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| 01:21.21 | iprouteth0 | how does one add dahdi in properly when compiling from source? |
| 01:22.09 | WIMPy | Have it installed befor you configure Asterisk. |
| 01:22.39 | iprouteth0 | so I'd just config, make, and make install it, bevore make menuselect with asterisk? |
| 01:23.02 | WIMPy | No, before you ./configure Asterisk. |
| 01:23.17 | iprouteth0 | right |
| 01:23.43 | iprouteth0 | cool, thanks. Now I just need to get my kernel source for the raspberry pi |
| 01:23.59 | WIMPy | Uh oh |
| 01:24.28 | WIMPy | But why do you want dahdi on it? |
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| 01:24.45 | iprouteth0 | end goal is to use bluetooth headset as endpoint, for which I believe zap channel is required for fxs |
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| 01:24.51 | iprouteth0 | or am I in error? |
| 01:25.18 | iprouteth0 | was able to add in chan_mobile without any trouble |
| 01:25.52 | WIMPy | BT should not have anything to do with dahdi and you can;t add any ports to the rpi anyway. |
| 01:26.02 | WIMPy | Well, maybe an Astribank. |
| 01:26.43 | WIMPy | Or the usual USB BRI dongles, off course, but that wouldn't be dahdi. |
| 01:26.54 | iprouteth0 | I'll have to keep reading up on my chan_mobile then. |
| 01:27.34 | iprouteth0 | curretly compiling without dahdi. Using asterisk certified for d30 |
| 01:27.36 | iprouteth0 | d40* |
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| 03:35.39 | ghostmediapro | is it possible to configure asterisk as a h323 gateway or gatekeeper |
| 03:50.03 | jpsharp | Yes. |
| 03:50.42 | jpsharp | But Asterisk's H323 stack isn't as well supported as the SIP and/or IAX stack. |
| 04:00.45 | *** join/#asterisk sertaconay (~sertacona@unaffiliated/sertaconay) |
| 04:02.58 | ghostmediapro | thanks for your response i'm looking sto setup a video confernce using polycom Viewstation fx, but it support only h323 |
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| 04:05.51 | jpsharp | I've used the H323 and SIP stacks to talk to Tandberg conference boxes, but never Polycom. |
| 04:10.44 | ghostmediapro | i've been googling for a while on a basic config for ooh323.conf |
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| 07:36.20 | bombev | Hi, all |
| 07:36.39 | bombev | any idea hot to delete my asterisk log via SSH |
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| 07:40.25 | ChannelZ | what, besides 'rm' ? |
| 07:43.37 | wdoekes | did you dial 0800-PORN ? |
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| 08:18.48 | BorjaGVO | Hi, anyone knows what specs does the .wav files recorded by mixmonitor have? (8,000 Hz 16 bit PCM?? No compression at all?) |
| 08:22.19 | kaldemar | 8 kHz, 16-bit, mono |
| 08:23.11 | BorjaGVO | alright |
| 08:23.12 | BorjaGVO | thanks |
| 08:23.52 | kaldemar | if you want something else, choose the format that MixMonitor uses. see "core show application MixMonitor" and "core show file formats" |
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| 08:25.09 | kaldemar | some people also use the command option for MixMonitor to handle the file after recording. |
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| 08:26.44 | [sr] | hi |
| 08:26.49 | [sr] | hi WIMPy |
| 08:27.47 | _omer | http://pastebin.com/BfjPc8RA (Freepbx is creating config files in wrong folder. anyhelp please) |
| 08:29.18 | kaldemar | ~freepbx |
| 08:29.18 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 08:31.00 | _omer | thanks |
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| 11:11.05 | lantizia | Hey I've got full => notice,warning,error,debug,verbose set but for some reason I've got only a few megs worth of logs on a very busy system |
| 11:11.08 | lantizia | am i missing something? |
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| 11:26.47 | hwt | hi, is the source IP of an incoming call available from the dialplan somehow? |
| 11:27.01 | kaldemar | what kind of a call? |
| 11:28.38 | hwt | SIP |
| 11:28.46 | ^rage^ | hwt: sip header |
| 11:29.15 | hwt | and just fetch the Via? that doesn't necessarily work if it's NATed |
| 11:29.38 | kaldemar | hwt: ${CHANNEL(peerip)} |
| 11:29.44 | hwt | kaldemar: great. thanks! |
| 11:30.01 | ^rage^ | kaldemar: ok, you win ;) |
| 11:30.36 | kaldemar | there is also CHANNEL(recvip) |
| 11:30.39 | ^rage^ | anybody use linksys/cisco spa ip phones? ;) |
| 11:30.48 | hwt | ^rage^: yes. |
| 11:31.13 | ^rage^ | hwt: how you implemented conferences? |
| 11:31.32 | ^rage^ | more that 3 persons |
| 11:31.44 | hwt | ^rage^: asterisk. |
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| 11:33.12 | ^rage^ | hwt: ok, i know about confbridge/meetme/conference. |
| 11:33.41 | hwt | ^rage^: that's pretty much what there is in the FOSS world |
| 11:33.43 | ^rage^ | hwt: this phones have softkeys like 'conf'. |
| 11:33.58 | hwt | ^rage^: if you're willing to shell out a lot of money, you can get hardware boxes from Cisco, Polycom, etc. |
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| 11:34.24 | hwt | ^rage^: they are only for three-way conferencing. media is bridge by the phone itself. |
| 11:35.02 | ^rage^ | hwt: when this softkey pressed, ip phone make invite to some external conference server |
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| 11:35.33 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:36.50 | ^rage^ | cisco use for this <meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI> |
| 11:37.18 | ^rage^ | and make invite to x-cisco-serviceuri-meetme-${EXTEN}@sipproxy |
| 11:38.56 | ^rage^ | simply, just cut x-cisco-serviceuri-meetme- we get peer |
| 11:40.02 | ^rage^ | linksys/cisco spa use another way: they send sip notify with event: conference |
| 11:41.40 | ^rage^ | yep, i can setup openser/kamailio as frontend.. |
| 11:42.15 | ^rage^ | but it seems too complicated. |
| 11:42.55 | leifmadsen | it is :) |
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| 12:34.49 | drzed | hi there! |
| 12:36.52 | drzed | how do i create a context so that an incomming call with no suffix is put a sip phone |
| 12:37.20 | drzed | i read that using ._,1,Dial(SIP ... is not a good idea? |
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| 12:37.47 | anonymouz666 | samsung galaxy note II |
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| 12:43.44 | jeffspeff | So, i just installed * 10.9.0 last night, and within menuselect i did not choose the deprecated package of app_mysql but for some reason the system added app_mysql.conf to /etc/asterisk does that mean it was compiled anyways? |
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| 12:47.04 | kaldemar | jeffspeff: no. you'll find the compiled and installed modules under /usr/lib/asterisk/modules/ |
| 12:47.28 | kaldemar | drzed: suffix to what? |
| 12:47.44 | jeffspeff | kaldemar, that's what i thought. it's not there, just seems odd that it would make a config file for an absent module |
| 12:48.01 | drzed | kaldemar: oh i found out that s,1,Dial( does |
| 12:48.57 | drzed | what i itend |
| 12:49.32 | kaldemar | jeffspeff: the "samples" target in the makefile copies all sample configs despite the modules you have selected. |
| 12:49.41 | jeffspeff | oh |
| 12:49.43 | jeffspeff | ty |
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| 13:00.27 | [TK]D-Fender | drzed, No. Firstly, Asterisk has no way to pattern match a SUFFIX. Next, "s" is not a "pattern" of any kind for SIP. |
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| 13:00.40 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:01.18 | [TK]D-Fender | jeffspeff, when you "make samples", it's an entire folder of configs for all possible modules that gets copied over, not selective |
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| 13:42.20 | lantizia | Hey I've got full => notice,warning,error,debug,verbose set but for some reason I've got only a few megs worth of logs on a very busy system... am i missing something? |
| 13:44.39 | pabelanger | lantizia: *CLI> logger reload |
| 13:44.46 | lantizia | yeah done that already |
| 13:44.53 | pabelanger | tail -f /var/log/asterisk/full |
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| 13:56.45 | b3nt_pin | away |
| 13:56.54 | b3nt_pin | not really |
| 13:57.19 | b3nt_pin | sorry folks, more caffeine req'd this morning |
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| 14:21.08 | mirela666 | Hi, is there a way to pass variables to Originated call |
| 14:21.21 | [TK]D-Fender | mirela666, Depends how you "originate"d it |
| 14:21.33 | [TK]D-Fender | mirela666, CLI = no, AMI=yes |
| 14:21.43 | [TK]D-Fender | mirela666, app_originate=no |
| 14:22.13 | mirela666 | with app |
| 14:22.28 | mirela666 | [TK]D-Fender: thx for quick resp. |
| 14:24.54 | mirela666 | so only solutions are writing script for AMI or trying with SHARED() func |
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| 14:33.44 | cusco | or use DBGet() DBPut() |
| 14:34.06 | cusco | that is what I'm using in .call files |
| 14:35.10 | [TK]D-Fender | mirela666, or call-files |
| 14:36.04 | [TK]D-Fender | cusco, He jsut wants to pass a var. DB might work, but fails on concurrency unless you dial a variable exten and set a pile of keys instead including the real dest. |
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| 14:36.11 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:36.39 | cusco | in my case, each call file has a unique identifier that I use as a key |
| 14:36.51 | cusco | and I remove it when the call ends |
| 14:37.04 | cusco | in the call file I Set: it |
| 14:38.07 | [TK]D-Fender | cusco, You can jsut pass actual vars ni the call-file method anyway. Actually DB only really seems to pay off if using app_originate .... which is a locking app anyway... far from ideal. |
| 14:40.00 | mirela666 | I've send Originate on exten and in ${EXTEN} i sent all variables i needed but I have reached limit of EXTEN length L| |
| 14:40.24 | mirela666 | with delimiter ofcours |
| 14:46.10 | [TK]D-Fender | mirela666, Yeah, simple things you might be able to encode, but AMI and call files allow you to set multiple anyway.... I'd aim for the simplest means for multiple values in that case. Also saves on a ton of parsing afterwards, plus the impact on CDR's because of the exten.... |
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| 14:54.56 | cusco | in call files I can't set variables in the first channel and get them back at the second channel (extension/context specified in the call file) |
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| 14:55.18 | cusco | Im trying to dial some cisco peer, and they are not sending us 'ANSWER' |
| 14:55.20 | cusco | :| |
| 14:55.30 | [TK]D-Fender | cusco, Should be available as-is through both.... |
| 14:55.34 | cusco | he can't figure it out, I don't think it is a problem in asterisk |
| 14:55.41 | drmessano | Anyone using Asterisk 11 + XMPP? |
| 14:56.17 | cusco | [TK]D-Fender: if the variable is set in the call file, it is, but if it is set in the dialplan of CHANNEL: Local/bla...., the extension where it goes to when answered, does not have varibales set in the first channel |
| 14:58.10 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:58.26 | [TK]D-Fender | cusco, Oh no, those should follow standard inheritance rules... |
| 14:59.07 | dose | Hi Everybody, any idea how to force a peer to add a domain to a dialled number because my peer doesn't do it, please see example below there is extension *074577 without domain or IP: |
| 14:59.09 | dose | From: 46044 <sip:46044@toto.com>;tag=bbfadsdsac To: *074577 <sip:*074577>;tag=as67ds0e62 |
| 14:59.39 | dose | so asterisk doesn't know what to do with it |
| 15:00.29 | dose | I can see the call arriving on asterisk with sip set debug on and I get an error 401 unauthorized |
| 15:05.27 | *** join/#asterisk deo (~dnepangue@112.198.79.211) |
| 15:14.59 | *** part/#asterisk deo (~dnepangue@112.198.79.211) |
| 15:15.20 | *** join/#asterisk deo (~dnepangue@112.198.79.211) |
| 15:16.04 | kozerski | hi |
| 15:17.06 | *** join/#asterisk Glonet (~Glonet@server.glonet.be) |
| 15:20.44 | *** join/#asterisk sruffell (sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:20.44 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:24.05 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 15:24.52 | *** join/#asterisk blee (~blee@72.188.117.219) |
| 15:26.43 | *** join/#asterisk hurdman (~ygcheny@bender.r0b0t.fr) |
| 15:26.48 | hurdman | hello folks |
| 15:27.22 | hurdman | have you had issue with digium 4xE1 or 8xE1 pci-E card, with asterisk and sandy bridge server ( debian for example ) ? |
| 15:27.33 | hurdman | because mine reboot.... without error |
| 15:28.05 | hurdman | and never with a stress test, but ever when i make a lot of call with asterisk :'( |
| 15:28.05 | [TK]D-Fender | hurdman, What tells you that the card is responsible? |
| 15:28.33 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 15:28.54 | hurdman | i have the same cards |
| 15:28.58 | hurdman | 4 and 8 E1 |
| 15:29.08 | hurdman | with "old" motherboard and same code |
| 15:29.12 | hurdman | no problem at all |
| 15:29.17 | hurdman | only with sandy bridge :/ |
| 15:30.14 | hurdman | i don't think it's the card [TK]D-Fender , i think my server only crash with asterisk+dahdi calls , and never with stress ( memory, cpu, disk ) tests |
| 15:30.40 | [TK]D-Fender | hurdman, You should probably be proving the origin of your crash..... |
| 15:31.14 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 15:31.22 | hurdman | that's with , i want to know if someone else have had issue with digium 4xE1 or 8xE1 and sandy bridge mother board ;) |
| 15:31.53 | [TK]D-Fender | hurdman, If it was gloabal you'd see something from Digium on it. |
| 15:32.03 | [TK]D-Fender | hurdman, Go get real debug. |
| 15:32.13 | hurdman | i have no message, from kernel or other things, the ipmi see nothing |
| 15:32.16 | hurdman | the server juste reboot |
| 15:32.34 | hurdman | [TK]D-Fender: i search since 3 days ^^" i have done real debug |
| 15:32.53 | hurdman | but nothing to see .... with strace, gdb , no core dump .... nothing, only a reboot |
| 15:33.01 | [TK]D-Fender | hurdman, nothing to show = nothing to help you with. |
| 15:33.13 | hurdman | snif |
| 15:38.23 | *** join/#asterisk navaismo (~Airo@189.191.94.52) |
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| 16:07.23 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 16:07.37 | cusco | [TK]D-Fender: meaning if I set those variables first in the call file, and re-set them in the local channel, I can inherance them in the second channel? |
| 16:08.04 | [TK]D-Fender | You shouldn't have to reset them |
| 16:08.10 | [TK]D-Fender | STANDARD inheritance should carry |
| 16:09.21 | cusco | I mean, re-set because the values I want to set are only available in the first channel dialplan. I noticed that any var that I set in the call file will be available in both channels... |
| 16:09.23 | *** join/#asterisk zamba (marius@flage.org) |
| 16:09.46 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:10.42 | *** join/#asterisk seraphie (~erin@76.73.167.231) |
| 16:11.10 | [TK]D-Fender | cusco, Shouldn't be. |
| 16:11.34 | *** join/#asterisk ulogic (421fc7ab@gateway/web/freenode/ip.66.31.199.171) |
| 16:13.01 | *** join/#asterisk aboven (~aboven@host-85-201-12-229.brutele.be) |
| 16:13.46 | aboven | What does "directmedia" mean (asterisk 1.8) ? |
| 16:14.33 | [TK]D-Fender | aboven, says the device is allowed to reinvite to a compatible other leg of the call directly so audio doesn't pass through your server |
| 16:19.58 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 16:20.23 | aboven | If directmedia has a wrong configuration, is it possible that I've "no audio" with a DID in early media ? |
| 16:25.53 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 16:25.57 | [TK]D-Fender | aboven, If the networking on each end is not righ or * isn't right then yes that can be responsible. |
| 16:26.18 | aboven | thx |
| 16:26.26 | [TK]D-Fender | aboven, Prove that it is works without allowing it first |
| 16:26.38 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 16:29.37 | aboven | Is there a website where it's possible to find a minimal configuration of asterisk ? |
| 16:30.06 | jmetro | exten => _X.,1,Dial(SIP/provider.com/+1${EXTEN},30,TtkKM(somemacro)) |
| 16:30.06 | jmetro | Hm.. Could I remove the TtkKm(somemacro) and this dial function and have this dial still work? |
| 16:30.28 | jmetro | in this* not and this |
| 16:36.42 | *** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com) |
| 16:36.58 | Katty | runs through with knitting needles |
| 16:37.22 | *** join/#asterisk ironm (~ironm@84-73-66-195.dclient.hispeed.ch) |
| 16:38.43 | Katty | carrar: ESPRESSO |
| 16:38.47 | Katty | carrar: !!! |
| 16:39.00 | jpsharp | As my wife says "I have two knitting needles, you have two eyes. Coincidence?" |
| 16:39.00 | Katty | boingboing, boingboingboing.... boing. |
| 16:39.12 | Katty | jpsharp: i approve of your wife. |
| 16:40.22 | jmetro | <PROTECTED> |
| 16:40.56 | jpsharp | Yes. Basic dial is Dial(SIP/foo). No options are "required". |
| 16:41.02 | jmetro | Awesome. =D |
| 16:41.43 | Katty | Dial(CAN/jpsharp) |
| 16:41.52 | Katty | shakes the string a bit |
| 16:42.11 | jmetro | same=> Hangup |
| 16:42.22 | jpsharp | get soup in the ear, having forgot to clean the can. |
| 16:42.54 | Katty | that's unfortunate. |
| 16:49.42 | *** join/#asterisk TimeRider (~steve@178.147.83.52) |
| 16:50.14 | ledoktre | Greetings, anyone here able to offer some thoughts / advice on ways to possibly setup the following. Calls coming in to * box ring 4 select phones, but ANY phone can pick up call (if the 4 people are away/out/on phone). |
| 16:50.54 | jpsharp | Pickup groups. |
| 16:53.02 | jpsharp | Or directed call pickup. |
| 16:53.02 | jmetro | You can have it dial Sip/Extension@context/Extension@Context/Extension@context |
| 16:53.34 | jpsharp | That gets the phones ringing, but he wants to any phone to be able to answer the call if those 4 phones are unmanned. |
| 16:53.40 | ledoktre | jpsharp: I read about that, and I must have misunderstood it. That allows you to have any incoming call answerable by anyone, but only a few selected phones actually ring? |
| 16:53.54 | jmetro | Oh, i thought he meant only the 4 phones incase the other 3 were off. |
| 16:54.02 | ledoktre | Yes, I don't want all phones ringing. I am already ringing phones in that manner for the 4 users |
| 16:54.24 | ledoktre | There are 14 total actually. I just need the 4 to ring, but anyone can pick up if 4 are unavailable |
| 16:55.11 | ledoktre | reading here : http://www.freepbx.org/support/documentation/howtos/how-to-use-callgroups-and-pickgroups |
| 16:55.59 | *** join/#asterisk Praise- (~Fat@unaffiliated/praise) |
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| 17:07.32 | *** join/#asterisk n8ideas (~joshua@65.112.207.3) |
| 17:09.20 | drmessano | Anyone using res_xmpp in Asterisk 11? I can't seem to authenticate my Google clients |
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| 17:13.46 | *** join/#asterisk Dovid (~dovid@float-gw.core-fw-t.mypbxmanager.net) |
| 17:17.41 | ledoktre | jpsharp: thanks for the lead. I will test it out here in a bit. |
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| 17:28.46 | n8ideas | anyone have any thoughts on ODBC voicemail performance? |
| 17:29.05 | n8ideas | getting a bunch of problems with "app_voicemail.c: SQL Direct Execute failed!" |
| 17:29.13 | n8ideas | which basically means the voicemail disappears inot the ether |
| 17:30.05 | jpsharp | storing voicemail files in a database blob makes me twitch. |
| 17:30.20 | n8ideas | Yeah... I know |
| 17:30.34 | n8ideas | but for geographically dispersed systems and HA, it's pretty much the only option |
| 17:30.42 | n8ideas | the code is extremely brittle, though |
| 17:30.55 | n8ideas | if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) |
| 17:31.07 | n8ideas | if that fails, there's no retry logic... it just bombs and the voicemail is gone |
| 17:31.12 | jpsharp | Maybe store them in IMAP rather than ODBC? |
| 17:31.34 | n8ideas | not perfect for my model, and it would seem it might have the same underlying issue |
| 17:33.58 | n8ideas | seems also like some form of retry logic or a better sense of how to optimize database writes would be helpful too |
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| 18:04.25 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 19:12.42 | tm1000 | chirp chirp |
| 19:12.50 | tm1000 | listens to tumbleweeds |
| 19:14.31 | *** join/#asterisk sustav (vpp@conference/digium/x-yifnajksxenmaftl) |
| 19:14.57 | jmetro | Note: There is a special level of hell reseved for you, if 0 is not a valid option on your auto attendant.. |
| 19:24.48 | jpsharp | I've had a couple of auto attendants that hang up if you dial 0. |
| 19:26.57 | jmetro | Awful |
| 19:28.23 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 19:30.31 | cusco | what is auto attendant? |
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| 19:45.27 | *** join/#asterisk troutowi (~troutowi@141.216.135.29) |
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| 19:47.58 | troutowi | hi, im having an issuw with Asterisk 10.9. I installed from source, then set asterisk to run as a non root user, set all permissions to folders, etc. asterisk starts as the non root user. but if i try to go and get into the CLI, by 'sudo asterisk -r' , it tells me im connected, but immediately takes me back to the bash shell |
| 19:49.37 | troutowi | and running 'sudo asterisk -rx "COMMAND" ' works correctly, except for sip commands, which asterisk says does not exist |
| 19:54.29 | [TK]D-Fender | I would check to see if * isn't crashing in circles..... |
| 19:54.48 | [TK]D-Fender | maybe that's why you can connect long enough to get a "blip" but nothing "survives". |
| 19:55.06 | [TK]D-Fender | And would explain why SIP connecds don't work ... probably doesn't live long enough to even load... |
| 19:55.18 | jpsharp | Or chan_sip.so didn't build. |
| 19:55.57 | troutowi | if i try 'asterisk -r' as root, ill get a Illegal instruction (core dumped) |
| 19:56.31 | [TK]D-Fender | troutowi, stop the daemon process and try running it by hand |
| 19:56.40 | [TK]D-Fender | See what it warns you about |
| 19:57.38 | troutowi | 'sudo asterisk -vvvvvvc' : |
| 19:57.39 | troutowi | <PROTECTED> |
| 19:57.39 | troutowi | <PROTECTED> |
| 19:58.16 | [TK]D-Fender | ~pb |
| 19:58.16 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:58.17 | [TK]D-Fender | ^^^^ |
| 19:58.21 | [TK]D-Fender | don't flood in here |
| 19:58.26 | troutowi | ok |
| 19:58.55 | troutowi | http://pastebin.com/nrUkSxS8 |
| 20:00.56 | troutowi | http://pastebin.com/C2W6JUkS |
| 20:01.05 | [TK]D-Fender | troutowi, "asterisk -gvvvvvvvvvc" |
| 20:01.06 | troutowi | ^ the permissions and chown's i performed |
| 20:01.40 | [TK]D-Fender | chown -R root:asterisk /etc/asterisk |
| 20:01.48 | [TK]D-Fender | why is etc ROOT there? |
| 20:01.51 | [TK]D-Fender | Shouldn't be.... |
| 20:01.55 | troutowi | its not |
| 20:01.59 | troutowi | i changed after the fact |
| 20:02.05 | troutowi | it is asterisk:asterisk |
| 20:02.22 | troutowi | [TK]D-Fender, asterisk -gvvvvvvvvvc had the same output |
| 20:03.38 | [TK]D-Fender | troutowi, "asterisk -U asterisk -G asterisk -gvvvvvvvvvc" |
| 20:04.03 | [TK]D-Fender | And check that you have a proper asterisk.conf, and so forth in your etc folder |
| 20:04.20 | [TK]D-Fender | PB it while you're at it.... and "ls -la /etc/asterisk" |
| 20:05.02 | *** join/#asterisk TimeRider (~steve@109.178.207.161) |
| 20:08.02 | troutowi | http://pastebin.com/McFevBU4 |
| 20:08.12 | *** join/#asterisk [CYB]iDeFiX (admin@conference/digium/x-lwqdvepshgtymekd) |
| 20:09.09 | troutowi | one second, for the conf files |
| 20:11.31 | troutowi | http://pastebin.com/A8kq1xDP < asterisk.conf |
| 20:12.12 | jmls1 | anyone got any comments about the Cisco 79xxG range of phones ? Do they work well with asterisk / sip ? We have used a load of the 7940/7960 and although extremely reliable, they don't do certain things like auto-answer if specified in a sip-header etc |
| 20:12.25 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 20:13.06 | drmessano | They sure don't support all features in SIP |
| 20:13.14 | troutowi | [TK]D-Fender, the extconfig.conf is all commented out, never been touched, would u still like to see it |
| 20:14.18 | jpsharp | jmls1: I've used several hundred 7940s with Asterisk and they've worked well, other than the auto-answer issue. |
| 20:14.32 | jmls1 | drmessano: even the latest 79x5G models ? |
| 20:15.06 | jmls1 | jpsharp: yeah, we're very happy with the 7940's, that's why I was kinda hoping the later versions had better sip support |
| 20:15.39 | [TK]D-Fender | troutowi, [directories](!) <- remove the (!) and retest |
| 20:16.03 | *** join/#asterisk Galen_ (~Galen@rrcs-76-79-170-42.west.biz.rr.com) |
| 20:16.13 | jmls1 | on that note, has anyone got a problem with 7940 and asterisk-11 with call hold ? It often takes 4-5 presses to get a call back from hold, if at all |
| 20:16.19 | troutowi | ok |
| 20:17.43 | troutowi | same result [TK]D-Fender |
| 20:17.48 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 20:17.48 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 20:18.04 | [TK]D-Fender | troutowi, modules.conf please |
| 20:18.22 | [TK]D-Fender | And following that please show the latest attempt |
| 20:18.27 | [TK]D-Fender | (include in the PB) |
| 20:18.30 | troutowi | will do |
| 20:19.22 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 20:20.41 | troutowi | http://pastebin.com/XgFZvLkh |
| 20:22.24 | [TK]D-Fender | Illegal instruction (core dumped) |
| 20:22.30 | [TK]D-Fender | ouch... yeah, that's kinda fatal.. |
| 20:22.37 | troutowi | yeah... |
| 20:23.11 | drmessano | jmls1: There is definitely a list of features they only support on SCCP. Even newer models. By design |
| 20:23.18 | [TK]D-Fender | I'd recommend doing a tarball extract & fresh compile |
| 20:23.55 | troutowi | any special flags for configure? ive just been using what it picks for me |
| 20:23.57 | jmls1 | drmessano: understood. Do you know if they support auto-answer on sip header ? |
| 20:24.01 | [TK]D-Fender | hold on... |
| 20:24.19 | drmessano | jmls1: No idea. I wouldn't expect so. The feature set is pretty standard |
| 20:24.46 | jmls1 | dammit. even aastra phones support sip-header auto-answer ;) |
| 20:24.57 | [TK]D-Fender | troutowi, Nope..... fresh out of ideas right now.... |
| 20:25.03 | drmessano | Cisco isn't in the business of making phones for generic SIP PBX'es |
| 20:25.08 | [TK]D-Fender | troutowi, Keep those PB's handy for others to reference.... |
| 20:25.21 | troutowi | ok, appreciate the help [TK]D-Fender |
| 20:25.45 | drmessano | Cisco makes phones for CCM and their Small Business PBX's, but they ALSO offer SIP firmware so the phones are useful elsewhere |
| 20:26.02 | drmessano | They are not trying to compete with other SIP phones |
| 20:26.15 | jpsharp | Technically, you have to buy an additional license to use SIP on their phones. |
| 20:26.45 | jmls1 | jpsharp: yup |
| 20:27.08 | [TK]D-Fender | troutowi, Go check /usr/lib/asterisk/modules to see if perms along the way lok funky or something snuck in that shouldn't have |
| 20:27.09 | drmessano | A Cisco phone is really a stupid purchase for a SIP endpoint. You can get a fully supported, feature-rich SIP endpoint for 25% on a comparable feature-set |
| 20:27.26 | drmessano | Polycom, Aastra, etc |
| 20:27.55 | jmls1 | drmessano: we've tried several brands, and the cisco has outlasted all of them, despite being the oldest model |
| 20:27.57 | n8ideas | anyone had any experience with optimizing mysql for realtime voicemail? |
| 20:28.14 | jmetro | polycoms ftw |
| 20:28.15 | jmls1 | every single one of our Aastra phones have broken (50+) |
| 20:28.28 | [TK]D-Fender | "broken"? |
| 20:28.40 | jmls1 | physically broken |
| 20:28.55 | jpsharp | I'd love to have a 7940 on my desk. |
| 20:28.58 | drmessano | Cisco are great phones. They just make for a shitty SIP endpoint |
| 20:28.59 | [TK]D-Fender | WTF are you doing to these ppor phones? |
| 20:29.10 | jmls1 | [TK]D-Fender: don't blame me ! |
| 20:29.14 | jmls1 | it's our bloody agents |
| 20:29.20 | troutowi | [TK]D-Fender, all modules are 750, and owned by asterisk |
| 20:29.24 | jmetro | i have a 10 year old aastra that still pairs with 10 yaer old handsets.. |
| 20:29.50 | [TK]D-Fender | Ok, I'd recommend a fresh compile from a freshly extracted tarball. |
| 20:29.59 | jmls1 | the d40 is nice, just don't know about longevity |
| 20:30.03 | [TK]D-Fender | And ... it's checkout time here, BBIAB |
| 20:30.10 | troutowi | later |
| 20:30.15 | drmessano | ..and without a full spec on SCCP, you can only get so much using SCCP/Skinny firmware on a Cisco |
| 20:30.23 | drmessano | So Cisco is a bad choice |
| 20:30.25 | [TK]D-Fender | troutowi, Keep at it.... |
| 20:30.32 | troutowi | will do :) |
| 20:30.48 | jmls1 | we have a couple of polycoms in for eval |
| 20:31.30 | jmls1 | the snom was discounted because it "looks like an ikea phone". :P |
| 20:31.40 | drmessano | Snom's are great phones |
| 20:31.49 | jmls1 | so I've heard ! |
| 20:32.01 | drmessano | Very sexy feature set for the price. |
| 20:33.19 | drmessano | ~polycommunist |
| 20:33.19 | infobot | A polycommunist is someone who believes Polycom phones can do no wrong... that Polycoms are so over and above anything else that what you are using is surely crap, and the mere fact you mention another brand name is of great insult to the channel, the community, and the world. They may also be getting a 10% kickback. |
| 20:33.31 | jmls1 | lol |
| 20:33.39 | drmessano | I wrote that like 5 years ago |
| 20:33.44 | drmessano | But Polycoms are awesome |
| 20:34.23 | drmessano | My IP331 made me breakfast on Saturday |
| 20:34.27 | jmetro | i just wish aastra handsets didnt fall off the freaking base when you tap the desk |
| 20:34.37 | drmessano | They dont have a hook? |
| 20:35.08 | drmessano | I flipped all the hooks on the Polycoms we have here. Not one disaster yet |
| 20:35.10 | jmetro | ive got a 6757i, it might as well be made of ice, attached to an ice phone, with an ice cord. |
| 20:35.26 | jmls1 | 9133i. pos. |
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| 21:49.34 | *** join/#asterisk AkkerKid (~AkkerKid@50-73-146-237-ip-static.hfc.comcastbusiness.net) |
| 21:49.55 | AkkerKid | Hi! Anyone know how to create a DNS SRV record for VPNs? |
| 21:51.59 | *** join/#asterisk rdegges (u4891@gateway/web/irccloud.com/x-ttkprdgxjeqfqymt) |
| 21:52.24 | rdegges | Sup! |
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| 22:05.09 | *** part/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
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| 22:21.09 | jeffspeff | anybody have any experience using cisco 8945 phones? |
| 22:21.40 | cusco | hmm I have a linksys near me |
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| 23:36.26 | ChannelZ | is Linksys 9xx |
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