| 00:08.37 | f2Knight | ClintGoudie-Nice, When using AsteriskRT, if you create a sip-account in the database, when that client connects it works. What I want to do is create a siptrunk from my AsteriskRT box to an existing account on another asterisk box., My Goal is to not have to have sip accounts in the sip.conf file at all.... |
| 00:08.52 | *** part/#asterisk wizard171 (~wizard171@h60.198.91.75.dynamic.ip.windstream.net) |
| 00:13.33 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 00:18.48 | nestAr | at this point, have the WiFi VoIP phones caught up? Or am I still better off with an ATA and a cordless phone |
| 00:18.59 | Maliuta | ata |
| 00:19.14 | Cyd | what did the homeless man get for christmas? |
| 00:19.25 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 00:20.08 | *** join/#asterisk xheliox (jeff@pdpc/supporter/student/xheliox) |
| 00:21.59 | nestAr | figured as much.. the ata+cordless solution served me well in the past.. |
| 00:24.52 | IsUp | nestAr: i have same solution |
| 00:25.01 | IsUp | have u ever used voicemail on your cordless phone? |
| 00:25.17 | IsUp | i cant receive Voicemail notifications (indicator on phone, led or icon) |
| 00:26.10 | nestAr | IsUp: it always worked for me with my grandstream |
| 00:26.34 | nestAr | the ht286.. i would get icon on the phone, blinking light at the base, and the duh duh duh dialtone on off-hook |
| 00:27.21 | nestAr | i was using a different extension for the ata than my desk phone (eg 1029 for desk, 1129 for ata), but in sip.conf, i put 1029 for the voicemail of 1129 |
| 00:27.37 | *** join/#asterisk path (~luis@190.196.69.196) |
| 00:29.54 | IsUp | i am using Alcatel dect phone, it doesnt give me dialtone |
| 00:30.05 | IsUp | i am typing number and pressing dial key |
| 00:31.21 | nestAr | not sure |
| 00:31.27 | nestAr | no experience with those units.. |
| 00:31.33 | nestAr | i was a bit more lo-tech |
| 00:32.15 | nestAr | anyone used the polycom kirk stuff? |
| 00:32.19 | nestAr | might be in my budget |
| 00:32.28 | nestAr | (if only i knew what my budget was) |
| 00:38.58 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:38.58 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 01:15.08 | f2Knight | Q: Does anyone have a working asterisk Realtime setup?? |
| 01:25.16 | leifmadsen | f2Knight: sure, I have several |
| 01:25.34 | leifmadsen | I've documented it here |
| 01:25.37 | leifmadsen | ~newbook |
| 01:25.38 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
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| 01:29.56 | f2Knight | leifmadsen, I am just starting to play with it, I already have several static boxes in the field, but I am playing with 1.8 - SVN and wanted to mess with realtime for a change. I got my IP-Phones registered and what not, I can call to the asterisk demo, but what I really want is to register to my own servers, but without using entries in the sip.conf, if that is possible. I would like to keep it in the database. My reason is that I w |
| 01:29.57 | f2Knight | ant to try and get a small group of systems to pull the information from a centeralized server. |
| 01:32.25 | socomm | is it true that asterisk does not scale very well |
| 01:33.15 | russellb | that's a pretty loaded statement ... |
| 01:33.37 | russellb | some aspects of scaling are not as easy as we would like them to be. However, Asterisk is being used successfully in extremely large environments. |
| 01:34.00 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 01:34.03 | russellb | I know of an instance with over 100,000 users and 400 servers, for example. |
| 01:34.35 | russellb | or another large company that has over 2000 remote locations, each that has an asterisk instance that connects back to a central large install |
| 01:34.36 | leifmadsen | russellb: I know of a particular government that is using it for an IVR system with several OC3's attached to it |
| 01:34.44 | russellb | yes, that too |
| 01:34.56 | leifmadsen | so yes, it scales |
| 01:35.05 | leifmadsen | and thus, your statement is untrue :) |
| 01:35.25 | leifmadsen | there are of course things we would prefer were easier to do |
| 01:36.23 | Kobaz | oh |
| 01:36.27 | Kobaz | leif |
| 01:36.39 | Kobaz | so this is what i've been slaving on the last week |
| 01:36.41 | Kobaz | http://pastebin.com/sfy0gDpD |
| 01:36.57 | russellb | and i know of another company that has over 3 million customers and routes over 1 billion minutes per month that usese asterisk as part of their infrastructure to provide PBX features |
| 01:37.36 | Kobaz | russellb: you'll probably be interested too |
| 01:37.43 | russellb | looks |
| 01:37.59 | Kobaz | so, you can link one channel to another via reference |
| 01:38.15 | Kobaz | so if you have channels that need to track each other across renames (transfers and etc), you can always get the current channel name |
| 01:38.24 | Kobaz | Set(TIE(foo)=SIP/123); |
| 01:38.38 | Kobaz | NoOp(${TIE(foo)}) |
| 01:38.46 | russellb | SIP/123-abc12323jadf ? |
| 01:38.51 | Kobaz | yeah |
| 01:38.55 | russellb | ok |
| 01:38.59 | Kobaz | well, it's not by prefix now |
| 01:39.02 | leifmadsen | Kobaz: like Bridge() I guess? |
| 01:39.13 | Kobaz | actually you would have to do Set(TIE(foo)=SIP/123-abc12323jadf)) |
| 01:39.15 | leifmadsen | I don't think I follow |
| 01:39.18 | Kobaz | but i can add prefix search too |
| 01:39.26 | Kobaz | you can track bridges, yes |
| 01:39.40 | Kobaz | any time a channel is masq'd the TIE will follow |
| 01:39.53 | russellb | neat code ... i think i'm going to need some use cases, though |
| 01:40.13 | Kobaz | say you have SIP/A calls SIP/B and SIP/C is calling Local/bar@baz which does some stuff and needs to know where SIP/A is |
| 01:40.30 | Kobaz | so on Local/bar@baz you set a tie to SIP/A-234234 |
| 01:40.31 | leifmadsen | ya I don't quite follow what it is used for :) |
| 01:40.45 | leifmadsen | tries to follow |
| 01:40.56 | Kobaz | so all of a suddon, SIP/A gets transfered |
| 01:41.14 | Kobaz | actually, it's more like |
| 01:41.28 | Kobaz | you want to monitor SIP/B's calls |
| 01:41.36 | Kobaz | SIP/B creates a new call, tranferss it to A |
| 01:41.46 | Kobaz | so SIP/B becomes SIP/A via masq on the attended transfer |
| 01:41.56 | Kobaz | anyway, i needed that type of thing so i wrote it |
| 01:42.03 | Kobaz | i think it's really cool |
| 01:42.09 | Kobaz | i can draw it out |
| 01:42.31 | leifmadsen | ya I'm going to suggest lots of documentation and use cases be submitted with the code :) |
| 01:42.38 | Kobaz | haha |
| 01:42.41 | leifmadsen | I can see how it could be useful, but it's not immediately obvious to me :) |
| 01:42.45 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 01:42.45 | Kobaz | yeah it's kind of crazy |
| 01:42.48 | leifmadsen | when I "get it" though I'll be on board ;) |
| 01:42.52 | Kobaz | took me a while to get it right in my head |
| 01:43.01 | Kobaz | it's about 200 lines less than it used to be |
| 01:43.13 | Kobaz | i'm putting in list locking so i can get rid of the big tie lock |
| 01:44.26 | Kobaz | here it is |
| 01:44.36 | Kobaz | http://pastebin.com/Aw2P5Xda |
| 01:44.53 | Kobaz | actually, there's more to it :P |
| 01:45.14 | russellb | still doesn't get it :-( |
| 01:45.56 | leifmadsen | indeed... |
| 01:46.16 | russellb | the "why" part, not just "what" is really important here (assuming you want to push this upstream) |
| 01:46.26 | russellb | because you're exposing some internal implementation details with this |
| 01:47.14 | russellb | anyway, the code looks neat, i just need to wrap my head around it |
| 01:47.50 | russellb | channel tracking, datastores, masquerades, ... that all makes up a party in my book, regardless of what it's doing :-p |
| 01:48.41 | Kobaz | hehe |
| 01:48.43 | Kobaz | haha |
| 01:48.48 | Kobaz | okay |
| 01:48.50 | Kobaz | this should explain it |
| 01:49.13 | Kobaz | aughh so slow, i need to move my office backups to after 10pm or something |
| 01:49.19 | Kobaz | http://pastebin.com/ven2dbHP |
| 01:49.36 | Kobaz | okay so, A calls B... B calls some other channel... B transfers that new call to A |
| 01:50.05 | Kobaz | meanwhile there's a local channel that got spawned along the way, that needs to know where the originating leg of SIP/B-123123 is |
| 01:50.50 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 01:50.54 | Kobaz | so before the transfer, it's SIP/B-123123 |
| 01:51.01 | Kobaz | after the transver it's SIP/A-213 |
| 01:51.14 | Kobaz | it's for call hijacking |
| 01:52.59 | Kobaz | basically, i have stuff external to asterisk (perl scripts) |
| 01:53.13 | Kobaz | that need to be involved in a masq... it needs to know the new channel name |
| 01:53.16 | Kobaz | (in dialplan) |
| 01:53.27 | Kobaz | i can't store the channel name, because it changes |
| 01:53.59 | Kobaz | i can't use my group variables, because again, the name changes, and in dialplan SIP/B-123123 doesn't know when it's transfered |
| 01:55.03 | *** join/#asterisk shapr (~shapr@nat/digium/x-hfhdvzpywjkliscz) |
| 01:55.09 | shapr | Good Morning #asterisk! |
| 01:55.13 | leifmadsen | evening :) |
| 01:55.29 | shapr | godmorgon leifmadsen! |
| 01:55.34 | Kobaz | leifmadsen: follow yet? :) |
| 01:56.00 | leifmadsen | Kobaz: lol not at all :) |
| 01:56.05 | leifmadsen | I'm looking at it all, and it somewhat makes sense |
| 01:56.11 | Kobaz | and i can't use AMI to get masq events, because the stuff i need to happen needs to happen very very fast |
| 01:56.29 | leifmadsen | looking at the pastebin since it has channel names |
| 01:56.56 | Kobaz | local/abc ties to SIP/B-123123 which has a tieback to local/abc, so it can inform it of updates |
| 01:58.43 | Kobaz | i have it all clear in my head, i don't quite know how to write it out yet |
| 01:58.54 | leifmadsen | ok, so when you do Set(TIE(foo)=${CHANNEL}) then later do NoOp(${TIE(foo)}), it is saying SIP/A-213 because that's where SIP/B-12345 (${CHANNEL}) is now associated with? |
| 01:59.16 | Kobaz | yeah |
| 01:59.22 | leifmadsen | I can already tell this is going to require a blog post, wiki page, and some graphics with arrows and such :) |
| 01:59.47 | leifmadsen | (and SIP/B-12345 is now gone because that was the channel transferred and masqueraded) |
| 01:59.54 | leifmadsen | but you need to know where the called party is |
| 01:59.56 | Kobaz | the 27, 8x10 color glossy pictures, with the circles and arrows, and a paragraph on the back of each one, explaining what each one was... |
| 02:00.00 | Kobaz | yeah |
| 02:00.05 | leifmadsen | yes :) |
| 02:00.08 | Kobaz | i don;t think it really exposes implementaion details |
| 02:00.12 | leifmadsen | when I get it I'll be able to draw it |
| 02:00.14 | Kobaz | because the channel gets renamed |
| 02:00.27 | Kobaz | that happens, it's visible in "userland" |
| 02:00.37 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 02:00.37 | shapr | I'm glad Arlo Guthrie references are still in style. |
| 02:00.39 | leifmadsen | what I'm picturing in my head is associating a "person" with a "channel" no matter where they get transferred |
| 02:00.45 | Kobaz | it's not some internal thingy, you can see it rename from ami and core show channels and such |
| 02:00.48 | Kobaz | yes |
| 02:00.51 | Kobaz | precisely |
| 02:01.04 | Kobaz | i drew it out on the board here as channels are just boxes |
| 02:01.16 | Kobaz | and the person (the "call") is inside the box |
| 02:01.17 | leifmadsen | "Bob was SIP/B-321, but he got transferred and now Bob is associated with SIP/A-123. Oh wait, Bob was transferred again and is now talking with SIP/C-456" |
| 02:01.22 | Kobaz | and you just throw the box away, and keep the person |
| 02:01.37 | Kobaz | yeah |
| 02:01.44 | Kobaz | and he could get transfered around 20 different ways |
| 02:01.46 | leifmadsen | gotcha -- pictures are going to be very useful here :) |
| 02:01.48 | leifmadsen | indeed |
| 02:01.53 | Kobaz | you'll always have an exact up to date instant reference to the channel |
| 02:01.58 | leifmadsen | he could get transferred into a conference room for example |
| 02:02.02 | Kobaz | yeah |
| 02:02.06 | leifmadsen | or Bridge()'d |
| 02:02.18 | Kobaz | and i dont want to listen for events on ami and then pipe them into my dialplan |
| 02:02.23 | leifmadsen | gotcha |
| 02:02.31 | Kobaz | i want it all right in dialplan, because the dialplan that;s running, has other stuff to do in the meantime |
| 02:02.51 | Kobaz | i think asterisk is missing a lot of these introspection type featurres |
| 02:03.15 | Kobaz | this one module removed like 200 lines of external code to track channels (which was buggy anyway) |
| 02:03.26 | Kobaz | transfer tracking is one of the banes of any asterisk user |
| 02:03.32 | kuku | Any reason why an inbound call would go to fast busy after two rings when connected to a TDM400p, even when asterisk is not running ? |
| 02:03.57 | Kobaz | from within dialplan i want to be able to get anything and everything about the system and any channel |
| 02:04.08 | Kobaz | right now there's some stuff for that, but not enough i think |
| 02:05.35 | Kobaz | leifmadsen: russellb: i was talking to a guy yesterday on -dev... and he was having some call stealing trouble, and one of my thoughts, before i found a different way. was hey, this guy could use TIE |
| 02:06.05 | leifmadsen | :) |
| 02:06.16 | Kobaz | he wasn't tracking calls through transfers, so he didn't need an up to date channel ref, but if he did want his app to survive transfers, it would be perfect |
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| 02:07.25 | leifmadsen | well I could see how this might be useful. It's certainly an edge case but with some good examples could make it more useful |
| 02:07.39 | leifmadsen | is so happy KVM is finally working |
| 02:07.49 | Kobaz | yeah |
| 02:07.54 | Kobaz | this is a very specific thing |
| 02:08.01 | Kobaz | it's not like, everyone is going to be using it tomorrow |
| 02:08.11 | leifmadsen | :) |
| 02:08.15 | leifmadsen | with good examples it could be useful |
| 02:08.17 | Kobaz | maybe they will be |
| 02:08.18 | Kobaz | yeah |
| 02:08.25 | leifmadsen | I could see some advantages to maybe pairing it up with CEL |
| 02:08.30 | leifmadsen | or even CDR() |
| 02:08.34 | Kobaz | good example is if SIP/A calls SIP/B but does an Originate right before calling |
| 02:08.51 | leifmadsen | I see it as potentially useful with CEL and tracking call progress |
| 02:08.56 | Kobaz | and now you have this "unrelated" channel floating in space that needs to do stuff on SIP/A |
| 02:09.22 | Kobaz | yeah |
| 02:09.29 | Kobaz | cel could use a lot of improvements |
| 02:09.42 | Kobaz | i have my own version of CEL in perl, i call it CAL |
| 02:09.50 | Kobaz | channel action logging, heh |
| 02:10.05 | Kobaz | maybe when i move to 1.8, i can link my cal to the cel |
| 02:10.29 | Kobaz | but my stuff tracks transfers on a per-call basis |
| 02:10.44 | Kobaz | follow the history in one series of records tied to one unique id |
| 02:15.15 | russellb | sounds like the linkedid in CEL |
| 02:16.03 | Kobaz | yeah, this created was before cel |
| 02:16.59 | Kobaz | i like the one id versus linking personally, but you can generate one from the other |
| 02:17.18 | Kobaz | oh man, did i really botch that sentence up that bad |
| 02:21.02 | Kobaz | and i added lots of console logging |
| 02:21.10 | Kobaz | i should port that to trunk, it's really cool |
| 02:21.32 | Kobaz | well, not really more, but extra data to existing logging to make it more useful |
| 02:21.48 | leifmadsen | you seem tired :) |
| 02:21.56 | Kobaz | hah |
| 02:21.59 | Kobaz | you have no idea |
| 02:22.16 | Kobaz | last week, up at 9am, sleep at 3am, all week |
| 02:22.23 | leifmadsen | been there done that :) |
| 02:22.31 | leifmadsen | well peas out homey |
| 02:22.37 | Kobaz | yeah |
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| 02:40.22 | *** join/#asterisk inluck (~inluck@142.162.125.31) |
| 02:40.34 | inluck | Anybody around? |
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| 02:41.38 | drmessano | Yes, was that your question? |
| 02:42.06 | inluck | Not at all :b I'm trying to figure something out in regards to voicemail. |
| 02:42.22 | inluck | I'm in the process of setting up two redundant asterisk switches, which I have completed |
| 02:42.34 | inluck | but I need to make a change to how voicemail is handled on incoming calls |
| 02:42.49 | inluck | I need to pbx to drop the call if the extention isn't registered |
| 02:43.04 | inluck | instead of forwarding to voicemail |
| 02:43.27 | inluck | would you happen to know where I could find some documentation on this. |
| 02:43.41 | inluck | I've been searching google and haven't found the right set of keywords I guess |
| 02:48.12 | inluck | no suggestions? |
| 02:58.10 | shapr | inluck: How does the call reach that particular switch if the extension is not registered to that switch? |
| 03:00.17 | inluck | I get my inbound DIDs from DIDWW, which will allow me setup a ring or hunt group to a selection of sip address |
| 03:00.45 | inluck | so if the same extention is on two pbxes |
| 03:00.55 | inluck | using DNS SRV records, the ata will go look to the other switch |
| 03:01.01 | inluck | if the main one goes offline |
| 03:01.23 | inluck | I need to have the pbx hangup on incoming calls to an extention if it isn't registered |
| 03:02.39 | inluck | if I can't do it that way |
| 03:03.19 | inluck | then I can setup an external server to monitor both pbx and update DIDWW's forwards manually when it detects an outage |
| 03:04.33 | inluck | or if I could make it ring anyways |
| 03:04.39 | inluck | as if it was registered |
| 03:04.53 | inluck | that could also work |
| 03:07.21 | shapr | I'm confused, when you say "if extension is not registered" do you mean the extension is currently logged in? or that it has an entry in sip.conf? or what? |
| 03:08.03 | inluck | by registered, I mean that if I did a "sip show peers" the extention shows as registered with an IP address , the endpoint is online and functioning |
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| 04:51.55 | ScarEye | hello everyone, is it possible to use fring to connect to my asterisk box at home that's connected to Google Voice to make calls? If so does anyone have a link they can point me to? |
| 04:53.44 | shapr | ScarEye: I don't know about fring, but Asterisk 1.8 supports google voice calls. |
| 04:53.54 | shapr | er, outgoing at least... not sure about incoming calls. |
| 04:54.36 | ScarEye | outgoing is all I am looking for |
| 04:55.29 | ScarEye | like, HTC phone with Fring that connects to my asterisk box at home which has a GV account to make outgoing calls. |
| 04:55.35 | ScarEye | Don't care about incomming. |
| 05:02.38 | ScarEye | shapr: Any howto's: you know of? |
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| 06:36.25 | Bloudermilk | Hey all. I know about make samples to generate all the sample config files. Is there a minimalistic version of that to only generate enough config for asterisk to run |
| 06:36.26 | Bloudermilk | ? |
| 06:37.59 | ChannelZ | not that I'm aware of |
| 06:38.33 | kaldemar | asterisk should run with the samples but yes, they shouldn't be used. and no, there's no such command. you must edit the samples by hand or start from scratch. |
| 06:44.59 | Bloudermilk | Yikes |
| 06:45.28 | Bloudermilk | Are there any key things to remove from the samples? Or maybe a guide to use when starting from scratch? |
| 06:45.54 | Bloudermilk | It just occurred to me that there's a good chance the samples have enabled way more services than I need |
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| 06:59.53 | kaldemar | Bloudermilk: well, no guide that i'd know of. modules.conf is handy in the way that you can disable whole modules with it if you don't need them. |
| 07:01.11 | Bloudermilk | kaldemar: I'll take a look at the comments in that file and see what I can do. Thanks |
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| 07:10.08 | kaldemar | other than that, the sample files are full of useful comments. |
| 07:10.34 | kaldemar | does not appreciate the post-1.8.0 state of documentation in the tarballs |
| 07:10.58 | Bloudermilk | Sweet |
| 07:11.06 | Bloudermilk | I'll just read through all 50 of them |
| 07:11.06 | Bloudermilk | hahah |
| 07:11.53 | kaldemar | there is a huge PDF file under doc/ in the source tree that has all kinds of documentation, in 1.8.0 and earlier there were separate text files. |
| 07:11.59 | kaldemar | other that that: |
| 07:12.03 | kaldemar | ~newbook |
| 07:12.03 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 07:12.14 | kaldemar | ~docs |
| 07:12.15 | infobot | Asterisk documentation is available at http://wiki.asterisk.org (Official Asterisk Documentation Wiki), the Voip-Info wiki at http://voip-info.org (~voip-info) or Asterisk: The Future of Telephony (~book) |
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| 07:13.00 | kaldemar | the official wiki is sloooow. |
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| 07:14.08 | Bloudermilk | I've always used voip-info, but it's notoriously inaccurate and/or dated |
| 07:14.17 | Bloudermilk | I'll check those other resources out |
| 07:14.18 | Bloudermilk | Thanks |
| 07:16.03 | hobodave | kaldemar: that's Confluence for ya |
| 07:17.52 | kaldemar | hobodave: yeah, i know. arent the AST.pdf and AST.txt generated from the confluence wiki? |
| 07:18.28 | kaldemar | yes they are. |
| 07:18.32 | hobodave | kaldemar: I am not the person to ask :) I just noticed that it was running Confluence |
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| 07:18.48 | kaldemar | wonders what AST.txt is supposed to be read with |
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| 07:22.30 | Bloudermilk | I don't suppose there are any cloud based centos users here... The TTY9 problem is biting me in the ass |
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| 07:56.46 | hobodave | damn |
| 07:57.02 | hobodave | I just upgraded from 1.6.2 to 1.8 and it went well except for res_snmp |
| 07:57.27 | hobodave | I can't view any of my asterisk info via snmp, and I updated the MIBs based on the latest ont he wiki |
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| 08:03.40 | hobodave | fixed it |
| 08:03.50 | hobodave | permissions on /var/agentx/ got jacked |
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| 08:31.31 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 08:31.33 | schmidts | good morning |
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| 08:54.25 | Bloudermilk | Hey all. If i've set up an IAX friend "foo", should Dial(IAX2/1234@foo) work? |
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| 08:55.02 | WIMPy | Not sure, but iax2/foo/1234 will. |
| 08:55.20 | Bloudermilk | WIMPy: Where 1234 is the extension? |
| 08:55.27 | WIMPy | yes |
| 08:56.03 | Bloudermilk | WIMPy: thanks |
| 08:57.03 | Bloudermilk | Gah blast, what is "Call rejected by 10.179.41.215: No authority found"? |
| 08:57.23 | kaldemar | Bloudermilk: afaik, with 1234@foo asterisk will assume that foo is a host, not a peer in iax.conf. |
| 08:57.49 | WIMPy | thinks so as well, but I'm not sure. |
| 08:57.58 | kaldemar | and 1234 is the username. |
| 08:58.27 | kaldemar | i'm sure now, just checked. |
| 08:59.04 | Bloudermilk | I see |
| 08:59.39 | Bloudermilk | So with an IAX2 friend "foo", and extension 1234, I should use Dial(IAX2/foo/1234) ? |
| 09:00.44 | WIMPy | yes |
| 09:01.33 | WIMPy | For peers configured in iax2.conf (or sip.onf) use iax2/peer/ext. |
| 09:03.29 | schmidts | sorry for offtopic but : http://wearefuntastic.net/imageserver/_ihroidnhotlinka/img/radadararinjeaection.jpg ROFL!!!! |
| 09:03.57 | Tozz_ | so old ;) |
| 09:04.26 | schmidts | but still good ;) i never see this before so sorry |
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| 09:13.49 | WIMPy | Are you shure you want to name your child ');drop databse;'? |
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| 09:29.03 | schmidts | wimpy thats exactly what i thought when i read the above one :D |
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| 09:30.16 | schmidts | http://xkcd.com/327/ |
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| 09:51.40 | littleball | hello, |
| 09:51.56 | littleball | anyone want to try IAX mobile software phone ? |
| 09:52.28 | littleball | for android and blackberry |
| 09:52.43 | Tozz_ | no thanks! |
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| 09:58.36 | joachim_- | Good day! Sun is shining :) |
| 09:58.58 | joachim_- | anyone know how to get asterisk to send out SNMPTRAPS? |
| 09:59.36 | tzafrir | schmidts, wow, I just searched for "bobby tables", and it seems that this name has become popular |
| 10:01.04 | Tozz_ | joachim_-: u could use AGI scripts |
| 10:01.20 | littleball | for android phone users, you can download mobile IAX2 softphone from android market by searching 'MPhoneGG'. |
| 10:01.37 | littleball | you can connect to your own asterisk server |
| 10:01.57 | schmidts | tzafrir me too :D |
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| 10:27.27 | schmidts | Does someone use Cisco SPA 5xx phones? Do you know if there exists an IPV6 Firmware for them? |
| 10:33.28 | *** join/#asterisk mta59066 (~mehmet@92.44.155.133) |
| 10:34.47 | mta59066 | hello, anybody know how do I specify what format sound files will be used? Right now it uses gsm even though I have both gsm and ulaw. Switching to a different language doesn't work because that language only has ulaw. |
| 10:35.51 | kaldemar | mta59066: by the used codec |
| 10:38.09 | mta59066 | kaldemar: ok I understand, if I am connecting with gsm than it uses the gsm sound files |
| 10:38.32 | mta59066 | kaldemar: thanks |
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| 12:31.22 | orn | quit |
| 12:33.50 | orn | support: never mind, figured it out |
| 12:39.40 | Dovid | morning all |
| 12:41.49 | shapr | howdy |
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| 12:54.35 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:54.35 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:54.56 | leifmadsen | sip:polycom@leifmadsen.com |
| 12:56.54 | xheliox | hmm? |
| 12:56.58 | leifmadsen | yes. |
| 13:00.56 | schmidts | leifmadsen you know this is quite old: Asterisk PBX 1.6.2.8 :D |
| 13:00.58 | *** join/#asterisk Thorn (~Thorn@unaffiliated/thorn) |
| 13:01.38 | Thorn | hello |
| 13:02.42 | Thorn | will asterisk complain if it can't parse the dtmf caller id packet? I have about 20% incoming calls without caller id, there're no error messages |
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| 13:24.34 | Benwa | hi, is there something else than free-pbx for asterisk with debian ? i mea an other web interface for users ? |
| 13:24.56 | mzb | asterisk-gui? |
| 13:25.01 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:25.17 | mzb | not sure if it does users in the same way |
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| 13:28.08 | Benwa | mzb: more 'user-friendly' than freepbx ? |
| 13:28.38 | mzb | no idea |
| 13:28.53 | mzb | long time since I tried it |
| 13:29.04 | Benwa | k thanks |
| 13:29.06 | mzb | which version of FReePBX are you talking about? |
| 13:29.29 | mzb | and are you talking about using or installing (on Debian) ? |
| 13:29.54 | Benwa | 2.5.2.3 |
| 13:30.05 | mzb | 2.8 is a lot easier to understand |
| 13:30.09 | Benwa | well both |
| 13:30.20 | Benwa | mzb: ok, i will try it |
| 13:30.20 | mzb | even more so if you don't install all the modules |
| 13:30.47 | mzb | fwiw I have a script that install dahdi+asterisk+freepbx on debian |
| 13:30.58 | mzb | (latest stable versions of all) |
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| 13:31.26 | Benwa | mzb: oh nice, can i have a look at it ? |
| 13:31.40 | Benwa | debian squeeze ? |
| 13:31.53 | mzb | any debian ... it automatically does dependencies |
| 13:31.56 | mzb | BUT |
| 13:32.13 | mzb | if you already have a database on that machine you'll have to trick it |
| 13:32.36 | Benwa | i've got a fresh squeeze install |
| 13:32.42 | Benwa | sooo np |
| 13:32.49 | mzb | ok |
| 13:32.51 | mzb | hang on |
| 13:33.12 | Benwa | yep :) |
| 13:34.54 | mzb | read this while I upload it (the download link isn't right yet) : http://www.voipcoop.org/viewforum.php?f=30 |
| 13:36.41 | *** join/#asterisk neurosys (~neurosys@50.20.65.126) |
| 13:37.36 | mzb | Benwa: refresh the page .. link should work |
| 13:38.05 | mzb | keep in mind it's very young |
| 13:38.29 | mzb | if it fails the first run I'd appreciate the log |
| 13:38.37 | mzb | then just run it again |
| 13:38.47 | mzb | select "EASY" |
| 13:38.57 | mzb | and then choose a password |
| 13:39.09 | Benwa | ok |
| 13:39.17 | mzb | go and make yourself a coffee/whatever ... |
| 13:39.24 | Benwa | :) |
| 13:39.40 | Benwa | squeeze almost installed ... few minutes left :) |
| 13:39.48 | mzb | it's getting there ... and when I say "EASY" I'm not joking ;) |
| 13:40.05 | Benwa | yes i see |
| 13:40.09 | Benwa | astersik 1.6 ? |
| 13:40.14 | mzb | nope |
| 13:40.16 | mzb | 1.8 |
| 13:40.27 | Benwa | from svn ? |
| 13:40.31 | mzb | you *can* choose 1.6 in intermediate, but it doesn't work |
| 13:40.41 | mzb | you can choose 1.8 from svn |
| 13:40.48 | mzb | ie 1.8 branch |
| 13:41.00 | mzb | which is handy if you want to keep up with fixes |
| 13:41.13 | mzb | default is to build from tarballs |
| 13:41.39 | Benwa | thanks a lot |
| 13:41.55 | mzb | if you choose advanced you can select svn ... but I'm almost sure that anything other than 1.8 + 2.8 won't work |
| 13:41.57 | mzb | np |
| 13:42.16 | mzb | minimal modules are loaded by default, btw |
| 13:42.47 | mzb | so the interface is very clean (and error free!) when you first see it |
| 13:42.58 | Benwa | :) |
| 13:43.34 | mzb | feedback appreciated |
| 13:44.37 | *** join/#asterisk neurosys (~neurosys@50.20.65.126) |
| 13:44.55 | Benwa | sure !! :) |
| 13:46.29 | Benwa | is asterisk 1.8 works with the last squeeze kernel ? cause i remember i had to compile a new kernel for 1.8 a few months ago |
| 13:46.57 | mzb | I haven't found any kernel-related problems yet |
| 13:47.08 | Benwa | mmmh, no, i remember now, i had a special kernel, sorry ... |
| 13:47.23 | mzb | however freedoh will halt if a kernel upgrade is pending |
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| 13:51.32 | mzb | Benwa: almost 1am here so time for bed, but my irc is always on ... pls let me know how you go |
| 13:52.17 | mzb | gnite |
| 13:54.20 | Benwa | mzb: good night (3PM here :) ) |
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| 14:27.29 | *** join/#asterisk Flashtek (~Flashtek@irc6.flashtek-uk.com) |
| 14:28.40 | Flashtek | I have a quick question.. if I want to simply download a tarball, unpack, ./configure and make, how could i avoid the menuconfig to enable specific modules etc..?? |
| 14:32.55 | leifmadsen | Flashtek: just don't run make menuselect |
| 14:33.12 | Flashtek | but if i don't, I done get things like mysql_cdr |
| 14:33.16 | leifmadsen | right |
| 14:33.31 | leifmadsen | well you shouldn't get cdr_mysql since that is in addons |
| 14:33.38 | leifmadsen | and none of the addons are selected by default |
| 14:33.46 | Flashtek | it's included in the 1.8 tarball... |
| 14:33.53 | leifmadsen | it's available in the 1.8 tarball |
| 14:34.02 | Flashtek | sure sure.. |
| 14:34.04 | leifmadsen | addons, afaik, are not enabled by default |
| 14:34.21 | Flashtek | ok.. so how can i enable them without needing menuconfig ? |
| 14:34.39 | Flashtek | as in, automate build |
| 14:35.25 | leifmadsen | hold on I have that documented somewhere |
| 14:35.28 | Flashtek | kk |
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| 14:35.30 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:36.41 | leifmadsen | Flashtek: http://ofps.oreilly.com/titles/9780596517342/ch03.html#Installing_id292994 <-- Scripting menuselect |
| 14:36.48 | *** join/#asterisk fofware (~fofware@wdctf.siup.gov.ar) |
| 14:36.48 | leifmadsen | I knew I documented it |
| 14:37.47 | Flashtek | leifmadsen: cheers |
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| 14:53.36 | *** join/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 14:54.13 | asteriskmonkey | if asterisk has a default of 3 seconds for a recording, why would there be 1 and 2 second recordings in a db? |
| 14:58.04 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 15:02.53 | leifmadsen | asteriskmonkey: someone hung up early? |
| 15:03.33 | asteriskmonkey | i think i see my problem.. i had in my head that there was a built in min message length of 3 for some reason.. it looks like its 0 by stock instead :P |
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| 15:23.42 | *** join/#asterisk StaRetji (~BigAll@91.143.222.166) |
| 15:25.11 | StaRetji | folks, every few restarts of pc asterisks doesn't recognize keystrokes from my dahdi interfaces (ordinary phone connected). There is nothing show in asterisk CLI. How to debug? What could it be |
| 15:25.13 | StaRetji | > |
| 15:25.16 | StaRetji | ? |
| 15:29.08 | *** join/#asterisk tash (~Tommy@66.64.111.226) |
| 15:30.08 | tash | I'm using cepstral's TTS engine with Asterisk. I'd like to enter some pauses using ssml for every space in a sentence. Does anyone in here have experience doing that? |
| 15:31.29 | *** part/#asterisk porche (~kursad@91.188.208.204) |
| 15:32.44 | StaRetji | maaaan, what is with this irc channel, I mean wtf, nobody never answers |
| 15:32.50 | StaRetji | baah.... |
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| 15:35.58 | SuPrSluG | StaRetji: have you turned on a verbose level? default is 0 |
| 15:37.26 | StaRetji | SuPrSluG: thx for reply, yes, it's set to 15 |
| 15:37.49 | tash | StaRetji: I feel your pain, lol...but I think a lot of people are busy and/or just stay logged in all the time |
| 15:38.01 | *** part/#asterisk Flashtek (~Flashtek@irc6.flashtek-uk.com) |
| 15:38.24 | StaRetji | it working fine, then after restart it will not, then I restart several times and tones got recognized |
| 15:38.34 | *** join/#asterisk anil-lim (~chatzilla@122.162.142.232) |
| 15:38.34 | StaRetji | tash: hehe, sorry for that comment |
| 15:38.54 | *** join/#asterisk wizard171 (~wizard171@h60.198.91.75.dynamic.ip.windstream.net) |
| 15:39.02 | tash | no apologies necessary ... I often feel the same way, but yeah ... people aren't always watching their IRC client |
| 15:39.26 | SuPrSluG | sound like a dtmf issue? |
| 15:39.26 | *** join/#asterisk Defraz (~Defraz@63.226.95.152) |
| 15:39.45 | StaRetji | SuPrSluG: yes, i tried googling, nothing comes up |
| 15:39.52 | StaRetji | at least nothing I could apply |
| 15:40.08 | StaRetji | one link says about recompiling dahdi-tools, which I did |
| 15:40.08 | n3hxs | Then there are those like me that don't know and don't respond. |
| 15:40.29 | tash | n3hxs: true true ... and I'm in that boat 99% of the time :P |
| 15:40.47 | StaRetji | hehe, folks, sorry, maaan, I regret i wrote that lol |
| 15:42.29 | n3hxs | You were not the first and won't be the last. ;) |
| 15:42.31 | tash | for real, no worries |
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| 15:44.40 | StaRetji | For example, tones are not recognized at the moment. Then I reboot, check and do that several times until it starts working. So I thought if I can fix it without rebooting, then I guess it will work on next reboot |
| 15:50.02 | tash | StaRetji: can you state the problem again? I joined the room after you stated it I think... |
| 15:50.42 | StaRetji | tash: Every few restarts of pc asterisks doesn't recognize keystrokes from my dahdi interfaces (ordinary phone connected). There is nothing show in asterisk CLI. |
| 15:50.47 | StaRetji | core set verbose 15 |
| 15:51.01 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:51.34 | tash | when it is working, do you see the DTMF digits in the asterisk CLI? |
| 15:51.40 | StaRetji | <PROTECTED> |
| 15:51.49 | tash | I can't remember if the CLI shows RTP/DTMF or not |
| 15:51.54 | StaRetji | that would be it, even though I called 600 for echo |
| 15:52.18 | tash | I always use wireshark which shows the RTP packets (then again, I'm using VOIP, not POTS) |
| 15:52.35 | tash | and my calls are outbound to phones not on our network |
| 15:53.27 | StaRetji | weird thing it worked. I rebooted, now it doesn't work. Let me try reboot several times again to see if it will work |
| 15:57.21 | tash | I'm thinking the reboot might just be coincendetal to it working |
| 15:57.23 | tash | I could be wrong |
| 15:57.33 | tash | I think you need to find the underlying issue as opposed to a reboot ... |
| 15:57.46 | tash | are you placing calls outbound? who is your provider? etc etc etc? |
| 15:58.10 | StaRetji | well, it's possible, I changed conf files so many times, copy them back that I'm not sure anymore |
| 15:58.43 | StaRetji | yes, I call another sip server, but I have problem calling from that sip server back, |
| 15:58.54 | StaRetji | it's local calls |
| 16:03.15 | tash | if it were me, I'd install tshark on my Asterisk server (tshark is a packet capturing tool) ... start tshark while calls are going through ... when done testing open the .pcap file on a windows box using Wireshark ... look at telphony calls, etc etc etc ... you can see whether DTMF is passed or not. But that is way too much stuff to talk about in IRC |
| 16:04.09 | nestAr | anyone used the polycom kirk stuff? |
| 16:04.32 | *** join/#asterisk madsara (~madsara@rtr0.circleeight.net) |
| 16:05.06 | madsara | Hey, you guys know if it's possible to disable a dadhi channel without issuing a dahdi restart? Seems that the restart drops calls. |
| 16:05.36 | madsara | I need to be able to (hopefully) add and remove channel 2 from group 5 on the fly. |
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| 16:13.27 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
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| 16:17.53 | JerJer | madsara: have not tried it in a while, but the asterisk config should be able to be reloaded without restarting |
| 16:18.36 | madsara | JerJer: I can do it via a "dahdi restart", which keeps non dahdi channels bridged, but drops anything dahdi realted. |
| 16:19.23 | JerJer | module reload chan_who's your dahdi.so ? |
| 16:19.24 | madsara | Oh, possibly config reload /etc/asterisk... |
| 16:19.50 | JerJer | :) |
| 16:20.31 | madsara | Oh ,yeah, looks like that works. |
| 16:21.12 | JerJer | i presume dahdi restart is more of a brute force thing |
| 16:21.56 | JerJer | not to be confused with dahdi sit on lap :) |
| 16:23.36 | StaRetji | tash: thx for the tip. It sound like a too much for me, but will consider it. Cheers... |
| 16:28.51 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 16:35.09 | madsara | JerJer: It definitely is more brutal. |
| 16:35.59 | madsara | It appears (although I should test this again), that if I have a channel commented out upon "dadhi restart" I can't re-enable it via "module reload", I have to issue a restart, after which time I can toggle it on and off via the config and a module reload. |
| 16:36.14 | madsara | Again, I should test that a few more times, but it's a production environment. |
| 16:40.24 | *** join/#asterisk Andrew_Majosi (6274247f@gateway/web/freenode/ip.98.116.36.127) |
| 16:42.53 | Andrew_Majosi | Q: No audio with Twinkle nor Blink clients, but connect fine from Internet. Everything works locally on local network. All ports are open on both sides (I think). What should I check for? |
| 16:44.01 | Andrew_Majosi | I use TLS for blink, no VPN. |
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| 16:57.05 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 16:57.07 | krion | hi |
| 16:58.08 | krion | i can't manage to get colored prompt with asterisk -r |
| 16:58.33 | krion | is the cli == logger ? |
| 16:59.05 | krion | because i don't get timestamp too (sorry for the low level of my question no daddi no zaptel e1 or ivr stuff :)) |
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| 17:00.14 | wizard171 | krion, all that stuff is usually in "asterisk.conf" ... |
| 17:02.47 | krion | looks like it depend on colored shell, but my shell is colored |
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| 17:19.10 | *** join/#asterisk proute (~AnthonyCB@mail.sysun-technologies.com) |
| 17:19.13 | proute | Hello |
| 17:20.04 | proute | I use asterisk 1.6.2.16 and sometimes, Asterisk crash (sip connection fail for all extensions) but I can connect on cli. I found nothing in my logs. do you have any idea about this problem? |
| 17:21.42 | proute | It seems that sip freeze but why....? |
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| 17:34.19 | KamikazeMicrowav | proute, are your sip phones registered? Can you see them with "sip show peers" on cli? |
| 17:34.24 | draeath | Ideally, what kind of kernel should asterisk be run on? (tickless, particular tick rate, realtime-patched, etc) |
| 17:34.59 | proute | I think found the solution with asterisk 1.6.2.16.1 (stack buffer overflow in SIP channel driver) |
| 17:35.21 | Qwell | a buffer overflow wouldn't cause a hang |
| 17:36.17 | proute | in cli, sip phone are registered |
| 17:36.29 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:36.35 | proute | but sip phone can't make a call |
| 17:36.43 | proute | I must restart asterisk |
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| 17:58.36 | brainiac | I have a TDM2400P with echo and static on it. It's usually heard by the maker of outgoing calls, but sometimes also by the callee. Is there some kind of diagnostic I can perform to help me know what's wrong? |
| 17:59.02 | citywok | your card is broken. i have two i'd be happy to sell you :P |
| 17:59.26 | brainiac | does the entire card have to be replaced? |
| 17:59.39 | citywok | i'm kidding about it being broken (although it is possible i suppose) |
| 17:59.53 | _Corey_ | brainiac: If still in warranty, I would seriously call Digium |
| 18:00.13 | citywok | does it have the echo cancellation module? |
| 18:00.28 | brainiac | Thanks. This happened once before, what could cause this? |
| 18:00.48 | brainiac | I don't know if the e/c module is enabled... I'll check |
| 18:00.57 | citywok | we had bad wiring on one of ours and it drove us nuts for a month with all kinds of cross talk / echo |
| 18:01.09 | citywok | eventually we repunched both cards from scratch and that helped |
| 18:02.41 | brainiac | citywok: do you mean you repunched the block? |
| 18:03.32 | citywok | yea where the tdm was terminated |
| 18:03.49 | citywok | where the amp cables went |
| 18:05.40 | brainiac | ok |
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| 18:08.48 | brainiac | citywok: thank you |
| 18:09.06 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 18:10.01 | citywok | brainiac: sure, although i'm not sure i provided a _TON_ of help heh. but if you need another tdm2400p or two... i've got a pair sitting on a shelf. |
| 18:10.31 | *** join/#asterisk teddy_salad (~bclark@eniccpsc01-v410.mid.sta.suddenlink.net) |
| 18:14.54 | brainiac | ok |
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| 18:26.28 | *** join/#asterisk smash- (smash@161.sub-72-102-19.myvzw.com) |
| 18:26.54 | smash- | Hey, I got a question. Has the issue with cisco pix been patched or are pix still trouble some to pass RTP throuhg? |
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| 18:40.15 | brainiac | citywok: Is there a type of diagnostic I can perform on the TDM2400P card? I have a customer who states that he suddenly started having problems on 2 (only 2) of his lines |
| 18:40.43 | citywok | i haven't played with those things since the days of zaptel & * 1.4, so i'm not going to be a whole lot of help software wise |
| 18:42.55 | brainiac | ok, thx anymay |
| 18:43.16 | Qwell | brainiac: simple fix |
| 18:43.27 | Qwell | swap the lines in the ports. does it follow the lines, or the ports? |
| 18:43.38 | chazzam | eh, if its fxs modules, can try fxstest. otherwise make sure to run fxotune, and yeah, what Qwell said |
| 18:44.00 | chazzam | always starts with fxotune for fxo module problems |
| 18:44.32 | Qwell | it sounds like citywork was right, with the lines needing to be reterminated. |
| 18:50.12 | brainiac | <PROTECTED> |
| 18:50.12 | brainiac | 12:43 < chazzam> eh, if its fxs modules, can try fxstest. otherwise make sure to run fxotune, and yeah, what Qwell said |
| 18:50.15 | brainiac | 12:43 * chazzam always starts with fxotune for fxo module problems |
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| 18:59.34 | *** join/#asterisk nny (~Scott@174.107.201.103) |
| 19:00.23 | nny | when trying to add cdr_adaptive_odbc in makemenuselect, it is greyed out. unixodbc and libtooldl are both installed, what else should be done? |
| 19:00.41 | nny | i have also added the c file for cdr-adaptive_odbc* |
| 19:03.18 | *** join/#asterisk ccesario_ (~ccesario@187.75.139.188) |
| 19:06.24 | leifmadsen | nny: did you re-run ./configure after installing the deps? |
| 19:06.47 | leifmadsen | -dev packages and not just standard pkgs too right? |
| 19:08.15 | nny | leifmadsen: can try -dev, do you mean for unixodbc or libtooldl? |
| 19:09.34 | nny | leifmadsen: tried it with a fresh set of files from asterisk-addons, moved the c file to /cdr/ makemenuselect has XXX next to adaptive_odbc, is there a way to correct this? |
| 19:11.10 | leifmadsen | both |
| 19:11.27 | nny | leifmadsen: ok thanks, installing dev for both now |
| 19:11.42 | leifmadsen | always need -dev pkgs to compile |
| 19:12.42 | nny | leifmadsen: yeah, wasn't aware the compilation required the dev packages in this instance, re running ./configure and menuselect now |
| 19:13.16 | nny | leifmadsen: odd, still XXX 1. cdr_adaptive_odbc with -devel packages for both libtooldl and unixODBC |
| 19:13.41 | nny | leifmadsen: i'll try both x86_64 and x86 devel |
| 19:13.48 | leifmadsen | you shouldn't use both |
| 19:13.49 | nny | eri386 |
| 19:13.54 | leifmadsen | that can cause problems |
| 19:13.55 | Qwell | why are you copying a file from addons? |
| 19:14.01 | leifmadsen | if you're on x86_64, just install those |
| 19:14.04 | Qwell | that's not going to work.. |
| 19:14.07 | leifmadsen | indeed |
| 19:14.16 | nny | leifmadsen: that's what I did |
| 19:14.30 | nny | leifmadsen: i haven't tried i386 yet, suggestion only |
| 19:14.39 | leifmadsen | that's a bad suggestion :) |
| 19:14.41 | nny | Qwell 1.4 doesn't support adaptive_odbc natively |
| 19:14.45 | nny | leifmadsen: i see |
| 19:14.48 | leifmadsen | you can't just copy the file over there |
| 19:14.51 | nny | leifmadsen: doesn't allow me to select it |
| 19:14.52 | nny | i didn't |
| 19:14.54 | leifmadsen | cdr_adaptive_odbc isn't part of 1.4 |
| 19:14.54 | Qwell | if it's in -addons, why not just install -addons? |
| 19:14.56 | nny | it's from digium |
| 19:15.01 | nny | oh joy |
| 19:15.04 | leifmadsen | what's from digium? |
| 19:15.08 | nny | the c file |
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| 19:15.13 | nny | emailed from the support team |
| 19:15.14 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
| 19:15.19 | leifmadsen | o.O |
| 19:15.31 | nny | during the last astercon when the web server was down |
| 19:15.40 | Qwell | http://svn.digium.com/community/tilghman/branches/1.4/ |
| 19:15.50 | nny | yup that's it |
| 19:16.00 | Qwell | get that, make, copy the .so |
| 19:16.29 | nny | that's the one I have |
| 19:17.58 | nny | XXX 1. cdr_adaptive_odbc |
| 19:18.08 | Qwell | you aren't listening. |
| 19:18.08 | Qwell | get that, make, copy the .so |
| 19:18.10 | nny | fresh untar of asterisk-addons-1.4.12 and the file you linked |
| 19:18.13 | nny | i can't make it |
| 19:18.20 | Qwell | step 1) GET THAT |
| 19:18.20 | nny | make menu select has it XXX out |
| 19:18.24 | nny | i DID |
| 19:18.24 | Qwell | tell me when you're done |
| 19:18.27 | nny | stop using caps |
| 19:18.29 | Qwell | the whole thing |
| 19:19.19 | nny | oh, you mean use the entire folder "1.4" above that tree |
| 19:19.47 | nny | get that is pretty vague, last time I did this the server was down and I had to obtain the file through another channel |
| 19:20.16 | Qwell | I would have linked the file if I intended you to get a file. |
| 19:22.50 | citywok | holy crap, * + Exchange Unified Messaging = awesome |
| 19:23.02 | nny | citywok interesting |
| 19:23.09 | citywok | voicemail transcription ftw |
| 19:23.28 | citywok | also, it took 5 minutes to make it work |
| 19:24.03 | nny | what folder does func_shared.c belong in? |
| 19:24.19 | nny | the rest are obvious |
| 19:24.24 | Qwell | the directory you checked out. |
| 19:24.32 | Qwell | 3 steps. follow them exactly. |
| 19:24.36 | Qwell | get that, make, copy the .so |
| 19:24.41 | nny | nm |
| 19:26.17 | nny | XXX 1. cdr_adaptive_odbc |
| 19:26.26 | nny | svn checkout http://svn.digium.com/community/tilghman/branches/1.4/ |
| 19:26.32 | nny | in addons folder |
| 19:26.35 | nny | make menuselect |
| 19:27.08 | nny | unless that's intentionally XX out |
| 19:27.16 | nny | and magically still compiles |
| 19:27.34 | Qwell | svn checkout http://svn.digium.com/community/tilghman/branches/1.4/; cd 1.4; make; cp *.so /usr/lib/asterisk/modules/ |
| 19:28.11 | nny | makes sense, also completely different than how we had to do it last time |
| 19:28.20 | Qwell | You did it wrong last time. |
| 19:28.27 | nny | well, maybe if the SVN server was up |
| 19:28.36 | nny | last time |
| 19:29.02 | Corydon76-home | nny: did you re-run the configure and make install in Asterisk after installing unixodbc-devel? |
| 19:29.08 | paulc | now let's all smile and be thankful because we have the right answer and can move on successfully :-) |
| 19:29.15 | nny | Corydon76-home: yes |
| 19:29.40 | Corydon76-home | Then it should have removed the XXX from cdr_adaptive_odbc |
| 19:30.03 | nny | Corydon76-home: apparently that is the wrong method, and separate compilation of the specific addon is the proper way. |
| 19:30.03 | nny | Corydon76-home: it didn't but thanks |
| 19:30.27 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-52-237-42.dhcp.embarqhsd.net) |
| 19:30.35 | nny | Corydon76-home: i am trying Qwells method now |
| 19:30.57 | Corydon76-home | What, copying the .c file into the Asterisk tree? |
| 19:31.03 | Qwell | Corydon76-home: running make |
| 19:31.08 | Corydon76-home | That should work, too |
| 19:31.14 | nny | in theory |
| 19:31.19 | nny | as soon as i correct asterisk's path |
| 19:31.39 | Corydon76-home | make ASTSRC=/path/to/asterisk/source |
| 19:32.55 | nny | http://pastebin.com/WNjjYT7m |
| 19:33.04 | nny | Â #error You must include stdio.h before file.h! |
| 19:33.31 | nny | always wondered why this isn't backported.. another conversation i assume |
| 19:33.33 | Corydon76-home | Okay, so cdr_adaptive_odbc compiled fine, but res_config_curl didn't |
| 19:33.41 | nny | yeah not sure why I need that |
| 19:33.47 | Corydon76-home | You're looking at the backport |
| 19:34.06 | Corydon76-home | Backports are not included in the main package |
| 19:34.11 | nny | yeah I assume so. |
| 19:34.42 | Corydon76-home | You probably don't need res_config_curl |
| 19:35.07 | Corydon76-home | It's an alternative realtime backend for those with shitty odbc drivers |
| 19:35.17 | nny | oddly I wouldn't be using 1.4 if this patch was available for 1.6 or ported over: https://issues.asterisk.org/view.php?id=15168 I have not see any evidence it is |
| 19:35.27 | nny | yeah this is just to allow custom fields in cdr afaik |
| 19:36.11 | Qwell | that is in 1.8 |
| 19:36.12 | Corydon76-home | We don't backport features, because it can seriously fuck up older versions |
| 19:36.17 | nny | makes sense |
| 19:36.56 | nny | Qwell: odd, tried 1.8, freepbx's format of queues.conf still caused an "Invalid" when showing queue members |
| 19:37.13 | Corydon76-home | and not just to allow custom fields, but also to populate them intelligently |
| 19:37.17 | nny | unless it is implemented in 1.8 different than the patch for 1.4 |
| 19:37.59 | nny | Corydon76-home: yeah understandable |
| 19:38.38 | nny | Qwell: I can fire up the VM and post a line from queues.conf that angers 1.8. is it possible digium's repos of 1.8 may not support it, but source does? |
| 19:38.54 | Corydon76-home | It was one of my very few awesome ideas |
| 19:39.20 | nny | Corydon76-home: a good one, it's used here to allow additional cdr fields to be populated bya piece of crm software |
| 19:39.41 | nny | Corydon76-home: and (afaik) also allows for multiple mysql databases to be populated at once (i.e. direct mirror) |
| 19:40.06 | nny | that may be not entirely correct, paraphrasing what I read 3 months ago |
| 19:40.25 | Corydon76-home | That's correct |
| 19:40.53 | Corydon76-home | although if they're mirrors, I don't know why you don't just use replication |
| 19:41.02 | nny | Corydon76-home: well not mirrors |
| 19:41.19 | nny | Corydon76-home: second db that is stripped and used to track recordings |
| 19:41.32 | nny | Corydon76-home: that's the theory, right now importing the data from cdr as needed |
| 19:41.46 | nny | Corydon76-home: which you can imagine isn't as elegant when you're doing it every X amount of hours |
| 19:41.46 | Corydon76-home | One of the main things is to allow each database to drop fields it doesn't want |
| 19:42.00 | nny | Corydon76-home: yeah that's the intention |
| 19:42.12 | Corydon76-home | I mean on the CDR |
| 19:42.45 | Corydon76-home | Adding custom fields is only half of cdr_adaptive_odbc. It's also dropping standard fields. |
| 19:42.50 | nny | Corydon76-home: yeah, one database will be full CDR, other is going to be just unique ID and some other fields pattern matched to file names from (cough*) freepbx |
| 19:43.00 | nny | don't shoot me, I ama vanilla guy at heart |
| 19:43.24 | Corydon76-home | Renaming fields was an addon, as was filters and (now) negative filters |
| 19:43.52 | nny | so I guess the bigger question is why 1.8 doesn't jive with queue hints |
| 19:43.57 | nny | or rather, my 1.8 |
| 19:44.07 | nny | doesn't 1.8 include adaptove by default? |
| 19:44.11 | nny | adaptive* |
| 19:44.20 | Corydon76-home | Yes, as does 1.6.2 |
| 19:44.38 | Corydon76-home | 'filter src = 12345' and 'filter src != 54321' |
| 19:44.57 | nny | ahh gotcha |
| 19:45.19 | *** join/#asterisk Tim_Toady (~moi@178.128.24.211.dsl.dyn.forthnet.gr) |
| 19:45.58 | nny | some of this dialplan is (sadly) ghostwritten by freepbx, would be a large undertaking to go through and rewrite each context for negative filters... the current field additions are done in a specific context. If this was vanilla I would have a field day |
| 19:46.38 | nny | I have preached the virtues of a scratch setup in this situation as hard as I can.. it's not an option for me |
| 19:47.39 | Corydon76-home | The negative filters are only in cdr_adaptive_odbc.conf |
| 19:47.46 | nny | Corydon76-home: yeah |
| 19:47.51 | nny | Corydon76-home: working on trying 1.8 |
| 19:48.15 | nny | Corydon76-home: and seeing it https://issues.asterisk.org/view.php?id=15168 (<-- what that adds to 1.4) works with fpbx |
| 19:48.31 | Corydon76-home | Just occurred to me... negative filters need to be allowed to have multiple settings |
| 19:49.40 | nny | ok |
| 19:50.00 | nny | member=Local/1265@from-queue/n,0,John Gacy,HINT:1265@ext-local |
| 19:50.06 | nny | should this work in 1.8? |
| 19:51.04 | nny | John Gacy (Local/1265@from-queue/n) (Invalid) has taken no calls yet |
| 19:51.09 | nny | is what I get |
| 19:51.43 | nny | gonna dig up the examples of how it's suppose to work with 1.8 and compare them, just curious if this is wrong |
| 19:54.02 | nny | Qwell: does that seem right to you? |
| 19:55.37 | Qwell | no, it's lowercase |
| 19:55.52 | nny | hint vs HINT? |
| 19:56.53 | nny | well i'll be |
| 19:57.00 | nny | Qwell: I owe you a coke |
| 19:57.48 | nny | now to figure out why freepbx does that.. Corydon76-home thanks for the adaptive tips, that's useful. Qwell thanks again |
| 20:01.57 | *** join/#asterisk jamko (~chatzilla@173.160.6.201) |
| 20:03.52 | jamko | Anyone know how to manipulate the way asterisk sends the P-Asserted Identity header? I want to change the Privacy token from "id" to "none" |
| 20:08.32 | l2trace99 | jamko: http://tinyurl.com/4c9r7dw |
| 20:08.35 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 20:10.55 | WIMPy | jamko: CALLERID(pres)? |
| 20:11.02 | jamko | l2trace99: Very good you clown. However I am looking for the setting which sets this by default, without having to modify the sip header for each call. |
| 20:15.20 | jamko | l2trace99: Nothing of real intelligence to offer here? Didn't think so... jackass. |
| 20:19.47 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 20:27.55 | *** part/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 20:40.30 | *** join/#asterisk xheliox (jeff@pdpc/supporter/student/xheliox) |
| 20:46.51 | *** part/#asterisk hexanol (~etienne@modemcable094.94-70-69.static.videotron.ca) |
| 20:50.47 | *** join/#asterisk xheliox (jeff@pdpc/supporter/student/xheliox) |
| 20:52.37 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 20:52.39 | [sr] | howdy |
| 20:52.43 | [sr] | hi WIMPy :) |
| 20:53.40 | *** part/#asterisk nny (~Scott@174.107.201.103) |
| 20:54.49 | *** join/#asterisk xheliox (jeff@pdpc/supporter/student/xheliox) |
| 20:56.23 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 21:02.53 | WIMPy | Hi [sr]. What's up in the wild south? |
| 21:03.32 | [sr] | ei WIMPy, nothing new, some work, less money every day |
| 21:03.49 | [sr] | i'm sure our country has been in your news several times :) |
| 21:04.24 | WIMPy | I prefer not to read news. Always gives you a bad mood. |
| 21:09.30 | [sr] | always! |
| 21:09.54 | [sr] | ei WIMPy, any idea on bypass the problem on compile LCR with asterisknow (asterisk1.8) |
| 21:10.17 | [sr] | i've sent an email and Andread didn't answered... you also replied to that email with some patch, can't remember which |
| 21:10.24 | WIMPy | What problem? |
| 21:10.30 | *** join/#asterisk ront (c2e659e1@gateway/web/freenode/ip.194.230.89.225) |
| 21:10.41 | [sr] | it doesn't compile |
| 21:11.30 | [sr] | the problem are the old libraries os centOS :( |
| 21:11.39 | WIMPy | I did an old one that went in to the branch callled asterisk_1_8. |
| 21:12.18 | ront | i'm going in sane trying to find the cause of this error. I'm trying to get T.38 working with a SIP trunk. |
| 21:12.19 | [sr] | does it compile with the libraries present in asterisknow 1.7? centos |
| 21:12.31 | ront | my providor supports it |
| 21:12.31 | WIMPy | And I also did a merger from the development branch. My versions are still up at [15:58] SuPrSluG has joined #asterisk (~SuPrSluG@8.22.96.106) |
| 21:12.36 | WIMPy | And I also did a merger from the development branch. My versions are still up at [15:58] mr_ian has left IRC (Ping timeout: 255 seconds) |
| 21:12.41 | WIMPy | Argh. |
| 21:13.01 | ront | my * is not negotiating T.38 prperly |
| 21:13.05 | WIMPy | Ah, shit. Script crashed. |
| 21:13.14 | [sr] | lol happens |
| 21:13.32 | WIMPy | http://voice.yeti.dk/patches |
| 21:13.45 | ront | it gets a reinvite by my providor an then sends "100 trying" but never sends a "200 OK" |
| 21:13.46 | WIMPy | So what happens when you compile? |
| 21:13.53 | ront | can some one give me some pointers? |
| 21:14.19 | wizard171 | ront, what version of asterisk? |
| 21:15.14 | [sr] | WIMPy: hum wait, now i'm missing some headers, i did the test on another machine |
| 21:15.18 | [sr] | allow me some minutes |
| 21:15.28 | mzb | leifmadsen: there's a very simple way to script asterisk+menuselect |
| 21:17.16 | ront | wizard171 1.6.2.13 |
| 21:17.29 | mzb | mind you, not as elegant as menuselect/menuselect --enable ;) |
| 21:19.02 | ront | wizard171 i've got a "SIP trunk > Firewall/Router (NAT) > Asterisk setup" |
| 21:19.15 | [sr] | WIMPy: with your branch, http://pastebin.com/W3JSqwbX |
| 21:19.16 | BlackBishop | what's that thing in 1.8 with more verbose logging of a call ? |
| 21:19.26 | chazzam | debug? |
| 21:19.30 | BlackBishop | I want to store in csv/sql what datacard calls where |
| 21:19.31 | chazzam | :p CEL ? |
| 21:19.51 | BlackBishop | yeah .. I think |
| 21:19.58 | ront | wizard171 then i use iaxmodem > hylafax for Fax to Email |
| 21:21.24 | WIMPy | [sr]: That looks to me as if you're using a pre 1.8 version of Asterisk. |
| 21:21.50 | [sr] | WIMPy: wait wait... oh dumb me, this machine has asterisk 1.6x :| |
| 21:22.01 | [sr] | ops, didn't said nothing!! |
| 21:22.22 | [sr] | i want 100% i was on a asterisk 1.8 box |
| 21:22.54 | wizard171 | ront, your other calls work (inbound from provider)? just not faxes, right? |
| 21:23.02 | ront | yes |
| 21:23.20 | ront | on G711 it works sometimes |
| 21:24.07 | WIMPy | [sr]: The version from the development branch is supposed to work on both versions, but last time I checked it didn't work at all. |
| 21:24.13 | ront | since its behind NAT i fowarded ports 4000 -4500 though |
| 21:24.57 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-52-237-42.dhcp.embarqhsd.net) |
| 21:25.15 | *** join/#asterisk espiceland (~espicelan@207.98.195.107) |
| 21:26.14 | WIMPy | [sr]: That's also in the december version I put up. |
| 21:26.17 | wizard171 | ront, and in sip.conf your t38pt_udptl=??? |
| 21:26.35 | ront | @wizard171 its talking fine to the providors asterisk, but when the providor sends a 200 reinvite for T38, my * never replys with 200 OK and port info etc.. it simply sends a 100 trying and eventually 488 not accepted here |
| 21:26.46 | [sr] | WIMPy: i believe that version is the one in the 1.8 branch right? as least it has 6 week of age |
| 21:26.47 | ront | wizard171 yes |
| 21:27.03 | ront | its t38pt_udptl=yes |
| 21:27.36 | WIMPy | [sr]: No, it's somewhere between the 1.8 and the development branch. |
| 21:28.03 | BlackBishop | chazzam: yup, seems more verbose than the CDR |
| 21:28.12 | WIMPy | And both will only work up to Asterisk 1.8.3, because of the new stream format handling. |
| 21:29.14 | [sr] | WIMPy: i'll do the tests on the machine (asterisknow 1.7x) i have with asterisk 1.8 and let you know the result! |
| 21:29.19 | wizard171 | ront, do you get any messages about it on console or logs? are they sending with "error correction" or do you know? maybe you have a SIP dialog of failed session you could pastebin somewhere ? I am looking for what inbound asked for ... |
| 21:29.45 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:29.46 | [sr] | WIMPy: tell me, what's best, install the misdn headers of centOS (that are i suppose misdn 1.x), or compile a freash new misdn 2.x ? |
| 21:30.27 | WIMPy | If you have a kernel >=2.6.26, that should be fine. |
| 21:30.49 | ront | i've got a wireshark capture |
| 21:30.56 | [sr] | it's the centOS kernel, 2.6.18 :( how i hate centOS! |
| 21:31.03 | [sr] | this could be a debian install!! |
| 21:31.09 | ront | wizard171 i've got a wireshark capture, i will postbin |
| 21:31.10 | LemensTS | Can you call a Polycom 321 and talk over its intercom (intercom paging) ? Ive done it with a Polycom 501 a few years ago, I had to edit the cfg files on the phone via tftp boot server... |
| 21:31.27 | _Corey_ | LemensTS: sure |
| 21:31.46 | WIMPy | [sr]: That version sounds like debian :-) |
| 21:31.46 | *** join/#asterisk anil-lim (~chatzilla@122.162.142.232) |
| 21:31.55 | LemensTS | Corey: thanks I figured it was the same |
| 21:32.52 | _Corey_ | LemensTS: yeah, hasn't changed really |
| 21:36.28 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 21:37.33 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 21:40.26 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 21:42.11 | ront | wizard171 i'm switching to my IRC client so we can have private chat.... |
| 21:42.24 | *** join/#asterisk adeel (~adeel@c-67-174-36-109.hsd1.ca.comcast.net) |
| 21:42.39 | *** join/#asterisk eerie (hoax@gateway/shell/bshellz.net/x-lgpjoufjbqjjogie) |
| 21:44.44 | *** join/#asterisk ront (~Adium@194-230-89-225.static.adslpremium.ch) |
| 21:46.26 | *** join/#asterisk Micc (~quassel@c-24-18-20-54.hsd1.wa.comcast.net) |
| 21:57.02 | Micc | is mixmonitor supposed to work after the channel is parked and picked up? |
| 22:02.20 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
| 22:04.09 | *** join/#asterisk StaRetji (~BigAll@91.143.222.166) |
| 22:06.00 | Kobaz | <PROTECTED> |
| 22:06.14 | Kobaz | how do i get fastagi to exit with 0 instead of -1 |
| 22:08.12 | *** join/#asterisk manji (~manjiki@ppp-94-65-220-14.home.otenet.gr) |
| 22:11.26 | wizard171 | Kobaz, my PHP ones give whatever I give to "exit(0)", zero in my case ... not sure if you are giving one or not ... or what yours are written in ... |
| 22:12.07 | Kobaz | well your exit code in fastagi isn't going to have anything to do with the ext code of the script |
| 22:13.55 | Kobaz | since it's over a socket.. the agi call will have no idea |
| 22:14.42 | wizard171 | Kobaz, yeah, I can see that .. as I just tested it ... :) I am now looking to see how to influence it ... |
| 22:16.56 | Kobaz | heh |
| 22:19.38 | wizard171 | Kobaz, I've run across something that says AGI returns -1 if the script receives or requests "hangup" and zero otherwise ... |
| 22:20.08 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 22:21.37 | [sr] | WIMPy: going to sleep, chat later! |
| 22:21.57 | WIMPy | gn |
| 22:24.31 | Kobaz | ah |
| 22:24.39 | Kobaz | i think i need to add a function then |
| 22:24.44 | Kobaz | SET_AGI_STATUS |
| 22:24.45 | Kobaz | or something |
| 22:24.55 | Kobaz | i want a "this script finished correctly" flag |
| 22:26.57 | wizard171 | Kobaz, yeah, my bad, I actually have a note in my code that says I can't rely on the "exit" value ... go figure ... its dated back in 2008! (its hell getting old ...) |
| 22:27.20 | seanbright | there already is an AGISTATUS i think |
| 22:27.28 | Micc | There is a bug from 2008 about MixMonitor not recording after a transfer and some new audiohooks code was introduced to solve the problem. Would that also apply to parking, which is transfer then a pickup/bridge I believe? |
| 22:28.20 | Kobaz | seanbright: oh, that'll do it |
| 22:28.32 | *** join/#asterisk blee_ (~blee@99-117-188-231.lightspeed.dybhfl.sbcglobal.net) |
| 22:28.38 | Kobaz | Micc: in which asterisk version? |
| 22:29.07 | Micc | Kobaz, 1.6.2.17-rc2 |
| 22:29.18 | Kobaz | oh wow |
| 22:29.55 | Micc | Kobaz, the bug I was talking about was in some version of 1.4. |
| 22:31.46 | Kobaz | oh okay |
| 22:33.08 | Micc | Is voip-info.org still the best place for asterisk information? |
| 22:33.28 | fauxalliance | Micc, google is... |
| 22:33.37 | _Corey_ | I'd recommend https://wiki.asterisk.org/wiki/display/AST/Home |
| 22:33.37 | *** join/#asterisk Lantizia (~Lantizia@erebus.seaquake.net) |
| 22:34.02 | Micc | google always comes up with voip-info. |
| 22:34.10 | fauxalliance | Micc, o? |
| 22:34.52 | Micc | wiki.asterisk.org is great, I've not seen that before. |
| 22:35.36 | _Corey_ | A lot of the voip-info content is very dated |
| 22:35.37 | Micc | fauxalliance, yeah whenever I google asterisk {dialplan command} voip-info is usually the first thing and everything else is garbage. |
| 22:35.53 | Micc | _Corey_, yes thats why I asked. |
| 22:36.24 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 22:37.05 | LemensTS | _Corey_: you still here? |
| 22:37.16 | _Corey_ | I'm here |
| 22:38.18 | LemensTS | _Corey_: cool another question. On polycoms if you use the auto-answer capability in the cfg file to do intercom-paging, can the person who has the phone that you page talk back to you? |
| 22:38.35 | LemensTS | 2 way intercom-paging I suppose it would be called |
| 22:38.37 | _Corey_ | It's really a function of how you page them on the asterisk side |
| 22:38.42 | _Corey_ | so, yes, you can |
| 22:38.59 | Micc | Theres also a mute option I think in the polycom config. |
| 22:39.19 | LemensTS | Cool, I was hoping they could just talk and not have to press a button |
| 22:39.31 | _Corey_ | Just dial them after setting the alert-info, and voila |
| 22:39.36 | _Corey_ | intercom |
| 22:39.39 | Micc | LemensTS, yes I do that a lot. |
| 22:39.53 | LemensTS | How long can they just talk before it cuts them off? |
| 22:40.01 | _Corey_ | Indefinitely |
| 22:40.08 | Micc | till someone hangs up. |
| 22:40.11 | paulc | core show calls --> 2934210 calls processed... almost 3 million since last restart.. that "warms the cockles of my heart" as my old grandmother used to say :-) |
| 22:40.12 | _Corey_ | unless you set an absolute timeout or something |
| 22:40.30 | LemensTS | Ah I gotchya, its just auto-answering their speaker phone, I understand |
| 22:40.39 | LemensTS | duh |
| 22:40.39 | LemensTS | lol |
| 22:40.55 | Micc | paulc, I bet the bill for all those calls warms something else. |
| 22:42.28 | _Corey_ | :) |
| 22:42.45 | *** join/#asterisk jamko (~chatzilla@173.160.6.201) |
| 22:43.30 | leifmadsen | mzb: eh? |
| 22:43.30 | *** join/#asterisk Corydon76-home (gold@c-69-137-80-31.hsd1.tn.comcast.net) |
| 22:43.30 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 22:43.31 | paulc | Micc: we pay for 1 leg (outbound) and the 2nd leg is internal SIP.. it's a click-2-call type service.. The marketing people love it apparently |
| 22:43.59 | mzb | leifmadsen: nm, I do it a *nasty* way by comparison ;) |
| 22:44.29 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-156-107.dsl.stlsmo.sbcglobal.net) |
| 22:45.00 | adeel | i'm having a little trouble wrapping my head around domain support in *...is there any documentation someone can refer me to that'll give some examples as to when to use it and why? |
| 22:46.15 | leifmadsen | mzb: oic :) |
| 22:46.25 | leifmadsen | not really |
| 22:46.31 | leifmadsen | just whatever is in sip.conf.sampl.e |
| 22:46.48 | leifmadsen | and perhaps the voip-info wiki (although I would expect it to be out of date) |
| 22:47.04 | leifmadsen | adds it to the list of things to discuss in the next version of A:TDG |
| 22:47.24 | Kobaz | Asterisk: The dang gathering? |
| 22:48.15 | leifmadsen | yes |
| 22:48.25 | Kobaz | sweet |
| 22:48.43 | adeel | the sip.conf.sample doesn't really explain why you'd set it...just how to set it |
| 22:48.59 | Kobaz | and i think domains arent fully supported too |
| 22:49.04 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 22:49.11 | leifmadsen | true |
| 22:49.21 | leifmadsen | if you don't know whether you need it, you probably don't need it :) |
| 22:49.34 | leifmadsen | it's essentially to route and control calls (at least incoming calls) based on domain |
| 22:49.49 | adeel | ah....hmmm.... |
| 22:49.56 | leifmadsen | it would be useful if you were controlling multiple incoming domains to the same server |
| 22:50.12 | Kobaz | i want to control incoming calls based on explicit matching |
| 22:50.14 | adeel | and you wanted to process them separetly |
| 22:50.15 | leifmadsen | you can ping oej when he comes online as he likely wrote it |
| 22:50.20 | leifmadsen | yes |
| 22:50.22 | Kobaz | glares at sip.conf |
| 22:50.27 | leifmadsen | it's just a lot easier to do it all in the dialplan |
| 22:50.34 | leifmadsen | heads out for a bit |
| 22:51.50 | adeel | on a slightly different note, i noticed in the Polycom Admin manual that the polycom's support a SIP Warning header, that can display a 3 sec pop up on the screen....any way to implement that in *? |
| 22:54.06 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 22:54.16 | chandoo | hi |
| 22:54.30 | paulc | adeel: ooh that's sexy.. must be in newer firmware than I've played with in the past.. I should think about getting me a new Polycom phone to play with.. |
| 22:55.07 | chandoo | i am using allvoi service for a while now, i recently got new softphone software, what are the setting of STUN for |
| 22:55.17 | adeel | paulc, yeah i thought so too....i recently found a 650 in my closet, so i just upgraded to 3.2.2, and saw that they had a little screen shot showing a call parking notification |
| 22:55.44 | chandoo | i am making calls from my iphone without stun configuration, why is stun really needed |
| 22:55.49 | paulc | Hmm. I think my one at home is a 650? I'll have to poke around.. |
| 22:56.04 | adeel | paulc, 650's have the color backlight |
| 22:56.17 | Kobaz | 650? they dont have color |
| 22:56.20 | Kobaz | 670 does |
| 22:56.30 | Kobaz | 650 just has a backlight |
| 22:56.35 | adeel | Kobaz, err, yeah, sorry, just the backlight |
| 22:56.37 | paulc | ah no.. mine's 6 line but not HD.. or backlit.. |
| 22:56.47 | adeel | probably the 601 |
| 22:56.49 | nestAr | 550's have backlight too |
| 22:56.49 | paulc | the backlight is nice - my 2nd phone at work has it.. |
| 22:57.32 | adeel | btw, the pic is on page 4-76 of the admin manual...http://supportdocs.polycom.com/PolycomService/support/global/documents/support/setup_maintenance/products/voice/spip_ssip_vvx_Admin_Guide_SIP_3_2_2_eng.pdf |
| 22:58.18 | *** join/#asterisk Poincare (~jefffnode@2001:6f8:14ee:4:230:48ff:fe86:1622) |
| 22:58.40 | paulc | goes off for a quick look before a meeting drags him away from his desk... |
| 22:59.53 | voxter | Anyone here use lumenvox/ |
| 23:06.58 | nestAr | anyone familiar with the snom m3 setup? |
| 23:10.18 | _Corey_ | adeel: That could be pretty sweet... "Agent logged into queue X" for example |
| 23:11.16 | adeel | _Corey_, yep yep |
| 23:11.36 | adeel | _Corey_, X number of people waiting in queue |
| 23:12.15 | _Corey_ | I'm wondering what kind of message we'd need to throw at a phone tho... I'm not finding too many examples. |
| 23:12.19 | _Corey_ | maybe a NOTIFY |
| 23:12.36 | adeel | _Corey_, according to the polycom manual, it's a SIP Warning header |
| 23:12.54 | _Corey_ | Well... the question is what kind of SIP message to stick that header into |
| 23:13.03 | adeel | probably a notify |
| 23:13.32 | _Corey_ | I'm sure I could rig something with sipsak if I can find an example |
| 23:13.36 | adeel | Warning Yes Only warning codes 300 to 399 |
| 23:13.50 | adeel | from page B-4 |
| 23:14.09 | adeel | http://sofia-sip.sourceforge.net/refdocs/sip/group__sip__warning.html |
| 23:14.19 | _Corey_ | here's someone trying it with a notify |
| 23:14.21 | _Corey_ | http://www.mail-archive.com/asterisk@uc.org/msg07945.html |
| 23:17.32 | adeel | hmmm....it probably should be the same kind of a message * uses for other 30* messages |
| 23:17.53 | _Corey_ | I was about to leave for the day but I'm going to try it :) |
| 23:17.59 | adeel | hehe |
| 23:18.16 | adeel | i have a polycom handy, and can install sipsak real quick, if you need me to try |
| 23:18.30 | _Corey_ | nah, I have a couple 550s on my desk |
| 23:18.41 | adeel | _Corey_, are they running the 3.2.2 firmware? |
| 23:18.52 | _Corey_ | 3.2.3 i think |
| 23:18.58 | adeel | ah ok, then you should be fine |
| 23:19.02 | _Corey_ | it had the option in the sip.cfg, it's disabled by default |
| 23:21.31 | adeel | let me know if it works |
| 23:22.46 | _Corey_ | I'm just modifying a script i had to do VM notifies |
| 23:30.23 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 23:35.57 | _Corey_ | adeel: Have you found any other examples? I'm wondering if it's supported in a NOTIFY at this point (keep getting "400 Bad Request" from the phone) |
| 23:37.45 | adeel | _Corey_, nothing really...the only thing i came across was the SIP RFC, and they had it sent as an options message |
| 23:38.24 | _Corey_ | I checked the RFC but didn't see a reference to OPTIONS specifically |
| 23:38.30 | _Corey_ | which section are you looking at? |
| 23:39.12 | _Corey_ | oh, 11.2 |
| 23:39.15 | _Corey_ | i see |
| 23:39.17 | adeel | _Corey_, i just did a full text search on the word 'warning' and saw something showing an OPTIONs |
| 23:40.02 | adeel | llow, Accept, Accept-Encoding, Accept-Language, and Supported header |
| 23:40.02 | adeel | <PROTECTED> |
| 23:40.02 | adeel | <PROTECTED> |
| 23:40.02 | adeel | <PROTECTED> |
| 23:40.02 | adeel | <PROTECTED> |
| 23:40.03 | adeel | <PROTECTED> |
| 23:40.05 | adeel | <PROTECTED> |
| 23:40.07 | adeel | <PROTECTED> |
| 23:40.18 | _Corey_ | yeah, I saw it thanks |
| 23:42.07 | _Corey_ | well, the Polycom doesn't bark when you sent it via an OPTIONS msg, though it doesn't do anything else either :) |
| 23:42.30 | adeel | hmmm...might be worth getting some clarification from polycom |
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| 23:49.03 | _Corey_ | I have a suspicion this could be an in-dialog response in an OK or something |
| 23:49.53 | adeel | possibly, or a subscribe event maybe? |
| 23:50.22 | _Corey_ | I don't think so on the SUBSCRIBE because I can successfully send those to the phone |
| 23:50.53 | adeel | try hanging the warning header off the subscribe, see what happens |
| 23:51.37 | _Corey_ | hmm |
| 23:51.48 | adeel | actually, i think you might be right...it would make the most sense to send this near the end of the in-dialog response |
| 23:51.59 | adeel | especially if the example Polycom is using is for call parking |
| 23:52.23 | _Corey_ | Yeah, I'm not sure but I suspect sending spontaneous messages to phones may not be possible using this method |
| 23:52.40 | adeel | probably |
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| 23:53.04 | _Corey_ | I don't think I could use SUBSCRIBE... I don't think the phone will respond to it |
| 23:53.12 | adeel | brb |
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