| 00:07.36 | *** join/#asterisk p3nguin_ (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
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| 00:59.36 | ritztech | anyone know if i can shorten the Call coming in I See the Starting simple switch on 'DAHDI/5-1' RIGHT away but takes like a few seconds |
| 01:00.13 | [TK]D-Fender | ritztech: in your dahdi configs do "usecallerid=no" |
| 01:00.33 | [TK]D-Fender | ritztech: Oh, and I found out how I'm going to queue things up for that project. Code pending tomorrow |
| 01:00.47 | ritztech | swibby :) |
| 01:01.37 | ritztech | ohhh My boss was getting to me so we had to find someone real quick He did some sort of Bash and Quefile Directortys with the Tilda ~ and made it a loop.sh kinda thing |
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| 03:06.34 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 03:07.57 | ruben23 | hi guys, i ahve installed asterisk on my ubuntu desktop laptop now the problem is whenh i do login using sip directly to my asterisk on laptop-it wont allow..any idea..? im kee[ getting error. |
| 03:11.34 | *** join/#asterisk deeperror (~deeperror@d149-67-49-94.try.wideopenwest.com) |
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| 03:15.18 | ManxPower | ruben23, don't run a sip client and sip server on the same host. |
| 03:15.53 | ManxPower | If you MUST do that, you might be able to change the SOURCE port on the sip CLIENT to something other than 5060 (do not change the destination port) |
| 03:15.59 | p3nguin_ | Or if you do, changes the ports. |
| 03:16.53 | p3nguin_ | When I used to test Asterisk on my desktop system, I would just change my client's port to 5061 and leave Asterisk on the normal 5060. |
| 03:17.10 | ManxPower | is in a good mood. Just managed to do live updates of extension status (in use, ringing, etc) on a web page without using Flash or Java |
| 03:17.21 | ManxPower | p3nguin_, you were changing the source port. |
| 03:17.51 | p3nguin_ | well, yeah. |
| 03:18.41 | ManxPower | p3nguin_, I don't use softphones, but as I understand it (and I know this applies to Sipura ATAs and hardphones) they don't exactly make source port .vs. destination port very clear in their config screens |
| 03:19.12 | ruben23 | p3nguin_: i should chnage my softphones..? |
| 03:19.26 | ruben23 | port into different. |
| 03:20.55 | p3nguin_ | In the same way that I use two different ports for an SPA-3102... you can't use the same port on your softphone running on the Asterisk box. |
| 03:23.43 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 03:25.42 | ManxPower | ruben23, not should, MUST |
| 03:33.42 | ruben23 | ok thanks guys it works.. |
| 03:34.44 | ManxPower | ruben23, remember in SIP there really isn't a "server" and a "client" in the more traditional sense of the words. All sip devices/programs are both servers and clients. You would not try to run two web servers on the same port at the same time. |
| 03:40.26 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 03:40.40 | ruben23 | ok noted |
| 03:44.20 | AliRezaTaleghani | hi all |
| 03:44.30 | AliRezaTaleghani | i have a problem |
| 03:44.34 | AliRezaTaleghani | :( |
| 03:46.33 | AliRezaTaleghani | can't find the place of the dailplan which executed when the caller in the queue will pass to an agent |
| 03:46.47 | AliRezaTaleghani | let explain my idead |
| 03:46.54 | ruben23 | <PROTECTED> |
| 03:47.24 | AliRezaTaleghani | ruben23: what do u mean? :) i can't understant |
| 03:47.24 | p3nguin_ | Once the call is in the Queue(), dialplan is not progressing. |
| 03:48.12 | AliRezaTaleghani | p3nguin_: uhum.. so how can i play some thing (like the Agent ID or recorded Name) as the agent want to answer the caller? |
| 03:50.45 | p3nguin_ | I don't know. |
| 03:51.34 | AliRezaTaleghani | p3nguin_: umm :) by the way... tnx |
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| 04:20.10 | *** join/#asterisk FabiOne (~fabi@151.13.190.23) |
| 04:20.16 | FabiOne | hi all |
| 04:35.04 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
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| 05:27.03 | *** join/#asterisk mike-ekim (~itp@72.46.242.59) |
| 05:27.17 | mike-ekim | how can I have two of the same extensions on the same server |
| 05:28.13 | p3nguin_ | Use different contexts. |
| 05:28.26 | mike-ekim | can you link me to an online example please? |
| 05:28.44 | mike-ekim | i think the terminology im using is making a good search difficult |
| 05:29.33 | mike-ekim | and - would they be able to both register on the same pbx? |
| 05:30.04 | p3nguin_ | Extensions do not register. |
| 05:30.13 | [TK]D-Fender | mike-ekim: You want 2 DEVICES to register to the same peer? |
| 05:30.16 | p3nguin_ | Extensions are the phone numbers used to call devices. |
| 05:30.50 | mike-ekim | I mean devices, not extensions |
| 05:30.56 | [TK]D-Fender | Or do "something" |
| 05:31.24 | mike-ekim | if multiple devices are using the same username to log in, how will asterisk authenticate? |
| 05:31.32 | p3nguin_ | If you want one extension to call two phones, that's not hard to do. |
| 05:31.46 | mike-ekim | cause i might have extension 100 for customer-a and extension 100 on customer-b too |
| 05:32.33 | p3nguin_ | exten => 1234,1,Dial(SIP/00001111eeff&SIP/00112233abcd) |
| 05:33.41 | [TK]D-Fender | mike-ekim: An EXTENSION is a number you dail. What it does depends on your DIALPLAN |
| 05:33.50 | [TK]D-Fender | mike-ekim: And that's what contexts are for |
| 05:34.00 | mike-ekim | oh, well I meant a SIP user, sorry |
| 05:34.04 | [TK]D-Fender | mike-ekim: So that you keep their "100" away from someone else's |
| 05:34.14 | mike-ekim | right |
| 05:34.18 | [TK]D-Fender | mike-ekim: Don't use NUMERIC SIP usernames then |
| 05:34.18 | mike-ekim | I understand contexts |
| 05:34.23 | mike-ekim | ok fine |
| 05:34.38 | [TK]D-Fender | mike-ekim: And no, obviously they CAN'T share the same name |
| 05:34.52 | mike-ekim | ok, no problem. |
| 05:34.54 | mike-ekim | Thanks |
| 05:35.03 | [TK]D-Fender | NEXT !@@!@@!#!!@' |
| 05:35.08 | [TK]D-Fender | ... (c) BKW |
| 05:35.56 | mike-ekim | exit |
| 05:36.09 | mike-ekim | quit |
| 05:42.17 | [TK]D-Fender | BAI BAI\ |
| 05:42.29 | p3nguin_ | bi |
| 05:42.36 | p3nguin_ | (sexual) |
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| 06:21.57 | *** join/#asterisk ritztech (~ritztech@ip70-189-221-21.lv.lv.cox.net) |
| 06:23.43 | ritztech | anyone know how to make the Process quciker on a Outbound call FXO i answer the call so i know theres timing somewhere (using .call files so maybe an entry in there) ? |
| 06:24.01 | ritztech | exten => s,1,playback(beep) thats the first part of the context takes 7 seconds |
| 06:24.09 | ritztech | is it like a global setting |
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| 07:16.31 | ritztech | i treid this but it killed the call to even come in |
| 07:16.31 | ritztech | <PROTECTED> |
| 07:37.17 | ritztech | found it it was callprogress=no BUT NOW if the call fails it wont work haha damm back to the drawing board |
| 07:50.09 | ChannelZ | if you're using call files, it will ring the Channel, and then once you answer it calls the Extension or runs the Application |
| 07:50.16 | *** join/#asterisk [SySteM] (~antoine@aqu33-6-88-168-80-163.fbx.proxad.net) |
| 07:50.46 | [SySteM] | Hello.. need some help please. |
| 07:50.58 | p3nguin_ | Funny how that works. |
| 07:51.23 | [SySteM] | Im on asterisk 1.4, impossible to put option "hH" (hangup on *) on a dial exten |
| 07:51.34 | ChannelZ | If what you're calling happens to be slow (if it's a POTS line with a lot of digits) there could be substantial delay between you answering and hearing a ring on the remote end |
| 07:52.04 | p3nguin_ | [system]: What? |
| 07:52.09 | p3nguin_ | You aren't making sense yet. |
| 07:52.35 | [SySteM] | Im on a AGI script (php) |
| 07:52.48 | ChannelZ | Show us your boobs |
| 07:52.54 | p3nguin_ | moobs |
| 07:52.54 | ChannelZ | I mean show us your Dial statement |
| 07:53.13 | [SySteM] | $AGI->exec("Dial","SIP/*4780007770000@sipbroker","hH"); |
| 07:54.21 | [SySteM] | Dial destination not on my serv, but can i have control on the hH |
| 07:54.25 | ChannelZ | I suppose that depends on the PHP lib you're using |
| 07:54.30 | p3nguin_ | And when you press the button defined in features.conf, it doesn't hangup? |
| 07:54.53 | [SySteM] | <PROTECTED> |
| 07:54.59 | [SySteM] | not really complicate :) |
| 07:56.07 | [SySteM] | ho yes ! |
| 07:58.15 | ChannelZ | unless 1.4 is different features.conf shouldn't have anything to do with it, the h flags specify * |
| 08:02.33 | ritztech | well POTS is BOOOOOOBYS |
| 08:02.49 | ritztech | haha well POTS im using but its just 4 digit Dialing to my MITEL system |
| 08:02.54 | ritztech | but i think i found it |
| 08:03.22 | ritztech | It works INSTANT when i set callprogress=no |
| 08:04.06 | ritztech | BUT if the call is a Busy tone it wont Retry the call (according to my .call file) so its it OR theres no way |
| 08:04.24 | ritztech | i did try putting busydetect=yes and also a busycount=2 but that still didnt work |
| 08:04.44 | [SySteM] | i try by AGI directly : DIAL SIP/num hH and DIAL SIP/num,hH |
| 08:04.49 | [SySteM] | no one running :( |
| 08:05.31 | ChannelZ | does dtmf otherwise work on that channel? |
| 08:05.50 | ritztech | im assuming it does |
| 08:06.04 | ritztech | Attempting call on dahdi/g0/8930 for 8930@from-internal:1 |
| 08:06.07 | ritztech | 5 seconds later |
| 08:06.21 | ritztech | Executing [s@macro-playPage:20] Playback("DAHDI/1-1", "/var/spool/asterisk/pages/8930-1284882265 |
| 08:06.35 | ritztech | <PROTECTED> |
| 08:06.53 | ChannelZ | I meant System |
| 08:07.24 | ChannelZ | but to you, analog call progress is hit-or-miss at best |
| 08:08.26 | ritztech | i was reading that somewhere too haha |
| 08:09.08 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:09.38 | ritztech | but i have wait for retrys = 10 in my .call files and if I dont have callprogress=yes then it will fail IF the dialed ext answers as BUSY or fails |
| 08:09.56 | ritztech | so i might have to live with the 5 - 7 second answer delay |
| 08:18.59 | *** join/#asterisk Intel`` (~clarencec@94.200.7.26) |
| 08:19.17 | Intel`` | guys is there a way to not let asterisk send out 503 error? |
| 08:30.41 | ChannelZ | yeah.. fix whatever it is causing them |
| 08:32.09 | Intel`` | ChannelZ: i wanted to use Hangup(503) so it will automatically hangup if it detect 503. but im using asterisknow not sure where to put it |
| 08:48.30 | Intel`` | ok not Hangup(503) but should be Hangup(PRI_CAUSE) |
| 08:49.23 | Intel`` | i get a lot of cause(31,17,1,27,16,19,18,21) and i wanted asterisk to act on it by just hanging up |
| 08:49.32 | *** join/#asterisk linagee (~linagee@about/linux/staff/linagee) |
| 08:49.51 | Intel`` | not by unregistering the account |
| 08:59.23 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 08:59.33 | elliot98 | waves to all |
| 08:59.56 | elliot98 | how do I set the port for the host when using the mysql application? |
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| 09:12.35 | *** join/#asterisk wuffi600 (~keck0f@g230078157.adsl.alicedsl.de) |
| 09:12.46 | wuffi600 | hi. |
| 09:25.15 | wuffi600 | if is start asterisk daemon i get message "Starting Asterisk PBX: Unable to set high priority". what does this mean? |
| 09:27.18 | ectospasm | it generally means that Asterisk wasn't able to get real-time priority |
| 09:27.22 | ectospasm | are you starting it as root? |
| 09:27.40 | ectospasm | wuffi600: ^^^ |
| 09:29.01 | elliot98 | wuffi600: are you running using root or user? |
| 09:31.49 | *** join/#asterisk lore20 (5d929700@gateway/web/freenode/ip.93.146.151.0) |
| 09:31.57 | lore20 | hello |
| 09:32.22 | lore20 | i am using asterisk 1.4 and i found out that if i set the "t" or the "T" option in the Dial() command |
| 09:32.36 | lore20 | asterisk doens't attempt to "native-bridge" the two peers |
| 09:33.18 | lore20 | i have read somewhere that transfer can also be handled with the SIP-INFO DTMF that always go throught the sip proxy also when there is a direct rtp stream |
| 09:33.46 | ectospasm | Asterisk is not a proxy, unless you allow reinvites it will always be in the media path |
| 09:34.01 | lore20 | i allowed reinvite and both proxy supports that |
| 09:34.25 | lore20 | without the transfer option i get a direct-bridge |
| 09:34.36 | ectospasm | even then I think Asterisk will overwrite the SDP, some proxies don't like that |
| 09:35.38 | lore20 | now i am working with two sip clients, without nat |
| 09:36.28 | lore20 | the problem is that the sip proxy is connected with a wifi bridge, so i would like to limit the rtp sessions that useless go back and forward to the bridge |
| 09:37.09 | lore20 | that as a low bandwidth |
| 09:37.12 | lore20 | *has |
| 09:37.56 | lore20 | with "sip proxy" i meant "asterisk" |
| 09:38.23 | wuffi600 | ectospasm, no, is runs as use asterisk. that seems to be to problem. thank you for the hint. |
| 09:38.48 | wuffi600 | elliot: thanx |
| 09:39.18 | ectospasm | wuffi600: yeah, Asterisk tends to have better performance when run as root. Some would argue that's a security concern... |
| 09:41.43 | wuffi600 | there is a nother thing: i would like to allow all my sip-clients to do videocalls. so i added videosupport=yes in the [general] section. now clients can do video calls, but in their video windows there is only black screen (no picture). all firewalls are complete open (it's a test-installation). |
| 09:42.38 | wuffi600 | do i have to device what codecs to use for video on the client or will a supported codec be used atomatically? (linphone/qutecom/x-lite) |
| 09:42.53 | wuffi600 | or do i have to define in sip.conf what codecs are allowed? |
| 09:44.46 | elliot98 | how to do you set the port for the database when using the mysql application? |
| 09:44.59 | ectospasm | wuffi600: are they the same type of endpoints? |
| 09:45.19 | ectospasm | elliot98: for Asterisk or for mysql? |
| 09:45.26 | wuffi600 | ectospasm, i've tested linphone to linphone, xlite to xlite. |
| 09:45.36 | ectospasm | You need to have Asterisk match MySQL |
| 09:45.49 | ectospasm | wuffi600: softphones generally suck |
| 09:46.09 | wuffi600 | ectospasm, is there a diagnostic reference softphone fro debugging? |
| 09:46.28 | ectospasm | wuffi600: not that I know of. You may find something on voip-info.org |
| 09:47.20 | wuffi600 | ectospasm, if there is no codec denied and no codec allowed for video in sip.conf. should it work? or do i have to allow videocodecs explicitely? |
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| 09:52.19 | ectospasm | wuffi600: I always explicitly disallow all, and only allow what I want |
| 09:52.26 | ectospasm | ...it's supposed to include all codecs if you don't disallow all nor allow any specific codec, but I don't know if it will include video codecs |
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| 10:18.35 | elliot98 | ectospasm: for mysql application |
| 10:19.04 | elliot98 | ectospasm: I see options to set the host, username, password, etc., however, I do not see any option for the port |
| 10:44.36 | Gugge | i havent tried, but maybe set the host to ip:portr |
| 10:44.40 | Gugge | ip:port |
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| 11:32.33 | Jasnejac | hi all. has anyone got app_konference running with 1.8 beta5 yet? |
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| 13:07.29 | neurosys | using openser(milkfish) remote phones cant xfer. i get SIP/2.0 481 Call leg/transaction does not exist in the debug. Cant help but think im missing something very simple. but cant think of it |
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| 13:21.59 | Dougy | http://paste2.org/p/993983 |
| 13:22.02 | Dougy | im currently stuck here |
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| 13:59.04 | Dougy | :( |
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| 14:42.51 | elliot98 | regarding choosing a port for the mysql function, some of the code needed to be changed |
| 14:46.22 | elliot98 | do you think the change should be submitteD? |
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| 14:52.58 | russellb | elliot98: attach patches to a report on https://issues.asterisk.org |
| 14:53.53 | elliot98 | it is for 1.4 version of asterisk |
| 14:54.02 | elliot98 | are patches still allowed for that version? |
| 14:54.38 | elliot98 | patches that add features, that is |
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| 14:55.25 | russellb | no |
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| 15:08.39 | ruben23 | hi guys how do i used the features of sending emailo with voicemail on asterisk, do i need to have a mail server also on my asterisk..? |
| 15:09.57 | [TK]D-Fender | ruben23: You need an MTA. * calls the standard "sendmail" binary for this to send it. Most distros would have this by default |
| 15:10.21 | [TK]D-Fender | ruben23: And unless you need to set it to go through an ISP's mail proxy it may work directly |
| 15:10.50 | *** join/#asterisk mateu (~mateu@li133-62.members.linode.com) |
| 15:23.39 | Dougy | I have a simple set up.. 2 extensions (adding a third soon), and one queue that when you dial the # all the phones ring.. can i remove everything but queues.conf sip.conf and extensions.conf from /etc/asterisk? |
| 15:23.43 | Dougy | what other configs are key |
| 15:23.44 | Dougy | that i need to keep |
| 15:24.32 | [TK]D-Fender | Dougy: lots indications.conf, rtp.conf, and a pile of others. No VM? might need that too, asterisk.conf obviously.... |
| 15:24.49 | [TK]D-Fender | Dougy: I advise against going psycho on your configs... |
| 15:25.08 | Dougy | hehe |
| 15:25.11 | [TK]D-Fender | Dougy: Because several modules will prevent * from loading if their configs are bad or missing |
| 15:25.16 | Dougy | well, i thought maybe i could clean a lot of it up |
| 15:25.19 | Dougy | but in that case, i will let it be |
| 15:25.46 | Dougy | I guess I ened to figure out my outgoing calling first since its all kinds of screwed up |
| 15:25.53 | Dougy | [TK]D-Fender: did you see my above pastebin? i guess i should update it tho |
| 15:26.17 | [TK]D-Fender | Dougy: What I would advise is specifically disabling things you absolutely won't need, like pbx_aels.so, and protocols you have no intention of using (H.323, IAX, Skinny, SCCP, or whatever). |
| 15:26.39 | Dougy | http://paste2.org/p/994152 |
| 15:26.44 | Dougy | any idea how i could rectify that? |
| 15:26.45 | [TK]D-Fender | Dougy: Yes... perhaps you should fix your existing problems before creating MORE :p |
| 15:26.56 | *** join/#asterisk ritztech (~ritztech@ip70-189-221-21.lv.lv.cox.net) |
| 15:27.01 | Dougy | [TK]D-Fender: just a bit :P |
| 15:28.13 | [TK]D-Fender | Dougy: [Sep 19 09:21:21] NOTICE[14175]: chan_sip.c:15228 handle_request_invite: Call from '1000' to extension '1001' rejected because extension not found. <--- where do YOU see a "1000" in your dialplan that it could possible call? |
| 15:28.33 | [TK]D-Fender | Dougy: I believe I see what you THINK should match it... but look at that one CLSOELY |
| 15:28.35 | Dougy | well, i'm trying to figure it out |
| 15:28.38 | [TK]D-Fender | CLOSELY even |
| 15:28.40 | Dougy | i thought (getting a config line) |
| 15:28.51 | Dougy | exten => _XXXX.,1,Dial(SIP/${EXTEN}) |
| 15:28.58 | Dougy | i thought taht would let me be able to dial a 4 digit extension |
| 15:29.28 | Dougy | http://paste2.org/p/994156 |
| 15:31.22 | [TK]D-Fender | Dougy: Nope. Look at the pattern again, what does each char represent? |
| 15:31.39 | Dougy | i am total newbie here |
| 15:31.45 | Dougy | the X represents a number, does it not? |
| 15:32.02 | [TK]D-Fender | Dougy: Each one is a single digit. and the last? |
| 15:32.10 | Dougy | the dot you mean? |
| 15:32.13 | [TK]D-Fender | (0-9) |
| 15:32.18 | [TK]D-Fender | Dougy: Yes, the dot |
| 15:32.26 | Dougy | I originally had it without the dot |
| 15:32.27 | Dougy | it didnt work |
| 15:32.40 | Dougy | I just mooched the dot off an example in hopes it'd fix it to be honest |
| 15:33.13 | [TK]D-Fender | Dougy: Go read what it means. |
| 15:33.13 | Dougy | holy crow |
| 15:33.18 | Dougy | now it works without the dot, fml |
| 15:33.36 | Dougy | i must have changed something else that made it work without dot |
| 15:34.12 | Dougy | so [TK]D-Fender my next and i guess last issue for now is.. when i call out, i can only call out from software phone.. i cant call out from the hardware phone |
| 15:34.14 | [TK]D-Fender | Dougy: if you say so... |
| 15:34.27 | [TK]D-Fender | Dougy: What "hardware phone"? |
| 15:34.31 | Dougy | cisco 7960G |
| 15:34.37 | Dougy | 3cx software phone works fine |
| 15:34.42 | [TK]D-Fender | Dougy: Guess you didn't configure that one right then |
| 15:35.10 | Dougy | I guess that's what I get for trying to clone a trixbox seutup without the trixbox |
| 15:35.22 | Dougy | [Sep 19 11:06:33] WARNING[14568]: chan_sip.c:13161 handle_response_invite: Received response: "Forbidden" from '"1000" <sip:1000@199.15.253.105>;tag=as3854c31c' |
| 15:37.15 | Dougy | hrm |
| 15:37.46 | [TK]D-Fender | Dougy: Some auth is wrong... |
| 15:38.00 | [TK]D-Fender | Dougy: And congratulations on your newfound liberation from TrashBox |
| 15:38.11 | Dougy | I am trying to get used to diff pbx systems |
| 15:38.18 | Dougy | i tried freeswitch but that just overwhelmed me |
| 15:38.35 | Dougy | ive gone trixbox -> asterisk cli -> pbx in a flash -> freeswitch -> asterisk cli (now) |
| 15:39.10 | Dougy | so [TK]D-Fender you think my issue is an improper password? |
| 15:40.54 | Dougy | I checked sip.conf and the SIPMAC.cnf and it seems fine |
| 15:42.37 | *** join/#asterisk wierdo (jimmy@77.78.3.197) |
| 15:43.57 | [TK]D-Fender | Dougy: pastebin your sip.conf masking only passwords, enable SIP debug at * CLI and capture the call attempt. |
| 15:44.38 | [TK]D-Fender | Dougy: Wait... 1000 REFUSED you... you tried CALLING the cisco? |
| 15:44.54 | [TK]D-Fender | No wait.. I got that backwards. |
| 15:44.58 | [TK]D-Fender | PB as before.. |
| 15:48.02 | Dougy | http://paste2.org/p/994182 |
| 15:51.29 | Dougy | [TK]D-Fender: it scrolls so much on debug i cant even copy it all from my putty window |
| 15:52.04 | [TK]D-Fender | Dougy: Thats what SCROLLBACK is for... |
| 15:52.18 | Dougy | what is? |
| 15:52.21 | Dougy | eek |
| 15:52.36 | [TK]D-Fender | Dougy: Your permit/deny settings are suspect. |
| 15:53.13 | Dougy | why would it work from a software phone and not a hardware phone in same LAN then |
| 15:54.01 | ritztech | penguins geeeesh |
| 15:54.34 | [TK]D-Fender | ritztech: Morning... I'm prepping for a band practice shortly, and mid-afternoon should be back to finazlize things... |
| 15:55.05 | Dougy | [TK]D-Fender: i set 0.0.0.0/0.0.0.0 for permit/deny on extension 1000 and it still doesnt work |
| 15:55.33 | [TK]D-Fender | Dougy: Trash both, reload. pastebin |
| 15:55.38 | Dougy | trassh both? |
| 15:55.41 | Dougy | as in remove the line entirely? |
| 15:56.41 | [TK]D-Fender | jsut comment out for now. |
| 15:57.04 | Dougy | [TK]D-Fender: http://paste2.org/p/994191 |
| 15:57.44 | *** join/#asterisk tzafrir_laptop (~tzafrir@91.135.102.198) |
| 15:58.32 | jamko | Fender: What is the name of your band? |
| 15:59.05 | Dougy | hey, outgoing worked |
| 15:59.06 | Dougy | cool |
| 15:59.13 | Dougy | now to fix caller ID and voicemail after lunch |
| 15:59.14 | Dougy | cheers! |
| 16:00.39 | [TK]D-Fender | jamko: I was in one until 2 weeks ago, left and am looking at a new project. First meet-up today and I only know the drummer (wro brought me in). Don't ahve a name yet and not sure if this mix will be it |
| 16:01.46 | jamko | Fender: Typical situation.. lol .. You play a strat? |
| 16:02.47 | *** join/#asterisk BANSAL (~bansal@117.199.117.233) |
| 16:03.42 | [TK]D-Fender | jamko: Nope, nick has nothing to do with music. I play an Ibanez RG-350DX, and a Hohner HR-1000 neck-thru (custom job) |
| 16:03.54 | [TK]D-Fender | jamAs well as keys, bass, and vocals |
| 16:04.40 | carrar | but can you do it all jimmie hendrix style! |
| 16:05.01 | carrar | behind the head backwards! |
| 16:05.25 | [TK]D-Fender | carrar: Yes. But I have no intention of lighting my guitar on fire afterwards :p |
| 16:05.31 | carrar | haha |
| 16:05.33 | carrar | WUSS!! |
| 16:05.47 | jamko | TK: Nice. You have some recordings on the Internet somewhere? |
| 16:06.19 | carrar | Does Alison wear tight spandex and sing in your band? |
| 16:06.31 | [TK]D-Fender | jamko: A few of my random noodling, a few vids on facebook (not hosted by me) |
| 16:07.00 | Dougy | back |
| 16:07.02 | [TK]D-Fender | carrar: lulz |
| 16:07.25 | carrar | I had to holdback on that comment :) |
| 16:08.02 | jamko | TK: anything on youtube or something searchable? I was at a jam until 3am last night. Slept in, lol.. |
| 16:08.27 | Dougy | [TK]D-Fender: off my last pastebin, it should force caller ID, shouldnt it? |
| 16:08.29 | Dougy | force set* |
| 16:08.52 | [TK]D-Fender | jamko: Actually... YES.. http://www.youtube.com/user/LessThan4Canada |
| 16:09.02 | [TK]D-Fender | jamko: I'm only in 2-3 of these as their lineup changed. |
| 16:09.41 | [TK]D-Fender | jamko: I'm in "All The Young Dudes - Less Than 4", |
| 16:09.48 | [TK]D-Fender | jamko: Hrm.. none of the others so far... |
| 16:10.23 | [TK]D-Fender | jamko: Guess they never updated much... |
| 16:11.44 | jamko | Would you be lead? |
| 16:12.16 | [TK]D-Fender | jamko: In this one, yes. Tall one in black on the left |
| 16:12.50 | carrar | What, no orange asterisk shirt on? |
| 16:13.27 | [TK]D-Fender | carrar: nothing gets the chicks wild like.. NEEEEEEERRRRRRRRRRRRRRRDDDDDDDDDDDD!!!!!!!! |
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| 16:15.05 | carrar | nice waitress outfits there |
| 16:15.56 | carrar | 2:11 |
| 16:16.33 | carrar | I say fire your video person |
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| 16:27.11 | [TK]D-Fender | carrar: Our bassist did it on his Canon P&S (dual meaning) |
| 16:27.15 | jamko | TK: Nice job on the Van Halen.. Found the FB page. They are a decent cover band. |
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| 17:07.58 | neurosys | using openser(milkfish) remote phones cant xfer. i get SIP/2.0 481 Call leg/transaction does not exist in the debug. Cant help but think im missing something very simple. but cant think of it |
| 17:17.21 | drmessano | Wouldn't that be a #openser issue? |
| 17:20.54 | neurosys | drmessano: Surely. But figured one of your guys would have run into this easily enough |
| 17:21.36 | [TK]D-Fender | jamko: link it... |
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| 17:41.56 | Dougy | http://paste2.org/p/994290 .. I want to make it redirect to voicemail after 60 seconds of ringing, how can I do that? |
| 17:42.05 | Dougy | I gathere I need to add it to the extension, but how or where |
| 17:50.51 | p3nguin | dougy: Which extension? |
| 17:52.30 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 17:52.37 | ManxPower | Long shot, but does anyone here have GIF/PNG/etc files of the icons that Polycom phones use for their buddy status? |
| 17:54.53 | Dougy | p3nguin: right now it set up so that all incoming calls to this number go to queue 7000 and it rings all the phones |
| 17:55.02 | Dougy | i want it to ring 7000 for 60sec then go to voicemail for extension 9999 |
| 17:55.14 | [TK]D-Fender | Dougy: then tell Queue() to timeout after 60 seconds |
| 17:55.19 | [TK]D-Fender | Dougy: "core show application queue" |
| 17:55.29 | [TK]D-Fender | Dougy: Otherwise they'll wait around in queue forever |
| 17:56.04 | Dougy | right |
| 17:56.06 | Dougy | i'm asking how to do that |
| 17:56.19 | Dougy | timeout = 60 |
| 17:56.21 | Dougy | is that all? |
| 17:56.31 | ManxPower | Dougy, "that" is standard dialplan stuff you need to know before you do anything else. |
| 17:56.39 | ManxPower | Dougy, "core show application queue" |
| 17:56.46 | Dougy | oh neat |
| 17:56.47 | Dougy | sec |
| 17:57.22 | Dougy | flags |
| 17:57.52 | p3nguin | <PROTECTED> |
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| 17:58.49 | Dougy | heeey that worked |
| 17:58.54 | Dougy | exten => 7000,3,Queue(inqueue,n) |
| 17:59.00 | Dougy | then exten => 7000,4,VoiceMail(9999) |
| 17:59.02 | Dougy | success |
| 17:59.16 | ManxPower | I don't see the 60 second timeout on that |
| 17:59.34 | Dougy | it's in queues.conf |
| 17:59.35 | p3nguin | He used option n plus the timeout INSIDE the queue. |
| 17:59.36 | Dougy | timeout = |
| 17:59.46 | Dougy | p3nguin: is the way i did it a good way? or? |
| 17:59.49 | p3nguin | It's the wrong way, but it gave him the results he wanted. |
| 18:00.12 | Dougy | ok well, can you show me the appropriate method? is it just adding ,n,60 ? |
| 18:00.33 | p3nguin | <PROTECTED> |
| 18:00.36 | Dougy | yeah |
| 18:00.38 | Dougy | i see it |
| 18:00.43 | Dougy | so i need to use pipes then |
| 18:00.50 | p3nguin | commas should be fine. |
| 18:00.56 | Dougy | thats what i was asking hehe |
| 18:00.57 | Dougy | the pipes threw me off |
| 18:01.35 | p3nguin | Queue(inqueue,,,,60) |
| 18:02.11 | Dougy | testing |
| 18:02.20 | Dougy | is there a minimum value it can be set to? |
| 18:02.28 | p3nguin | 1, I presume |
| 18:02.34 | Dougy | exten => 7000,3,Queue(inqueue,,,,3) |
| 18:02.38 | Dougy | now at 37 seconds in |
| 18:02.40 | Dougy | still holding music |
| 18:02.53 | Dougy | oh |
| 18:02.56 | Dougy | do i have one too many commas |
| 18:03.50 | p3nguin | Did you run dialplan reload? |
| 18:04.33 | Dougy | i ran just 'reload' - i assumed (probably my mistake) that it'd reload * |
| 18:05.28 | p3nguin | If you change the dialplan, you should run dialplan reload after saving the changes. |
| 18:05.37 | Dougy | k, ran it |
| 18:05.53 | Dougy | still just keeps ringin |
| 18:06.37 | p3nguin | That doesn't make much sense... if you set a timeout for the application, it should exit and proceed with the dialplan. |
| 18:07.04 | Dougy | well, it didn't. hehe. :( :( |
| 18:07.09 | Dougy | i'll just use my 'improper' way |
| 18:07.56 | *** join/#asterisk ayrjola (~ayrjola@a85-156-216-139.elisa-laajakaista.fi) |
| 18:08.17 | Dougy | so i guess last thing i need to figure out is.. |
| 18:08.34 | Dougy | p3nguin: trixbox by default has the ability to dial *97 and go through voicemail etc |
| 18:08.46 | Dougy | i googled and cant find lines i'd need to add to have it available |
| 18:09.41 | p3nguin | exten => *97,1,VoiceMailMain(${CALLERID(num)}@default) |
| 18:09.41 | p3nguin | something similar, I assume, is what you're after. |
| 18:10.10 | Dougy | more then likely |
| 18:10.11 | Dougy | tries |
| 18:10.41 | Dougy | thats exactly it |
| 18:10.42 | Dougy | :-) |
| 18:10.44 | Dougy | thank you |
| 18:10.46 | p3nguin | You could also leave out the caller id part of that and you should be prompted for which mailbox you're looking for. |
| 18:11.36 | Dougy | exten => *97,1,VoiceMailMain() |
| 18:11.36 | Dougy | <PROTECTED> |
| 18:11.42 | Dougy | or leave the @default ? |
| 18:11.54 | p3nguin | You're supposed to include the vm context. |
| 18:12.12 | p3nguin | exten => *97,1,VoiceMailMain(@default) |
| 18:12.12 | Dougy | so (@default) ? |
| 18:12.46 | p3nguin | It'll say "Comedian mail... Mailbox?" |
| 18:13.13 | Dougy | that is exactly what it says |
| 18:13.17 | Dougy | if i leave it as VoiceMailMain() |
| 18:13.27 | p3nguin | If you use the caller ID method, it'll always take you to the mailbox matching the caller ID. |
| 18:13.36 | Dougy | yeah |
| 18:13.48 | Dougy | i'll leave *97 as direct, *98 as general |
| 18:13.50 | Dougy | thank you p3nguin :) |
| 18:14.13 | p3nguin | I prefer *86 (*VM). |
| 18:14.56 | Dougy | for.. general? |
| 18:14.57 | Dougy | or |
| 18:15.13 | p3nguin | to check voicemail. |
| 18:15.52 | Dougy | i have vmail.cgi set up |
| 18:15.53 | Dougy | also |
| 18:17.25 | Dougy | i think i am good to go :D |
| 18:18.19 | Dougy | p3nguin: can you set 2 different caller ID's for same extension? one for outgoing (for example to call my cell phone from pbx) and a different one for internal |
| 18:18.44 | Dougy | oh, duh |
| 18:19.24 | p3nguin | You're not going to have the same extension making calls to both of those locations independently. |
| 18:19.37 | Dougy | exten => _NXXXXXXXXX,1,Set(CALLERID(all)="Cerios" <7189067177>) |
| 18:19.37 | Dougy | exten => _XXXX/1000,1,Set(CALLERID(all)="Douglas Haber" <1000>) |
| 18:19.39 | Dougy | should work? |
| 18:20.06 | p3nguin | That's two different extensions. |
| 18:20.17 | Dougy | well, |
| 18:20.33 | p3nguin | Take off the quotes on the CID names, and you should be fine. |
| 18:20.34 | Dougy | I have exten => _XXXX,1,Dial(SIP/${EXTEN} also |
| 18:21.01 | p3nguin | You'll never be able to match both ways. |
| 18:21.07 | Dougy | how come |
| 18:21.25 | *** join/#asterisk digitalirony (~digitalir@my.ass.looks.just.like.your-face.info) |
| 18:21.32 | p3nguin | _XXXX/1000 says what to do if you match _XXXX from a caller ID of 1000. |
| 18:21.42 | Dougy | yeh |
| 18:21.50 | p3nguin | If if matches that, it sets the caller ID to what you set. |
| 18:21.56 | Dougy | i cant call extension 1001 if i set the _XXXX/1000 |
| 18:21.57 | Dougy | call fails |
| 18:21.59 | Dougy | if i remove it, it works |
| 18:22.14 | p3nguin | my point has been made. |
| 18:22.18 | Dougy | indeed |
| 18:22.25 | Dougy | so i should just give up on my thought then |
| 18:22.38 | p3nguin | You've got two priority 1's. |
| 18:22.46 | Dougy | oh, derr |
| 18:22.53 | Dougy | change the Dial line to 2, or n? |
| 18:23.32 | p3nguin | When you have two of the same priority, one with callerid match and one without, if you match the callerid, you can never use the other same-numbered priority. |
| 18:23.41 | Dougy | aha |
| 18:24.32 | Dougy | Got SIP response 405 "Method Not Allowed" back from 67.80.51.230 |
| 18:24.37 | Dougy | what does that mean, everything works but isee that |
| 18:24.38 | Dougy | googles |
| 18:25.58 | p3nguin | I use two different contexts for calls. One for outgoing, one for "internal" phones. The one for outgoing matches 7, 10, and 11-digit numbers, and sets the caller ID number to be sent outbound. internal numbers do not change the callerid, and the values from sip.conf are used. |
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| 19:19.18 | ManxPower | Long shot, but does anyone here have GIF/PNG/etc files of the icons that Polycom phones use for their buddy status? |
| 19:53.14 | *** join/#asterisk TehRabbitt (~Teh@pool-71-172-89-155.nwrknj.fios.verizon.net) |
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| 19:58.55 | TehRabbitt | where is a good place for a cheap IAX2 DID? |
| 20:00.14 | ManxPower | TehRabbitt, Is there a reason you want IAX2 rather than SIP? |
| 20:00.39 | ManxPower | IIRC both Teliax and Vitelity support IAX2, but I never use it. |
| 20:01.09 | TehRabbitt | ManxPower, bad experiances with DTMF tones and SIP here |
| 20:02.32 | drmessano | IAX2 is a bad move |
| 20:02.53 | TehRabbitt | hm |
| 20:03.16 | *** join/#asterisk Tim_Toady (~moi@77.49.122.124.dsl.dyn.forthnet.gr) |
| 20:03.21 | drmessano | No ITSP really supports IAX2 well.. IAX2 between asterisk boxes is great, but an ITSP's support of IAX2 is likely not going to be via the same controlled, up to date Asterisk version you would use to peer to another box |
| 20:04.00 | drmessano | e.g. Les.net uses an old 1.2 box for their IAX2 "trunks" |
| 20:06.41 | TehRabbitt | so what would you reccomend? |
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| 20:16.26 | ManxPower | Use SIP |
| 20:16.43 | ManxPower | I suspect all the "IAX2" supporting ISPs just use Asterisk to convert the call to SIP anyway. |
| 20:16.52 | ManxPower | TehRabbitt, what version of Asterisk? |
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| 20:55.57 | wuffi600 | hi. |
| 20:58.51 | Heldwin | hello, i am using asterisk recently, but I have a question. With it, I want all calls to my server and only redirecting defined one to my phone with a box ATA (complete filter to unknown call). is it possible ? |
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| 21:10.14 | wuffi600 | could you recomment a video-capable sip-client to use with asterisk? (i've already tested qutecom, x-lite, ekiga, linphone |
| 21:12.47 | ManxPower | wuffi600, the best one is the Polycom VVX 1500 |
| 21:13.09 | ManxPower | If I remember correctly, "1500" is both the model number and the retail price. |
| 21:19.22 | wuffi600 | ManxPower: :) ; i'm looking for a softphone, free of charge.. |
| 21:20.15 | ManxPower | wuffi600, And I want 10 million dollars. I think we both have about the same chance of getting what we want. |
| 21:20.20 | *** join/#asterisk p3nguin (bssd5fPY3x@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 21:20.31 | ManxPower | If the X-Lite/Egiga stuff does not work for you, you won't find anything better. |
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| 23:48.06 | Besticles | We got anybody in here that does some FastAGI with Asterisks? |
| 23:50.48 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 23:52.55 | Besticles | I am wondering why I am not getting a Result= response back from Asterisk after I send Exec Dial SIP/SERVER. I'm looking at documentation, and it appears that it should be. |
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| 23:53.44 | WIMPy | Buffers somewhere? |
| 23:54.08 | russellb | you're not going to get the result until the call is over |
| 23:54.17 | russellb | because the Dial() application runs as long as the two channels are up and talking |
| 23:54.34 | ManxPower | it has always been this way, and will always be this way. |
| 23:54.40 | russellb | and EXEC blocks until the application stops running |
| 23:54.47 | ManxPower | That is why I NEVER Dial from an AGI. |
| 23:55.19 | thehar | Dial(DAHDI/russelb,rt) |
| 23:55.21 | russellb | or just use a development paradigm that lets you do other things from your script while dial is running |
| 23:55.30 | *** join/#asterisk underdog (~whyareyou@abel.33ad.org) |
| 23:55.57 | ManxPower | russellb, like .call files? |
| 23:56.14 | underdog | hey guys...quick question...I've searched and searched (database and grep) for the blacklist values....where are those numbers stored? Thanks |
| 23:56.27 | russellb | hm? no, i just mean writing your AGI script in such a way that it can do other things while waiting on the result to come back if you ened to |
| 23:56.41 | russellb | underdog: in astdb somewhere ... |
| 23:56.44 | ManxPower | underdog, they are likely stored in the AstDB and not accessable other than via Asterisk |
| 23:56.57 | underdog | that makes sense |
| 23:57.06 | underdog | I dumped the Asterisk DB and nothing |
| 23:57.11 | underdog | greped the whole syste and nothimg |
| 23:57.44 | underdog | thanks russellb, ManxPower |
| 23:57.47 | russellb | np. |
| 23:57.52 | ManxPower | underdog, "asterisk db"? That stinks like FreePBX |
| 23:58.23 | underdog | just connected to it via mysql and dumped it to a text file and grep'd it |
| 23:58.34 | ManxPower | underdog, that is not AstDB |
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| 23:59.11 | underdog | yeah, I know now...I wasn't aware of an internal AstDB (which that's what it sounds like asterisk uses for core functions) |
| 23:59.24 | russellb | yup |
| 23:59.30 | ManxPower | underdog, you must be using some king of GUI |