| 00:00.08 | Ritzerisk | tricky but i have no idea how to keep the dtmf digits in memory and save it as the filename |
| 00:01.26 | silvertip257 | hello [TK]D-Fender - what's up? Do you [anyone in here is welcome to add] have time for me to explain my WRT54/asterisk + Ekiga situation? Thanks. |
| 00:02.51 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 00:03.02 | angryuser | silvertip257 ekiga it's like openwrt ? |
| 00:03.25 | silvertip257 | angryuser: ekiga is a Linux softphone |
| 00:03.51 | silvertip257 | angryuser: http://www.gnomemeeting.org/ |
| 00:03.54 | *** join/#asterisk implicit (n=bayan@ip68-105-71-13.sd.sd.cox.net) |
| 00:03.57 | angryuser | oh sorry sounded like one distrib i knew |
| 00:04.24 | angryuser | my bad, what it the problem ? |
| 00:04.36 | silvertip257 | no problem - I'm trying to get the Ekiga softphone client to register with Asterisk |
| 00:05.13 | angryuser | register it then ;=) |
| 00:05.24 | silvertip257 | I have confirmed the * PBX is listening on the right ports and passwords/usernames are right |
| 00:05.33 | silvertip257 | angryuser: har har har |
| 00:05.37 | angryuser | no cli output ? |
| 00:05.52 | angryuser | you said 'no problem' |
| 00:05.53 | lanning | iax or sip? |
| 00:05.58 | silvertip257 | angryuser: no sip debug output when I use ekiga to try and register |
| 00:06.14 | angryuser | no traffic at all ? |
| 00:06.49 | silvertip257 | angryuser: I'm running wireshark (on the client - just to confirm what's going on) - CLI doesn't say I have traffic and debug+verbose = 3 |
| 00:07.24 | angryuser | and on ekiga side ? Sip packets are sent ? |
| 00:07.53 | silvertip257 | runs more tests |
| 00:08.12 | silvertip257 | falls over dead |
| 00:08.24 | *** join/#asterisk coppice (n=chatzill@27.202.17.210.dyn.pacific.net.hk) |
| 00:09.02 | silvertip257 | mutters ${obscenities} ... angryuser what magical powers do you possess? |
| 00:09.37 | angryuser | it's hypnose over vpn direct into you heads |
| 00:09.56 | angryuser | :) |
| 00:10.17 | silvertip257 | angryuser: hehe ... it reg'd |
| 00:10.38 | angryuser | have fun |
| 00:10.41 | silvertip257 | what in the world .. must have been the firewall rules I changed earlier ... but it still didnt register until you said try again |
| 00:11.00 | silvertip257 | angryuser: thanks for being here - good to have someone to talk to while this * PBX makes an ass outta me |
| 00:14.07 | angryuser | it is better than hitting my head with exchange wall for 5 hours, little offtopic, what do you prefer for a lot of users, 3 party or exchange ? |
| 00:14.30 | angryuser | lot is 150 for me |
| 00:14.43 | jaytee | angryuser, you using Exchange UM with *? |
| 00:14.47 | silvertip257 | angryuser: I don't have much of an opinion - I'm more of a networking guy, not a teleco engineer |
| 00:15.30 | angryuser | jaytee nope, not yet but *in project* ad+ex+* |
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| 00:35.11 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 00:36.26 | angryuser | im off |
| 00:37.25 | phix | :D |
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| 00:52.10 | Sparkie- | anyone around with experience using chan_skinny ? |
| 01:01.53 | TrentCreek | fat |
| 01:13.06 | *** join/#asterisk Kira (n=chatzill@210.176.243.218) |
| 01:13.47 | Kira | Hello. |
| 01:14.24 | jaytee | hi |
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| 01:15.17 | Kira | I am new to VoIP, SIP, etc. but I am becoming the infrastructure support of my workplace. |
| 01:15.53 | Kira | So the first thing I would like to ask is, what is the relationship between Asterisk, Trixbox, and FreePBX? |
| 01:16.18 | JT | freepbx is a horrible configuration gui for asterisk |
| 01:16.28 | JT | trixbox is centos + freepbx + a couple of other things |
| 01:17.27 | Kira | Ah. |
| 01:17.58 | jaytee | Kira, underneath the GUI in Freepbx and Trixbox and a few other spinoffs is the core which is Asterisk. |
| 01:18.15 | bobbym | is there any logic why my providers can get the destination number that i want to call from the Dial(SIP/destination@provider) command? they are asking me to change the FROM field in sip so they can complete the call: |
| 01:18.17 | bobbym | ? |
| 01:18.38 | Kira | So if there is anything wrong with my PBX server, there are like 3 ways to solve things. |
| 01:18.45 | bobbym | i mean they want me to put the destination in the from field? |
| 01:19.10 | Kira | So if there is anything wrong with my PBX server, there are like 3 ways to solve things. |
| 01:19.11 | jaytee | Kira, no. Three points of failure is more likely. |
| 01:19.18 | Kira | oops |
| 01:19.23 | Kira | :S |
| 01:19.27 | Kira | Anyway. |
| 01:20.31 | jaytee | Asterisk configuration is done using .conf files which are just text files that get parsed when Asterisk loads. Freepbx glues a MySQL database on top of that with a web based gui and sort of locks down the flexibility of the dialplan. |
| 01:20.59 | Kira | I have a Trixbox machine that used to be behind NAT firewall (the firewall used to forward one of our public static IPs to the LAN IP of the Trixbox machine). |
| 01:21.10 | jaytee | and Trixbox adds a few more bells and whistles (some of which are nice features to have when they actually work) |
| 01:21.29 | Kira | We now want to move the Trixbox directly to our broadband connection. |
| 01:21.52 | Kira | I have changed a few configuration files until our SIP phones can successfully register with the server. |
| 01:22.07 | Kira | However, when we try to make calls, we get the message that all circuits are busy. |
| 01:22.10 | jaytee | you want to connect Asterisk from your DMZ? why? |
| 01:23.19 | Kira | jaytee: it used to be in the DMZ anyway (the firewall used to forward all request for that public IP to the NAT IP). |
| 01:23.58 | jaytee | with no other port protection if it's in the DMZ |
| 01:24.11 | Kira | no port protection |
| 01:24.33 | jaytee | kinda makes ya vulnerable in more ways than just using SIP/NAT |
| 01:25.01 | *** join/#asterisk RoyK (n=roy@ip-157-60-149-91.dialup.ice.no) |
| 01:25.59 | jaytee | as far as the all circuits are busy problem I wouldn't know where to start since it's a Trixbox. |
| 01:26.44 | Kira | jaytee: probably, but SonicWALL doesn't seem to support port forwarding of secondary WAN IPs. |
| 01:26.56 | jaytee | but if it used to be behind a NAT'd firewall you might want to reference this to see if there's something that will help you reverse things. |
| 01:27.03 | jaytee | ~sipnat |
| 01:27.03 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 01:29.23 | jaytee | and you might try asking in the #trixbox channel cuz most people in here only run Asterisk. |
| 01:29.57 | jaytee | although the #trixbox channel is usually dead or full of zombie accounts |
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| 01:32.29 | Kira | jaytee: aah |
| 01:32.31 | Kira | I see |
| 01:32.35 | Kira | Thanks. :) |
| 01:32.41 | JT | as if a sonicwall can't port forward |
| 01:32.59 | JT | any router/nat device that cannot port forward has one rightful place: the bin |
| 01:33.00 | *** join/#asterisk AJayMN (n=me@72-61-170-183.area2.spcsdns.net) |
| 01:33.28 | Kira | JT: It looks like SonicWALL can port forward for the *primary* WAN IP, but not for secondary WAN IPs. |
| 01:33.39 | JT | you should port forward and forget about silly router "DMZ" IPs |
| 01:33.52 | JT | s/IPs/modes/ |
| 01:34.20 | JT | if that's definitely the case, you should get a different router |
| 01:36.10 | jaytee | our Cisco PIX can forward any of our outside static IP addresses. I can't believe SonicWALL would sell a product like that that can't. Kinda lame. |
| 01:36.26 | JT | i think they run linux |
| 01:36.27 | *** part/#asterisk silvertip257 (n=chatzill@unaffiliated/silvertip257) |
| 01:36.33 | JT | and i know linux definitely can do it |
| 01:37.06 | jaytee | yeah, well what's a the core of IOS? a super tightly stripped down version of the Unix kernel. |
| 01:37.21 | Kira | It *does* matter which WAN IP I let the SIP server use, right? (I don't mean just having to tell every SIP client to connect to a new IP; I mean, if I put the SIP server on a different IP, it might simply NOT work because the VoIP service provider is expecting me to come from a particular WAN IP) |
| 01:37.47 | JT | jaytee: which unix kernel? |
| 01:38.31 | JT | can't you simply update the voip provider as to the new ip if they even need to be personally notified at all? |
| 01:38.40 | jaytee | I think it's a custom SRV5 the same as Nortel uses in their Meridian systems |
| 01:40.35 | JT | also, this is a very good argument for using DNS, re your sip clients |
| 01:45.30 | AJayMN | i tried to delete the voicemail directory and it wont delete.. im getting rm: cannot lstat 'voicemail/device/201': No such file or directory |
| 01:45.48 | AJayMN | yet if i goto voicemial/device/ and do a ls there is no directory 201 listed |
| 01:45.55 | AJayMN | is this a problem due to symbolic links? |
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| 02:08.40 | Yourname`` | Does anyone know what timezone the Vitelity CDRs are from? |
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| 02:26.41 | TedNJ38 | Does anyone know what ports I should forward to my linux box if I want to pull the xml files for my phone from outside my network? |
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| 02:48.43 | _ShrikE | ~seen khronos |
| 02:48.47 | jbot | khronos <n=khronos@aquaman.perryinstitute.org> was last seen on IRC in channel #asterisk, 14d 15h 57m 51s ago, saying: 'K.'. |
| 02:49.00 | `Sean | Anyone here using Cisco 7970s? |
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| 03:12.25 | nadio | Does asterisk support other protocols? |
| 03:12.41 | nadio | for example jabber/msn ? |
| 03:12.47 | nadio | or even skype |
| 03:12.52 | russellb | jabber, yes |
| 03:12.54 | russellb | the others, no |
| 03:13.29 | C4colo | 1.6 might be able to be hacked to support yahoo voice |
| 03:13.58 | C4colo | as it is sip over tcpip instead of udp |
| 03:14.11 | C4colo | the audio stream is still udp, but the sip signalling is tcp |
| 03:14.23 | nadio | ok |
| 03:14.41 | TJNII | hacked? Doesn't * just support sip over TCP/IP? |
| 03:14.56 | russellb | that doesn't mean it interoperates with yahoo ... |
| 03:15.02 | russellb | SIP is a very ... loose "standard" |
| 03:15.07 | C4colo | I saw the settings in 1.6 for the first time |
| 03:15.12 | C4colo | never saw them in 1.4 |
| 03:15.20 | russellb | it's new in 1.6 .. |
| 03:15.55 | russellb | http://svn.digium.com/view/asterisk/trunk/CHANGES?view=markup ... check that out for more new things in 1.6 |
| 03:15.56 | TJNII | I thought sip signalling over TCP support was older than that. Well, OK, then. |
| 03:19.02 | JT | TJNII: btw you shouldn't refer to TCP as tcpip |
| 03:19.14 | JT | tcpip is usually used in a much more broad sense |
| 03:19.26 | CoffeeIV | I'm installing the latest asterisk (1.4.21.1) on CentOS 4.3. WHen I do "make menuconfig" it keeps telling me I need ncurses, but I have installed ncurses and ncurses-devel and re-did the configure commands, and did a make clean -- any ideas ? |
| 03:24.32 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135) |
| 03:26.18 | CoffeeIV | the answer to my problem, was to do "make distclean" and then re-do the configure and make menuconfig steps |
| 03:32.22 | MikeJ | yahoo is over tls not tcp |
| 03:42.38 | [TK]D-Fender | TedNJ38: Whatever port the protocol that is used to retrieve them uses. I'm sure google can answer that in about 2 seconds flat |
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| 04:00.52 | marc7 | where could I purchase a G.729 codec that I could load into asterisk? |
| 04:01.34 | Juggie | ~g729 |
| 04:01.37 | jbot | [~g729] G.729(.a /.ab /.b) is a patent-encumbered ITU-standard voice codec operating at 8kbps offering quality similar to GSM. For Asterisk to transcode G.729 licenses (per channel) must be bought from http://store.digium.com |
| 04:02.01 | marc7 | thank-ee |
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| 04:27.12 | kamanashisroy | What is the meaning of forking cdr if it has the same uniqueid ! |
| 04:30.03 | kamanashisroy | ?? |
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| 04:44.27 | marc7 | kamanashisroy: starts the timer over again? |
| 04:44.58 | kamanashisroy | marc7: yes .. But I expected to use it for callback billing :( |
| 04:46.24 | marc7 | i wonder if there's a club for all of us who have had our hearts broken by the notion that we'd be able to use this CDR table for some good, when in reality, the only thing we can do is build our own billing system. |
| 04:47.54 | kamanashisroy | :) .. the truth is always rude .. |
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| 06:25.14 | ionix | hey, can I bridge two zaptel channel without doing any kind of transcoding? On a TDM401 |
| 06:25.49 | ionix | Basically I want to answer all incoming calls, if it's a fax, send it to a fax via an analog hole, else in the IVR |
| 06:26.17 | ionix | and same for the fax. Enable the fax to dial directly to the outside line, without asterisk transcoding or encoding anything |
| 06:33.21 | dlynes | ionix: no transcoding is involved there..it's a bridge |
| 06:38.41 | ionix | yes that's what I want |
| 06:39.02 | *** part/#asterisk Kira (n=chatzill@210.176.243.218) |
| 06:39.07 | ionix | but since I want asterisk to answer first and check if it's a fax, can I then just bridge the two zaptel channels? |
| 06:39.43 | ionix | in the extensions.conf, I couldn't fine how to bridge channels. Do I just Call() it? |
| 06:40.30 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
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| 06:54.50 | dominic1 | does anybody out there here me'? |
| 06:54.53 | dominic1 | hear |
| 06:55.25 | C4colo | no, but I can see your text |
| 06:55.33 | gnorbert | :) |
| 06:56.07 | dominic1 | good answer, short question: Is it possible to see that a call was transferred by the internal function of a phone? |
| 06:56.20 | C4colo | 0.o |
| 06:56.29 | C4colo | you want to see in the CDR info? |
| 06:56.47 | C4colo | or in the settings of the phone? |
| 06:56.53 | C4colo | or capture it in the dialplan? |
| 06:57.21 | dominic1 | if a call has been transferred by the phone, I want to set specific parameters in asterisk. |
| 06:57.51 | dominic1 | I need that to tag a bounced back unattended transfer |
| 06:58.58 | dominic1 | external caller calls person1, person1 initiates a unnatended transfer with the phone to person2, person2 doesn't answer and the call will bounce back to person1 with the callerid(name) UNANSWERED |
| 07:01.33 | gnorbert | Hi, I should play a sound file in a meetme conference, and I already know, that it can be solved by call files. But how can I play a sound file from a call file? Playback seems doesn't work for me. |
| 07:04.28 | dlynes | ionix: it'll still get bridged |
| 07:04.39 | dlynes | ionix: Just do a Dial() |
| 07:05.01 | *** join/#asterisk Keypad (n=Keypad@125-238-132-39.broadband-telecom.global-gateway.net.nz) |
| 07:05.23 | gnorbert | http://pastebin.com/d2e208ee2 |
| 07:05.38 | Keypad | Any one able to help a complete astricks noob ? |
| 07:06.10 | dlynes | Keypad: just ask first...we don't know if we can help, if you don't tell us what it is you're asking |
| 07:06.15 | gnorbert | If you're really noob, then maybe, but I afraid, you know even more, then me. :) |
| 07:06.17 | Keypad | fair enough :) |
| 07:06.33 | Keypad | I want to connect my Linksys VOIP gateway to my astricks box |
| 07:06.46 | creativx | its asterisk |
| 07:06.49 | creativx | not tricks.. |
| 07:06.49 | creativx | :) |
| 07:06.51 | dlynes | Keypad: which gateway? Linksys PAP2-NA? |
| 07:07.20 | Keypad | Its a SPA3102 |
| 07:08.26 | dlynes | Keypad: Everything you ever wanted to know about configuring Linksys and Sipura voip devices can be found here: http://forum.voxilla.com/linksys-sipura-voip-support-forum/ |
| 07:08.53 | dlynes | Keypad: it's pretty much the master list of information for those devices |
| 07:11.10 | gnorbert | Does somebody know, how can I play sound file from a call file? |
| 07:11.15 | Keypad | Thanks for that I booked marked it |
| 07:12.37 | Keypad | I am a bit confused on how to start connecting these guys together |
| 07:12.50 | Keypad | Since its a router |
| 07:13.40 | Keypad | First thing id like is to be able to use my phone line from any where in the network using a softphone |
| 07:14.49 | dlynes | Keypad: 3102 has nothing to do with softphones...it's for analog phones |
| 07:15.30 | *** join/#asterisk damjan (n=damjan@217.16.95.15) |
| 07:15.39 | Keypad | I wanted to use the PSTN Line on it |
| 07:16.02 | Keypad | if thats possable |
| 07:16.27 | C4colo | if it has an fxo you can use it as a trunk on the pbx |
| 07:16.40 | C4colo | I have no idea how to configure this, just that I know it can be done |
| 07:17.05 | Keypad | Yeah It does |
| 07:17.15 | Keypad | I have no idea how its done as well :( |
| 07:17.21 | dlynes | Keypad: if the 3102 is like the sipura 3000, it'll have one fxo port (for plugging in your telco line), and one fxs port (for plugging in your analog telephone) |
| 07:17.47 | dlynes | Keypad: Just follow the instructions on that forum for setting up a Sipura 3000 in that case, if you can't find anything for setting up a Linksys SPA-3102 |
| 07:19.33 | dlynes | Keypad: it's a bit of a pain in the ass to get it to work with asterisk if it's like the sipura 3000, but it does work, and when you do have it working, it works like a charm |
| 07:19.42 | gnorbert | Can somebody help? :) |
| 07:20.01 | dlynes | gnorbert: if i actually used agi, I probably could |
| 07:20.11 | dlynes | gnorbert: but, unfortunately, i don't use agi |
| 07:20.21 | creativx | gnorbert: isnt this your third day of asking the same questions |
| 07:20.23 | Keypad | dlynes: Thats good :) |
| 07:20.58 | dlynes | Keypad: when i set up my sipura 3000 to work with asterisk, I followed the instructions on the voxilla forum, and they worked like a charm |
| 07:21.23 | gnorbert | creativx: Actually, it is, thanks for notice it. :) But I always got nearly the same answer, mainly nothing. :) |
| 07:21.25 | dlynes | Keypad: i misread the instructions the first time, however...so teh first time, they didn't work so well...but that was my fault |
| 07:21.34 | *** join/#asterisk kamh (n=q@host-81-190-236-85.wroclaw.mm.pl) |
| 07:21.58 | Keypad | dlynes: bad luck, lol. Should I be able to access my web gui for the device out of the internet port ? |
| 07:22.01 | dlynes | gnorbert: there's a few people on here during the day, US East Coast time that use AGI, if it helps |
| 07:22.21 | dlynes | Keypad: only if you have it enabled |
| 07:22.28 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 07:22.38 | *** join/#asterisk af_ (n=getsmart@88-149-230-143.dynamic.ngi.it) |
| 07:22.42 | dlynes | Keypad: and if it's like every other linksys device, it'll be port 8080 for external connections, not port 80 |
| 07:22.52 | Keypad | oh ? |
| 07:22.54 | gnorbert | dlynes: I hope it does. :) |
| 07:22.57 | dlynes | Keypad: and it might be https, not http |
| 07:23.01 | gnorbert | Thank you. |
| 07:23.40 | dlynes | gnorbert: otoh, you could always try reading the agi documentation too |
| 07:23.50 | dlynes | gnorbert: i think it mentions what all the commands are for agi |
| 07:24.18 | Keypad | dlynes: that dident work :S |
| 07:24.23 | Keypad | brb |
| 07:24.32 | dlynes | gnorbert: yeah...type 'agi show' at the cli |
| 07:24.38 | dlynes | gnorbert: it'll show you all the agi commands |
| 07:24.42 | gnorbert | Thanks. |
| 07:24.57 | creativx | gnorbert: yeah.. maybe you should try to look at different ways to reach your goal |
| 07:24.58 | creativx | =) |
| 07:25.36 | dlynes | gnorbert: so i would think 'stream file', or 'control stream file' would be what you're looking for |
| 07:25.39 | MCooper | I have a question about queues... right now I have queues working, and it rings all phones.. GREAT.. but it has MOH, and I would rather the customer here a ring instead. |
| 07:25.41 | gnorbert | creativx: I guess it's the third way. :) |
| 07:25.42 | MCooper | any ideas? |
| 07:26.01 | creativx | MCooper: get a ringing tone as moh |
| 07:26.23 | creativx | MCooper: record -> playback beep -> use output file as moh |
| 07:26.30 | MCooper | creativx, Yes I have thought about that... |
| 07:26.43 | creativx | MCooper: i ended up doing that. works like a charm |
| 07:26.48 | creativx | i cant remember why i had to do it that way |
| 07:26.51 | creativx | but it was something stupid |
| 07:26.52 | creativx | hehe |
| 07:26.53 | MCooper | creativx, Awesome...:) |
| 07:27.11 | creativx | its no problem getting asterisk to make your dial tone either |
| 07:27.12 | gnorbert | dlynes: Thanks, I make a try, but really that sounds the most hopefully. :) |
| 07:27.13 | creativx | and recording it |
| 07:27.23 | MCooper | creativx, well the clients over here... do not understand getting placed in a queue with moh instead of a ring. |
| 07:27.45 | creativx | MCooper: hehe, then give them a ring :) i can't stand moh actually.. a ring is much better |
| 07:28.10 | *** join/#asterisk nuonguy (n=john@c-24-6-187-202.hsd1.ca.comcast.net) |
| 07:28.30 | MCooper | creativx, Yep.. that is why I figured it would have been an option... but everything I read.. was well.... discouraging... |
| 07:28.31 | MCooper | hahha |
| 07:28.47 | creativx | hehe |
| 07:29.03 | dlynes | MCooper: rarely, if ever, do customers ever make sense |
| 07:29.26 | MCooper | dlynes, hahaha Now you are being cynical... :) |
| 07:29.26 | creativx | except they tend to sponsor your growing wealth |
| 07:29.27 | dlynes | oops...one too many evers in there :) |
| 07:29.43 | dlynes | creativx: nah...they sponsor your growing poverty |
| 07:29.52 | dlynes | creativx: they always want something for nothing, or close to nothing |
| 07:30.03 | creativx | then you have the wrong kind of customers :-) |
| 07:30.09 | MCooper | dlynes, well not with what I am doing.. but its still a dangerous world here... |
| 07:30.11 | dlynes | creativx: when you give them a bill, they always bitch and whine and complain |
| 07:30.26 | creativx | dlynes: hehe.. not our customers :) |
| 07:30.37 | dlynes | creativx: you live in a utopia :) |
| 07:31.01 | creativx | dlynes: norway actually, but that's somewhat the same.. he he j/k |
| 07:31.04 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 07:31.05 | MCooper | is jealous of creativx |
| 07:31.21 | dlynes | Frankly, I probably like Vancouver better :) |
| 07:31.27 | dlynes | but probably cause I'm from here :) |
| 07:31.41 | creativx | i guess they pay up because if they dont we cut their service off.. no monatas no software, no software no income for them |
| 07:31.54 | MCooper | dlynes, I am from Seattle.. but work in Baghdad |
| 07:32.01 | MCooper | hahah Long daily commute... |
| 07:32.03 | dlynes | MCooper: dood |
| 07:32.08 | creativx | heheh |
| 07:32.13 | MCooper | dlynes, hahaha |
| 07:32.18 | dlynes | MCooper: You took a wrong turn at Albuquerque |
| 07:32.29 | MCooper | dlynes, Yep.. and ended up in a war zone.. |
| 07:32.56 | MCooper | Man... but things here are really improving.. contrary to what the media would like to say... |
| 07:33.35 | dlynes | Didn't think they were improving or getting worse |
| 07:33.37 | dlynes | Just staying the same |
| 07:33.53 | creativx | still hot i assume |
| 07:34.05 | MCooper | dlynes, Getting a lot better... |
| 07:34.06 | gnorbert | "[Jul 22 09:33:30] WARNING[28796]: pbx_spool.c:245 apply_outgoing: Unknown keyword 'stream file' at line 5 of /var/spool/asterisk/outgoing/proba.call" :) |
| 07:34.11 | MCooper | creativx, 122 the outher day... |
| 07:34.14 | MCooper | other |
| 07:34.37 | MCooper | stream file... interesting... |
| 07:35.03 | MCooper | creativx, thanks for the solution.. i will use it... it would be the simplest. |
| 07:35.06 | dlynes | gnorbert: I just gave you an idea |
| 07:35.16 | dlynes | gnorbert: you need to do some legwork to figure out what to do with it |
| 07:35.28 | dlynes | gnorbert: Like I said before, I've never used AGI |
| 07:35.28 | creativx | MCooper: it works fine here with us.. you cant hear its actually moh either |
| 07:35.39 | gnorbert | I know, just wanted to give myself a reason to continue asking. :) |
| 07:36.13 | dlynes | gnorbert: but maybe if you pastebin your code, it'll be something blatantly obvious |
| 07:36.21 | gnorbert | Before somebody tells me after my next question, that there was already an answer on this. :) |
| 07:37.00 | dlynes | gnorbert: actually, wait just a cotton pickin' minute |
| 07:37.09 | dlynes | gnorbert: this is in a call file, not an AGI script |
| 07:37.20 | dlynes | gnorbert: of course it's an unknown keyword |
| 07:37.44 | gnorbert | Yes, just I thought, it's the same, you were so sure in it. :) |
| 07:38.00 | dlynes | gnorbert: how do you expect an agi keyword to work in a call file? |
| 07:38.12 | gnorbert | ^ |
| 07:38.23 | kamanashisroy | CDR(amaflags)=BILL; is not working in ael .. any clue ? after doint it when I do verbose(CDR(amaflags)) .. I get DOCUMENTATION |
| 07:39.22 | dlynes | gnorbert: yeah, but a call file can connect you to another channel, and when you connect yourself to that channel, you can use the Playback() application to play the sound |
| 07:39.32 | dlynes | gnorbert: i didn't realize you were using call files |
| 07:40.00 | dlynes | kamanashisroy: because 'BILL' is not a valid value, more than likely |
| 07:40.08 | dlynes | kamanashisroy: it's 'BILLING' |
| 07:40.31 | kamanashisroy | dlynes: I see, I was following channelvariables.txt |
| 07:40.50 | gnorbert | dlynes: But can I connect to the same meetme conference from two extensions? |
| 07:41.17 | dlynes | kamanashisroy: don't know what that file is, or where it is |
| 07:41.36 | kamanashisroy | dlynes: it is in asterisk distro .. in doc directory .. |
| 07:41.56 | dlynes | kamanashisroy: I determined what it was, by either creating a Master.csv file, or by reading the source code....I can't remember which, it was so long ago |
| 07:42.15 | kamanashisroy | dlynes: thanks :) |
| 07:42.40 | dlynes | kamanashisroy: 'Use the source, Luke!' |
| 07:43.06 | dlynes | gnorbert: yes |
| 07:43.07 | kamanashisroy | dlynes: I know .. I did the same earlier .. but was lost watching the doc :( |
| 07:43.24 | dlynes | gnorbert: the Page() function does that quite effectively |
| 07:43.43 | dlynes | gnorbert: using N extensions |
| 07:43.47 | gnorbert | dlynes: Thanks, now I go back to my .conf files. :) |
| 07:44.07 | Keypad | urh im lost |
| 07:44.08 | dlynes | gnorbert: erm Page() application I mean |
| 07:44.32 | gnorbert | Ok, thanks. :) |
| 07:44.47 | dlynes | gnorbert: show application page, from the cli |
| 07:45.29 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 07:47.58 | *** join/#asterisk SwK (n=SwK@user-24-236-97-87.knology.net) |
| 07:49.02 | Keypad | whats all this auth id stuff |
| 07:49.13 | *** join/#asterisk mrnick (n=basement@kulnet-nat-2.kulnet.kuleuven.be) |
| 07:52.16 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:52.51 | Keypad | whats the first thing I should do ? |
| 07:53.04 | Keypad | get it so that when I pick up the handset it does something ? |
| 07:55.02 | Keypad | Whats the difference between caller id and auth id ? |
| 07:56.10 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 07:58.37 | C4colo | that is like asking what is the difference between your social security number and your IRC nick |
| 07:59.21 | C4colo | callerid is the info displayed on the caller id screen on the phone when a call is received, such as "Joe Smith" <1234567890> |
| 07:59.36 | C4colo | authid is the user id on the system for authentication purposes |
| 07:59.47 | Keypad | oh thanks |
| 08:03.41 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:06.51 | Keypad | Urh I have no idea on what im doing |
| 08:07.37 | C4colo | we all started there at some point |
| 08:07.40 | *** join/#asterisk ToTo (n=ToTo@207.176.6.180) |
| 08:07.51 | Keypad | :) |
| 08:07.54 | C4colo | you want a dialplan to look at? |
| 08:08.05 | Keypad | I think I am doing things wrong |
| 08:08.39 | C4colo | I think I have an example dialplan you could use as a reference |
| 08:10.17 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 08:10.49 | Keypad | Yeah that would be cool |
| 08:11.02 | C4colo | yea, if I can find it |
| 08:11.13 | C4colo | I can't remember where I saved it |
| 08:11.53 | C4colo | ah found it |
| 08:14.17 | C4colo | that file has a few cool things in it, including some example php agi scripts |
| 08:14.32 | C4colo | you will need to install php and asterisk::agi |
| 08:14.42 | C4colo | for it to work, but they are good as an example |
| 08:15.35 | C4colo | that's probably later, the extensions.conf is a good place to start, that one has a lot of features and shows a lot of the applications available |
| 08:15.48 | C4colo | also, google is your friend |
| 08:16.00 | C4colo | search for: asterisk dial command |
| 08:16.15 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:16.16 | C4colo | or: asterisk voicemail.conf |
| 08:16.34 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 08:16.46 | C4colo | you will usually find info about the various settings for each file or application |
| 08:17.52 | C4colo | some of the sound files referenced in extensions.conf may not exist and may need to be recorded |
| 08:19.14 | Keypad | Hmm I think I am hooking up my 3102 wrong |
| 08:19.35 | C4colo | probably |
| 08:19.54 | C4colo | I know people who have been working with asterisk for years who couldn't get the 3102 to work like it should |
| 08:21.00 | C4colo | the fxs ports are pretty easy, as those are just straight-forward provisioning |
| 08:21.21 | C4colo | but the fxs may take you quite some time, and lots of fiddling |
| 08:21.42 | Keypad | lol, it helps to tick the enable webserver on WAN tickbox |
| 08:22.12 | C4colo | heh yea |
| 08:22.23 | C4colo | that would be the first thing |
| 08:22.25 | Keypad | now I dont need my laptop on my lap |
| 08:22.47 | C4colo | yea, I just set up HTTP provisioning for SPA2102s for work |
| 08:23.38 | C4colo | I actually set up a computer with two ethernet ports so that I could plug into the LAN and the WAN side at the same time |
| 08:23.50 | *** part/#asterisk RoyK (n=roy@ip-157-60-149-91.dialup.ice.no) |
| 08:23.57 | C4colo | got tired of moving the cable back and forth trying to get it to work |
| 08:24.06 | C4colo | the good news is that it all works now |
| 08:24.10 | Keypad | Is that like automating setting up the phone ? |
| 08:24.24 | C4colo | yea, it downloads an xml file |
| 08:24.29 | C4colo | has all the settings in it |
| 08:24.45 | C4colo | and then it checks in every so often to grab the config in case there are any changes |
| 08:24.56 | talntid | and puts its logfiles |
| 08:25.36 | Keypad | wow thats cool |
| 08:26.14 | Keypad | I am just playing with Asterisks atm. I work at a school with a old PBX |
| 08:26.39 | Keypad | after watching a video I knew how usefull it would be for our school |
| 08:26.50 | Keypad | since our phone system is only in 1 block in our whole school |
| 08:27.04 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 08:27.14 | Keypad | where as our network spreads across the whole school |
| 08:27.26 | joobie | y0 |
| 08:31.03 | *** join/#asterisk BBHoss (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 08:35.42 | joobie | guys, what functions would i use in the dialplan so that i can (a) automatically pick up a call with asterisk and push the user to an ivr (b) prompt the user to specify a password in the ivr (c) if the password is correct then prompt for an extension number to monitor (d) then monitor the extension ? |
| 08:36.17 | Keypad | haha when I pick up the phoen now it makes a different dial tone ! |
| 08:36.46 | Keypad | omg some girl talked to me |
| 08:37.15 | Keypad | and said mailbox |
| 08:37.18 | Keypad | xD |
| 08:37.46 | *** join/#asterisk kkjoe (n=opera@217.7.238.226) |
| 08:37.48 | C4colo | the applications would be: Answer, Playback, Background, Authenticate, and the last part varies based on your version of asterisk |
| 08:37.54 | C4colo | Keypad, that would be Allison Smith |
| 08:38.17 | C4colo | http://www.theivrvoice.com ... you can have her record your prompts for a nominal fee if you want consistancy |
| 08:39.10 | joobie | C4, sup bro |
| 08:39.12 | joobie | thanks :P |
| 08:39.23 | Keypad | Hahah |
| 08:39.26 | Keypad | Thats cool |
| 08:39.32 | Keypad | I got it to work somewhat |
| 08:39.43 | C4colo | cool |
| 08:40.41 | C4colo | chanspy |
| 08:40.50 | C4colo | if you just want to listen |
| 08:41.11 | C4colo | I was trying to look up the bridge application to see what version they were/did implement it in |
| 08:41.49 | joobie | it was before 1.4 |
| 08:41.57 | C4colo | bridge()+ |
| 08:41.59 | joobie | i read something about a few features that came as of 1.4.. additional features that is |
| 08:42.00 | C4colo | ? rather |
| 08:42.03 | kkjoe | does anyone has a clue why i can`t call sip addresses with an snome phone an asterisk ? if i try to call 20@192.168.3.91 with an snom 360 phone i get invalid context 20, why there isn`t the complete number 20@192.168.3.91 taken as the extension ? |
| 08:42.03 | joobie | ahhh |
| 08:42.11 | joobie | C4 |
| 08:42.19 | joobie | what about ExtenSpy() and Monitor() ? |
| 08:42.30 | C4colo | monitor is for recording |
| 08:42.30 | joobie | so many other options too.. why do you opt for Chanspy() ? |
| 08:42.35 | *** join/#asterisk ToTo (n=ToTo@209.8.41.213) |
| 08:42.35 | C4colo | extenspy might work, never used it |
| 08:42.37 | *** join/#asterisk torrikft (n=pako@m85-94-184-212.andorpac.ad) |
| 08:42.39 | joobie | ahh |
| 08:43.44 | joobie | c4, will chanspy effect the ability to monitor constantly across multiple calls? |
| 08:44.09 | joobie | like say i dialed '123' to chanspy on 123.. and 123 makes a call. then hangs up.. then makes another, etc etc.. will chanspy hold that constant stream across multiple calls? |
| 08:45.27 | C4colo | hmm |
| 08:45.29 | C4colo | let me see |
| 08:46.38 | C4colo | seems to work |
| 08:46.42 | C4colo | just goes silent |
| 08:46.51 | C4colo | then picks up when the channel is bridged |
| 08:47.11 | C4colo | I don't know how extenspy works though, like I said, never used it |
| 08:48.03 | *** join/#asterisk Vec (n=Vec@nsabfw1.nsab.se) |
| 08:49.21 | joobie | ahh k |
| 08:49.52 | torrikft | mornin all |
| 08:51.39 | torrikft | may i ask a question about a2billing in this chan? |
| 08:55.57 | joobie | hmm .. guys anyone know of a free repo of wav / mp3 phrases for IVR? specifically, "please enter your password" |
| 08:56.02 | joobie | soemthing along those lines? |
| 08:56.15 | C4colo | vm-password |
| 08:56.27 | C4colo | says "Password" |
| 08:57.34 | joobie | is that a native wave in asterisk c4? |
| 09:01.09 | *** join/#asterisk XnOSX (n=XnOSX@212.145.173.80) |
| 09:01.30 | Keypad | Does any one use Asterisk GUI ? |
| 09:02.13 | Keypad | I maded a extentsion |
| 09:02.22 | Keypad | lol good england* |
| 09:02.27 | Keypad | but I cant delete it now |
| 09:02.45 | joobie | c4, ahh got it.. ta |
| 09:03.04 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.139) |
| 09:05.28 | *** join/#asterisk Great_Anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-eddd88857750ea40) |
| 09:06.48 | joobie | Unable to find a codec translation path from g729 to ulaw |
| 09:06.56 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-7-16.revip2.asianet.co.th) |
| 09:07.03 | joobie | is there a way around that? im playing a .gsm file via Playback and getting flooded with mesages simialr to that |
| 09:07.15 | joobie | then after that is says "Unable to find a codec translation path from g729 to gsm" |
| 09:14.38 | Keypad | Hmm, I got my phone connected, now I need a way to dial numbers out onto the PSTN |
| 09:16.18 | gr0mit | Keypad, you will need a voip provider |
| 09:16.23 | gr0mit | or an interface card |
| 09:17.05 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 09:18.37 | Keypad | I have a VOIP gateway thing |
| 09:18.40 | Keypad | that can connect to PSTN |
| 09:19.01 | Keypad | Its a SPA-3102 |
| 09:20.02 | gr0mit | ah ok. good luck! |
| 09:20.36 | Keypad | lol |
| 09:20.37 | Keypad | :( |
| 09:20.39 | Keypad | ty |
| 09:20.51 | Keypad | I got it to have a dialtone |
| 09:21.12 | *** join/#asterisk Whisk (n=Whisk@82-44-94-242.cable.ubr04.croy.blueyonder.co.uk) |
| 09:21.32 | Keypad | and I can call 7000# and get a voice prompt ! |
| 09:22.00 | Whisk | ~centos52bug |
| 09:22.01 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 09:22.04 | Keypad | I think I am gonna waste all my cell phone money calling into my house :) |
| 09:25.08 | C4colo | Keypad: sprint to home / my favs / friends list |
| 09:26.22 | Keypad | omfg |
| 09:26.32 | Keypad | vodaphone suck |
| 09:26.45 | Keypad | you get charged if your waiting to call some one |
| 09:27.05 | Keypad | or maybe because my box picked up the call then hanged up instantly |
| 09:27.08 | Keypad | it screwed up |
| 09:27.34 | joobie | "set_format: Unable to find a codec translation path from g729 to gsm" anyone know how to fix that error? |
| 09:30.02 | Keypad | " Goooddd byeeee " :) |
| 09:30.04 | Keypad | <3 |
| 09:32.26 | Keypad | Man I am loving this :) |
| 09:33.34 | joobie | sweet it works |
| 09:33.40 | joobie | g729 codec |
| 09:33.47 | joobie | damn licensed technology sux |
| 09:34.14 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 09:37.36 | *** join/#asterisk cabbiepete (n=cabbiepe@83-244-133-169.cust-83.exponential-e.net) |
| 09:39.51 | Keypad | Whats the best way to start calling out |
| 09:41.27 | *** join/#asterisk cabbiepete (n=cabbiepe@83-244-133-169.cust-83.exponential-e.net) |
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| 09:45.03 | *** join/#asterisk contactdq1 (n=dqasem@rrcs-71-43-203-146.se.biz.rr.com) |
| 09:46.13 | contactdq1 | hi everyone, i could use some help setting up some astra phones....i'm get a 404 when i sip debug in the cli and was wondering what i should do. |
| 09:52.07 | *** join/#asterisk Jerjer[mobile] (n=PhatJ@24-231-253-65.dhcp.aldl.mi.charter.com) |
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| 09:57.35 | *** part/#asterisk RoyK (n=roy@ip-157-60-149-91.dialup.ice.no) |
| 09:57.35 | *** join/#asterisk badcfe (i=christia@peter.mindslice.net) |
| 09:58.01 | badcfe | hello. i Set(CALLERID(number)=123) in dialplan and then Dial. no effect for the outgoing call. |
| 09:58.05 | badcfe | what do i miss here |
| 09:58.34 | joobie | badcfe, you can specify it in the sip.conf too. have you tried that? |
| 09:58.49 | badcfe | i need variable and not just 123 |
| 10:00.05 | joobie | not sure then bad |
| 10:00.16 | joobie | all i know is i tried to set it in extensions.conf and it didnt take effect |
| 10:00.23 | joobie | had to do it in sip.conf and then it was fine |
| 10:00.34 | badcfe | oh thats saad |
| 10:00.38 | joobie | hang around though - some experienced peeps in this chan - no doubt someone will be able to help |
| 10:01.09 | badcfe | yes. hope its my lucky day |
| 10:08.35 | *** join/#asterisk spike008t (n=spikie@ven69-2-82-228-116-153.fbx.proxad.net) |
| 10:08.37 | spike008t | hi all |
| 10:08.50 | *** part/#asterisk kkjoe (n=opera@217.7.238.226) |
| 10:09.47 | spike008t | Does someone have already use the iaxclient? (sorry if I post my question here but nobody is in the iaxclient chan :S) |
| 10:10.02 | torrikft | spike i use iax |
| 10:10.21 | torrikft | whats up? |
| 10:11.12 | spike008t | I've got a problem with the version 2.0.2 on windows. In fact I can hear the other but nobody can hear me |
| 10:13.18 | C4colo | spike008t: that is probably not an issue with the client |
| 10:13.39 | C4colo | start troubleshootig by looking at your firewall, nat settings, and RTP port settings |
| 10:13.57 | C4colo | sometimes it is as simple as adding nat=yes to the user details in sip.conf |
| 10:14.45 | spike008t | C4colo: But in fact I'm writing my client on c#... And I'm testing on local network... |
| 10:15.11 | spike008t | my serv and the other phone are also in.. |
| 10:15.31 | C4colo | then firewall and nat settings would be out |
| 10:15.35 | C4colo | check rtp port settings |
| 10:16.20 | *** join/#asterisk gnorbert (n=gnorbert@228-177.static.ew.hu) |
| 10:16.21 | spike008t | okay I'll going to see it... And before that, i was in the 1.0 version and everything was good... |
| 10:16.45 | C4colo | I know nothing of the client, just offering where I would look |
| 10:18.54 | spike008t | C4colo: ok thank's. I'll see it |
| 10:19.37 | Keypad | can some one explain these Calling Rule things |
| 10:19.50 | Keypad | do I need to change them for my contery ? |
| 10:23.12 | gnorbert | Hi, does somebody have an idea, how could I open a file called 345.wav, if the extension is 12345 and so on? (So to see only the last 3 numbers of extension. For example is it possible to use ${EXTEN} %1000?) I ask it here, because I shouldn't restart the server until I don't have it. |
| 10:29.46 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279551595.dsl.bell.ca) |
| 10:30.02 | *** join/#asterisk friendly12345 (n=friendly@ppp59-167-145-230.lns4.mel6.internode.on.net) |
| 10:30.29 | Keypad | Does any one know how to make it so that when I pick up my phone I can push 1 or call our some how |
| 10:35.50 | badcfe | joobie: i found out |
| 10:36.08 | badcfe | joobie: have to do a SetCallerPres(allowed) |
| 10:36.52 | *** join/#asterisk macros73 (n=cs@c-67-186-22-161.hsd1.pa.comcast.net) |
| 10:37.13 | styelz | joobie: yo |
| 10:39.22 | DarKnesS_WolF | tzafrir_laptop: there ? |
| 10:39.36 | tzafrir_laptop | yes |
| 10:39.55 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 10:44.27 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 10:50.37 | gnorbert | Anybody? |
| 10:56.11 | joobie | badcfe, nice |
| 10:56.23 | joobie | styelz, sup :P |
| 11:00.51 | gnorbert | Can somebody tell, what can be the problem with this extensions.conf? (It writes "User cannot be found at given address, when I try to call it on ext 01000) |
| 11:00.57 | gnorbert | http://pastebin.com/dd49e20d |
| 11:02.31 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:07.24 | gnorbert | Anybody? :) |
| 11:08.35 | styelz | i think you need to prepend your exten with _ when using pattern matching |
| 11:08.46 | styelz | like _01XXXX |
| 11:08.52 | gnorbert | Sounds well. :) |
| 11:09.01 | styelz | or just use 01000 instead |
| 11:09.14 | gnorbert | Thank you, I really missed it. |
| 11:09.20 | styelz | welcome |
| 11:15.44 | Keypad | how come all the User Extensions I make are all grayed out ? |
| 11:15.47 | Keypad | and I cant change them |
| 11:21.41 | Keypad | fuck |
| 11:21.42 | Keypad | I broke it |
| 11:22.58 | Keypad | It plays the busy tone every time I call any extention now |
| 11:24.21 | Rico29 | l |
| 11:26.55 | *** join/#asterisk Segnale007 (n=Segnale0@host141-4-dynamic.18-79-r.retail.telecomitalia.it) |
| 11:38.11 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 11:43.44 | *** join/#asterisk redax (i=redax@r6.hu) |
| 11:44.02 | redax | hi, |
| 11:44.21 | redax | is it possible to configure the S0 bus length in mISDN ? |
| 11:44.58 | redax | similar to zaptel, where I can configure the buslength with some 0-1-2 |
| 11:45.38 | tzafrir_laptop | redax, buslength? where? (in zaptel) |
| 11:46.12 | tzafrir_laptop | In some specific driver? |
| 11:46.20 | redax | in zaptel.conf the span=1,1,0,ccs,ami |
| 11:46.29 | redax | the third digit imho |
| 11:46.49 | redax | yes. in zaphfc... |
| 11:48.14 | tzafrir_laptop | ah. OK. LBO, as it is called in zaptel.conf |
| 11:58.29 | *** join/#asterisk op3r (n=Op3r@222.127.34.153) |
| 11:58.39 | Keypad | Can some one explain why I cant delete phone extentions from the web gui |
| 11:59.05 | creativx | Keypad.. #asterisk |
| 11:59.10 | creativx | which web gui |
| 12:00.08 | Keypad | The one that comes with now |
| 12:00.21 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:00.22 | tzafrir_laptop | using the gui |
| 12:00.38 | tzafrir_laptop | (or by editing users.conf |
| 12:00.41 | tzafrir_laptop | ) |
| 12:00.55 | Keypad | but the GUI boxes are all grayed out |
| 12:01.12 | Keypad | and when I click on the box it select them all |
| 12:01.13 | tzafrir_laptop | That's the sort of answer you'll get asking here and not in #asterisknow or #asterisk-gui ;-) |
| 12:01.31 | tzafrir_laptop | Keypad, do you use Firefox 3? |
| 12:01.38 | Keypad | Yeah tzafrir_laptop |
| 12:01.51 | Keypad | Let me guess |
| 12:01.54 | Keypad | I need to use IE |
| 12:01.57 | Keypad | or something |
| 12:02.49 | tzafrir_laptop | http://bugs.digium.com/12533 |
| 12:02.54 | tzafrir_laptop | Get the patch from there |
| 12:03.24 | tzafrir_laptop | I've already updated my packages |
| 12:03.29 | Keypad | Thanks for that |
| 12:03.38 | Keypad | tzafrir_laptop |
| 12:04.20 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 12:08.05 | gnorbert | Does someody have any idea, what can be the problem? |
| 12:08.07 | gnorbert | http://pastebin.com/d2bdbcafd |
| 12:10.08 | Keypad | Urhh I hate it working for a while then I broke everything |
| 12:11.12 | Keypad | I wonder what I did wrong |
| 12:11.36 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:12.53 | *** join/#asterisk ThoMe (i=tm@tm.muc.de) |
| 12:12.59 | ThoMe | 'lo |
| 12:13.07 | ThoMe | how i can kill a call? -- <SIP/12-b58f36b8> Playing 'vm-for' (language 'de') -- <SIP/12-b58f36b8> Playing 'vm-Old' (language 'de') -- <SIP/12-b58f36b8> Playing 'vm-messages' (language 'de') |
| 12:13.08 | Keypad | I added a provider and that broke everything |
| 12:13.15 | ThoMe | this call: SIP/12-b58f36b8 |
| 12:13.22 | ThoMe | is it posible to kill this? :) |
| 12:14.14 | tzafrir_laptop | gnorbert, create_addr: No such host: 5060 |
| 12:14.38 | ThoMe | tzafrir_laptop: ideas? :-) |
| 12:14.52 | tzafrir_laptop | gnorbert, What exactly do you have configured for that peer? |
| 12:14.56 | gnorbert | tzafrir_laptop: That's in my sip.conf. If that's not good, where shall I do that? |
| 12:15.12 | gnorbert | A minute, then I pastebin my sip.conf too. |
| 12:15.13 | gnorbert | :) |
| 12:15.57 | gnorbert | tzafrir_laptop: http://pastebin.com/dd0bf32f |
| 12:17.37 | *** join/#asterisk M1s3ry (n=M1s3ry@216.207.245.1) |
| 12:17.54 | tzafrir_laptop | gnorbert, you have no peer called "5060" . hence 5060 is interpereted as a hostname. In the syntax [username@]hostname |
| 12:18.56 | tzafrir_laptop | ThoMe, soft hangup SIP/12-b58f36b8 |
| 12:19.00 | gnorbert | tzafrir_laptop: Then where shall I make new peer? |
| 12:19.06 | ThoMe | oh, ok |
| 12:19.30 | tzafrir_laptop | Or call to a host. Or whatever. What did you actually want to call to? |
| 12:20.20 | gnorbert | I wanted to call a meetme conference of the server |
| 12:21.00 | gnorbert | So shall I write the ip address instead of 5060? |
| 12:21.05 | op3r | hello again. I was trying chanspy using 1.2 and I am hearing beep after i put in the extension. this is the config http://pastebin.com/d4c31def I am doing it wrong? |
| 12:23.49 | op3r | anyone? |
| 12:27.10 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 12:27.18 | tzafrir_laptop | gnorbert, define a peer (or firend, or whatever) for that server, and use that |
| 12:28.18 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
| 12:29.17 | gnorbert | Isn't userA a friend in extensions.conf? |
| 12:29.20 | gnorbert | I guess it is.. |
| 12:29.41 | gnorbert | And when I try with that, it gives the same failure message. |
| 12:30.24 | gnorbert | Sorry, if I misunderstood it.. |
| 12:37.08 | *** join/#asterisk ToTo (n=ToTo@209.8.41.213) |
| 12:38.05 | *** join/#asterisk teknoprep (n=cmr@unaffiliated/teknoprep) |
| 12:41.22 | dominic1 | hi, I need some help |
| 12:41.43 | dominic1 | can anybody tell me how I can initiate a attended transfer via management interface? |
| 12:45.05 | gnorbert | tzafrir_laptop: Could you explain it a little more circumstancially? |
| 12:45.44 | *** join/#asterisk ToTo (n=ToTo@207.176.6.212) |
| 12:47.10 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 12:47.21 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:47.51 | *** join/#asterisk kio (n=kio@38.98.68.18) |
| 12:51.49 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 12:55.48 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 12:55.50 | gnorbert | tzafrir_laptop: Now it gives only one notice instead of two notices and one warning after changing Channel to ip address. :) |
| 12:56.06 | gnorbert | So it improved a lot. :) |
| 12:56.30 | dominic1 | how can I set a active call on hold by asterisk? |
| 12:56.43 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:56.50 | tzafrir_laptop | gnorbert, sorry, busy. But your question appears to be a pretty basic one regarding the usage of sip trunks |
| 12:58.21 | gnorbert | tzafrir_laptop: Sorry, didn't want to bother you and thanks for help anyway. :) |
| 13:06.02 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:06.10 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:06.56 | dominic1 | if I want to start a transfer my system only waits for 2 digits.... |
| 13:06.59 | dominic1 | why that? |
| 13:07.45 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-104-82-57.pskn.east.verizon.net) |
| 13:08.34 | *** join/#asterisk sgtpepper (n=ncorrare@200.61.187.185) |
| 13:09.11 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 13:09.40 | sgtpepper | can anyone help me with an Issue with libmfcr2, I'm trying to get the right parameters to connect asterisk with a panasonic PABX, and I get the following error: http://pastebin.com/m6e5b72e5 |
| 13:09.57 | sgtpepper | outward calls work, inward calls does not |
| 13:10.02 | tzafrir_laptop | dominic1, attended transfer? |
| 13:10.36 | dominic1 | yes |
| 13:14.50 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:15.11 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:15.55 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.fr) |
| 13:17.42 | *** join/#asterisk coppice (n=chatzill@27.202.17.210.dyn.pacific.net.hk) |
| 13:20.31 | *** join/#asterisk [intra]lanman (n=lanman@va-71-0-90-168.dyn.embarqhsd.net) |
| 13:21.36 | *** join/#asterisk moy (n=moy@nat/ibm/x-622ba78a236b3956) |
| 13:22.27 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 13:27.05 | gnorbert | Does somebody have an idea, what can be the problem with this? |
| 13:27.05 | gnorbert | http://pastebin.com/d6068f02b |
| 13:28.25 | davevg-btwtech | Default vs default |
| 13:30.51 | gnorbert | davevg-btwtech: Still the same message after correction |
| 13:32.18 | davevg-btwtech | is SIP/172... userA? |
| 13:32.27 | davevg-btwtech | in the call file |
| 13:33.13 | *** join/#asterisk tobias (n=tobias@cpe-069-134-204-037.nc.res.rr.com) |
| 13:33.45 | gnorbert | Still the same. |
| 13:33.47 | davevg-btwtech | you also have a user set in extensions.conf instead of sip.conf |
| 13:33.50 | [TK]D-Fender | gnorbert: Why the hell is * calling ITSELF? |
| 13:34.23 | gnorbert | [TK]D-Fender: I have to play sound in a meetme conference. |
| 13:34.54 | [TK]D-Fender | gnorbert: you've clearly picked a bad way. Don't use SIP to call * from * |
| 13:35.27 | [TK]D-Fender | gnorbert: Go read up on the complete list of * channel types. Hopefully you'll be able to pick out the one to use on your own. |
| 13:40.18 | *** part/#asterisk friendly12345 (n=friendly@ppp59-167-145-230.lns4.mel6.internode.on.net) |
| 13:50.20 | *** join/#asterisk ArashHemmat (n=arash@91.184.77.232) |
| 13:50.45 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 13:51.32 | *** join/#asterisk freezey (n=freezey@rrcs-72-43-189-130.nyc.biz.rr.com) |
| 14:00.09 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 14:08.22 | *** join/#asterisk jpcansa (n=jpbenavi@190.10.19.145) |
| 14:10.07 | jpcansa | hi, why would I suddenly lose a zap channel? i run "zap show channels" and i have all my channels up, and then, next day i run "zap show channels" and I have lost one or more channels |
| 14:10.11 | jpcansa | any idea? |
| 14:10.25 | *** join/#asterisk XnOSX (n=XnOSX@212.145.173.80) |
| 14:10.47 | jpcansa | if I restart i get all the channels back |
| 14:10.48 | [TK]D-Fender | jpcansa: Done a "zap destroy channel" by any chance? |
| 14:11.34 | jpcansa | Fender, yes, how do i get the channel back up? |
| 14:11.58 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-e539dd19d3a47071) |
| 14:12.50 | [TK]D-Fender | jpcansa: Restart *. |
| 14:13.28 | [TK]D-Fender | jpcansa: And stop using that option. Thats a way to take a channel out of commission, and is NOT a way to gently disconnect a call |
| 14:13.50 | jpcansa | Fender: so, zap destroy channels will remain the channel down? |
| 14:13.56 | *** join/#asterisk ManxPower (n=manxpowe@7.sub-75-249-205.myvzw.com) |
| 14:14.17 | *** part/#asterisk ManxPower (n=manxpowe@7.sub-75-249-205.myvzw.com) |
| 14:14.33 | *** join/#asterisk gfather (n=enforcer@86.108.47.162) |
| 14:15.02 | *** join/#asterisk ManxPower (n=manxpowe@7.sub-75-249-205.myvzw.com) |
| 14:15.07 | gfather | guys can i coonect the asterisk to a mobile phone for outgoing calls ? |
| 14:15.21 | [TK]D-Fender | jpcansa: Yes, so stop./ |
| 14:15.51 | jpcansa | Fender: how can i kill a channel then, some people left calls on hold by mistake, and then it gets stucked in *, even if it is not holding anymore in the phone, so the zao channel remain unusable, like busy |
| 14:15.52 | [TK]D-Fender | gfather: Only if you have some other piece of hardware that can talk to the cell. |
| 14:15.54 | ManxPower | gfather: Maybe, depending on the phone and the adapter. |
| 14:16.11 | [TK]D-Fender | jpcansa: "soft hangup [channel]" |
| 14:16.17 | gfather | what mobiles are supported or can work |
| 14:16.18 | gfather | ? |
| 14:16.39 | [TK]D-Fender | gfather: its not the PHONE, its a question of the interface |
| 14:17.04 | ManxPower | gfather: Asterisk does NOT support connecting cell phones to it. Asterisk supports SIP and analog POTS interfaces. Find a device that will convert your cell phone into one of those two interfaces. |
| 14:17.13 | [TK]D-Fender | gfather: If you have a special adapter that lets you plug a regular phone into it (like the dock-n-talk), you can use it with an FXO card |
| 14:17.32 | ManxPower | Those devices exist for some phones -- none of them that I've heard of people using worked all that well (mostly hangup detection issues) |
| 14:17.39 | *** join/#asterisk javb (n=javb@tdev210-201.codetel.net.do) |
| 14:17.59 | jpcansa | Fender: will * hungup a zap channel if it detects a hung up from the outside? |
| 14:18.12 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 14:18.13 | gfather | i see |
| 14:18.14 | javb | hello, MeetMe is not requesting for the password or pin to enter, mmmm, im using meetme(600,P,1234) ... any idea? |
| 14:18.30 | gfather | so its has allot of problem , or meduim ones ? |
| 14:18.52 | ManxPower | jpcansa: YES! |
| 14:18.54 | *** join/#asterisk spokra (n=spokra@host093-179-153.sea0.speakeasy.net) |
| 14:18.58 | [TK]D-Fender | gfather: Cross your fingers & say a prayer |
| 14:18.58 | ManxPower | jpcansa: If it can. |
| 14:19.10 | gfather | <[TK]D-Fender> loooooooooool |
| 14:19.21 | gfather | man i need a solution |
| 14:19.23 | gfather | damn :( |
| 14:20.09 | jpcansa | Manxpower: why it wouldnt can? |
| 14:20.27 | [TK]D-Fender | gfather: Go make sure you can get a phone thats compatible with a device like this : http://www.cellantenna.com/Dockingstations/dockntalk.htm |
| 14:20.33 | ManxPower | jpcansa: if your telco does not use North American hangup indications |
| 14:20.55 | gfather | <[TK]D-Fender> ill do |
| 14:21.09 | ManxPower | javb: you didn't do something stupid like adding extra spaces to the MeetMe command, did you? |
| 14:21.42 | jpcansa | Manxpower: where can i change those settins in * to meet my telco? |
| 14:21.42 | gfather | <[TK]D-Fender> is the Dock-N-Talk is good , or there are better ones |
| 14:22.09 | ManxPower | jpcansa: This is an incredibly complicated thing, and I am not interested in spending the next few hours with you to try and fail to make it work well. |
| 14:22.13 | [TK]D-Fender | gfather: No idea, go look for reviews yourself. Voxilla used to sell it so I presume its probably about as good as any other on par. |
| 14:22.23 | ManxPower | Sorry, you'll have to find a more patient person |
| 14:22.35 | ManxPower | gfather: Few people use those devices with Asterisk |
| 14:23.16 | jpcansa | good, thx anyway |
| 14:23.23 | gfather | <ManxPower> yes i understand that , but becouse its the first time i hear about these |
| 14:23.30 | gfather | and its the only solution i have |
| 14:23.44 | [TK]D-Fender | gfather: Too bad for you I guess. Get searching |
| 14:23.51 | *** join/#asterisk moy (n=moy@nat/ibm/x-e83015f2125d9d42) |
| 14:24.15 | gfather | <[TK]D-Fender> yes i see that |
| 14:24.26 | gfather | but how ill now its gonna work with asterisk |
| 14:24.36 | ManxPower | gfather: Do a google search, dude. |
| 14:24.47 | [TK]D-Fender | gfather: it will work, its a questio of which model will work with your PHONE. |
| 14:24.51 | ManxPower | Search the mailing list archive, even! |
| 14:25.05 | gfather | im searching guys :) |
| 14:25.46 | hi365 | anyone ever have issues tring to import the cdr in mysql? im getting: 'Row 1 doesn't contain data for all columns' for most of the lines |
| 14:26.07 | [TK]D-Fender | hi365: pastbin is your friend. |
| 14:26.34 | *** join/#asterisk angryuser (n=sldf@78.115.236.230) |
| 14:26.51 | hi365 | [TK]D-Fender: http://pastebin.ca/1079504 |
| 14:28.03 | *** join/#asterisk Segnale007 (n=Segnale0@host92-1-dynamic.9-79-r.retail.telecomitalia.it) |
| 14:28.20 | [TK]D-Fender | hi365: You know without seeing your broken data and broken SQL file that your last pastebin was completely worthless, right? |
| 14:28.48 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 14:28.59 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:29.00 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:29.02 | hi365 | [TK]D-Fender: hmm, its 175MB. how can i post it? |
| 14:29.17 | [TK]D-Fender | hi365: You know which row #'s.... |
| 14:29.55 | ManxPower | hi365: How about pasting the first 50 error messages? |
| 14:30.15 | [TK]D-Fender | ManxPower: we see the error messges, now to see whats CAUSING them. |
| 14:30.24 | hi365 | the first 50 mysql error's? or the first 50 lines of the dump? |
| 14:30.34 | ManxPower | How about both? |
| 14:30.48 | [TK]D-Fender | hi365: the script, and the source data, and your TABLE. You know... all the stuff that is wrong. |
| 14:31.58 | hi365 | could it be that all the sql lines are mising a line breaks? cause head -n49 shows no data (just the beging of the dump file) while head -n50 seem to run thru a ton of records |
| 14:32.26 | hi365 | (im just rtrying to figure out how i show you guys a couple of lines without posting all 175MB) |
| 14:32.48 | [TK]D-Fender | hi365: You'd better show us something complete, because you are wasting our time otherwise. |
| 14:33.06 | *** join/#asterisk LakeSolon (n=blake@64-83-198-152.dhcp.stcd.mn.charter.com) |
| 14:33.13 | ManxPower | hi365: The key to getting help here is to FOLLOW THE INSTRUCTIONS OF THE PERSON HELPING YOU. |
| 14:34.42 | hi365 | ManxPower: i get that, but can someone give me instructions as on how to share the dump? its 18mb zipped! where can i past that? |
| 14:34.48 | hi365 | can i email it to you guys? |
| 14:35.06 | seanbright | head -n100 myfile > anotherfile |
| 14:35.14 | seanbright | then pastebin anotherfile |
| 14:35.24 | seanbright | glad i could help |
| 14:35.40 | *** join/#asterisk [intra]lanman (n=lanman@va-71-0-90-168.dyn.embarqhsd.net) |
| 14:37.37 | hi365 | here is the forst 51 lines: http://www.2shared.com/file/3635145/a9e317f9/past.html |
| 14:37.48 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 14:37.50 | [TK]D-Fender | hi365: every line is apparently broken, so Oh I don't know... maybe the FIRST 5 ought to do. |
| 14:37.51 | seanbright | hmmm |
| 14:38.01 | seanbright | it's gzipped |
| 14:38.19 | ManxPower | hi365: we don't damn com;pressed files. What is wrong with you? |
| 14:39.05 | hi365 | nothing. just having a hard day |
| 14:39.52 | hi365 | give me a minut or two |
| 14:41.13 | dominic1 | any idea how I can execute Set(DEVSTATE(Custom:100)=RINGING) from the AMI??? |
| 14:41.27 | ManxPower | I suspect once you try using UNCOMPRESSED files to work with your errors will go away |
| 14:41.37 | ManxPower | dominic1: You are using 1.6? |
| 14:41.45 | dominic1 | no 1.4 with backported devstate |
| 14:42.10 | hi365 | i am |
| 14:42.30 | tzafrir_laptop | dominic1, execute? to what channel? |
| 14:43.03 | [TK]D-Fender | dominic1: READ : http://www.voip-info.org/wiki/index.php?page_id=4398&tk=6c62aea9dd41f946c9fb&comments_page=1 |
| 14:43.52 | dominic1 | hi fender, I wanted to use the original port from digium |
| 14:44.24 | [TK]D-Fender | dominic1: so far, TFB |
| 14:44.50 | [TK]D-Fender | dominic1: Guess that backport didn't include to B5 mod |
| 14:45.20 | dominic1 | what's B5? |
| 14:46.05 | [TK]D-Fender | dominic1: * 1.6.0-Beta5 |
| 14:46.16 | [TK]D-Fender | dominic1: http://www.voip-info.org/wiki/index.php?page_id=5060 |
| 14:46.32 | [TK]D-Fender | dominic1: the 1.4 devstate patch was from way back last year |
| 14:50.10 | dominic1 | is there a big risk to get a unstable system whe executing many cli commands? |
| 14:50.37 | ManxPower | dominic1: but you are not doing that |
| 14:50.51 | dominic1 | ?? |
| 14:50.52 | tzafrir_laptop | I figure it depends on what those commands are . |
| 14:51.09 | drfreeze | Wow, the digium video on their website has a whole new definition for 'pause' |
| 14:51.11 | tzafrir_laptop | e.g. system(rm -rf /) is not a good choice |
| 14:51.20 | dominic1 | why @Manx |
| 14:51.38 | TrentCreek | \ |
| 14:51.46 | ManxPower | dominic1: I have not seen any indication you are using any CLI commands in any of this conversations. |
| 14:51.55 | drfreeze | I thought it meant stop playing. To them it means dump everything you have downloaded, reset to the beginning, start a new video download and start playing from the beginning |
| 14:52.06 | ManxPower | You asked about AMI, but that is not CLI (hence the difference in letters) |
| 14:52.52 | dominic1 | at the moment I am setting devicestates in the dialplan and that's not very good |
| 14:52.52 | [TK]D-Fender | dominic1: Now why are you looking to do a lot of these changes via AMI? |
| 14:52.58 | tzafrir_laptop | you want to use the CLI command? Isn't there also a respective manager command? |
| 14:53.05 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:53.19 | tzafrir_laptop | (or you could use the manager command "Command") |
| 14:54.40 | dominic1 | cause I have a lot of virtual numbers which have their own devstate |
| 14:55.03 | [TK]D-Fender | dominic1: Fine, but why is *AMI* being used to set them? |
| 14:56.05 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 14:57.12 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 14:57.56 | hi365 | ManxPower: just respond here :) |
| 14:58.54 | dominic1 | following problem: Number1 is ringing, user picks up STATE = INUSE, Number1 is ringing a second time then state is RINGINUSE, the number1 is called a third time, state is again ringinuse. How will I be able to indicate when the state is no more ringinuse? |
| 14:59.37 | dominic1 | with the first ringinuse that's not a problem I set a channelvar and check if this channel did the devstate set to ringinuse |
| 14:59.59 | dominic1 | then I set the dev to state inuse on hangup or ringing if the other channel hangs up |
| 15:04.02 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 15:04.02 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:04.13 | [TK]D-Fender | dominic1: devstate should have nothing to do with hanging up channels unless the your logic is severely broken. |
| 15:04.30 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.121) |
| 15:06.00 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 15:06.36 | dominic1 | if I set devstate manually I need to execute it after hangup to set the new devstate |
| 15:06.50 | [TK]D-Fender | dominic1: clearly. |
| 15:09.53 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 15:11.32 | *** join/#asterisk |dennis| (n=Dennis@200.32.231.18) |
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| 15:15.42 | Qwell | anybody happen to have a server running CentOS 5.1? |
| 15:15.51 | Qwell | extra points for 5.0 |
| 15:16.38 | dlynes | Qwell: Yeah...just not running asterisk :) |
| 15:16.52 | Qwell | dlynes: that's fine - could you get me the output of `cat /etc/redhat-release`? |
| 15:17.57 | dlynes | Qwell: CentOS release 5.2 (Final) |
| 15:18.04 | Qwell | You fail. :p |
| 15:18.16 | davevg-btwtech | CentOS release 5 (Final) |
| 15:18.17 | dlynes | Qwell: Well...5.2...not close enough? |
| 15:18.22 | Qwell | dlynes: nope.. |
| 15:18.32 | Qwell | davevg-btwtech: what's that from? |
| 15:18.38 | davevg-btwtech | my home * server :) |
| 15:18.43 | Qwell | I mean, what version? |
| 15:18.58 | davevg-btwtech | 5.0 i believe, have not updated in a while |
| 15:19.14 | Qwell | could you see what Release: says in `lsb_release -a`? |
| 15:19.22 | errr | Qwell: CentOS release 5 (Final) |
| 15:19.40 | seanbright | Release: 5 |
| 15:19.47 | Qwell | seanbright: why did I expect that? |
| 15:19.56 | Qwell | silly centos |
| 15:20.07 | davevg-btwtech | same here; Release 5 |
| 15:20.11 | seanbright | silly Qwell |
| 15:20.14 | Qwell | very ambiguous :) |
| 15:20.17 | Qwell | thanks guys |
| 15:20.19 | Juggie | LSB Version: :core-3.1-ia32:core-3.1-noarch:graphics-3.1-ia32:graphics-3.1-noarch |
| 15:20.20 | Juggie | Distributor ID: CentOS |
| 15:20.20 | Juggie | Description: CentOS release 5.2 (Final) |
| 15:20.20 | Juggie | Release: 5.2 |
| 15:20.40 | seanbright | Juggie: pastebin |
| 15:20.40 | seanbright | heh |
| 15:20.45 | Juggie | nevar! |
| 15:20.49 | dlynes | Qwell: Teaches you to not use Debian :) |
| 15:21.32 | Juggie | Qwell, what weer you looking for exactally? |
| 15:21.34 | Qwell | so, could one reasonably expect that 5.1 would show 5.1 and not just 5? |
| 15:22.02 | davevg-btwtech | i can install it into a vmware session if needed, i think i have a 5.1 dvd somewhere around here |
| 15:22.09 | seanbright | dlynes: lsb_release -a | grep "^Release" |
| 15:22.11 | Qwell | davevg-btwtech: nah, I could do that |
| 15:22.17 | Juggie | Qwell, i'd bet if you did a legit 5.1 install, it would. |
| 15:22.51 | Juggie | sometimes yum upgrades do weird things |
| 15:23.10 | Qwell | yeah... |
| 15:23.31 | Qwell | it seems really odd to me that they'd seemingly arbitrarily version a release |
| 15:23.37 | seanbright | i'm pretty sure i installed 5.1 though |
| 15:23.45 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:23.45 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:23.46 | dlynes | seanbright: Release: 4.0 |
| 15:23.59 | seanbright | dlynes: ouch... and you're running 5.2? |
| 15:24.01 | Juggie | what would the sed command be to remove the indent at the beginning of every lnie in a file. |
| 15:24.03 | Qwell | so, they changed the format between 4 and 5. awesome |
| 15:24.09 | dlynes | seanbright: no...that's Debian Etch 4.0 |
| 15:24.10 | [TK]D-Fender | Qwell: Description: CentOS release 5.2 (Final) <- I installed 5.1, but yum's up to 5.2 |
| 15:24.26 | seanbright | Juggie: sed -er 's/^\s//g' |
| 15:24.39 | seanbright | err |
| 15:24.47 | seanbright | Juggie: sed -r -e 's/^\s//g' |
| 15:25.13 | Juggie | so, sed -r -e 's/^\s//g' filename |
| 15:25.30 | dlynes | seanbright: I'm running a mix of Debian Etch 4.0R1, R3, Centos 4.6, and Centos 5.2 |
| 15:25.50 | Juggie | i have a mix of centos 4.6 & 5.2 |
| 15:26.03 | Qwell | gentoo++ |
| 15:26.07 | dlynes | seanbright: Oh yeah...and Slackware 10.0, 11.0, and 12.0 :) |
| 15:26.37 | Juggie | seanbright, that sed no workey, solaris doesnt like -r :( |
| 15:26.50 | dlynes | but i've kinda discontinued slackware...too much of a pain in the ass to upgrade multiple machines |
| 15:26.52 | Sparkie- | anyone around with experience using chan_skinny and cisco phones? |
| 15:27.02 | dlynes | points at Qwell. |
| 15:27.12 | Juggie | seanbright, any other ideas? :) |
| 15:28.37 | dlynes | Juggie: sed -e 's/^\s*(.*)/$1/g' < filename > newfilename |
| 15:29.07 | dlynes | Juggie: might have to escape the parentheses...can't remember |
| 15:29.20 | Qwell | with $1, yeah |
| 15:29.27 | Qwell | erm, $1? not \1? |
| 15:29.39 | dlynes | Qwell: \1 for most regex parsers, but $1 for sed |
| 15:29.41 | Qwell | backreference, right? |
| 15:29.45 | Qwell | \1 works in sed |
| 15:29.48 | Juggie | so the right command is? :P |
| 15:29.53 | bijit | where does the server ip for asterisk? Is it the ip of the machine asterisk is installed and running? |
| 15:29.54 | Juggie | is just telling this to his gf |
| 15:30.01 | Juggie | she is the one with solaris boxes |
| 15:30.12 | Corydon76-dig | Juggie: use the backslash |
| 15:30.16 | dlynes | Qwell: hrm...last time I tried, had to use a '$'. |
| 15:30.23 | [TK]D-Fender | bijit: ... huh? |
| 15:30.33 | dlynes | Qwell: But i've never tried on Centos, either |
| 15:30.40 | bijit | hi [TK]D-Fender |
| 15:30.53 | dlynes | Qwell: Only used it on Slackware, Debian, and Solaris |
| 15:31.21 | Corydon76-dig | Perl is the only engine I've ever seen that used $ for backreferences |
| 15:31.40 | Juggie | dlynes, how would you do it if you just wanted to replace indents at the beginning of the line. |
| 15:31.41 | bijit | I am trying to find out if i need to add the server ip on any conf file or its just the machine ip asterisk is on... |
| 15:33.41 | dlynes | Juggie: you just wanted to strip the indents, right? |
| 15:33.49 | Juggie | dlynes, only at the beginning of the line. |
| 15:33.52 | Juggie | not any after that |
| 15:33.53 | [TK]D-Fender | bijit: Usually only the WAN IP if its behind NAT to sip.conf |
| 15:34.17 | dlynes | Juggie: sed -e 's/^\s*\(.*\)$/\1/g' < filename > newfilename |
| 15:34.34 | dlynes | Juggie: assuming Solaris sed supports the '\s' token |
| 15:34.41 | dlynes | Juggie: not all awks support that token |
| 15:35.42 | Juggie | ok that works for me on linux, i'll see if it works for her on solaris |
| 15:36.02 | Juggie | some day i'll learn regex |
| 15:36.34 | dlynes | Juggie: if solaris awk doesn't support it (I believe they use AT&T awk, not gnu awk), it might only support [ \t], instead, where '\t' would get replaced by a tab |
| 15:36.56 | dlynes | Juggie: also know that space in the range atom |
| 15:37.04 | dlynes | Juggie: s/know/note/ |
| 15:38.03 | *** join/#asterisk SwK (n=SwK@user-24-236-97-87.knology.net) |
| 15:38.08 | dlynes | Juggie: regex isn't that difficult as long as you stick to the basics...it's just difficult remembering the nuances that change from one regex interpreter to another, sometimes |
| 15:38.24 | *** join/#asterisk raz (n=y@unaffiliated/raz) |
| 15:38.28 | raz | ~nat |
| 15:38.28 | jbot | extra, extra, read all about it, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 15:38.36 | raz | ~natfaq |
| 15:38.40 | [TK]D-Fender | ~sipnat |
| 15:38.41 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:38.43 | [TK]D-Fender | ^^^^^^^^^^^ |
| 15:38.46 | raz | ah thx |
| 15:38.58 | raz | still chewing on my problem :\ |
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| 15:41.21 | Juggie | dlynes, works for me, but not for her on solaris. |
| 15:41.43 | bijit | [TK]D-Fender: thanks |
| 15:41.51 | dlynes | Juggie: And my solaris box is setting in my storage closet, or I'd be able to help |
| 15:41.55 | Juggie | :) |
| 15:41.58 | dlynes | s/setting/sitting/ |
| 15:42.05 | Juggie | well she managed to figure out how to get rid of it in vi |
| 15:42.12 | Juggie | but for a bigger file that would not work |
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| 15:43.01 | dlynes | Juggie: :%s/^\s*\(.*\)$/\1/g <-- this'll work in vim (and probably vi) |
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| 15:49.38 | *** part/#asterisk Drag0n` (n=Drag0n@net20.quickoffice.com) |
| 15:52.15 | seanbright | why do you need the back reference?? |
| 15:52.39 | seanbright | s/^[\t\n\r ]*//g |
| 15:52.41 | seanbright | done |
| 15:53.02 | dominic1 | porn: tested funcdevstate cli. Set the devstate about 100000 from asterisk-java |
| 15:53.07 | dominic1 | this function is rock stable |
| 15:53.09 | *** join/#asterisk mltlnx (n=mltlnx@75.138.164.68) |
| 15:53.20 | seanbright | Juggie: tell her to install GNU sed since all others suck |
| 15:53.21 | seanbright | :) |
| 15:53.27 | dominic1 | so I can add a kind of SIMPLE to my asterisk |
| 15:53.49 | dominic1 | just need to remove the warning message when changing the devstate |
| 15:55.32 | seanbright | Juggie: sed -e 's/^[[:space:]]*//g' |
| 15:55.39 | seanbright | Juggie: that should work with non-GNU sed as well |
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| 15:58.51 | twisted | has a new found fondness for JSON |
| 16:00.42 | dlynes | seanbright: [:space:] is a gnuism, \t doesn't work in all regex parsers |
| 16:00.50 | [TK]D-Fender | twisted: I noticed that FF3 uses it for its Bokmarks, but never delved into the file. What else uses it that you've encountered? |
| 16:01.04 | dlynes | seanbright: if it's a gnuism, it definitely won't work in AT&T vi, or AT&T sed |
| 16:01.06 | seanbright | dlynes: i ran with --posix and it still works |
| 16:01.12 | seanbright | but thanks for playing |
| 16:01.16 | seanbright | :) |
| 16:01.22 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 16:01.25 | dlynes | seanbright: posix v what? |
| 16:01.36 | seanbright | sed --posix -e 's/^[\t ]*//g' |
| 16:01.54 | seanbright | where "--posix" == "disable all GNU extensions." |
| 16:02.09 | dlynes | seanbright: like i said...it doesn't work in all regex parsers...and I said [:space:] is a gnuism, not [\t] |
| 16:02.17 | seanbright | dlynes: i ran with --posix |
| 16:02.25 | seanbright | dlynes: not sure how much clearer i can be on this |
| 16:02.27 | seanbright | <-- wins |
| 16:02.30 | *** join/#asterisk atis_work (n=atis_wor@c158.csc.lv) |
| 16:02.48 | dlynes | seanbright: you did sed --posix -e 's/^[:space:]*//g'? |
| 16:02.52 | twisted | [TK]D-Fender: well, twitter, myspace, facebook, etc. |
| 16:02.55 | seanbright | dlynes: yes |
| 16:03.11 | [TK]D-Fender | twisted: For App devel? |
| 16:03.29 | Juggie | .. \t doesnt work w/ --posix |
| 16:03.45 | seanbright | where is there a \t in the statement i just pasted? |
| 16:03.57 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:04.16 | Juggie | its not you posted 1 a few minutes ago w/ \t, the [:space:] works |
| 16:04.27 | seanbright | right, the \t was wrong |
| 16:04.29 | dlynes | seanbright: seanbright>sed --posix -e 's/^[\t ]*//g' |
| 16:04.45 | seanbright | 11:55 < seanbright> Juggie: sed -e 's/^[[:space:]]*//g' |
| 16:04.46 | seanbright | 11:55 < seanbright> Juggie: that should work with non-GNU sed as well |
| 16:04.54 | seanbright | ok, how about this |
| 16:04.56 | seanbright | stfu |
| 16:04.57 | seanbright | :) |
| 16:05.12 | seanbright | dlynes: you are right, i am wrong. constantly and consistently. |
| 16:06.06 | PakiPenguin | hi guys, |
| 16:06.09 | PakiPenguin | i have a weird issue |
| 16:06.12 | seanbright | but the real moral of the story |
| 16:06.21 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 16:06.25 | seanbright | stop using antiquated utilities |
| 16:06.29 | PakiPenguin | asterisk is not saving voicemail messages more then 60 sec , althought i have the message parameter set to 360 in voicemail.conf |
| 16:06.30 | seanbright | it's cute... or something |
| 16:06.47 | seanbright | but ridiculously impractical |
| 16:07.18 | Juggie | its solaris not my fault :) |
| 16:07.27 | seanbright | GNU sed compiles on solaris |
| 16:07.27 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 16:07.39 | Juggie | i'm sure it does but she dosnt maintain the servers |
| 16:07.42 | ManxPower | I thought the option was "messagelength" or "maxmessage" or something like that. |
| 16:07.44 | seanbright | pfft |
| 16:07.48 | seanbright | likely |
| 16:07.51 | dlynes | seanbright: gnu environments completely bastardize a solaris install, too |
| 16:07.58 | ManxPower | The voicemail.conf.sample should have an example |
| 16:08.22 | seanbright | dlynes: thanks for your input |
| 16:10.08 | seanbright | never gets tired of being right |
| 16:10.10 | seanbright | it's like crack |
| 16:10.49 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:11.24 | TrentCreek | it kills |
| 16:11.52 | twisted | [TK]D-Fender: yes, and data propogation |
| 16:12.15 | [TK]D-Fender | twisted: Cool. |
| 16:12.39 | [TK]D-Fender | "I would never do a drug named after a part of my ass" - Dennis Leary |
| 16:15.22 | *** join/#asterisk jpcansa (n=jpbenavi@201.237.61.30) |
| 16:16.49 | Nugget | heh |
| 16:20.01 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 16:20.22 | shtoom | Hi, when I try to access Graphs in gui I am getting Tool Error what is the reason ? |
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| 16:28.04 | *** join/#asterisk mighty-d (i=500@63.58.83.190.static.coldecon.com) |
| 16:28.07 | mighty-d | Hi |
| 16:28.29 | *** join/#asterisk fcois93 (n=fcois93@bagnolet.acropolistelecom.net) |
| 16:28.39 | fcois93 | hello all |
| 16:28.47 | fcois93 | I need help for realtime... |
| 16:29.13 | fcois93 | I need to do realtime static with the realtime (real) database syntax! |
| 16:29.17 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
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| 16:29.22 | outtolunc | ponders replying to shtoom something like 'it is telling you your tool is broken' <G> |
| 16:29.29 | fcois93 | did you undersdant what I need? |
| 16:29.56 | fcois93 | in fact, the realtime database in static is hard to admin |
| 16:30.01 | fcois93 | but i nee static |
| 16:30.19 | fcois93 | so if it is possible to have the dynamic syntaxe with the static realtime... |
| 16:30.24 | fcois93 | any idea? |
| 16:30.26 | mighty-d | im thinking on buying an ALIX2C2 card to get a pbx for 25-40 extensions and 10-15 concurrent calls, the card comes with a compact flash port and my vendor is suggesting me to buy a 300x CF card, the price is 70% up from the 133x card, do you know the performance comparisson between 133x and SATA 7200 RPM hard disk drive? |
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| 16:31.23 | shtoom | outtolunc: can you tell me which tool graphs uses ? nothing is changed on the server for months all of sudden I am getting this. |
| 16:31.24 | *** part/#asterisk zydoon (n=zydoon@41.225.155.169) |
| 16:31.50 | shtoom | outtolunc:Any specific area to look for ? |
| 16:32.10 | ManxPower | mighty-d: Where is the Asterisk question? |
| 16:32.29 | outtolunc | shtoom: it was a joke, but if you have issues with something 'not working all the sudden' i would check for a packager log (like yum) that might have run and loaded something new (and in your case, incompatible) |
| 16:32.41 | [TK]D-Fender | shtoom: GUI's are NOT supported in this channel |
| 16:33.02 | jmacz | Hi, I have a question regarding t.38 passthrough behavior. If I have a SIP trunk that sends me t.38 fax packages, may I asnswer this with a fax machine attached to a t.38 enabled ATA? Will this scenario work in Asterisk? |
| 16:33.04 | mighty-d | ManxPower, im worried to see if asterisk can perform well on a cf 133x card |
| 16:33.28 | shtoom | [TK]D-Fender: #asterisk-gui is dead silent. |
| 16:33.48 | [TK]D-Fender | mighty-d: Get the SATA drive... |
| 16:33.50 | ManxPower | shtoom: This is NOT 2nd level support for AsteriskGUO |
| 16:33.56 | ManxPower | or Asterisk GUI |
| 16:33.57 | [TK]D-Fender | shtoom: Not our problem. |
| 16:34.30 | mighty-d | [TK]D-Fender, the card doesnt come with a sata port |
| 16:34.35 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 16:34.55 | [TK]D-Fender | mighty-d: get a mini-pci card for it |
| 16:35.09 | mighty-d | [TK]D-Fender, so you are saying the CF is not worth it? |
| 16:35.09 | shtoom | outtolunc: I remember installing iftop thanks for you help , I'll uninstall and check it. |
| 16:35.50 | [TK]D-Fender | mighty-d: card is pricy, and there is the read/write lifespan to consider |
| 16:36.16 | coppice | pricy? :-\ |
| 16:37.09 | coppice | $20 for a 4G card + a couple of dollars for a cf to IDE adaptor isn't much |
| 16:37.27 | Sparkie- | anyone around with experience using chan_skinny and cisco phones? I'm having a bit of trouble making the speeddial's work on a 7960 |
| 16:37.39 | [TK]D-Fender | coppice: He's looking for a 300x card... |
| 16:37.45 | Qwell | Sparkie-: give up on speeddials with 1.4 |
| 16:37.54 | ManxPower | I didn't know chan_skinny support speed dials. Maybe it was added at some point? |
| 16:37.57 | Sparkie- | :( |
| 16:38.07 | coppice | so double it. its still not much |
| 16:38.10 | Qwell | ManxPower: oh, it supports it |
| 16:38.10 | Sparkie- | Yea, it says it was added. Guess it doesn't work |
| 16:38.29 | mighty-d | [TK]D-Fender, no, you got me wrong, please excuse me, im trying to figure if i can go with a 133x card |
| 16:38.32 | Qwell | Sparkie-: use 1.6 if you need chan_skinny |
| 16:38.35 | ManxPower | Qwell: not any better than any other feature of chan_skinny? |
| 16:39.03 | jaytee | I gave up on chan_skinny. My Asterisk server kept losing too much weight and started having fainting spells like an anorexic supermodel. |
| 16:39.10 | [TK]D-Fender | mighty-d: Why are you looking to use such a weak system anyways? What are you going to use along with *? |
| 16:39.21 | coppice | the slower 32G SD cards are about $100 now. amazing really |
| 16:40.17 | Sparkie- | Well, I tried chan_sccp (the b version from sourceforge) and it works ok, but the 7914 addons that I have attached to it randomly lose their speeddials and then randomly they return |
| 16:40.21 | ManxPower | mighty-d: like ALL asterisk scaling issues, the answer depends on your needs and what you are doing. If you are doing lots of call recording that is different than not doing much call recording, etc |
| 16:40.48 | mighty-d | [TK]D-Fender, its an small office rackable install, what do you think is weak the CF approach or everything including the ALIX2C2? |
| 16:40.51 | ManxPower | Most of just buy an over powered system and then not worry about performancve |
| 16:41.06 | ManxPower | mighty-d: what the HECK is an ALIX2C2? |
| 16:41.11 | [TK]D-Fender | mighty-d: It realy just not recommended... |
| 16:41.23 | [TK]D-Fender | ManxPower: AMD 500mhz x86 embedded board |
| 16:41.52 | [TK]D-Fender | ManxPower: Just like Soekris... only cheaper :) |
| 16:41.55 | mighty-d | ManxPower, well i wanted to answer you but [TK]D-Fender was faster :) |
| 16:41.57 | ManxPower | [TK]D-Fender: Ah, so mostly just "overinfo" |
| 16:42.11 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 16:42.15 | ManxPower | [TK]D-Fender: also just as underpowered for transcoding? |
| 16:42.51 | [TK]D-Fender | ManxPower: "ill advised" |
| 16:43.22 | mighty-d | [TK]D-Fender, so, your advice is not to go with ALIX2C2 at all? |
| 16:44.09 | [TK]D-Fender | mighty-d: Correct |
| 16:44.49 | mighty-d | [TK]D-Fender, ok, thanks a lot... i think im going to take it :). |
| 16:46.11 | jeev | Fender |
| 16:46.13 | jeev | y0h |
| 16:46.21 | jeev | high 5 |
| 16:48.08 | [TK]D-Fender | ManxPower: I might use something like that for a home install w/ pure voip, and no weighty transcoding. |
| 16:48.34 | [TK]D-Fender | ManxPower: but FFS, not a 40 person office! |
| 16:48.53 | jeev | is still holding his hand in the air |
| 16:49.03 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:51.20 | [TK]D-Fender | prepares for some tameshigiri |
| 16:52.37 | jeev | what the hell is your problem |
| 16:52.38 | jeev | gimme 5 |
| 16:52.56 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:54.06 | SwK | anyone familiar with Teldori? |
| 16:55.04 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 16:55.14 | *** join/#asterisk BrokenNoze (n=magic@host81-137-172-233.in-addr.btopenworld.com) |
| 16:55.49 | BrokenNoze | Hi, has anyone been having issues with MixMonitor recently? when i put a call on hold the audio goes out of sync |
| 16:56.31 | [TK]D-Fender | severs jeev's carotid, and both femeral and brachial arteries. |
| 16:56.39 | [TK]D-Fender | jeev: 5 free...on the house! |
| 16:56.59 | [TK]D-Fender | sits and watches jeev exsanguinate |
| 16:57.13 | ManxPower | [TK]D-Fender: Thanks, he was straiting to get annoyinh. |
| 16:57.18 | PakiPenguin | how long duration of a message can be stored in a longblob ( talking about voicemail storage in mysql d b ) |
| 16:57.24 | ManxPower | perhaps it's time for more coffee. |
| 16:57.38 | PakiPenguin | ? |
| 16:58.19 | ManxPower | PakiPenguin: that is the wrong question. The right question is "how big of a blob can mysqld handle"? AFTER YOU HAVE THE ANSWER TO THAT, then you can ask how much space the VM format that you have uses. |
| 16:58.41 | ManxPower | Once you have the answer to both of those questions your problem should be solved. |
| 16:59.00 | PakiPenguin | ManxPower, my voicemail messages bigger then 60 sec are not stored in the db :( |
| 16:59.05 | ManxPower | I'll bet you can guess the correct place to ask about MySQL, right? |
| 16:59.10 | PakiPenguin | :) |
| 16:59.12 | PakiPenguin | yes! |
| 16:59.14 | PakiPenguin | :p |
| 16:59.27 | ManxPower | PakiPenguin: Yes, and that is what you will have to deal with WHEN YOU'VE gotten then information you need. |
| 16:59.29 | BrokenNoze | No one else had MixMonitor issues? |
| 16:59.42 | PakiPenguin | ManxPower, alright |
| 17:00.05 | ManxPower | My guess is that you are using the WAV format for VM. |
| 17:01.03 | *** join/#asterisk fedya (n=fedya@rrcs-71-43-222-2.se.biz.rr.com) |
| 17:02.11 | [TK]D-Fender | PakiPenguin: Free just for you today only! http://www.google.ca/search?hl=en&sa=X&oi=spell&resnum=0&ct=result&cd=1&q=how+big+of+a+blob+can+mysql+handle&spell=1 |
| 17:02.19 | ManxPower | PakiPenguin: if you had included the little bit about using REALTIME you might have had your question answered today |
| 17:02.28 | PakiPenguin | :) |
| 17:02.36 | ManxPower | [TK]D-Fender: Are you going to be giving him fish for the rest of his life? |
| 17:03.08 | [TK]D-Fender | ManxPower: Sure, why not. Mercury poisoning is CUMULATIVE you know ;) |
| 17:03.33 | ManxPower | [TK]D-Fender, the #Asterisk Google Proxy |
| 17:04.17 | PakiPenguin | :) |
| 17:04.23 | PakiPenguin | ah |
| 17:04.38 | *** join/#asterisk jonsmith1982 (n=jon@82-47-175-34.cable.ubr01.donc.blueyonder.co.uk) |
| 17:06.04 | PakiPenguin | thanks [TK]D-Fender |
| 17:10.06 | *** part/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 17:10.51 | jonsmith1982 | hi, i would like to setup asterisk, at first i want to just use some kind of interface on my pc to use skype (or some other voip service). could anyone please recommend a interface on linux which would replace the ip-phones used in so many of the asterisk examples/tutorials. |
| 17:11.53 | ManxPower | ~skype |
| 17:11.54 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, all of which are poor implementations of convoluted hacks. In general, forget about using Skype with Asterisk... |
| 17:12.06 | [TK]D-Fender | jonsmith1982: Soft-phones are unrecommeded, but here are some to look for : Ekiga, kiax, twinkle |
| 17:12.17 | [TK]D-Fender | jonsmith1982: And forget skype.. |
| 17:12.29 | jonsmith1982 | yeah that was just an example. |
| 17:13.21 | ManxPower | jonsmith1982: we don't really deal with random example. |
| 17:13.28 | ManxPower | What specific thing so you want to do? |
| 17:13.38 | ManxPower | with Asterisk? |
| 17:13.48 | jonsmith1982 | experiement with it. |
| 17:14.12 | ManxPower | Then go ahead and experiment with it. Read the book. Try stuff. |
| 17:14.13 | [TK]D-Fender | jonsmith1982: Then go download the book, insall it, install a softphone and get busy |
| 17:14.15 | jonsmith1982 | thanks [TK]D-Fender |
| 17:14.15 | [TK]D-Fender | ~book |
| 17:14.16 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:14.17 | jeev | Fender |
| 17:14.18 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 17:14.22 | jeev | i moved my hand and said SYKEEEEEE |
| 17:14.24 | jeev | when you tried. |
| 17:14.46 | SwK | no one has heard of teldori |
| 17:14.58 | ManxPower | jeev: Put down the booze |
| 17:15.05 | jeev | i dont drink |
| 17:15.12 | ManxPower | then put down the drugs |
| 17:15.17 | ManxPower | or get some better ones |
| 17:15.23 | TrentCreek | just Ask Jeevs |
| 17:16.36 | ManxPower | #asterisk-cli for non-GUI questions |
| 17:16.44 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:16.48 | *** join/#asterisk aksyn (n=aksyn@jayfenton.plus.com) |
| 17:17.18 | raz | hmm how can i make two distinct inbound numbers (both SIP) refer to the same dialplan? i know i can just point both to the same context but what about the extensions? all my exten-lines are prefixed with one of the two numbers. do i just put "s" there? |
| 17:17.50 | ManxPower | exten => _patternmatchfordid2,1,Goto(did1,1) |
| 17:18.13 | raz | aah cool |
| 17:18.20 | *** join/#asterisk spokra (n=spokra@host093-179-153.sea0.speakeasy.net) |
| 17:18.20 | ManxPower | raz: "s" is matched when there IS no known dialed number, usually only seen on FXO signalled ports. |
| 17:18.22 | raz | thx that was trivial |
| 17:19.54 | *** join/#asterisk jgoddess (n=monkey@phrank.aus.us.siteprotect.com) |
| 17:22.05 | sgtpepper | has anyone ever saw this -- Got SIP response 483 "Too Many Hops" back from 190.2.12.177 |
| 17:25.27 | jonsmith1982 | [TK]D-Fender, why would using "Soft-Phones" be unrecommended? |
| 17:26.10 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
| 17:29.28 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 17:30.21 | jonsmith1982 | ip-phones will surely have similar software on the anyway. |
| 17:31.36 | [TK]D-Fender | jonsmith1982: because sound card suck. Mics suck. using some unintuitive shitty little GUI is unintuitive. |
| 17:32.09 | *** part/#asterisk ecrist (n=ecrist@t111-gw.c7200-1.bdr6.mpls.iphouse.net) |
| 17:33.01 | jonsmith1982 | i can't see how that different from ip-phones. |
| 17:33.16 | jonsmith1982 | differs* |
| 17:33.26 | errr | yeah software is just the same as a physical ip phone |
| 17:36.16 | jonsmith1982 | are there any "open" ip-phones? |
| 17:37.11 | errr | iirc there is an open snom project |
| 17:37.12 | jonsmith1982 | like openmoko i suppose the shitty unituitive gui wont be of any use on those :) |
| 17:37.22 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 17:38.57 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:40.02 | ManxPower | jonsmith1982: Then use a softphone and find out for your self. Just remember that hardphones work better and don't let the softphones give you a bad impressions of Asterisk or VoIP |
| 17:41.14 | ManxPower | [TK]D-Fender gave you his advice. Take the advice or don't take the advice, but don't go arguing with one of the most knowledgeable Asterisk people in the world about if softphones are bad. |
| 17:41.28 | tzafrir_laptop | jonsmith1982, two projects I can think of: astfin and openmoko |
| 17:41.47 | tzafrir_laptop | None of them really aim for a hardware voip phone |
| 17:42.29 | [TK]D-Fender | jonsmith1982: What part of "every audio component is inferior", and "ease of use sucks" was not painfully clear before? Are we at a new understanding now? |
| 17:42.47 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:42.48 | ManxPower | [TK]D-Fender: he' |
| 17:42.53 | ManxPower | he's not worth arguing with |
| 17:43.11 | [TK]D-Fender | jonsmith1982: on the least expensive side you could (and probably should) get an ATA to use with a regular phone at least. |
| 17:43.59 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 17:46.34 | SwK | hmmm so anyone head of teldori? hah |
| 17:46.53 | SwK | or is it just one of the plethora of asterisk UIs the never made it anywhere |
| 17:47.26 | ManxPower | SwK: Do you REALLY think a channel that is dedicated to NON-GUI Asterisk use would know anything about an Asterisk GUI? |
| 17:48.13 | SwK | ManxPower, theres enuff asterisk people that hang out in here that someone might know something about it... |
| 17:48.13 | tzafrir_laptop | <[TK]D-Fender> jonsmith1982: because sound card suck. Mics suck. using some unintuitive shitty little GUI is unintuitive. |
| 17:48.28 | tzafrir_laptop | those are actually pretty bad arguments. |
| 17:48.29 | Nugget | http://macnugget.org/stuff/asterisk-irc.txt <-- this channel |
| 17:48.45 | ManxPower | best of luck |
| 17:48.59 | SwK | dont be a dick just cause i asked a question about something you arent interested in |
| 17:49.00 | ManxPower | For people that do not want GUI talk, you can join #asterisk-cli |
| 17:49.43 | tzafrir_laptop | For the cost of a decent hard phone you can get a simple full-duplex sound card, a good head set and not be stuck with the one forced upon you by the phone vendor |
| 17:49.51 | ManxPower | Well, not interested in and totally off topic as well. |
| 17:49.53 | [TK]D-Fender | tzafrir_laptop: having your head chained to your PC via a headset, goofing around with sound card settings na dhaving to pop up interfaces and mouse around for stuff thats 2nd nature on any phone is a pretty good argument. |
| 17:50.33 | SwK | hows it off topic since this is the general asterisk channel... this isnt #asterisk-${SOMETHING_SPECIFIC} |
| 17:51.11 | tzafrir_laptop | Next: the user interface of software is much better than those of separate devices. e.g: I can copy a hpone number to them |
| 17:51.21 | ManxPower | Maybe because this channel existed and was and still used for non-gui talk. There are plenty of resources for GUI people. |
| 17:51.28 | tzafrir_laptop | It is much easier to them them and such |
| 17:51.39 | ManxPower | Heck they even seem to ask on the asterisk-users mailing list -- fortunatly they seem to be ignored there most of the time. |
| 17:52.09 | tzafrir_laptop | The main atvantage of a hardware phone is that there's a much much higher chance for it to actually work and be available |
| 17:52.25 | ManxPower | Makes about as much sense as asking a car mechanic for fashion advice. |
| 17:52.35 | SwK | 1) I didnt ask for help with the thing 2) I asked if anyone heard of it... 3) if someone has heard of it I'm perfectly willing to take further conversation off channel as to not annoy you... thank you have a nice day.... |
| 17:52.35 | [TK]D-Fender | ManxPower: I'd ease up on this one. Its a "what do you think of" question, not a "help X is broken in my unsupported GUI" |
| 17:52.40 | SwK | <PROTECTED> |
| 17:52.52 | ManxPower | SwK: How many responses did you get? |
| 17:53.04 | ManxPower | [TK]D-Fender: I'm starting to agree with you. |
| 17:53.16 | [TK]D-Fender | tzaYes... having a working and available phone is somewhat impotant to me ;) |
| 17:53.30 | [TK]D-Fender | tzafrir_laptop: ^ |
| 17:53.39 | [TK]D-Fender | \important even ;) |
| 17:54.04 | [TK]D-Fender | SwK: Your question is fine. No, yours is the first time I've ever heard about that interface |
| 17:55.07 | SwK | [TK]D-Fender, i know dood... he's just being cranky again and acting like i'm a n00b |
| 17:55.54 | [TK]D-Fender | SwK: Common backlash to a GUI question, and a bit harsh for this one. |
| 17:56.03 | [TK]D-Fender | SwK: What do you see as its upsides? |
| 17:56.43 | SwK | [TK]D-Fender, I know... i just get tired of people acting like i'm a noon when I been around longer then a large portion of people in this channel... |
| 17:57.03 | [TK]D-Fender | SwK: Its ok, try to let go a bit. |
| 17:57.16 | [TK]D-Fender | SwK: So, does this one stand out in any special way? |
| 17:57.51 | SwK | I dont see anything as the products upside... I have a family member thats the sr technology fellow at a mid sized .edu that deployed this thing around 1.0.10 (guessing by time frame its been deployed) and the vendor they used disappeared a month or so ago and now they are having issues... |
| 17:58.31 | SwK | so I was wondering if anyone was familiar with it, and if it was someone that I am familiar with I was going to try to hook them up with some paying work... |
| 18:00.55 | [TK]D-Fender | SwK: Ancient version says that nearly a dead-end right from the start. |
| 18:02.03 | SwK | yeah |
| 18:05.37 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 18:07.12 | *** join/#asterisk |dennis| (n=Dennis@200.32.231.18) |
| 18:11.34 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:11.42 | *** join/#asterisk nicox_ (n=nicox@213-33-18-237.adsl.highway.telekom.at) |
| 18:15.28 | rwaite | wow calm down cowboys |
| 18:15.45 | rwaite | somebody is gonna have a heart attack |
| 18:16.26 | SwK | nah i just get tired of people being asshats on irc (i dunno why i let irc get me cranky after all these years it just does) |
| 18:16.46 | rwaite | you know what you need is a hooker |
| 18:17.04 | SwK | ofcourse arguing on irc is like winning the shortbus race... even if you win you're still a short bus kid |
| 18:19.10 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
| 18:21.34 | *** join/#asterisk funxion (n=x@63.214.236.169) |
| 18:21.35 | Nugget | irc is like multiplayer notepad. |
| 18:21.59 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:22.05 | tzanger | that's my favourite analogy |
| 18:22.17 | tzanger | although the stealing letters mintrubbing flash game is fun |
| 18:22.26 | [TK]D-Fender | cuts Nugget and flushes the Cut&paste buffer |
| 18:22.39 | [TK]D-Fender | FTW! |
| 18:22.41 | funxion | it seems that since I've put a digium card in my * box it is now autoloading the drivers for that card, does anybody how to keep it from loading the wct4xxp module? |
| 18:22.51 | tzanger | http://web.okaygo.co.uk/apps/letters/flashcom/ |
| 18:23.02 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 18:23.11 | [TK]D-Fender | funxion: remove the kernel module. |
| 18:23.26 | funxion | not sure I understand |
| 18:23.41 | EmleyMoor | Not directly related to Asterisk, but what does Automatic connection in the SIP stack do on a Nokia N95? |
| 18:23.46 | funxion | from modules.conf? |
| 18:24.06 | [TK]D-Fender | funxion: Zaptel compiles and copies over a loadable kernel module. BLOW IT AWAY |
| 18:24.50 | SwK | MOO! |
| 18:25.03 | funxion | remove /lib/modules/2.6.18-6-686/misc/wct4xxp? |
| 18:25.19 | [TK]D-Fender | funxion: \o/ |
| 18:25.26 | funxion | thanks |
| 18:25.30 | funxion | sry I'm slow |
| 18:25.58 | [TK]D-Fender | funxion: s'ok... I've seen too many "dead halts" lately... slow I can live with (mostly) |
| 18:29.20 | funxion | lol |
| 18:29.22 | funxion | that sux |
| 18:33.52 | EmleyMoor | If a SIP connection is set to connect automatically on an N95, over a mobile data connection, does the phone try to bring the connection up to connect the SIP account, or does it only bring the SIP account up when the connection is made for some other reason or on demand? |
| 18:34.14 | EmleyMoor | (I appreciate that's not related to Asterisk, other than that you can connect it to Asterisk) |
| 18:34.36 | [TK]D-Fender | EmleyMoor: Google-able. |
| 18:41.31 | *** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) |
| 18:43.38 | *** join/#asterisk ManxPower (n=manxpowe@1.sub-70-220-14.myvzw.com) |
| 18:44.08 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 18:46.36 | *** join/#asterisk ManxPower (n=manxpowe@35.sub-70-220-26.myvzw.com) |
| 18:46.54 | ManxPower | stupid isp |
| 18:51.49 | *** join/#asterisk Peri (n=redanti@ap1.aci.on.ca) |
| 18:54.03 | *** join/#asterisk CrashHD (n=IceChat7@65.74.161.225) |
| 18:54.59 | unpaidbill | aww manx gettin fresh on the list |
| 18:55.34 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 18:56.34 | drako | whats the best php gateway for agi? |
| 18:57.13 | *** join/#asterisk stimpie (n=stimpie@84-104-6-222.cable.quicknet.nl) |
| 18:57.23 | CrashHD | anyone have issues where calls just drop |
| 18:57.25 | CrashHD | ? |
| 18:57.29 | CrashHD | running 1.4.21.1 |
| 18:57.31 | *** join/#asterisk CoryC (n=IceChat7@oh-71-55-58-7.dhcp.embarqhsd.net) |
| 18:57.48 | CrashHD | or what the reason behind it may be? |
| 18:58.22 | [TK]D-Fender | drako: ..."gateway"? |
| 18:58.28 | *** join/#asterisk hmmhesays (n=hmmhesay@70-57-192-175.farg.qwest.net) |
| 18:59.09 | GhOnDiE | drako are you using a device over nat? |
| 18:59.18 | *** join/#asterisk emiller (n=ed@UNUSED-216-37-164-100.UNUSED.epix.net) |
| 18:59.43 | drako | [TK]D-Fender, i mean, the framework, interface, etc. |
| 19:00.07 | [TK]D-Fender | drako: PHP-AGI seems to be used more than anything else. |
| 19:00.10 | emiller | if i wanted to reduce the number of rings before voicemail picks up, in which config could i do that in? say the phone rings 10 times, and i want to reduce it to 6. |
| 19:00.28 | [TK]D-Fender | drako: Given how little there is to do in AGI I can't imagine there could be much more to offer. |
| 19:00.45 | [TK]D-Fender | emiller: extensions.conf |
| 19:00.46 | GhOnDiE | hi emiller that would be set in your extensions.conf file |
| 19:00.59 | emiller | thanks guys, ill do some scouring |
| 19:01.44 | GhOnDiE | this is a good starting place http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf |
| 19:02.09 | hmmhesays | anyone ever work with an old nuera sgx-100? I'm trying to find a manual for one |
| 19:02.11 | drako | [TK]D-Fender, yes, just need something to start with. |
| 19:02.26 | drako | ill try PHP-AGI |
| 19:02.46 | *** join/#asterisk jmacz (n=jmacz@correo.motovalle.com) |
| 19:05.41 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 19:05.55 | GhOnDiE | hi hmmhesays have you tried to contact the company direct? |
| 19:06.06 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 19:06.10 | hmmhesays | I would skip the PHP-AGI class and just use a simple php socket connection to do your stuff |
| 19:06.23 | hmmhesays | GhOnDiE, yeah its an old gateway and they can't find a manual |
| 19:06.30 | GhOnDiE | lol typical |
| 19:06.58 | hmmhesays | pretty much. I have everything figured out except the routing table. Its an empty text area that you have to fill in |
| 19:07.02 | emiller | ah, found it, exten = s,n,Dial(SIP/200&SIP/201&SIP/202,40) changing it to 30. Thanks [TK]D-Fender and GhOnDiE |
| 19:07.26 | *** join/#asterisk CrashSys (i=Kumba@azrael.crashsys.com) |
| 19:07.41 | GhOnDiE | 30 would be 30 seconds |
| 19:07.47 | emiller | yup. |
| 19:07.57 | GhOnDiE | depends on where you are but here in the uk thats about 10 rings |
| 19:08.03 | GhOnDiE | :p |
| 19:08.06 | emiller | god, i hate our ISP. always dropping my ssh sessions |
| 19:08.17 | CrashSys | If I have a call ring in, start recording with monitor, then pass that call through a local channel, and start recording again, would I get two recordings in the end? |
| 19:08.20 | GhOnDiE | that website is a wealth of info and i am on it all the time |
| 19:08.59 | GhOnDiE | hi CrashSys, i have never tried that but i guess you would??? |
| 19:09.13 | [TK]D-Fender | CrashSys: Yes, each channel's inherent recording is separate |
| 19:09.48 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 19:09.55 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 19:12.30 | *** join/#asterisk hi365_m (n=hi365@213.151.63.7) |
| 19:17.01 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 19:20.08 | *** join/#asterisk mags2 (n=mags2@ampulex.whoi.edu) |
| 19:21.36 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:24.57 | *** join/#asterisk Tond (n=t@CPE001a70b2449d-CM00194747ae5e.cpe.net.cable.rogers.com) |
| 19:25.22 | CrashHD | what does this mean? : "chan_sip.c:1952 retrans_pkt: Cancelling retransmit of OPTIONs"? |
| 19:25.47 | oej | Maybe we got a reply |
| 19:25.57 | CrashHD | ahh |
| 19:26.02 | CrashHD | so a sip options was send |
| 19:26.16 | CrashHD | another one queued up because it hadn't got a response? then we get one so we cancel the request? |
| 19:26.18 | Tond | Hi, i use Regexten=1111 in my iax peer 2222 (which is set to friend) and when the peers registers and i dial it using Dial(IAX2/1111) it says host not available |
| 19:26.30 | CrashHD | oej, btw, your blog posts are a must read |
| 19:26.35 | CrashHD | I enjoy them |
| 19:26.39 | [TK]D-Fender | CrashHD: If you're retransmitting, of course that means you've already transmitted :) |
| 19:26.52 | *** join/#asterisk Shotygun (n=thorn@82.166.243.195) |
| 19:26.55 | Tond | does that regexten command work? if so how do i call the extensoin i have registered using it? |
| 19:27.07 | CrashHD | tk: just clarifying and restating so I understand |
| 19:27.10 | [TK]D-Fender | Tond: Yes it works. |
| 19:27.23 | [TK]D-Fender | Tond: If just has NOTHING to do with what you think its for. |
| 19:27.47 | [TK]D-Fender | Tond: Go look it up on the WIKI |
| 19:27.49 | [TK]D-Fender | ~wiksi |
| 19:27.52 | [TK]D-Fender | ~wiki |
| 19:27.54 | [TK]D-Fender | ~wikis |
| 19:27.55 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 19:28.02 | Shotygun | Hi, here is a bit OT question, Does anyone here know any tool to test out the quality of SIP termination by setting a probe on the terminating side and probe over the originiation side? Looking for SIP Termination <-> DID Provider quality test. |
| 19:28.10 | Tond | TKD: oh.. lol k, tnx,, i did, and thought it was going to do what i wanted it to |
| 19:28.51 | Tond | http://www.voip-info.org/wiki/view/Asterisk+sip+regexten nothing is here though |
| 19:29.49 | *** join/#asterisk skaboy` (i=pinguino@87.22.143.193) |
| 19:29.52 | *** join/#asterisk Jerjer[mobile] (n=PhatJ@24-231-253-65.dhcp.aldl.mi.charter.com) |
| 19:30.08 | *** join/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 19:30.43 | Tond | can u quickly explain what regexten does? I can't seem to find any doc on it |
| 19:30.55 | *** part/#asterisk skaboy` (i=pinguino@87.22.143.193) |
| 19:31.38 | angryuser | the try of the day :TK maybe talkative ? |
| 19:32.06 | [TK]D-Fender | Tond: http://www.voip-info.org/wiki/view/Asterisk+sip+regcontext |
| 19:32.14 | CrashHD | any sip network testing tools out there? to load/quality test connections between asterisk and sip phone? |
| 19:32.52 | CrashHD | oej, can you see any issue with running 20+ asterisk daemons on a single linux instance? timing problems or otherwise? |
| 19:33.00 | Shotygun | CrashHD: There is SIPp for stress testing & somewhat beyond. |
| 19:33.17 | CrashHD | shotygun: thank you I'll give it a look |
| 19:33.27 | *** join/#asterisk skaboy` (i=pinguino@2001:470:904e:a:0:0:0:0) |
| 19:35.31 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 19:35.45 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 19:36.41 | CrashHD | ~centos52bug |
| 19:36.42 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 19:37.33 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 19:37.59 | dioedu | hello, is there some way to do a 3rd party call with another application than meetme ? |
| 19:38.05 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 19:38.09 | dioedu | out of telephone... |
| 19:38.48 | [TK]D-Fender | dioedu: huh? |
| 19:39.36 | dioedu | i wanna do a 3rd party call |
| 19:40.37 | dioedu | and i don't know the application that i need to use |
| 19:41.08 | dioedu | to join the 3rd party to the call... |
| 19:41.21 | M1s3ry | dioedu, we're not sure of the functionality your looking to do |
| 19:41.33 | M1s3ry | are you trying to do a conference call? explian in more detail plz |
| 19:43.07 | [TK]D-Fender | dioedu: What phone are you using? |
| 19:43.56 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 19:46.08 | dioedu | [TK]D-Fender, that's the problem... my phone don't have this feature... and i wanna know if there is some way to do that |
| 19:46.27 | *** join/#asterisk FoneHome (n=ben@75.145.224.205) |
| 19:46.27 | [TK]D-Fender | dioedu: just answer questions directly please. |
| 19:46.37 | dioedu | in this case X-Lite |
| 19:46.57 | dioedu | but i have another cases... |
| 19:47.16 | FoneHome | [TK]D-Fender: Wanted to thank you for your help the other day on the BLF issues. Problem resolved, changed to peers and upgraded to 1.4.21 seemed to solve the problems |
| 19:47.29 | [TK]D-Fender | FoneHome: You're welcome |
| 19:48.13 | dioedu | M1s3ry, sorry my poor explain... but i wanna add another channel in the current call |
| 19:49.39 | FoneHome | ODD Issue: every now and then asterisk will iniate a random call to extension 101, the caller id shows up as asterisk, but the line is dead when you pick it up, I have seen this happen before when a POTS line was unplugged and replugged in however there has been none of that. Anyone have any ideal what to look for as a call? I am using TDM800 card with 8 FXO adapters and echo cancellation. |
| 19:49.45 | dioedu | like some ip phones do with the conference button (i.e. grandstream) |
| 19:49.49 | GhOnDiE | dioedu so you want to do a 3 way call |
| 19:49.53 | GhOnDiE | ? |
| 19:49.57 | dioedu | :p |
| 19:50.00 | dioedu | yes |
| 19:50.10 | GhOnDiE | ok well thats the same as pressing conf button |
| 19:50.17 | dioedu | sorry... is not a 3rd party... a 3 way |
| 19:50.22 | [TK]D-Fender | dioedu: Yes, it DOES appear to support conference |
| 19:50.29 | [TK]D-Fender | dioedu: I just did it myself with X-Lite. |
| 19:50.45 | dioedu | let me see... |
| 19:51.33 | dioedu | but if the phone doesn't have this facility... i can't do that... right ? |
| 19:51.59 | ManxPower | As I understand it, GS BT101 does not have the conference/3-way feature in it's firmware |
| 19:52.10 | ManxPower | ~gs |
| 19:52.10 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 19:52.23 | [TK]D-Fender | dioedu: HUH? I just did this with X-Lite. It is capable. Please be precise about the circumstances you are asking about. |
| 19:52.53 | *** join/#asterisk asdf_13 (n=i13@pool-151-198-7-77.nwrk.east.verizon.net) |
| 19:54.32 | *** join/#asterisk aksyn (n=aksyn@jayfenton.plus.com) |
| 19:55.05 | Tond | i am having a very wierd problem, if i put R or r in my Dial command, the ring back sounds very wierd choopy and many times just plays the first ring and then silence until someone picks up |
| 19:55.10 | Tond | any ideas why? |
| 19:55.15 | ManxPower | [TK]D-Fender: X-LITE or X-PRO? For some reason I thought only the pay version supported 3-way calls |
| 19:55.27 | [TK]D-Fender | ManxPower: X-Lite. |
| 19:55.34 | ManxPower | Tond: that would be expected. That's why you should not use those options |
| 19:55.48 | [TK]D-Fender | ManxPower: I read what he asked and I tested it personally immediately following his question. |
| 19:55.50 | dioedu | [TK]D-Fender, My X-Lite doesn't have the button "conf" enabled |
| 19:55.53 | Tond | Ya, but then my router won't play any ringback either |
| 19:55.54 | ManxPower | Asterisk will provide ringing sound by default. If it's not doing that then it's a problem |
| 19:56.04 | *** join/#asterisk schemathings (n=schemath@173.7.254.35) |
| 19:56.06 | tzanger | ManxPower: why would that be expected? I use r when dialing my cell and never get weird ringback |
| 19:56.11 | [TK]D-Fender | Tond: "router"? Routers don't make sound |
| 19:56.19 | tzanger | [TK]D-Fender: if the fans go, they do :-) |
| 19:56.30 | Tond | ManxPower> ast is doing it by default.. |
| 19:56.32 | ManxPower | Tond: tzanger I'll bet you would if you were using a compressed codec and the ringback was inband because of an answer somewhere |
| 19:56.38 | [TK]D-Fender | tzanger: Actually... if the fans go, they STOP ;) |
| 19:56.50 | dioedu | [TK]D-Fender, than the answer is... you don't know if we can do the 3-way calls out of the phone... |
| 19:57.18 | ManxPower | dioedu: no, the answer is we don't know why your conference button is not enabled. Contact X-Lite to find out why. |
| 19:57.20 | tzanger | ManxPower: hmm, depends on where you are I suppose... I don't think ringback is a continuous tone, but yeah, g729 ringback could very likely sound like ass |
| 19:57.23 | [TK]D-Fender | dioedu: WTF is "the phone". I have just told you TWICE that YES, X-Lite can do 3-way calls |
| 19:57.25 | tzanger | gsm ringback isn't bad |
| 19:57.26 | ManxPower | It has NOTHING to do with ASterisk |
| 19:57.49 | [TK]D-Fender | ManxPower: Yes, you just need to know how to use it |
| 19:57.52 | ManxPower | dioedu: you are using a Softphone. "Phone" is a generic term here and pretty much useless. |
| 19:58.06 | Tond | TKD> well i thought Asterisk will send session in progress and then, router will transmit that to the seitch to get the ring |
| 19:58.07 | ManxPower | [TK]D-Fender: OK. How do you make the conference button not be greyed out? |
| 19:58.34 | ManxPower | Tond: no. "r" overrides all other sounds and just plays a ringing audio sound. |
| 19:58.49 | [TK]D-Fender | ManxPower, dioedu : place call 1. Place call 2 (#1 "holds" automatically), Press "Conf". DONE. |
| 19:58.53 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 19:58.55 | Tond | ManxPower> i even tried it without R or r, and it is still doing the same thing |
| 19:58.56 | dioedu | ManxPower, i'm not asking about x-lite... or another softphones or hardphones, i wanna know if there is some way to do 3-way calls with asterisk |
| 19:59.14 | dioedu | if there is some application to do that |
| 19:59.16 | ManxPower | Tond: Then we need to find out why you are having problem. |
| 19:59.20 | cesar_CR | hello guys what does mean this error ??? [Jul 22 13:53:18] WARNING[15354]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 19:59.22 | ManxPower | dioedu: yes. It is called meetme |
| 19:59.24 | [TK]D-Fender | dioedu: * doesn't do "3-way calls". DEVICES do 3-way calls |
| 19:59.34 | dioedu | ok |
| 19:59.39 | dioedu | thanks |
| 19:59.43 | dioedu | genius |
| 19:59.46 | ManxPower | technically MeetMe is an n-way conferencing app. |
| 19:59.49 | [TK]D-Fender | dioedu: And for CONFERNCING, there is MeetMe, and app_conference |
| 19:59.51 | ManxPower | not a 3-way app |
| 20:00.08 | tzanger | Tond: if you removed the r/R options and it's still occuring, take a look at your provider |
| 20:00.19 | ManxPower | but yes, asterisk does not do 3-way calls. DEVICES create 3-way calls. |
| 20:00.45 | dioedu | ok... |
| 20:01.08 | ManxPower | cesar_CR: I'll answer your question on #asterisk-cli |
| 20:01.08 | Tond | tzanger> it is not going to a provider, it is going to a local IAX peer that has been registered |
| 20:01.27 | tzanger | Tond: describe the connection |
| 20:01.33 | asdf_13 | hey is it normal that the IVR doesn't respond immediately to button presses? I usually have to wait till the end of the message for it to recognize that I press a number |
| 20:01.35 | tzanger | you are using what to place the call which is going to the iax2 peer |
| 20:01.47 | ManxPower | asdf_13: only if you have a poorly designed IVR |
| 20:02.20 | [TK]D-Fender | asdf_13: then you are likely using PLAYBACK, where you should be using BACKGROUND |
| 20:02.21 | asdf_13 | ManxPower: ummm... what would define a poorly designed IVR? |
| 20:02.37 | asdf_13 | [TK]D-Fender: Oh i see |
| 20:02.46 | ManxPower | asdf_13: Any of a million things, but without more info I really can't say. TK's idea is one possibility |
| 20:03.08 | asdf_13 | ManxPower: hey I appreciate the help. thanks |
| 20:03.19 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 20:03.31 | ManxPower | overlapping options/extensions is also another thing that could be bad |
| 20:03.31 | asdf_13 | [TK]D-Fender: thanks. i'll check it out |
| 20:03.32 | Tond | Ok, I am calling a DID that goes through my PRI to Asterisk, and asterisk will forward that DID to the IAX peer using AGI. I have noticed that this problem only occurs when dial is done from AGI. If i directly forward it in Extensoins.conf, i will get perfect ringbacks |
| 20:03.37 | [TK]D-Fender | ManxPower: Forced wait implies the dialplan is in the way. He didn't say interdigit, or post-dial delay. So I read into it a little. |
| 20:03.58 | cesar_CR | ManxPower, ok |
| 20:04.14 | ManxPower | [TK]D-Fender: yeah, but if he used Playback for an IVR, dog knows what other stuff he did to the poor innocent helpless dialplan. |
| 20:04.17 | [TK]D-Fender | Tond: "dial" is "dial" |
| 20:04.23 | Tond | i know |
| 20:04.28 | Tond | that is why i am so confused |
| 20:04.46 | Tond | i look in the consol and i see it is dialing it the right way without r option |
| 20:05.11 | dioedu | and X-Lite doesn't have 3-Way calling support... just eyeBeam and bria... |
| 20:05.11 | *** join/#asterisk Jerjer[mobile] (n=PhatJ@24-231-253-65.dhcp.aldl.mi.charter.com) |
| 20:05.14 | ManxPower | Tond: somewhere your answering the line and so any future ringback will be inband audio, which sounds like shit. |
| 20:05.58 | Tond | ManPower, yaa! That's it I think! |
| 20:06.10 | Tond | ManxPower, sorry |
| 20:06.13 | Tond | :) |
| 20:06.15 | ManxPower | Tond: never answer unless you have to. |
| 20:06.26 | ManxPower | and be careful of using apps that automatically answer |
| 20:06.37 | Tond | Ok, tnx |
| 20:07.01 | [TK]D-Fender | dioedu: last ^&%#$ing time : I just did it in X-Lite MYSELF <- AND I just gave you bloody step-by-step instructions! |
| 20:07.09 | ManxPower | Tond: I'll bet if you switch to ulaw or alaw as the codec for all parts of the call it will sound fine. |
| 20:08.27 | *** part/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:08.33 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:09.53 | Tond | ManxPower> hrm.. I will try that |
| 20:10.11 | jeev | hmm |
| 20:10.36 | ManxPower | Tond: both codes handle ringback audio just fine. If switching codecs makes it sound good, then you DO have an answer somewere in the call path |
| 20:11.27 | *** join/#asterisk mgdm_ (n=michael@serenity.mgdm.net) |
| 20:11.27 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 20:11.47 | oej | Seems like a very common question here is about SIP retransmits. |
| 20:11.54 | oej | I wrote this document to explain: http://svn.digium.com/view/asterisk/branches/1.4/doc/sip-retransmit.txt?view=co |
| 20:12.01 | oej | Maybe we can teach jbot about it |
| 20:12.02 | oej | :-) |
| 20:12.12 | Tond | ManxPower> Thanks dude, i made it so it won't answer the call |
| 20:12.24 | Tond | ManxPower> It worked like a charm |
| 20:12.25 | Tond | :) |
| 20:14.15 | [TK]D-Fender | ~sipretransmit |
| 20:14.16 | jbot | [~sipretransmit] to learn more about SIP retransmits, please read : http://svn.digium.com/view/asterisk/branches/1.4/doc/sip-retransmit.txt?view=co |
| 20:14.20 | [TK]D-Fender | oej: There |
| 20:17.23 | ManxPower | who "owns" jbot (i.e. can make it join channels)? /msg me. |
| 20:18.54 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 20:22.16 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 20:26.58 | dioedu | lol |
| 20:29.50 | [TK]D-Fender | ok, checkout time, heading home. Later all |
| 20:30.40 | neurosys | wow les.net has some great international rates |
| 20:31.05 | *** join/#asterisk sniper_sniper (i=michofr@80.77.188.63) |
| 20:32.03 | *** join/#asterisk mintee (n=mintone@75.150.132.150) |
| 20:32.37 | mintee | hey all, anyone know off the top of your heads what number you can dial that will read back your CallerID? |
| 20:32.53 | mintee | i know there was an MCI number I used last year, but I forgot what it was. |
| 20:34.05 | *** join/#asterisk Jerjer[mobile] (n=PhatJ@24-231-253-65.dhcp.aldl.mi.charter.com) |
| 20:34.13 | *** join/#asterisk VaNNi (n=VaNNi___@38.98.61.142) |
| 20:37.46 | ManxPower | mintee: 985-246-3704 option 93, the 9 and the 1 are added to the callerid when it comes in, so you can ignore that, other than that the callerid readback is what was received |
| 20:38.48 | Shotygun | Interesting, is there any echo test option of MCI as well? |
| 20:39.32 | ManxPower | echo test is option 91 on my system services application |
| 20:39.38 | ManxPower | same number as the readback |
| 20:40.59 | ManxPower | and nobody has tried it. LOL! |
| 20:42.02 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 20:42.08 | TrentCreek | me |
| 20:42.47 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 20:43.33 | ManxPower | I see that someone did now |
| 20:43.51 | jaytee | I just tried that number |
| 20:43.57 | jaytee | with the option 93 |
| 20:44.04 | *** join/#asterisk SwK (n=SwK@user-24-236-97-87.knology.net) |
| 20:45.21 | ManxPower | The DID and BTN stuff are set internally. If calling from the outside, those would be empty |
| 20:46.42 | *** join/#asterisk watchy (n=watchy@adsl-69-152-41-251.dsl.ltrkar.swbell.net) |
| 20:46.44 | watchy | yo |
| 20:46.55 | oej | Thanks for the jbot fix! |
| 20:46.55 | watchy | ERROR[5422]: asterisk.c:2982 main: Asterisk has detected a problem with your Zaptel configuration and will shutdown for your protection. You have options: |
| 20:47.01 | watchy | what in the hell does that mean |
| 20:47.36 | ManxPower | Here's the full info: +1-985-246-3704. Quiet Line Test: 90, Echo Test 91, Milliwatt Test 92, Caller*ID Readback 93, |
| 20:47.44 | Shotygun | ManxPower: the 985-246-3704 is yours? |
| 20:47.51 | ManxPower | Shotygun: yes. |
| 20:47.59 | jaytee | watchy, it means your zaptel configuration is wrong or your hardware is flaky. |
| 20:48.07 | Shotygun | ManxPower: Based on what info you pulling out the caller id readback ? |
| 20:48.34 | ManxPower | Shotygun: ${CALLERID(num)} and ${CALLERID(name)}. |
| 20:48.59 | ManxPower | As I said, when the call comes into the system, we automatically preprocess the callerid to add a 9 a 1 and some dashes to make our users happy. |
| 20:49.11 | ManxPower | you're not really supposed to do that, but it does work in our enviroment. |
| 20:50.32 | ManxPower | Telco -> PRI -> Tellabs commercial echo canceling system -> Asterisk |
| 20:50.48 | *** join/#asterisk udzinari (n=david@host-88-210-253-165.adsl.caucasus.net) |
| 20:51.16 | Shotygun | ManxPower: Thanks to you I caught some issue with my termiating service that I confirmed with my own echo test asterisk, thanks =) |
| 20:51.29 | ManxPower | Shotygun: what issue? |
| 20:51.51 | Shotygun | ManxPower: Nothing serious, just that if I mark the cli is unavailable it makes up one on his own |
| 20:51.57 | Shotygun | instead of showing forbidden or whatever |
| 20:51.59 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:52.24 | Shotygun | I wasn't aware of this behaviour |
| 20:52.49 | ManxPower | Shotygun: I also have a DID setup to make a blast of calls to a local cab company that you can never get thru to during holidays. Dialing direct from the cell = 20 mins to get thru, dialing via call blast DID, 20 seconds to get thnru. |
| 20:53.16 | *** join/#asterisk tessier__ (n=treed@wsip-68-15-4-27.sd.sd.cox.net) |
| 20:53.18 | tessier__ | Hello all! |
| 20:53.26 | Shotygun | lol |
| 20:53.33 | ManxPower | sequentially, not in paralell, but it can so about 2 calls per second because we are on PRI |
| 20:53.45 | Shotygun | I have done something similar but for waking up somebody =) |
| 20:53.48 | *** join/#asterisk aksyn (n=aksyn@jayfenton.plus.com) |
| 20:53.57 | jaytee | ManxPower, the perfect example of "when all else fails, get a bigger hammer." |
| 20:54.12 | hi365_m | is there any way to use sed or grap or something to show an asterisk context (i.e. jsut that context)? |
| 20:54.17 | ManxPower | jaytee: Well a faster dialer, at least |
| 20:54.25 | jaytee | yep indeedy |
| 20:54.36 | watchy | hmm |
| 20:54.43 | watchy | i think it must be my hardware jaytee |
| 20:55.10 | watchy | im trying some wildcat wxp100s or whatever they are called |
| 20:55.36 | jaytee | some people call them clone pieces of [expletive deleted] |
| 20:55.48 | watchy | yea they are |
| 20:56.05 | ManxPower | jaytee: I call them "not manufactured in 5 years" |
| 20:56.07 | jaytee | ya get what ya pay for and often you get less |
| 20:56.26 | Shotygun | I hope you guys don't mind, gonna repeat a question I posted 30 minutes ago: Does anyone here know any tool to test out the quality of SIP termination by setting a probe on the terminating side and probe over the originiation side? Looking for end to end (Caller->SIP termination->PSTN world->SIP origination[DID]->Callee) voice quality measurement. |
| 20:56.43 | jaytee | to paraphrase P.T. Barnum, "There's a sucker born every minute and Ebay will get every one of them" |
| 20:56.47 | watchy | well my boss is a complete idiot |
| 20:56.50 | tessier__ | I have two accounts with teliax. So I have two auth lines under [authentication] but all calls get charged to the last auth line I specify. Anyone know how to fix this? |
| 20:56.59 | watchy | he sold a phone system and we didnt order the equipment till yesterday |
| 20:57.07 | watchy | they wanted the phone system monday |
| 20:57.07 | rwaite | i have a problem with echo but its only internal. using sip phones and a tdm400p. when calls come in from the pots on the tdm card, i can hear echo of myself, but the calling party hears no echo |
| 20:57.12 | rwaite | what could be causing this? |
| 20:57.13 | *** join/#asterisk Greek-Boy (n=email@41.222.89.77) |
| 20:57.14 | ManxPower | tessier__: We don't do GUIs here? |
| 20:57.14 | jaytee | Shotygun, wireshark on each end |
| 20:57.21 | watchy | the only pots cards i have a freakin wildcards |
| 20:57.33 | tessier__ | ManxPower: I'm not using a gui. I'm talking about in sip.conf |
| 20:57.44 | jaytee | Shotygun, sorry, that won't work for voice "quality" |
| 20:57.46 | ManxPower | tessier__: are you using 1.6? |
| 20:57.50 | Shotygun | jaytee: It's going through the PSTN world, do you think comparing RTP packets will be reliable? I suspect they will be very diff at the two points. |
| 20:57.57 | Shotygun | It's good for SIP to SIP check. |
| 20:58.05 | ManxPower | because in 1.4 and lower [authentication] is a sip userid |
| 20:58.10 | jaytee | yeah, SIP to SIP but not voice quality |
| 20:58.14 | tessier__ | ManxPower: Nope. 1.4.16 |
| 20:58.29 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 20:58.34 | jaytee | and with that many points in between that you can't measure at you're pretty much screwed. |
| 20:58.52 | Shotygun | If you add latency, jitter, packet loss & checksum of rtps then you can measure quality, but that only applies to SIP to SIP. |
| 20:58.55 | ManxPower | Put your sip.conf masking ONLY passwords (you can remove any IP phone stuff too) and I'll help you on #asterisk-cli. |
| 20:59.11 | ManxPower | I think you guys mean RTP, not SIP! |
| 20:59.20 | Shotygun | I need something that can do like voice analysis at codec level or something fancy like that |
| 20:59.24 | Shotygun | ManxPower: You are right |
| 20:59.34 | ManxPower | anyone else )not using a gui) is welcome to join #asterisk-cli as well. |
| 21:02.50 | cesar_CR | ManxPower, I am going there again.. great help! |
| 21:05.56 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 21:06.15 | *** join/#asterisk pabelanger (n=pabelang@OTWAON23-1242541604.sdsl.bell.ca) |
| 21:10.07 | angryuser | ManxPower i am not sure i can handle questions 'where do i click' ;) |
| 21:10.10 | *** join/#asterisk gramulhaozin (n=charles@c-76-110-242-178.hsd1.fl.comcast.net) |
| 21:12.18 | watchy | well * starts with a tdm |
| 21:12.33 | *** part/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 21:12.36 | watchy | i guess this vostro 200 dont like X100s |
| 21:13.53 | *** join/#asterisk mikeshank (n=sam@c-68-37-250-134.hsd1.pa.comcast.net) |
| 21:16.00 | *** join/#asterisk Dovid (n=Dovid@tony09-121-90.inter.net.il) |
| 21:16.21 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 21:16.24 | [TK]D-Fender | watchy: Dell is often trouble all by itself, and you compound it with the use of shitty cards. YGWYPF |
| 21:16.39 | Dovid | afternoon TK |
| 21:17.40 | watchy | tk: dont make me slap u son, this is what my idiot boss sells |
| 21:18.30 | [TK]D-Fender | watchy: You're right. thats misdirected anger. Slap HIM :p |
| 21:18.38 | watchy | haha |
| 21:18.41 | *** join/#asterisk SwK (n=SwK@user-24-236-97-87.knology.net) |
| 21:18.52 | watchy | well what sucks is he sold a phone system 2 months ago that was scheduled to go in monday |
| 21:18.59 | watchy | but he didnt order the parts till yesterday |
| 21:19.01 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 21:19.04 | watchy | they are highly pissed |
| 21:19.30 | outtolunc | as well they should be <G> |
| 21:19.41 | angryuser | the famous dell speed delivery ;) |
| 21:19.43 | *** join/#asterisk LevonK (n=levon@nat/yahoo/x-37e9d0f76837f506) |
| 21:20.51 | watchy | i actually get my dells quick |
| 21:21.01 | watchy | but im waiting on polycoms and sangoma cards |
| 21:22.00 | Dovid | watchy: sounds like u have a great boss ;) |
| 21:22.47 | mikeshank | Hi all. I'm really new to this stuff and pretty confused at this point. Here's my scenario, I have asterisk installed, a grandstream gxp-2000 and broadvoice as my service. My problem is I cant get the phone to actually ring, but when i dial my number I can see its routed to asterisk because I get a warning/error saying it cannot find an extension (my number). |
| 21:23.32 | Dovid | 2 problems 1) You are using grandstream 2) You are using broadvoice ;) |
| 21:24.03 | Dovid | chances are you do not have it configure properlt which is why it cant get to your phone. look at the error. it cant get to urphone |
| 21:24.06 | Dovid | ur phone* |
| 21:24.20 | watchy | i want hugz |
| 21:25.07 | mikeshank | Dovid: you mean the phone is not confgured correctly |
| 21:25.24 | `Sean | is there any Stable Fax over IP soloutions Now? |
| 21:26.56 | mikeshank | Dovid: also, if my setup blows, what in your opinion should i be using |
| 21:27.38 | ManxPower | ~ipphones |
| 21:27.43 | ManxPower | ~sipphones |
| 21:27.53 | ManxPower | ~phones |
| 21:27.54 | jbot | phones is probably http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 21:28.03 | Dovid | mikeshank: no -ur asterisk is not set up correclty |
| 21:28.10 | Dovid | post your extensions.conf |
| 21:28.12 | Dovid | ~pb |
| 21:28.12 | 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 |
| 21:28.13 | ManxPower | mikeshank: there is the answer to one part of that question |
| 21:28.58 | Dovid | ManxPower: As per the * users list if you have the latest firmware (and only the latest firmware) you can use it as a door stopper |
| 21:29.33 | TrentCreek | dammed huricane |
| 21:29.44 | TrentCreek | damn you all to hell |
| 21:30.17 | TrentCreek | shutting down........... |
| 21:30.35 | Dovid | TrentCreek: Move !!! LA seems to be nice |
| 21:31.03 | TrentCreek | yeah, if I want to live in Mexio |
| 21:31.14 | CrashHD | anyone with experience/knowledge of running asterisk in vmware? |
| 21:31.15 | TrentCreek | Los Angeles, mexico |
| 21:31.35 | ManxPower | What's wrong with Mexico? |
| 21:31.45 | TrentCreek | mexico needs to stay there |
| 21:31.49 | Dovid | hehe |
| 21:32.00 | TrentCreek | If I wanted to live there I would move, not go to LA |
| 21:32.16 | ManxPower | Modern cities, varied people's and climate, great food. |
| 21:32.16 | mikeshank | Dovid: http://pastebin.com/m21d1b98c |
| 21:32.47 | Dovid | mikeshark: what context do you have set for your number in sip.conf ? |
| 21:33.27 | Dovid | mikeshark: post your sip.conf (with out the passwords ;) ) |
| 21:34.09 | Dovid | also you do not have anything there to send the call to your phone. |
| 21:34.41 | Dovid | as well as under phones you have include incoming all that says is that if some one dials s from the phoens extension it will play the welcome message |
| 21:35.44 | mikeshank | Dovid: http://pastebin.com/m6d97259c |
| 21:36.39 | Dovid | mikeshark: Here is your issue |
| 21:36.56 | Dovid | under the broadvoice section in sip.conf you have the context set to context=from-broadvoice |
| 21:37.06 | Dovid | however you do not have such a context |
| 21:37.43 | ManxPower | in sip.conf context=whateverisheremustalsobeinextensions.conf |
| 21:38.02 | Dovid | if you switched it to context=default then when they called it will go to the default context is extensions.conf. as per your configurations it will pick up, say hello world and then hang up |
| 21:40.47 | cesar_CR | hey guys I've got asterisk running thanks to #asterisk-cli !!!! |
| 21:40.56 | [hC] | cesar_CR: hows it going man? |
| 21:40.57 | mikeshank | Dovid: the context for [sip.broadvoice.com] change to default |
| 21:41.09 | cesar_CR | hey [hC] great !!! |
| 21:41.20 | Dovid | mikeshark: under context= under the broadvoice section |
| 21:41.26 | [hC] | cesar_CR: hey do you work in san jose, i forget? I will be there in a couple weeks. |
| 21:41.33 | ManxPower-Away | [hC]: mind keeping an eye on #asterisk-cli as well as here? |
| 21:41.50 | [hC] | ManxPower-Away: ? I didnt know #asterisk-cli existed.. |
| 21:42.05 | Dovid | neither did i |
| 21:42.05 | ManxPower-Away | [hC]: it didn't until last Wed |
| 21:42.09 | cesar_CR | yes I'm here in San Jose |
| 21:42.30 | cesar_CR | [hC], great guys there :D |
| 21:43.11 | [hC] | cesar_CR: we should get together for an imperial or two |
| 21:43.41 | mikeshank | Dovid: no dice, changed the context= default still seeing the same error, says call from 'mynumber' to extension 'mynumber' rejected because extension not found |
| 21:43.44 | cesar_CR | great are you here ?? did you try pilsen ? |
| 21:43.50 | *** part/#asterisk pabelanger (n=pabelang@OTWAON23-1242541604.sdsl.bell.ca) |
| 21:43.51 | cesar_CR | [hC], ? |
| 21:44.13 | [hC] | cesar_CR: I will be there on aug 7th i think.. I have tried pilsen, of course... I lived there for 3 years! |
| 21:44.17 | [hC] | cesar_CR: and i go back every 2 months |
| 21:44.37 | Dovid | mikeshank: Ok |
| 21:44.47 | Dovid | seems like they are sending the call to the number |
| 21:44.57 | Dovid | try putting this in under default |
| 21:45.19 | cesar_CR | [hC], great I love pilsen for me is better, aug, great, not in sept, cause I will not be here |
| 21:46.37 | Dovid | http://www.pastebin.ca/1079999 |
| 21:46.45 | Dovid | so we can see where they are sending it to |
| 21:48.06 | mikeshank | so i should comment out what i have in default and add this? |
| 21:48.49 | *** join/#asterisk Psykick (n=anon@125-237-108-243.jetstream.xtra.co.nz) |
| 21:48.51 | Dovid | mikeshark: add it |
| 21:48.52 | Psykick | hi guys |
| 21:49.27 | *** join/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 21:49.32 | Psykick | is it possible in manager.conf to permit only certain IP addresses to connect to as opposed to defining that for each user? |
| 21:49.39 | *** join/#asterisk Ksilebo|Home (n=ksilebo@c-76-120-84-103.hsd1.co.comcast.net) |
| 21:49.56 | *** join/#asterisk dmz (n=dmz@216.16.220.180.dyn-cm-pool48.pool.hargray.net) |
| 21:50.48 | *** join/#asterisk udzinari (n=david@host-88-210-253-165.adsl.caucasus.net) |
| 21:50.51 | Dovid | psykick: yes |
| 21:51.02 | GhOnDiE | psykick i guess you could probably do it in the [general] section |
| 21:51.28 | Psykick | Dovid: same syntax as is used in the users sections? |
| 21:51.42 | Dovid | hOnDiE has a point |
| 21:51.46 | GhOnDiE | permet 192.168.0.1/255.255.255.0 |
| 21:51.48 | Dovid | prob under the general section |
| 21:51.53 | GhOnDiE | in genral section i believe |
| 21:51.56 | mikeshank | Dovid: now when i call it doesnt seem to connect, just disconnect |
| 21:51.59 | GhOnDiE | permit even |
| 21:52.14 | Dovid | what comes up in the CLI ? |
| 21:52.16 | Psykick | GhOnDiE: just a comma seperated list right? |
| 21:52.16 | ManxPower-Away | why don't you look at the manager.conf.sample included in Asterisk and see if it mentions it |
| 21:52.23 | mikeshank | Dovid: nothing |
| 21:52.31 | Dovid | try set verbose 9 |
| 21:52.34 | Dovid | and call in again |
| 21:52.46 | seanbright | Psykick: i don't think it will work in the [general] section |
| 21:53.09 | Psykick | seanbright: neither ... I've checked voip-info.org ... doesn't mention being able to do it |
| 21:53.25 | seanbright | Psykick: just glanced at where the file is read and don't see it there either |
| 21:53.25 | mikeshank | where you have _x, should i be replacing that with _1NXXNXXXXXX, |
| 21:53.26 | Psykick | seanbright: only thing I can do is bind connection to localhost |
| 21:53.44 | GhOnDiE | psykick give it a try in the general section |
| 21:53.48 | GhOnDiE | see what happens |
| 21:53.55 | seanbright | (even though it probably won't work) |
| 21:54.03 | GhOnDiE | it will either work or not |
| 21:54.07 | GhOnDiE | no harm in trying |
| 21:54.07 | Dovid | mikeshank: u dont need to . includes anything after that |
| 21:54.09 | seanbright | (probably won't) |
| 21:54.14 | Dovid | so _X. will match 1 or 1234 |
| 21:54.15 | GhOnDiE | :P |
| 21:54.25 | Dovid | _XXXX will only match a 4 digit number |
| 21:54.41 | Dovid | _XXXX. will require a min. of 4 numbers (or 5 not sure) |
| 21:54.46 | *** join/#asterisk kensuke_ (i=be0210a1@gateway/web/ajax/mibbit.com/x-f9b79653841a937d) |
| 21:55.01 | seanbright | Psykick: yeah, just looked at the code more closer. in 1.4.x it won't work that way. |
| 21:55.08 | seanbright | more closer? jeez |
| 21:55.09 | GhOnDiE | ok |
| 21:55.10 | seanbright | more closely. |
| 21:55.15 | Psykick | I think best option might be to bind to localhost .. and SNAT connections for select IPs |
| 21:55.23 | kensuke_ | Hi, i have a asterisk 1.2 on centos 32, and now probe migrate to 1.4 on centos x86_64 |
| 21:56.25 | kensuke_ | on the logs i see this: NOTICE[12866] chan_iax2.c: Avoiding IAX destroy deadlock |
| 21:56.45 | kensuke_ | someone can say what is the cause for this notice? |
| 21:56.51 | kensuke_ | ( sorry for my english ) |
| 21:57.00 | Psykick | kensuke_: if you are going to upgrade ... upgrade to 1.4.5 |
| 21:57.11 | seanbright | kensuke_: yes, the notice happens because asterisk is Avoiding IAX destroy deadlock |
| 21:57.19 | seanbright | bows |
| 21:57.30 | seanbright | 1.4.5? |
| 21:57.36 | kensuke_ | Psykick: i install 1.4.20.1 |
| 21:57.44 | mikeshank | Dovid: weird, just drops the call, nothing in the cli |
| 21:57.46 | kensuke_ | why 1.4.5? |
| 21:58.05 | seanbright | you mean 1.4.21.1 i assume |
| 21:58.13 | Psykick | seanbright: yeah ... I've had nothing but problems with IAX with 1.4.19, 1.4.20, 1.4.20.1 and 1.4.21.1 |
| 21:58.14 | seanbright | unless you were referring to libpri |
| 21:58.36 | Psykick | asterisk deadlocks and crashes |
| 21:58.47 | Psykick | or won't accept registrations |
| 21:58.53 | kensuke_ | Psykick: and yo know the cause? |
| 21:58.57 | seanbright | Psykick: any response to the issues you've posted on the bug tracking? |
| 21:59.05 | seanbright | s/tracking/tracker/ |
| 21:59.20 | Psykick | seanbright: other than they have been fixed and committed to 1.6 branch |
| 21:59.45 | seanbright | Psykick: if they are 1.4 issues, the fixes should be being committed to the 1.4 branch |
| 22:00.06 | Psykick | seanbright: they aren't in 1.4 as I've been checking the changelogs and nothing yet |
| 22:00.45 | Psykick | seanbright: things are pretty stable with 1.4.5 ... at least ... the issues I was having with the more recent versions aren't happening |
| 22:00.56 | seanbright | Psykick: what is your mantis username? |
| 22:01.00 | Psykick | seanbright: as soon as I see them in the changelogs then I'll upgrade |
| 22:01.06 | kensuke_ | Psykick: the calls are interrupted on iax extensions... |
| 22:01.09 | Psykick | seanbright: Psykick I believe |
| 22:01.10 | mikeshank | Dovid: how do i set verbose 9 |
| 22:01.20 | seanbright | Psykick: no such user |
| 22:01.24 | Psykick | mikeshank: core set verbose 9 |
| 22:01.53 | Psykick | seanbright: lemme just double check |
| 22:01.57 | Dovid | mike: set verbose 9 |
| 22:01.57 | seanbright | Psykick: cool |
| 22:02.00 | Dovid | from the CLI |
| 22:02.05 | mikeshank | do i issue that when im in the cli? noob |
| 22:02.16 | seanbright | mikeshank: core set verbose 9 |
| 22:02.25 | *** part/#asterisk mags2 (n=mags2@ampulex.whoi.edu) |
| 22:02.33 | Dovid | mikes: yes |
| 22:02.38 | kensuke_ | Psykick: thanks, i probe the 1.4.5 |
| 22:02.42 | kensuke_ | bye |
| 22:03.09 | Psykick | seanbright: mantis username rangib |
| 22:04.02 | *** join/#asterisk javb (n=javb@190.80.224.32) |
| 22:04.05 | Psykick | there are 3 or 4 bugs that I've been following ... not recently though ... busy with other things at the moment |
| 22:04.11 | seanbright | ohhh |
| 22:04.14 | seanbright | you haven't posted any |
| 22:04.16 | seanbright | gotcha |
| 22:04.22 | Psykick | not recently no |
| 22:04.23 | javb | does someone here knows a link to a howto on how to install Asterisk-Stat including the LAMP server? |
| 22:04.25 | seanbright | well if you crash again, please post so they can be looked at |
| 22:04.49 | Psykick | seanbright: bug #12795 |
| 22:04.58 | *** join/#asterisk |dennis| (n=Dennis@200.32.217.34) |
| 22:05.03 | Psykick | that was one of them |
| 22:05.26 | seanbright | U branches/1.4/channels/chan_iax2.c |
| 22:05.30 | seanbright | it was fixed in 1.4 |
| 22:05.59 | Psykick | seanbright: other bug was 12717 |
| 22:06.18 | seanbright | also committed to 1.4 |
| 22:06.28 | seanbright | but you're still crashing? |
| 22:06.33 | seanbright | even with 1.4.21.1? |
| 22:06.35 | Psykick | seanbright: correct ... |
| 22:06.38 | *** part/#asterisk LevonK (n=levon@nat/yahoo/x-37e9d0f76837f506) |
| 22:06.38 | seanbright | hrmm |
| 22:07.03 | Psykick | seanbright: thing is ... its a customers system ... and they were getting rather peeved that at it |
| 22:07.07 | seanbright | well if you could get another backtrace with DONT_OPTIMIZE and all that and post it, that would be great |
| 22:07.12 | seanbright | right |
| 22:07.19 | Psykick | seanbright: so have reverted them to 1.4.5 |
| 22:07.24 | seanbright | gotcha |
| 22:07.27 | seanbright | well that's a shame |
| 22:07.52 | Psykick | seanbright: they are keen to run another server with latest code on it to help out the community |
| 22:08.32 | Psykick | seanbright: they run a conferencing server so they need it stable |
| 22:08.38 | seanbright | yeah |
| 22:08.42 | seanbright | ah well |
| 22:08.57 | mikeshank | Dovid: http://www.pastebin.ca/1080034 |
| 22:09.40 | Psykick | seanbright: as they've said though ... they are willing to help out the community by running another server ... but they need to make arrangements with colo but more so ... need money coming in so they can get another server |
| 22:09.53 | seanbright | Psykick: great. |
| 22:10.00 | seanbright | ok, i'm off |
| 22:10.06 | Psykick | seanbright: later |
| 22:10.42 | Dovid | mike: OJ. so your carrier is sending the call to your phone number@your box instead of to the s extension |
| 22:10.52 | Dovid | so in place of exten => S,....... |
| 22:10.54 | Dovid | do |
| 22:11.05 | Dovid | exten => your_number,1,................. |
| 22:12.56 | *** part/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 22:14.22 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 22:14.37 | mikeshank | Dovid: in the incoming context? |
| 22:14.37 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 22:14.47 | Dovid | in default |
| 22:14.53 | Dovid | cause thats what u have set now in sip.conf |
| 22:15.05 | mikeshank | in place of _1NXXNXXXXXX |
| 22:15.06 | Dovid | and remove the exten => X.,... that u put it for me |
| 22:15.16 | Dovid | mike: I will PB |
| 22:15.28 | *** join/#asterisk mltlnx (n=mltlnx@75.138.164.68) |
| 22:17.22 | Dovid | mike: my bad. under incoming |
| 22:18.03 | mikeshank | Dovid: np, and remove the _X. lines |
| 22:18.43 | Dovid | mike: |
| 22:18.45 | Dovid | use this |
| 22:18.46 | Dovid | http://pastebin.com/m10a8edcb |
| 22:18.53 | Dovid | keep the s so that if u have another carrier it will go there |
| 22:19.08 | Dovid | what happens is asterisk tells ur carrier to send the calls to the s@your_IP |
| 22:19.27 | Dovid | however your carrier seems to be ignoring that and sending the call to your_number@your_IP |
| 22:19.36 | Dovid | so you need to create an "extension" to get the call |
| 22:20.56 | mikeshank | Dovid: replace with my number but still getting the same error, so how do i create an extension for my number |
| 22:21.58 | mikeshank | Dovid: scratch that, I assume exten => your_number creates the extension, but it dosent seem to work |
| 22:23.49 | Dovid | well replace your_number with your number |
| 22:24.06 | Dovid | mike: did u do a reaload ? |
| 22:24.32 | mikeshank | Dovid: yes, did both |
| 22:24.40 | Dovid | what do u see in ur CLI now ? |
| 22:25.17 | Dovid | mike: follow what i put on the PB |
| 22:25.38 | mikeshank | handle_request_invite: call from 'my number' to extension 'my number' rejected because extension not found |
| 22:26.59 | Dovid | mike: pb ur extensions.conf and sip.conf again |
| 22:29.51 | *** join/#asterisk hi365_m (n=hi365@213.151.63.7) |
| 22:30.02 | mikeshank | Dovid: http://pastebin.com/m67594009 |
| 22:31.09 | Dovid | mike: in the error in the cli did ur number have a 1 infront of it ? |
| 22:31.28 | mikeshank | Dovid: no |
| 22:31.42 | Dovid | mike: nm |
| 22:31.48 | Dovid | i am spacing ouit |
| 22:45.48 | *** join/#asterisk Katty (n=angela@adsl-209-30-144-78.dsl.stlsmo.swbell.net) |
| 22:45.52 | Katty | hewwoes. |
| 22:46.17 | Dovid | hi katty |
| 22:46.20 | *** join/#asterisk LevonK (n=levon@nat/yahoo/x-37e9d0f76837f506) |
| 22:47.16 | Katty | how's things |
| 22:47.33 | Dovid | have not been here in a while |
| 22:47.40 | Dovid | katty: have a look at asterisk-cli |
| 22:47.45 | Dovid | not too active as of now |
| 22:47.53 | Katty | baroo? |
| 22:48.28 | Katty | pokes file |
| 22:48.49 | Katty | hmmhesays: oh. |
| 22:48.52 | Katty | hmmhesays: are you still here? |
| 22:53.04 | Katty | wow. so quiet :< |
| 22:53.36 | Dovid | lol |
| 22:53.37 | *** part/#asterisk LevonK (n=levon@nat/yahoo/x-37e9d0f76837f506) |
| 22:54.26 | Katty | not even [TK]D-Fender is talking. |
| 22:54.29 | Katty | something's horribly wrong. |
| 22:54.38 | Dovid | lol |
| 22:54.45 | Dovid | he was b4 |
| 22:54.52 | Katty | he doesn't shut up (= |
| 22:54.56 | Katty | ever. |
| 22:55.53 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 22:56.19 | Dovid | he only talks when spoken to |
| 23:02.14 | *** join/#asterisk hi365_m (n=hi365@213.151.63.7) |
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| 23:02.51 | *** join/#asterisk raz (n=y@unaffiliated/raz) |
| 23:03.04 | raz | is there a way to get rid of the implicit "beep" that the Record() command issues? |
| 23:03.09 | raz | i'd like to send my own custom beep anyhow ;) |
| 23:03.40 | Dovid | raz: not that i know of. try changing the cource code ;) |
| 23:03.46 | raz | dang |
| 23:03.51 | raz | i thought it might be one of the default soundfiles |
| 23:03.54 | mikeshank | Dovid: think i got it, my sip context was set to default, changed it to context=phone and got hello world |
| 23:03.57 | Dovid | prob. is |
| 23:04.05 | Dovid | mike: :) |
| 23:04.38 | Dovid | mike: last thing i told ya was to check sip.conf but its good when u learn the hardway. u never forget |
| 23:04.40 | mikeshank | Dovid: yeah, only problem shouldnt my phone be ringing :) |
| 23:04.44 | Dovid | i have wasted many of hours |
| 23:04.56 | Dovid | mike: No. because you dont have anything in there to call your phone |
| 23:05.03 | Dovid | look at what you have |
| 23:05.05 | Dovid | answer |
| 23:05.07 | Dovid | play file |
| 23:05.11 | Dovid | hang up |
| 23:05.23 | Dovid | you would need to change ur configs |
| 23:05.27 | Dovid | on line 2 change it to |
| 23:05.59 | Dovid | mike: Change exten => my_number,2,Playback(hello-world) |
| 23:06.25 | Dovid | to: exten => my_number,2,Dial(SIP/Sam) |
| 23:06.38 | Dovid | so now instead of playing a file it will call your phone |
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| 23:08.18 | mikeshank | Dovid: time for a beer |
| 23:08.23 | Dovid | I agree |
| 23:08.26 | Dovid | u did well for urself |
| 23:08.45 | Dovid | what makes me most proud is that there still is hope. some people still use the CLI and not gui's |
| 23:08.49 | Dovid | asterisk-cli is new |
| 23:08.53 | Dovid | #asterisk-cli |
| 23:09.01 | mikeshank | Dovid: thanks again for all your help, didnt think it was gonna happen |
| 23:09.06 | Dovid | join it. its for peole that actually write their own configs ;) |
| 23:09.13 | Dovid | i have had many of such nights |
| 23:09.50 | *** join/#asterisk rwaite (n=richardw@d118-75-139-184.try.wideopenwest.com) |
| 23:09.52 | mikeshank | Dovid: so, whats the deal with broadvoice, they really suck |
| 23:10.06 | Dovid | mike: when i tested them 4 years ago they did |
| 23:10.06 | mikeshank | just signed up yesterday |
| 23:10.10 | Dovid | i never looked back |
| 23:10.18 | Dovid | for messing around its ok |
| 23:10.30 | Dovid | i had lots of call quality issues with them |
| 23:10.31 | rwaite | hi all, i've been having a problem where voicemail messages are being "lost" in the Old folder... is there a way to disable sending messages to the Old folder? |
| 23:10.41 | Dovid | calls just not going through |
| 23:10.48 | mikeshank | i want to use it for my small business |
| 23:11.05 | Dovid | mikeshank: what kind of traffic do u think u will have ? |
| 23:11.14 | mikeshank | not much ;) |
| 23:11.23 | Dovid | try myphonecompany.com |
| 23:11.26 | Dovid | $5.00 |
| 23:11.32 | Dovid | for a number |
| 23:11.43 | Dovid | they have some issues but for that money its worth it |
| 23:11.44 | mikeshank | just tired of paying 150 mo. for a business line |
| 23:11.48 | Dovid | they dont advertise it |
| 23:11.57 | Dovid | use that only for incoming |
| 23:12.03 | Dovid | for outgoing use voipjet.com |
| 23:12.11 | Dovid | they are the cheapest for outbound |
| 23:12.23 | rwaite | i guess not |
| 23:13.03 | Dovid | rwaite: saw some1 else that had the issue. dont remember what the issue was |
| 23:13.15 | mikeshank | Dovid: thanks, i look into both of them |
| 23:13.32 | Dovid | np |
| 23:13.40 | Dovid | thats ur best bang for ur buck |
| 23:13.57 | Dovid | i work for an ITSP but we are more expensive ;) |
| 23:13.59 | rwaite | Dovid: i tried simply symlinking Old to INBOX, which "works" but then * will say 1 new, 1 old |
| 23:14.27 | Dovid | why would u want to link it ? |
| 23:14.38 | Dovid | thats prob. whats cause ur issue |
| 23:14.43 | rwaite | um, no. |
| 23:14.48 | Dovid | cause it deletes it from the INBOX that may be whats doing it |
| 23:14.54 | Dovid | if my sleepy head is correct |
| 23:15.03 | rwaite | i symlinked them so that when asterisk moves the voicemail to old, it gets "moved" back to inbox |
| 23:15.22 | Dovid | try taking that off |
| 23:15.26 | Dovid | and test it |
| 23:15.31 | rwaite | the problem is voicemails i have not heard yet are being moved into the old folder |
| 23:15.36 | Dovid | if it works then, maybe write a script that will move it |
| 23:15.47 | Dovid | thats wierd and should not happen |
| 23:15.54 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 Zaptel troubles:type ~centos52bug -=- Asterisk 1.6.0-beta9 (2008/05/14), 1.4.21.2 (2008/07/22), *-Addons 1.6.0-beta4 (2008/06/04), 1.4.7 (2008/06/04), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.6 (2008/07/22) -=- #asterisknow or #asterisk-gui for AsteriskNOW -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX |
| 23:16.01 | rwaite | maybe a cron job that moves the messages every 5 minutes |
| 23:16.14 | rwaite | woot new release |
| 23:16.17 | Dovid | yea. but b4 u do that test it and see what happens |
| 23:16.34 | rwaite | Dovid: what i just described to you, the symlink, was in effort to fix the issue |
| 23:16.49 | rwaite | the problem existed before i did any mucking around |
| 23:16.53 | Dovid | rwaite: try getting the latest version. it should not happen |
| 23:17.58 | Qwell | woot netsplit |
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| 23:22.04 | profxavier | good quality VOIP phone (name + model), for business use ? |
| 23:22.27 | Qwell | ~phones |
| 23:22.28 | jbot | well, phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 23:22.48 | Dovid | I like Polycom |
| 23:24.09 | profxavier | name + model ? |
| 23:24.18 | Qwell | depends on what you need |
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| 23:25.46 | profxavier | thanks you have been lots of help |
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| 23:31.49 | angryuser | snom at end ? i dont like that, they are better then spa ;) |
| 23:33.04 | TJNII | mutters somethuing about his snom resetting whenever someone calls |
| 23:34.22 | angryuser | Qwell i had a good exp with siemens sip phones the have pretty unique uptions like 2 dect>1base>2sipaccount>whatever , ithink we should add them for home and small office use |
| 23:36.47 | TJNII | Of course, the final concensus was that this phone is defective, hence why it was on eBay. |
| 23:36.54 | TJNII | But when it does work it is nice. |
| 23:37.35 | mmlj4 | i loathe polycom |
| 23:37.59 | Shotygun | I agree, Snom are great phones. |
| 23:38.28 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 23:40.48 | drfreeze | Hey - we are setting up a new PRI system and trying to dial out |
| 23:41.02 | drfreeze | My PRI channels are listed here: http://pastie.textmate.org/private/kc8bdxmrr9mzauaopaqhcw |
| 23:41.35 | drfreeze | What should my dial string look like? |
| 23:41.39 | drfreeze | I have tried: ZAP/1-1/15129499683|| |
| 23:41.46 | drfreeze | and ZAP/g1/15129499683|| |
| 23:41.50 | drfreeze | and neither work |
| 23:42.27 | drfreeze | the latter gives error: Unable to create channel of type 'ZAP' (cause 0 - Unknown) |
| 23:43.06 | drfreeze | the former Unable to create channel of type 'ZAP'cause 34 - Circuit/channel congestion |
| 23:45.20 | GhOnDiE | you trying to dial out on just chan 1? |
| 23:46.07 | wwalker | GhOnDiE: or a magic rotation. what do you suggest for drfreeze ? |
| 23:46.53 | GhOnDiE | gues it depends on how he has it setup |
| 23:47.00 | wwalker | drfreeze lost his connection just now but I'm in the room with him |
| 23:47.17 | GhOnDiE | may only have that specific chan available for outgoing circuit |
| 23:47.18 | GhOnDiE | ? |
| 23:47.21 | wwalker | he has a PRI with 11 voice channels into a TE-122b |
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| 23:47.58 | GhOnDiE | ok well would be better to rotate it for load ballance function |
| 23:48.38 | wwalker | cool. is that a zapata.conf setting, or is there a dial target that will just rotate the calls? |
| 23:48.54 | wwalker | right now we just want to verify that it's possible to make an outbound call at all |
| 23:49.05 | wwalker | they are installing a new PRI... |
| 23:49.19 | GhOnDiE | this page should help you http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels |
| 23:49.38 | GhOnDiE | would be a r option in zapta.conf |
| 23:50.27 | GhOnDiE | make a group of all channels and then do a dial(Zap/r1.... for instance |
| 23:50.33 | GhOnDiE | depends on how you setup the group |
| 23:51.26 | GhOnDiE | anyway good luck with it, im off to bed. |
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