| 00:00.48 | EmleyMoor | draygon: I do |
| 00:07.21 | EmleyMoor | At work, someone referred to using Asterisk and Festival togeether as "Festerisk" the other day |
| 00:07.25 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 00:07.41 | EmleyMoor | This keyboard seems to be doing odd things |
| 00:10.11 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 00:10.23 | EmleyMoor | "Festerisk" is just the kind of hybrid word my dad would have liked! |
| 00:12.54 | *** join/#asterisk xenonex (n=xenonex@89.218.237.221) |
| 00:15.46 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.12.64.27) |
| 00:16.14 | EmleyMoor | I actually use Festival for all my non-personal "public-side" announcements, and sometimes for internal ones if I need to set them up quickly with no time to record my own. |
| 00:18.12 | TJNII | I use festival in AGIs because simply typing in a string I want in the prompt is easier than pre-recording and mapping a bunch of sound files. |
| 00:18.35 | TJNII | Plus it works for my 411 script, which pulls from a SQL database. |
| 00:20.07 | EmleyMoor | My "known marketing number" trap is a bit of a masterpiece, I think |
| 00:20.12 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 00:21.20 | EmleyMoor | First, they get instructed by Festival, then by Jay Benham then they end up leaving Voicemail (if they bother!) |
| 00:21.54 | TJNII | Hahaha. I do have a "queue" in one script that consists of for($count = 0; $count < 120; $count += 1) {do_command("EXEC playback \"custom/tech_music\"");} |
| 00:22.11 | TJNII | Where tech_music is a 30 second clip of bad hold music. |
| 00:22.45 | TJNII | (It also plays some sound files telling you about "priority service," but I didn't want to flood the channel |
| 00:23.23 | MatBoy | I think the airplane with my 2x 410 card crashed or something :S |
| 00:23.32 | MatBoy | *cards |
| 00:23.41 | jblack | Did they also assure you that you're call was important to them while dumping you off on bad music? |
| 00:24.31 | TJNII | Of course. |
| 00:24.33 | EmleyMoor | I am tempted to add "The importance of your call is yet to be determined." |
| 00:24.48 | *** join/#asterisk Katty (n=The@adsl-75-59-138-34.dsl.stlsmo.sbcglobal.net) |
| 00:24.58 | jblack | EmleyMoor: Nice!. |
| 00:25.30 | Katty | allo. |
| 00:25.47 | EmleyMoor | Since I got Asterisk configured correctly I have not had one unwanted phone call wake me up, apart from the necessary evil of work calls. |
| 00:26.07 | jblack | set up a hold that makes them randomly perform simple addition, hanging up if they either don't answer, or answer incorrectly. |
| 00:26.14 | riddlebox | does anyone have a Grandstream HT488? I set it to periodically subscribe to MWI, but after like an hour the phone doesnt work, I have to unplug it then plug it back in? If I set MWI to not subscribe the phone works perfectly? |
| 00:26.15 | *** join/#asterisk cardiff (n=cardiff@76-10-153-160.dsl.teksavvy.com) |
| 00:26.49 | jblack | "Please enter the sum of 3+4 to continue holding." |
| 00:27.31 | billytwowilly | anyone familiar with the asterisk appliance here? |
| 00:28.11 | TJNII | I thought about making the script with the "queue" call the user back if they hung up either in the "queue" or the string of pointless IVR menus beforehand, but I thought better of it. |
| 00:28.50 | jblack | Why did you decide against it? |
| 00:28.50 | TJNII | "I'm sorry we were disconnected. We care about your call. Let me place you where you left off." |
| 00:29.07 | TJNII | Because people might come for my head. :) |
| 00:29.09 | billytwowilly | How hard is it to configure the appliance to auto route numbers to specific phones on the lan? i.e can I have 15 people with 15 phones, get 15 dids and then associate each did with a phone on the net so the asterisk appliance passes calls through automagically? |
| 00:29.22 | jblack | Oh, you don't mean as a feature... You mean.... call hold stalking. |
| 00:29.23 | jblack | Vicious! I love it! |
| 00:29.34 | TJNII | Yea |
| 00:29.45 | jblack | "I told you your call was important to me." |
| 00:29.53 | *** join/#asterisk voiceperu (n=al@190.42.38.201) |
| 00:30.03 | voiceperu | hellooo |
| 00:30.14 | TJNII | It's not really a feature when the "queue" is just an hour of sounds. |
| 00:30.51 | jblack | tell me the sounds aren't that "the computer is thinking really hard" boo-de-beeps. |
| 00:31.50 | TJNII | This is all in my "tech support" script. It looks at the calling context, and reacts based off that. If it is from a phone I maintain, it drops you into a real queue. If it is from outside, if messes with you for at least an hour and a half. |
| 00:32.13 | *** join/#asterisk kimosabe (n=nat@adsl-69-155-128-143.dsl.hstntx.swbell.net) |
| 00:32.21 | voiceperu | can anybody test my asterisk server? |
| 00:33.49 | kimosabe | can i recieve pri on my ds3 to my cisco router while recieving internet bandwith and then from my cisco send the pri to asterisk box |
| 00:34.53 | *** join/#asterisk St1ckm4n (i=St1ckm4n@75.145.72.133) |
| 00:37.41 | St1ckm4n | does anyone here have much experience with asterisk in a call center enviroment? |
| 00:38.49 | voiceperu | can anybody type 190.43.129.89 |
| 00:38.56 | voiceperu | in a browser |
| 00:40.25 | jblack | I'm thinking about using the "youuuuu, you got what I neeeed, but you say he just a friend" song in loop as my hold music |
| 00:40.51 | voiceperu | sorry |
| 00:41.03 | voiceperu | type 190.42.38.201 |
| 00:41.24 | TJNII | That is a good song. |
| 00:41.36 | TJNII | I have American Pie |
| 00:41.44 | TJNII | And Safety Dance |
| 00:42.27 | TJNII | But when I really feal cruel, I use this: http://www.amazon.com/Beatles-Bossa-Brazilian-Tropical-Orchestra/dp/B00000G7GF |
| 00:42.33 | jblack | How about I'm too sexy? |
| 00:42.36 | *** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) |
| 00:42.45 | Idle | what variable is used for the timestamp? |
| 00:42.56 | TJNII | And I use the clips I downloaded off Amazon, so you never get more than 35 seconds of a song. |
| 00:43.00 | jblack | Perhaps everything at http://www.blender.com/guide/articles.aspx?id=786 |
| 00:43.38 | TJNII | God that list is bad. |
| 00:44.54 | jblack | Ohhhhh. ice ice baby |
| 00:50.37 | TJNII | I have the entire bible in WAV files, but in never made its way into the moh directory. |
| 00:51.41 | jblack | If I were rich, I'd hire allison to read /usr/share/dict/words. |
| 00:52.35 | rkeene | I've got a design question -- I'm going to be migrating from a legacy system and I want to change as little as possible. I'm basically only migrating *PART* of the legacy system (the part under my control), so users are used to dialing 8-XXXX to get someone else on base, and I want to preserve this functionality (going out to the PSTN if it's not a SIP phone)... Is there any way to do this without enumerating all of my extensions in the dialplan ? |
| 00:52.40 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.12.64.27) |
| 00:53.04 | jblack | I can't believe walk like an egyptian didn't make it onto the list |
| 00:53.30 | TJNII | rkeene: It can be done with a pattern match and a goto |
| 00:53.44 | *** part/#asterisk cardiff (n=cardiff@76-10-153-160.dsl.teksavvy.com) |
| 00:54.17 | *** join/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
| 00:54.18 | rkeene | TJNII, How so ? |
| 00:54.36 | rkeene | All the pattern matching I've read about was sequential, numerical, or string based |
| 00:55.16 | ThatKidKel | cdr_pgsql.so question.. is the spool=pgsql.spool only available after applying the patch? what is recommended for catching cdrs when the database is unavailable?? |
| 00:55.36 | jblack | OH YES! |
| 00:55.38 | jblack | http://youtube.com/watch?v=p-At6wk_fQs |
| 00:55.43 | TJNII | Put your extensions in one context. Don't give users access it. Then something like exten => _8XXXX,1,Goto(hidcontext,${EXTEN:1},1) |
| 00:56.00 | *** part/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
| 00:56.34 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-7addb4c2794e0f4d) |
| 00:57.17 | TJNII | jblack: When video phones become standard..... |
| 00:57.26 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 00:57.54 | rkeene | TJNII, "Put my extensions in one context" ? |
| 00:58.18 | voiceperu | where i can find this in my debian ? |
| 00:58.21 | voiceperu | You don't have permission to access /html/admin/modules/recordings/popup.php on this server. |
| 00:58.34 | TJNII | I think I may have misunderstood you, what do you mean by "not enumerating all my extensions" |
| 00:58.36 | rkeene | (Enumerate them in a context, or specify that they belong to a particular context when defining them) |
| 00:58.37 | voiceperu | i dont see html folder |
| 00:59.47 | TJNII | Oh, you want _8XXXX to transfer out of your system? |
| 01:00.20 | rkeene | TJNII, i.e., I don't want to have something like: exten => 81001,1,Macro(blah) exten => 81003,1,Macro(blah) in my dial plan to have 1001, and 1003 be local and 1002 be external. |
| 01:00.52 | rkeene | I guess I could, but I already have the list of extensions in so many places :-P |
| 01:01.20 | TJNII | Oh, yea. So do you want 81003 and 1003 to work? |
| 01:01.34 | TJNII | Or just 81003? |
| 01:01.58 | rkeene | Just 8XXXX |
| 01:02.25 | rkeene | But 81002 is external (via PSTN trunk), while 81001 is internal |
| 01:04.08 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 01:04.13 | *** join/#asterisk zen-froglet (n=zen@starlight.chat.za.net) |
| 01:04.33 | zen-froglet | morning. |
| 01:04.43 | TJNII | You'll need to sort the 4 digit extensions so they, and only they, are accessable through one context, if you haven't already. Don't include this context in the context of your phones. Then, put that pattern match in the context of the phones and point it at the "hidden" context. |
| 01:04.57 | zen-froglet | what would be the recommended app to use for receiving Faxes with Asterisk ? |
| 01:05.12 | TJNII | So it will match any 5 digit extension starting with 8, and then look in your already defined extensions. |
| 01:06.34 | rkeene | So basically I will have to list out all of my extensions in the dial plan, correct ? |
| 01:07.22 | TJNII | Well, yes and no. |
| 01:07.38 | rkeene | (I have 300 extensions) |
| 01:07.41 | TJNII | You will need to seperate the 4 digit extensions from the 5 digit extensions |
| 01:07.54 | rkeene | They will all be dialed as 5 digit extensions |
| 01:07.59 | TJNII | Right. |
| 01:08.29 | TJNII | But unless you want to support 4 digit extensions, you must seperate them from any 5 digit extensions in the dialplan contexts. |
| 01:08.48 | TJNII | I guess, how do you have it set up now? Do all 300 extensions work? |
| 01:09.04 | rkeene | And to do that I will need to put the list of all valid extensions in the dial plan ? |
| 01:09.13 | *** join/#asterisk RoyK (n=roy@ip-77-55-149-91.dialup.ice.no) |
| 01:09.35 | rkeene | There are over 300 extensions under 8XXXX right now, I want a subset of them (300) to resolve to SIP accounts, and the rest (around 4500) to resolve to PSTN terminations |
| 01:10.10 | rkeene | So if I dial 81002 from my SIP phone, I get the PSTN 81002, but if I dial 81001 I get the SIP 81001 |
| 01:11.05 | voiceperu | hellooo , why i cant play my recordings in freepbx? |
| 01:11.09 | voiceperu | i get this |
| 01:11.10 | TJNII | Is there a rhyme or reason to which exten goes where? |
| 01:11.13 | voiceperu | You don't have permission to access /html/admin/modules/recordings/popup.php on this server |
| 01:11.21 | TJNII | If not, how were you planning on not enumerating each one? |
| 01:11.26 | rkeene | No, they are a random subset |
| 01:11.31 | TJNII | voiceperu: #freepbx |
| 01:11.41 | rkeene | I've already enumerated them in the SIP configuration, I was hoping I could reference that list somehow |
| 01:12.01 | TJNII | yea, you've done it as 4 digit extensions correct? |
| 01:12.19 | TJNII | All in a context? |
| 01:12.53 | rkeene | No, in the SIP configuration they are the 5 digit (8XXXX) values.. but merely for convience and I could change them to the 4 if that would be easier |
| 01:13.03 | rkeene | Yes, all in the same context, even |
| 01:13.25 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 01:13.27 | *** join/#asterisk stansmith (n=stansmit@d60-65-93-136.col.wideopenwest.com) |
| 01:13.35 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 01:13.54 | TJNII | Okay, so you've done it in the SIP configs, but you haven't done the dialplan yet? |
| 01:14.15 | rkeene | Right |
| 01:14.33 | rkeene | (Well, the SIP config was generated from a script, pulling the information from LDAP) |
| 01:14.39 | *** part/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
| 01:14.40 | TJNII | So your dialplan is currently blank. |
| 01:15.37 | rkeene | No, it just doesn't allow for dialing 8XXXX when you want the legacy system's 8XXXX |
| 01:15.53 | TJNII | Okay, so you have a dialplan with 4 digit extensions defined |
| 01:15.54 | rkeene | I can dial 81234 and get SIP phone 81234 |
| 01:16.02 | TJNII | Okay |
| 01:16.05 | rkeene | No, it uses the 5 digit values |
| 01:16.20 | TJNII | All right, I think I'm getting it. |
| 01:16.49 | TJNII | You have extensions for all the sip phones, but not the extensions the go the the PSTN. |
| 01:17.13 | rkeene | Right (since I don't control the PSTN) |
| 01:17.28 | rkeene | My interface to the PSTN is a PRI card |
| 01:18.20 | TJNII | Okay, so for simplicity put all these in a context, like [siplocal] |
| 01:18.36 | TJNII | Then define another, say [default] |
| 01:18.46 | TJNII | include => siplocal and then do |
| 01:19.05 | TJNII | exten => _8XXXX,1,Dial(however you get to the PSTN) |
| 01:19.30 | TJNII | It will match your local extens first, and the pattern match will grab the rest and send it to the PSTN. |
| 01:19.44 | *** join/#asterisk BobLutz (n=stansmit@d60-65-93-136.col.wideopenwest.com) |
| 01:19.48 | BobLutz | Hello |
| 01:19.55 | TJNII | Just make sure asterisk gets to the pattern match AFTER your defined extens. |
| 01:20.09 | rkeene | But what will cause me dialing 81001 (where 81001 is a SIP account) to get directed to the SIP account ? |
| 01:20.49 | TJNII | Because your defined extensions are first |
| 01:21.14 | TJNII | so it will run your, already defined, exten => 81001 before ever getting to the pattern match. |
| 01:21.25 | rkeene | So in my extensions.conf I would have [siplocal] *with nothing in it* [default] exten => _8XXXX,1,... ? |
| 01:21.42 | TJNII | siplocal will contain all the sip extens you said you already made. |
| 01:21.45 | rkeene | (Since my goal is to avoid listing all of the extensions in the extensions.conf again, since they are already in the sip.conf) |
| 01:22.04 | rkeene | Right, my goal is to AVOID listing all the extensions in the sip.conf and the extensions.conf |
| 01:22.07 | TJNII | Well, sip.conf tells the system what phones are connected to it |
| 01:22.34 | TJNII | sip.conf is not part of the dialplan |
| 01:22.46 | rkeene | Right, I don't want to enumerate this in my dial pla |
| 01:22.51 | rkeene | dial plan, rather. |
| 01:22.55 | TJNII | You're going to have to |
| 01:23.02 | rkeene | Oh, okay |
| 01:23.10 | TJNII | If there is no rhyme or reason to which are SIP and which are PSTN |
| 01:23.48 | rkeene | I just thought there might be some way for the dial plan to examine the list of SIP addresses, for convience |
| 01:24.05 | TJNII | Not that I know of. |
| 01:24.16 | TJNII | You could probably make something that does, though. |
| 01:24.31 | TJNII | Well, are the sip addresses going to change often? |
| 01:24.35 | BobLutz | Oh boy |
| 01:24.47 | rkeene | No... but I can easily generate the dialplan from a script as I do the sip.conf :-P |
| 01:24.59 | TJNII | That would probably be easist. |
| 01:25.26 | TJNII | Make a script that creates extensions.siplocal.conf and include that file into your extensions.conf |
| 01:26.01 | rkeene | Can I include files from sip.conf, also ? |
| 01:26.17 | TJNII | I think so. |
| 01:26.55 | BobLutz | Wait - You can have an "include" in extensions.conf, but the file that is included can be dynamically created - without having to reload the dialplan? |
| 01:27.23 | rkeene | BobLutz, I assume you will have to reload the dial plan |
| 01:27.32 | TJNII | Yea, you do |
| 01:27.42 | BobLutz | Oh, I misunderstood what you guys were saying - that would of been extremely powerful though |
| 01:28.21 | rkeene | But reloading the dialplan isn't that hard |
| 01:28.22 | TJNII | There are dynamic dialplan solutions, I know nothing about them though. |
| 01:28.33 | rkeene | (From a script... that generates the dial plan :-P) |
| 01:28.49 | TJNII | Heheheh. There you go. |
| 01:28.56 | rkeene | (asterisk -r -x 'dialplan reload') |
| 01:30.19 | BobLutz | Yea... |
| 01:31.32 | rkeene | The list comes from an LDAP server, and gets sprinkled into an XML file for the phonebook's directory and the extension list :-P |
| 01:33.07 | TJNII | You could potentially roll something with agi, but since it is updating the sip.conf anyways, I don't think it would be worth the effort. |
| 01:33.18 | TJNII | I don't know any means to do a dynamic sip.conf. |
| 01:33.36 | *** part/#asterisk RoyK (n=roy@ip-77-55-149-91.dialup.ice.no) |
| 01:35.51 | JackEStorm | I have my sip.conf and queue stuff (along with cdr) in sql via realtime |
| 01:37.09 | rkeene | JackEStorm, I would consider it, but from what I've been told it wouldn't help (without "potentially something with agi") |
| 01:37.09 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 01:38.43 | TJNII | Realtime may do it, voip-info has an LDAP example |
| 01:40.43 | TJNII | None the less, you can probably do the scripted dialplan to get it working now, and then figure out if realtime will work better later. |
| 01:40.59 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 01:41.03 | zen-froglet | what would be the recommended app to use for receiving Faxes with Asterisk ? |
| 01:41.14 | TJNII | I haven't played with LDAP in a couple years, but when I did I thought it was kind of an uphill battle. |
| 01:41.16 | zen-froglet | other than rxfax and spandsp ? |
| 01:41.32 | kimosabe | if not mistaken asterisk cant do faxes yet |
| 01:41.36 | rkeene | Yeah, the scripted dial plan is trivial |
| 01:41.57 | *** join/#asterisk efort (n=efort@74-86-100-202.lx-vs.net) |
| 01:42.14 | TJNII | Well, maybe it won't do LDAP. I don't know. |
| 01:42.25 | zen-froglet | kimosabe: my previous Asterisk box would accept faxes and email to me |
| 01:42.50 | zen-froglet | perhaps not a specific function of Asterisk, but it ran within my system |
| 01:43.43 | *** join/#asterisk AndyGraybeal (n=AndyGray@128.177.27.78) [NETSPLIT VICTIM] |
| 01:47.31 | *** join/#asterisk youngproguru (n=root@cpe-76-180-239-199.buffalo.res.rr.com) |
| 01:50.40 | *** join/#asterisk RobH (n=RobH@69.18.84.191) |
| 01:54.51 | *** join/#asterisk Corydon76-dig (i=one@pdpc/supporter/bronze/Corydon76-home) [NETSPLIT VICTIM] |
| 01:54.51 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 01:54.58 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) [NETSPLIT VICTIM] |
| 01:56.12 | *** join/#asterisk BeeBuu (n=beebuu@218.13.81.79) |
| 01:57.37 | *** join/#asterisk Alpha_AI (n=Ben@210.11.97.57) |
| 01:57.40 | Alpha_AI | Hello there |
| 01:58.07 | BobLutz | Alpha_AI, Hey! |
| 01:59.10 | Alpha_AI | hey boblutz, how r ya? |
| 01:59.32 | BobLutz | Im thinking if i should modify app_swift.c or not |
| 01:59.40 | BobLutz | Alpha_AI, you know a lot of C? |
| 01:59.42 | Alpha_AI | oh yip |
| 01:59.47 | Alpha_AI | no i dont |
| 02:00.00 | Alpha_AI | mostly c |
| 02:00.01 | BobLutz | I know a little..I think enough to do what I need to do, but im scared! |
| 02:00.02 | Alpha_AI | oops |
| 02:00.04 | Alpha_AI | mostly delphi |
| 02:00.15 | BobLutz | I was reading something last night, Delphi is real fast, no? |
| 02:00.23 | Alpha_AI | it is pretty fast |
| 02:00.31 | Alpha_AI | fast enough for me |
| 02:00.39 | BobLutz | Faster than C (gcc) according to that page I read |
| 02:00.48 | Alpha_AI | oh yip |
| 02:00.59 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 02:01.54 | *** join/#asterisk zapa (n=hzavala@206.132.198.6) |
| 02:02.36 | Alpha_AI | BobLutz, do you know much about sip server? |
| 02:02.49 | BobLutz | :-/ , I only use sip for testing |
| 02:02.53 | BobLutz | What are you trying to do? |
| 02:03.00 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-1ef75968c4ff3bf7) |
| 02:03.34 | Alpha_AI | well i have to pay $635 a month if i get 128 simultaneous channels |
| 02:03.59 | BobLutz | for sip? |
| 02:04.05 | Alpha_AI | which i believe i need. So instead of paying someone to give me channels i thought i will try and give myself free channels |
| 02:04.07 | Alpha_AI | yeah for sip |
| 02:04.18 | drmessano | Free channels? |
| 02:04.53 | BobLutz | Alpha_AI, you can give yourself unlimited free channels (depends on your hardware) ... getting that out over the internet is something else though |
| 02:05.13 | drmessano | No.. |
| 02:05.19 | Alpha_AI | no according to the sip providers, they are the ones that provide the channels |
| 02:05.27 | *** join/#asterisk youngproguru (n=root@cpe-76-180-239-199.buffalo.res.rr.com) |
| 02:05.27 | drmessano | Getting them terminated with an ITSP is something else |
| 02:05.37 | BobLutz | Is what I meant ^ |
| 02:05.44 | drmessano | You can have unlimited free channels on the internet |
| 02:06.35 | Alpha_AI | lets say i got a did number, to get more people to call that one did number i will need more channels. they cost $5 per month with the group im looking at |
| 02:07.09 | drmessano | ok... |
| 02:07.16 | drmessano | So how do you propose to do that for free? |
| 02:09.50 | Alpha_AI | . |
| 02:10.10 | drmessano | ? |
| 02:10.15 | Alpha_AI | i dont know |
| 02:10.22 | drmessano | Sounds like a plan |
| 02:10.27 | Alpha_AI | im looking at a free open source sip server at the moment |
| 02:10.28 | drmessano | Let me know how it turns out |
| 02:10.35 | drmessano | Ok |
| 02:10.37 | drmessano | Thats great |
| 02:10.42 | drmessano | You have SIP |
| 02:10.45 | drmessano | Then what |
| 02:11.01 | TJNII | Step 1: Require hundreds of simultaneous PSTN lines. Step 2: ??? Step 3: Profit! |
| 02:11.10 | drmessano | FTW |
| 02:11.12 | TJNII | Step 2 is always a bitch. |
| 02:11.17 | BobLutz | LOL |
| 02:11.43 | drmessano | ~freelines |
| 02:11.44 | billytwowilly | step 2 is hire phone sex operators to man the hundreds of simultaneous PSTN lines |
| 02:12.01 | BobLutz | :-o |
| 02:12.05 | *** join/#asterisk seanbright-home (n=seanbrig@c-69-251-175-43.hsd1.md.comcast.net) |
| 02:12.08 | drmessano | ~freelines |
| 02:12.09 | jbot | Step 1: Require hundreds of simultaneous PSTN lines. Step 2: ??? Step 3: Profit! |
| 02:12.09 | TJNII | I like the cut of your jib. |
| 02:12.52 | drmessano | ~freelines |
| 02:12.54 | drmessano | ~free lines |
| 02:12.55 | jbot | Step 1: Require hundreds of simultaneous PSTN lines. Step 2: ??? Step 3: Profit! |
| 02:12.56 | drmessano | better |
| 02:12.59 | drmessano | needed the space |
| 02:13.52 | drmessano | "ATT wants $500 a month for a T1" "I am trying to work out how to do it for free" "?????" |
| 02:14.13 | BobLutz | drmessano, Hey, come on, we dont want no black-on-black in here |
| 02:14.29 | drmessano | black on black? |
| 02:14.39 | BobLutz | um.. |
| 02:15.22 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 02:15.24 | billytwowilly | BobLutz only goes for the inter-racial shenanigans.. |
| 02:16.10 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.or.comcast.net) |
| 02:16.15 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 02:17.01 | drmessano | I see |
| 02:17.13 | BobLutz | ast_exists_extension() is deprecated? |
| 02:17.34 | drmessano | Where did you see that? |
| 02:17.41 | BobLutz | http://lists.digium.com/pipermail/asterisk-dev/2008-January/031466.html |
| 02:18.10 | ZPertee | how do I find an asterisk consultant in my area? |
| 02:19.05 | droops | where is your area? |
| 02:19.27 | plik | haha, voip manufacturer Snom moved offices last week and their phone lines still aren't connected! |
| 02:19.41 | BobLutz | 0wn3d? |
| 02:19.50 | drmessano | 0wn3d? |
| 02:19.54 | ZPertee | droops: North East Ohio |
| 02:20.07 | drmessano | pwn3d <-- the non-lame spelling |
| 02:20.11 | plik | nsh, just a typical teecom cock-up |
| 02:20.47 | droops | ZPertee, http://www.voip-info.org/wiki/view/Asterisk+consultants+USA#OHIO |
| 02:20.58 | SteveTotaro | they should have prepared a little better for a move |
| 02:21.23 | SteveTotaro | rent the telco closet for another month and send the calls via sip |
| 02:22.02 | SteveTotaro | or just forward the calls, but don't move until the circuits are in place to forward to |
| 02:22.45 | SteveTotaro | snom is nice but too european looking in style for my tastes |
| 02:23.00 | plik | yeah, we all know how it should be done, or could be resolved - you'd think theyd have a clue too :) |
| 02:23.47 | SteveTotaro | well they are a phone maker, so it doesn't really surprise me |
| 02:24.07 | SteveTotaro | they don't deal with the telcos all the time like some of us |
| 02:24.59 | drmessano | Where is Snom located? |
| 02:25.10 | SteveTotaro | germany i believe |
| 02:25.10 | plik | germany somewhere |
| 02:25.11 | TJNII | Berlin, I believe |
| 02:25.26 | SteveTotaro | tear that wall down |
| 02:25.40 | BobLutz | ha |
| 02:25.44 | droops | i think they did |
| 02:25.49 | SteveTotaro | how are you doin DRM? |
| 02:25.50 | plik | berlin, yes... TJNII Wins!! |
| 02:26.02 | SteveTotaro | I was correct too |
| 02:26.11 | drmessano | Ah |
| 02:26.12 | SteveTotaro | just a little broader in terms |
| 02:26.20 | drmessano | Spreken Ze Deutsch? |
| 02:26.31 | drmessano | Im good, Steve |
| 02:26.35 | drmessano | Been working my new job |
| 02:26.55 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-61864d40278afa84) |
| 02:26.59 | efort | how can I tell if asterisk is registering with my ITSP? |
| 02:27.04 | SteveTotaro | falling in like a well oiled cog? |
| 02:27.24 | drmessano | Yeah.. and I made twice as much this week as I did last week working for my old company |
| 02:27.25 | SteveTotaro | sip show registry |
| 02:27.27 | drmessano | How is that? lol |
| 02:27.37 | SteveTotaro | or iax show reg |
| 02:28.28 | SteveTotaro | money is one thing, but is the fit right do you think? |
| 02:28.39 | efort | SteveTotaro: thanks, it's a sip trunk |
| 02:29.05 | SteveTotaro | np |
| 02:30.15 | drmessano | I think it is.. Seems to be a good bunch.. They all work like a big family.. and I see to be right in there |
| 02:30.52 | SteveTotaro | can you slip in your clandestine asterisk agenda? |
| 02:30.54 | *** join/#asterisk Asterlinktechy (n=chito@125.60.231.8) |
| 02:31.15 | SteveTotaro | soon they will all be addicted |
| 02:31.32 | drmessano | I am going to |
| 02:31.51 | drmessano | I plan to wait a few weeks and push the issue |
| 02:31.53 | efort | what would keep it from registering? what needs done to get * to register? it's static ip on both ends. using Vitelity which I've been happy with so far |
| 02:33.00 | *** join/#asterisk DonAlex (n=DonAlex@glanforn.demon.co.uk) |
| 02:33.13 | DonAlex | Evening all :) |
| 02:33.55 | SteveTotaro | so vitelity worked and now it does not? |
| 02:34.47 | *** join/#asterisk macaruchi (n=macaruch@tdev253-190.codetel.net.do) |
| 02:34.52 | macaruchi | Hi! |
| 02:35.20 | SteveTotaro | has anyone used druid? is it better than the other free GUIs in your opinion? |
| 02:35.24 | macaruchi | <PROTECTED> |
| 02:35.39 | Asterlinktechy | I use vitelity for long time and its works well on m box :) |
| 02:35.47 | BobLutz | drmessano, you like C? |
| 02:35.59 | Alpha_AI | it seems that IAX can deliver more channels than SIP can |
| 02:36.19 | macaruchi | i use install_prereq and everything fine |
| 02:36.33 | SteveTotaro | alpha, i think you are going to run into audio issues with that iax attitude |
| 02:37.13 | drmessano | BobLutz: C is not my favorite.. I like Thiamin and E is a close second.. Zinc is up there too |
| 02:37.16 | efort | no I have yet to get the trunk configured. they tell me I'm not registering and that's what I see also but I've been happy with them so far because their support has been good and I can generally get someone on the phone if I really need to |
| 02:37.26 | BobLutz | LOL |
| 02:37.33 | drmessano | IAX can not deliver more channels than SIP |
| 02:37.42 | SteveTotaro | i have used iax.cc/vitelity for years, i just use their forward function to my cell nowdays |
| 02:38.12 | SteveTotaro | sip debug |
| 02:38.19 | SteveTotaro | see what is going on |
| 02:38.32 | SteveTotaro | and then pastebin |
| 02:38.36 | SteveTotaro | ~pb |
| 02:38.36 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 02:40.26 | DonAlex | Hey peeps... would anyone like to make a guess why when making dialplan rules via AsteriskNow interface my config is just ignoring them? It saves it to extensions.conf and reloads but just ignores anything I put. |
| 02:40.50 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-b3102908aebe9ba9) |
| 02:41.06 | efort | sip debug shows only "remote unix connection" then "remote unix connection disconnected" |
| 02:44.21 | efort | I don't think that's related though, I see that all the time like being scanned |
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| 02:46.24 | BobLutz | ast_exists_extension() --> Where could I find such a method in doxygen? |
| 02:46.50 | droops | DonAlex, you might try #asterisknow or #asterisk-gui |
| 02:47.38 | BobLutz | Nevermind ---> pbx.c |
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| 02:50.49 | DonAlex | droops: Well not 100% sure it is related to the gui or just me mucking up the file somehow ;) |
| 02:51.04 | DonAlex | droops: but I will anyway , thanks |
| 02:51.55 | Alpha_AI | is iax ready for production use yet? |
| 02:52.46 | drmessano | I use it in production |
| 02:52.50 | drmessano | So do others |
| 02:54.42 | DonAlex | Ahhh and another thing to pick someones brains about but just what IS the difference between q931 and 931e ? |
| 02:55.29 | Qwell | an e |
| 02:56.22 | drmessano | q in the front, e on the back |
| 02:56.29 | drmessano | oh |
| 02:56.44 | drmessano | Be all technical about it, Qwell |
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| 03:00.09 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:01.30 | *** part/#asterisk youngproguru (n=root@cpe-76-180-239-199.buffalo.res.rr.com) |
| 03:06.05 | znoG_ | does anyone know how I could possibly lookup the IP address of the originating call? the idea is to only allow outbound calls if the source IP of the IAX/SIP user is within the local subnet. |
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| 03:22.03 | TJNII | znoG_: wouldn't it be easier to do that on a per-user basis with contexts? |
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| 03:37.54 | TJNII | ~free lines |
| 03:37.54 | jbot | Step 1: Require hundreds of simultaneous PSTN lines. Step 2: ??? Step 3: Profit! |
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| 04:00.34 | Alpha_AI | Hello |
| 04:01.24 | Alpha_AI | Im installing Asterix now through VMserver and its asking me to create a new partitition saying this will cause the loss of ALL DATA on this drive |
| 04:01.24 | Qwell | what is asterix? |
| 04:01.25 | Alpha_AI | does that mean i will lose everything on the drive or just everything in the VM? |
| 04:01.29 | Alpha_AI | asterisk |
| 04:01.47 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 04:02.06 | Alpha_AI | i hate it how someone says 'what is asterix?' |
| 04:02.10 | Alpha_AI | ya know what im talking about |
| 04:03.16 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 04:03.16 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 04:03.16 | Qwell | if you can't put forth a little effort to spell the name correctly, why would anybody want to put forth effort to help you? |
| 04:03.36 | lmadsen | Qwell: but I didn't even ask my question yet |
| 04:03.43 | Qwell | lmadsen: good! |
| 04:03.52 | lmadsen | question time! :) |
| 04:03.59 | Qwell | only if you spell it right :p |
| 04:04.10 | lmadsen | what causes Asterisk to write a disposition of ANSWERED when looking in a SIP trace? |
| 04:04.29 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-6da9f8bc0f383707) |
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| 04:13.38 | russellb | lmadsen: the other side answering the call ... 200 OK |
| 04:13.45 | lmadsen | russellb: thx! |
| 04:13.47 | russellb | np |
| 04:13.55 | lmadsen | what should I expect... just 180 Ringing? |
| 04:14.30 | russellb | maybe :-p |
| 04:14.31 | drmessano | Hmm |
| 04:17.40 | lmadsen | hrmm... I think a reinvite is screwing up my CDRs |
| 04:28.22 | lmadsen | codefreeze: long shot... but... ping? |
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| 04:57.47 | b11d` | hey chaps |
| 04:58.36 | b11d` | . |
| 04:58.41 | b11d` | whoops |
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| 05:26.49 | TripleX1 | anyone up ? |
| 05:27.01 | BBHoss | depends |
| 05:27.17 | TripleX1 | hehe |
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| 05:39.55 | TripleX1 | which v you running ? |
| 05:49.22 | *** join/#asterisk illumini (n=michael@zhaz.scanningsystems.com.au) |
| 05:50.03 | illumini | Hello, I'm a first time user setting up AsteriskNow (latest). When I call another extension it goes straight to voicemail, what have I done wrong? |
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| 06:31.31 | *** mode/#asterisk [+o codefreeze] by irc.freenode.net |
| 06:33.13 | illumini | where can I download sample Asterisk config setups, they're mentioned on the website but no links provided |
| 06:33.39 | TripleX1 | make samples |
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| 06:50.11 | mattman99 | illumini - re your voicemail problem, sounds like the phone you are calling is not registered |
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| 07:35.03 | pascals | Goodmorning. |
| 07:36.00 | pascals | I have a quadBRI card and can accept incoming calls, but outgoing calls give the following error: chan_zap.c:8800 zt_pri_error: 4 Write to 30 failed: Unknown error 500 |
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| 07:45.07 | awk | hmm, any sugestions as I can determine if 30 channels on a PRI are used.. |
| 07:45.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:45.13 | awk | anyone have a script ? |
| 07:45.24 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:45.25 | pascals | Found my problem |
| 07:46.10 | pascals | zapata.conf listed too many channels available. Asterisk 1.0 didn't mind, 1.2 doesn't seem to handle that the same way |
| 07:47.07 | pascals | FYI: I had the gsm optimization bug on my Suse 10.3 box with gcc 4.2.1, solved it by setting -O6 to -O2 in codecs/gsm/Makefile. |
| 07:49.07 | awk | I need some script that will tell me when i've used up all available channels on the PRI... i'm passing everything to a quintum on sip trunk 8000 and 7000 so I could use some like, asterisk -rx "show channeks" | grep 8000 and pass that to a placement and 7000 and then get the sum of X and Y... and pass to Z and if Z = 30 then sendmail |
| 07:51.54 | *** join/#asterisk oej (n=olle@62.97.243.70) |
| 07:52.37 | rkeene | Anyone have any thoughts on the voicemail password being the same as the SIP password ? |
| 07:54.35 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 07:55.17 | cmantito | hmm ... sip show peers shows ~95% of the peers "UNREACHABLE"...can you say upstream transit problems? ;p |
| 07:55.38 | awk | rkeene depends on what type of security or what type of pols are in place... |
| 07:55.50 | awk | most of my clients share the same... |
| 07:56.07 | awk | cmantito how is the peers connected |
| 07:56.17 | cmantito | over teh interwebs ;p |
| 07:56.25 | cmantito | from various locatiosn |
| 07:56.26 | awk | ok and half of them are down? |
| 07:56.29 | awk | err 95%? |
| 07:56.34 | cmantito | actually, specifically, all the comcast users are down |
| 07:56.35 | cmantito | *cough* |
| 07:56.41 | cmantito | ...and a few Covads users |
| 07:57.06 | awk | well its simple can you see reg attempts, if not can you ping their ip's or g/w? |
| 07:57.27 | awk | if I was you I would use something like vqmanager and monitor endpoints... |
| 07:57.46 | cmantito | I can see them re/deregistering, and I'd bet money any second now I'll get an SMS from Zabbix letting me know of transit problems somewhere |
| 07:57.57 | cmantito | err, not deregistering |
| 07:58.02 | cmantito | but, become unregistered ;p |
| 07:58.08 | awk | unreachable :) |
| 07:58.13 | awk | after a timeout of 2000ms, or something |
| 07:58.19 | cmantito | yeah, that's the word |
| 07:58.27 | cmantito | sorry, 4 am and I'm brought outta bed for this ;p |
| 07:58.42 | awk | go back to bed, nothing you can do.. send the NOC a mail and go sleep |
| 07:59.12 | cmantito | haha, and the Jabber server's AIM transport just died |
| 07:59.23 | cmantito | and the winner is ... Failur(3)! |
| 08:00.09 | cmantito | wow, 50% packet loss from here. Yayyy.. |
| 08:00.14 | cmantito | gnight lol |
| 08:01.52 | awk | just run a mtr on your endpoint |
| 08:02.00 | awk | and when you wake up see what type of packet los you acing |
| 08:02.01 | awk | facing |
| 08:02.08 | cmantito | that's what I'm doing |
| 08:02.14 | cmantito | this datacentre has been nothing but problems. |
| 08:03.17 | awk | heh, :) looks like its time to change.. |
| 08:03.31 | cmantito | we're certainly trying ^_^ |
| 08:03.43 | *** join/#asterisk tengulre11 (n=tengulre@124.42.50.9) |
| 08:03.43 | awk | just make sure you don't make the same mistake next time! |
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| 08:03.56 | cmantito | yeah. |
| 08:04.01 | cmantito | that's for damned sure. |
| 08:04.12 | tengulre11 | Oh,yes !@#$#^^ |
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| 08:19.22 | DonAlex | Anyone any idea why asterisk would ignore my dial rules. They are these : [ Context 'numberplan-custom-1' created by 'pbx_config' ] |
| 08:19.22 | DonAlex | <PROTECTED> |
| 08:19.22 | DonAlex | <PROTECTED> |
| 08:19.22 | DonAlex | <PROTECTED> |
| 08:19.22 | DonAlex | <PROTECTED> |
| 08:19.56 | DonAlex | Just ignores them completely and thinks any number dialled is an extension? |
| 08:21.20 | DonAlex | It is very frustrating. There is only one trunk set up and it is pointing to the right zap interface. |
| 08:27.20 | cmantito | and I'm going to bed. Happy piday guys. |
| 08:28.30 | DonAlex | Ok moving on then.. |
| 08:28.55 | DonAlex | anyone can fill in the details of the difference between q931 and q931e ? |
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| 08:43.41 | FabiOne | good morning |
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| 09:20.05 | Chris-NB | hi |
| 09:20.21 | Chris-NB | anyone tried to setup a mail to fax gateway with asterisk? |
| 09:20.56 | Chris-NB | receive an mail, convert to fax/tiff and send a fax via ISDN |
| 09:23.18 | FabiOne | i've a problem with outgoing call through a HFC-S PCI ISDN card |
| 09:23.29 | FabiOne | i receive and make call |
| 09:23.57 | FabiOne | but only 1 at a time |
| 09:24.14 | rkeene | awk, The SIP clients are restricted to a private subnet |
| 09:25.33 | FabiOne | i think it's a conf problem in zapata.conf or dialplan |
| 09:25.57 | FabiOne | i use dial(zap/1/${EXTEN}) |
| 09:26.12 | FabiOne | i think the "1" is wrong.. |
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| 09:36.25 | eric2 | Chris-NB I've tried.. but have yet to succeed |
| 09:36.44 | Chris-NB | eric2, so no stable/reliable solution? |
| 09:36.53 | eric2 | not with 711 |
| 09:37.10 | eric2 | any network jittering and the fax get's messed up |
| 09:37.29 | eric2 | people claim to be able to do it on an internal LAN but over the internet is unreliable |
| 09:37.42 | eric2 | best way is with t.38 from what I've read |
| 09:37.53 | Chris-NB | I try to send a fax via ISDN, not SIP/IAX/..... |
| 09:38.22 | Chris-NB | with tx_fax, hylafax, spandsp, asterfax ... or whatever |
| 09:38.30 | eric2 | I'm all SIP |
| 09:38.35 | eric2 | look at callweaver |
| 09:38.47 | eric2 | another software piece that might help |
| 09:39.38 | Chris-NB | okay, thanks! I'll try |
| 09:39.42 | Chris-NB | or look |
| 09:40.40 | eric2 | If you do get som'n going, let me know |
| 09:41.01 | eric2 | but then again, I don't have any ISDN stuff here |
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| 09:54.55 | roffer | Hello, i have a question, i run Asterisk 1.4.4 and got a problem vith making a connection out with my sip provider. i can make internal calls and answer incoming calls from my sip provider but not make any calls out. i get this error. chan_sip.c:11848 handle_response_invite: Failed to authenticate on INVITE to '"unknown" anyone that can help me with this problem ? |
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| 10:15.38 | roffer | Hello, i have a question, i run Asterisk 1.4.4 and got a problem vith making a connection out with my sip provider. i can make internal calls and answer incoming calls from my sip provider but not make any calls out. i get this error. chan_sip.c:11848 handle_response_invite: Failed to authenticate on INVITE to '"unknown" anyone that can help me with this problem ? |
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| 10:43.02 | DarKnesS_WolF | anyone having problems for snom behind NAT ? |
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| 11:03.19 | Tili | has anyone used asterisk in hong kong? |
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| 11:09.27 | *** join/#asterisk kodomo (n=memyself@emu.net.informatik.tu-muenchen.de) |
| 11:09.31 | kodomo | hi folks |
| 11:09.40 | *** join/#asterisk duckz (n=duckz@81-180-102-217.etth.opensys.ro) |
| 11:10.18 | kodomo | did he update to the newest zaptel/asterisk versions and now I'm getting loads of lost interrupts, when I load ztdummy |
| 11:10.53 | kodomo | any pointers appreciated (currently heavy googling ;) ) |
| 11:16.10 | *** join/#asterisk Worf (n=worf@84.119.67.68) |
| 11:17.56 | nixguy | if i wanna have som statistics for calls from asterisk |
| 11:18.16 | nixguy | anyone have any tips on how to get me started?, the google search words are to wide to give any good results.. |
| 11:20.17 | *** join/#asterisk Rienzilla (i=rien@sinas.rename-it.nl) |
| 11:20.27 | Rienzilla | wooyah this channel is large :) |
| 11:21.34 | Worf | nixguy: hmm ... /var/log/asterisk/cdr-csv/ contains a spreadsheet with your calls ... i guess you have to parse that ... |
| 11:21.56 | nixguy | Worf: k thnx |
| 11:22.43 | JT | finally |
| 11:22.50 | JT | after 3 months, this pri is lit |
| 11:22.51 | JT | :) |
| 11:23.46 | nixguy | any recomendations? |
| 11:23.55 | nixguy | what are you guys using to parse those logsfiles? :) |
| 11:24.42 | Worf | hehe - i could dump my thoughs here now, but i'm a asterisk noob. there should be some solutions allready :) |
| 11:24.59 | Rienzilla | Hey everyone, I have a question which I cannot find on google. I have a frontoffice here with 4 people who are responsible for both answering phonecalls and answering real-life questions at the desk. Therefore I am looking for a queueing solution for asterisk which will not immediately connect callers to an agent's handset, but rather have the handset ring for people that are present (So one of the employees can pick the call up). I do need other fea |
| 11:27.40 | DrAk0 | Rienzilla, what? |
| 11:27.53 | Worf | Rienzilla: from what i found out you can log in with the same account multiple times, and all phones will ring simultaneousely then. and the first one who picks up, wins... |
| 11:29.19 | Rienzilla | but if I log in, i'm permanently off hook, no? |
| 11:29.38 | Rienzilla | (I found a callbacklogin thing, but that is deprecated according to documentation) |
| 11:30.29 | DrAk0 | Rienzilla, i don't get your point |
| 11:30.41 | DrAk0 | Rienzilla, you don't want the call to ring on all desks? |
| 11:30.45 | Rienzilla | no |
| 11:30.54 | DrAk0 | Rienzilla, in which one you want it to ring? |
| 11:30.56 | Rienzilla | I only want the call to ring on the desks of users that are present |
| 11:31.05 | DrAk0 | Rienzilla, ok |
| 11:31.07 | DrAk0 | thats easy |
| 11:31.35 | Rienzilla | (I have snom 360 sip phones and asterisk 1.4.9 pbx, if that helps) |
| 11:31.50 | DrAk0 | Rienzilla, agentcallbacklogin |
| 11:32.24 | Rienzilla | DrAk0: yes I saw that on voip-info but it is deprecated according to the docs there |
| 11:32.36 | DrAk0 | Rienzilla, i use it |
| 11:32.40 | Rienzilla | ok |
| 11:32.40 | DrAk0 | on asterisk 1.4.x |
| 11:32.46 | DrAk0 | and works pretty good |
| 11:32.51 | Rienzilla | ok good |
| 11:32.56 | Rienzilla | I'll try that then |
| 11:33.04 | Worf | time for my noob question ... i set up asterisk on my router but i think i messed something up badly, because when dialing out nobody can hear me. it works when being called and it works when configuring the phone to not log in on my own asterisk but on sipgate directly ... any hints where to start looking? firewall issue? misconfigured asterisk? how do i actually track down such a problem? |
| 11:33.33 | Rienzilla | and one other thing. Is there an easier way for users to park calls and pick them up than to manually transfer them to an extension? |
| 11:33.58 | DrAk0 | Rienzilla, hold button on the phone |
| 11:34.14 | Rienzilla | yes but the users here want to be able to pick up each other's call |
| 11:34.34 | tzanger | Rienzilla: program a speed dial |
| 11:34.39 | Rienzilla | right now they can press hold, and then someone else can pick up that line by simply choosing the line x button on their handset |
| 11:35.00 | tzanger | Rienzilla: that's because you've got a key system |
| 11:35.10 | Rienzilla | tzanger: the speed dial is one thing, but it would be nice if the handsets indicated that there were calls parked on a specific extension |
| 11:35.10 | tzanger | you want shared line appearances, which are (I think) still beta |
| 11:35.24 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 11:35.41 | tzanger | Rienzilla: you can do that with my long-ago merged patch that allows you to get the parked extension with ${PARKEDAT} |
| 11:36.06 | Rienzilla | ok |
| 11:36.10 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:36.15 | Rienzilla | and how do you announce that to handsets? |
| 11:36.26 | Rienzilla | or is that handset specific or impossible :) |
| 11:36.53 | tzanger | that's up to your specific implementation |
| 11:37.19 | DrAk0 | that sounds not simple:P |
| 11:37.32 | Rienzilla | bah :) |
| 11:37.39 | DrAk0 | im wondering if there is any 3 way call easy implementation |
| 11:37.50 | Rienzilla | well I'll give it a shot :) |
| 11:37.54 | DrAk0 | and a good CTI for a call center |
| 11:38.14 | Rienzilla | I might be able to program my phones to poll where calls are parked regularly |
| 11:38.22 | Rienzilla | or something like that |
| 11:38.38 | DrAk0 | Rienzilla, which phones are you using |
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| 11:50.47 | Worf | hmm ... i suspect my problem has to do with the various possible nat settings in asterisk ... |
| 12:01.59 | JT | ~SIPNAT |
| 12:02.00 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
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| 12:05.44 | roffer | Hello, i have a question, i run Asterisk 1.4.4 and got a problem vith making a connection out with my sip provider. i can make internal calls and answer incoming calls from my sip provider but not make any calls out. i get this error. chan_sip.c:11848 handle_response_invite: Failed to authenticate on INVITE to '"unknown" anyone that can help me with this problem ? |
| 12:06.56 | roffer | and im allso confued, i used asterisk gui to setup sip account, and it puts it under user.conf, all other doc i can find they put it in sip.conf ? |
| 12:06.59 | Worf | JT: thanks ... |
| 12:09.19 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 12:09.22 | casix | hello |
| 12:09.35 | casix | I'm making a call back system with an ivr |
| 12:11.15 | casix | when the .call file is readed it calls to the destination. After call (when is ringing) asterisk doesn't wait until it is answered. Asterisk continues executing. Can I make asterisk wait until it is answered?? |
| 12:14.09 | Rienzilla | DrAk0: great. callbacklogin works like a charm |
| 12:14.26 | Rienzilla | only annoying thing is that my snom's register calls which are answered by another handset as a missed call |
| 12:19.34 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:23.27 | DrAk0 | Rienzilla, it happens even with soft phones |
| 12:23.53 | *** join/#asterisk yang (i=yang@static-ip-62-75-255-125.inaddr.intergenia.de) |
| 12:24.29 | yang | If someone wants to dial +XXXX number, can I simply add that to the dialplan - exten => _+.,1,Dial(SIP/gsm/${EXTEN}) |
| 12:26.41 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
| 12:26.58 | DonAlex | Guys.. |
| 12:27.08 | DonAlex | why is it asterisk is ignoring my dialrules? |
| 12:27.21 | DonAlex | keeps trying to make every number dialled and extension? |
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| 12:28.04 | *** join/#asterisk pithen (n=pithen@mail.graphlogic.com) |
| 12:28.23 | pithen | is the xorcom guy here? (sorry, can't remember your nick) |
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| 12:36.05 | tzanger | tzafrir I believe works for xorcom |
| 12:36.34 | DarKnesS_WolF | pithen: what is with ur xorcom ? |
| 12:38.35 | DonAlex | Yes h does but not here atm.. |
| 12:38.40 | DonAlex | what seems to eb the probelm ? |
| 12:38.48 | yang | pithen: tzafrir |
| 12:39.03 | lirakis | Does any one know how asterisk generates the "tag=" for To: responses? Is it some type of hash on a part of the invite? |
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| 12:51.12 | pithen | thanks guys- actually i think I just got the answer I was looking for from one of their vendors |
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| 12:54.19 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 13:03.41 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 13:08.00 | tzanger | coppice: oh hai, I solved that weird 3rd harmonic problem, although I still dont' understand why it was consistently a 3rd harmonic |
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| 13:08.20 | coppice | 3rd rate, of course |
| 13:08.51 | tzanger | coppice: the 32 timeslots coming back were consistently missing the LSB |
| 13:09.20 | Lsodi | Hi, I would like to record calls on asterisk server, can someone recommend software for recording? |
| 13:09.25 | *** join/#asterisk SteveTotaro (n=root@pool-71-166-110-9.bltmmd.east.verizon.net) |
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| 13:09.57 | coppice | is this ulaw or alaw? |
| 13:10.25 | tzanger | coppice: alaw |
| 13:10.26 | SteveTotaro | anyone try druid yet? i have tried so many GUIs, I am curious if this is worth the time and effort of testing? |
| 13:10.41 | tzanger | coppice: basically I send to the codec and it sends back one bit time late... for all 32 timeslots |
| 13:11.01 | tzanger | I was rotating the bits *in* each timeslot, instead of rotating the entire stream |
| 13:11.35 | coppice | missing the LSB is normal on robbed bit T1s, and it just reduces the quality a little. I'm not sure if alaw coding falls apart. alaw does something tricky with the first 2 segments of the pseudo-log curve |
| 13:11.37 | tzanger | so it was 0765432107654321076543210, and I was rotating it per-byte instead of per-stream :-) |
| 13:11.47 | *** join/#asterisk svenna_ (n=svenna@p548D35CF.dip0.t-ipconnect.de) |
| 13:11.49 | tzanger | yeah I read a lot of alaw theory |
| 13:12.02 | tzanger | they linearize the first bit of the code |
| 13:12.44 | tzanger | and they invert the even bits, as you described :-) |
| 13:14.02 | tzanger | after they fixed the fpga, audio I got abck from the codec on the far end of the tdm bus was clean :-) |
| 13:14.25 | coppice | the inverting the even bits is really odd. they need inverting, but it should have been an aspect of the comms channel, and not the codec |
| 13:14.27 | tzanger | the fpga sees it as a stream of bits, whereas the blackfin sees a stream of 8 bit words |
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| 13:15.00 | coppice | blackfins seem to be flavuor of the month :-) |
| 13:15.04 | tzanger | yeah that blew me away when I read that, but you explanation was very good |
| 13:15.44 | tzanger | I like 'em |
| 13:16.11 | tzanger | If I spent the time I'm sure I could have bit-rotated the stream nicely or played with the dma settings to delay the bit correctly, but fuck it, there's an fpga guy, make him work a little :-) |
| 13:16.29 | tzanger | I already had to deinterlace the damned bit stream |
| 13:16.34 | tzanger | gotta love old hardware |
| 13:16.53 | coppice | they are the only really successful attempt to made a general purpose + DSP core that does both jobs well. however, people complain that it looks like ADI ran out of cash when Intel pulled out, and the thing never got finished |
| 13:17.03 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
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| 13:17.28 | tzanger | OMAP was interesting |
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| 13:18.06 | coppice | OMAP is a general purpose core + a DSP core. that's the route a lot have taken. the blackfin is the only successful combined core |
| 13:18.58 | tzanger | yes... what I don't like about OMAP is the dual cores. it was interesting, but looks like a bigger pain in the ass than it's worth |
| 13:19.16 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:19.17 | tzanger | trance first thing in the morning... nice |
| 13:19.33 | Katty | morning! |
| 13:19.43 | tzanger | woot |
| 13:19.55 | Katty | tzanger: quick! tell me if you can record a zap line. |
| 13:20.01 | coppice | there is good and bad. you can run Linux on the ARM, while the DSP core is not handcapped by the timing vagary of linux |
| 13:20.03 | Katty | tzanger: from start to finish of the call! |
| 13:20.33 | tzanger | Katty: yes of course you can |
| 13:20.42 | tzanger | coppice: indeed |
| 13:20.52 | tzanger | coppice: actually I've been buried in realtime all week |
| 13:20.54 | Katty | tzanger: is it record, monitor, or mixmonitor? |
| 13:21.04 | tzanger | monitor or mixmonitor is what I use |
| 13:21.07 | coppice | hm. it looks like the UK courts have finally fallen into line with the rest of .eu over patentability |
| 13:21.41 | tzanger | actually had a design specc'd out that recorded every single call, internal or external, catalogued it, converted it and archived it for sbarnes-oaxley or however you spell it |
| 13:22.14 | tzanger | coppice: on x86 anyway I was able to see max 32us latency on a shitty via c7, throwing everything short of a kitchen sync at it |
| 13:22.33 | coppice | sarbanes-oxtongue |
| 13:23.21 | coppice | at the interrupt level, maybe. not at the apps level. also, that latency is based on big assumptions about other peripherals |
| 13:23.32 | *** join/#asterisk BobLutz (n=miles@d60-65-93-136.col.wideopenwest.com) |
| 13:24.08 | tzanger | coppice: that is at the apps level |
| 13:24.38 | BobLutz | Morning all |
| 13:24.38 | tzanger | that's a userspace app running latency tests with network, interrupts and flat-out CPU |
| 13:26.01 | tzanger | 1.2GHz Via C7 |
| 13:27.56 | coppice | you are imposing some severe constraints on the machine's activities if it can always respond so crisply |
| 13:28.05 | Katty | tzanger: can i pastebin some stuff and have you give it a quick look see? |
| 13:28.11 | Katty | tzanger: in regards to mixmonitor |
| 13:28.11 | tzanger | I can try |
| 13:28.37 | tzanger | coppice: naturally, it's an embedded system, but I'm not doing anything overy weird to it |
| 13:28.45 | tzanger | obviously zero power saving |
| 13:28.57 | tzanger | (although I have a hard time calling any x86 "embedded" :-) |
| 13:28.58 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:29.10 | BobLutz | lol |
| 13:29.25 | coppice | geode? |
| 13:30.06 | BobLutz | Qwell: Should I not ask questions in #asterisk-dev ? |
| 13:30.23 | *** join/#asterisk svenna_ (n=svenna@p548D35CF.dip0.t-ipconnect.de) |
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| 13:30.23 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:31.06 | tzanger | even geode I don't refer to as embedded |
| 13:31.08 | tzanger | too PC like |
| 13:31.30 | DonAlex | Awww crap.. |
| 13:31.35 | DonAlex | now why is this happening.. |
| 13:31.36 | DonAlex | handle_request_invite: Call from '' to extension |
| 13:31.42 | *** join/#asterisk atb_ (n=d@196.25.36.129) |
| 13:31.53 | DonAlex | with every bloody number.. |
| 13:32.06 | DonAlex | what macro got deleted or something? |
| 13:33.34 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 13:33.56 | atb_ | hi there, i know it aint the right place but possibly sombody can assist as ive been searching around for the past two days and cannot fix such a smalle problem, anybody here had any experience with a2billing integration and care to assist ? |
| 13:34.49 | Katty | tzanger: [from-pstn-recorded] |
| 13:35.15 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:36.09 | Katty | tzanger: http://angela.sleekgeek.org/2008/03/14/record-incoming-zap-line/ |
| 13:36.18 | tzanger | hahah sleekgeek |
| 13:36.33 | Katty | yes, that's our website. |
| 13:36.44 | Katty | tzanger: that's what i've got in mind. |
| 13:37.28 | tzanger | that looks about right. I would probably sanitize the caller id first, checking for empty or invalid and replacing iwth 'unknown' |
| 13:37.49 | tzanger | also you're mixing , and | but it's not a federal crime, only a misdemeanor |
| 13:37.57 | yang | If someone wants to dial +XXXX number, can I simply add that to the dialplan - exten => _+.,1,Dial(SIP/gsm/${EXTEN}) |
| 13:38.55 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
| 13:39.07 | ZaVoid | morning guys |
| 13:39.10 | Katty | tzanger: good idea. |
| 13:39.14 | ZaVoid | i got a strange thing herei can't get right |
| 13:39.24 | Katty | tzanger: why is mixing a misdemeanor? |
| 13:39.37 | ZaVoid | i'm trying to dial out... and do exten => s,5,Background("beep") after the call is answered |
| 13:39.39 | Katty | tzanger: i thought if you told someone first... :/ |
| 13:39.45 | ZaVoid | but it doesn't always play the beep at the right time |
| 13:39.57 | ZaVoid | i'm guessing i'm using the wrong command to wait for supervision |
| 13:40.46 | De_Mon | yang _XXXX would be the better way to handle 4 digit extensions _+ is just plain dangerous |
| 13:41.11 | yang | De_Mon: it can be any amount of digits I made it _+. |
| 13:41.26 | tzanger | Katty: it's just inconsistent |
| 13:41.32 | Rienzilla | hmm |
| 13:41.34 | tzanger | like telling someone you like them but then ignoring them |
| 13:41.39 | tzanger | drives people batty :-) |
| 13:41.41 | Rienzilla | can you detect in your dialplan whether a queue has agents logged in? |
| 13:41.53 | *** join/#asterisk noneo (n=artur@82.152.82.149) |
| 13:42.05 | De_Mon | yeah, I just realized + isn't a valid pattern, still half a sleep I spose. |
| 13:42.25 | yang | De_Mon: its not valid? |
| 13:43.11 | yang | De_Mon: any idea how can he otherwise dial the number from his mobile phone, which aere enterred as +XXXXXX |
| 13:44.01 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 13:44.34 | ZaVoid | its as if the BEEP never plays sometimes |
| 13:44.48 | hmmhesays | ~beep |
| 13:44.48 | jbot | beep is probably the protocol formally known as bxxp |
| 13:45.38 | BobLutz | In app_read.c, What does "int res" represent? |
| 13:45.58 | BobLutz | "int res = 0" line 89 to be exact |
| 13:47.48 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:47.48 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:49.02 | awk | hello putnopvut, with bug 12127 you had any issue with v3 patch? v2 was shoddy, and caused locks |
| 13:49.20 | awk | busy building a rpm package with the new v3, and see if it works... |
| 13:49.30 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:49.48 | putnopvut | awk: v2 caused locks? Did you mention it on the bug? |
| 13:50.47 | awk | we gave the info to jvandal... if he doesn't report it i will... |
| 13:51.07 | putnopvut | awk: wait... |
| 13:51.10 | putnopvut | do you mean 12098? |
| 13:51.31 | awk | also, you closed that thread on 1.4.17 iax io threads... I can re-produce that, just instaled a version now on a few clients with debug info... |
| 13:51.58 | putnopvut | awk: I have no idea what you're talking about with regards to iax io threads. |
| 13:52.06 | *** join/#asterisk oej (n=olle@ti400720a080-9600.bb.online.no) |
| 13:52.09 | awk | hmm, iax runs out of threads.. |
| 13:52.17 | awk | will show you bug id, wait... |
| 13:54.05 | awk | http://bugs.digium.com/view.php?id=11790&nbn=3 |
| 13:54.22 | awk | russel was dealing with it.. sorry.. |
| 13:54.22 | *** join/#asterisk lftsy (n=lftsy@anj75-1-81-57-0-22.fbx.proxad.net) |
| 13:54.43 | putnopvut | awk: Okay. |
| 13:55.01 | putnopvut | awk: So v2 on 12127 was causing locks? Are you sure you don't mean issue 12098? |
| 13:55.21 | awk | I have the lock file... |
| 13:56.32 | putnopvut | Could you upload it to the issue, please? That would be very helpful. |
| 13:56.56 | awk | hmm, sorry the lock file I have is for issue 12098.. only 1 client has issue with v2 |
| 13:57.12 | putnopvut | awk: Ah, that makes more sense. |
| 13:57.35 | awk | do you also need that? |
| 13:57.52 | putnopvut | awk: no, I think I've cleared up the locking problem with v2. |
| 13:58.03 | awk | great, let me try that patch... |
| 13:58.03 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:58.09 | putnopvut | jvandal reported that v3 doesn't crash or lock, but he has a lot of stuck channels. |
| 13:58.16 | putnopvut | I'm not sure if it's related to my patch or not though. |
| 13:58.23 | awk | let me quickly ask him on msn |
| 13:59.38 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 14:00.21 | awk | joel: I'm doing more test |
| 14:01.11 | awk | simplify dialplan and will send on ticket, with the patch asterisk doesnt crash, no locks, etc, but have 'zombie' channels, |
| 14:01.15 | awk | thats what he says... |
| 14:02.18 | putnopvut | I'll try to see why there could be zombie channels with that patch. If I find out why, I'll report on the bug. |
| 14:02.45 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 14:03.00 | awk | thanks... will keep updates on ticket too.. just doing some more tests myself.. |
| 14:03.14 | putnopvut | awk: Thanks for testing :) |
| 14:03.55 | *** join/#asterisk oej (n=olle@ti400720a080-9600.bb.online.no) |
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| 14:19.58 | *** join/#asterisk jks2 (n=jks@87.57.88.86) |
| 14:21.04 | jks2 | using * 1.4.x how can I store voicemail message in a mysql database? (the mysql vm interface seems to be gone and the realtime interface seems to allow me only to store voicemail users in the db) |
| 14:21.11 | ZaVoid | so how do i wait for an answer? or progress? to move on in my context? |
| 14:21.14 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582425.dsl.bell.ca) |
| 14:21.44 | *** join/#asterisk op3r (n=Op3r@222.127.88.164) |
| 14:22.11 | op3r | hello whats the command again to show how many channels you have for the e1? zap show status? |
| 14:22.20 | op3r | or zap show channels? |
| 14:22.55 | sysreq | jks2: you have to compile 'odbc voicemail storage' support and then use mysql through odbc.. |
| 14:23.03 | sysreq | jks2: http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage |
| 14:24.04 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 14:24.30 | jks2 | sysreq, okay, I was looking at that -- no way to go directly to MySQL instead of through unixodbc, right? |
| 14:25.17 | sysreq | jks2: i don't think so, no. |
| 14:25.36 | sysreq | also, beware that (taken from that page, at the bottom).. "I tried adding ODBC message storage to a 1.2.5 system already using MySQL for RealTime... Not a good idea, but using ODBC for both Realtime and msg storage seems good so far. -X1Z" |
| 14:25.52 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 14:26.06 | jks2 | hmm, but 1.2.5 is a long time ago |
| 14:26.07 | sysreq | i don't know if it still applies to 1.4 though. |
| 14:26.30 | sysreq | yeah, i agree.. but just keep that in mind if you ever go through problems ;p |
| 14:26.40 | jks2 | hehe, I'll do that :) |
| 14:27.45 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 14:28.09 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 14:28.28 | JayTee52 | I have 2 SIP phones setup on a new server to test, extensions 5154 and 5146. I can call 5146 from 5154 but when I try to call from 5146 to 5154 I get a fast busy and a 603 on the phone display. |
| 14:28.37 | ZaVoid | anyone got any ideas? i want to dial using SIP.. and when the far end users picks up run a macro.. but not till they pick up the phone |
| 14:29.05 | roffer | Hello, i have a question, i run Asterisk 1.4.4 and got a problem vith making a connection out with my sip provider. i can make internal calls and answer incoming calls from my sip provider but not make any calls out. i get this error. chan_sip.c:11848 handle_response_invite: Failed to authenticate on INVITE to '"mynumber" anyone that can help me with this problem ? |
| 14:29.08 | *** join/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
| 14:29.11 | [TK]D-Fender | ZaVoid: use a "call file" or "AMI Originate" |
| 14:29.19 | *** join/#asterisk fedya (n=fedya@75.112.143.226) |
| 14:29.21 | methods | anyone use the perl agi lib's ? |
| 14:29.32 | methods | sorry i mean Pg |
| 14:29.32 | *** join/#asterisk ddunavant (n=David@pool-71-178-115-160.washdc.east.verizon.net) |
| 14:29.41 | ZaVoid | i am doing that fender |
| 14:30.15 | ZPertee | If the first line of my dialplan is exten => s,1,Wait(10) does that mean that it will wait 10 seconds before answering? |
| 14:30.18 | DarKnesS_WolF | [TK]D-Fender: hello dude :) how are u ? |
| 14:30.41 | [TK]D-Fender | ZPertee: Depends on what kind of channel you are answering |
| 14:31.07 | [TK]D-Fender | ZPertee: Actually.... I should say "yes" to that/// |
| 14:31.15 | [TK]D-Fender | DarKnesS_WolF: Getting by |
| 14:31.29 | JayTee52 | On the server console I get the message: Warning[10793]: app_dial.c:1115 dial_exec_full: Dial argument takes format (technology/[device:] number1) |
| 14:31.40 | ZaVoid | fender http://pastebin.com/d20a92db4 |
| 14:31.45 | ZPertee | [TK]D-Fender: ok thanks. it will be a zap channel. |
| 14:31.58 | DarKnesS_WolF | [TK]D-Fender: have u ever had a snom phone behind nat ? i'm become unreachable in 1 min and 4 sec |
| 14:32.12 | [TK]D-Fender | DarKnesS_WolF: Read up : |
| 14:32.13 | [TK]D-Fender | ~sipnat |
| 14:32.14 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:32.54 | [TK]D-Fender | ZaVoid: Depends on what your channel is doing... |
| 14:33.04 | ZaVoid | i'm playing Background BEEP |
| 14:33.06 | ZaVoid | then reading digits |
| 14:33.15 | [TK]D-Fender | ZaVoid: the one you're CALLING. |
| 14:33.15 | ZaVoid | but someitmes the BEEp plays before teh call is answered |
| 14:33.20 | ZaVoid | yep |
| 14:33.30 | ZaVoid | sometimes it plays after its answered.. seems kinda random |
| 14:33.36 | DarKnesS_WolF | [TK]D-Fender: i did read it :( nop the phone is NATED the asterisk on public |
| 14:33.53 | [TK]D-Fender | DarKnesS_WolF: And you're not showing me anything... |
| 14:34.05 | [TK]D-Fender | ZaVoid: And you still aren't showing much either. |
| 14:34.12 | ZaVoid | yeah i know one sec |
| 14:34.34 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.227.137) |
| 14:35.17 | JayTee52 | ok, disregard my question. I figured it out. damn typos! grrrrrrr arrrggghhh |
| 14:35.30 | DarKnesS_WolF | [TK]D-Fender: sure ask to show u anything :-) the phone register after 1 min exactly it become unreachable |
| 14:35.50 | [TK]D-Fender | DarKnesS_WolF: Try ahrder. |
| 14:35.53 | [TK]D-Fender | harder* |
| 14:36.16 | ZaVoid | fender: http://pastebin.com/d7ce6b298 |
| 14:36.32 | roffer | need some help with auth, i get Failed to authenticate on INVITE to, i have tried auth in sip.conf and register, but cannot make a outgoing call. i can call intern and call inn to the sip privider configurd with asterisk gui inside users.conf ? anyone ? |
| 14:36.39 | DarKnesS_WolF | [TK]D-Fender: i think if itis 1 min then i need to add option for like defaultexpire or so |
| 14:37.09 | [TK]D-Fender | ZaVoid: I don't see the context for your Channel.... |
| 14:37.23 | [TK]D-Fender | DarKnesS_WolF: Show first, comment second.... |
| 14:37.38 | ZaVoid | line 7 |
| 14:37.49 | *** join/#asterisk JunK-Y (n=junky@modemcable153.55-201-24.mc.videotron.ca) |
| 14:37.59 | ZaVoid | [anicallback-leg2] |
| 14:38.34 | DarKnesS_WolF | [TK]D-Fender: yes what u wanna to see ? there is nothing on logs :-D tell me show me that and this and i'll do |
| 14:39.29 | [TK]D-Fender | ZaVoid: and your call-file says : Channel: Local/17328530514@anicallback-leg1/n |
| 14:39.37 | ZaVoid | # |
| 14:39.37 | ZaVoid | Context: anicallback-leg2 |
| 14:39.38 | [TK]D-Fender | ZaVoid: Leg1 <- |
| 14:39.45 | *** join/#asterisk draygon-w (n=draygon@gateway5-pnap.exigo.com) |
| 14:39.47 | ZaVoid | ahh |
| 14:39.59 | [TK]D-Fender | ZaVoid: I said your CHANNEL, not where you dump it after it ANSWERS! |
| 14:40.23 | *** join/#asterisk VJFROMGT (n=vjfromgt@pool-96-232-13-228.nycmny.east.verizon.net) |
| 14:40.31 | ZaVoid | so i'm still answering in anicallback-leg1 then |
| 14:40.31 | DarKnesS_WolF | [TK]D-Fender: Mar 14 16:13:39 NOTICE[4134]: chan_sip.c:10078 handle_response_peerpoke: Peer '187' is now REACHABLE! (372ms / 5000ms) |
| 14:40.34 | DarKnesS_WolF | Mar 14 16:14:43 NOTICE[4134]: chan_sip.c:11867 sip_poke_noanswer: Peer '187' is now UNREACHABLE! Last qualify: 372 |
| 14:40.51 | VJFROMGT | suddenly i hear no audio , nat looks ok |
| 14:40.54 | VJFROMGT | any suggestions? |
| 14:41.23 | [TK]D-Fender | ZaVoid: I want to see what it dials.... |
| 14:41.29 | ZaVoid | ok |
| 14:42.02 | VJFROMGT | if i restart the machine problem gest solved for a few hours |
| 14:42.04 | rkeene | Anyone else have a problem with the "tor2" module causing a kernel panic ? http://www.rkeene.org/viewer/tmp/asterisk/zaptel-causing-crash.txt.htm |
| 14:42.37 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:42.41 | yang | hey [TK]D-Fender can my dialplan start as exten =>_+. for numbers starting with + |
| 14:43.11 | [TK]D-Fender | yang: I might think so, but have never had to work with "+". Go try. |
| 14:44.12 | ZaVoid | Local/17328530514@anicallback-leg1-aff9,2" |
| 14:45.09 | ZaVoid | because if read callfiles correctly... If the call answers, connect it here * Context: <context-name> Context in extensions.conf |
| 14:45.29 | ZaVoid | so it sholdn't do anythign in the context until after the call actually connects. |
| 14:46.24 | *** join/#asterisk Entr4nced (n=IMG001@67-129-213-34.dia.static.qwest.net) |
| 14:46.36 | jasonWoot | does anyone have any experience with intuitivecreation's AMA suite? |
| 14:47.09 | roffer | need some help with auth, i get Failed to authenticate on INVITE to, i have tried auth in sip.conf anything else i can try ? |
| 14:47.38 | JunK-Y | rkeene: which kernel? |
| 14:47.42 | ZaVoid | so this happens before my phone even rings... so i'm getting false progress messages i guess |
| 14:47.43 | ZaVoid | <PROTECTED> |
| 14:47.43 | ZaVoid | <PROTECTED> |
| 14:47.45 | [TK]D-Fender | ZaVoid: "Channel: Local/17328530514@anicallback-leg1/n" <- show me what this is calling..... |
| 14:48.05 | ZaVoid | you mean the DIAL command fender? |
| 14:48.33 | [TK]D-Fender | ZaVoid: I want to see the whole damn dialplan for EVERYTHING this thing is supposed to be using AND the CLI output to match |
| 14:48.38 | rkeene | JunK-Y, 2.6.21.5-smp |
| 14:49.02 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 14:50.18 | JunK-Y | rkeene: report this with OS on bugs.digium.com please. |
| 14:50.56 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 14:51.40 | BobLutz | Does anyone here know where AST_FRAME_DTMF is defined? |
| 14:52.06 | *** part/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
| 14:52.59 | coppice | tzanger: what about an OC768 PCIE card? |
| 14:53.12 | JunK-Y | BobLutz: frame.h |
| 14:53.28 | tzanger | coppice: nah, that means I'd need something to connect it to :-) |
| 14:53.53 | coppice | make 2 |
| 14:54.24 | putnopvut | BobLutz: it's in include/asterisk/frame.h |
| 14:55.07 | BobLutz | JunK-Y: putnopvut: Thanks! |
| 14:57.50 | rkeene | JunK-Y, Done: http://bugs.digium.com/view.php?id=12213 |
| 14:58.57 | Qwell | ~nowwhat |
| 14:58.58 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 15:00.16 | roffer | need some help with auth, i get Failed to authenticate on INVITE to, i have tried auth in sip.conf anything else i can try ? |
| 15:00.45 | *** join/#asterisk RoyK (n=roy@box36.fortel.no) |
| 15:06.54 | anonymouz666 | Ashley Dupre is just....amazing. |
| 15:07.28 | anonymouz666 | I'd install an asterisk box for her for free. |
| 15:07.29 | anonymouz666 | lol |
| 15:08.05 | tzanger | haha |
| 15:08.11 | tzanger | she is not really good looking |
| 15:08.26 | tzanger | she's got a nice enough face and yeah she'd draw second looks,but amazing? |
| 15:09.16 | anonymouz666 | yeah, perfect. |
| 15:09.20 | anonymouz666 | IMHO |
| 15:09.49 | BobLutz | Is there a specific reason app_swift.c was never merged into the Asterisk project? |
| 15:10.10 | Qwell | BobLutz: because the author chose not to contribute it |
| 15:10.54 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:11.01 | BobLutz | Qwell: Interesting ... |
| 15:11.07 | *** join/#asterisk bmg505 (n=leon@196-209-76-155-tbnb-esr-2.dynamic.isadsl.co.za) |
| 15:11.13 | file | rkeene: that is already fixed in the latest Zaptel |
| 15:11.33 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 15:12.03 | Qwell | file: the tor2 thing? |
| 15:12.14 | Qwell | might want to poke sruffell before he looks into 12213.. |
| 15:12.34 | file | pretty sure kpfleming fixed it with revision 3863 |
| 15:12.57 | file | but... could be wrong |
| 15:14.34 | tzanger | struffel? |
| 15:17.47 | jks2 | sysreq, seems to be a problem yes...asterisk just crashes every time I run the voicemail app :-| |
| 15:20.21 | *** join/#asterisk ddunavant (n=David@pool-71-178-115-160.washdc.east.verizon.net) |
| 15:26.46 | casix | I'm making a auto-dial with ivr syste. When the .call file is readed it calls to the destination. After call (when is ringing) asterisk doesn't wait until it is answered. Asterisk continues executing. Can I make asterisk wait until it is answered?? |
| 15:26.51 | *** part/#asterisk C4colo (n=DJpyro@67.41.154.214) |
| 15:26.58 | agx | I've 2 xDSL and 2 public's IP to the asterisk box: but there is seems no way it accepts 2 IP in externip= for using SIP trunks |
| 15:28.18 | casix | but i'll have the same problem |
| 15:28.27 | rkeene | file, I'm using the latest Zaptel |
| 15:28.37 | rkeene | file, And so I must conclude that it is not fixed |
| 15:29.05 | file | you mentioned 1.4.9, the latest is 1.4.9.2 |
| 15:29.05 | sysreq | jks2: err.. and that's with odbc storage and mysql realtime? i guess you're going to have to use mysql through odbc for your realtime as well ;\ |
| 15:29.34 | *** join/#asterisk ming_zym (n=ming_zym@220.181.54.88) |
| 15:29.43 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:29.43 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:30.41 | jks2 | sysreq, yes, odbc storage and mysql realtime :-| |
| 15:31.00 | rkeene | file, Arg, regression |
| 15:31.37 | rkeene | file, Close the bug with extreme prejudice |
| 15:32.08 | jks2 | sysreq, just a bit odd that voicemail simply makes asterisk crash... mysql realtime works fine, but asterisk crashes when the voicemail app is started |
| 15:32.57 | sysreq | jks2: .. but then using odbc everywhere will allow you to have a more homogenous solution; plus if you decide to switch to another database system along the way, you'll only have to modify you odbc configs. |
| 15:33.11 | jks2 | oh well, I don't think I'll ever do that |
| 15:33.20 | jks2 | doesn't odbc introduce an overhead? |
| 15:34.44 | sysreq | i don't know, i've rarely used it |
| 15:34.55 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:35.09 | sysreq | but i wouldn't think so |
| 15:35.22 | jks2 | I would think so :-| |
| 15:37.14 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 15:38.30 | BobLutz | "if(f->frametype == AST_FRAME_DTMF) { ... }", if I want to read in more then 1 DTMF, I should change "AST_FRAME_DTMF" to "AST_FRAME_DTMF_BEGIN", right? |
| 15:42.35 | file | begin is a frame to indicate the start of a single DTMF, and an end is a frame to indicate the end of a single DTMF (the same DTMF as the begin), if you need to read in multiple DTMF digits you need to probably store the digits from end frames in a buffer |
| 15:43.56 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 15:44.04 | BobLutz | file: I am trying to modify app_swift.c (not official * code i know) to have functionality similiar to app_read.c, I am combing back through app_read.c, and I think I see my mistake, thank you |
| 15:46.22 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 15:46.50 | Zeeek | Anyone feel like talking? Maybe asterisk 1.6? I dunno. http://voipusersconference.org |
| 15:47.20 | Zeeek | Call (724) 444-7444 and enter 22622# 1# |
| 15:47.35 | Qwell | Zeeek: are the commercials over? |
| 15:47.39 | Zeeek | never |
| 15:47.44 | Zeeek | how can I pay for gas? |
| 15:48.03 | Zeeek | or the vegetarian restaurant where I get gas? |
| 15:49.00 | JunK-Y | Zeeek: can we call via IAX2? :) |
| 15:49.05 | Zeeek | sure |
| 15:49.10 | Zeeek | via your provider :) |
| 15:49.29 | JunK-Y | what about if my providers are zap only? :) |
| 15:49.38 | Zeeek | Call (724) 444-7444 and enter 22622# 1# |
| 15:49.48 | Zeeek | it's worth it |
| 15:50.03 | BobLutz | I called a couple weeks ago |
| 15:50.21 | Zeeek | BobLutz and were you treated with the proper respect ? |
| 15:50.35 | BobLutz | Someone was like "Who is this from Ohio?" |
| 15:50.41 | BobLutz | I got scared and hung up |
| 15:50.47 | Zeeek | Because we can't see who you are |
| 15:50.57 | Katty | HORAY! |
| 15:50.59 | Katty | MY PRI WORKS! |
| 15:51.00 | Zeeek | we see what you telco reports as an area code |
| 15:51.09 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:51.09 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 15:51.31 | Zeeek | Today, Barak Obama shares his dialplan secrets with us |
| 15:51.45 | Strom_C | Zeeek: usually, if you're running a conference, etiquette dictates that you don't reveal information about the callers |
| 15:52.11 | Zeeek | I have no other way to ask the person to speak UNLESS they register with Talkshoe |
| 15:52.23 | Zeeek | If they do, I see a pseudo and have NO other info about them |
| 15:52.31 | Zeeek | which is a GoogThing (tm) |
| 15:52.50 | Zeeek | as asterisk freaks, you are all able to change your CID anywway, no? |
| 15:53.01 | Strom_C | thats not the point |
| 15:53.11 | Strom_C | the point is you don't announce "Oh look, someone from Ohio" |
| 15:53.19 | Zeeek | ok here is the point: I HAVE NO WAY to ask someone to speak |
| 15:53.36 | Strom_C | good. Let them speak when they're ready. |
| 15:53.52 | JayTee52 | Katty, I'm getting ready to use 2 PRI circuits on our Asterisk box. Is your PRI an incoming PRI from your telco provider? |
| 15:54.08 | Strom_C | does a "telco provider" provide telephone companies? |
| 15:54.14 | Katty | JayTee52: yes, it is. |
| 15:54.19 | Katty | JayTee52: i'll be blogging it today, too. |
| 15:54.20 | Strom_C | or did you mean just 'telco'? |
| 15:54.22 | Strom_C | :P |
| 15:54.24 | Katty | JayTee52: would you like the link when i'm done? |
| 15:54.30 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:54.30 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 15:54.30 | JayTee52 | yes, please |
| 15:54.32 | Katty | k |
| 15:54.35 | Katty | hi blitz (= |
| 15:54.48 | coppice | only the desparate would blog about a PRI |
| 15:54.48 | blitzrage | Katty: hi! |
| 15:54.57 | blitzrage | Katty: you know that blitzrage == lmadsen right? :) |
| 15:54.58 | Katty | coppice: i blog everything, dear. |
| 15:55.09 | Katty | coppice: my memory requires it |
| 15:55.13 | coppice | that's even worse |
| 15:55.17 | Katty | hehehe |
| 15:55.20 | Katty | oh well. |
| 15:55.21 | Katty | it's handy |
| 15:55.27 | Katty | it will continue. |
| 15:55.33 | Strom_C | what the hell ever happened to notepaper? :P |
| 15:55.42 | Katty | i can't ready my own handwriting. |
| 15:55.47 | Katty | i type fast that i write anyway ;) |
| 15:56.02 | Strom_C | and clearly not very accurately... |
| 15:58.05 | anonymouz666 | Katty: I can't just write without a keyboard |
| 15:59.11 | *** join/#asterisk CrashSys (n=kumba@216-199-37-76.tpa.fdn.com) |
| 16:00.09 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 16:00.10 | CrashSys | Simple question about global var's. If I have status=1 as a global car, from a channel can I change it to status=0, and will it save that? |
| 16:00.15 | *** join/#asterisk af_ (n=getsmart@88-149-230-191.dynamic.ngi.it) |
| 16:00.41 | CrashSys | s/car/var |
| 16:00.55 | BobLutz | CrashSys: I beleive so...let me check something |
| 16:01.14 | DarKnesS_WolF | mmm been hours i'm wih this problem :-s |
| 16:01.17 | CrashSys | well I can do it the amish method... just try it on a test DID... |
| 16:01.33 | BobLutz | CrashSys: I do something like that in my extensions.conf and it works |
| 16:02.47 | Katty | does the [from-pstn] bit in extensions.conf handle incoming calls the same way with a pri as regular analog ports? |
| 16:02.54 | Katty | s,1, s,2, s,3 etc? |
| 16:03.06 | BobLutz | CrashSys: status will only be 0 for that channel that changed it though |
| 16:03.07 | DarKnesS_WolF | Katty: yes i think so |
| 16:03.19 | Katty | DarKnesS_WolF: do you know about DIDs? |
| 16:03.30 | CrashSys | BobLutz: What if I want to change the global var for all channels after that? |
| 16:03.48 | DarKnesS_WolF | Katty: nop :-) |
| 16:03.49 | CrashSys | isn't there a setglobal? |
| 16:03.56 | Katty | DarKnesS_WolF: okie dokie |
| 16:04.03 | CrashSys | katty: replace the s with the DID the carrier is sending |
| 16:04.09 | DarKnesS_WolF | Katty: what i thikn u'll need to create siperated gorups for ur DID's PRI channels |
| 16:04.19 | CrashSys | so if the carrier is sending the last 4 digits as the did, you would replace S with the last 4 digits... |
| 16:04.22 | DarKnesS_WolF | then u make context for each one |
| 16:04.36 | DarKnesS_WolF | and treat them the way u like |
| 16:04.46 | BobLutz | CrashSys: For some reason, I think you can NOT change the global var for all channels.. |
| 16:04.47 | CrashSys | so it would be <did>,1,command; <did>,n,command |
| 16:05.09 | CrashSys | BobLutz: well that makes the possibility of a night button difficult... |
| 16:05.17 | BobLutz | night button? |
| 16:05.35 | BobLutz | CrashSys: You were correct |
| 16:05.39 | BobLutz | page 434 |
| 16:05.40 | CrashSys | yes... old-school KSU/PBX feature... |
| 16:05.45 | *** join/#asterisk seanbright (i=seanbrig@65.207.74.18) |
| 16:05.48 | CrashSys | hit a button, and the system plays a "night menu" |
| 16:06.00 | CrashSys | I wish I had a better memory sometimes :( |
| 16:06.06 | Strom_C | CrashSys: don't do it with globals |
| 16:06.10 | BobLutz | Set(GLOBAL(var)=val) |
| 16:06.10 | Strom_C | do it with database settings |
| 16:06.25 | BobLutz | CrashSys: Are you talking about a GotoIfTime()? |
| 16:06.27 | Strom_C | that way, night mode will stay on until you explicitly turn it off |
| 16:06.34 | CrashSys | strom: you mean astdb? |
| 16:06.43 | CrashSys | or mysql? |
| 16:06.45 | Strom_C | astdb |
| 16:06.46 | Katty | CrashSys: dankou. |
| 16:06.53 | CrashSys | ok... |
| 16:06.58 | Strom_C | mysql is clearly overkill for your application |
| 16:07.04 | CrashSys | yes, I was about to say :) |
| 16:07.13 | CrashSys | I've never tried to store anything within astdb |
| 16:07.19 | Strom_C | it's super-easy |
| 16:07.24 | CrashSys | I shall now search voip-info.org for enlightenment :D |
| 16:07.45 | CrashSys | dbput/dbget? |
| 16:07.54 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 16:08.46 | Strom_C | jesus, that's ancient syntax |
| 16:08.49 | Strom_C | use the DB() function |
| 16:09.08 | CrashSys | one sec, let me fire up a CLI |
| 16:09.12 | Strom_C | is the wiki seriously still that out-of-date? |
| 16:09.22 | CrashSys | Yes, there is still pre 1.0 stuff on the wiki :) |
| 16:09.31 | CrashSys | that's what, 4 years old now? |
| 16:09.33 | Strom_C | does it still say things like "NOTE!!!! As of 0.8.4...." |
| 16:09.54 | CrashSys | the astdb page is pre 1.2 release :) |
| 16:10.01 | Strom_C | wigh |
| 16:10.04 | Strom_C | er, sigh |
| 16:10.06 | Strom_C | what a mess |
| 16:10.25 | CrashSys | :) Has digium hired a guy to do nothing but correct voip-info.org yet? |
| 16:10.43 | CrashSys | Imagine the money you'd save on tech support calls by having accurate 1.2/1.4/1.6 pages in there |
| 16:10.50 | CrashSys | or they |
| 16:13.34 | BobLutz | When you Google anything Asterisk, voip-info.org is the first always, lol |
| 16:14.49 | *** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 16:16.22 | *** join/#asterisk robeph (n=robf@router.asteriasgi.com) |
| 16:16.32 | bkruse | BobLutz: That is a huge reference, even moreso before there was a lot of documentation |
| 16:16.39 | bkruse | like 'the book', asteriskguru, etc |
| 16:16.48 | robeph | [TK]D-Fender: I found it today... hahah the problem was so...simple yet not readily apparent |
| 16:16.49 | BobLutz | I frequent the site a lot, people in here tell me not to |
| 16:17.17 | x86 | why would someone want to use SER (or OpenSER) with Asterisk? |
| 16:17.20 | seanbright | running 1.4.18 - i have a SIP phone set up and working, making calls out over a PRI via Zap. when the party on the other end answers the call, i get 1 CDR record. when the party on the other end does not answer, i get 2 CDR records. i would like to only have 1 in either case... |
| 16:17.52 | robeph | [TK]D-Fender: their dns was broken, but this didn't effect any normally dns requisite services, since host file or direct IP use bypassed this problem all together, what the problem was is that for some reason SRV was enabled in sip.conf and every time it'd try to dial it would send srv requests like they would soon be out of style. |
| 16:18.10 | robeph | [TK]D-Fender: turning this off...everything works fast and new. |
| 16:18.57 | robeph | though I'm not particularly sure of the byproduct of turning SRV requests off for sip will be. does anyone? |
| 16:20.03 | coppice | jameswf: http://www.xkcd.com/ |
| 16:32.36 | *** join/#asterisk Entr4nced (n=IMG001@67-129-213-39.dia.static.qwest.net) |
| 16:32.56 | *** join/#asterisk bipolar (i=bflong@216-164-162-138.pa.subnet.cable.rcn.com) |
| 16:33.24 | [TK]D-Fender | robeph: Seems to cause miraculous speed-ups for people with maligned DNS servers ;) |
| 16:34.19 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 16:34.41 | *** join/#asterisk VaNNi (n=VaNNi@lgb-static-216.70.165.200.mpowercom.net) |
| 16:35.07 | robeph | [TK]D-Fender: :p.. well the 36 second timeout must be set somewhere, which would account for that exact time wait that each dial received.. |
| 16:35.19 | robeph | I just wondered what the purpose of using it at all would be in the case of SIP |
| 16:35.42 | robeph | in this particular case the failover/static load balancing of SRV records seems silly |
| 16:36.40 | robeph | though I assume it may have some uses if using sip trunking, but i guess its not very bright in assumptive reasoning :p |
| 16:49.42 | draygon-w | yang did you get the + thingy resolved? |
| 16:50.21 | yang | draygon-w: i wasn't able to test it |
| 16:51.14 | yang | draygon-w: from what i saw on the CLI the customer succesfully connected, but i am not sure |
| 16:51.59 | *** join/#asterisk af_ (n=getsmart@88-149-230-191.dynamic.ngi.it) |
| 16:53.16 | *** join/#asterisk dikdust (n=dikdust@77.43.42.95) |
| 16:53.18 | dikdust | hi |
| 16:54.31 | dikdust | is there a way to look if I have putted mysql support in * ? guess I haven't compiled asterisk-addons package .-. |
| 16:54.59 | dikdust | grep sql on log => [Mar 14 17:50:10] VERBOSE[10958] logger.c: cdr_sqlite.so => (SQLite CDR Backend) |
| 16:57.54 | *** join/#asterisk xenonex (n=xenonex@92.47.14.21) |
| 17:01.14 | JunK-Y | type cdr status and look if you see mysql? |
| 17:04.25 | *** join/#asterisk minthome (n=mintee@c-68-45-231-166.hsd1.nj.comcast.net) |
| 17:04.27 | *** join/#asterisk quigon (n=matias@190.3.121.15) |
| 17:06.35 | blitzrage | codefreeze: ping |
| 17:07.54 | dikdust | thanks JunK-Y ;) there isn't mysql support ... and with asterisk-addons can't see mysql support ... |
| 17:09.13 | dikdust | damn guess I have taked 1.2 asterisk addons package .-. |
| 17:10.19 | BobLutz | Is it possible to have a SIP channel go through SSH? |
| 17:12.25 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
| 17:12.44 | codefreeze | blitzrage: here am i |
| 17:12.53 | blitzrage | codefreeze: !! |
| 17:12.58 | blitzrage | codefreeze: I have a CDR question for you............ |
| 17:13.26 | *** part/#asterisk quigon (n=matias@190.3.121.15) |
| 17:15.13 | blitzrage | codefreeze: ok, so I've got a call that comes in from a CCM to asterisk, and then asterisk does a call forward (just another Dial()) back out to another CCM gateway. I have the exchange: INVITE, 100 Trying, 180 Ringing. If the incoming leg to Asterisk then hangs up, I send the CANCEL, get a 488 back, then send a couple more CANCELs, and the exchange is done - however... in this scenario, my CDR disposition shows as 'ANSW |
| 17:15.13 | blitzrage | ERED' |
| 17:15.37 | blitzrage | I have two CDR records, one for the incoming that shows NO ANSWER, and the 2nd leg shows as ANSWERED... it's almost like they are swapped... |
| 17:16.49 | codefreeze | blitzrage: CCM = Corba Component Module? |
| 17:17.09 | file | Cisco Call Manager |
| 17:17.17 | blitzrage | heh... what file said :) |
| 17:17.33 | blitzrage | I don't get a 200 OK or anything... at least not from the 2nd call leg |
| 17:17.56 | codefreeze | OK, I was hoping for Cute, Cuddly... something, but that'll do... |
| 17:18.08 | blitzrage | heh |
| 17:19.00 | *** join/#asterisk xenonex (n=xenonex@82.200.211.5) |
| 17:20.05 | codefreeze | blitzrage: I have no quick answer to that one. This sounds like the code that does this is in chan_sip somewhere. |
| 17:20.42 | blitzrage | ya... :( |
| 17:20.46 | *** join/#asterisk hohum (i=dcorbe@dh-451-1-109.lax1.trit.net) |
| 17:20.48 | blitzrage | not entirely sure where to start debugging that one |
| 17:20.52 | codefreeze | Because of masquerading, I'd be not shocked if some sort of swapping might occur in edge cases |
| 17:21.16 | blitzrage | ya... I did canreinvite=no because I thought that was causing it, but same issue |
| 17:21.31 | blitzrage | seems like the first leg should have been set to ANSWERED, and the 2nd to NO ANSWER (at least that would make more sense to me) |
| 17:22.45 | codefreeze | It all comes down to the philosophy that CDR's store 3 events, and it gets real tricky sometimes picking out which channel & CDR an event applies to. |
| 17:23.21 | *** join/#asterisk roffer_ (n=roffer@39.84-234-228.customer.lyse.net) |
| 17:23.58 | roffer_ | need some help with auth, i get Failed to authenticate on INVITE to, i have tried auth in sip.conf anything else i can try ? |
| 17:24.03 | codefreeze | especially down in the channel drivers, where it's juggling all that stuff on an event-driven basis. |
| 17:25.14 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 17:26.25 | blitzrage | ya... I'm wondering if I should file a bug with the traces to at least determine if it's a bug, or something I'm doing stupid, or if this is normal.... |
| 17:29.40 | dikdust | ok solved I' m a jerk .-. |
| 17:29.52 | dikdust | thanks a lot to everybody := |
| 17:29.54 | dikdust | :) |
| 17:32.06 | roffer_ | i have problem with auth when i try dial out with sip provider. log here http://pastebin.com/d3a776261 every thing else vork local, inbound. |
| 17:32.33 | codefreeze | blitzrage: it won't HURT to report this bug, it sounds like one to me. I'm torn: I spent a week or two in the chan_zap code, trying to straighten out transfer issues; but there's dozens of overlapping scenarios, and fixes to one can throw another off... |
| 17:32.36 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 17:33.33 | anonymouz666 | codefreeze: what to do then? :) |
| 17:33.56 | codefreeze | blitzrage: I've worked on the CEL stuff (no, Juggie don't close 10099 yet), but that was classed as 'recreational' time, so I threw myself into just trying to straighten out what's there... |
| 17:35.40 | codefreeze | The whole thing has grown to be a mess, and needs to be re-engineered, but community input is definitely needed, and we need a spec, and agreement as to direction. |
| 17:36.24 | anonymouz666 | re-engineered means rewrite most of code? |
| 17:36.38 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
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| 17:37.16 | blitzrage | codefreeze: sounds like a very important discussion for astridevcon |
| 17:37.55 | blitzrage | codefreeze: ya, I think I'll file a bug and see if it's just the way I'm interpreting the data, or if it's some other underlying issue. Both channel legs are SIP, and there is no transfer going on. Just a call comes into Asterisk, and then Asterisk doesn't another Dial() back out |
| 17:38.25 | blitzrage | however, I DO a Playback() which might be causing the one ANSWERED disposition, but I think the bug is it being swapped on the CDR channels. I'll let you know when I've filed the bug. |
| 17:38.43 | blitzrage | might try taking out that playback to see if I end up with a disposition of NO ANSWER on both then |
| 17:38.54 | *** join/#asterisk R0land (n=Roland@193.227.191.90) |
| 17:39.03 | codefreeze | anonymouz666: Well, sort of. The CDR struct as it is now, probably should go. CEL is fundamentally the right route, I think, but I haven't thought all the detail through for it. What info you need to tie the segments together, and how to store/provide that info is what chiefly concerns me. |
| 17:39.21 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
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| 17:40.06 | R0land | hello all! |
| 17:41.24 | R0land | im facing a certain issue at hand, 1-Zap/7 was active, and asterisk was continually logging status of zap/7 without connecting any cable to line No:7 |
| 17:41.45 | R0land | what might be the cause of that! |
| 17:44.11 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:44.27 | *** join/#asterisk svenna_ (n=svenna@p548D35CF.dip0.t-ipconnect.de) |
| 17:44.47 | BobLutz | file: The light bulb just turned on, thanks again! |
| 17:45.44 | file | BobLutz: hmm? |
| 17:47.09 | BobLutz | file: I was asking about AST_FRAME_DTMF vs AST_FRAME_DTMF_BEGIN earlier |
| 17:47.12 | BobLutz | http://www.pastebin.ca/942632 |
| 17:47.17 | file | ah |
| 17:47.38 | file | AST_FRAME_DTMF is really AST_FRAME_DTMF_BEGIN in disguise, it's there for old times sake |
| 17:47.42 | BobLutz | I format my code like a no0b...but I graduated only a couple months ago |
| 17:47.45 | file | or is it AST_FRAME_DTMF_END... |
| 17:47.58 | file | yeah, it's end |
| 17:48.20 | JunK-Y | END yeah (based on the define) |
| 17:48.21 | BobLutz | Well..For some reason, AST_FRAME_DTMF wasnt picking up the first DTMF...AST_FRAME_DTMF_BEGIN does |
| 17:49.17 | BobLutz | I need to clean that code up big time, but I would like to submit it (contribute?) to the Asterisk project |
| 17:49.32 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:49.50 | file | are you the original author of what you are going to submit? |
| 17:50.22 | BobLutz | eh...I took will@loopfree.net's app_swift.c and combined it with app_read.c from Asterisk |
| 17:50.37 | BobLutz | Dynamic TTS + Read() |
| 17:51.09 | BobLutz | I dont know if app_swift.c is maintained anymore..I need to e mail him |
| 17:51.59 | *** part/#asterisk R0land (n=Roland@193.227.191.90) |
| 17:52.01 | file | that would be no then... which makes it more difficult, as in order to contribute you must sign a license agreement for what you are contributing... and since you are not the original author that is unhappy |
| 17:52.07 | robeph | what I should do is stick a small bit of code to warn on SRV timeout, so no one will spend 10 hours of trying to figure why asterisk receives Dial, waits 36 seconds, then sends invite.. |
| 17:52.20 | robeph | when dns isn't around. |
| 17:52.30 | BobLutz | file: bah..oh well, im too young with my eyes too wide to know any better |
| 17:52.40 | BobLutz | file: thanks for your help though |
| 17:52.46 | robeph | you have to sign license agreement for submitting code thats already gpld? |
| 17:53.04 | robeph | odd..but okay. |
| 17:53.29 | file | Digium has business edition which is not open source, in order to use contributions in that they have to be properly licensed |
| 17:53.55 | BobLutz | Hmm.. I wonder if that is why will@loopfree.net never submitted app_swift.c originally |
| 17:54.07 | file | it is possible |
| 17:54.14 | robeph | BobLutz: do what a lot do and simply toss your patch up on a seperate server with a what and why, if people find it useful they'll grab it, stick it up on some of the message list / newsgroups, people will flame you and praise you if its good... or both if not, or sometimes niether and ignore you =) |
| 17:54.28 | BobLutz | probably ignore me |
| 17:54.29 | BobLutz | lol |
| 17:55.01 | robeph | I can't tell you how many "after market" patches I shove down the throat of my open source apps on my machines =\ |
| 17:55.25 | robeph | probably 10% of what I build is patched in some form with various tweaks. =s |
| 17:55.34 | robeph | prolly why all my stuff is always falling apart but hey.... |
| 17:55.52 | file | the positive thing about having it put in is that it's no longer up to you to maintain it, and it potentially gets a wider audience... but some individuals still dislike licensing it, which is fine |
| 17:56.26 | robeph | file: do not underestimate the audience of asterisk users ML and newsgroups... those are usually the people that count anyhow ;) |
| 17:56.30 | BobLutz | I kinda always felt like a chump in the back of my mind for using all open source software, I had always wanted to contribute back |
| 17:57.08 | file | BobLutz: hehe |
| 17:57.22 | *** join/#asterisk Uni (n=mpartin@smtp.academicsuperstore.com) |
| 17:57.30 | robeph | BobLutz: hahah well don't do that, you're being gracious often times just using it ;).... the idea behind it is often just to create something useful and by it being useful you're using it to its purpose. but if you CAN contribute, doing so can do no harm, unless you write code like me then you get bricks through your window late at night.... |
| 17:57.48 | BobLutz | haha |
| 17:58.46 | Uni | hey all, got a problem, trying to add support for "if you know the ext of your part you can dial it..." and so I used Background(greeting) followed by a WaitExten(10,) but if I dial any numbers asterisk just hangs up the call, any ideas? |
| 17:59.54 | *** join/#asterisk RobH (n=RobH@216.207.245.1) |
| 17:59.57 | Uni | guess the relevant config section may help: http://pastebin.com/m711a6906 |
| 18:01.08 | robeph | seriously though, if you aren't the original licensee or you use multiple licensed libs etc. Get yourself a lil host, a free one for all it matters (not geocities... ) make a lil index.html with links to the directories with your tarballs for each lil project and an explination of what each is, why its done, what functionality it provides, why its useful, etc. then post the link in your sig on newsgroups / mailing lists during your normal |
| 18:01.23 | robeph | people will see and be curious, I often follow links from sigs to such things |
| 18:01.36 | robeph | found lotsa neat stuff , others do too, its free, and easy to get noticed if you do good work |
| 18:04.28 | BobLutz | robeph: Yea this works out cause I have a domain name, but nothing on the site |
| 18:05.04 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 18:05.25 | robeph | I mean nothing complex, I'd model it like cpan for example, just your name, explination of what it is, cut and dry, no pretty graphics, this is what you get download it or don't, here's how it works, have fun good day, gpl, bye =) |
| 18:05.39 | BobLutz | haha |
| 18:06.01 | *** join/#asterisk DonAlex (n=DonAlex@host86-137-212-179.range86-137.btcentralplus.com) |
| 18:06.08 | DonAlex | Waaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa |
| 18:06.14 | DonAlex | I want to cry.. :( |
| 18:06.23 | robeph | you already have. |
| 18:06.31 | *** part/#asterisk TripleX1 (n=TripleX1@modemcable132.108-83-70.mc.videotron.ca) |
| 18:06.38 | DonAlex | What the hell am I doing wrong that asterisk is ignoring my dialrules?! |
| 18:06.52 | robeph | care to be a bit more informative? |
| 18:06.58 | DonAlex | Every number I type it thinks is an extension. |
| 18:06.59 | BobLutz | DonAlex: `dialplan reload` |
| 18:07.16 | BobLutz | DonAlex: Misread, sorry |
| 18:07.17 | DonAlex | BobLutz: Done and Done. |
| 18:07.31 | robeph | paste the dialplan? |
| 18:07.33 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 18:07.43 | jameswf | ~pb |
| 18:07.44 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:07.57 | robeph | yeah not in here, that'd be bad. do what he says. |
| 18:08.13 | robeph | in sial or rafb pastebins? :'( |
| 18:08.21 | robeph | oh nm there is rafb |
| 18:08.24 | DonAlex | sure the snippet it is liek 3 lines |
| 18:08.25 | DonAlex | ;) |
| 18:08.52 | robeph | well if its ignoring it, the rest may be revealing as to why, ie things overriding it etc. |
| 18:09.31 | DonAlex | http://pastebin.com/m4ee84f70 |
| 18:09.40 | DonAlex | oh the owhole thing huh |
| 18:09.43 | DonAlex | no worrie |
| 18:11.37 | DonAlex | http://pastebin.com/d34785fc6 |
| 18:12.28 | DonAlex | That any help ? |
| 18:14.33 | DonAlex | It is not like they are complicated rules.. I only JUST started configuring things!!?! |
| 18:14.39 | robeph | heh |
| 18:14.50 | DonAlex | robeph: *sighs* |
| 18:14.55 | robeph | I still am not sure what you're passing and what you're getting back, and what you expect to get back |
| 18:15.32 | DonAlex | robeph: Well essentiall I am trying to dial an outside line. 123 is the speaking clock here int he UK |
| 18:15.44 | DonAlex | robeph: a good way to make sure we are outside.. |
| 18:16.21 | DonAlex | robeph: so _123 is if it matches 123 right? |
| 18:16.30 | DonAlex | robeph: add a 9 and send it out. |
| 18:16.38 | BobLutz | DonAlex: You might want to try setting timeouts |
| 18:16.54 | BobLutz | Set(TIMEOUT(digit)=3) for example |
| 18:16.57 | robeph | k that 123 made no sense to me sense 123 here in us is ...well nothing.. or part of a (cc)-npa |
| 18:17.04 | DonAlex | bob: Timeouts for what exactly? It is recognising 123 as an extension |
| 18:17.12 | BobLutz | 1-2-3 |
| 18:17.17 | DonAlex | in fact it recognises ANY number as a bloody extension?! |
| 18:17.37 | robeph | it is an extension... heh sort of |
| 18:17.41 | DonAlex | wait lemme show a real number |
| 18:17.42 | DonAlex | :) |
| 18:18.14 | DonAlex | [Mar 14 18:17:56] NOTICE[464]: chan_sip.c:13753 handle_request_invite: Call from '500' to extension '07050653748' rejected because extension not found. |
| 18:18.21 | DonAlex | Grrrr |
| 18:18.25 | DonAlex | ;) |
| 18:19.10 | DonAlex | opppps before you lambast me.. |
| 18:19.23 | DonAlex | [Mar 14 18:18:59] NOTICE[464]: chan_sip.c:13753 handle_request_invite: Call from '500' to extension '907050653748' rejected because extension not found. |
| 18:19.36 | DonAlex | does the same with a 9 as well.. as per the dial rules. |
| 18:19.40 | DonAlex | Hmmms |
| 18:20.24 | DonAlex | wonder if parked calls is to blame it is the only other bloody thing that is included I don't quite understand. |
| 18:20.49 | robeph | why include it? |
| 18:21.02 | rkeene | I'm having a weird problem with my voicemail after the most recent crash |
| 18:21.28 | robeph | also yo u even sure you're in the right context? |
| 18:21.37 | DonAlex | robeph: Dunno.. it did that by default. That's asterisk now for you? |
| 18:21.46 | robeph | it did what? |
| 18:21.51 | DonAlex | custome-numberplan-1 no? |
| 18:21.52 | rkeene | No voicemail menus have output ... It *ACTS* like it's working, as far as I can tell.. but I can't hear things like "Enter password" |
| 18:22.18 | robeph | DonAlex: yeh yio usure its dialing in that context |
| 18:22.31 | DonAlex | robeph: I mean [numberplan-custom-1] |
| 18:22.36 | DonAlex | robeph: of course |
| 18:22.37 | robeph | yes |
| 18:22.38 | robeph | ok |
| 18:22.47 | DonAlex | robeph: How do I make sure it is doing that? |
| 18:23.01 | DonAlex | robeph: I mean it is the Only Dialplan there is?! |
| 18:23.28 | BobLutz | context != dialplan |
| 18:23.31 | robeph | from console try dial 07050653748@numberplan-custom-1 |
| 18:23.34 | robeph | oh |
| 18:23.36 | robeph | dur |
| 18:23.46 | *** join/#asterisk steliosk (n=Stelios@athedsl-281704.home.otenet.gr) |
| 18:23.47 | robeph | heh |
| 18:23.51 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:24.12 | robeph | i guess I dunno then :p |
| 18:24.32 | DonAlex | robeph: The asterisk console? |
| 18:24.38 | robeph | DonAlex: well not from bash |
| 18:24.55 | bkruse | gui questions belong in #asterisk-gui, stop flooding #asterisk because you are not understanding :] |
| 18:25.10 | DonAlex | rob : Hmmmmm "No such command 'dial 07050653748@numberplan-custom-1' (type 'help' for help)" |
| 18:25.30 | DonAlex | robeph: Or am I being dense and do no need dial ? |
| 18:25.32 | robeph | no dial command? |
| 18:25.45 | DonAlex | robeph: apparently not. |
| 18:25.52 | robeph | odd.. dunno the internals well enough to know why you don't have that heh |
| 18:25.58 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
| 18:26.03 | bkruse | you do not have a console driver loaded. |
| 18:26.10 | bkruse | alsa/oss |
| 18:26.37 | DonAlex | bkruse: Ahh could be.. but this is and embedded box so not sure how to go about doing that |
| 18:26.53 | DonAlex | XR1000 if anyone is interested ;) |
| 18:26.57 | bkruse | DonAlex: it is it. |
| 18:27.10 | bkruse | DonAlex: What 'embedded box' ? |
| 18:27.23 | DonAlex | bkruse: Xorcom XR1000 |
| 18:27.26 | bkruse | an AA50? |
| 18:27.29 | bkruse | oh....no idea then |
| 18:27.32 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:27.43 | bkruse | my guess is it does not have alsa/oss (that would be normal in an embedded environment) |
| 18:27.49 | robeph | they really should include a console debugging dial app |
| 18:27.53 | DonAlex | bkruse: ugg fortunately not.. ;) Had even more headaches with that one :P |
| 18:28.05 | robeph | that isn't related to sound drivers heh |
| 18:28.15 | bkruse | robeph: nah, you can use originate I believe.... |
| 18:28.21 | robeph | ah? |
| 18:28.24 | DonAlex | robeph: Well I am guessing they think it will just 'work' and of course they are short on space :) |
| 18:28.35 | DonAlex | robeph: originate? |
| 18:28.42 | robeph | DonAlex: he said it not I |
| 18:28.44 | bkruse | type 'originate' at your asterisk cli. |
| 18:28.53 | robeph | bkruse: I do have dial, don't have originate =s |
| 18:28.53 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:28.56 | DonAlex | Ahh yes.. |
| 18:28.58 | robeph | does it take same args? |
| 18:29.02 | DonAlex | that's there.. |
| 18:29.04 | bkruse | robeph: you do not have originate on your asterisk CLI? |
| 18:29.08 | DonAlex | any idea of the synatax? |
| 18:29.31 | bkruse | originate iax2/bkruse extension 9037whatever@numberplan-custom-1 |
| 18:29.40 | bkruse | type 'originate' at the CLI! It tells you the syntax! |
| 18:29.44 | robeph | hahah |
| 18:29.53 | robeph | pbx*CLI> originate |
| 18:29.54 | robeph | No such command 'originate' (type 'help' for help) |
| 18:29.57 | robeph | ahhh ok =) |
| 18:30.02 | robeph | who does originate belong to |
| 18:30.03 | DonAlex | bkruse: It tells me a lot more than that ;) Hadly a man page though |
| 18:30.18 | bkruse | robeph: not sure |
| 18:30.31 | robeph | what does it do, route call between two devices? |
| 18:30.41 | robeph | or device / ext |
| 18:30.43 | DonAlex | bkruse: How are extensions specified then ? 500@<ip> |
| 18:31.08 | bkruse | 500@context, it has to be on the local box |
| 18:31.21 | *** join/#asterisk echo--- (n=echo@64.184.118.232) |
| 18:31.37 | robeph | bkruse: what did BobLutz mean that I was wrong in calling that a context |
| 18:32.11 | DonAlex | bkruse: local, what like a FXS port? |
| 18:32.17 | znoG_ | is there some way to tell the IP address of a call made by a SIP/IAX user? |
| 18:32.31 | BobLutz | robeph: I just said a dialplan isnt a context really |
| 18:32.34 | BobLutz | a dialplan has a context |
| 18:32.48 | robeph | BobLutz: I meant point the ext at the context in his dialplan |
| 18:32.49 | robeph | ok |
| 18:32.50 | robeph | :p |
| 18:32.54 | robeph | ie ext@context |
| 18:32.54 | BobLutz | sorry haha |
| 18:33.12 | robeph | I was confused I was lik I couldv'e sworn I knew what I meant to say here... |
| 18:33.47 | BobLutz | happy pi day everyone |
| 18:34.26 | robeph | anyhow DonAlex you wanna ' originate proto/device 07050653748@numberplan-custom-1 ' I'm guessing without having the help text for that function |
| 18:34.41 | robeph | BobLutz: 3.1415926535898? |
| 18:34.48 | robeph | I think thats it anyhwo |
| 18:34.58 | BobLutz | :) |
| 18:35.27 | robeph | I have an ex gf who knew it to a disturbing 80 something places, why I do not know, it was likely just a display of deeper rooted mathmatical obsessions |
| 18:35.35 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-88-173.vif.net) |
| 18:35.53 | DonAlex | rats.. being kicked out of the office.. Will be back in a few hours.. Thanks for your help guys.. I will be looking up the originate when I get home. |
| 18:36.02 | DonAlex | Laters all |
| 18:36.32 | BobLutz | haha wow, 80? |
| 18:36.51 | robeph | yeh |
| 18:36.59 | robeph | + - some |
| 18:37.44 | robeph | in truth it was really odd, cos aside from that lil nuance, she was a rather normal girl, worried more of fashion and chick flicks than science et al. |
| 18:37.50 | robeph | :p |
| 18:37.58 | robeph | *shrug* just one of those oddities.. |
| 18:41.06 | rkeene | Hmm, it seems that Playback() no longer works on my Asterisk system.. any ideas ? |
| 18:41.20 | robeph | when'd it stop working? |
| 18:41.53 | CrashSys | Hey strom, is AstDB persistent? |
| 18:42.00 | rkeene | After the asterisk box crashed and came back up |
| 18:42.28 | BobLutz | rkeene: is the module loaded? |
| 18:42.31 | rkeene | (I had restarted the asterisk daemon recently, and all my configuration files are version controlled...) |
| 18:42.53 | rkeene | BobLutz, I would guess so.. it worked this mornig |
| 18:43.30 | rkeene | (Which module) |
| 18:43.36 | robeph | app_playback2.so |
| 18:43.44 | BobLutz | 2? |
| 18:43.53 | robeph | I guess, thats what I use |
| 18:44.00 | robeph | :p |
| 18:44.04 | BobLutz | haha |
| 18:44.09 | robeph | it works |
| 18:44.17 | robeph | so.. I mean i dunno if there's another version heh |
| 18:44.19 | BobLutz | 1.4.17 --> app_playback.so for me |
| 18:44.31 | robeph | heh I'm using 1.2.24 |
| 18:44.37 | rkeene | Module Description Use Count |
| 18:44.42 | rkeene | app_playback.so Sound File Playback Application 1 |
| 18:44.47 | rkeene | (I'm using 1.4.18) |
| 18:45.51 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:46.56 | BobLutz | rkeene: Does modules.conf autoload/ |
| 18:47.18 | rkeene | [modules] |
| 18:47.18 | rkeene | autoload=yes |
| 18:47.26 | BobLutz | word... |
| 18:47.41 | BobLutz | No kind of error? |
| 18:48.55 | rkeene | No error that I've been able to find |
| 18:51.10 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 18:54.27 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 18:54.44 | *** join/#asterisk svenna_ (n=svenna@p548D35CF.dip0.t-ipconnect.de) |
| 18:56.25 | robeph | rkeene: not sure if it'll help, but turn debug on? |
| 18:56.45 | *** join/#asterisk docelmo (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net) |
| 18:58.39 | BobLutz | What exactly is an ASTOBJ? |
| 19:00.02 | russellb | an object management api ... |
| 19:00.57 | BobLutz | Reading through the source.. just trying to wrap my head around it |
| 19:01.08 | russellb | which has been deprecated in favor of astobj2 |
| 19:01.49 | BobLutz | d'oh |
| 19:02.01 | russellb | :) |
| 19:02.09 | russellb | astobj2 is documented pretty well |
| 19:02.33 | BobLutz | Is there better documentation then the doxygen? |
| 19:02.50 | russellb | include/asterisk/astobj2.h |
| 19:02.51 | rkeene | robeph, Seems to be related to the Zaptel module |
| 19:02.59 | rkeene | Because I unloaded that, and it is working now |
| 19:03.07 | russellb | there are some things in the header file that aren't showing up in doxygen for some reason |
| 19:03.08 | robeph | ah |
| 19:03.11 | robeph | strange |
| 19:03.18 | BobLutz | russellb: thanks |
| 19:03.27 | CrashSys | all hail the mighty russell |
| 19:04.02 | russellb | BobLutz: no problem |
| 19:04.06 | russellb | CrashSys: greetings sir |
| 19:04.17 | CrashSys | Now I need a beer |
| 19:04.17 | rkeene | Perhaps even the "tor2" module, since I can load many Zaptel modules, but not "tor2" |
| 19:04.26 | rkeene | (Well, I can.. but then playback() quits working) |
| 19:05.17 | *** join/#asterisk codejunky (i=jan@88.198.12.5) |
| 19:05.24 | CrashSys | Can we write an app_beer module? monitors my stock of stella artois in the fridge, and re-orders if it gets low |
| 19:06.06 | CrashSys | Some sort of predictive ordering algorithm... |
| 19:06.33 | Qwell | ~nowwhat |
| 19:06.34 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 19:06.56 | codejunky | Hello, I have connected my asterisk to my voip provider with the sip protocol and connected a phone (hardware) to my asterisk. Now the problem is that if I dial out I do not hear the one I am calling for 5-10 seconds after he answers the call. Any ideas what I can do? |
| 19:08.09 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 19:08.17 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 19:08.36 | *** join/#asterisk svenna_ (n=svenna@p548D35CF.dip0.t-ipconnect.de) |
| 19:09.52 | nny_1 | ok kids.. here's a loaded question.. Does have anyone have an opinion on the price of this system: |
| 19:09.52 | nny_1 | 170 Snom 300 Phones |
| 19:09.52 | nny_1 | Quad T1 Card (Digium) |
| 19:09.52 | nny_1 | Server - Dual Core, 4Gb Ram |
| 19:09.52 | nny_1 | System load average 20% |
| 19:09.53 | nny_1 | 65,000.00 |
| 19:10.15 | [TK]D-Fender | Snom... BLEH |
| 19:10.20 | CrashSys | 65K? |
| 19:10.22 | CrashSys | .... |
| 19:10.24 | nny_1 | yes |
| 19:10.35 | jasonWoot | is that Yen? |
| 19:10.40 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 19:10.45 | nny_1 | [TK]D-Fender: USD |
| 19:10.47 | nny_1 | oops mt |
| 19:10.49 | nny_1 | USD |
| 19:11.07 | jasonWoot | ¥ |
| 19:11.09 | Qwell | err...what's a snom 300 cost? |
| 19:11.12 | CrashSys | Seems like a high price to me... |
| 19:11.23 | nny_1 | [TK]D-Fender: heh only for the web interface, easier to test remotely |
| 19:11.27 | nny_1 | CrashSys: installed |
| 19:11.36 | CrashSys | Including cabling? |
| 19:11.37 | nny_1 | CrashSys: basically we are installing it |
| 19:11.45 | nny_1 | CrashSys: existing infrastructure |
| 19:11.55 | [TK]D-Fender | nny_1: BLEH <- And massively over-priced. |
| 19:11.58 | CrashSys | So your just deploying to the desktop and server rack? |
| 19:12.06 | nny_1 | CrashSys: yes |
| 19:12.07 | bkruse | Qwell: for some strange reason, I heard the music video from ~nowwhat, but it was not from my pc..... ha ha :] |
| 19:12.16 | Qwell | bkruse: heh |
| 19:12.42 | [TK]D-Fender | nny_1: Snom 300 = $99 USD (retail). 170 x $99 =16830$ USD for phones. So basically you're selling a server for almost 50K$ |
| 19:12.51 | Qwell | ... |
| 19:12.58 | nny_1 | [TK]D-Fender: 2 hours per phone setup and testing |
| 19:13.04 | Qwell | 2 hours?! |
| 19:13.05 | *** join/#asterisk snapple42 (n=snapple4@h216-18-80-132.gtconnect.net) |
| 19:13.05 | Qwell | PER PHONE? |
| 19:13.08 | Qwell | no |
| 19:13.19 | Qwell | 5 minutes - tops |
| 19:13.20 | [TK]D-Fender | nny_1: Man I gotta get me some customers like that ;) |
| 19:13.21 | nny_1 | Qwell: 1 hour in house testing, physical deployment in room |
| 19:13.24 | nny_1 | [TK]D-Fender: hehe |
| 19:13.27 | CrashSys | You are doing something wrong if a large deployment like that takes you 2 hours per phone set-up/install |
| 19:13.28 | Qwell | ...you've gotta be kidding me |
| 19:13.29 | nny_1 | it's for a hotel |
| 19:13.30 | Deeewayne | Qwell, maybe its mini-mall style testing |
| 19:14.02 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 19:14.17 | CrashSys | 65K is high |
| 19:14.23 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582425.dsl.bell.ca) |
| 19:14.25 | bkruse | Qwell: Agreed, we should start a business |
| 19:14.29 | nny_1 | Qwell: ha good luck |
| 19:14.51 | nny_1 | Qwell: this isn't in HHI, and if you have ever been here, you'd know it's a deal compared to the rest of the industry here |
| 19:15.18 | Qwell | nny_1: feel free to send some of that money to the developers :) |
| 19:15.28 | nny_1 | Qwell: heh |
| 19:15.33 | CrashSys | an IPOffice on 8 phone with sip trunks was $10K... |
| 19:15.38 | Qwell | small, unmarked bills would be best |
| 19:15.52 | nny_1 | CrashSys: yeah our 10 phone system is 6k |
| 19:15.57 | nny_1 | installed |
| 19:15.58 | St1ckm4n | we payed about 15k for our nortel switch that capped out at 25 phones |
| 19:16.05 | CrashSys | nny: Well the definition of good customer service is they are happy when they leave... so if they are tickled pink to pay $65K for it and you can deliver their needs, sell it... |
| 19:16.25 | nny_1 | CrashSys: yeah this is all prelim.. we mainly deal with smaller systems atm |
| 19:16.39 | nny_1 | we are probably going to knock down the installation time |
| 19:17.00 | nny_1 | we have a metric that doesn't scale well, and that's one of the things I am looking at |
| 19:17.16 | CrashSys | cant imagine you'd have more then 40 work hours in laying the system out and setting up provisioning files |
| 19:17.47 | CrashSys | Specially for a simpler hotel set-up... most PITA thing is billing/wake-up calls |
| 19:17.47 | nny_1 | CrashSys: true |
| 19:17.51 | bkruse | besides, snom's auto provisiong ROCKS |
| 19:17.53 | nny_1 | CrashSys: yeah indeed |
| 19:17.56 | nny_1 | bkruse: agree |
| 19:17.57 | *** join/#asterisk quigon (n=matias@190.3.121.15) |
| 19:18.01 | CrashSys | Polycom's... |
| 19:18.03 | nny_1 | hey i haven't sent an invoice yet, lol |
| 19:18.06 | nny_1 | yeah |
| 19:18.15 | nny_1 | polycom is on the slab too, (we normally use them) |
| 19:18.15 | bkruse | nny_1: if you can do it, do it, we are all just jealous |
| 19:18.31 | CrashSys | If you want to sub it out, let me know :) |
| 19:18.38 | nny_1 | :) |
| 19:18.39 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 19:18.40 | bkruse | i would say 95% is putting the phones in place, provisioning is 5% with snom's awesome auto provisioning, you just have to be efficient |
| 19:18.47 | nny_1 | bkruse: fully agreed |
| 19:18.57 | nny_1 | bkruse: i have the ability to do the provisioning |
| 19:19.00 | Qwell | walk to a room. plug it in. |
| 19:19.08 | Qwell | that's...err...not hard |
| 19:19.11 | nny_1 | yeah lol |
| 19:19.37 | bkruse | Qwell: Like I said, we need to start a business. QwellKruzTeleJoshComVoxMart - Install asterisk and phones, only 70k! |
| 19:19.53 | CrashSys | bkruse: you hiring? |
| 19:20.02 | nny_1 | well my main concern is how this stacks against 1.) say someones trixbox deployment (shudder) 2.) a traditional analog system and 3.) nortel/mitel/etc |
| 19:20.19 | nny_1 | analog, er --> FXS |
| 19:20.40 | bkruse | CrashSys: of course, it takes a lot of man power for provisioning! or...I could be completely lying. :P |
| 19:20.43 | nny_1 | and heh, hell yeah start the damn business, the market is horribly inflated, obviously :) |
| 19:20.53 | CrashSys | I'm good at phone deployment! |
| 19:21.04 | CrashSys | I even make sure the cords dont go under the phone stand! |
| 19:21.06 | bkruse | nny_1: asterisk > nortel/mitel/avaya in analog deployment |
| 19:21.08 | bkruse | CrashSys: hired |
| 19:21.16 | nny_1 | bkruse: agreed |
| 19:21.25 | CrashSys | sahweet |
| 19:21.25 | bkruse | nny_1: actually, the market is not, you have to know HOW to market |
| 19:21.26 | Qwell | but, you probably aren't beating them on price |
| 19:21.33 | bkruse | Qwell: never |
| 19:21.40 | Qwell | not at 65k |
| 19:21.50 | nny_1 | right now ALL of our SMB systems kick the shit out of the local market |
| 19:21.52 | bkruse | nortel systems (analog) for that many are under 30k easy |
| 19:21.59 | nny_1 | mind you we are also in the process of doing an 800 phone system |
| 19:22.06 | bkruse | So..... |
| 19:22.10 | nny_1 | we even beat the local telco on pricing |
| 19:22.13 | CrashSys | For a straight analog switch you wont be beating the mitel's... |
| 19:22.22 | bkruse | CrashSys: In that price range, no |
| 19:22.24 | nny_1 | er mitel IP rather |
| 19:22.29 | Qwell | except that his solution *would* be less than 30k |
| 19:22.38 | Qwell | if he wasn't charging 4x for the phones |
| 19:22.45 | CrashSys | True |
| 19:22.59 | nny_1 | whos charging 4x per phone? |
| 19:23.02 | bkruse | Qwell: Absolutely. |
| 19:23.07 | nny_1 | fender hit it on the head |
| 19:23.10 | nny_1 | 170 per phone |
| 19:23.15 | nny_1 | 7k server |
| 19:23.20 | bkruse | Not to mention that snom does annual pricing agreements, which you should look into.... |
| 19:23.26 | nny_1 | heh and 34k for install |
| 19:23.39 | nny_1 | yeah we get the snoms for less than 99 |
| 19:23.41 | CrashSys | 7 |
| 19:23.49 | CrashSys | 7K for a server? You aren't buying a dell are you? |
| 19:23.54 | bkruse | nny_1: exactly. That what 'mass purchasing' is. |
| 19:23.56 | nny_1 | no thats retail |
| 19:24.01 | CrashSys | Cause that's one way you can guarantee some service calls :D |
| 19:24.03 | nny_1 | bkruse: indeed i know this |
| 19:24.08 | nny_1 | CrashSys: no dell |
| 19:24.10 | bkruse | CrashSys: Even so, 2950's are ~4k with 4 gigs of ram and rails |
| 19:24.15 | nny_1 | CrashSys: and thats not what we pay for those items |
| 19:24.26 | nny_1 | CrashSys: we are a business :) |
| 19:24.32 | bkruse | CrashSys, Qwell: so basically he gets even more money in pocket getting them even cheaper |
| 19:24.51 | nny_1 | bkruse: precisely |
| 19:24.53 | CrashSys | I'm just joe blow trunk slammer... but i'd ethically have an issue with myself charging $65K for that job... |
| 19:25.07 | CrashSys | Plus at $65K you wont win the bid compared to a TDM Interconnect |
| 19:25.08 | nny_1 | CrashSys: you or a company? |
| 19:25.18 | nny_1 | CrashSys: no bidding here |
| 19:25.24 | nny_1 | CrashSys: we are their provider |
| 19:25.41 | Qwell | so then charge $800k |
| 19:25.45 | CrashSys | If they'll pay, take 'em... |
| 19:25.45 | nny_1 | CrashSys: and remember, this is a discussion, no invoice estimate or anything has left my desk |
| 19:26.04 | nny_1 | so you would charge what? |
| 19:26.32 | nny_1 | that's why i am having this discussion, i am not out for someone's bloood or to overcharge, however, our 2 hour metric usually includes training, which this doesn;t need |
| 19:26.45 | nny_1 | and we are evolving the process |
| 19:27.28 | bkruse | Qwell: You know how hard we could hit the market and what an advantage we would have? lol. qwell + bkruse > all |
| 19:27.40 | nny_1 | bkruse: what are you waiting for? |
| 19:27.42 | nny_1 | bkruse: do it |
| 19:27.47 | bkruse | nny_1: I work for digium. |
| 19:27.49 | nny_1 | bkruse: we need more asterisk companies out there |
| 19:27.51 | nny_1 | bkruse: lOL |
| 19:28.01 | CrashSys | nny: shooting from the hip i'd ballpark a job like that around 30K |
| 19:28.06 | nny_1 | bkruse: well.. then you guys know what the deal is |
| 19:28.06 | bkruse | I will wait for the buyout, then do something like that :P |
| 19:28.12 | bkruse | yes, we do :] |
| 19:28.15 | bkruse | I have a couple things up my sleeve.... |
| 19:28.15 | CrashSys | without spending 3-4 hours and a site-survey crunching the hard numbers... |
| 19:28.32 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 19:28.52 | nny_1 | bkruse: you guys have siwtchvox systems priced pretty damn close to ours |
| 19:29.08 | bkruse | CrashSys: Honestly, with competing with cisco, I have heard of many cases where the company went with cisco instead of an asterisk install because the price was TO low, we cannot criticize nny_1 for having some of the market figured out. |
| 19:29.24 | nny_1 | bkruse: indeed |
| 19:29.36 | nny_1 | bkruse: we lost the local township system to Cisco JUST for that reason |
| 19:29.42 | CrashSys | Well, then just say $500/phone |
| 19:30.05 | CrashSys | That's on par for digital KSU/PBX |
| 19:30.07 | bkruse | nny_1: People will be people. Thinking logically is not always the standpoint to view from. In the marketing biz, perception is your best friend. |
| 19:30.26 | nny_1 | FWIW in the precise local market we have one competitor, and we undercut their NOrtel BCM prices by 10% at least |
| 19:30.32 | bkruse | Figuring people out is a gift and a talent that you must possess |
| 19:30.41 | CrashSys | Just hope there's no ESI's or Telrad's up in your neck of the woods |
| 19:30.49 | keith4_ | what are my options for boot servers for polycom phones? all examples I see seem to be ftp |
| 19:31.00 | bkruse | keith4_: how about res_phoneprov.c in trunk? |
| 19:31.03 | nny_1 | heh we sell enough, I'll invite you all down.. this market is it's own unique metropolis |
| 19:31.14 | bkruse | keith4_: You can do it all from the GUI with minimal tweaking....... |
| 19:31.23 | keith4_ | GUI? |
| 19:31.24 | nny_1 | my business partner is an old Fed/ GIS guy and the statistics here are mind blowing |
| 19:31.28 | bkruse | nny_1: You haven't even seen the market outside the US. |
| 19:31.30 | bkruse | keith4_: web interface |
| 19:31.32 | nny_1 | bkruse: indeed |
| 19:31.44 | [TK]D-Fender | keith4_: Go read the admin guide... |
| 19:31.46 | keith4_ | bkruse: looking to do provisioning using dhcp |
| 19:32.00 | nny_1 | bkruse: mind you, we are looking at *this* one vs* the global market |
| 19:32.12 | nny_1 | bkruse: don't get me wrong, 65k was prelim, we hope to get it down to under 50 |
| 19:32.15 | keith4_ | [TK]D-Fender: polycom's admin guide? |
| 19:32.22 | [TK]D-Fender | keith4_: Clearly. |
| 19:32.27 | keith4_ | ugh |
| 19:32.42 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 19:33.02 | CrashSys | nny_1: In the florida area an average of $500/station is a pretty good "dip-stick" to measure an install by... |
| 19:33.04 | bkruse | nny_1: Sure, but I am talking about business outside the US. I do not think you fully realize that people take what is made, and sell it for hundreds of thousands to countries that do not have information like you do. I know people in india that make hundreds of thousands / year because they understand the market and can read english and translate. |
| 19:33.05 | CrashSys | unless your cisco... |
| 19:33.06 | keith4_ | I started to, but it's full of so much useless crap |
| 19:33.12 | nny_1 | CrashSys: thank you |
| 19:33.51 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 19:33.53 | CrashSys | But for $500/station, it's a full-features KSU/PBX from Telrad or ESI or Samsung, etc... |
| 19:34.09 | CrashSys | Hotel systems are a lot cheaper... almost half usually |
| 19:34.23 | CrashSys | Because it's all SLT's and just a billing modules/wake-up call |
| 19:34.31 | nny_1 | bkruse: I understand, thank you... i hope to learn that first hand soon :) |
| 19:34.44 | *** join/#asterisk bkruse (n=bkruse@216.207.245.1) |
| 19:34.44 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 19:34.48 | bkruse | sorry about that |
| 19:34.55 | CrashSys | Mitel's big into hotel systems... |
| 19:35.29 | ZPertee | anyone use ipcomms free DID service? |
| 19:36.10 | [TK]D-Fender | keith4_: Its the 4th chapter, and blatantly obvious. Seek therapy... |
| 19:36.29 | bkruse | [TK]D-Fender: haha, try not to be too harsh, but I completely agree. |
| 19:36.36 | keith4_ | [TK]D-Fender: if you don't know the answer, just say so :-P |
| 19:36.42 | nny_1 | CrashSys: bkruse etc .. going to rewrite this, let you know what craooens |
| 19:36.44 | nny_1 | crappens* |
| 19:36.49 | keith4_ | I didn't say "where can I learn for myself how to provision polycoms" ;-) |
| 19:37.08 | nny_1 | and thank you all for the input, I appreciate the discussion |
| 19:37.25 | [TK]D-Fender | keith4_: And no, I will not be so easily goaded into satisfying your sloth-like tendencies :p |
| 19:37.33 | keith4_ | awwww |
| 19:37.37 | [TK]D-Fender | pwned |
| 19:37.38 | bkruse | nny_1: The global market. For example, I know of an asterisk installer in brazil that does outrageous installs, you have to think outside the box of the US government and regulations and think from their perspective, where no one knows asterisk, and pbx is a monopoly, one that can be easily taken over |
| 19:37.52 | keith4_ | all I have is this lousy PDF that doesn't have an index in the sidebar |
| 19:37.53 | bkruse | jbot: [TK]D-Fender++ |
| 19:39.37 | keith4_ | [TK]D-Fender: FYI, it's chapter 3, not 5 |
| 19:39.39 | keith4_ | er, not 4 |
| 19:40.21 | nny_1 | bkruse: heh i recently dealt with an install in Panama. The company used trixbox on two boxes (IAX2 to US). The boxes were way subpar (VIA Chipset, Harddrives on static wrap, unmounted) MiniITX systems. 8 or so Snom Phones, $10,000. We ended up wiping Trixbox, setting up proper dialplans from scratch. The hardest part wasn't the price, they had NO recourse for getting some of their money back or the situation resolved before we stepped in |
| 19:40.45 | bkruse | nny_1: My point exactly |
| 19:40.54 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:41.02 | nny_1 | oh and the US box is no throwing hard drive errors, so I had to give him a quote on preplacing it, as the box itself is basically trash |
| 19:41.07 | nny_1 | now* |
| 19:41.43 | [TK]D-Fender | keith4_: Depends which version you're looking at, and it sure appears you found it jsut fine... |
| 19:41.46 | bkruse | nny_1: dell 2950's are your friend |
| 19:41.54 | znoG_ | sorry to repeat, i just find it hard to believe no-one knows how to do this :) is there some way to tell the IP address of a call made by a SIP/IAX user? |
| 19:42.08 | keith4_ | [TK]D-Fender: yah, but now all I have is lots of reading to do, and no easy answers |
| 19:42.14 | bkruse | znoG_: sip debug |
| 19:42.24 | bkruse | It might even be in cdr-csv |
| 19:43.06 | [TK]D-Fender | keith4_: its in bllody big print as to what protocols is supports for the boot server. Get some new eyes... this is just sad... |
| 19:43.14 | [TK]D-Fender | ashdsdklsd |
| 19:43.15 | keith4_ | it's aggravating that polycom abbreviates "Soundpoing IP" as "SIP" everywhere |
| 19:43.44 | nny_1 | bkruse: yeah we are a growing company, we aren't trying to rape the market, or inflate it anymore than it already is... My business partner and I had a discussion on how we could lower the pricing on this quote, and what it looked like vs. other quotes we have seen. We came to two conclusions, one is that our quote could* be lowered, although we have played that game before and lost due to the lower rate exhibiting the product as "cheap" and 2.) we are al |
| 19:43.45 | keith4_ | oh... yah I found the tftp stuff a while ago |
| 19:43.50 | lirakis | when you install a new kernel, do you just need to recompile zaptel stuff? or * too |
| 19:44.10 | keith4_ | just zaptel |
| 19:44.14 | nny_1 | bkruse: we are looking at dells for the HA setup (800 phones) here as well |
| 19:44.22 | nny_1 | bkruse: heheh well |
| 19:44.24 | bkruse | :] |
| 19:44.30 | bkruse | Ya, you got it down. |
| 19:44.34 | *** join/#asterisk quigon (n=matias@190.3.121.15) |
| 19:44.41 | nny_1 | bkruse: thanks.. |
| 19:44.44 | bkruse | Exactly, lower rate = cheap |
| 19:44.56 | bkruse | you completely understand what 75% of people in the market do not |
| 19:45.02 | bkruse | the other 25% are making $$$, and lots of it. |
| 19:45.12 | jasonWoot | keith if you figure out how to get rid of softkeys on the polycoms, let me know... going on 6 months now |
| 19:45.15 | nny_1 | bkruse: agree |
| 19:45.36 | nny_1 | brb |
| 19:45.56 | keith4_ | jasonWoot: sure, as soon as I figure out what a softkey is |
| 19:46.20 | BobLutz | Is there some kind of naming convention for Asterisk code? (AST_DECLARE_APP_ARGS(), AST_APP_ARG() --> are these in the same source file?) |
| 19:46.50 | [TK]D-Fender | jasonWoot: Like which? |
| 19:47.41 | bkruse | naming convention and what file they reside in are two totally different question |
| 19:47.45 | bkruse | BobLutz: ever heard of ctags? it rocks. |
| 19:47.56 | BobLutz | ctags? |
| 19:48.26 | bkruse | !google vim + ctags |
| 19:49.00 | BobLutz | bkruse: I see the vim ctags, but I know not how to use vim :-/ sucks to be me i guess |
| 19:49.17 | BobLutz | err ctags in the source code... I didnt know what they wre called |
| 19:49.37 | CrashSys | Hmmm |
| 19:50.05 | bkruse | BobLutz: when you find someone you want to the find the function/origin you can use ctrl+] |
| 19:50.32 | *** join/#asterisk hmmhesays (n=hmm@24-119-176-74.cpe.cableone.net) |
| 19:51.03 | BobLutz | bkruse: lol that is so useful! |
| 19:51.15 | bkruse | BobLutz: you have no idea. |
| 19:51.17 | BobLutz | lol |
| 19:51.22 | BobLutz | i really dont.. |
| 19:51.22 | bkruse | :] |
| 19:51.31 | bkruse | especially in big projects like asterisk |
| 19:51.52 | keith4_ | can I reboot a polycom remotely? |
| 19:52.02 | bkruse | keith4_: web interface |
| 19:52.07 | keith4_ | tried that |
| 19:52.12 | keith4_ | web interface sucks |
| 19:52.13 | bkruse | actually, just make a change in the web interface in the admin panel, it has to reboot |
| 19:52.17 | CrashSys | keith: if there has been a provisioning change, you can use sip notify... otherwise web interface |
| 19:52.18 | bkruse | no it does not, you suck. |
| 19:52.19 | keith4_ | oh duh |
| 19:52.24 | keith4_ | no, it sucks |
| 19:52.32 | *** join/#asterisk joshaidan (n=brianj@S0106001c1023e838.tb.shawcable.net) |
| 19:52.34 | bkruse | keith4_: or, you cannot use the web interface and need to admit it |
| 19:52.44 | *** join/#asterisk retloc (n=retloc@199-117-163-66.dia.static.qwest.net) |
| 19:52.45 | znoG_ | bkruse: wouldn't sip debug only apply to SIP clients? |
| 19:52.53 | [TK]D-Fender | "Denial, its not just a river in Egypt" |
| 19:52.55 | keith4_ | i prefer the snom web interface to the polycoms |
| 19:53.11 | nny_1 | keith4_: the snom's and the polycoms are night and day as far as phones go |
| 19:53.12 | retloc | My extensions.conf keeps clearing itself after every edit. Permissions are 777 and I am using nano for editing. |
| 19:53.16 | [TK]D-Fender | keith4_: Shouldn't be touching the web interface on a Polycom EVER anyways. |
| 19:53.17 | retloc | Any thoughts on this |
| 19:53.27 | bkruse | znoG_: oh, I thought you said sip |
| 19:53.27 | nny_1 | [TK]D-Fender: heh agreed |
| 19:53.30 | keith4_ | [TK]D-Fender: how else would you recommend rebooting a phone that's several miles away? |
| 19:53.43 | [TK]D-Fender | keith4_: the WIKI is your friend. |
| 19:53.44 | keith4_ | nny_1: agreed |
| 19:53.46 | znoG_ | bkruse: i need to find out the IP of the user in either SIP or IAX (depending on how they logged in) |
| 19:53.50 | keith4_ | the wiki is a mess |
| 19:54.10 | CrashSys | Keith: make a silly change in a polycom provisioning file, like changing registration number 8's port to 5061... it'll reboot |
| 19:54.15 | keith4_ | as wiki's tend to be |
| 19:54.24 | retloc | Even if I copy extensions.conf off of one of my identical working dialers it still clears the file completely |
| 19:54.29 | [TK]D-Fender | keith4_: http://www.voip-info.org/wiki-Polycom+reboot+hardphone+script |
| 19:54.35 | jasonWoot | polycom won't reboot via web interface if ext is in use |
| 19:54.47 | [TK]D-Fender | keith4_: Like a second google search on it doesn't turn it up as the FIRST result, nice and blatant. |
| 19:55.31 | nny_1 | heheh damn telemarketers.. "Can I speak to the decision maker?" (Accented voice)... who the hell responds to that?? |
| 19:55.33 | keith4_ | I need 130 lines of perl to reboot a phone? not acceptable |
| 19:55.43 | nny_1 | I need to instill telemarketer hell as defined in the wiki |
| 19:56.11 | [TK]D-Fender | keith4_: You have a very serious reading dysfunction.... |
| 19:56.19 | retloc | For the solutions it was found |
| 19:56.27 | retloc | MY HDD was ful |
| 19:56.32 | retloc | Clearing recordings |
| 19:56.40 | *** join/#asterisk FuriousGeorge (n=brian@ool-4354d18c.dyn.optonline.net) |
| 19:56.46 | FuriousGeorge | hey all |
| 19:56.55 | nny_1 | retloc: was that a haiku attempt :) |
| 19:57.17 | keith4_ | [TK]D-Fender: all of these methods seem to use the fact that the phone is periodically checking its config file for updates.... i need to reboot the phone to have it re-dhcp to tell it which tftp server to use... so that's not very helpful, now is it? |
| 19:58.08 | [TK]D-Fender | keith4_: Keep reading, and it might all sink in... |
| 19:58.23 | FuriousGeorge | i was just thinking: if there were some elegant way to automatically check in on listening channels that have been open for 5 hours or more, somehow determine if there is activity, and if not perform a soft hangup, i could probably go without rebooting asterisk daily |
| 19:59.13 | FuriousGeorge | its not a bid deal, but it kinda bothers me. i dont restart postfix or apache daily |
| 20:01.15 | FuriousGeorge | and it rarely just crashes and dumps a core, that i could file a bug report wity |
| 20:01.17 | FuriousGeorge | *with |
| 20:01.37 | nny_1 | something to consider on dealing with pricing vs. cost vs. competition too is what you can do to increase the price with VAR based things, extended warranties, 24 hours support, etc... You can charge X for the system on a base level, but add the things that you have to address in the market. |
| 20:02.09 | keith4_ | damn |
| 20:02.13 | keith4_ | i'm such an asshole |
| 20:02.15 | nny_1 | the reason a cisco phone system is high dollars is because you can call Cisco at 4 am, and tell them what you had for lunch, and for that price they'll listen |
| 20:02.45 | FuriousGeorge | nny_1: is that to me? |
| 20:02.57 | nny_1 | FuriousGeorge: heh no, part of a prior discussion |
| 20:03.06 | FuriousGeorge | heh |
| 20:03.08 | nny_1 | FuriousGeorge: what kind of system are you dealing with |
| 20:03.10 | nny_1 | ? |
| 20:03.38 | nny_1 | phones/ channels/ version of asterisk? I haven't seen the issues you mentioned, so I am curious (furious?) as to what is causing your specific reboot needs |
| 20:04.03 | FuriousGeorge | nny_1: one sec, pls... darned clients always interrupt |
| 20:05.34 | keith4_ | [TK]D-Fender: why can't you just say "sip notify" instead of letting me ramble on like an idiot for 10 minutes? |
| 20:05.34 | nny_1 | heh |
| 20:05.41 | [TK]D-Fender | keith4_: What and prevent this ultimate reality from smacking you upside the head? Never! ;) |
| 20:06.03 | keith4_ | in my defense, you sent me to a wiki page called "reboot hardphone SCRIPT" |
| 20:06.53 | [TK]D-Fender | keith4_: Sorry... doesn't hold water. You're every attempt at reading anything for information shows the same pattern. Thime to correct the problem, not the symptom. |
| 20:06.53 | *** join/#asterisk cardiff (n=cardiff@76-10-153-160.dsl.teksavvy.com) |
| 20:07.04 | [TK]D-Fender | your* |
| 20:07.04 | keith4_ | lol |
| 20:07.09 | [TK]D-Fender | time* |
| 20:07.13 | keith4_ | the solution would be a cup of coffee |
| 20:07.17 | [TK]D-Fender | WOW.... I'm f-ing fried today... |
| 20:07.26 | nny_1 | it is Friday |
| 20:07.29 | keith4_ | but it's nearly 5 here, so it's not going to happen |
| 20:08.20 | nny_1 | beer it is |
| 20:08.42 | keith4_ | no |
| 20:08.46 | keith4_ | beer is what started this problem |
| 20:08.50 | nny_1 | lol |
| 20:08.51 | keith4_ | 2 of them, at lunch, to be specific |
| 20:09.16 | keith4_ | [TK]D-Fender: i'm not usually this dumb, but i've been drinking at work today |
| 20:13.43 | *** join/#asterisk xenonex (n=xenonex@82.200.211.5) |
| 20:16.26 | bkruse | either you work rocks, or it sucks horribly |
| 20:16.57 | *** part/#asterisk cardiff (n=cardiff@76-10-153-160.dsl.teksavvy.com) |
| 20:17.15 | *** join/#asterisk ThatKidKel (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 20:17.28 | ThatKidKel | anyone know where one can buy or download a quality NPA NXX list? |
| 20:19.06 | _ShrikE | ThatKidKel: nanpa |
| 20:19.10 | keith4_ | bkruse: no specific policy about drinking during lunch |
| 20:19.19 | keith4_ | and the boss is out today... so.... |
| 20:19.28 | ThatKidKel | _ShrikE.. I'm all over there site, and can't find it |
| 20:19.28 | nny_1 | bkruse: btw have you heard of any changes in the Digium support offerings? We are under the understanding that you can buy 24/7 support, but only with AstBiz... which I have nothing against, except there are quite a few situations where we would rather use our own base distro |
| 20:19.43 | ThatKidKel | _ShrikE.. Only Area Codes |
| 20:19.55 | bkruse | nny_1: AstBiz? You can purchase 24/7 support for any particular product |
| 20:20.20 | bkruse | nny_1: http://www.digium.com/en/services/maintenance.php |
| 20:20.51 | nny_1 | bkruse: yeah they update that, but AFAIK for config support, you have to use AstBiz |
| 20:21.08 | _ShrikE | ThatKidKel: you can try maponics |
| 20:21.29 | BobLutz | This Asterisk programming is mad hard |
| 20:21.40 | nny_1 | BobLutz: heh is that sarcasm? |
| 20:21.52 | BobLutz | no |
| 20:22.06 | BobLutz | struct stuff *ps !#$# |
| 20:22.09 | bkruse | BobLutz: Programming asterisk? It is if you do not know how to program, or the layout and some of the API's in asterisk |
| 20:22.22 | Qwell | BobLutz: what, you've never used structs and pointers? |
| 20:22.23 | bkruse | but talk about open source, you have all the examples you could ever want! |
| 20:22.32 | bkruse | Qwell: That is why programming asterisk is hard :] |
| 20:22.37 | BobLutz | I know a little bit of C |
| 20:22.40 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 20:22.44 | BobLutz | Its kinda weird going from OOP java to C |
| 20:22.47 | bkruse | BobLutz: would it help if I rewrite asterisk in visual basic? |
| 20:22.52 | BobLutz | LOL |
| 20:23.02 | bkruse | :] |
| 20:23.04 | nny_1 | bkruse: lol actually, I would prefer it in the original BASIC |
| 20:23.06 | BobLutz | Im a fast learner...I didnt even know what a keyboard was a couple years ago |
| 20:23.07 | russellb | i'm almost done with my bash rewrite |
| 20:23.17 | bkruse | russellb: good good |
| 20:23.47 | BobLutz | app_read.c is extremely helpful though, if you guys see Mark, tell him I said great job |
| 20:24.25 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 20:24.26 | russellb | try channels/chan_zap.c |
| 20:24.29 | russellb | that's the best place to start |
| 20:24.44 | russellb | *** KIDDING *** |
| 20:24.58 | BobLutz | LOL, i was opening it up in my text editor |
| 20:25.04 | *** join/#asterisk harryv (n=fufufu@0x55508034.adsl.cybercity.dk) |
| 20:25.39 | russellb | nah, apps/ and funcs/ are the best place to start in general ... |
| 20:25.49 | BobLutz | If I can get this module to do what I want, I will probably try to do some janitor work in my free time |
| 20:25.49 | russellb | some of them, anyway |
| 20:25.50 | nny_1 | bkruse: long story short, and openly stating the fact that additional fees are expected, we want to be able to tell our customers that on top of our already **stellar** support we offer, they can contact digium directly after we have all fled to Mexico and still have a support channel. Shit right now I already have the framework, site and software setup for a proper support channel, I just hate to reinvent such an already prefectly round whee; |
| 20:25.52 | anonymouz666 | I wonder if someone really understand chan_zap.c :) |
| 20:25.53 | nny_1 | wheel* |
| 20:25.55 | russellb | app_queue, app_dial, for example, are more complicated |
| 20:26.08 | BobLutz | app_read is nice...barely at 200 lines |
| 20:27.07 | Qwell | "if you guys see Mark, tell him I said great job" |
| 20:27.26 | Qwell | I wonder how much of his code is still left |
| 20:27.30 | russellb | heh |
| 20:27.35 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 20:27.49 | Qwell | probably a crapton |
| 20:27.54 | russellb | probably |
| 20:27.56 | BobLutz | the only file i havent seen his name on is astobj2.h |
| 20:28.02 | russellb | he's the last person who changed 23% of the lines of code |
| 20:28.09 | Qwell | russellb: damn |
| 20:28.10 | russellb | kpfleming is the last person who changed the most, at 26% |
| 20:28.12 | anonymouz666 | Qwell: did he still program? |
| 20:28.20 | Qwell | anonymouz666: occasionally.. |
| 20:28.27 | russellb | http://bugs.digium.com/svnstats/asterisk/trunk/ |
| 20:28.43 | Qwell | 1.8%? O.o |
| 20:28.52 | BobLutz | lol |
| 20:28.53 | Qwell | + 0.1%. w00t |
| 20:28.59 | russellb | ha |
| 20:29.28 | Qwell | Marko? |
| 20:29.29 | Juggie | 0%, woot :) |
| 20:29.33 | russellb | no, me, heh |
| 20:29.36 | Qwell | oh |
| 20:29.58 | Qwell | russellb: I'm sure it would be much different in 1.0 |
| 20:30.01 | russellb | i wonder if the svn mv to create the main/ dir got kpfleming some stats :-p |
| 20:30.12 | Qwell | hmm, not sure how svn handles moves |
| 20:30.17 | russellb | no idea |
| 20:30.31 | Juggie | if Qwell would stop commiting my patches i'd have 2 :p |
| 20:30.47 | Qwell | wmeadows has 0 lines per change |
| 20:30.51 | Qwell | wtf did he change? |
| 20:31.13 | bkruse | nny_1: Then contact sales@digium.com and strike a support deal! |
| 20:31.23 | nny_1 | bkruse: good call, will do :) |
| 20:31.46 | bkruse | nny_1: That, and you can probably get around it by purchasing our hardware (which you will probably have to do) and getting support through those |
| 20:31.48 | Qwell | wow, I did not expect that at all. 33% of my changes are in main/, and 27.8% in channels/ |
| 20:31.52 | Qwell | ...7% in apps |
| 20:32.10 | bkruse | but MANY companies have deals to where they have 24/7 support with digium that they offer when they cannot figure things out. |
| 20:32.22 | nny_1 | bkruse: by hardware do you mean the card or the entire PBX? |
| 20:32.23 | russellb | i have made over 150 commits between 2 and 3 AM |
| 20:32.25 | russellb | O.O |
| 20:32.32 | bkruse | nny_1: cards |
| 20:32.37 | bkruse | russellb: I do not doubt that at all |
| 20:32.39 | nny_1 | bkruse: cool that we already do |
| 20:33.12 | keith4_ | do I really have to "contact my polycom reseller" to get the damn firmware? |
| 20:33.46 | hmmhesays | or google |
| 20:33.53 | hmmhesays | which firmware are you looking for? |
| 20:34.06 | keith4_ | SIP 3.0 |
| 20:34.15 | keith4_ | also, the "distribution zip" file |
| 20:34.18 | keith4_ | for the boot server |
| 20:34.39 | keith4_ | the SIP 3.0 administrator's guide says "Copy all files from the distribution zip file to the phone home directory." |
| 20:34.47 | keith4_ | like I'm supposed to a) know what that file is or b) have it |
| 20:35.41 | hmmhesays | I would guess that is the firwmware file |
| 20:35.51 | keith4_ | me too |
| 20:36.02 | keith4_ | but to login to the polycom portal, you apparently have to be a reseller partner |
| 20:36.08 | keith4_ | which is idiotic |
| 20:36.41 | keith4_ | I even tried to open an account, and one of the options under "what are you?" is "phone owner"... and then you can login, but can't download anything |
| 20:36.43 | keith4_ | fucking polycom |
| 20:37.01 | *** join/#asterisk EvilDeshi (n=Skunk@75-135-93-93.dhcp.mdsn.wi.charter.com) |
| 20:37.17 | EvilDeshi | anyone around that can help me resolve this issue I am having with realtime using the odbc handler? |
| 20:37.36 | Qwell | keith4_: so ask your reseller? |
| 20:37.49 | keith4_ | I did. they haven't gotten back to me |
| 20:37.57 | keith4_ | telephonydepot |
| 20:38.21 | Yourname`` | So if manager dialing is crappy and slows everything down like asterisk taking up 100% CPU, loads going upto 70... what am I doing wrong? |
| 20:42.40 | nny_1 | good question, is there a way to break down app cpu usage? |
| 20:43.02 | FuriousGeorge | nny_1: im bacm |
| 20:43.07 | FuriousGeorge | and back too |
| 20:43.17 | BobLutz | nny_1: `top` ? |
| 20:43.37 | EvilDeshi | do I have to use odbcinst to get odbc to work with mysql and realtime? |
| 20:43.55 | nny_1 | BobLutz: lol |
| 20:44.08 | nny_1 | BobLutz: well.. yeah.. i knew that :) |
| 20:44.13 | BobLutz | lol |
| 20:45.38 | FuriousGeorge | i have two 'medium volume servers' on running 1.2.twenty-something, thats been around for four years, another one on 1.4.18... today, they both use tyan motherboards w/ opterons on nforce chipsets, and sangoma cards, but in the past ive used tdm400p |
| 20:45.43 | FuriousGeorge | i use snom phones |
| 20:45.53 | FuriousGeorge | the 1.2 box tends to have internal calls lock |
| 20:46.01 | nny_1 | BobLutz: it may be the crack i just smoked, but doesn't top *just* show asterisk usage? I was thinking something like "app_que" is using X resources |
| 20:46.04 | FuriousGeorge | the 1.4 tends to have inbound pots calls lock |
| 20:46.30 | BobLutz | `man top` ? |
| 20:46.37 | nny_1 | BobLutz: indeed |
| 20:46.39 | BobLutz | lol |
| 20:46.39 | FuriousGeorge | the former is sip<->sip, the latter is pots->sip. if i reboot them nightly they will be good for 6 months |
| 20:46.52 | FuriousGeorge | if i dont they will start experiencing hung channels after a few days |
| 20:47.22 | nny_1 | BobLutz: looks like i have some reading to do, my top-fu is only basic |
| 20:47.28 | keith4_ | htop |
| 20:47.53 | FuriousGeorge | err, i meant to say 'if i restart * nightly' |
| 20:48.52 | bkruse | FuriousGeorge: Bug reports means bugs get fixed (with proper information) |
| 20:49.04 | bkruse | saying "zaptel doesn't work" does not help.... |
| 20:50.04 | FuriousGeorge | bkruse: i started out by saying, before i got interrupted, that im not sure how to bug report these. there is no core dump |
| 20:50.13 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 20:50.22 | FuriousGeorge | and when it happens i just quickly softhangup the channels and cron asterisk to reboot at night |
| 20:50.48 | FuriousGeorge | i keep saying reboot instead of restart, but you get the idea |
| 20:51.44 | nny_1 | time to flee, later all |
| 20:51.47 | FuriousGeorge | later |
| 20:51.56 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 20:52.18 | bkruse | FuriousGeorge: Console output or anything? |
| 20:53.57 | FuriousGeorge | bkruse: just the occasional 'maximum retries exceeded' but those always tend to happen. I must confess i dont grasp the lingo of the CLI, but i don't see anything that jumps out at me. |
| 20:54.10 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:56.24 | FuriousGeorge | chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 3c42 |
| 20:56.53 | file | that's bad... it means chan_sip sent out a packet somewhere but got no response |
| 20:58.14 | FuriousGeorge | file: this is a pretty basic setup, but they do some volume in calling. any practical cause for that? bad wiring? bad phones? all of the above? |
| 20:58.32 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 20:58.35 | BobLutz | its 5 |
| 20:58.40 | bkruse | file: Right, possibly a slow dns response? |
| 20:58.48 | bkruse | Or his other endpoint is just disappearing? |
| 20:58.50 | file | possibly configuration... possibly network... possibly routing... |
| 20:58.57 | file | without a sip debug it is hard to say |
| 20:58.58 | bkruse | I wonder how much packetloss he has to that host |
| 20:59.13 | BobLutz | file: russellb: Qwell: bkruse: Thanks for the direction with the Asterisk code |
| 20:59.15 | *** part/#asterisk BobLutz (n=miles@d60-65-93-136.col.wideopenwest.com) |
| 20:59.20 | bkruse | lol |
| 20:59.43 | russellb | func_shell is where it's at |
| 21:00.16 | bkruse | russellb: Totally. It is a good answer to "Wtf struct iax2_user *iax2_user, I do not have *'s in java!" |
| 21:00.34 | Qwell | multiplying a struct times an iax2_user?! |
| 21:00.35 | bkruse | BobLutz messaged me that :] |
| 21:00.36 | Qwell | you're mad! |
| 21:00.41 | file | FuriousGeorge: but it is not normal to get those... |
| 21:01.18 | FuriousGeorge | file: i can prevent it by rebooting asterisk |
| 21:01.20 | russellb | bkruse: lol! |
| 21:01.21 | FuriousGeorge | nightly |
| 21:01.29 | bkruse | Qwell: Exactly! a "structure" and "pointer" um, did you include the math library import java.classes.util.io.math.asterisk.string.res.channels.math.Properties.1948.newest.class.math ? |
| 21:01.30 | russellb | pointers are such silly |
| 21:02.49 | FuriousGeorge | i notice if i use answer-after: 0 with my snom phones i can reliably hang a sip channel using 1.2 |
| 21:03.35 | russellb | 1.2? |
| 21:03.38 | russellb | hmmm ... oh yeah! |
| 21:03.41 | russellb | i remember 1.2 ... |
| 21:03.47 | Qwell | 1.2 what? |
| 21:03.48 | FuriousGeorge | 1.2.20 |
| 21:03.49 | russellb | but then i moved on like 2 years ago |
| 21:03.56 | *** join/#asterisk darius_ (n=darius@humility.bourg.net) |
| 21:04.05 | file | my apartment PBX runs 1.6.0-beta4 |
| 21:04.11 | russellb | 63 changes to asterisk 1.2 since 1.2.20 |
| 21:04.19 | darius_ | Who's the iax supporting PSTN terminating voip provider of choice these days? |
| 21:04.42 | FuriousGeorge | i suppose i should upgrade |
| 21:04.58 | file | russellb: SHARK! |
| 21:05.21 | russellb | OMFG NO! |
| 21:05.23 | bkruse | FuriousGeorge: http://www.voip-info.org/tiki-index.php?page=Asterisk+v1.2 |
| 21:05.42 | bkruse | notice one of the first lines "On Nov. 15, 2005 Asterisk 1.2.0 finally saw the light of day!" |
| 21:05.49 | bkruse | today is March 14, 2008. |
| 21:06.07 | FuriousGeorge | bkruse: yeah, i know development stopped, but the 1.4.18 that i run, when its channels lock, its pots and sip |
| 21:06.28 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 21:06.46 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 21:07.04 | FuriousGeorge | so im scared if i upgrade, since the servers are almost identical, that i will get the worse version of the problem... i guess it doesnt matter since i restart daily, but i feel like that is just giving up on finding the nature of the issue |
| 21:11.49 | *** join/#asterisk plasmid (n=noway@c-76-124-171-163.hsd1.pa.comcast.net) |
| 21:13.03 | fujin | anyone know if it's possible to do a silent macro/gosub yet? |
| 21:13.13 | fujin | I'm using a macro for a local channel which is really really noisy |
| 21:14.12 | plasmid | I am having difficulty with my P2PT sipura box not seeing my pbx. I get this on the registration bit: Registration State:Can't connect to login server. I did a vi /etc/hosts.allow and my P2PT of IP 192.1681.107 is allowed. What the devil am I misisng here? Ports 5060-5061,10001~10021 UDP open on router. |
| 21:15.12 | plasmid | i changed the registraion details to that of the provider and it registers fine... but as soon as i change the proxy to my internal pbx (192.168.1.105) I get the above error. What gives? |
| 21:16.20 | EvilDeshi | anyone know how i can fix this issue [unixODBC][Driver Manager]Data source name not found, and no default driver specified? |
| 21:16.37 | fujin | configure odbc properly |
| 21:16.45 | EvilDeshi | I am not sure how |
| 21:17.00 | *** join/#asterisk RobH (n=RobH@216.207.245.1) |
| 21:20.14 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 21:21.51 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 21:23.12 | *** join/#asterisk zobia (n=laurashr@222.212.66.128) |
| 21:23.20 | zobia | Hello everyoen |
| 21:23.23 | zobia | everyone |
| 21:23.58 | zobia | anyone knows where there's zaptel 1.4.8 or 1.4.9 rpm for centos 4? |
| 21:24.11 | zobia | i search it for long time . still no luck/ |
| 21:24.39 | zobia | and if anyone can share the zaptel's .spec for 1.4.8 or 1.4.9 i am so appreciate. |
| 21:24.39 | _ShrikE | zobia: its not that hard to build it from source |
| 21:25.27 | zobia | _ShrikE: i use rpmbuild and checkinstall to build it . both failed. please help |
| 21:26.11 | zobia | _ShrikE: i am not good at the .spec file making. if you have any idea please let me know. thanks. |
| 21:26.44 | fujin | what do you need an rpm for? just install from source ;> |
| 21:27.22 | zobia | fujin: i need to rpm to make a autointall cd for zaptel and asterisk kickstart cd. |
| 21:28.03 | fujin | ew |
| 21:28.20 | *** join/#asterisk wordzilla (n=me@d58-106-139-71.sbr4.nsw.optusnet.com.au) |
| 21:29.54 | *** join/#asterisk CVirus (n=GoD@196.205.192.125) |
| 21:36.14 | zobia | no one knows? or someone can tell me how to make or .spec or find a .spec for zaptel 1.4.8 or 1.4.9? |
| 21:41.06 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 21:45.53 | *** join/#asterisk simNIX (n=user@82-204-21-111.dsl.bbeyond.nl) |
| 21:46.02 | *** join/#asterisk glaz (i=strke@glaciuz.com) |
| 21:46.29 | simNIX | greetings |
| 21:46.38 | glaz | Hi |
| 21:46.50 | glaz | Is it possible to convert a Cisco 7970 Phone to SIP? |
| 21:47.03 | glaz | Not sure I am asking the right question at the right place |
| 21:47.14 | _ShrikE | glaz: the answer is yes |
| 21:47.47 | glaz | _ShrikE: thanks, I can't find any docs on this, but I guess if you say yes I'll find a way to do it. |
| 21:48.43 | simNIX | anone perhaps know url on how to setup Asterisk + spa2100 ? (I asume I dont need zaptel ?) |
| 21:48.50 | _ShrikE | glaz: cisco intentionally does not provide much information on that phone |
| 21:48.58 | glaz | _ShrikE: why is that? |
| 21:49.36 | _ShrikE | I guess they dont want it to be very easy to get it working on other platforms |
| 21:50.03 | glaz | you did it? |
| 21:50.05 | _ShrikE | but I have one that does work with * and sip. |
| 21:51.03 | glaz | ok, I read that some people made it working with chan_sccp |
| 21:51.30 | _ShrikE | With that phone, sccp is easier than sip. IMHO |
| 21:51.37 | [TK]D-Fender | simNIX, start with : |
| 21:51.38 | [TK]D-Fender | ~book |
| 21:51.39 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 21:51.40 | [TK]D-Fender | ^^^^ |
| 21:52.07 | [TK]D-Fender | simNIX, And as for the SPA, there are dozens of quicky guides for devices like that. And Indeed you do not require Zaptel for just * + that ATA |
| 21:52.37 | [TK]D-Fender | simNIX, You need Zaptel for TDM hardware, MeetMe Conferences, and IAX2 Trunking Mode. |
| 21:53.10 | glaz | _ShrikE: I guess this is what I need: http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7970g_7971g-ge/english/5_0/sip/english/administration/guide/70sipag.pdf |
| 21:54.11 | high-rez | <PROTECTED> |
| 21:54.14 | high-rez | erps |
| 21:54.33 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-231.static.twtelecom.net) |
| 21:54.49 | CrazyTux | Hey guys has anyone intergrated an AGI script that simply connects to an HTTP (Server) For CNAM, |
| 21:55.01 | CrazyTux | That they wouldnt mind emailing me, or helping me out :) |
| 21:55.21 | CrazyTux | I looked into FastAGI -- Perl, etc, but it does not seem to be to complete. |
| 21:57.03 | glaz | _ShrikE: how did you do it? quickly |
| 21:58.40 | simNIX | Fender Ty |
| 21:59.15 | _ShrikE | glaz: i'm not sure that can be done "quickly" :) |
| 21:59.20 | glaz | when they say United IP phone, are they meaning SIP ? |
| 21:59.37 | _ShrikE | glaz: lemme see if I can find my old configs |
| 22:00.00 | glaz | _ShrikE: great! |
| 22:01.19 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 22:01.39 | EmleyMoor | Are there known problems with using Festival with cache? |
| 22:02.19 | EmleyMoor | I ask because I have it on and my FXO had a stuck call on it - stuck in a Festival call |
| 22:02.41 | EmleyMoor | Only noticed because a call that would normally have gone over the FXO went over IAX instead |
| 22:02.59 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 22:03.48 | EmleyMoor | I did read it may be a problem - but short of conducting some more testing, I will be unable to tell |
| 22:04.52 | EmleyMoor | What the asterisk console logs - is it kept anywhere? |
| 22:08.03 | glaz | _ShrikE: any luck? |
| 22:09.24 | *** join/#asterisk BobLutz (n=stansmit@d60-65-93-136.col.wideopenwest.com) |
| 22:10.00 | plasmid | when using a regular phone with a sipura box, is there a way to put the caller on hold? I dont' have one of those voip phones yet. |
| 22:11.57 | *** join/#asterisk russellb (n=mobile@asterisk/developer-and-stable-maintainer/drumkilla) |
| 22:11.58 | *** mode/#asterisk [+o russellb] by ChanServ |
| 22:25.33 | [TK]D-Fender | plasmid, yes, go read the users guide to see how. Its a * code. |
| 22:25.43 | [TK]D-Fender | plasmid, [flash] + feature code |
| 22:26.03 | plasmid | [TK]D-Fender, i've been trying to find this guide... lol.. because that's not the only feature i would like to use. I also want to transfer, etc... |
| 22:26.13 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:26.35 | plasmid | freepbx.org documentation doesn't show the codes... is there a wiki? |
| 22:27.15 | [TK]D-Fender | plasmid, its not FreePBX's job to document other hardware... |
| 22:27.41 | [TK]D-Fender | plasmid, That's like asking Ford for a 1989 Tercel owner's guide... |
| 22:28.07 | plasmid | [TK]D-Fender, ok. Is there a google garage sale for these unsupported codes? |
| 22:28.17 | [TK]D-Fender | plasmid, And no, Sipura/Linksys does not have a WIKI |
| 22:28.34 | plasmid | so in other words, regular phones cannot put the caller on hold. |
| 22:28.39 | [TK]D-Fender | plasmid, http://www.google.ca/search?hl=en&q=Sipura+ATA+users+guide&btnG=Google+Search&meta= |
| 22:28.54 | *** join/#asterisk jay21d (n=jjohnson@pool-71-180-24-188.tampfl.fios.verizon.net) |
| 22:29.37 | plasmid | hmm.. sounds like I have to dwell with PAP2T config codes. |
| 22:29.44 | plasmid | [TK]D-Fender, thanx for the info. |
| 22:30.37 | [TK]D-Fender | plasmid, And amazingly its onlyt he first link :) |
| 22:30.46 | [TK]D-Fender | plasmid, You should try a little harder... |
| 22:31.14 | plasmid | [TK]D-Fender, i was trying harder but the other way. Thinking it was a pbx documentation. |
| 22:31.16 | jay21d | Pocket Talk is a good SIP softphone that runs on Windows mobile |
| 22:31.49 | [TK]D-Fender | plasmid, Devices tend to have their own way of doing things. Read your device's manual first |
| 22:33.47 | plasmid | [TK]D-Fender, something aside... IS there a way to improve the quality of the calls? I heard that u can use a different codec at the expense of more CPU usage? |
| 22:34.09 | [TK]D-Fender | plasmid, if you're local to the server you should be using G.711 |
| 22:34.10 | plasmid | [TK]D-Fender, kinda newbish at this whole VOIP. |
| 22:34.41 | plasmid | [TK]D-Fender, and also Qos I take it. My router is crappy though at handling VOIP calls. Time for a new router I think. |
| 22:35.36 | plasmid | G.711a or G.71u? |
| 22:35.44 | plasmid | *G.711u |
| 22:36.07 | *** join/#asterisk xenonex (n=xenonex@82.200.211.5) |
| 22:37.15 | plasmid | Codec G.711a is used within Australia and Europe, while G.711u is used within US. nv <-- |
| 22:39.33 | *** join/#asterisk Dexter_81 (n=Dexter_8@host231-112-dynamic.3-87-r.retail.telecomitalia.it) |
| 22:39.46 | Dexter_81 | hi i'm italian |
| 22:40.16 | *** join/#asterisk russellb (n=mobile@asterisk/developer-and-stable-maintainer/drumkilla) |
| 22:40.17 | *** mode/#asterisk [+o russellb] by ChanServ |
| 22:41.16 | Dexter_81 | hi i'm italian, how to i can called a number using spoofing? |
| 22:41.59 | [TK]D-Fender | plasmid, Either depending on what the other side of most calls will be using. |
| 22:42.38 | plasmid | [TK]D-Fender, my PAP2T 26 pg guide does not mention a single tidbit on these codes u mentioned. |
| 22:42.40 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 22:43.15 | plasmid | [TK]D-Fender, i do remember that to call someone on the network I have to put the extension and then #. |
| 22:43.59 | *** join/#asterisk paci` (n=paci@cpe-066-057-116-114.nc.res.rr.com) |
| 22:44.03 | paci` | hey you guys around |
| 22:45.27 | [TK]D-Fender | plasmid, notmal people don't use residential devices like that for business-like functions like transfer/hold |
| 22:45.42 | paci` | how would i go about configuring asterisk to use skype |
| 22:46.20 | plasmid | [TK]D-Fender, agreed. |
| 22:46.38 | plasmid | ack.. skype.... they privy into your conversations. |
| 22:46.44 | plasmid | read the small print. |
| 22:46.45 | [TK]D-Fender | ~skype |
| 22:46.47 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, all of which are poor implementations of convoluted hacks. In general, forget about using Skype with Asterisk... |
| 22:47.25 | paci` | aw. |
| 22:47.35 | paci` | whats a good VoIP service to use with asteris |
| 22:47.43 | [TK]D-Fender | ~itsp |
| 22:47.44 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 22:47.45 | paci` | asterisk* |
| 22:48.00 | paci` | ~itsplist-us |
| 22:48.00 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, or http://www.jnctn.com, or http://www.bandwidth.com, or http://vitelity.net |
| 22:51.36 | paci` | [TK]D-Fender, if you odn |
| 22:51.37 | paci` | er |
| 22:51.48 | paci` | don't mind the highlight*, do you know any ones with a fixed rate by hand/ |
| 22:52.25 | [TK]D-Fender | paci`, most offer a variety of services. Go read |
| 22:52.25 | *** join/#asterisk RobH (n=RobH@69.18.84.191) |
| 22:52.49 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 23:00.02 | paci` | what would the conf file for using a bluetooth gateway be |
| 23:02.56 | paci` | or better yet, what is a free incoming gatewat |
| 23:02.59 | paci` | gateway* |
| 23:05.13 | [TK]D-Fender | paci` : gateway from where? |
| 23:05.23 | paci` | [TK]D-Fender, |
| 23:05.27 | paci` | like, not sure how to explain it |
| 23:05.30 | paci` | I think FWD has it |
| 23:05.33 | paci` | a free incoming number |
| 23:05.41 | [TK]D-Fender | paci`, www.ipkcall.com |
| 23:05.46 | paci` | that was it |
| 23:05.48 | paci` | ipkall |
| 23:16.50 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 23:18.25 | eric2 | I just had asterisk die on me |
| 23:18.30 | eric2 | out of the blue |
| 23:19.07 | drmessano | Did ya restart it? |
| 23:19.22 | eric2 | yes, but luckily I was on the phone when it happened |
| 23:19.31 | eric2 | otherwise it would still be dead |
| 23:19.36 | eric2 | very strange |
| 23:20.18 | drmessano | Waht time zone are you in? |
| 23:20.25 | eric2 | e.s.t |
| 23:20.33 | eric2 | new york/toronto |
| 23:20.37 | drmessano | Was it a business call? |
| 23:20.44 | eric2 | n |
| 23:20.48 | drmessano | oh |
| 23:21.00 | drmessano | Asterisk deserves the weekend off too |
| 23:21.00 | eric2 | but it's been cutting out for a second every now and then too |
| 23:21.11 | eric2 | no, residential customers are on the system too |
| 23:21.18 | eric2 | so it's gotta work 24/7 |
| 23:23.06 | drmessano | Anyone using a softphone on their blackberry? |
| 23:24.13 | Qwell | drmessano: buy me one, and I'll set one up |
| 23:24.16 | drmessano | lol |
| 23:24.23 | Qwell | You're a Dr. You can afford it. |
| 23:24.33 | drmessano | What color? |
| 23:24.40 | Qwell | orange |
| 23:24.50 | Qwell | no such thing? better make it a Neo1973 then |
| 23:24.55 | Qwell | those come in orange. kthx |
| 23:25.16 | [TK]D-Fender | Qwell, bkruse wrote an IAX one for it ;) |
| 23:25.23 | Qwell | [TK]D-Fender: I know |
| 23:25.27 | [TK]D-Fender | Qwell, That'll be 500$ please ;) |
| 23:25.31 | drmessano | lol |
| 23:26.35 | drmessano | I don't want a phone built on open specs.. "Open" is overrated |
| 23:26.47 | drmessano | OMG |
| 23:26.49 | drmessano | Listen to me |
| 23:26.55 | Qwell | Mr. DRM |
| 23:26.56 | drmessano | One day of using Vista.. and look at me |
| 23:27.02 | Qwell | GTFO |
| 23:27.17 | drmessano | Actually |
| 23:27.55 | drmessano | I sit down at my new desktop at work for the FIRST TIME.. and find a bug in Outlook due to a patch from Tuesday that makes it almost unusable |
| 23:27.57 | drmessano | Welcome to Windows |
| 23:28.26 | drmessano | I want a showstopper bug in 1.6 beta 6 |
| 23:28.32 | drmessano | Just so I feel better |
| 23:29.40 | drmessano | wow |
| 23:29.51 | drmessano | Les.net just added something called "Virtual PBX" |
| 23:30.05 | Qwell | eh? |
| 23:30.31 | alrs | drmessano: that fails to shock |
| 23:30.48 | drmessano | You dont even know what the feck it is yet.. |
| 23:31.01 | drmessano | It lets you create a 0 thru 9 single depth IVR.. you record the greeting and prompts, and point each option a different peer |
| 23:31.25 | drmessano | So you can point your DID or DIDs to the IVR and route calls to each peer based on the IVR response |
| 23:32.05 | drmessano | Very cool for say multiple stores with one main number |
| 23:32.27 | drmessano | So you dont have one PBX switching the calls |
| 23:32.28 | znoG_ | Question: I have a SIP client configured to only use ulaw/alaw. I tried to call someone and Asterisk says "No compatible codecs found". I enabled SIP debug and I see this line: Capabilities: us - 0x0 (nothing), peer - audio=0xc (ulaw|alaw) |
| 23:32.40 | znoG_ | why would Asterisk see NO codecs? |
| 23:34.43 | [TK]D-Fender | znoG_, Because maybe you didn't set any in your sip.conf |
| 23:35.35 | znoG_ | [TK]D-Fender: i am using res_config_ldap, however, for that user I have set the allowed codecs to alaw and ulaw, and disallow all |
| 23:35.44 | znoG_ | but maybe the allow line is not doing its thing |
| 23:36.08 | [TK]D-Fender | znoG_, if your disalloy=all follows it then it will override your allows. Order counts |
| 23:37.07 | Qwell | I don't know if you can order with ldap... |
| 23:37.56 | znoG_ | you can't |
| 23:38.06 | znoG_ | it's definately reading the disallowed codecs attribute |
| 23:38.12 | znoG_ | but the allowed one... doesn't look like it |
| 23:38.54 | [TK]D-Fender | znoG_, It can be reding BOTH, but like I said, order counts. |
| 23:39.08 | [TK]D-Fender | znoG_, If * processes the disallow second then you have no codecs. |
| 23:39.22 | znoG_ | [TK]D-Fender: yep, i understand that, however as Qwell said you can't order with LDAP |
| 23:39.57 | [TK]D-Fender | znoG_, This is that unique position they call "SOL" |
| 23:42.56 | drmessano | Remove the codecs youre not using |
| 23:43.34 | drmessano | Who is Strom Carlson? |
| 23:44.55 | Qwell | drmessano: Strom_C |
| 23:45.36 | drmessano | He just posted something on twitter about having some Digium shirts to get rid of |
| 23:45.38 | drmessano | :/ |
| 23:47.29 | St1ckm4n | does anyone here do any realtime monitoring of call center agents with asterisk 1.4? |
| 23:48.01 | znoG_ | [TK]D-Fender: SOL? |
| 23:48.11 | drmessano | Shit Outta Luck |
| 23:48.17 | BobLutz | whoa |
| 23:48.33 | drmessano | BobLutz: Are you 14? |
| 23:48.59 | BobLutz | drmessano, are you a doctor? |
| 23:49.09 | drmessano | Asked you first |
| 23:49.12 | BobLutz | damn |
| 23:49.13 | BobLutz | no |
| 23:49.30 | drmessano | Then i'm going with my second guess of 74 |
| 23:49.38 | BobLutz | lol |
| 23:49.56 | jameswf | ~drmessano |
| 23:49.56 | jbot | [drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway |
| 23:50.26 | BobLutz | lol, i dont remember the OB/GYN part |
| 23:50.35 | paci` | whats better, sip/iax |
| 23:50.42 | jameswf | Iax2 |
| 23:50.50 | jameswf | ~iax2 |
| 23:50.51 | jbot | well, iax2 is http://www.voip-info.org/wiki-IAX |
| 23:50.52 | [TK]D-Fender | St1ckm4n, Yes. |
| 23:51.19 | paci` | well, |
| 23:51.23 | drmessano | Links that point to the voip-info should be banned |
| 23:51.25 | paci` | ipkall offers sip and iax |
| 23:51.28 | paci` | which would be better? |
| 23:51.34 | [TK]D-Fender | paci`, SIP generally |
| 23:52.01 | paci` | what is FWD's sip server? |
| 23:52.02 | jameswf | unless you use nat then sip suxors |
| 23:52.38 | St1ckm4n | [TK]D-Fender: are you using a program that parses the manager api output or are you storing/reading it from a db |
| 23:52.45 | drmessano | fwd.pulver.com |
| 23:53.03 | jameswf | ~fwd |
| 23:53.04 | jbot | [~fwd] Free World Dialup, created by Jeff Pulver, is a free SIP server for P2P style that does not involve the PSTN (there is a charged option for this as well though). http://www.freeworlddialup.com/ |
| 23:53.26 | drmessano | ~ipkall |
| 23:53.28 | drmessano | :( |
| 23:53.43 | `Sauron | FWD offers IAX2 as well |
| 23:53.54 | drmessano | It doesnt work |
| 23:53.58 | paci` | `Sauron, FWD != IPKall |
| 23:54.14 | `Sauron | I never said they were the same |
| 23:54.25 | `Sauron | It was an FYI. :p |
| 23:54.28 | drmessano | FWD IAX2 is so unreliable, they even tell you not to use it |
| 23:54.34 | paci` | ah |
| 23:54.39 | [TK]D-Fender | St1ckm4n, Yup, I parse it the very dirty way. |
| 23:54.50 | paci` | mmm |
| 23:55.13 | paci` | how exactly |
| 23:55.15 | `Sauron | Hum, I thought iax was suggested over sip, at least back in the day when I set it up. |
| 23:55.22 | `Sauron | shrug |
| 23:55.30 | paci` | ;register => 1234:password@mysipprovider.com |
| 23:55.32 | drmessano | Its ALWAYS been "experimental" with them |
| 23:55.34 | paci` | ok, so that would be |
| 23:55.48 | paci` | register => NUMBER:NUMBER@fwd.pulver.com |
| 23:55.50 | paci` | ? |
| 23:55.50 | drmessano | FWD has never pushed using IAX2 over SIP |
| 23:56.10 | drmessano | number:password |
| 23:56.13 | St1ckm4n | [TK]D-Fender: I've been looking at FOP but it seems buggy, I've written a php page that connects to the manager api and just parses the show agents output but I don't like having to continuously query the server, was it a pain in the ass to do it the dirty way? |
| 23:56.33 | `Sauron | hum |
| 23:56.36 | `Sauron | ohwell |
| 23:56.38 | paci` | drmessano, the password to my FWD account? |
| 23:56.44 | drmessano | yes |
| 23:57.00 | paci` | would the NUMBER be the username |
| 23:57.03 | paci` | or sip # |
| 23:57.08 | drmessano | No, it would be the number |
| 23:57.54 | [TK]D-Fender | St1ckm4n, I *do* continuously query the server... thats the "dirty" part (in addition to the fact its pur text-parsing) |
| 23:57.56 | [TK]D-Fender | pur* |
| 23:58.12 | *** join/#asterisk frogonwheels (n=michaelg@203.59.141.93) |
| 23:58.26 | paci` | how can I test asterisk if its conected to my sip server |
| 23:58.32 | paci` | I havn't used it in a long time |
| 23:58.43 | drmessano | sip show registry |
| 23:59.07 | frogonwheels | I have asterisk running on openwrt: |
| 23:59.24 | paci` | [Mar 14 19:59:14] NOTICE[23880]: chan_sip.c:7425 sip_reg_timeout: -- Registration for '903316@mysipprovider.com' timed out, trying again (Attempt #3) |
| 23:59.25 | paci` | rawr |
| 23:59.26 | St1ckm4n | [TD]D-Fender: do you think there is any stability issues if you have 5 pc's continually querying the server ever 3-5 seconds |
| 23:59.35 | frogonwheels | if I start it up - I get no ctl file - but if I use -vv or -d it does create one. |
| 23:59.38 | paci` | oh |
| 23:59.38 | paci` | duh |
| 23:59.39 | paci` | rofl |
| 23:59.42 | paci` | i didnt change the host |
| 23:59.43 | frogonwheels | any clues why? |