| 00:00.27 | [TK]D-Fender | Again, stop worrying about configs and go look at what is actually happening. |
| 00:00.43 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 00:10.46 | leifmadsen | you know what blocks stuff? firewalls. |
| 00:11.42 | [TK]D-Fender | So far as we've seen there is no "blockage". |
| 00:12.19 | [TK]D-Fender | That implies something even tried going somewhere it was intended to but was obstructed. |
| 00:12.28 | [TK]D-Fender | We never got to see that |
| 00:14.33 | flan | Are we playing the how-to-learn-network-diagnosis-techniques game? |
| 00:18.52 | WIMPy | Does anyone have an idea, how a new call can end up at extension h immediately? |
| 00:19.06 | WIMPy | And no, "h" wasn;t dialled. |
| 00:21.24 | flan | Hangup/fallthrough-without-answer before it progressed past, or even to, the alerting stage, maybe. |
| 00:21.28 | flan | Just a guess. |
| 00:25.18 | *** join/#asterisk mattwj2002 (~matt@wikisource/pdpc.active.mattwj2002) |
| 00:26.18 | mattwj2002 | hi guys as far as google voice I heard that you can't have the phone registered with anything else |
| 00:26.27 | mattwj2002 | *phone number |
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| 00:27.27 | mattwj2002 | anyone here? |
| 00:27.28 | WIMPy | Hmm, yes, it says Spawn extension ... exited non-zero. But that's not what happens if it is a invalid extension. |
| 00:27.35 | mattwj2002 | hi WIMPy |
| 00:29.03 | p3nguin | Verbose output during the failing call does not show anything else? |
| 00:29.26 | WIMPy | No |
| 00:29.30 | *** join/#asterisk RickCogley (~anonymous@EM114-48-116-117.pool.e-mobile.ne.jp) |
| 00:31.00 | WIMPy | [2012-03-02 01:26:10] -- Accepting AUTHENTICATED call from 10.9.42.2: |
| 00:31.05 | WIMPy | [2012-03-02 01:26:10] == Spawn extension (dial-user, 11940128, -1) exited non-zero on 'IAX2/yeti-lmaa-11289' |
| 00:31.10 | WIMPy | [2012-03-02 01:26:10] -- Executing [h@dial-user:1] Verbose("IAX2/yeti-lmaa-11289", "1,==== Hangup dial-user:0 ") in new stack |
| 00:31.56 | WIMPy | wonders if the -1 is a clue. |
| 00:39.16 | mattwj2002 | hi WIMPy |
| 00:39.26 | mattwj2002 | hi p3nguin |
| 00:39.34 | p3nguin | Hi. |
| 00:39.50 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 00:39.54 | mattwj2002 | I have a question about google voice and asterisk if you would be so kind |
| 00:41.28 | mattwj2002 | what are the requirements for getting asterisk to work with an google voice account? I heard if you have an android phone associated with the account it won't work....do you know the details on that? |
| 00:42.18 | WIMPy | has absolutely no clue about GV. |
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| 00:55.22 | p3nguin | If he would have stuck around, I would have told him what he wanted to know. |
| 01:01.14 | flan | I'm curious on his behalf. |
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| 01:17.29 | p3nguin | flan: There's a wiki page for setting it up. And to make it work in conjunction with his android device, he would just need to set the priority in asterisk to be higher than the one on the android (which I believe is hard-set to priority 24) for asterisk to be able to take the inbound phone calls. |
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| 01:38.03 | dano0 | anyone around? |
| 01:39.06 | phix | [TK]D-Fender: looks like router firewall actually i can port scan port 5060 and it is closed (expected behaviour) but the rtp ports are filtered not closed |
| 01:40.14 | phix | indicating the router is blocking them, which is why the caller can hear me but i cant hear them |
| 01:41.19 | phix | what you think [TK]D-Fender ? or could something else be causing it |
| 01:56.08 | phix | yay I have closed status on the RTP ports now but it still isn't working :\ |
| 01:57.20 | phix | [TK]D-Fender: any ideas? and yes there is a "blockage", voice being sent to the Asterisk server is being blocked / dropped, voice being sent from the asterisk server is working fine |
| 01:57.58 | *** join/#asterisk fornax (~fornax@85.183.53.64) |
| 01:58.29 | phix | [TK]D-Fender: the problem is the logs are not saying why it is being blocked, probably because the voice isn't being blocked before it gets to the server |
| 01:59.02 | fornax | Can someone give me a hint were I can download the german sounds for asterisk? It seem to be deleted from all mirrors and even the gentoo ebuild does not work anymore. |
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| 05:08.48 | TeknoJuce | anyone around that has exp with 1.8 and unistim.conf nortel i2004 phones |
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| 06:12.13 | jgowdy | What branch is 10.2.0-rc4 in SVN? |
| 06:12.48 | jgowdy | I'm trying to figure out what branch to use in my git mirror of the SVN repository to track 10.x |
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| 06:37.35 | asphalt | When caller hangups, the AGI returns 0 instead of -1? Why is that? |
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| 06:55.03 | asphalt | Can any one tell why the AGI cmd returns 0 whereas I can see in console that the AGI returned -1, but in my application I receive it as 0 (it happens on hangup) |
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| 07:04.12 | eye-scuzzy | moin |
| 07:04.46 | eye-scuzzy | looks for acmepacket jedi |
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| 07:55.57 | tyman | What's the best way to make a second line (polycom phone) use a different outbound caller-id than line 1? |
| 07:56.55 | tyman | Does it have to be a second extension? |
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| 08:00.03 | p3nguin | Extension? No, extensions have nothing to do with your line keys. |
| 08:00.14 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 08:00.40 | p3nguin | You can assign a new sip device name to the second line key and configure the callerid value for the new device as you want. |
| 08:01.25 | *** join/#asterisk ninor (~ninor@unaffiliated/ninor) |
| 08:01.27 | ninor | re |
| 08:01.37 | tyman | Ah…yes... |
| 08:01.44 | tyman | thx p3nguin |
| 08:01.48 | ninor | are there any open source libs or mac apps that facilitate p2p voip calls? |
| 08:01.51 | p3nguin | For multi-line devices, I just append -a, -b, -c, etc. to the names. A phone by the name of 0000AAAAFFFF will have keys 1 and 2 named 0000AAAAFFFF-a and 0000AAAAFFFF-b. |
| 08:02.00 | ninor | i'm not into server only stuff like AIM |
| 08:02.37 | tyman | p3nguin: y…looks good |
| 08:02.40 | tyman | thx |
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| 08:16.00 | jofry | hi folks |
| 08:22.37 | kaldemar | ninor: for LAN scale, there are apps like blink that use bonjour for locating other endpoints. |
| 08:22.59 | ninor | nah over the net |
| 08:23.03 | ninor | i wanna voip chat with friends |
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| 08:23.08 | schmidts | good morning |
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| 08:25.31 | kaldemar | ninor: i think all use a server for locating peers at the moment. |
| 08:41.12 | *** join/#asterisk ziz212 (~chatzilla@203.115.2.22) |
| 08:42.06 | ninor | fuck that |
| 08:42.14 | ninor | i'm tired of this internet lite BS |
| 08:42.17 | ninor | actually |
| 08:42.24 | ninor | THAT's the etymology of Internet !!!!! |
| 08:42.26 | ninor | OMFG |
| 08:42.48 | ninor | i've been around since '94, and witnessed the strange transition of internet from internet to Internet. figured it was just typical biz |
| 08:42.51 | ziz212 | Dear all, Is there any way to write mysql query in dial plan to insert data to a mysql database? I have searched this and I cant find that in the web. If it is there pls let me know the location of the information. |
| 08:43.12 | ninor | but no, it's their standard model of 'formalizing' an identity as a cover/shell of another organization with different intents altogether |
| 08:43.19 | ninor | the Internet is not free. the internet was/is |
| 08:45.11 | ninor | the Internet is about metered bandwidth at the user level, port filtering, IP filtering, p2p elimination |
| 08:45.23 | ninor | man this is fucked up we have to tear down those walls |
| 08:45.47 | ninor | are you guys seeing anything like Skype trying to come in and sue other apps that implement internet voice features? |
| 08:51.29 | Henchman21 | i think theres an exec command |
| 08:51.46 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 08:54.06 | Henchman21 | <PROTECTED> |
| 08:54.49 | Henchman21 | maybe that could help you interface to a mysql dbase |
| 08:55.25 | Henchman21 | http://www.voip-info.org/wiki/view/Asterisk+cmd+System |
| 08:57.16 | kaldemar | ziz212: func_odbc is what you want. |
| 08:57.39 | Henchman21 | even better |
| 08:58.35 | kaldemar | ziz212: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Database_id287624.html |
| 08:59.28 | kaldemar | ziz212: http://svn.digium.com/svn/asterisk/tags/10.1.2/configs/func_odbc.conf.sample |
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| 09:00.27 | kaldemar | ziz212: configuring ODBC: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DB.html |
| 09:02.52 | ziz212 | @kaldemar : Thanks for the information. |
| 09:08.00 | ninor | it's strange to see you lazy castrated heifers have nothing to say about what i wrote |
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| 09:08.22 | ninor | maybe this is why it's slowly happening. hm ya, that just won't do |
| 09:08.42 | Dovid | hi all |
| 09:09.03 | krotos | hi all :) |
| 09:09.44 | ninor | :) |
| 09:11.26 | krotos | i've got two asterisk boxes (1.8.x), the first box called 'A' Send call's trought sip-trunk to the second box 'B'. B is connected with sip-provider in SIP. The 'A' Box send call to B that route it to sip provider. On the 'A' box i set CALLERID(num-pres) and CALLERID(name-pres) |
| 09:12.04 | krotos | if i set sendrpid=yes and trustrpid=yes on sip trunk that connect A and B Box can i see the same Channel Variabile on the B Box? |
| 09:13.07 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 09:13.25 | v0lZy | hello |
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| 09:17.43 | Dovid | has anyone used the read function in an agi with Cepstral? |
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| 09:27.00 | darrenlooby | Need some help diagnosing an issue. I have asterisk with opensips - and have working inbound and outbound external. No sound on internal calls - but, MOH works. When returning to the call from MOH, full sound is provided. Any clues? |
| 09:28.36 | krotos | i think you have some problem with nathelper |
| 09:28.42 | krotos | and opensips :) |
| 09:28.46 | krotos | and = in |
| 09:29.00 | *** part/#asterisk hurdman (~ygcheny@cylon.r0b0t.fr) |
| 09:29.13 | darrenlooby | Not sure where to start... :( |
| 09:29.14 | krotos | so here is asterisk channel, not opensips :) |
| 09:32.52 | darrenlooby | Ahh, ok... sorry, missed your subtle hints |
| 09:32.54 | darrenlooby | :) |
| 09:42.30 | krotos | darrenlooby: the first point to start i debugging sip packet |
| 09:42.56 | darrenlooby | Just giving that ago now :) |
| 09:43.09 | krotos | and understand where the stream go to, and the used ip and port |
| 09:43.13 | krotos | (rport) |
| 09:44.12 | darrenlooby | cheers |
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| 10:55.16 | WIMPy | Do I have a network issue or has Jira just gone down? |
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| 12:06.53 | Dovid | is it possible to have a variable that is an array ? |
| 12:13.04 | Dovid | found it. function array |
| 12:14.20 | Tim_Toady | well, the name is tricky on this one, as far as i know you cant have arrays, with array func you just set multiple variables at once |
| 12:18.21 | Dovid | it seems that you can't read it. u can only write it |
| 12:18.35 | Dovid | hats the max legnth of an Astersik variable? |
| 12:18.56 | Dovid | whats* |
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| 12:31.38 | kaldemar | Dovid: see HASH and HASHKEYS |
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| 12:42.25 | Dovid | wondering if i should use asterisk variables and then have it all in memory or store results in sql. |
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| 12:44.18 | kaldemar | Dovid: beware that variables are volatile. |
| 12:45.01 | kaldemar | my asterisk (10.1.2) seems to choke when variable value length is 4091 characters. |
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| 12:59.35 | fbnts | Hi, I am having a weird problem with Asterisk & SIP. I have a local asterisk server with SIP phones connected locally. I then have a SIP trunk through an online provider. When make an inbound or outbound call it appears that the outbound audio is trying to route directly to the remote gateway and not through my local asterisk |
| 13:00.42 | kaldemar | fbnts: set directmedia=no for the provider. |
| 13:02.47 | fbnts | thanks kaldemar, will try that. I'm sure I did have that at some point in my general section |
| 13:03.21 | Thazza | Hi All, I am editing my dialplan, I would like to do a GotoIf statement as long as 2 variables contain the following string.. I am lost as to how I would do this. Any assistance? |
| 13:06.05 | kaldemar | Thazza: 2 variables contain as in both of them contain a string at some position of the string? |
| 13:07.24 | Thazza | GotoIf(${Var1}="Home"&${Var2}="Redirect"?DialHome,s,1) |
| 13:08.11 | Thazza | kaldemar: The Vars only contain either this string or a different string. So can be only like 3 options for each of the 2 vars. |
| 13:09.29 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:09.53 | Thazza | For my example above, Var1, can be either "Home" or "Work" and Var2 can be either "Redirect" or "NoRedirect", so i need 4 GotoIf statements to check all options. |
| 13:10.19 | kaldemar | GotoIf($["${Var1}" =~ "string" && "${Var2}" =~ "string"]?true:false) |
| 13:10.38 | Thazza | Or perhaps a Case statement? |
| 13:10.57 | Thazza | What does the =~ do? |
| 13:11.12 | kaldemar | it's a regex match. |
| 13:11.41 | kaldemar | you can of course use == if you know exactly what the variables should contain. |
| 13:11.59 | kaldemar | sorry, not == but a single =. |
| 13:12.56 | Thazza | OK.. So * will handle it as long as I use the && in the middle. Guess I could do more than just 2 vars that way.. Can you do GotoIf in Case format? |
| 13:13.06 | kaldemar | sure. |
| 13:14.39 | Thazza | How would you do that? sorry, my brain can't think of how. |
| 13:14.46 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 13:14.58 | jacc0 | hi all! almost weekend over here!! |
| 13:15.22 | fbnts | kaldemar: I've just looked and the problem I am having is more on the asterisk server, its sourcing its SIP traffic to the online provider from its 192.168.x.x address rather than its public IP. Is there a way to specify which IP to use per peer in sip.conf? |
| 13:15.23 | kaldemar | Thazza: GotoIf($["${Var1}" =~ "string" && "${Var2}" =~ "string" && "${Var3}" = "string"]?true:false) |
| 13:15.40 | *** join/#asterisk garymc (~chatzilla@81.138.225.164) |
| 13:15.46 | kaldemar | fbnts: then you need to configure nat settings. |
| 13:15.48 | kaldemar | ~sipnat |
| 13:15.48 | infobot | [~sipnat] Quick guide on configuring Asterisk + 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 . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 13:16.12 | Thazza | kaldemar: Oh ok.. cool.. Thank you for your help.. Time to majorly re-write this DP. |
| 13:16.56 | [TK]D-Fender | <Thazza> OK.. So * will handle it as long as I use the && in the middle. Guess I could do more than just 2 vars that way.. Can you do GotoIf in Case format? <- no |
| 13:16.57 | *** join/#asterisk darkbasic (~quassel@2a01:7e00:e001:315::2) |
| 13:17.37 | [TK]D-Fender | Thazza, AEL templates a construct like that that gets parsed back to the messy GotoIf equivalent |
| 13:18.35 | Thazza | has yet to dive into the world of AEL. |
| 13:21.29 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 13:22.23 | *** join/#asterisk Andrioid (~andri@cpe.ge-1-1-0-116.vjnxj1.customer.tele.dk) |
| 13:22.59 | Andrioid | Is there any way to create/edit SIP users through AMI or CLI without rewriting entire config files? (v10) |
| 13:23.43 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 13:23.52 | Andrioid | And on a seperate note, is it possible to change a caller-id for a specific user from AMI or CLI ? |
| 13:24.23 | kaldemar | ugly imitation of a case statement: http://pastebin.com/index/6QxJnVr0 |
| 13:24.37 | [TK]D-Fender | Thazza, I wouldn't... |
| 13:24.48 | [TK]D-Fender | Andrioid, No |
| 13:25.25 | [TK]D-Fender | Andrioid, No you can't change peers without going through configs and reloading except for real-time DB. |
| 13:25.31 | Andrioid | [TK]D-Fender: so, I have to read the config file through AMI, change it, upload and reload? |
| 13:26.01 | Andrioid | what about Caller-ID changes on users - please tell me I can do that live :P |
| 13:26.04 | [TK]D-Fender | Andrioid, Not even sure of a way to read via AMI... |
| 13:26.21 | [TK]D-Fender | <Andrioid> what about Caller-ID changes on users - please tell me I can do that live :P <--- no that is the perr just the same |
| 13:26.30 | [TK]D-Fender | peer* |
| 13:26.31 | Andrioid | rats |
| 13:26.42 | kaldemar | Andrioid: mandger show command Setvar |
| 13:27.08 | [TK]D-Fender | kaldemar, that is for a call, not the overall peer |
| 13:27.19 | leifmadsen | Andrioid: it sounds like you're looking for Realtime |
| 13:27.22 | [TK]D-Fender | Not what he's asking for |
| 13:27.34 | [TK]D-Fender | (kaldemar) |
| 13:27.41 | kaldemar | [TK]D-Fender: it can be used to set a global variable that can be used to store caller id's for peers. |
| 13:27.55 | kaldemar | boom! changable caller id's. |
| 13:28.07 | leifmadsen | Andrioid: if you just want to change CallerID for a call, just set it in the dialplan when placing a call. You can do a lookup from the database for the peer using func_odbc if you want it configurable. |
| 13:28.14 | [TK]D-Fender | kaldemar, Don't think CDR will look at it like that, and you've have to read it in and apply.... |
| 13:28.18 | leifmadsen | Andrioid: the approach you're thinking does not exist |
| 13:28.38 | Thazza | just relised his DP, and gotoif just got really big. :-(.. it is probally best to do macro's? I will create a pastebin of the conditions. |
| 13:28.41 | Andrioid | leifmadsen: ok, thanks - we just got AMI access to our box - so my options are a bit limited |
| 13:29.02 | [TK]D-Fender | kaldemar, Fugly for sure and only marginally usable while making a kludge out of your dialplan |
| 13:29.41 | leifmadsen | Andrioid: there is no way to modify the existing configuration via AMI like you're approaching |
| 13:29.47 | [TK]D-Fender | Andrioid, then use AMI to install something else for you to communicate with |
| 13:30.17 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 13:30.40 | Andrioid | leifmadsen: we have a tool that can do this, but it's completely closed (and doesn't run on my machine, otherw. i would sniff it) |
| 13:30.56 | Andrioid | D-Fender: like? |
| 13:31.32 | leifmadsen | Andrioid: ok, so you've built a tool that does that, but I obviously know nothing about that, and we're talking vanilla asterisk here. |
| 13:31.51 | Andrioid | leifmadsen: I didn't build it, was shipped with the box - just started here a week ago :P |
| 13:31.58 | leifmadsen | if it does work like you're suggesting, it likely takes data from the database and writes flat files somehow |
| 13:33.06 | Andrioid | this has been helpful, at least I can stop searching for editPeer or somth like that :) |
| 13:34.34 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:35.43 | kaldemar | [TK]D-Fender: had to try it for myself. one setvar in sip.conf that tells the dialplan which global variable is used and one line in dialplan to set the caller id. |
| 13:37.04 | Andrioid | does that work? |
| 13:38.38 | kaldemar | yes it works. |
| 13:39.45 | Andrioid | ok, then I just have to hope they did it like that :) |
| 13:40.35 | kaldemar | you can confirm that by listening to the manager interface. |
| 13:42.05 | Andrioid | with 'manager set debug' right? |
| 13:42.13 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 13:43.53 | kaldemar | that shows something. what i had in mind was to dump the TCP port on the asterisk host. |
| 13:44.16 | Andrioid | don't have access to the box, only AMI - outsourcing :( |
| 13:44.18 | kaldemar | manager debug should give you a notification when an action is performed. |
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| 13:49.42 | *** join/#asterisk darkbasic_ (~quassel@2a01:7e00:e001:315::2) |
| 13:51.44 | Thazza | kaldemar: Are you still here? |
| 13:52.22 | Thazza | kaldemar: I have created a PasteBin of what I am trying to do. Is this the best way to do it? http://pastebin.com/bHyqmVgd |
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| 13:53.08 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:59.06 | kaldemar | Thazza: i don't know, that's a kind of a mess. |
| 13:59.58 | Thazza | kaldemar: Yeah i know.. Do you understand what I am trying to do? |
| 14:01.17 | Thazza | kaldemar: I labelled each ExecIf statement, and then connected True/False Bounce. |
| 14:01.51 | kaldemar | write it as a real dialplan. |
| 14:02.44 | [TK]D-Fender | <kaldemar> [TK]D-Fender: had to try it for myself. one setvar in sip.conf that tells the dialplan which global variable is used and one line in dialplan to set the caller id. <- not quite... global is all calls... |
| 14:02.59 | [TK]D-Fender | kaldemar, that means another device will have the same global |
| 14:03.13 | [TK]D-Fender | kaldemar, And globals don't survive * restarting, etc |
| 14:03.24 | [TK]D-Fender | kaldemar, Trust me, the dominos are falling fast on this one :) |
| 14:04.23 | Thazza | [TK]D-Fender: Do you understand what I am trying to do. Is multiply ExecIf statements the best for me? |
| 14:04.24 | kaldemar | [TK]D-Fender: no, all devices have their own variables. the variables have default values in the file under [globals]. |
| 14:05.19 | kaldemar | Thazza: write it as a real dialplan. your paste is so infested with syntax errors it hurts eyes to read. :) |
| 14:05.28 | kaldemar | Thazza: and OR is || in an expression. |
| 14:06.05 | *** join/#asterisk serafie (~erin@nat/digium/x-ulrvtwcdjbknvnjw) |
| 14:06.41 | [TK]D-Fender | kaldemar, Yes, but those don't survive reboot. |
| 14:06.46 | kaldemar | [TK]D-Fender: every peer has a setvar=myclid=thispeersclid, and dialplan has CALLERID(num)=${GLOBAL(${myclid})}. |
| 14:06.53 | Thazza | kaldemar: I will try. |
| 14:07.02 | [TK]D-Fender | kaldemar, And you have to pull it all throughout your dialplan. House of cards |
| 14:07.06 | *** join/#asterisk martijn_ (~martijn@dhcp-077-250-092-061.chello.nl) |
| 14:07.07 | kaldemar | [TK]D-Fender: yes, i know that. they won't even get written to the file with "dialplan save". |
| 14:07.57 | kaldemar | [TK]D-Fender: it's not a house of cards if there is a structure in the dialplan. |
| 14:08.22 | kaldemar | [TK]D-Fender: but like everything else, that can be used in a bad manner. |
| 14:11.01 | kaldemar | i'm not saying it's good practice either, but it works the way it does. |
| 14:12.10 | *** join/#asterisk Vince-0 (~AndChat@196.215.188.241) |
| 14:13.47 | *** join/#asterisk mpoole (~mpoole@minotaur.apache.org) |
| 14:14.15 | mpoole | hi guys, is there any truth to asterisk (1.8) 64bit being problematic? |
| 14:14.38 | *** join/#asterisk veris (~fabiano@189.11.198.98) |
| 14:14.58 | ayrjola | Hi, it seems that when asterisk gets rtp event (Cisco RFC 2833) 192 'Shift to voiceband data mode (192)' asterisk stops sending audio to deskphone. I have set faxdetect=no in sip.conf (using asterisk 1.8.7.0). Any idea to get asterisk to ignoge this rtp packet? I know problem is in Cisco, it shouldn't send that rtp event in middle of call, but I'm trying to find workaround to this problem from asterisk side. |
| 14:17.02 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.22.59) |
| 14:21.29 | *** join/#asterisk Alborracho (~chatzilla@201.218.12.210) |
| 14:21.34 | Alborracho | Hi everyone |
| 14:22.18 | Alborracho | I just installed asterisk 1.8 in a centos 6.2, but it seems that the file /etc/init.d/asterisk is missing, anyone knows where is it in the sources? |
| 14:23.45 | leifmadsen | Alborracho: make config |
| 14:23.49 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:24.05 | fbnts | Hi kaldemar, I tried the directmedia=no but I am still having the same problem with inbound audio. I have set the nat options in sip.conf and using TCP dump the data is being receive by asterisk but not forwarded to the IP Phone. I have tried setting up an echo test but when calling in its silent as though my audio is not being received by asterisk |
| 14:24.34 | Alborracho | leifmadsen: thanks a lot! |
| 14:24.54 | leifmadsen | Alborracho: it's described in the installation chapter of Asterisk: The Definitive Guide btw :) |
| 14:25.56 | anonymouz666 | stragen, I have seen cases when the "odbc show all" show the DSN is connected but cdr_adaptive_odbc stops inserting into DB (mysql) |
| 14:26.08 | anonymouz666 | just a reload on the module makes work again |
| 14:26.40 | Thazza | kaldemar: Here is my view of a Nested ExecIf statement http://pastebin.com/Ygv3Eai4 |
| 14:27.47 | leifmadsen | Thazza: this is already wrong on the first line: ExecIf($[NumberInSQL} |
| 14:27.57 | leifmadsen | Thazza: the use of && is invalid |
| 14:28.10 | leifmadsen | the use of || is also invalid |
| 14:28.24 | leifmadsen | that whole line is a mess |
| 14:28.26 | Thazza | leifmadsen: check out http://pastebin.com/bHyqmVgd |
| 14:28.27 | leifmadsen | simplifying it |
| 14:28.46 | leifmadsen | s/simplifying/simplify/ |
| 14:29.12 | leifmadsen | Thazza: that paste is still very difficult to read as youv'e got a lot of stuff that is unnecessary and wrong in there |
| 14:29.38 | leifmadsen | I can't tell if that is dialplan or AEL |
| 14:29.41 | Thazza | leifmadsen: Yeah i thought it would be.. I am looking for the best way to write that statement. |
| 14:29.51 | leifmadsen | use multiple lines |
| 14:29.53 | leifmadsen | make it simpler |
| 14:30.07 | leifmadsen | break it out into pieces instead of trying to do it all on one line |
| 14:30.13 | Thazza | leifmadsen: How do you do ExecIf on multiply lines? |
| 14:30.30 | leifmadsen | You don't -- use GotoIf() |
| 14:31.00 | leifmadsen | just to simpler blocks |
| 14:31.08 | leifmadsen | do less in more areas, not more in less areas |
| 14:31.23 | Thazza | leifmadsen: Didn't use GotoIf.. used ExecIf. |
| 14:31.31 | [TK]D-Fender | change that |
| 14:31.36 | leifmadsen | Thazza: I understand what you did, and I'm telling you what you should have done |
| 14:32.08 | leifmadsen | if you're running into issues with that complex of a statement, then you need to simplify it |
| 14:32.13 | Thazza | leifmadsen: I do not understand.. that is why i am asking for assistance.. No point telling someone who knows it is a mess to clean it.. |
| 14:32.32 | Thazza | leifmadsen: Check out this sort of flowchart http://pastebin.com/bHyqmVgd |
| 14:32.44 | leifmadsen | Thazza: I don't understand what you want from me; I just told you how to approach the problem. Don't use nested ExecIf() -- use multiple GotoIf() to execute simpler code blocks |
| 14:33.04 | leifmadsen | Thazza: I've looked at that link -- it's a mess, and isn't a flow chart |
| 14:33.10 | leifmadsen | it's pseudo-code |
| 14:33.26 | Thazza | leifmadsen: Sigh.. |
| 14:33.36 | leifmadsen | good luck in your endeavour |
| 14:33.43 | leifmadsen | I have nothing else I can offer you |
| 14:34.11 | [TK]D-Fender | <Thazza> leifmadsen: Didn't use GotoIf.. used ExecIf. <- stop using execif. What is hard to understand about this? |
| 14:34.39 | [TK]D-Fender | Thazza, As leifmadsen told you... use gotoif's to jump to execute blocks of code. |
| 14:34.40 | Thazza | leifmadsen: your advise is to move from one line to many lines, using gotoif. how do you use gotoif, when you have a double question. |
| 14:35.13 | [TK]D-Fender | Thazza, what is a "double question"? |
| 14:35.46 | Thazza | the steps are, I check 1 var, then i have to check other vars, however the other var questions need to be asked in the true and false of the first question. |
| 14:36.19 | leifmadsen | well your first execif can easily be broken out into 2 code paths |
| 14:36.21 | Thazza | Am I better just to ask the first question, and set a var with the answer, and then do the other questions. |
| 14:36.56 | [TK]D-Fender | Thazza, then maybe 1 execif for that part... or an IF() instead. |
| 14:37.10 | [TK]D-Fender | Thazza, Perhaps if you showed us actual code we could SHOW you |
| 14:37.23 | leifmadsen | [TK]D-Fender: he did show it |
| 14:37.27 | Thazza | [TK]D-Fender: That was the actual code. |
| 14:37.31 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 14:37.42 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
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| 14:37.47 | leifmadsen | Thazza: first, I don't understand if your problem is a logic problem, or a syntax problem |
| 14:37.48 | Thazza | I am trying to work out the best way to code it. So it was my view of the code. |
| 14:37.53 | leifmadsen | because as I pointed out, your syntax is wrong |
| 14:37.54 | [TK]D-Fender | Where? |
| 14:38.08 | Thazza | leifmadsen: probally logic. |
| 14:38.19 | leifmadsen | Thazza: I also told you a better way than multiple nested execif statements, and you seemed to ignore me |
| 14:38.33 | leifmadsen | the first execif can be broken into 2 separate paths already |
| 14:38.35 | Thazza | leifmadsen: I am happy to look at the other way.. |
| 14:38.37 | [TK]D-Fender | OMG not this : <Thazza> leifmadsen: Didn't use GotoIf.. used ExecIf. |
| 14:38.44 | [TK]D-Fender | http://pastebin.com/Ygv3Eai4 |
| 14:38.46 | [TK]D-Fender | rather |
| 14:39.08 | [TK]D-Fender | Please dera God tell me this isn't it... |
| 14:39.11 | [TK]D-Fender | dear* |
| 14:39.31 | leifmadsen | GotoIf($[${MAILBOX_EXISTS(${MACRO_EXTEN}@default)}" = "1"]?FirstPathTrue,1:FirstPathFalse,1) |
| 14:39.43 | leifmadsen | already half way there |
| 14:39.47 | [TK]D-Fender | that line is a cluster-fuck. |
| 14:39.51 | [TK]D-Fender | (his) |
| 14:40.06 | leifmadsen | Thazza: break it into smaller steps, that's my suggestion |
| 14:41.02 | leifmadsen | if you start breaking it out separately with GotoIf(), you're going to find it much easier to follow and read |
| 14:41.03 | Thazza | leifmadsen: Ok.. going to try again.. will have many lines split in 5 mins. |
| 14:41.18 | leifmadsen | Thazza: that's good -- I don't need a time estimate, I'm not your manager |
| 14:42.36 | jacc0 | lol |
| 14:43.18 | [TK]D-Fender | I may be available ... let me know what you're offering for me to be your boss. Part-time of course as I'm the boss and get to set the rules ;) |
| 14:44.04 | Thazza | Wish I had a manager for this, wish it was more than just for personal. |
| 14:44.48 | *** join/#asterisk phlax (~ryan@host-92-27-135-217.static.as13285.net) |
| 14:48.12 | phlax | hi i have 2 fxo modules on a TDM410P card. Until recently they worked fine. Now one of the lines generates "red alarms", and asterisk can no longer make/receive calls. I have swapped the lines/cables around etc and it seems it is a problem with the phone line. Any suggestions on tracking down the exact cause of the "red alarms"? |
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| 15:12.58 | Thazza | [TK]D-Fender: http://pastebin.com/XgHmdpte - Here is my DP |
| 15:13.17 | Thazza | leifmadsen: http://pastebin.com/XgHmdpte - Here is my DP |
| 15:13.45 | leifmadsen | looks better |
| 15:13.51 | leifmadsen | still syntax errors though |
| 15:13.51 | Thazza | kaldemar: http://pastebin.com/XgHmdpte - Here is my DP |
| 15:13.57 | leifmadsen | first line doesn't have a closing ) |
| 15:14.13 | leifmadsen | MAILBOX_EXISTS doesn't have an opening double quote |
| 15:14.22 | leifmadsen | ${EXTEN} will always be 's' |
| 15:14.23 | Thazza | leifmadsen: So is there a way to stop my double up of code. |
| 15:14.41 | leifmadsen | Thazza: when you run into issues of duplicate code, the answer is a GoSub() |
| 15:15.09 | [TK]D-Fender | More comments than code.... |
| 15:15.18 | leifmadsen | [TK]D-Fender: that's a good thing |
| 15:15.47 | [TK]D-Fender | leifmadsen, Not always..... |
| 15:16.07 | [TK]D-Fender | If I need a book to understand 1 line of code... just give up :p |
| 15:17.11 | leifmadsen | [TK]D-Fender: yes always |
| 15:17.16 | [TK]D-Fender | Thazza, exten s,n,GotoIf("$[NumberInSQL]" <> "NULL")?NoVMMobileFound:NoVMNoMobile <- wrong braces for variable reference |
| 15:17.20 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
| 15:17.21 | Thazza | So it looks better. |
| 15:17.21 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 15:17.43 | leifmadsen | Thazza: generally looks better, unfortunately you have lots of syntax issues |
| 15:17.49 | [TK]D-Fender | exten s,n(NoVMNoMobile),NoOp(Good We have No Voicemail Box, And Mobile was found, so Send Message saying this) |
| 15:17.50 | [TK]D-Fender | exten s,n,Goto(EndofMadness) |
| 15:18.14 | [TK]D-Fender | You jump to NoVMNoMobile only to jump on the NEXT LINE. And the only thing you effectively did was a NoOp |
| 15:18.30 | [TK]D-Fender | Which isn't really what I'd call "effective". |
| 15:18.47 | [TK]D-Fender | I'd say this whole pile is at least 5 times bigger than it need to be. |
| 15:19.02 | Thazza | [TK]D-Fender, The NoOp's are just there for comments for when i add the other 4 lines that the NoOp gets replaced by. |
| 15:19.22 | Thazza | Comments will be removed before it goes in my DP. |
| 15:19.59 | *** join/#asterisk zkn (~ergo@82.131.70.188.cable.starman.ee) |
| 15:21.34 | Thazza | Thank you for at least answering the question. You can't simplify * DP code, This was always what i thought I would have to do, just wanted to know if it could be easier.. I will turn sections into GoSub. |
| 15:25.15 | [TK]D-Fender | Thazza, AEL would give you a CASE structure that would cleaner for this one segment, and I'm sure you could shorten it up a little bit more regardless, but there is a certain size it will have to be.... |
| 15:27.50 | *** join/#asterisk stix (~stix@94.191.220.185.bredband.3.dk) |
| 15:29.36 | Thazza | [TK]D-Fender: So i should look into learning to use AEL? |
| 15:30.45 | [TK]D-Fender | Not really... |
| 15:31.16 | [TK]D-Fender | just saying in a few paces it can look a bit cleaner. just not worth the overhead and extra learning, risk of bugs, etc |
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| 15:50.49 | ke-esc | Hi All.. So I'm trying to follow the advice of keeping user phone extensions abstracted from devices. Devices are named with mac addresses, and the dialplan takes the user extension, say 199, and sends it to the proper device (this is all database driven). Problem is, when user logs into a device, I edit the mailbox column for the device, but its not reread until I reload SIP in asterisk, thus the MWI doesn't work.. any ways around this? |
| 15:51.42 | p3nguin | If you are using realtime SIP configuration, you probably aren't supposed to run sip reload very much. |
| 15:52.18 | p3nguin | Thank you for naming your devices correctly, though. |
| 15:53.54 | ke-esc | p3nguin, I don't want to use sip reload at all :) I'm just not sure how to trigger the MWI change |
| 15:54.53 | *** join/#asterisk Sorcier_FXK (~nsystem@unaffiliated/sorcierfxk) |
| 16:08.14 | leifmadsen | ke-esc: you need to have polling enabled in voicemail.conf and the peers need to be cached (configured via sip.conf) |
| 16:09.59 | ke-esc | leifmadsen, hmm, i actually just turned off caching of peers figuring that was what was preventing me from updating the mailbox column? |
| 16:10.18 | leifmadsen | no, if you don't have the peers cached, then asterisk has no idea where to send the MWI messages |
| 16:11.43 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:12.50 | ke-esc | ok giving it a try now |
| 16:18.50 | ke-esc | doesn't seem to be working... when i move mailbox 198 from one peer to another, the mwi still is on the old peer, but doesn't come on the new peer (i set pollfreq to 10) |
| 16:27.16 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v021-062.mobile.uci.edu) |
| 16:27.31 | ke-esc | Okay, this is getting a bit more complicated than I'd hoped- but what if I created a "ghost" vm box for each device that remains static... I then call a script on user login/logout and with externnotify to "transfer" voicemails from the users vm box to the device mailbox their logged into, or at least just the txt files used for MWI |
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| 16:41.21 | ke-esc | Haha, I think that is going to work! Combination of cached peers, vm polling, device mailboxes and some simple scripts |
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| 16:56.31 | Alborracho | leifmadsen: hi, do you know if /etc/init.d/asterisk need som kind of tweak? i did "service asterisk start" dont get any error, i looked over the /varlog/messages and everything its ok, the process is up and running, but when i try to log in i get "does /var/run/asterisk/asterisk.ctl exist?" i looked over the folder and that file doens exist |
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| 16:57.08 | leifmadsen | Alborracho: asterisk probably isn't starting or has an error -- try 'asterisk -cvvv' and see what you need to fix to make asterisk start. |
| 16:57.15 | leifmadsen | I've never modified the init script |
| 16:58.41 | Alborracho | i did "service asterisk stop" then "asterisk -cvvvv" and i get this "Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk -r' to connect." |
| 16:59.05 | Alborracho | the process is down i dont see it with "ps ax | grep asterisk" |
| 17:17.54 | p3nguin | Stop asterisk if it is running. Delete the pid file if it exists. Start asterisk again using asterisk -cvvv and see what happens. Pastebin the entire output. |
| 17:18.02 | p3nguin | alborracho: ^ |
| 17:18.08 | p3nguin | ~pb |
| 17:18.08 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 17:19.34 | Alborracho | p3nguin: let me see |
| 17:20.06 | Alborracho | the server atm its keeping me out of the console |
| 17:20.12 | p3nguin | Crap. I meant "delete the ctl file." |
| 17:20.18 | p3nguin | I said pid, but meant ctl. |
| 17:22.45 | *** join/#asterisk bowzak (~bowzak@95.170.203.162) |
| 17:23.55 | bowzak | hi all. can anyone recommend a good provider that can do a US and UK DID with a 20 line trunk ? I'm using elastix |
| 17:24.17 | p3nguin | There is no such thing as a 20 line trunk in VoIP. |
| 17:24.53 | bowzak | maybe i'm wording it wrong.. 20 channels |
| 17:26.51 | p3nguin | If Asterisk says 'Got SIP response 486 "Busy Here" back from 192.168.1.112', does this indicate that the call would have never been displayed on the phone as in incoming call? I'm trying to determine if the phone would show a missed call in this case. |
| 17:27.28 | [TK]D-Fender | p3nguin, could be a "full", or "DND", or "Reject". hard to say |
| 17:27.58 | p3nguin | I'd bet on DND (busy) being enabled, but I can't know for sure since I am 1000 miles from the phone. |
| 17:28.22 | p3nguin | With DND set, the phone would never show a missed call, right? |
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| 17:31.42 | fbnts | Hi, where is the best place to set the MONITOR_EXEC variable? Should I do it in the dialplan using Set(MONITOR_EXEC=/usr/local/bin/scriptname) or in the extension.conf file? |
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| 17:33.06 | glaz | in the dialplan fbnts |
| 17:33.15 | p3nguin | extensions.conf CONTAINS the dial plan. |
| 17:33.36 | glaz | I probably ment the [general] section of extension.conf |
| 17:33.41 | glaz | I guess... |
| 17:35.19 | p3nguin | I think you can do it as a global or within a calling context... but never in the general section. |
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| 17:38.16 | fbnts | ah sorry yeah what I meant was should I set it the global rather than in each individual dialplan flow. will give that a go |
| 17:38.59 | p3nguin | If you will use the same value for it in every other place in dial plan, I'd think setting a global would be the most efficient. |
| 17:42.50 | fbnts | Is there anything else I need to do to enable it? I have it set and am calling Monitor(wav,${CALLFILENAME},m) which is recording the two seperate files but not calling the script upon exit |
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| 18:05.13 | fbnts | Is there a special way to define MONITOR_EXEC variable in extensions.conf? |
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| 18:29.07 | p3nguin | Is there any way to permanently disable DND on the SPA-942? I need to make it so the person can't enable it with the soft key. |
| 18:35.08 | p3nguin | I found a user guide, but I can't find an admin guide. |
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| 19:01.12 | rossand | if I tcpenable=yes, can I set transport=tcp or transport=udp on a case by case basis? |
| 19:15.19 | Andee | you can even put transport=udp,tcp to allow a user/friend to use either at their own disgression iirc |
| 19:19.43 | rossand | Andee: excellent, that was what I thought. Just wanted to double check. Thanks so much. |
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| 20:11.30 | aiksa[LV] | hi, a quick late friday question; a bit of offtopic as other package that asterisk is concerned |
| 20:12.21 | aiksa[LV] | when using linphonec how should I specify targeted soundcard if I have more than one soundcard with similar name |
| 20:12.25 | aiksa[LV] | ? |
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| 21:11.29 | phix | hey, can I set fromuser in extensions.conf instead of sip.conf? or does fromuser in sip.conf always overwrite / take precedence? |
| 21:13.26 | [TK]D-Fender | phix, no. Yes. Respectively |
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| 21:16.14 | phix | [TK]D-Fender: ok, so if I don't set the fromuser in sip.conf but set set(CALLERID(NUM)=${SOMENUMBER}) will that have the same effect? |
| 21:16.52 | [TK]D-Fender | generally, yes |
| 21:19.24 | phix | yay it worked |
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| 21:37.16 | [TK]D-Fender | Checkout time, later all |
| 21:41.36 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 21:41.38 | [sr] | howdy |
| 21:42.13 | [sr] | does make menuselect saves a file to backup ? like the .config in the kernel |
| 21:42.23 | [sr] | a file with the result/confs of the menuselect |
| 21:42.56 | russellb | menuselect.makeopts |
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| 21:45.08 | [sr] | cool |
| 21:45.10 | [sr] | thanks :) |
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| 22:19.39 | ChrisInSydneyToo | p3nguin: You still looking for SPA942 stuff ? |
| 22:22.04 | p3nguin | I'd like to know how to permanently disable the ability to select DND on the phone itself. |
| 22:22.48 | p3nguin | The receptionist seems to hit the DND key "accidentally" and then calls are rejected left and right. |
| 22:23.11 | p3nguin | She was told to watch for it and if it gets enabled, disable it. |
| 22:23.15 | p3nguin | But it happened again. |
| 22:23.26 | p3nguin | So I would like to shut it off. |
| 22:23.39 | ChrisInSydney | Sell her a Snom. I know where it is on there ;-) |
| 22:23.50 | p3nguin | There's no selling involved. |
| 22:24.10 | p3nguin | They have phones. She was told about the key. I want to ensure it never happens again. |
| 22:24.30 | p3nguin | I went into the User configuration and set DND to no, but I think that just toggles it like the key does. |
| 22:24.31 | ChrisInSydney | I was beeing cheeky |
| 22:26.12 | dano0 | dnd is usually handled on the phone side |
| 22:27.01 | ChrisInSydney | http://www.cisco.com/en/US/docs/voice_ip_comm/csbpipp/ip_phones/administration/guide/spa_wip_admin.pdf |
| 22:27.14 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 22:27.39 | ChrisInSydney | STEP 1 Log in to the web administration interface. |
| 22:27.39 | ChrisInSydney | STEP 2 Click Admin Login and advanced. |
| 22:27.39 | ChrisInSydney | STEP 3 Click the Phone tab. |
| 22:27.39 | ChrisInSydney | STEP 4 Under Supplementary Services, under DND Serv, choose yes. |
| 22:27.48 | ChrisInSydney | or in your case no |
| 22:27.51 | p3nguin | I did that. |
| 22:28.18 | p3nguin | I was under the impression that was for DND controlled by a vertical service code, such as *34. |
| 22:28.33 | ChrisInSydney | did you use a bigger hammer, swear alot, you know the usual |
| 22:28.46 | ChrisInSydney | not that it does much,m but I feel better after doing that myself |
| 22:29.07 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 22:29.39 | ChrisInSydney | from memory, and I havent got much of that left, you can tie a service number to the DND key, or it can just work on the phone its self |
| 22:30.28 | ChrisInSydney | I am assuming you have the latest (last decade) firmware |
| 22:31.01 | *** join/#asterisk talntid (~t@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 22:31.10 | p3nguin | I've never used nor touched a 942, so I don't have the foggiest idea. |
| 22:31.34 | talntid | Besides faxing via asterisk... do guys guys use vFax services from vitelity, ringcentral... or who? |
| 22:31.55 | talntid | looking for alternatives, as neither vitelity or ringcentral can handle my requests |
| 22:32.40 | p3nguin | Clients have this tendency to use phones that I do not have on hand, so I can't poke around to find out what makes them tick. |
| 22:32.48 | ChrisInSydney | I have had one client with a single 942 |
| 22:33.09 | ChrisInSydney | but that got dumped last year some time. |
| 22:33.28 | Li7h | talntid, I use a service called faxsipit.com since a lot of our clients "need" a fax and it's worked great so far |
| 22:33.45 | Li7h | they provide an ata and it connects directly to an analog fax machine |
| 22:34.12 | p3nguin | If they ever have to get new phones, I'll do my best to make sure they get either the new Digium phones or Polycoms. |
| 22:34.30 | ChrisInSydney | new digium phones :O |
| 22:34.41 | talntid | hmm, that's nice info, Li7h |
| 22:34.42 | ChrisInSydney | when did they come out ? |
| 22:34.56 | ChrisInSydney | we're a little isolcated here |
| 22:34.59 | ChrisInSydney | we = I |
| 22:35.00 | talntid | I need email-to-fax service, but for another project, I do need that ATA service |
| 22:35.01 | ChrisInSydney | by choice |
| 22:35.53 | p3nguin | http://www1.digium.com/en/products/phones |
| 22:36.58 | p3nguin | I got the sneak peek email on Feb 1. |
| 22:37.12 | ChrisInSydney | http://www.digium.com/en/error/ |
| 22:37.16 | ChrisInSydney | I got a 404 |
| 22:38.06 | ChrisInSydney | p3nguin: Commercially, if you were a mechanic, I would suggest at this point to "Jack up the number plates" |
| 22:38.19 | p3nguin | wtf |
| 22:38.37 | ChrisInSydney | swap the phone out, |
| 22:38.54 | *** join/#asterisk slingr (santas@will.one.day.hack-the-pla.net) |
| 22:38.59 | ChrisInSydney | jack up the numberplates, replace everything in between |
| 22:39.47 | ChrisInSydney | sorry, too early on a Sat morning here |
| 22:40.31 | ChrisInSydney | P3nguin: I have a SPA525G2 here. I'll have a play and see what it does. May be of some help to you |
| 22:40.45 | p3nguin | It could have a similar DND setup. |
| 22:41.02 | ChrisInSydney | must go and attend to the 2 year old. Felix. |
| 22:41.06 | ChrisInSydney | probably |
| 22:41.15 | ChrisInSydney | he is trashing the place |
| 22:41.29 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:41.30 | ChrisInSydney | type soon & good luck :-) |
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| 23:13.31 | ChrisInSydney | Hi all, just a "quick" one.... |
| 23:14.56 | ChrisInSydney | Is there any way to force CONNECTEDLINE info to the calling handset without using an Answer() but prior to the called party picking up the call |
| 23:14.58 | ChrisInSydney | ?? |
| 23:15.07 | ChrisInSydney | eg: exten => s,n,Set(CONNECTEDLINE(name,i)=${NAME}) |
| 23:15.35 | ChrisInSydney | will not display on the handset unill an answer |
| 23:16.03 | WIMPy | Wenn, if you hafe an ,i there it won;t do anything. |
| 23:16.09 | ChrisInSydney | either in the dial plan or by the called bparty |
| 23:16.44 | WIMPy | You could also try REDIRECTIONG. |
| 23:16.59 | WIMPy | REDIRECTING |
| 23:17.00 | ChrisInSydney | WIMPy: 'splain |
| 23:17.25 | ChrisInSydney | WIMPy: you are typing like its Friday |
| 23:17.29 | ChrisInSydney | :D |
| 23:17.30 | WIMPy | But only if Asterisk works like normal telephon which I haven;t tested for that. |
| 23:17.47 | WIMPy | I guess I'm 17 minutes late then :-) |
| 23:18.24 | ChrisInSydney | hehe |
| 23:18.27 | WIMPy | You know the ,i is there to tell it not to do anything. |
| 23:18.39 | ChrisInSydney | ahh |
| 23:18.46 | ChrisInSydney | let me try |
| 23:18.51 | ChrisInSydney | without the i |
| 23:19.32 | WIMPy | The idea is ath you can set multiple values and only set the last one without ,i so all information is sent in one go. |
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| 23:20.40 | ChrisInSydney | WIMPy_: you are a F*&^%t genius. I dont care what p3nguin says about you , you're allright ;-) |
| 23:20.57 | ChrisInSydney | thanks heaps |
| 23:21.05 | WIMPy | What did he say? o.O |
| 23:21.24 | ChrisInSydney | nothing: just stirring |
| 23:21.53 | ChrisInSydney | not that it needs much this time of the day / week |
| 23:22.04 | WIMPy | Ok, so you make the drinks from now on :-) |
| 23:22.31 | ndespres | is everyone else drinking already, too? |
| 23:22.36 | ChrisInSydney | I'm a Aussie, isn't that what we are paid to do ? |
| 23:23.06 | WIMPy | But having CONNECTEDLINE without a connection seems a litle strange anyway. |
| 23:23.30 | WIMPy | Well, surely you've got the right time of year for that. |
| 23:23.34 | ChrisInSydney | When I dial a number, I would like the name of the person to show on the handset |
| 23:24.12 | ChrisInSydney | Yup, just heading into autum, however, its raining heaps at the moment |
| 23:24.51 | ChrisInSydney | bit of flooding too |
| 23:24.59 | WIMPy | Warm rain? |
| 23:25.09 | ChrisInSydney | tempid rain |
| 23:25.27 | ChrisInSydney | I'm still in a tee and shorts |
| 23:25.30 | WIMPy | So you can shower for free? |
| 23:27.45 | ChrisInSydney | yep. |
| 23:29.15 | ChrisInSydney | Its also Gay and Lesbian Mardi Gras this weekend too. Every year it rains heavily. All the "angry God botherers" are saying that "this is a sign" then 30 mins before it starts, it clears up., |
| 23:29.36 | ChrisInSydney | must go, 2year old on the rampage |
| 23:29.37 | WIMPy | LOL |
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| 23:54.47 | phix | ChrisInSydney: hehe |
| 23:55.20 | phix | ChrisInSydney: Never been to one, probably wont go tonight either |
| 23:56.00 | ChrisInSydney | phix: You in aus too ? |
| 23:56.07 | phix | yup |
| 23:56.15 | ChrisInSydney | sydney or where ? |
| 23:56.25 | phix | yeah Sydney |
| 23:56.55 | ChrisInSydney | Where ? I'm around the eastern suburbs beaches |
| 23:57.08 | phix | Sutherland area |
| 23:57.14 | ChrisInSydney | a shire lad |
| 23:57.19 | phix | yup |
| 23:57.24 | phix | not a lad though :) |
| 23:57.41 | ChrisInSydney | :) |
| 23:57.45 | phix | lads live in bankstown |
| 23:57.53 | ChrisInSydney | Yeah, the floats dont head quite that far south |
| 23:57.59 | ChrisInSydney | true |
| 23:59.03 | ChrisInSydney | Thats right, after the Cronulla riots, they were talking about building a new motorway from there to the shire...the Middle Eastern Distributor |
| 23:59.19 | phix | heh |
| 23:59.36 | ChrisInSydney | poor taste I know |
| 23:59.40 | phix | very :) |