| 00:03.24 | joako | LOL |
| 00:03.51 | joako | It seems they took away the firmware about 5 years ago. |
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| 00:24.45 | [TK]D-Fender | "the firmware"? |
| 00:25.02 | [TK]D-Fender | http://www.mitel.com/sip-phones |
| 00:25.12 | [TK]D-Fender | I see SIP firmware for some of their phones RIGHT HERE.... |
| 00:26.50 | joako | I am looking for the firmware for a Mitel 5304 phone. |
| 00:30.36 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Mitel+SIP+Firmware |
| 00:30.47 | [TK]D-Fender | they mention that you normally go through a reseller |
| 00:30.54 | [TK]D-Fender | and then mirror a direct download source |
| 00:30.58 | [TK]D-Fender | Go run with that |
| 00:32.15 | joako | I dont have a reseller :( |
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| 01:08.46 | someone11 | whois mjordan |
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| 03:56.23 | carrar | moof |
| 03:57.28 | ChannelZ | FOOM |
| 03:58.06 | carrar | ZOOM |
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| 05:48.19 | skyroveRR | Has anyone in here tried chan_celliax module? |
| 05:49.32 | skyroveRR | I'm trying to setup that module and I want to setup asterisk without any voip temporarily. |
| 05:50.11 | skyroveRR | I've setup all the necessary hardware, including the headset/microphone output, and have an AT compatible phone connected. |
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| 08:36.12 | ChannelZ | skyroveRR, That channel driver is ancient |
| 08:36.31 | skyroveRR | ChannelZ: then which is the latest and working one? |
| 08:37.12 | ChannelZ | I have no idea |
| 08:37.23 | skyroveRR | O.o |
| 08:38.42 | ChannelZ | Probably none, unless you run a similarly ancient version of Asterisk. The page I found (I had to search, I'd never even heard of it before) talks of Asterisk 1.2 and 1.4 |
| 08:39.15 | skyroveRR | ChannelZ: I saw the github repo, it says it's updated for 1.8.X. |
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| 11:33.58 | *** join/#asterisk Guest8893 (53ff29a0@gateway/web/freenode/ip.83.255.41.160) |
| 11:34.46 | Guest8893 | Hi, am new to Asterisk and I am looking to implement a feature in my installation. I wish all SIP messages to be saved to a file. I know from experience that Kamailio output all sip messages to the syslog. Am after something similar for Asterisk. |
| 11:35.12 | Guest8893 | I have enabled debugging and it now outputs to the Asterisk console, but I do not want that... also I want to avoid using debugging in a productive env |
| 11:47.17 | Guest8893 | I have debug=yes in sip.conf also in logger.conf I have messages => notice,warning,error,debug,dtmf |
| 11:47.19 | Guest8893 | Still no SIP messages |
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| 12:52.45 | rexwin_ | does logmein allow copy and paste to local machine? |
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| 13:39.31 | LemensTS | the Polycom VVX 400 is a 12 line phone, does that mean I can put 11 people on hold at one time using it (If that sip user has that many allowed connections in sip.conf of course) |
| 13:51.34 | WIMPy | Maybe. |
| 13:51.53 | WIMPy | As SIP doesn't have line, that maketing term could mean anything. |
| 13:52.27 | WIMPy | 12 accounts, 12 calls, some combination thereof, something else, ... |
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| 13:53.50 | LemensTS | Yea i've never really messed with the line buttons, I knew you could make each one their own sip account. I'm not sure how they were with putting people on hold? |
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| 13:54.55 | carrar | VVX 400 needs a faster video processor |
| 13:55.34 | carrar | but indeed it's a nice phone |
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| 14:00.37 | LemensTS | carrar: Do you know how hold works on that phone? |
| 14:00.59 | LemensTS | Does it put the held call on a line button? And if you hold more calls does it use more line buttons for those? |
| 14:01.48 | WIMPy | 'd wish that more phones had call buttons instead of those totally senseless line buttons. |
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| 15:24.54 | emk | Hey all, what is a good affordable (cheap) hardware option for adding some GSM(sim card lines) to My Asterisk Server? |
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| 16:29.22 | [TK]D-Fender | emk, chan_mobile |
| 16:29.34 | [TK]D-Fender | emk, Sangoma, etc also make DAHDI compatible cards for this |
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| 19:20.48 | someone7 | <PROTECTED> |
| 19:21.11 | someone7 | directly related to getting Asterisk monitored |
| 19:22.59 | someone7 | and this is the website I used for the tutorial although I added the text for all but the first item that I had to input which I used the CAT for http://sysadminman.net/blog/2010/blocking-asterisk-hackingscanning-attempts-with-fail2ban-1392 |
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| 21:19.43 | rexwin_ | extension to extension calls can't be heard. freepb runs above asterisk |
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| 21:30.48 | ChannelZ | Extensions on the same LAN? On the same LAN as the asterisk server? |
| 21:32.35 | rexwin_ | asterisk is over cloud and I test it from my two softphones located elsewhere but connected to cloud asterisk |
| 21:33.11 | rexwin_ | no is the answer to both of your questions |
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| 21:35.15 | rexwin_ | how to check audio is transmitted from my desktop softphone to my cloud asterisk |
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| 23:12.22 | phix | mornin |
| 23:12.47 | WIMPy | phix! |
| 23:13.09 | phix | I can make SIP calls fine on my mobile (android) via asterisk, however when someone calls me and I pick up I get a loud tone that plays and I can barely hear the caller |
| 23:13.13 | phix | any ideas what would cause that? |
| 23:13.16 | phix | WIMPy:! |
| 23:13.28 | phix | Me best asterisk pal! |
| 23:13.59 | WIMPy | wonders if that's a good or a bad thing in that context. |
| 23:30.43 | phix | All good |
| 23:30.46 | phix | so any ideas? |
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