTED> |
| 01:21.16 | Bananaskin | xim010 - Nail, Hit and Head ring a bell |
| 01:21.26 | Deeewayne | when you start asterisk, type: 'asterisk -vvvvvvvvvvvvvvgc' |
| 01:22.03 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 01:22.06 | rob0 | xim010: It's junk. Did you have a specific question? |
| 01:22.16 | xim010 | Oh well I will drop it in the trash and move on to anther project |
| 01:22.20 | eluizbr | Deeewayne: This is for me? |
| 01:22.30 | Deeewayne | after it crashes, type 'gdb /usr/sbin/asterisk /path/to/core.12345, where 12345 is the asterisk pid |
| 01:22.38 | Deeewayne | eluizbr: yes ^^ for you |
| 01:22.52 | xim010 | nope I just want it to work. I have a friend and his works fine but this POS is just recognized and nothing else will fly |
| 01:23.11 | rob0 | you have zaptel drivers installed? |
| 01:23.15 | eluizbr | Deeewayne: ok.. moment please |
| 01:23.23 | Bananaskin | xim010 you will spend more time and effort trying to get it to work right, when in reality you should have bought a Digium or Sangoma or other reputable card. Don't worry though I imagine most of us here have tried a X100p of sorts and instantly made it into a boomerang |
| 01:23.23 | Deeewayne | eluizbr: at the gdb prompt, type 'bt' then 'bt full', and post the output to pastebin |
| 01:23.31 | eluizbr | ok |
| 01:23.41 | Deeewayne | eluizbr: I may be in and out, but post it and I'll look at it in a couple minutes |
| 01:23.43 | *** join/#asterisk saftsack (n=saftsack@p4FC77327.dip.t-dialin.net) |
| 01:24.06 | xim010 | I will save my pennies and do it right next time |
| 01:24.15 | rob0 | The one good thing about x10xp is that it's a cheap way to dabble in zaptel hardware. |
| 01:24.34 | eluizbr | ok.. thanks |
| 01:24.36 | xim010 | that was my initial idea and all it has been is a PITA |
| 01:24.36 | Bananaskin | rob0 - true statement |
| 01:25.15 | Bananaskin | teknoprep has the 7940 booted after the flash ? |
| 01:25.15 | teknoprep | Bananaskin, not really having any luck with tihs |
| 01:25.19 | teknoprep | Bananaskin, yes |
| 01:25.28 | Bananaskin | ok, so whats it doing ? |
| 01:25.37 | teknoprep | Bananaskin, opening 10.10.10.254 |
| 01:25.43 | xim010 | can someone recommend a reputable card that is reasonable for dabbling |
| 01:25.45 | teknoprep | Bananaskin, i have to change that quick to 10.10.10.101 |
| 01:26.06 | Bananaskin | did u edit the sccp.conf ? |
| 01:26.27 | eluizbr | Deeewayne: http://pastebin.org/12584 |
| 01:26.53 | Bananaskin | xim010 look at a Digium TDM card Card with 1 FXO port will cost approx £70 or US140 |
| 01:27.08 | xim010 | OK thanks you |
| 01:27.10 | Bananaskin | can support up to 4 ports in any cfg of FXO and FXS |
| 01:27.17 | Bananaskin | modular design |
| 01:27.23 | squigly | does any one know of a provider i can register my asterisk against as a client so i can try and debug my problem? |
| 01:27.24 | xim010 | have a url for reference |
| 01:27.27 | xim010 | ? |
| 01:27.31 | Bananaskin | http://www.ipchitchat.com/products/telephony.htm |
| 01:27.37 | rob0 | digium.com |
| 01:27.40 | Bananaskin | bottom card |
| 01:27.51 | teknoprep | Bananaskin, yes |
| 01:28.02 | teknoprep | Bananaskin, i have to now look to see if everything is right |
| 01:28.02 | xim010 | thank you all again. |
| 01:28.10 | *** part/#asterisk xim010 (n=xim010@75-120-205-174.dyn.centurytel.net) |
| 01:30.45 | Bananaskin | brb |
| 01:32.43 | teknoprep | Bananaskin, almost there |
| 01:32.53 | teknoprep | i have no lines registered |
| 01:32.57 | teknoprep | but the phone is up and running |
| 01:33.08 | Bananaskin | ok, you are getting somewhere |
| 01:33.22 | teknoprep | yup |
| 01:33.23 | Bananaskin | you have the cfg in the /etc/asterisk die |
| 01:33.27 | Bananaskin | dir even |
| 01:33.31 | teknoprep | yes of course |
| 01:33.39 | teknoprep | let me change the name of this |
| 01:33.44 | Bananaskin | there was 1 example ext on the cfg |
| 01:34.29 | Bananaskin | 1 example device and 1 example extn |
| 01:34.45 | teknoprep | yup |
| 01:34.51 | teknoprep | do i need both the device and the ext ? |
| 01:35.03 | Bananaskin | yes |
| 01:35.14 | Bananaskin | device is for devices etc etc |
| 01:35.22 | teknoprep | ok |
| 01:35.27 | *** join/#asterisk RoyK (n=roy@ti211210a080-0234.bb.online.no) |
| 01:35.34 | Bananaskin | in the device section, chnage the 7970 reference to 7940 |
| 01:37.04 | Bananaskin | the auto login line is the ext number of the lines which are created below the devices |
| 01:37.16 | Bananaskin | in the example I have 135 |
| 01:37.29 | Bananaskin | change that to whatever you require |
| 01:37.49 | teknoprep | already did that |
| 01:37.52 | teknoprep | 1001 |
| 01:38.13 | Bananaskin | ok at bottom of the device section, there is a device => SEPMACADDRESS |
| 01:38.14 | *** join/#asterisk RoyK (n=roy@ti211210a080-0234.bb.online.no) |
| 01:38.15 | eluizbr | Deeewayne: http://pastebin.org/12584 |
| 01:38.32 | teknoprep | already did that too Bananaskin |
| 01:38.43 | teknoprep | i don't understand how to register the line on the phone |
| 01:38.47 | teknoprep | in the config it says siplines |
| 01:38.50 | Bananaskin | 2 secs |
| 01:39.00 | teknoprep | for the SEP<mac> file |
| 01:39.34 | Bananaskin | what cfg says sip lines ? |
| 01:39.47 | teknoprep | nvm |
| 01:39.49 | teknoprep | i skrewed up |
| 01:39.52 | Bananaskin | hmmm |
| 01:40.06 | teknoprep | hey whats that webaccess ? |
| 01:40.15 | teknoprep | <webaccess>1</webaccess> |
| 01:40.23 | teknoprep | does the phone have a web config? |
| 01:40.29 | Bananaskin | you can access the web server in the phone for stats |
| 01:40.35 | teknoprep | oh ok |
| 01:40.35 | Bananaskin | never mind that for now |
| 01:40.53 | Bananaskin | have you then cfg'd the extn in the sccp.conf file as well ? |
| 01:40.59 | teknoprep | <ipAddr1>192.168.168.2</ipAddr1> |
| 01:41.02 | teknoprep | whats that for |
| 01:41.06 | teknoprep | yes Bananaskin |
| 01:41.13 | teknoprep | now i need to configure the SEP<MAC> |
| 01:41.23 | Bananaskin | the ip is the asterisk server |
| 01:41.42 | Bananaskin | the sepmac.cnf.xml is fine atm |
| 01:41.50 | Bananaskin | apart from the ip |
| 01:42.29 | teknoprep | where do i put my LINE information? |
| 01:42.34 | teknoprep | in the sepmac.cnf.xml ? |
| 01:42.37 | Bananaskin | no |
| 01:42.46 | *** part/#asterisk RoyK (n=roy@ti211210a080-0234.bb.online.no) |
| 01:42.48 | teknoprep | that goes into the sccp.conf |
| 01:42.51 | Bananaskin | just change the ipaddress section to relate to your PBX |
| 01:42.57 | teknoprep | got ya |
| 01:43.09 | Bananaskin | there are logical steps here and you are jumpin all around the place |
| 01:43.13 | Bananaskin | :) |
| 01:43.17 | teknoprep | lol |
| 01:43.56 | eluizbr | Deeewayne: http://pastebin.org/12584 |
| 01:44.15 | Bananaskin | teknoprep did you change the ip address in the sepmac.cnf.xml |
| 01:44.23 | teknoprep | yup |
| 01:44.25 | teknoprep | all done |
| 01:44.30 | Bananaskin | ok, save and do no more with it for now |
| 01:44.33 | teknoprep | i needed to change ALL the ip's correct |
| 01:44.47 | teknoprep | i set the ntp server to time.mit.edu |
| 01:44.49 | Bananaskin | no need just yet |
| 01:44.55 | teknoprep | oh |
| 01:44.59 | teknoprep | well i already did lol |
| 01:45.01 | Bananaskin | the ips at the bottom are for services |
| 01:45.03 | Bananaskin | ok |
| 01:45.21 | Bananaskin | do u already have a 1001 in freepbx |
| 01:45.32 | teknoprep | yes |
| 01:45.36 | teknoprep | do i need to remove it? |
| 01:45.37 | Bananaskin | ok, delete it |
| 01:45.42 | teknoprep | or set it up as a custom extension? |
| 01:45.55 | Deeewayne | eluizbr: can you try to latest 1.4 code ? |
| 01:46.02 | Bananaskin | yep custom |
| 01:46.19 | eluizbr | 1.4.16.1 |
| 01:46.31 | Bananaskin | atthe section - This device uses custom technology. the dial string is SCCP/1001 |
| 01:46.40 | Bananaskin | for the 1001 extn, clearly |
| 01:46.57 | eluizbr | deeewayne: It is happening any version of the series co 1.4 |
| 01:46.59 | teknoprep | ok thats done |
| 01:47.09 | *** join/#asterisk nirz (n=nir@bzq-79-183-137-211.red.bezeqint.net) |
| 01:47.21 | *** join/#asterisk truz_`24 (n=truz_`24@74-138-37-35.dhcp.insightbb.com) |
| 01:47.29 | eluizbr | Deeewayne: With the 1.2 series is not true that |
| 01:47.38 | teknoprep | Bananaskin, thats all done |
| 01:47.44 | teknoprep | Bananaskin, reboot the phone? |
| 01:47.48 | Bananaskin | yep |
| 01:47.59 | teknoprep | i am getting arm spasms |
| 01:48.00 | Deeewayne | eluizbr: are you only trying versions or have you checked out the latest 1.4 code from subversion? |
| 01:48.01 | teknoprep | wtf |
| 01:48.54 | eluizbr | Deeewayne: Already tested several versions |
| 01:49.15 | eluizbr | Deeewayne: subversion too |
| 01:49.19 | teknoprep | well now it doesn't say no line registered |
| 01:49.34 | teknoprep | just gives me nothing when i hit the speakerphone button or pickup the handset |
| 01:49.45 | Bananaskin | in asterisk -r |
| 01:50.08 | Bananaskin | do unload chan_sccp.so |
| 01:50.12 | teknoprep | i have to reboot my asterisk server to get that back up and running |
| 01:50.20 | teknoprep | give it a minute |
| 01:50.21 | Bananaskin | then load chan_sccp.so |
| 01:51.19 | teknoprep | i need to make a drink |
| 01:51.19 | teknoprep | brb |
| 01:52.22 | Deeewayne | eluizbr: Try setting /etc/asterisk/modules.conf autoload=no and comment out 'load' lines, then restart asterisk |
| 01:52.23 | *** join/#asterisk implicit_ (i=implicit@gateway/tor/x-8450a4b8b3408c92) |
| 01:52.47 | eluizbr | ok.. |
| 01:52.52 | Bananaskin | Deeewayne u can use noload as well |
| 01:53.50 | teknoprep | Bananaskin, |
| 01:53.59 | teknoprep | Bananaskin, ty for this help bro... it this works out... w0ot w0ot |
| 01:54.06 | teknoprep | Bananaskin, i am really enjoying this too.... |
| 01:54.25 | Bananaskin | has the phone registered an extn ? |
| 01:54.33 | teknoprep | well i am rebooting my server |
| 01:54.37 | eluizbr | Deeewayne: Same mistake |
| 01:54.38 | teknoprep | because of the one issue i have |
| 01:54.41 | Bananaskin | k |
| 01:55.13 | teknoprep | Bananaskin, btw |
| 01:55.20 | teknoprep | Bananaskin, i was saying ty lol |
| 01:55.32 | Bananaskin | k :) |
| 01:55.42 | eluizbr | Deeewayne: The asterisk this running. I so can not access the CLI |
| 01:56.59 | Deeewayne | I just updated my 1.4 branch to revision 94667 and do not see this problem |
| 01:57.08 | *** join/#asterisk PepOSX (n=pepOSX@201.248.215.16) |
| 01:57.16 | teknoprep | -- SCCP: Alarm Message: Severity: Warning (1), 6: Name=SEP000A8A93D529 Load=8.0(7.0)File Not Found [2050/1661602314] |
| 01:57.16 | teknoprep | <PROTECTED> |
| 01:57.31 | teknoprep | no lines registered |
| 01:58.01 | Bananaskin | email me your sccp.conf |
| 01:58.24 | teknoprep | one sec |
| 01:58.27 | teknoprep | i will pastbin it |
| 01:58.30 | teknoprep | so that you can edit it easy |
| 01:59.43 | teknoprep | http://pastebin.ca/829316 |
| 02:00.31 | Bananaskin | first mistake |
| 02:00.37 | Bananaskin | line 65 |
| 02:00.44 | Bananaskin | autologin 1001 not 135 |
| 02:01.41 | teknoprep | ok changed |
| 02:01.51 | Bananaskin | do a |
| 02:02.10 | Bananaskin | unload chan_sccp.so then load chan_sccp.so |
| 02:02.18 | Bananaskin | from the asterisk console |
| 02:04.04 | Unst4ble | Has anyone got a sucessful Finarea SA trunk working in asterisk? |
| 02:04.07 | teknoprep | ok now its working sorta |
| 02:04.10 | teknoprep | i get dialton |
| 02:04.13 | teknoprep | but i can't call out |
| 02:04.24 | teknoprep | i dial a number and it just hangs up after a second |
| 02:04.37 | Bananaskin | 2 secs |
| 02:05.09 | Bananaskin | does it show the line appearance for 1001 |
| 02:05.37 | teknoprep | inbound works tho |
| 02:05.43 | teknoprep | Ext 1001 |
| 02:05.43 | Bananaskin | might be your context |
| 02:05.43 | teknoprep | yes |
| 02:05.53 | Bananaskin | what is your normal context for the phones |
| 02:05.57 | teknoprep | from-internal |
| 02:06.14 | Bananaskin | sure ? |
| 02:06.29 | teknoprep | yes |
| 02:06.37 | teknoprep | thats also the default in freepbx |
| 02:07.41 | Bananaskin | 2 secs |
| 02:07.51 | teknoprep | http://pastebin.ca/829316 |
| 02:07.56 | teknoprep | sorry wrong paste |
| 02:08.09 | teknoprep | Dec 22 21:07:26 WARNING[3677] chan_iax2.c: Unable to create translator path for unknown to g729 on IAX2/VoicePulse4-12 |
| 02:08.24 | teknoprep | thats where it hangs up |
| 02:08.28 | teknoprep | let me take off g729 |
| 02:08.32 | teknoprep | i don't have g729 on this box |
| 02:08.52 | Bananaskin | sccp.conf |
| 02:08.56 | Bananaskin | allow=g729 |
| 02:09.01 | Bananaskin | line 11 |
| 02:09.13 | teknoprep | yup |
| 02:09.16 | teknoprep | taking it off now |
| 02:09.23 | teknoprep | its working |
| 02:09.23 | teknoprep | w0ot |
| 02:09.25 | teknoprep | omfg |
| 02:09.27 | teknoprep | i love you man |
| 02:09.38 | teknoprep | W000000000000))))))HHOOOoooooooooOOOTTT |
| 02:09.50 | Bananaskin | :) |
| 02:10.25 | windsor510 | hi guys, thanks for the help. got iax2 working, able to connect to my * server behind my DSL/firewall and check/record voicemail while sitting here at a cafe (most likely another nat'd IP). |
| 02:10.48 | Bananaskin | u using idefisk ? or zoipper or whatever its called |
| 02:10.55 | windsor510 | yep, zoiper |
| 02:11.04 | Bananaskin | :) |
| 02:11.22 | windsor510 | working like a charm, I just have to go tidy up my config files after I take a brain-break. =) |
| 02:12.33 | *** join/#asterisk Coder365_ (n=me@wrlsmdm025.cbpu.com) |
| 02:12.51 | Coder365_ | has anyone had any trouble with gizmo+grandcentral |
| 02:13.00 | Coder365_ | (+ asterisk) |
| 02:13.52 | Bananaskin | teknoprep do u consider the audio quality better than sip ? |
| 02:14.05 | *** part/#asterisk eluizbr (n=eluizbr@201.78.140.135) |
| 02:15.10 | windsor510 | << never used any hardware with * servers (yet). sorry |
| 02:15.34 | Coder365_ | windsor510: was that directed to me? |
| 02:15.42 | windsor510 | yea, sorry I didnt prefix it |
| 02:15.50 | Coder365_ | gizmo is a sip trunk |
| 02:16.07 | Coder365_ | and grandcentral is a forwarding thing that forwards to gizmo |
| 02:16.12 | windsor510 | isn't grandcentral a hardware unit? |
| 02:16.15 | Coder365_ | No |
| 02:16.19 | windsor510 | oh, my bad |
| 02:16.22 | Coder365_ | :) |
| 02:16.34 | teknoprep | dude sccp kicks ass |
| 02:16.41 | teknoprep | yes its fucking kick ass |
| 02:16.45 | windsor510 | I'm an asterisk noob anyways. =P |
| 02:16.51 | windsor510 | what is sccp? |
| 02:16.55 | teknoprep | Bananaskin, everything is f'n perfect |
| 02:16.59 | Bananaskin | glad you approve teknoprep |
| 02:17.03 | Coder365_ | anyway, I can forward it to my cell just fine, and it'll connect. But, when i forwrad to my gizmo it wont terminate the call |
| 02:17.14 | Coder365_ | it'll ring and gizmo will answer, but it'll keep ringing |
| 02:17.19 | windsor510 | 9250=1000,matt, matt@somecompany.com |
| 02:17.19 | windsor510 | 9251=1000,joel,joel@somecompany.com |
| 02:17.19 | windsor510 | 9252=1000,gerald,gerald@somecompany.com |
| 02:17.20 | windsor510 | 9250=1000,matt, matt@somecompany.com |
| 02:17.21 | windsor510 | 9251=1000,joel,joel@somecompany.com |
| 02:17.23 | windsor510 | 9252=1000,gerald,gerald@somecompany.com |
| 02:17.25 | windsor510 | wow.. |
| 02:17.28 | windsor510 | sorry |
| 02:17.39 | teknoprep | OMFG |
| 02:17.48 | teknoprep | + everything is kick arse bro on the skinny look |
| 02:17.53 | windsor510 | unix style right click paste buffer.. didnt mean to spam you guys. =) |
| 02:18.12 | teknoprep | hey Bananaskin how do i add a new device now? |
| 02:18.28 | Bananaskin | do u have more than 1 cisco unit ? |
| 02:18.39 | Coder365_ | does anyone happen to have a fwd account laying around they can help me test some stuff out with? |
| 02:18.50 | teknoprep | i have 30 at one office all on SIP |
| 02:18.55 | teknoprep | i am going to transfer them over soon |
| 02:19.01 | teknoprep | say 2 or 3 at a time |
| 02:19.02 | windsor510 | I have a FWD account |
| 02:19.13 | Coder365_ | okay, hold on |
| 02:19.24 | teknoprep | do i just create a context for each one? |
| 02:19.26 | windsor510 | lemme fire up my client |
| 02:19.31 | windsor510 | 9250=1000,matt, matt@somecompany.com |
| 02:19.31 | windsor510 | 9251=1000,joel,joel@somecompany.com |
| 02:19.31 | windsor510 | 9252=1000,gerald,gerald@somecompany.com |
| 02:19.35 | windsor510 | aaaah.. sorry |
| 02:19.42 | Coder365_ | rofl |
| 02:20.07 | Bananaskin | ok, well all you need to is copy down the device statement, change the details to reflect the phone ie the auto login extn etc, and the macaddy, then add all the relevant extns as well. The do the custom extns in freepbx thats it |
| 02:20.26 | teknoprep | ok i got the last part |
| 02:20.44 | Bananaskin | each device statement relates to 1 phone |
| 02:20.57 | teknoprep | ok |
| 02:21.00 | Bananaskin | so you need X device statements for X phones |
| 02:21.04 | teknoprep | does it have to be [devices] |
| 02:21.10 | teknoprep | or can it be whatever i want it to be |
| 02:21.33 | Bananaskin | no it's devices |
| 02:21.36 | teknoprep | ok |
| 02:21.40 | teknoprep | then i put the lines in |
| 02:21.45 | teknoprep | [lines] |
| 02:21.51 | Bananaskin | exactly |
| 02:22.03 | Bananaskin | no need to repeat the devices or lines statements |
| 02:22.03 | teknoprep | and i need X [lines] for each [devices] that points to a specific [lines] |
| 02:22.11 | Bananaskin | only 1 occurance in the cfg |
| 02:22.17 | teknoprep | ? |
| 02:22.21 | Bananaskin | ie |
| 02:22.27 | Coder365_ | windsor510: 883524 |
| 02:22.33 | Coder365_ | lemmie know if it connects |
| 02:22.59 | teknoprep | can you show me your sccp.conf |
| 02:23.02 | teknoprep | so i can learn |
| 02:23.10 | Bananaskin | 2 secs |
| 02:24.56 | windsor510 | coder365_: ok, calling in a sec |
| 02:25.11 | Coder365_ | k |
| 02:26.33 | windsor510 | I get an error. I am able to call myself however, and I believe I am on the network. |
| 02:26.50 | Coder365_ | okay |
| 02:26.53 | Coder365_ | its at my end |
| 02:26.56 | Coder365_ | thanks |
| 02:26.59 | windsor510 | I'm also relatively confident in my being on the network, because I was chatting with a friend earlier. |
| 02:27.05 | Coder365_ | yeah |
| 02:27.07 | Coder365_ | its me |
| 02:27.09 | Coder365_ | gonna t/s |
| 02:27.23 | teknoprep | Bananaskin, you there bro? |
| 02:27.29 | windsor510 | k. I have settings for 'Wengophone' if you want them. |
| 02:27.35 | teknoprep | Bananaskin, must be taking out the secret info |
| 02:27.47 | Bananaskin | nah, 2 secs, wife nattering to me |
| 02:28.40 | teknoprep | lol |
| 02:28.50 | teknoprep | i am changing the office over to this tommorow |
| 02:28.54 | teknoprep | this is simple as hel |
| 02:29.02 | Bananaskin | http://www.pastebin.ca/829344 |
| 02:29.23 | Bananaskin | u can even add Ip Communicator as well (Cisco Softphone) |
| 02:29.43 | windsor510 | as a iax client? |
| 02:29.54 | Bananaskin | no.... :) sccp |
| 02:30.25 | windsor510 | Ah. |
| 02:30.43 | teknoprep | w0ot |
| 02:30.50 | teknoprep | is Ip Communicator free / |
| 02:30.55 | Bananaskin | nope :) |
| 02:30.58 | teknoprep | didn't think so |
| 02:31.08 | Bananaskin | if uu have a smartnet account it is |
| 02:31.47 | teknoprep | <PROTECTED> |
| 02:31.54 | teknoprep | if you set the |
| 02:31.55 | teknoprep | type = |
| 02:32.02 | teknoprep | it doesn't start over until you get to the |
| 02:32.19 | teknoprep | device => |
| 02:32.20 | Bananaskin | Mac address |
| 02:32.23 | Bananaskin | yep |
| 02:32.26 | teknoprep | then just start over again |
| 02:32.31 | teknoprep | thats EASY AS F**K |
| 02:32.32 | teknoprep | lol |
| 02:32.44 | Bananaskin | essentially, copy and paste then change the details |
| 02:32.48 | teknoprep | yup |
| 02:32.56 | teknoprep | add a SEP<mac> |
| 02:33.01 | teknoprep | which you odn't even need to change |
| 02:33.07 | Bananaskin | there is a lot you can have in the general section |
| 02:33.30 | teknoprep | i like the skinny firmware interface much better roo |
| 02:33.31 | Unst4ble | How do i enable NAT on external peers? |
| 02:33.32 | teknoprep | too |
| 02:33.32 | Bananaskin | btw, look at line 293 ad 294 |
| 02:33.38 | teknoprep | Unst4ble, nat=yes |
| 02:33.43 | Unst4ble | So when i do a sip show peers it says nat Y |
| 02:34.02 | Unst4ble | It's being overridden somewhere because when i do sip show peers i get nat N |
| 02:34.22 | windsor510 | I have a question (now I'm using iax, not sip), and I want to have one extension for people to call me, but stay registered in two locations, do I have to worry about being booted fom one soft-phone when connecting on another? |
| 02:34.29 | teknoprep | Bananaskin, ahh for call pickup |
| 02:34.36 | Bananaskin | teknoprep *8 you can assign call groups and pickup groups |
| 02:34.39 | teknoprep | Bananaskin, so a manager can do a call pickup if things get bussy |
| 02:34.54 | Bananaskin | yep |
| 02:35.01 | teknoprep | Bananaskin, very nice |
| 02:35.03 | Bananaskin | or diff depts |
| 02:35.17 | teknoprep | yeah thats not the groups i am targeting right now |
| 02:35.23 | Bananaskin | ie sales are call and pickupgroup 1 - complaints are 2 etc etc |
| 02:35.33 | teknoprep | so how does sccp perform over nat and over the inet |
| 02:35.34 | windsor510 | Unst4ble: you can do nat=yes in sip.conf in the [general] section, but maybe you need it in each entry for the sip connections. |
| 02:35.35 | teknoprep | is it reliable |
| 02:36.04 | Bananaskin | reliable as any other ip |
| 02:36.32 | teknoprep | if i have a remote ext |
| 02:36.35 | teknoprep | is it easy to configure |
| 02:36.39 | teknoprep | does it only use port 2000 |
| 02:37.16 | Bananaskin | no, it uses a few more, 2 secs |
| 02:40.17 | teknoprep | Bananaskin, do you have the 1.4 chan_sccp.so |
| 02:40.22 | teknoprep | or is that the one you sent me in the email |
| 02:40.33 | Bananaskin | I emailed it yes |
| 02:40.36 | teknoprep | ok |
| 02:40.44 | teknoprep | becuase at home i still use asterisk 1.2 |
| 02:40.53 | Bananaskin | I can send you 1.2 |
| 02:40.53 | teknoprep | but at the office i contract for i am using 1.4 |
| 02:40.55 | teknoprep | i have 1.2 |
| 02:41.16 | Bananaskin | is that what didn't work earlier the 1.4 on the 1.2 box ? |
| 02:41.31 | Bananaskin | cos you said that you were using 1.4 ? |
| 02:41.36 | teknoprep | yeah thats fine |
| 02:41.44 | teknoprep | i can use that chan_sccp.so at other places |
| 02:41.48 | teknoprep | i needed it |
| 02:42.15 | Bananaskin | k, it should work on 1.4s same as the 1.2 does |
| 02:43.31 | *** join/#asterisk Winkie (n=urmom@general-ld-220.t-mobile.co.uk) |
| 02:43.59 | Bananaskin | still trying to locate ithe info on the ports that sccp uses, it is tcp/2000 for the sccp there are other ports though (or at least I think) |
| 02:45.09 | Bananaskin | teknoprep - you may get some info here - http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123newft/123_1/ftskinny.htm |
| 02:47.12 | *** join/#asterisk RoyK (n=roy@ti211210a080-0234.bb.online.no) |
| 02:47.29 | *** join/#asterisk RoyK (n=roy@ti211210a080-0234.bb.online.no) |
| 02:52.05 | Coder365_ | im having trouble getting fwd to register on my asterisk machine |
| 02:52.55 | teknoprep | hey Bananaskin how do i get button 2 to have an extension on it? |
| 03:01.53 | Bananaskin | teknoprep on the device for that phone, you will see auto login line it has 1001 atm, add the other exnt you want |
| 03:02.07 | Bananaskin | look at my 7970 for the example |
| 03:02.51 | Bananaskin | line 66 |
| 03:08.30 | teknoprep | can i have the same extension 2x's ? |
| 03:08.39 | teknoprep | or would that be stupid with skinny |
| 03:09.43 | Bananaskin | erm, good question, don't know what would happen with inbounds, you can degine how many inbound calls you can handle on 1 ext though |
| 03:10.24 | teknoprep | yeah i saw that |
| 03:10.27 | teknoprep | thats very nice |
| 03:10.45 | teknoprep | instead of just 2 calls being answered in SIP |
| 03:10.51 | teknoprep | you can have as many as you want i guess |
| 03:11.03 | *** part/#asterisk Unst4ble (n=Unstable@87-194-206-186.bethere.co.uk) |
| 03:11.11 | Bananaskin | u can also have line appearances for example if there is a shared exnt the line appearance shows busy |
| 03:11.33 | teknoprep | hmm |
| 03:11.37 | teknoprep | you can share extensions? |
| 03:11.42 | teknoprep | thats crazy cool |
| 03:11.57 | teknoprep | so every phone could have ext 1000 voicemail extension |
| 03:12.16 | Bananaskin | 2 secs |
| 03:12.18 | teknoprep | say its an extension setup for global voicemail in a small office |
| 03:12.29 | Bananaskin | confirming my comment |
| 03:13.27 | teknoprep | i hope mwi works |
| 03:13.30 | teknoprep | testing it now |
| 03:13.31 | Bananaskin | does |
| 03:13.36 | teknoprep | hasn't lit yet tho |
| 03:13.44 | Bananaskin | takes 30 - 60 secs |
| 03:13.48 | teknoprep | hmm |
| 03:13.49 | teknoprep | thats ghey |
| 03:14.15 | Bananaskin | it so that the network ain't bombarded with broadcast traffic |
| 03:15.09 | teknoprep | hmm |
| 03:15.12 | teknoprep | still isn't lit |
| 03:15.19 | teknoprep | but it shows that i have a voicemail |
| 03:15.25 | teknoprep | next to the button |
| 03:15.28 | Bananaskin | on the screen ? |
| 03:15.30 | teknoprep | but the light isn't lit |
| 03:15.34 | teknoprep | yeah on the screen but no light |
| 03:15.55 | Bananaskin | 2 secs |
| 03:16.49 | piper69 | Bananaskin: you here |
| 03:16.55 | piper69 | Does anyone here know how to configure a Cisco ATA-186 for use with freeworlddialup? My adapter currently has firmware version 3.1.0. If this needs to be upgraded, does anyone have the upgrade file available? I am behind a NAT router. |
| 03:17.12 | Bananaskin | nope, this is a recorded message, please leave your questions after the .... |
| 03:17.15 | teknoprep | why do have this dire need to use such an old ATA |
| 03:17.51 | Bananaskin | I have 2 of em :) |
| 03:17.51 | teknoprep | heh |
| 03:17.51 | piper69 | :) |
| 03:17.51 | teknoprep | well i still don't have MWI light |
| 03:17.51 | Bananaskin | 2 secs |
| 03:17.51 | teknoprep | but it shows up on the damn screen |
| 03:17.52 | teknoprep | that sucks |
| 03:18.55 | teknoprep | i like how it keeps an open session with the server |
| 03:19.07 | teknoprep | every button press is pretty much recorded |
| 03:19.13 | teknoprep | even if you open up with speakerphone |
| 03:19.21 | Bananaskin | change the config to reflect - device - mwilamp = on |
| 03:19.29 | teknoprep | already did that |
| 03:20.29 | teknoprep | yay it works |
| 03:20.43 | teknoprep | i did it in the general configuartion area |
| 03:20.47 | Bananaskin | k |
| 03:21.48 | teknoprep | can't sccp be transcoded ? |
| 03:21.56 | Bananaskin | to what ? |
| 03:22.07 | teknoprep | from g729 sscp <-> iax2 gsm |
| 03:22.15 | teknoprep | my provider is iax2 gsm |
| 03:22.26 | Bananaskin | yep but your PBX transcodes |
| 03:22.26 | teknoprep | but when i enable g729 it doesn't work anymore |
| 03:22.30 | teknoprep | oh wait |
| 03:22.36 | teknoprep | i don't have g729 on my damn pbx lol |
| 03:22.41 | teknoprep | well not this one |
| 03:22.46 | Bananaskin | nothing to do with sccp |
| 03:22.50 | teknoprep | yeah |
| 03:22.52 | teknoprep | ok np |
| 03:23.21 | Bananaskin | teknoprep where you from ? |
| 03:23.42 | teknoprep | i live near philly now |
| 03:23.48 | Bananaskin | k |
| 03:24.32 | teknoprep | what about you? |
| 03:24.35 | Bananaskin | Ireland |
| 03:24.39 | teknoprep | ahh |
| 03:25.00 | teknoprep | that would be Philadelphia, Pennsylvania in the US |
| 03:25.01 | teknoprep | lol |
| 03:25.42 | Bananaskin | figured that :) |
| 03:25.45 | *** join/#asterisk Unst4ble (n=Unstable@87-194-206-186.bethere.co.uk) |
| 03:26.21 | Unst4ble | Hi all. Can asterisk spoof itself to make it look like an SPA-3102 to the SIp providor? |
| 03:26.55 | Unst4ble | I have a SIP account that i can connect to and make outgoing calls from my SPA 3102. But can't get a trunk configured for it in asterisk. |
| 03:36.55 | teknoprep | Bananaskin, i am very happy bro |
| 03:37.02 | teknoprep | Bananaskin, i am extremely happy man lol |
| 03:37.14 | teknoprep | Bananaskin, sccp works much better on these phones then sip will ever |
| 03:38.17 | *** part/#asterisk timeshell (n=Khoja@206.248.136.108) |
| 03:38.41 | piper69 | Bananaskin: do you have any guide on config ATA 186 SIP |
| 03:44.07 | teknoprep | Bananaskin, how do i reassign the buttons |
| 03:44.16 | teknoprep | Bananaskin, i want to put the buttons i want in the order i want |
| 03:44.40 | Bananaskin | teknoprep just change the order |
| 03:45.05 | Bananaskin | ie if it is autologin 1001, 1000 change to 1000,1001 |
| 03:45.13 | teknoprep | nono the options on the screen |
| 03:45.24 | Bananaskin | what options |
| 03:45.44 | teknoprep | the dynamic buttons on the screen |
| 03:45.48 | teknoprep | like end call |
| 03:45.48 | teknoprep | hold |
| 03:45.49 | teknoprep | park |
| 03:45.51 | teknoprep | that crap |
| 03:45.56 | Bananaskin | nope, u canny do that |
| 03:46.18 | teknoprep | that sucks |
| 03:46.20 | teknoprep | park is on page 2 |
| 03:46.32 | Bananaskin | thats all part of the firmware |
| 03:46.45 | Bananaskin | whats wrong with ##70 ? |
| 03:46.51 | teknoprep | thats what i usually do |
| 03:47.04 | teknoprep | actually i changed it to ##9 |
| 03:47.50 | teknoprep | how do i do a conference call |
| 03:48.32 | Bananaskin | use asterisk for conf calls |
| 03:48.38 | teknoprep | ? |
| 03:48.40 | Bananaskin | the conf facility |
| 03:48.46 | teknoprep | i used to conf call with the phone |
| 03:48.48 | teknoprep | it was much easier |
| 03:49.33 | Bananaskin | how did u manage a conf call with the 7940 ? |
| 03:49.57 | teknoprep | with SIP |
| 03:50.00 | teknoprep | there was a CONF button |
| 03:50.07 | teknoprep | you made another call after hitting that |
| 03:50.10 | teknoprep | and then hit join |
| 03:50.16 | teknoprep | it joined the calls together |
| 03:51.37 | Unst4ble | What would make a SIP connection work fine from an SPA 3102, but UNREACHABLE from asterisk trunks? |
| 03:51.55 | Bananaskin | well tbh, never used it, alwyas tfr callers to an actual conference |
| 03:52.04 | teknoprep | ? |
| 03:52.10 | teknoprep | you transfer calls to a conference |
| 03:52.14 | teknoprep | then join the conference? |
| 03:52.23 | Bananaskin | you can tfr calls to conf yes |
| 03:52.27 | Bananaskin | yep |
| 03:52.33 | Bananaskin | ##conf number |
| 03:52.33 | teknoprep | hmm |
| 03:52.41 | Bananaskin | blind transfer |
| 03:52.52 | teknoprep | yeah i understand what ## does lol |
| 03:53.12 | teknoprep | the phone is showing the wrong time by 1 hour |
| 03:53.15 | teknoprep | this will be my last question |
| 03:53.25 | teknoprep | i am using time.mit.edu to set the time |
| 03:53.33 | Bananaskin | its in the sepmacaddy.cnf.xml |
| 03:53.36 | Bananaskin | look for -1 |
| 03:53.44 | Bananaskin | or +1 |
| 03:53.49 | jql | the time is correct; the zone, though... |
| 03:55.55 | teknoprep | <timeZone>EST Standard/Daylight Time</timeZone> |
| 03:56.22 | *** part/#asterisk Coder365_ (n=me@wrlsmdm025.cbpu.com) |
| 04:04.40 | teknoprep | well this was a kick ass evening |
| 04:04.44 | teknoprep | thanx Bananaskin |
| 04:04.46 | teknoprep | again |
| 04:05.52 | piper69 | Bananaskin: check pm |
| 04:08.30 | *** join/#asterisk tobias (n=tobias@nat1.ppckernel.org) |
| 04:20.42 | *** join/#asterisk craigk (n=ckowald@58.174.150.119) |
| 04:37.39 | *** join/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum) |
| 04:39.27 | Cherebrum | I figured out that the Polycom IP500, IP501, IP600, IP601 all use the same chipset as the IP650 and IP550... If you add G722 codec support to the non HD Polycom phones then you can do HD audio! :) 16k sample rate with G722 and the same bandwidth as G711 at 8k! I've tested it with FreeSWITCH and it works great! :) |
| 04:39.55 | Cherebrum | you just have to add G722 in the config file for the IP50x/60x phones |
| 04:40.04 | Cherebrum | Enjoy! |
| 04:40.06 | *** part/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum) |
| 04:42.00 | WilliamK | wow only a few of us been here for hours.... |
| 04:42.29 | Unst4ble | Is there any way to force NAT on a trunk? nat-yes doesnt seem to be doing it |
| 04:42.50 | jql | you mean you want to set the externalip? |
| 04:43.02 | jql | or externip or whatever that damn var is called |
| 04:43.06 | Unst4ble | I have sip_nat.conf set up |
| 04:43.18 | WilliamK | :) |
| 04:43.18 | Unst4ble | but when i do a sip show peers it says nat isnt on |
| 04:44.06 | Unst4ble | I'm sniffing a call and it seems very one way. |
| 04:44.20 | Unst4ble | REGISTER sip:sip.voipstuntREGISTER sip:sip.voipstuntREGISTER sip:sip.voipstuntREGISTER sip:sip.voipstuntREGISTER sip:sip.voipstuntREGISTER sip:sip.voipstuntREGISTER sip:sip.voipstuntREGISTER sip:sip.voipstuntREGISTER sip:sip.voipstuntREGISTER sip:sip.voipstuntREGISTER sip:sip.voipstuntREGISTER sip:sip.voipstuntREGISTER sip:sip.voipstuntREGISTER sip:sip.voipstuntREGISTER sip:sip.voipstuntREGISTER sip:sip.voipstuntREGISTER sip:sip.voipstun |
| 04:44.29 | Unst4ble | no response from server |
| 04:44.37 | Qwell | err, what? |
| 04:44.44 | jql | what does the server see? |
| 04:44.51 | Qwell | how is he planning on "adding g722 codec support" to a 601? |
| 04:44.55 | Unst4ble | I don't know, it's external |
| 04:45.16 | Bananaskin | Unst4ble - its best not to paste a load of text to a irc channel, best to use pastebin.ca |
| 04:45.45 | Unst4ble | Sorry, didnt mean to copy that much. Only wanted 2 or so packets. |
| 04:45.58 | _x86_ | Qwell: does polycom offer an "hd voice" firmware option for the 601's? |
| 04:46.10 | Qwell | uhh...considering the phone doesn't support it - no |
| 04:46.12 | _x86_ | Qwell: or only the 550/650 will run "hd voice" eh? |
| 04:46.20 | _x86_ | Qwell: codec is in software, no? |
| 04:46.30 | Qwell | and there were hardware changes to support it |
| 04:46.45 | Qwell | the speakers and mics on the 650s are *FAR* superior |
| 04:46.54 | _x86_ | ah cool |
| 04:47.03 | Qwell | they sound *amazing*, even with g711, heh |
| 04:47.04 | _x86_ | so even if you use g711 it sounds better? |
| 04:47.09 | _x86_ | ah nifty |
| 04:47.11 | Qwell | yep, very much so |
| 04:47.16 | *** join/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum) |
| 04:47.19 | Cherebrum | https://ares.jasongarland.com/~jgarland/polycom/hdaudio.cfg |
| 04:47.22 | Qwell | I mean, Polycoms sound great normally, but.. |
| 04:47.24 | Cherebrum | there is a sample config |
| 04:47.26 | Cherebrum | :) |
| 04:47.34 | jql | and they added the "radio announcer" effect to the speakerphone. it's weird. even g711 calls sound "rich" on the 550 |
| 04:47.34 | _x86_ | too bad they dont offer "hd voice" on any other models |
| 04:47.37 | jql | hard to explain |
| 04:47.46 | Qwell | Cherebrum: that doesn't mean it's "supported" |
| 04:47.51 | Cherebrum | no.. but it works |
| 04:47.56 | _x86_ | 550/650 are the only "hd voice" capable phones eh? |
| 04:48.01 | Qwell | Cherebrum: I doubt it |
| 04:48.04 | Cherebrum | I tested it |
| 04:48.06 | Cherebrum | works great |
| 04:48.17 | Cherebrum | You can definately tell a difference |
| 04:48.18 | _x86_ | Cherebrum: what? g722 on a 501/601? |
| 04:48.19 | SwK | Qwell, enabling the built in echo canceller on the handset isnt supported but it does work heh |
| 04:48.27 | Cherebrum | call sip:moh@jasongarland.com to test it |
| 04:48.38 | MooingLemur | polycom 500/501 speakerphones have low-bitrate mp3-like distortion with g711 |
| 04:48.42 | MooingLemur | it's really bad |
| 04:48.46 | Cherebrum | SwK: only if you have 2.1.2 or newer I beleive |
| 04:48.52 | _x86_ | MooingLemur: yeah i know :( |
| 04:48.59 | Cherebrum | the HD only works with the handset |
| 04:49.03 | _x86_ | MooingLemur: my 601's don't sound a lot better than my 501's |
| 04:49.14 | Cherebrum | doesn't work with headset or speakerphone.. even on the 650 and 550 |
| 04:49.16 | _x86_ | but both the 501s and 601s i have sound better than my 301s |
| 04:49.28 | SwK | Cherebrum, what the echo can? |
| 04:49.30 | Cherebrum | the 501 and 601 have a difference chipset.. |
| 04:49.34 | Cherebrum | SwK: yes |
| 04:49.43 | SwK | Cherebrum, that works on everything back to like 1.3.x |
| 04:49.44 | Cherebrum | different then the 301 I mean.. |
| 04:49.45 | _x86_ | Cherebrum: echo cancellation doesn't work on speakerphone? |
| 04:49.50 | SwK | its just disabled by default int he configs |
| 04:50.08 | SwK | its great for masking that pesky echo on TDM400s sometimes |
| 04:50.51 | jql | I wish I could force every phone/telco to install echo cancellers. grr |
| 04:50.53 | Cherebrum | If you enable syslog on the phone... https://ares.jasongarland.com/~jgarland/polycom/syslog.cfg |
| 04:50.58 | Cherebrum | you can see what chipset the phone is using.. |
| 04:51.10 | Cherebrum | and then you go look it up on ti.com and you can see what codecs are supported by that chip |
| 04:51.20 | Cherebrum | the 501, 601, 550, and 650 all use the same TI chip |
| 04:51.27 | Cherebrum | that TI chip supports G.722 |
| 04:51.40 | Qwell | hmm |
| 04:52.12 | Cherebrum | By the way... FreeSWITCH supports G.722 at 16k and you can do a 16k conference. ;) |
| 04:52.23 | Cherebrum | it's sounds f'in awesome |
| 04:52.32 | Cherebrum | er it |
| 04:52.41 | *** join/#asterisk coppice (n=chatzill@235.202.17.210.dyn.pacific.net.hk) |
| 04:53.13 | Cherebrum | and you don't a damn zap* module for it to work... I'm running mine on a hosted Xen server and it sounds wonderful |
| 04:54.27 | Cherebrum | Speex will also do 16k |
| 04:54.46 | Cherebrum | If you use Google Talk with dingaling you can do 16k Speex |
| 04:54.53 | WilliamK | is that some sort of concrete spackle talk? :) |
| 04:55.08 | Cherebrum | ? |
| 04:55.17 | WilliamK | speex = spackle |
| 04:55.27 | WilliamK | kinda like peter piper picked pickled peppers |
| 04:55.29 | Cherebrum | speex is a codec |
| 04:55.29 | WilliamK | :) |
| 04:55.49 | WilliamK | I know what speex is... just insanely bored |
| 04:55.50 | WilliamK | :) |
| 04:56.14 | coppice | oh course you're bored. it christmas |
| 04:56.16 | Cherebrum | hmm... ultra-wideband speex looks like fun... 32k |
| 04:56.40 | Cherebrum | I should setup a 32k conference bridge |
| 04:57.03 | WilliamK | least I don't have relatives coming over, etc... |
| 04:57.16 | WilliamK | and I'm off my day-to-day gig till Jan 2nd |
| 04:57.17 | coppice | ultra-wideband speex sucks - I have Jean Marc Valin's word on that :-) |
| 04:57.23 | Cherebrum | oh yea? |
| 04:57.29 | Cherebrum | Any other codecs that do 32k? |
| 04:57.45 | coppice | MP3, vorbis |
| 04:58.08 | WilliamK | hey coppice, do you know if the latest spanDSP works with SVN (asking before I harzardly find out) |
| 04:58.13 | Qwell | why stop at 32k? |
| 04:58.17 | Qwell | go to 48k |
| 04:58.19 | Cherebrum | I don't have a softphone that does vorbis |
| 04:58.31 | coppice | when you get that wideband, speech oriented codecs tend to make less sense |
| 04:58.37 | Cherebrum | I don't have a softphone that does 48k either |
| 04:58.47 | WilliamK | I often wonder how I would sound if we used a full 45Mbps |
| 04:58.52 | Qwell | coppice: dumb question - how do mp3 or vorbis handle speech? |
| 04:59.01 | jql | at that point, go *stereo* |
| 04:59.06 | coppice | WilliamK: work with the SVN of what? |
| 04:59.08 | jql | yeah, stereo speech codecs |
| 04:59.17 | WilliamK | SVN of asterisk |
| 04:59.19 | WilliamK | latest ver |
| 05:00.19 | WilliamK | ooo and I'm eagerly thinking of getting another cell to wifi router...and trying voip over it |
| 05:00.23 | WilliamK | never done that before |
| 05:00.56 | coppice | MP3 and vorbis are generalised sound codecs. The remove what your ears can't recognise |
| 05:00.58 | coppice | Speech codecs are targeted to one kind of source sound. The further remove what your voice can't produce. |
| 05:01.00 | coppice | That's why they tend to sound awful for anything but a single voice. |
| 05:01.22 | coppice | So, MP3 and vorbis code speech well, but take a higher bit rate to do it |
| 05:01.36 | Qwell | makes sense |
| 05:01.54 | Qwell | is there a reason for that high bitrate with speech? |
| 05:02.16 | coppice | duh! if you remove less, you end up with more bits |
| 05:02.17 | Qwell | I guess your last sentence answered that |
| 05:02.21 | Cherebrum | <action application="set" data="absolute_codec_string=speex@32000h,G722@16000h,G722"/> <action application="conference" data="1000@32k"/> |
| 05:02.23 | Cherebrum | muhahahaha! |
| 05:03.32 | Cherebrum | sip:1032@jasongarland.com should work at 32k speex now |
| 05:04.00 | Cherebrum | now I just need to find a softphone that supports 32k speex |
| 05:07.30 | Cherebrum | hmmm... Twinkle for Linux does Ultra-Wideband Speex |
| 05:07.41 | *** part/#asterisk Unst4ble (n=Unstable@87-194-206-186.bethere.co.uk) |
| 05:15.30 | tzafrir_home | WilliamK, there's app_fax in addons of trunk |
| 05:15.44 | tzafrir_home | that is: in trunk of asterisk-addons |
| 05:15.54 | tzafrir_home | Didn't get to test it |
| 05:19.36 | *** part/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum) |
| 05:24.41 | *** join/#asterisk UserReg_CL (n=COB@200.120.24.117) |
| 05:25.12 | WilliamK | k |
| 05:25.12 | UserReg_CL | hi all !!! |
| 05:25.19 | UserReg_CL | (hola a todos) |
| 05:25.32 | WilliamK | I've just had a hard time trying to get a fax machine to work on inbound calls with the SPA-2002 |
| 05:25.35 | WilliamK | works great outbound |
| 05:25.59 | WilliamK | even a modem (postage machine) works fine over the SPA-2002 |
| 05:26.15 | UserReg_CL | where change time register in asterisk ? |
| 05:26.29 | WilliamK | had it working a while back, and then went to a newer ver of SVN branch 1.4, and it broke |
| 05:27.56 | WilliamK | UserReg, if I'm understanding you right and assuming you're using SIP.... sip.conf |
| 05:28.38 | WilliamK | However, beware - boredom for me leads to falling asleep |
| 05:28.41 | WilliamK | and I'm quite bored |
| 05:28.43 | WilliamK | :) |
| 05:28.55 | UserReg_CL | mmm |
| 05:28.57 | UserReg_CL | traslated... |
| 05:30.55 | WilliamK | UserReg, this will probably help you - http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf |
| 05:33.18 | UserReg_CL | thankn friend |
| 05:34.08 | tzafrir_home | WilliamK, modem? with the RTP traffic going back and forth to Asterisk? |
| 05:34.41 | tzafrir_home | or did you set it to noreinvite? (but in such a case you couldn't do fax detection and such) |
| 05:38.39 | WilliamK | tzafrir, E&M T1 on PSTN --- Asterisk -- SPA-2002 --- modem |
| 05:38.42 | WilliamK | and yes RTP |
| 05:39.11 | WilliamK | canreinvite=no has always been in the sip.conf file |
| 05:39.29 | WilliamK | can't remember off the top of my head what the SPA was set to |
| 05:40.00 | WilliamK | actually in sip.conf it's set to =yes now |
| 05:40.18 | WilliamK | don't remember when that got changed, probably when it was moved to the outside of the firewall |
| 05:42.11 | WilliamK | I had some rogue setting that was causing it to do a codec unknown 100 error, and then I read on serveral sites that T.38 isn't supported on the SPA-2002 |
| 05:42.21 | WilliamK | so hopefully monday the newer SPA-2102 will be here |
| 05:43.16 | UserReg_CL | mmm |
| 05:54.26 | *** join/#asterisk mmurdock (n=blah@c-24-10-190-87.hsd1.ut.comcast.net) |
| 05:54.35 | lucent | what causes a 404 when a trunk registers with us? |
| 05:54.45 | lucent | I mean, what are the possible causes |
| 05:54.54 | lucent | would an invalid user/pass do it |
| 05:55.07 | lucent | sorry... not "trunk" |
| 05:55.10 | lucent | peer |
| 06:04.35 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 06:07.15 | piper69 | lucent: are you related to lucent tech |
| 06:07.24 | piper69 | 5ess |
| 06:19.00 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 06:30.44 | *** join/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum) |
| 06:31.07 | *** part/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum) |
| 06:35.34 | *** join/#asterisk AJaymn (n=mypocket@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 06:35.59 | AJaymn | Anyone know of a wholesale provider in US less then .01 per min? |
| 06:36.06 | AJaymn | for outbound |
| 06:36.41 | *** join/#asterisk Maliuta (n=nikolai@ppp214-92.static.internode.on.net) |
| 06:39.56 | lucent | piper69: negatory |
| 06:42.24 | piper69 | lucent: what you mean |
| 06:42.41 | jql | AJaymn: if you find one, you lemme know. :) |
| 06:43.03 | AJaymn | jql: i found one but have to fill out FCC Form 499... |
| 06:43.25 | AJaymn | (havnt sat down to read the 50page form example sheets) lol |
| 06:43.43 | jql | ah, joy |
| 06:44.06 | AJaymn | VoicePulse advertises less then .01 BUT calls are rounded to the whole min |
| 06:44.28 | piper69 | lucent: i am a 5ESS DCS Engineer |
| 06:44.34 | _x86_ | twas the night before the night before christmas; not a creature was stirring, except the computer mouse |
| 06:44.47 | AJaymn | Im using Vitelity now that charges 1.34cents... But they do charge correctly.. they dont round it |
| 06:44.50 | jql | well, the government thinks it will only take on average 10 hours to fill out |
| 06:44.53 | jql | how nice |
| 06:45.10 | _x86_ | AJaymn: i know of a provider |
| 06:46.05 | lucent | piper69: 'lucent' is a nickname I picked for myself |
| 06:46.15 | lucent | :) |
| 06:46.32 | piper69 | <PROTECTED> |
| 06:46.35 | piper69 | lol |
| 06:47.09 | lucent | I've installed plasma televisions and such at the Lucent Tech establishment in Illinois, it is a very strange (beautiful?) building |
| 06:47.23 | lucent | coincidence only |
| 06:48.03 | piper69 | i hope you charged them over $250k |
| 06:48.06 | lucent | tonight (and the past 4-5 nights) I am struggling with asterisk to talk to a VegaStream 50 FXS |
| 06:48.09 | lucent | hah |
| 06:48.35 | piper69 | well they make bank every 1hr |
| 06:49.13 | lucent | I don't know specifics, I subcontract for my friend who works at the audio/visual firm |
| 06:49.23 | lucent | it must be many millions |
| 06:51.29 | lucent | there I am with a work cart and a $50,000usd christie roadster projector on top, wheeling it up to the door |
| 06:51.49 | lucent | you know, there's a speed hump in the road that blocks the access ramp |
| 06:52.15 | lucent | scared the willies out of me trying to navigate it with the projector and the cart |
| 06:56.43 | Tili | how do we call registered iax peers. Dial(IAX2/user) |
| 06:57.27 | *** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
| 06:57.27 | *** mode/#asterisk [+o mog] by ChanServ |
| 07:05.06 | jql | AJaymn: hmm... the answer was apparently icall |
| 07:16.31 | WilliamK | AJaymn, BroadWing if you have enough volume |
| 07:17.04 | WilliamK | aka now L3 |
| 07:19.11 | AJaymn | jql icall? what?! ha? lol |
| 07:19.34 | Tili | when using outgoing callback, how do we give max limit of call? |
| 07:19.56 | AJaymn | L3 would require a huge commitment |
| 07:20.17 | WilliamK | AJaymn, you might also look at Transcom (aggregator for alot of bell co's) |
| 07:20.29 | WilliamK | yeah most are going to require a commitment |
| 07:20.33 | WilliamK | pay or play |
| 07:20.52 | AJaymn | :P |
| 07:28.36 | *** join/#asterisk wglenncamp (i=wglennca@c-69-139-127-105.hsd1.ky.comcast.net) |
| 07:30.02 | wglenncamp | got an easy question for ya (hopefully).. I am trying to register a Polycom IP501 to an asterisk box. I am getting the error "device does not match ACL" |
| 07:30.05 | wglenncamp | any ideas? |
| 07:30.28 | jql | interesting error |
| 07:31.38 | wglenncamp | I have had a heck on a night with these phones.. :) I generally haven't had a problem with the Polycom's, but I purchased these used. And I updated the firmware on them tonight. That is when the horror started. |
| 07:32.09 | wglenncamp | Anyway, I got it worked out, and now this is what I got... Running bootrom 4 and SIP 2.2 |
| 07:32.51 | wglenncamp | Of course, I could be tired and the settings are hosed... But I can make outbound calls, just can't recieve them.. Wierd.. |
| 07:34.22 | lucent | wglenncamp: did the auth username change? |
| 07:35.19 | lucent | unrelated, I had a VegaStream 50 FXS that was saying the same thing to me, I had no clue what the username was |
| 07:35.23 | *** part/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 07:35.40 | lucent | now I sorted that out and at least it is registering |
| 07:35.44 | wglenncamp | nope, I verified them and they match. I'll check again though.. Like I said, I'm pretty tired. I may sleep on it if I don;t get it soon. |
| 07:36.00 | lucent | might be a good idea (to sleep on it) |
| 07:36.14 | lucent | makes sense though, there's two registrations - one inbound and one outbound |
| 07:36.30 | lucent | stuff i never understood about SIP 4 days ago |
| 07:36.33 | lucent | ;) |
| 07:37.28 | lucent | say if I do host=a.b.c.d does this tell asterisk to go and register with that IP? |
| 07:37.38 | lucent | I have host=dynamic at the moment |
| 07:37.43 | lucent | it's working okay |
| 07:38.07 | lucent | I want to accept registrations from a host |
| 07:38.29 | lucent | my question is, does host=dynamic correlate to the appropriate way to do this? |
| 07:46.52 | *** join/#asterisk c4t3l (n=c4t3l@c-98-200-2-74.hsd1.tx.comcast.net) |
| 07:47.04 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 07:47.40 | c4t3l | greetings all... are there any agi gurus in the house tonight? |
| 07:49.43 | c4t3l | i'm writing an agi script that will allow a caller to announce they're name and then call my cell and play their recorded name back to me and give the option of accepting or rejecting call |
| 07:50.11 | c4t3l | the prob is that I'm not sure how to actually connect the call once I accept it |
| 07:52.22 | *** join/#asterisk markgreene (n=Mark_Gre@71-12-185-155.dhcp.leds.al.charter.com) |
| 07:52.39 | markgreene | Does anyone here have fonebridge from red-fone? |
| 07:52.52 | c4t3l | sorry, not I |
| 07:53.24 | markgreene | c4t3l: thanks for letting me know that you do not have it.. lol |
| 07:53.31 | c4t3l | :) |
| 07:54.03 | markgreene | I am a bit frustrated b/c their website is down and i need their version of zaptel |
| 07:54.37 | c4t3l | which version number, maybe I have it in my repo, or is it patched by them? |
| 07:55.35 | markgreene | No it's not plain old zaptel. I could download that from a number of sites. It's a version they wrote to include a better version of TDMoE |
| 07:56.22 | c4t3l | hmm, those jerks! |
| 07:56.40 | markgreene | Yeah well it is pretty great actually. But the fact that their site is down is not |
| 07:56.48 | markgreene | It's been down for over 10 hours |
| 07:57.14 | _x86_ | no, they are jerks for not committing the patch to the main zaptel branch ;) |
| 07:57.29 | markgreene | Actually I don't know why they didn't |
| 07:57.41 | tzafrir | where can I find their code? |
| 07:58.24 | markgreene | LOL - On their site, that's offline! It went down about 10 hours ago when I was half way through installing one of their products |
| 07:58.26 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 07:58.36 | c4t3l | yikes |
| 07:59.07 | markgreene | I am really excited about getting it working. It's going to expand my options a ton |
| 07:59.21 | _x86_ | what's wrong with standard TDMoE? |
| 08:00.00 | tzafrir | markgreene, how does it compare to a dedicated PC serving TDMoE? |
| 08:00.07 | tzafrir | Is that possible with Zaptel? |
| 08:00.27 | markgreene | Well the MAIN reason I can't use it is because this fone-bridge won't work with it because it's only programemd to work with thier version. And their version uses some kind of multi channel... something |
| 08:01.09 | markgreene | tzafrir: I dont' know how it compares. I've never uesed one. This is my first |
| 08:01.17 | _x86_ | regular TDMoE can certainly handle multiple channels lol |
| 08:01.55 | _x86_ | what is red-fone? |
| 08:02.21 | tzafrir | A dedicated TDMoE device |
| 08:02.31 | markgreene | http://www.voip-info.org/wiki/view/Redfone |
| 08:02.34 | _x86_ | nifty |
| 08:02.38 | markgreene | It's the name of the company that makes it |
| 08:02.43 | markgreene | But yes the device is pretty neat |
| 08:02.53 | markgreene | Or, in theory it is. I have not set it up yet |
| 08:05.17 | markgreene | Alright guys I'm out then |
| 08:05.20 | markgreene | later |
| 08:05.23 | *** join/#asterisk sergey (n=sergey@91.189.233.71) |
| 08:05.36 | *** part/#asterisk markgreene (n=Mark_Gre@71-12-185-155.dhcp.leds.al.charter.com) |
| 08:08.40 | WilliamK | #1 way of telling if someone is bored or tired of being spammed.... spending hours training SpamAssin's filter |
| 08:09.05 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 08:09.41 | jql | I gave up on handling email so long ago... I surrendered my communication rights to Yahoo long ago |
| 08:09.46 | _x86_ | #1 way of telling someone is bored: they've started reading up on learning Cobol |
| 08:10.06 | _x86_ | jql: gmail is much better |
| 08:10.29 | jql | I have that too, for "official" correspondence |
| 08:10.41 | jql | but my primary has been yahoo for way longer |
| 08:11.28 | _x86_ | i used to use yahoo but i've long since switched ;) |
| 08:14.35 | WilliamK | x86, funny you mention that... I used to do night ops stuff for a company using Cobol |
| 08:14.59 | _x86_ | i'd never put myself in that situation |
| 08:16.22 | _x86_ | lol |
| 08:18.56 | WilliamK | I only did it for 89 days |
| 08:19.07 | _x86_ | contract? |
| 08:19.17 | WilliamK | day before my "eval term" came up, I told them I was quitting |
| 08:19.32 | WilliamK | couldn't stand my dad's friend (higher level mgr) nor the supervisor under him |
| 08:19.51 | WilliamK | was working at an insurance company doing their nightly tasks |
| 08:20.09 | WilliamK | lemme tell you, it sucked |
| 08:20.17 | WilliamK | more less the politics |
| 08:22.35 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 08:29.47 | _x86_ | heh |
| 08:29.51 | _x86_ | politics-- |
| 08:30.34 | tzafrir | spamassisin works great here |
| 08:30.42 | tzafrir | And didn't need much tuning |
| 08:34.59 | *** join/#asterisk alrs (i=non-knav@pozug.com) |
| 08:37.14 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 08:37.24 | fujin | hey uh |
| 08:37.29 | fujin | howcome I get colour with -c |
| 08:37.31 | fujin | but not without? |
| 08:37.43 | fujin | I'd like colour, but for it to be daemonized so that if I reattach later it has colour |
| 08:38.27 | lucent | use with 'screen' may support such a thing |
| 08:38.38 | fujin | uh |
| 08:38.40 | fujin | next idea |
| 08:38.52 | fujin | I've seen it started from init.d as a daemon with colour before |
| 08:39.26 | lucent | yeah I don't really think my idea was ideal |
| 08:40.04 | fujin | nm, I'll play with it |
| 08:40.47 | WilliamK | x86, he didn't especially like it when I said I was going to work for another ISP right after he finished my training, etc... but I didn't care for someone who was B.S.'ing his way through things either |
| 08:44.50 | fujin | is there a particular command line flag which gives colour? |
| 08:45.03 | fujin | asterisk -c gives me colour, but nothing 'asterisk' just forks without colour |
| 08:45.37 | fujin | oh, I suppose i can -cF |
| 08:45.49 | fujin | egh that doesn't work. |
| 08:47.21 | WilliamK | so where does everyone do most of their traffic termination trading nowdays (websites) |
| 09:00.22 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 09:00.32 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 09:05.30 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 09:38.39 | tzafrir | asterisk -n disables color |
| 09:39.10 | tzafrir | color is currently disabled by defualt when in remote mode. Which is kind of strange |
| 09:39.30 | tzafrir | asterisk -c forces asterisk not to detach |
| 09:39.43 | tzafrir | A very lame way to get color. |
| 09:40.36 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 09:54.19 | *** join/#asterisk vetetix (n=vetetix@83.222.34.219) |
| 10:05.31 | *** join/#asterisk qdk (n=qdk@195.242.194.41) |
| 10:20.48 | *** join/#asterisk droemel (n=droemel@p548EAFB2.dip0.t-ipconnect.de) |
| 10:23.53 | Juggie | i've seen color using a remote console before |
| 10:23.59 | Juggie | i think |
| 10:24.11 | Juggie | usually on the local console of the box. |
| 10:25.11 | mvanbaak | remote console as in: asterisk -r |
| 10:25.12 | mvanbaak | ??? |
| 10:25.43 | mvanbaak | that has color here |
| 10:27.31 | Juggie | yeah, works for me |
| 10:29.39 | *** join/#asterisk saftsack (n=saftsack@p4FC7468F.dip.t-dialin.net) |
| 10:33.02 | mvanbaak | same here |
| 10:34.10 | DarKnesS_WolF | good evening |
| 10:35.16 | *** join/#asterisk Dovid (n=Dovid@bzq-79-178-61-156.red.bezeqint.net) |
| 10:35.43 | Dovid | any one here from Australia ? |
| 10:46.55 | *** join/#asterisk rvhi (n=chatzill@66.175.65.82) |
| 10:47.23 | rvhi | anyone knows how to convert a wav file to mp3 with sox? |
| 10:49.41 | tzafrir | mvanbaak, is that trunk? |
| 10:49.54 | tzafrir | rvhi, sox file.wav file.mp3 |
| 10:50.39 | tzafrir | You'll only lose quality in the process, though... |
| 10:51.19 | rvhi | sox: Failed writing msg0045.mp3: Sorry, no MP3 encoding support |
| 10:52.45 | mvanbaak | tzafrir: yes, trunk |
| 10:52.58 | mvanbaak | Asterisk SVN-trunk-r92206 built by root @ asterisk.vanbaak.info on a i686 running Linux on 2007-12-10 18:59:05 UTC |
| 10:53.02 | tzafrir | I think that issue was fixed there |
| 11:00.30 | tzafrir | rvhi, so don't use sox. Use lame or whatever |
| 11:01.36 | mvanbaak | or ditch mp3 and go for some more open format like ogg |
| 11:09.08 | *** join/#asterisk saftsack (n=saftsack@p4FC7468F.dip.t-dialin.net) |
| 11:09.53 | tzafrir | mvanbaak, hmm, did you see that commit yesterday to chan_sip in 1.2? It's not in 1.2.26.2 , right? |
| 11:10.08 | tzafrir | .1? .2? |
| 11:12.08 | mvanbaak | 1.2.26.2 is not there yet |
| 11:12.23 | mvanbaak | I think they should release it |
| 11:12.30 | mvanbaak | same as 1.4.something |
| 11:21.08 | Dovid | any one here from Australia ? |
| 11:21.43 | Dovid | I need a toll Free Number there checked as well as some one in Argentina |
| 11:22.06 | *** join/#asterisk PiterPunk (n=piterpk@cardinal.lizella.net) |
| 11:22.33 | *** part/#asterisk PiterPunk (n=piterpk@cardinal.lizella.net) |
| 12:22.11 | *** join/#asterisk Winkie (n=urmom@general-ld-220.t-mobile.co.uk) |
| 12:30.12 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 12:33.29 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 12:36.00 | squigly | Dovid hello? |
| 12:36.11 | squigly | oh he quit |
| 12:58.39 | *** join/#asterisk suvir (n=suvir@ppp-124.120.128.44.revip2.asianet.co.th) |
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| 13:39.37 | ronr_ | anyone knows what causes the error led LOS on the athrea ac 2032/t? |
| 13:44.42 | ronr_ | it's loss of signal, so I think my cable is broken |
| 13:46.56 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 13:46.57 | *** join/#asterisk PepOSX (n=pepOSX@201.248.215.16) |
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| 14:52.41 | *** mode/#asterisk [+o blitzrage] by ChanServ |
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| 15:18.29 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
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| 15:59.04 | piper69 | Good morning all |
| 16:11.14 | *** join/#asterisk teknoprep (n=tekno@unaffiliated/teknoprep) |
| 16:14.04 | *** join/#asterisk Qapf (n=Qapf@mail.oldworlddoor.com) |
| 16:16.17 | tzafrir | Good evening |
| 16:16.31 | piper69 | tzafrir: its morning here |
| 16:17.02 | tzafrir | piper69, it's evening here |
| 16:17.23 | piper69 | lol |
| 16:17.25 | tzafrir | And I bet you think that this is not a normal weekday |
| 16:17.28 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 16:17.31 | piper69 | where is here |
| 16:17.54 | tzafrir | piper69, that's what you have whois for |
| 16:18.28 | *** join/#asterisk Faustov (n=faustov@unaffiliated/faustov) |
| 16:18.40 | Faustov | hi all |
| 16:19.06 | piper69 | Faustov: hi |
| 16:20.50 | piper69 | tzafrir: Iseral |
| 16:21.26 | tzafrir | Right |
| 16:21.56 | piper69 | tzafrir: i used to fly in "Al aresh" |
| 16:23.01 | piper69 | tzafrir: nice to meet you any ways ;) |
| 16:23.15 | Faustov | tzafrir: regarding your last advice, about setting up incoming calls (context=xyz in sip.conf, then [xyz] exten => Dial(SIP/0001) in extensions.conf) - are you sure that is all i need to do to get incoming calls working? |
| 16:24.06 | piper69 | ok guys i have Cisco ATA 186 , i want someone please to take a look at the configuration at tell me if this is right for SIP |
| 16:24.15 | tzafrir | Faustov, that depends to what number you called |
| 16:24.47 | [TK]D-Fender | Faustov, and that exten... HAS no exten. |
| 16:24.48 | tzafrir | That extension needs to handle that number |
| 16:25.55 | Faustov | hmmm |
| 16:26.17 | Faustov | how do i do that? |
| 16:26.39 | piper69 | http://24.219.82.161 |
| 16:26.45 | piper69 | is this is correct |
| 16:27.28 | [TK]D-Fender | Faustov, pastebin your config and show us the complete CLI output of a failed attempt at verbose 10 & sip debug enabled |
| 16:27.46 | [TK]D-Fender | piper69, does it work? |
| 16:28.01 | Faustov | [TK]D-Fender: the funny part is, at core set debug 20 i dont get anything when i try to call the number assigned by the voip provider |
| 16:28.20 | piper69 | [TK]D-Fender: i don't know where to put my info |
| 16:28.23 | Faustov | i'll pastebin the cfg parts |
| 16:28.32 | [TK]D-Fender | Faustov, and I said with "sip debug" enabled which has nothing to do with "core debug" |
| 16:28.53 | [TK]D-Fender | Faustov, And typically you have to register with your ITSP for them to know where to send calls to. |
| 16:28.57 | c4t3l | are there any AGI gurus here |
| 16:29.06 | [TK]D-Fender | piper69, Does it WORK? |
| 16:29.18 | *** join/#asterisk implicit (i=implicit@gateway/tor/x-ffe604413ea5bcf0) |
| 16:29.29 | piper69 | [TK]D-Fender: can you take a look at the web config and please tell me if this look like where i put my info |
| 16:29.39 | c4t3l | I have a lil question regarding channel assignment in the AGI variable list |
| 16:29.46 | piper69 | [TK]D-Fender: http://24.219.82.161 |
| 16:30.30 | Faustov | [TK]D-Fender: http://pastebin.ca/829928 <--- here's my cfg |
| 16:31.34 | [TK]D-Fender | Faustov, And how does your ITSP know to send calls to you in the first place? |
| 16:31.42 | [TK]D-Fender | Faustov, Did you give them your IP, etc? |
| 16:32.24 | Faustov | [TK]D-Fender: no... doesn't * register with them in the first place? |
| 16:32.35 | Faustov | using user and pass from sip.conf? |
| 16:32.45 | [TK]D-Fender | Faustov, that's what the "REGISTER" command in sip.conf is for. |
| 16:32.55 | [TK]D-Fender | Faustov, And you don't have one. |
| 16:33.01 | Faustov | oh |
| 16:33.07 | c4t3l | I'm trying to connect an inbound call to agi, have agi call out to cell phone, play message and accept DTMF to accept or reject call. can anyone point me close to the right direction? |
| 16:33.15 | [TK]D-Fender | Faustov, read this sample for some inspiiration : |
| 16:33.18 | [TK]D-Fender | ~jerjerguide |
| 16:33.18 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 16:33.26 | piper69 | [TK]D-Fender: will this ATA work for SIP |
| 16:34.31 | [TK]D-Fender | piper69, .... go read the WIKI. These are ridiculous question you could have answered with a 4-word Google query |
| 16:35.28 | teknoprep | hey all |
| 16:35.30 | Faustov | [TK]D-Fender: i've been following that guide, altho it doesn't say anything about registering with the sip provider to allow incoming calls, just basic setup |
| 16:35.38 | piper69 | [TK]D-Fender: please don't give me this crap i have been working on this ata for the past 5 night to get it to this point, if you are willing to help me i will really appreciate it |
| 16:35.40 | teknoprep | [TK]D-Fender, did you ever run asterisk inside of vserver or qemu ? |
| 16:35.59 | Faustov | [TK]D-Fender: register => user[:secret[:authuser]]@host[:port][/extension] <---- i put this in the sip account context? |
| 16:36.23 | [TK]D-Fender | Faustov, Wow.. you're right, he didn't include the REGISTER |
| 16:36.35 | [TK]D-Fender | Faustov, Well go read the WIKI ro sample sip.conf on this. its all in there |
| 16:36.37 | c4t3l | looks like I'm on my own here... later |
| 16:36.39 | [TK]D-Fender | teknoprep, nope |
| 16:36.41 | blitzrage | Faustov: registrations go in the [general] section |
| 16:36.46 | Faustov | ok |
| 16:36.52 | Faustov | [TK]D-Fender: that's where i got this line from :P |
| 16:37.12 | [TK]D-Fender | c4t3l, what makes the AGI begin to execute in the first place? |
| 16:37.15 | teknoprep | [TK]D-Fender, i was thinking that would be a good solution to running many asterisk boxes on one machine since it uses actual hardware and not virtual hardware |
| 16:37.41 | [TK]D-Fender | teknoprep, Don't you have enough problems already? :) |
| 16:37.51 | teknoprep | [TK]D-Fender, lol no not really |
| 16:38.06 | [TK]D-Fender | Faustov, and "just basic setup" doesn't say anything much at all. |
| 16:38.12 | teknoprep | [TK]D-Fender, won't ever know if it works if you don't try it |
| 16:38.15 | [TK]D-Fender | teknoprep, Then keep at it, you're almost there! |
| 16:38.24 | teknoprep | [TK]D-Fender, lol |
| 16:38.48 | [TK]D-Fender | piper69, http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+ATA18x <----------------- |
| 16:39.14 | c4t3l | [TK]D-Fender, the agi is called from my dialplan (ivr), I've discovered a way to beat my cell provider in terms of monthly cost by using DISA and a VOIP account |
| 16:39.54 | [TK]D-Fender | c4t3l, So you want to call *, then have it hangup and call you back to count as a "free incoming call" right? |
| 16:40.02 | piper69 | [TK]D-Fender: well i read it all , the H.323 keep failing all the time |
| 16:40.03 | c4t3l | CID doesn't pass thru tho, and my VIOP provider will not ever pass thru CID :( so th AGI is just to read the CID to me |
| 16:40.35 | [TK]D-Fender | piper69, well you were asking about SIP, so whats this about H.323 now? |
| 16:41.12 | c4t3l | <[TK]D-Fender> not exactly, i want it to ask the originating caller to record name, then call me on cell and play said name and give me the option of accepting or rejecting the call |
| 16:41.17 | [TK]D-Fender | piper69, but clearly yes, * can work jsut fine witht he device and I've already linked you to the page that tells you all about using it with * |
| 16:41.42 | [TK]D-Fender | c4t3l, you don't need AGI for that at all |
| 16:41.50 | piper69 | [TK]D-Fender: this is were i am getting confused , because someone told me i have to have H.323 to get it to work, my question was for you to check and see my web Server for that ATA i post here and tell me do i have a hope of getting it to work |
| 16:42.05 | [TK]D-Fender | c4t3l, just use Record before diaing out with the "M" option. "show application dial" |
| 16:42.23 | c4t3l | did I miss something? AHHHHHHHHHHHHHH! |
| 16:42.45 | [TK]D-Fender | piper69, the page I linked you to clearly says it can talk SIP, MGCP, H.323, and SCCP. |
| 16:42.51 | [TK]D-Fender | piper69, so go read. |
| 16:42.53 | c4t3l | <[TK]D-Fender> once again thank you |
| 16:43.06 | [TK]D-Fender | c4t3l, np |
| 16:45.45 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 16:45.57 | Faustov | [TK]D-Fender: ok, now that * has registered - what else can i check? |
| 16:45.59 | blitzrage | gentoo?! BAH! :D |
| 16:46.12 | Faustov | gentoo4life |
| 16:46.15 | blitzrage | hahaha |
| 16:46.22 | Faustov | my * is running on gentoo :> |
| 16:46.31 | blitzrage | now that I've started a religious war... I'm out! heh |
| 16:46.36 | Faustov | :)) |
| 16:46.39 | [TK]D-Fender | Faustov, Try calling with SIP debug enabled and see what happens |
| 16:47.45 | [TK]D-Fender | Faustov, And has it indeed registered? |
| 16:48.32 | Faustov | yes, because i entered the pass incorrectly to see if it gonna whine |
| 16:48.36 | Faustov | and it whined |
| 16:48.39 | Faustov | so no the pass is correct |
| 16:48.59 | Faustov | and yay i noticed one packet with number-im-calling-from in the debug log |
| 16:49.04 | Faustov | lets see if it actually works now |
| 16:50.08 | [TK]D-Fender | Faustov, and the proper way to check is "sip show registry" |
| 16:54.38 | Faustov | [TK]D-Fender: i'll pastebin the output in a moment |
| 16:56.04 | teknoprep | does anyone here work with sccp in 1.4 of asteirks |
| 16:57.35 | Faustov | [TK]D-Fender: can i somehow change the registry expire timeout? looks like the default is 120 seconds |
| 16:58.26 | Faustov | [TK]D-Fender: http://pastebin.ca/829954 <--- i set the debug for only the ip of my provider, because there was too much noise |
| 16:59.29 | Faustov | [TK]D-Fender: so, i'm getting INVITEs, but looks like i'm not answering to them? |
| 17:05.38 | [TK]D-Fender | Faustov, put "nsecure=port,invite" into your peer entry and reload |
| 17:06.55 | Faustov | [TK]D-Fender: you mean "insecure"? |
| 17:07.02 | [TK]D-Fender | Faustov, just do it |
| 17:07.07 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:07.11 | Faustov | yeah, obviously :> |
| 17:07.16 | [TK]D-Fender | Faustov, your provider doesn't want the proxy auth on incoming calls |
| 17:07.37 | *** join/#asterisk adjohn (n=adjohn@p5087-ipad408marunouchi.tokyo.ocn.ne.jp) |
| 17:09.12 | piper69 | what does it mean "Set UID0: to your-login-line1"? is here where i put my sip phone number? |
| 17:11.38 | Faustov | [TK]D-Fender: ok, it works... but all mixed up for some reason |
| 17:12.17 | Faustov | phone number assigned to provider account 1 -> context 3 |
| 17:12.19 | Faustov | wtf |
| 17:12.23 | Faustov | i'll recheck my cfg |
| 17:12.39 | [TK]D-Fender | piper69, this is explained quite well in the first obvious externally linked guide in that WIKI page I gave you... |
| 17:15.16 | piper69 | [TK]D-Fender: where i can't see it man |
| 17:15.34 | [TK]D-Fender | piper69, http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+ATA18x |
| 17:16.03 | piper69 | [TK]D-Fender: i am on that page |
| 17:16.09 | *** join/#asterisk ManxPower (n=manxpowe@159.sub-75-201-35.myvzw.com) |
| 17:16.30 | [TK]D-Fender | piper69, well then go look for the first blatantly obvious externally link guide on setting it up. |
| 17:16.32 | piper69 | [TK]D-Fender: talks about that ATA 186 /188 and SIP H.323 |
| 17:17.07 | [TK]D-Fender | piper69, maybe some of it, but you are beginning to look completely blind. |
| 17:17.31 | Faustov | [TK]D-Fender: is there a limit to how many accounts i can register? |
| 17:17.38 | [TK]D-Fender | piper69, Actually alomst none of that page... |
| 17:17.47 | piper69 | [TK]D-Fender: yes because i only spend last night reading this page over and over again |
| 17:18.10 | [TK]D-Fender | piper69, A detailed practical explanation how to configure ATA-18x with Asterisk (including password reset procedure): |
| 17:18.10 | [TK]D-Fender | http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt |
| 17:18.21 | [TK]D-Fender | pip is that link not bloody well obviou enough for you?! |
| 17:19.01 | [TK]D-Fender | piper69, under a giant header called "Configuration". |
| 17:19.19 | drmessano | h.323 will be deprecated from Asterisk before he gets that working |
| 17:19.32 | drmessano | "What do you mean it won't work in Asterisk 1.8?" |
| 17:19.39 | [TK]D-Fender | drmessano, this page has virtually nothing to do with H.323. Leave it alone. |
| 17:20.32 | *** join/#asterisk Mmurdock (n=vnjyjta@111.sub-75-227-198.myvzw.com) |
| 17:20.34 | piper69 | [TK]D-Fender: i read that page too sir, plus i am not using the asterisk box with this ATA |
| 17:20.53 | ManxPower | piper69: If you are not using Asterisk, why are you here? |
| 17:21.47 | piper69 | ManxPower: Because i first want to make sure that the ATA is working then i will connect it to asterisk, i am trying to avoid trouble shoot both |
| 17:21.50 | piper69 | you get me |
| 17:22.05 | ManxPower | piper69: you really can't test it without a server to connect to. |
| 17:22.54 | [TK]D-Fender | piper69, if you want to see if its working, then use it with *. You can't control your ITSP, but you can control your * box. |
| 17:22.55 | piper69 | ManxPower: yes i can, are you telling me that i can't just hook my ATA to the router after i config it. to make calls |
| 17:23.13 | [TK]D-Fender | piper69, and until you try and connect it somewhere you won't know if its right |
| 17:23.44 | ManxPower | piper69: not unless the "router" is a voip server as well. |
| 17:23.48 | [TK]D-Fender | piper69, thats like trying to "test" a car without actually driving it! |
| 17:24.03 | ManxPower | you could do "direct ip calls" but almost nobody even knows how to do that. |
| 17:24.40 | [TK]D-Fender | piper69, You are completely backwards. Go follow the guide. You are wasting your time and ours. |
| 17:25.00 | piper69 | ok thanks |
| 17:25.01 | *** join/#asterisk aurax (n=axaxax@192.115.235.250) |
| 17:25.20 | aurax | hello, can anyone explain to me how can i connect to phone that uses sccp protocol? |
| 17:25.35 | piper69 | aurax: you can't |
| 17:25.45 | piper69 | lol |
| 17:25.50 | drmessano | [TK]D-Fender he's been asking for help with H.323 for days, I could care less what your page says.. I don't see it happening. |
| 17:26.00 | piper69 | aurax: is that a Cisco |
| 17:26.13 | aurax | yeah |
| 17:26.17 | ManxPower | aurax: SCCP/Skinny is not all that common with Asterisk. You can do it, but not many people can help you. |
| 17:26.30 | piper69 | aurax: model |
| 17:26.31 | aurax | i see |
| 17:26.36 | [TK]D-Fender | drmessano, 11:24 <piper69> ok guys i have Cisco ATA 186 , i want someone please to take a look at the configuration at tell me if this is right for SIP |
| 17:26.36 | aurax | better get ip phone that uses sip ? |
| 17:26.42 | ManxPower | aurax: your best bet is the wiki and the mailing lists. |
| 17:26.47 | Faustov | [TK]D-Fender: soo, i got 3 accounts with the same SIP provider (each account has a number assigned), for each i got a separate context for incoming calls, but for some reason, always the last context is being chosen, no matter which number i dial - what could be wrong? |
| 17:26.47 | [TK]D-Fender | drmessano, Let it go. He's asking about SIP. |
| 17:26.54 | ManxPower | aurax: 90% of asterisk users use SIP. |
| 17:27.45 | aurax | yes |
| 17:27.45 | drmessano | [TK]D-Fender: I've been here the last four days, and you really don't need to tell me what to do.. kthx |
| 17:27.50 | aurax | i can see... |
| 17:28.17 | [TK]D-Fender | drmessano, Fine then you can actually read what he asked an hour ago properly like the rest of us. |
| 17:28.45 | ManxPower | drmessano: [TK]D-Fender is one of the few people here that helps people AND knows what he is talking about. Not good to piss him off. |
| 17:29.13 | piper69 | ManxPower: [TK]D-Fender he is mean a55 whole |
| 17:29.18 | piper69 | thats what it is |
| 17:29.24 | drmessano | I'm not trying to piss anyone one.. just don't pick a fight with me either. Helping doesn't give you the right to be a jerk. |
| 17:29.39 | piper69 | help people is not asult them infront of evey one |
| 17:29.55 | aurax | it's been a while since i saw someone actually typing kthx... how old are you, 12? |
| 17:30.24 | drmessano | lol |
| 17:30.28 | drmessano | Yeah, 12.. you got me |
| 17:30.43 | Faustov | guys, stop it... |
| 17:30.50 | Faustov | i'd appreciate some help :> |
| 17:30.53 | [TK]D-Fender | piper69, I walked you through every little baby-step from Google, to the Wiki, heck to the specific link on the wiki that you could text-search for your key-word of "uid0" and come up with a 100% clearly understandable 1-sentence answer to your question. How is that not helping? |
| 17:31.30 | piper69 | [TK]D-Fender: go read what you typed |
| 17:31.30 | [TK]D-Fender | piper69, And STILL you can't even SEE it. And I did this in an attempt to help you with a model I've never even used before. |
| 17:31.45 | [TK]D-Fender | piper69, And still managed to find the answer for in under 1 minute flat of actually trying. |
| 17:31.46 | piper69 | go read how you treated me |
| 17:31.58 | piper69 | and because i need your help i listen to all this crap |
| 17:32.03 | piper69 | i even told you that |
| 17:32.10 | aurax | lol |
| 17:32.12 | aurax | this is funny |
| 17:32.14 | [TK]D-Fender | piper69, You come in here asking for help and can't seem to show that you're actually trying. |
| 17:32.38 | Faustov | aurax: no, it's not... the only person helping me around is preocupied by flamewar :D |
| 17:32.53 | piper69 | you know what , just for get it and i don't want to be like you |
| 17:33.01 | piper69 | thank you for everything any how |
| 17:33.34 | [TK]D-Fender | Faustov, make sure to specific "/[extension]" part in your "register" statements. |
| 17:33.43 | piper69 | aurax: if this is a cisco device i will help as much as i can , i spent the last 4 days trying to figuer it out |
| 17:33.51 | [TK]D-Fender | Faustov, and see how they come in. "s" is a bad thing... |
| 17:33.53 | Faustov | [TK]D-Fender: so i have to have them in both places? |
| 17:33.59 | Faustov | k |
| 17:34.01 | *** join/#asterisk shmnx (n=shaman@200-193-227-89.bsace703.dsl.brasiltelecom.net.br) |
| 17:34.45 | piper69 | aurax: is it ATA or IPphone |
| 17:35.01 | [TK]D-Fender | UID0: This is the Username for line #1. This is the same as the username in your "sip.conf" file for a SIP peer. Thus, you would enter "2299" if your sip.conf looked like our example listed at the top of this document. |
| 17:35.03 | [TK]D-Fender | ^^^^^^^^^ |
| 17:35.10 | [TK]D-Fender | Geez |
| 17:35.54 | Faustov | [TK]D-Fender: nevertheless, re-registering every 105 seconds doesn't seem right to me, should i change it? |
| 17:36.05 | [TK]D-Fender | Its all spelled out in gory detail in the Guide I linked : http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt |
| 17:36.19 | [TK]D-Fender | Faustov, Odds are its your ITSP that's asking for it. |
| 17:36.33 | [TK]D-Fender | Faustov, Guess they want to know ASAP that you are still around. |
| 17:37.15 | Faustov | weird |
| 17:39.33 | [TK]D-Fender | piper69, First 2 links of this Google search give a 100% beginning-to-end list of settings to configure your ATA :http://www.google.ca/search?hl=en&q=%22uid0%22+Cisco+ATA+186+setup&btnG=Google+Search&meta= |
| 17:39.36 | Faustov | [TK]D-Fender: To: <sip:990193@sip.easycall.pl> <--- does this mean i can change s,1,Dial to 990193,1,Dial? |
| 17:39.45 | piper69 | how can i test if my sip number is working |
| 17:39.50 | [TK]D-Fender | Faustov, Yes, exactly |
| 17:40.01 | [TK]D-Fender | piper69, Try dialing it. |
| 17:40.38 | piper69 | [TK]D-Fender: or you can give me yours to say thank you |
| 17:40.39 | piper69 | :) |
| 17:41.02 | [TK]D-Fender | piper69, well what is this magical term you call "SIP number"? |
| 17:41.10 | Faustov | oh crap |
| 17:41.15 | Faustov | i see what i messed up |
| 17:41.24 | Faustov | it's /extension not /context |
| 17:41.33 | [TK]D-Fender | Faustov, \o/ |
| 17:42.53 | ManxPower | Faustov: that would do it. |
| 17:45.17 | ManxPower | [TK]D-Fender: I'm much less grumpy since I took a vacation from helping people here. |
| 17:46.50 | [TK]D-Fender | ManxPower, Yes, escaping the source of aggravation would tend to help that :) |
| 17:47.10 | piper69 | ok i assigend it to a phone number phone number and when i call that number it goes right away to the vm |
| 17:47.44 | *** join/#asterisk Bananaskin (n=Banana@user-5af0486a.wfd99.dsl.pol.co.uk) |
| 17:47.46 | ManxPower | [TK]D-Fender: some people are beyond your help. |
| 17:47.55 | [TK]D-Fender | ManxPower, Entirely true. |
| 17:48.00 | teknoprep | Bananaskin, whats up man |
| 17:48.03 | ManxPower | Just remember you are under no obligation to help ANYONE. |
| 17:48.08 | teknoprep | Bananaskin, hopefully this fixes alot of my stuff bro |
| 17:48.18 | piper69 | Bananaskin: welcome back |
| 17:48.22 | Bananaskin | hey |
| 17:48.28 | MacWinner | is there a way to originate a call but give a time limit for it? |
| 17:48.34 | piper69 | Bananaskin: damn you are loved here |
| 17:48.35 | piper69 | :P |
| 17:48.47 | Bananaskin | only by those that want to use sccp :) |
| 17:48.56 | [TK]D-Fender | ManxPower, Yeah well I do walk away when thing reach a certain point. Usually the simpler the need, the longer I give, hoping that they'll come to it. |
| 17:49.38 | piper69 | Bananaskin: no i am a SIP guy now thanks to you |
| 17:49.49 | MacWinner | i guess, more specifically, how do you enforce a time limit on a call? |
| 17:49.51 | [TK]D-Fender | MacWinner, Originate the call with a channel type that lets you set the rules. Naturally there is only 1, so go think about it a bit :) |
| 17:49.53 | Bananaskin | Ah, did sip work for you ? |
| 17:50.14 | piper69 | Bananaskin: yes and no |
| 17:50.29 | piper69 | the file you send me didn't work |
| 17:50.43 | piper69 | Bananaskin: it does the same thing |
| 17:51.07 | Bananaskin | hmmm |
| 17:51.43 | piper69 | Bananaskin: it's not the v3.1.0 |
| 17:53.10 | teknoprep | Bananaskin, this is too fun |
| 17:54.34 | MacWinner | tkd, which channel type are you referring to? |
| 17:55.02 | [TK]D-Fender | MacWinner, Go through the list. The answer should be pretty clear. |
| 17:57.19 | Faustov | [TK]D-Fender: ok, so i got the register with /6000 which is defined in extensions.conf in [stations-a] as exten => 6000,1,MeetMe(${EXTEN},MdDix), but in sip debug i get 404 for Looking for 6000 in stations-b (domain 192.168.127.253) |
| 17:57.22 | MacWinner | local channel? |
| 17:57.55 | Faustov | [TK]D-Fender: so the big question, if context for that provider account is stations-a, then why does it look in stations-b? |
| 17:58.19 | [TK]D-Fender | Faustov, Well you put it in [stations-a] , and it's looking in [stations-b]. Well what more do we really need to say? |
| 17:58.27 | ManxPower | Faustov: did you look at the registration examples in sip.conf.sample? |
| 17:58.37 | [TK]D-Fender | Faustov, pastebin your sip.conf and sip debug for the incoming call.... |
| 17:58.51 | Faustov | ManxPower: yes, i got my lines from there |
| 17:58.52 | [TK]D-Fender | ManxPower, 'register' doesn't set the context.... |
| 17:58.55 | Faustov | [TK]D-Fender: k, one moment |
| 17:59.04 | [TK]D-Fender | MacWinner, Correct. |
| 17:59.07 | Bananaskin | teknoprep I must write up the process properly for sccp implementation |
| 17:59.19 | piper69 | [TK]D-Fender: do you know why i get voice mail when i dail the phone number assigend to my SIP before it ring on the phone connected to my ATA |
| 17:59.38 | teknoprep | Bananaskin, what i am doing works great |
| 17:59.46 | teknoprep | Bananaskin, its quite easy |
| 18:00.31 | [TK]D-Fender | piper69, You haven't shown me anything so I guess I'd have to say "no" |
| 18:01.04 | piper69 | [TK]D-Fender: i config the ata with my sip number |
| 18:01.13 | Faustov | [TK]D-Fender: http://pastebin.ca/829992 <--- sip.conf |
| 18:01.26 | Bananaskin | teknoprep thats cos the legwork is done |
| 18:01.27 | [TK]D-Fender | piper69, what is this term "SIP number"? It is not a self-defining thing....... |
| 18:01.41 | [TK]D-Fender | piper69, and you are still not showing me anything useful. |
| 18:01.52 | [TK]D-Fender | Faustov, and the sip debug of the call please |
| 18:02.05 | ManxPower | [TK]D-Fender: just put him on /ignore. He does not want to use the troubleshooting procedures that are needed. |
| 18:02.44 | teknoprep | Bananaskin, ? |
| 18:02.48 | Bananaskin | yo |
| 18:02.52 | teknoprep | Bananaskin, what you mean legwork? |
| 18:02.52 | [TK]D-Fender | ManxPower, I'm not done yet.... |
| 18:03.05 | teknoprep | Bananaskin, can you send me a copy of the cisco softphone ? |
| 18:03.09 | ManxPower | [TK]D-Fender: you can't force him to show you his sip.conf |
| 18:03.12 | Bananaskin | you are now familiar with the majority of the steps |
| 18:03.24 | teknoprep | Bananaskin, i'd like to play with that for nat'd softphones |
| 18:03.31 | [TK]D-Fender | ManxPower, that isn't even what I'd want first.... |
| 18:03.32 | ManxPower | and we both know nobody can help him without it. |
| 18:03.36 | Bananaskin | whereas to show someone from scratch is long winded |
| 18:03.48 | Faustov | [TK]D-Fender: http://pastebin.ca/829998 <--- extensions.conf, sip debug coming up |
| 18:03.52 | [TK]D-Fender | ManxPower, actually thats often incorrect |
| 18:03.56 | teknoprep | Bananaskin, i might put a full how-to for both 1.2 and 1.4 * on voip-info |
| 18:04.05 | piper69 | [TK]D-Fender: i don't understand what do you want |
| 18:04.22 | Bananaskin | In process of writing a howto atm, hence my statement |
| 18:04.28 | [TK]D-Fender | piper69, show me something where I can actually see the problem. |
| 18:05.23 | Faustov | [TK]D-Fender: http://pastebin.ca/830000 <--- debug messages |
| 18:06.05 | Faustov | [TK]D-Fender: obviously it should look in [stations-ostc], but goes to the other context instead |
| 18:06.13 | teknoprep | Bananaskin, at home if i dial the number it goes through right away |
| 18:06.17 | teknoprep | Bananaskin, but not here |
| 18:06.26 | teknoprep | Bananaskin, i think i skrewed up the number check when dialing numbers |
| 18:06.29 | [TK]D-Fender | Faustov, Found peer 'easycall3' - Looking for 6000 in stations-mind (domain 192.168.127.253) - Look what peer it matched, look at what context it uses and realize that exten isn't in that context. |
| 18:06.33 | Bananaskin | hit the # |
| 18:06.40 | teknoprep | i don't have to do that at home |
| 18:06.50 | Bananaskin | otherwise it will wait 5 secs by deafult |
| 18:07.03 | Bananaskin | did you have a dialplan.xml at home? |
| 18:07.28 | Faustov | [TK]D-Fender: ye i see it "found peer easycall3" - but it's wrong :( why 3 if i'm calling 2? |
| 18:07.39 | *** part/#asterisk ManxPower (n=manxpowe@159.sub-75-201-35.myvzw.com) |
| 18:07.55 | [TK]D-Fender | Faustov, because * is matching the last entry it finds. |
| 18:08.23 | [TK]D-Fender | Faustov, jsut sent them all to the same context and let the extens do their work. |
| 18:08.42 | Faustov | [TK]D-Fender: looks like it, wanted to do it if everything else fails |
| 18:08.57 | [TK]D-Fender | Faustov, Congratulations.... everything else fails :) |
| 18:10.32 | Faustov | [TK]D-Fender: heheh... well in exchange, if a miracle happens and i dont have to work 12h/day, i'll write a patch for this, i think it lays in *'s capabilities |
| 18:11.06 | [TK]D-Fender | Faustov, chan_sip is going under large modification already in trunk.... |
| 18:11.14 | [TK]D-Fender | Faustov, you've got a lot of catching up to do.... |
| 18:11.38 | teknoprep | Bananaskin, sometimes the phones get stuck in calling out |
| 18:11.45 | teknoprep | Bananaskin, when i dial a # that doesn't exist |
| 18:11.51 | teknoprep | Bananaskin, have you had this problem before? |
| 18:12.15 | Faustov | [TK]D-Fender: still, very glad to hear that |
| 18:12.37 | *** part/#asterisk vn (n=nostalge@modemcable083.222-80-70.mc.videotron.ca) |
| 18:13.08 | Faustov | ok, works |
| 18:13.11 | Faustov | and i'm starving |
| 18:13.29 | Faustov | thanks for help and merry xmas! |
| 18:16.38 | piper69 | [TK]D-Fender: so did you had time to take a look |
| 18:17.07 | [TK]D-Fender | piper69, Look at what? I don't see you linking me to anything useful... |
| 18:17.29 | piper69 | [TK]D-Fender: check your pm please |
| 18:18.02 | [TK]D-Fender | piper69, all I have in PM is that 1 line you sent me an hour ago. |
| 18:18.16 | [TK]D-Fender | piper69, And please don't PM me this stuff. |
| 18:18.34 | piper69 | [TK]D-Fender: well what do you want to know, this is my config |
| 18:19.16 | piper69 | i asked why is when i call my sip number , it goes right away to voice mail without ringing |
| 18:19.30 | [TK]D-Fender | piper69, and I don't see it, and as I mentioned to ManxPower, that isn't even what I would want to see first. Go look at what Faustov provided for me to help him with his issue |
| 18:20.16 | [TK]D-Fender | piper69, And your config won't tell me what the full error is necessarily. I said I need to see that actual PROBLEM. |
| 18:20.56 | *** join/#asterisk wglenncamp (n=wglennca@c-69-139-127-105.hsd1.ky.comcast.net) |
| 18:21.40 | piper69 | lol, this is confusing as hell |
| 18:21.45 | wglenncamp | Anyone here using Polycom Phones? I have a question... I recently loaded bootrom 4 on an IP501, and it seems to boot a little slower now. Any ideas (sorry that this is a little OT) |
| 18:22.42 | [TK]D-Fender | piper69, ok, I've spent all this time hoping you would actually use your eyes and see what it takes to debug these things on your own given time. This is clearly not working so I guess I'm going to have to spell it out for you. |
| 18:23.00 | [TK]D-Fender | piper69, I want to see the complete CLI output of a failed call at verbose 10, and with SIP debug enabled. |
| 18:23.13 | [TK]D-Fender | piper69, THAT is what will say "Hey I failed because of reason X!" |
| 18:23.56 | [TK]D-Fender | piper69, Just showing me your config is like showing me a recipe and not letting me notice that you set the oven to 1000 C and set it on FIRE. |
| 18:24.38 | [TK]D-Fender | wglenncamp, which SIP image are you running? |
| 18:27.12 | wglenncamp | 2.2 |
| 18:27.46 | [TK]D-Fender | wglenncamp, Which SIP app files are you specifically using in your <mac>.cofg for that phone? |
| 18:28.21 | wglenncamp | 1.233.2.32 |
| 18:28.39 | [TK]D-Fender | wglenncamp, thats the filename? |
| 18:28.53 | wglenncamp | no way.. MAC.cfg |
| 18:28.59 | wglenncamp | They are working.. Just slow |
| 18:29.18 | wglenncamp | I thought you were asking for the version |
| 18:29.36 | [TK]D-Fender | wglenncamp, no, the files specifically, just like I asked |
| 18:31.02 | wglenncamp | phone200.cfg, server.cfg, sip.cfg |
| 18:31.07 | wglenncamp | That is for extension 200 |
| 18:31.29 | wglenncamp | Then I set the phone params on the phonexxx.cfg file and the server info on server.cfg |
| 18:31.48 | wglenncamp | It just seems to take a while for the network init. |
| 18:32.08 | wglenncamp | Before it starts to load the SIP junk. It could just be me.. I am a bit impatient. |
| 18:32.44 | [TK]D-Fender | wglenncamp, Please try again... I asked exactly what sip image files were being called in your <mac>.cfg file for that phone.... |
| 18:32.58 | *** join/#asterisk squish102 (n=squish10@cpe-069-132-197-093.carolina.res.rr.com) |
| 18:33.04 | wglenncamp | They are working fine though. No, probs. Just a little slow on boot... that's all |
| 18:33.43 | [TK]D-Fender | ARGH |
| 18:33.58 | wglenncamp | What?!?! |
| 18:34.23 | [TK]D-Fender | wglenncamp, I asked you for 1 excruciating simple and specific thing and you are dancing around it. |
| 18:34.41 | wglenncamp | Dude, I'm not dancing... So chill out |
| 18:34.42 | [TK]D-Fender | wglenncamp, And showing me everything ELSE |
| 18:35.13 | [TK]D-Fender | wglenncamp, When you mechanic asks to look under the hood, you don't start giving him a detailed tour of the glove compartment and trunk... |
| 18:36.38 | wglenncamp | And, I don't take my car to a mechanic that has poor social skills. I take my business elsewhere to someone that would help me understand the issue instead of getting frustrated with me. |
| 18:36.58 | *** join/#asterisk Ahmuck (n=Ahmuck@p37n22.ruraltel.net) |
| 18:37.17 | Ahmuck | hey, i have a new slogan for asterisk - wanna here it? |
| 18:37.31 | wglenncamp | If the mechanic clearly understood, that I didn't understand his question, then he would reword it, or help get to the right answer. |
| 18:37.38 | aurax | is anyone experienced with loading chan_sccp.so? |
| 18:37.59 | Ahmuck | the new asterisk slogan - "your pbx on steroids" :-) |
| 18:39.42 | [TK]D-Fender | wglenncamp, You're <mac>.cfg specifies exactly which SIP image file names to load for you phone, that is what I have been asking for. |
| 18:39.50 | wglenncamp | sip.ld |
| 18:39.58 | wglenncamp | Is that what you need? |
| 18:40.04 | [TK]D-Fender | wglenncamp, Yes. |
| 18:40.47 | [TK]D-Fender | wglenncamp, With SIP 2.2 they broke the SIP application into several model specific components as well as including a composite SIP.LD (huge) that encompasses all model as they used to. |
| 18:41.28 | [TK]D-Fender | wglenncamp, Set your phone to use the model-specific versions as layed out in the admin guide for your 501 instead of loading the bloated composite image. |
| 18:41.40 | wglenncamp | I see. I'll check it out. Thanks |
| 18:44.03 | Ahmuck | is there a way to simulate the phone service? |
| 18:44.19 | Ahmuck | hook up x number of POTS lines to another machine? |
| 18:45.57 | aurax | is anyone experienced with loading chan_sccp.so? |
| 18:45.57 | [TK]D-Fender | Ahmuck, simulate the phone serve to whom? |
| 18:46.29 | [TK]D-Fender | Ahmuck, um... you want * to look like the telco to another PBX by any chance? |
| 18:47.51 | *** join/#asterisk A500mg (n=x@ACaen-151-1-9-230.w86-215.abo.wanadoo.fr) |
| 18:47.53 | A500mg | hi |
| 18:49.00 | Ahmuck | [TK]D-Fender: yes |
| 18:49.08 | Ahmuck | for testing |
| 18:49.35 | [TK]D-Fender | Ahmuck, well this involves hardware expense which if only for testing is kinda wasteful, but here goes : |
| 18:50.24 | [TK]D-Fender | Ahmuck, Digium TDM /400P /800P /2400P depending on the number of ports required. With this you'd get FXS modules to cover the number of ports you require. |
| 18:51.17 | [TK]D-Fender | Ahmuck, Other models : Sangoma A200 series, Rhino has a modular card as well. You could also use ATAs like the Linksys SPA-2102 or larger gatways lke the SPA-8000 |
| 18:59.06 | A500mg | question: AEX800/TDM800 works with OSLEC ? |
| 18:59.43 | [TK]D-Fender | A500mg, And device using the Zaptel channel driver. |
| 19:01.18 | tzafrir | A500mg, it should. OSLEC simply uses the standard Zaptel EC interface |
| 19:01.32 | A500mg | ok :) |
| 19:01.49 | A500mg | and if i use a b410p, a tdm01b, and oslec, no problem ? |
| 19:02.31 | [TK]D-Fender | A500mg, Same answer......... |
| 19:02.59 | A500mg | b410p use zaptel or misdn ? |
| 19:03.21 | tzafrir | b410p: should. Though it is misdn, so you'll have to rely on misdn's support of OSLEC. |
| 19:03.34 | tzafrir | It should support it but I have no idea how wel lit works |
| 19:04.58 | A500mg | Can I use b410p with classic echo canceller, and in the same time a tdm400 with oslec ? |
| 19:08.41 | tzafrir | yes |
| 19:08.44 | A500mg | (I don't know if I can use oslec with misdn, I've never used misdn ...) |
| 19:08.45 | [TK]D-Fender | A500mg, Zaptel uses a single routine period |
| 19:09.18 | [TK]D-Fender | A500mg, if your card uses Zaptel, then thats the final answer. Whether MISDN supports anything seperate or is even relevent is another matter. |
| 19:09.32 | A500mg | ooh ok |
| 19:09.42 | A500mg | no relation between zaptel and misdn |
| 19:10.11 | A500mg | if i configure oslec with zaptel, no effect on misdn echo cancelation |
| 19:10.46 | A500mg | ok ok :) |
| 19:27.00 | *** join/#asterisk Mmurdock (n=vnjyjta@111.sub-75-227-198.myvzw.com) |
| 19:28.25 | *** join/#asterisk Deeewayne (n=Deeewayn@ool-43522b13.dyn.optonline.net) |
| 19:28.25 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:28.32 | Qwell | Deeewayne: hey |
| 19:28.51 | Deeewayne | Qwell: howdy! |
| 19:29.04 | *** join/#asterisk tobias (n=tobias@nat1.ppckernel.org) |
| 19:29.10 | Qwell | how's the Easterly coast? |
| 19:31.30 | wglenncamp | [TK]D-Fender, Thanks for your help. The new sip file sped things up. |
| 19:31.32 | wglenncamp | Thanks again |
| 19:31.44 | wglenncamp | Next time, I'll read the manual |
| 19:32.12 | wglenncamp | er, I mean release notes |
| 19:32.29 | [TK]D-Fender | wglenncamp, Always a good thing to do. |
| 19:33.45 | wglenncamp | I just assumed it all worked the same as the old versions. |
| 19:34.01 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:35.01 | Deeewayne | Qwell: pretty good, but raining :-( |
| 19:35.07 | Mmurdock | Wglenncamp: are you moving from 1.2 to 1.4? |
| 19:35.20 | [TK]D-Fender | wglenncamp, there's a great saying for that. |
| 19:35.26 | [TK]D-Fender | ~assume |
| 19:35.26 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 19:44.53 | *** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
| 19:44.53 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:46.13 | wglenncamp | Mmurdock, no I was updating bootroms and sip on my polycom phones |
| 19:47.34 | WilliamK | 2.2 is alot nicer on poly's |
| 19:47.35 | WilliamK | :) |
| 19:48.52 | [TK]D-Fender | Espcially given the massive improvements to thee microbrowser, dial prefix prepening, Ring on CW, etc that they added. |
| 19:49.32 | Mmurdock | Wglenncamp: ah, cool. |
| 19:55.28 | *** join/#asterisk seanwg123 (n=seanwg12@bas1-calgaryqa-1242361325.region2.highspeedunplugged.bell.ca) |
| 19:59.56 | *** part/#asterisk Bananaskin (n=Banana@user-5af0486a.wfd99.dsl.pol.co.uk) |
| 20:00.23 | seanwg123 | anyone know if its possible to make routes like '+14412995959'? |
| 20:00.29 | seanwg123 | sorry extensions? |
| 20:01.21 | [TK]D-Fender | seanwg123, Don't see why no. Have you tried? |
| 20:01.32 | seanwg123 | yah |
| 20:01.35 | piper69 | question: i signup with sipphone.com and they provided my sip number (i configure it in my ATA). i also signup with IPKALL.com to get a phone number to tie it to my sip number, now when i call the phone number provided by ipkall i get the voice mail |
| 20:01.42 | seanwg123 | it seems to treat + as regex |
| 20:01.54 | [TK]D-Fender | seanwg123, Show us the full attempt and your dialplan |
| 20:03.00 | [TK]D-Fender | piper69, If you're configuring your ATA direct with another provider, we'll never know why. We can't debug what the device is doing. |
| 20:03.10 | [TK]D-Fender | seanwg123, Pastebin it all please. |
| 20:03.12 | [TK]D-Fender | ~pb |
| 20:03.13 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste |
| 20:03.46 | [TK]D-Fender | piper69, You'd be advised to configure * to connect to your ITSP, and the ATA to *. |
| 20:05.27 | piper69 | [TK]D-Fender: ok , i am waitting for timeshell he said he will helo me setup my * |
| 20:06.03 | piper69 | [TK]D-Fender: i will not be using FX decives , just my ATA |
| 20:06.22 | [TK]D-Fender | piper69, well you have no debug to show for whats going on with your Cisco, and * isn't even involved. We can't help you with this. |
| 20:07.02 | [TK]D-Fender | piper69, and what do you mean with "FX" devices? |
| 20:07.29 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:07.38 | piper69 | [TK]D-Fender: i completely understand your point of view, but i am sure that at 1 point in your life you were stuck with something and you tried to go to difrenet toom to get help |
| 20:08.21 | [TK]D-Fender | piper69, I don't ask my accountant to fix my car you know. And you have nothing to offer us to even try to help you. |
| 20:09.42 | piper69 | [TK]D-Fender: show me another channl , talk talk about voip and i will GO |
| 20:10.05 | piper69 | abviouslly you guys are not accountant |
| 20:11.25 | [TK]D-Fender | piper69, Go find yourself a certified Cisco tech, try in #cisco, or get Google-ing for information. |
| 20:12.09 | [TK]D-Fender | piper69, and this isn't #voip either. Few of us here use that model and fewer still can debug anything other than its use with Asterisk. |
| 20:12.39 | [TK]D-Fender | piper69, Yes, we do know more about spcific models, but that isn't one of the more supported ones. |
| 20:13.01 | piper69 | [TK]D-Fender: so are you the spoke man of the room |
| 20:13.03 | piper69 | ?? |
| 20:13.55 | [TK]D-Fender | piper69, Do you know how many years I've been in here and how much time I spend supporting this channel and * in general? That I also consult in it, go to the occasional conference, etc? |
| 20:14.50 | piper69 | [TK]D-Fender: do you know how many years i am a 5ESS DCS switch engineer |
| 20:14.52 | [TK]D-Fender | piper69, or perhaps that I have redone if not created most of the jbot information snippets for support & tehnical info, written up support documents for * as well? |
| 20:15.39 | piper69 | [TK]D-Fender: maybe all your life you will not even enter the control room for that equipment , but still that why we all are diffrent , you know something and i know something |
| 20:15.49 | [TK]D-Fender | piper69, yes, your experience is with 5ESS DCS. That is a good thing of course, but you asked me about being at all worthy of being some kind of spokesperson for #asterisk. I think I have answered that quite well. |
| 20:16.31 | piper69 | so you could easily forget or ignore my questions if you think you can't help me |
| 20:16.50 | piper69 | thats why i ask in the room , hoping if someone knows something about it |
| 20:17.41 | wglenncamp | You may be able to find you answer at #trixbox. ;) |
| 20:18.09 | piper69 | i really hate to cut this short , but i need to go somewhere now. |
| 20:18.17 | piper69 | wglenncamp: i will try that thank you sir |
| 20:18.38 | [TK]D-Fender | wglenncamp, Thats just mean.... |
| 20:18.47 | wglenncamp | :) |
| 20:18.57 | jer | if i do a: dialplan add extension 1234,1,Dial,SIP/1234 into foo ... and the context 'foo' doesn't exist, does that create the context 'foo' ? |
| 20:19.22 | [TK]D-Fender | jer, you mean from * CLI? |
| 20:19.25 | jer | yup |
| 20:19.47 | [TK]D-Fender | jer, Unusual, but it would make perfect sense that it would. Go try |
| 20:19.55 | jer | i suppose i could do that |
| 20:20.16 | [TK]D-Fender | jer, when in doubt you should always try first..... |
| 20:20.19 | jer | ah, failed to add it |
| 20:20.30 | [TK]D-Fender | jer, its not like the answer would take even as long as your question :) |
| 20:20.57 | jer | [TK]D-Fender, this is true, my apologies |
| 20:25.22 | [TK]D-Fender | jer, I do wonder though.... why would you even want to use the CLI to add stuff to your dialplan in such a temporary manner? |
| 20:25.44 | jer | [TK]D-Fender, i'm just poking around |
| 20:26.27 | [TK]D-Fender | jer, have fun then... |
| 20:27.58 | _x86_ | re all |
| 20:38.07 | *** join/#asterisk ZX81 (n=ZX81@121.90.169.122) |
| 20:40.25 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
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| 21:37.42 | tzafrir_home | mpg123 1.0 was released... |
| 21:44.33 | WilliamK | wow |
| 21:44.35 | WilliamK | that took a while |
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| 22:38.56 | *** join/#asterisk knarfly (n=vtserije@c-75-74-155-198.hsd1.fl.comcast.net) |
| 22:39.34 | knarfly | I installed the asterisk-gui but cannot find the readme file...anyone know where it is...Oh yes, I run FreeBSD not linux |
| 22:41.47 | *** join/#asterisk implicit_ (i=implicit@gateway/tor/x-97f0b983b03ea056) |
| 23:01.22 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 23:01.40 | *** join/#asterisk supjigator (n=sysgod@152.53.16.10) |
| 23:01.45 | *** part/#asterisk supjigator (n=sysgod@152.53.16.10) |
| 23:17.17 | *** join/#asterisk tobias (n=tobias@nat1.ppckernel.org) |
| 23:19.38 | wglenncamp | another polycom question.. :) My IP501 phones MWI is working, and when I click on messages, it shows an overview of what's on the server, but when I click connect on the phone it just hangs up. Any ideas? |
| 23:20.29 | *** join/#asterisk HybridStorm (n=HybridSt@adsl-066-156-078-028.sip.asm.bellsouth.net) |
| 23:20.35 | [TK]D-Fender | wglenncamp, Go look in sip.cfg for "OneTouch" and your mwi contact. By defaul it tries to dial the username you register as. |
| 23:21.44 | wglenncamp | When I set the Onetouch, it phone doesn't do anything after that |
| 23:22.08 | wglenncamp | Like it doesn't know where to dial, but I have it set in phone1.cfg |
| 23:22.33 | [TK]D-Fender | wglenncamp, pastebin the appropriate tags from each |
| 23:22.35 | [TK]D-Fender | ~pb |
| 23:22.35 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste |
| 23:23.13 | wglenncamp | ok, one sec. |
| 23:24.41 | HybridStorm | what is the best way to do failover support in asterisk? |
| 23:25.24 | wglenncamp | http://pastebin.ca/830430 |
| 23:25.45 | [TK]D-Fender | HybridStorm, What you attempting to maintain in this "failover" and what role * plays is the key factor in answering that question. |
| 23:26.39 | [TK]D-Fender | wglenncamp, So you want it to dial 8500 for VMM? |
| 23:27.25 | wglenncamp | yes |
| 23:27.46 | HybridStorm | [TK]D-Fender, what I am looking to do is have a hot spare asterisk box on call should my first one lose connectivity within my data center space and have it swap over automatically. |
| 23:28.58 | wglenncamp | Maybe a dialplan issue? Possibly? |
| 23:30.03 | Greek-Boy | [TK]D-Fender I remember when I asked you what asterisk trunk is you said something about asterisk 1.6 |
| 23:30.09 | Greek-Boy | is "trunk" a code name? |
| 23:30.53 | [TK]D-Fender | wglenncamp, Looks fine.... pastebin a failed attempt to use at verbose 10 SIP debug enabled. |
| 23:32.18 | [TK]D-Fender | Greek-Boy, Not so much... I'm not sure the best way to word that. Any of the core developers would be able to elaborate better. Try asking Qwell, file, Corydon, etc when they're around. |
| 23:32.38 | [TK]D-Fender | wglenncamp, Could be, thats why I want to see exactly what comes up. |
| 23:32.53 | wglenncamp | nothing to debug, the phone doesn't even attempt to dial |
| 23:32.59 | wglenncamp | just tried it |
| 23:34.13 | [TK]D-Fender | wglenncamp, So you push "Messages" and nothing seems to happen at all? |
| 23:34.32 | Greek-Boy | will do |
| 23:34.33 | wglenncamp | right |
| 23:34.41 | [TK]D-Fender | wglenncamp, pastebin your <mac>.cfg please |
| 23:38.00 | wglenncamp | http://pastebin.ca/830444 |
| 23:38.19 | wglenncamp | I pasted my <mac>.cfg and phone1.cfg |
| 23:39.19 | [TK]D-Fender | wglenncamp, So you see nothing with SIP debug enabled at * CLI? |
| 23:39.38 | *** join/#asterisk saftsack (n=oliver@p4FC7468F.dip.t-dialin.net) |
| 23:39.40 | file | trunk is where new development appears. |
| 23:40.03 | [TK]D-Fender | wglenncamp, Have you attempted any other ke-remapping? |
| 23:40.20 | wglenncamp | nope. Everything is default. |
| 23:40.30 | wglenncamp | I can dial the extension directly from the phone though |
| 23:40.36 | file | (At least for Asterisk... Zaptel is a different story...) |
| 23:41.23 | [TK]D-Fender | wglenncamp, take a look in Menu > Status > Platform > Configuration directly on the phone |
| 23:41.46 | [TK]D-Fender | wglenncamp, Verify what files and in which order they are listed |
| 23:42.30 | [TK]D-Fender | wglenncamp, And when you showed me sip.cfg and & phone1.cfg, were those only portions of those files? |
| 23:43.31 | wglenncamp | no, they were the whole file. But, I also have another file that gets used.. phone<extension>.cfg |
| 23:43.42 | wglenncamp | and server.cfg |
| 23:44.37 | [TK]D-Fender | wglenncamp, I think you're missing enough of the nested tags in those 2 files to make the onles you left functional... |
| 23:45.14 | [TK]D-Fender | wglenncamp, For instance a stock phone1.cfg is encased in a <phone1> tage, etc.... |
| 23:45.41 | [TK]D-Fender | wglenncamp, you might want to scrap your preakdown method and try to do them off the base files as provided. |
| 23:45.59 | *** join/#asterisk dannz (n=dannz@pluto.codev.co.nz) |
| 23:46.06 | [TK]D-Fender | wglenncamp, Tag heirarchy issues are the trickiest to debug |
| 23:46.13 | wglenncamp | I see. |
| 23:46.41 | wglenncamp | Weird thing is that this setup works fine in a (cough cough) .... |
| 23:46.44 | wglenncamp | ...trixbox |
| 23:46.47 | wglenncamp | :) |
| 23:47.05 | [TK]D-Fender | wglenncamp, Decapitation works wonders for coughs like that! |
| 23:48.03 | wglenncamp | I looked at the platform info, and they looked right though. All of the files that are supposed to be listed are listed |
| 23:48.19 | wglenncamp | But, I didn't realize order was a big concern |
| 23:49.06 | *** join/#asterisk ZX81 (n=ZX81@121.90.90.231) |
| 23:49.08 | [TK]D-Fender | wglenncamp, Order can matter if a tage repeats, etc. Listing a minimal inner/outer tags, etc |
| 23:49.30 | [TK]D-Fender | wglenncamp, Unless you really know what you're doing, rocking the boat will break things in funny ways.... |
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