| 00:00.21 | d-k-t | indeed |
| 00:00.58 | coppice | well, I should actually bother to get permanent resident status in HK, instead of just unconditional stay. then there are other visa options. I just haven't faced the hassle of the permanent residence process |
| 00:01.35 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 00:03.22 | coppice | well, my olympics tickets are for the events in HK, so I don't need a visa to see those :-) |
| 00:03.43 | d-k-t | I'm skipping it all |
| 00:03.52 | d-k-t | it's bound to be a nightmare |
| 00:04.25 | coppice | it will be fine in HK. I think Beijing is gonna end up like Moscow, clouded in political stupidity |
| 00:04.42 | d-k-t | might be able to get some cheap flights out of china around that time, the airlines will need to find a way to get some people on planes going out to bring in more people |
| 00:05.16 | d-k-t | anyway, time to go to work |
| 00:05.41 | d-k-t | cya |
| 00:05.48 | coppice | bye |
| 00:08.55 | tzafrir_laptop | coppice, here? |
| 00:09.12 | coppice | where? |
| 00:13.52 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 00:15.17 | *** join/#asterisk [cfdisk] (n=cfdisk@68-116-156-85.dhcp.ftwo.tx.charter.com) |
| 00:17.43 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 00:19.21 | resin0008 | tzafrir_laptop |
| 00:19.30 | resin0008 | do you know how hints work? |
| 00:21.00 | tzafrir_laptop | basically, why? |
| 00:21.08 | resin0008 | cuz it doesnt make sense to me at all |
| 00:21.10 | coppice | go on. give him a clue :-) |
| 00:21.22 | resin0008 | i don't understand how that should be in the diaplan |
| 00:21.43 | resin0008 | i would think, asterisk would need a separate realtime element to keep things hinted |
| 00:22.12 | resin0008 | but apparantly it does work from the diaplan, so i'll just explain what i'm trying to do |
| 00:22.40 | resin0008 | and a quick question, when does the "hint" priority get executed |
| 00:22.41 | resin0008 | ? |
| 00:23.09 | resin0008 | and what does it do when it does get executed |
| 00:24.50 | resin0008 | is it simple? |
| 00:25.06 | resin0008 | or, do you know where theres clear documentation on how it works |
| 00:25.40 | *** join/#asterisk nobesnickr (n=pmccaffr@ip72-201-157-30.ph.ph.cox.net) |
| 00:27.12 | tzafrir_laptop | "hint" is a priority (actually it is translated to priority -1, but that's an implementation issue) |
| 00:27.44 | resin0008 | ok, well, that at least fits it into the way a diaplan works better |
| 00:27.48 | nobesnickr | hello all, i seem to be having a strange issue with MeetMe where after about 60 seconds the conference simply dies with "Quitting Time..." as the only CLI that show |
| 00:27.58 | resin0008 | which is "step" based |
| 00:28.12 | jaytee | still confuses me |
| 00:28.21 | resin0008 | yah, well, me too |
| 00:28.26 | resin0008 | tzafrir, where did you read that? |
| 00:33.23 | resin0008 | tzafrir_laptop: where did you read that info |
| 00:34.25 | tzafrir_laptop | resin0008, probably on some mailing-list post :-) |
| 00:34.45 | resin0008 | you got any links to any good documentation on hints |
| 00:34.56 | resin0008 | developer type docs |
| 00:37.10 | nobesnickr | does anyone have any experience with ztdummy or meetme? |
| 00:37.18 | resin0008 | dangin |
| 00:37.23 | resin0008 | dang it |
| 00:37.28 | resin0008 | tzafrir does :) |
| 00:37.41 | tzafrir_laptop | One of the first hits in the Yahoo search for "asterisk hints": http://www.voip-info.org/wiki/index.php?page=Asterisk+standard+extensions |
| 00:37.43 | resin0008 | but i was lookin for his help on somethin else |
| 00:37.56 | resin0008 | yes thanks, thats useless |
| 00:38.03 | resin0008 | lol |
| 00:38.39 | resin0008 | it doesnt say WHEN or HOW the hint priority is executed |
| 00:39.17 | jaytee | maybe if you ask in asterisk-dev? |
| 00:39.43 | tzafrir_laptop | It is not something that is executed. It is a device (or list of devices) that are connected with that extension |
| 00:42.07 | resin0008 | but every other line in the dialplan sits dormant until a call steps into it. |
| 00:43.02 | resin0008 | so if it's a special exception-to-the-rule thing that's ok, i just want to know that i'm not crazy for thinking it's unusual |
| 00:43.04 | resin0008 | and confusing |
| 00:43.13 | resin0008 | and i still can't find any official documentation |
| 00:43.38 | resin0008 | here's my purpose |
| 00:43.48 | resin0008 | Asterisk is stepping through the diaplan and the agent presses #0# which parks a call in slot 1 because of my dialplan. |
| 00:43.56 | resin0008 | This will change the state of the parking space "1" from "available" to "in use". |
| 00:44.15 | resin0008 | I want to make my light for my second line key on all my phones blink . This should be simple right? |
| 00:45.43 | resin0008 | i don't think it's appropriate for asterisk-dev |
| 00:45.49 | resin0008 | u know what, on second thought, it might be |
| 00:47.33 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 00:48.28 | *** join/#asterisk Shotygun (n=thorn@82.166.246.116) |
| 00:51.45 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 00:51.48 | _ShrikE | resin0008: you may be able to accomplish that with devstate |
| 00:53.27 | resin0008 | please explain |
| 00:54.33 | _ShrikE | devstate lets you get or set device state in the dialplan and subscribe to it with a hint |
| 00:55.11 | resin0008 | well, that sound proper, but first i would like to get an idea of how a hints work, then we can move on to subscribing to it |
| 00:55.30 | *** part/#asterisk RoyK (n=roy@ip-88-4-149-91.dialup.ice.no) |
| 00:56.20 | _ShrikE | extstate could work as well |
| 00:56.40 | resin0008 | _ShrikE: can you walk me through understanding ay of this? |
| 00:57.41 | _ShrikE | there is documentation out there that explains these things much better than I can. Try the wiki and google. |
| 00:57.57 | plla | Hello, is there a template system in users.conf ? |
| 00:58.51 | *** join/#asterisk Gwayne (n=Gwayne@bb116-14-25-104.singnet.com.sg) |
| 00:59.07 | resin0008 | _ShrikE ? Do you understand how it works and have you used it? Or are you just saying that "theoretically based on what the wiki says hints and devstate should do, then I should be able to do what I described?" |
| 00:59.10 | resin0008 | cuz i've read the wiki |
| 01:01.33 | _ShrikE | I use devstate quite a bit yes. |
| 01:01.52 | resin0008 | please explain how you use it if you don't mind |
| 01:04.15 | _ShrikE | I use it in lots of ways, voicemail.. busy lamp fields.. etc.. I have not used extstate that much however. |
| 01:04.46 | resin0008 | can you please show some diaplan that demonstrates it's usage? |
| 01:05.58 | _ShrikE | http://www.voip-info.org/wiki/view/Asterisk+func+Devstate |
| 01:06.46 | resin0008 | i meant your own diaplan |
| 01:09.13 | _ShrikE | you dont need my dialplan. read up and write your own. |
| 01:11.01 | *** join/#asterisk jsmith (n=jsmith@72.21.36.138) |
| 01:11.01 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 01:11.05 | resin0008 | thanx |
| 01:11.39 | jsmith | resin0008: To answer your question though -- yes, you can have extension 123 in multiple places in your dialplan, and in sip.conf you can specify which context the hints are located in |
| 01:12.02 | resin0008 | Ahhhhh, that might make a big difference in my understanding, let me ponder that |
| 01:12.40 | *** join/#asterisk digitalirony (n=eric@216.207.245.1) |
| 01:12.44 | digitalirony | hello |
| 01:12.47 | resin0008 | every time asterisk changes the state of SIP/jsmith, it looks in sip.conf to see where the hints are. it goes to that context and processes all hints for that extension? |
| 01:12.49 | jsmith | resin0008: See the "subscribecontext" setting in sip.conf |
| 01:12.55 | digitalirony | question |
| 01:13.17 | resin0008 | so would it not make sense to have a context setup specifically to list all your hints and do nothing else |
| 01:13.23 | resin0008 | just to abstract it away from everything else? |
| 01:13.27 | digitalirony | instead of setting Ring_DEBOUNCE in the wctdm24xxp.h file how can you pass this to zaptel to test settings? |
| 01:13.37 | resin0008 | why not abstract it to it's own file "subscriptions.conf |
| 01:13.45 | jsmith | resin0008: Pretty close... when a particular peer subscribes to extension 123's state, Asterisk looks at the subscribecontext setting for that peer (or the global one), and uses that to find the hint |
| 01:13.48 | resin0008 | because it has absolutely nothing to do with the diaplan |
| 01:14.16 | jsmith | resin0008: It's up to you -- you can put all your hints in a single context, or have them scattered throughout your dialplan... it's up to you |
| 01:14.16 | resin0008 | how does a peer subscribe to SIP/jsmith's state |
| 01:14.52 | resin0008 | i mean, thats a misleading concept. really, you tell asterisk who gets the updates in the dialplan with your hints statements don't you |
| 01:14.54 | jsmith | resin0008: It depends on the phone... in essence, the phone sends a SIP SUBSCRIBE message to Asterisk saying "Hey, I wanna know when extension 123 changes state" |
| 01:15.09 | resin0008 | how would you even get your phone to do that? |
| 01:15.20 | resin0008 | in the polycoms, is there a section for that? |
| 01:15.24 | jsmith | resin0008: Asterisk then keeps track of that, so it knows who to send a SIP message to when the state changes |
| 01:15.24 | _ShrikE | polycom refers to it as buddies |
| 01:15.33 | jsmith | resin0008: Yeah, buddies in the Polycom way... |
| 01:15.40 | _ShrikE | you set buddy watch to yes |
| 01:16.29 | resin0008 | ok, so i modify my phone configs so that they're subscribing to another devices state (whats the syntax, because i will be subscribing to a parking-space)? |
| 01:16.31 | digitalirony | anyone know how to pass options to zaptel when you load it I.E RING_DEBOUNCE for testing for the right setting? |
| 01:17.04 | resin0008 | then i put the subscribecontext=subscriptions |
| 01:17.41 | resin0008 | then in subscriptions i put "exten => something,hint,SIP/phone1&SIP/phone2etc |
| 01:17.53 | resin0008 | im almost more confused now |
| 01:18.34 | resin0008 | so the phone actually does send a subscribe message to asterisk for parking lot space 1. cool that makes sense |
| 01:18.35 | jsmith | resin0008: No, you've got the right idea |
| 01:19.02 | jsmith | resin0008: From the CLI, you can type "sip show hints" to see a list of who is subscribed to which hints |
| 01:19.13 | resin0008 | so why even need the diaplan |
| 01:19.28 | resin0008 | why can't asterisk just give my phone the update directly |
| 01:20.06 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 01:20.08 | resin0008 | If i have to configure all my phones to subscribe in their config files, why do i also have to do it from the diaplan. seems redundant |
| 01:20.43 | jsmith | resin0008: Well, it's because extensions in the dialplan don't have a state. To be able to determine state, you need to tie the extension to a device that *does* have state. |
| 01:20.46 | resin0008 | if there was another step in there, i would think it would be simple permission checking |
| 01:20.49 | _ShrikE | You are creating a hint in the dialplan so you have something to subscribe to with you phone. |
| 01:21.06 | resin0008 | oh, that might be it _ShrikE |
| 01:21.28 | resin0008 | thinking... |
| 01:21.57 | resin0008 | nah, asterisk natively aware of devices as devices |
| 01:21.57 | _ShrikE | devstate and extstate are tools that allow you to set/read the state of of devices or extensions via hints in the dialplan |
| 01:22.56 | resin0008 | SIP/5001 is a device, and it has a changing state while it is registered to the box, so asterisk could just send the notifications when this changes |
| 01:23.28 | jsmith | resin0008: Right... but phones don't subscribe to the state of a *phone*, they subscribe to the state of an *extension* |
| 01:23.39 | resin0008 | oh ok then |
| 01:23.50 | jsmith | resin0008: See, on lesser phones systems, a phone *is* an extension. But not with Asterisk. We're more flexible than that. |
| 01:24.09 | jsmith | resin0008: One extension might ring *two* phones, or two extensions might ring the *same* phone |
| 01:24.13 | resin0008 | i know i know, it's just understanding which one we're talking about and when thats tough for this particular topic |
| 01:24.28 | jsmith | resin0008: When I say *extension*, I mean a named set of actions in the dialplan. |
| 01:24.48 | jsmith | resin0008: When I say *phone* or *endpoint* or *device*, I'm talking about something external to Asterisk |
| 01:25.43 | resin0008 | alright, so it seems stupid for a phone to request the state of an "extension" in the diaplan |
| 01:26.25 | resin0008 | i would think asterisk should just interpret that to be the state of that "device" instead. |
| 01:27.13 | resin0008 | oh well, that's another discussion really |
| 01:27.45 | jsmith | Well, that's all find and dandy, but then you couldn't subscribe to the state of a parking lot, for example |
| 01:28.17 | jsmith | (or the state of something arbitrary, as you can with custom extension states) |
| 01:28.26 | resin0008 | well, anybody got the syntax of the polycom buddy statement? |
| 01:29.03 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 01:29.29 | iceyp | hey guys, any one know what causes this with an IAX2 call: CAUSE : No authority found |
| 01:29.34 | resin0008 | seems like you could embed the technology into there like: |
| 01:29.35 | resin0008 | <line1> |
| 01:29.35 | resin0008 | <subscribeuddy>SIP/5001</subscribebuddy> |
| 01:29.35 | resin0008 | </line1> |
| 01:29.43 | iceyp | looks to be an auth issue, but i cannot find any problems with the auth |
| 01:29.57 | jsmith | resin0008: You put a contact in the directory, and then enable buddy watch for that contact |
| 01:30.33 | resin0008 | or <subscribeuddy>IAX/5001</subscribebuddy> or <subscribebuddy>PL/5001</subscribebuddy> and then asterisk could interpret the part before the / |
| 01:30.55 | resin0008 | ok jsmith, thanks for your help |
| 01:31.09 | resin0008 | i think i get it now for the most part. is that documented anywhere in a thorough manner? |
| 01:31.15 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 01:31.16 | resin0008 | and don't say the wiki or google please lol |
| 01:31.23 | resin0008 | hang on iceyp, let me se |
| 01:31.40 | iceyp | thanks resin0008 |
| 01:31.42 | jsmith | resin0008: Right, but that's not the SIP way... Polycom doesn't know whether the phone is gonna be used with Asterisk or some other phone system. |
| 01:32.11 | iceyp | brb smoko |
| 01:32.14 | jsmith | iceyp: Yes, it's an authentication problem. Typically it means you don't have an IAX user (or friend) configured correctly on the receiving side |
| 01:33.01 | resin0008 | no but i mean, let polycom send a generic sip_subscribe , but have a delimiter that asterisk can recognize and use to split technology from UID |
| 01:33.51 | resin0008 | people who use asterisk just need to know, if i want my polycom to subscribe to extension "5000", i have to put sip_5000 in the config file or iax_5000 or meetme_5000 |
| 01:34.06 | resin0008 | cuz what they have to do now is WAY more convoluted and confusing |
| 01:34.32 | resin0008 | i mean, i came in and everyone was acting like this was simple, but it's not at all. there are several things that all have to be aligned |
| 01:34.52 | resin0008 | and its not documented anywhere |
| 01:35.15 | resin0008 | it's like SLA, theres 1 tutorial on it and it's incomplete and not intuitive at all |
| 01:35.29 | resin0008 | and thats why nobody uses it and everybody hates it |
| 01:36.00 | resin0008 | but ok, you've enlightened me a fair amount just now. I will go try to set this up real quick i think |
| 01:36.49 | resin0008 | the last thing i'll need to figure out is how to make a busy state for that contact in my directory cause my second line key to light up |
| 01:36.57 | resin0008 | im sure i'll be back for that |
| 01:38.21 | *** join/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 01:38.54 | Titanous | why am I getting 'IMAP user not set for mailbox 200' I'm trying to disable IMAP |
| 01:38.55 | jsmith | resin0008: Well, you're preaching to the choir when it comes to documentation. Feel free to join me in #asterisk-doc and help write Asterisk docs |
| 01:40.21 | *** join/#asterisk Entranced (n=entrance@191.23.119.70.cfl.res.rr.com) |
| 01:42.02 | [TK]D-Fender | resin0008: No thats not why noboody uses SLA. |
| 01:42.06 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
| 01:44.19 | [TK]D-Fender | resin0008: Ready to cope with some more "enlightenment"? |
| 01:44.48 | *** part/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 01:49.02 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 01:51.12 | MrNaz | ok |
| 01:51.27 | MrNaz | now that i have asterisk installed and running properly, how do i test it to see if its working? |
| 01:51.43 | iceyp | can someone make a guest connection for me to iax2 |
| 01:51.54 | iceyp | just want to test anonymous call and not sure how i can do this myself |
| 01:56.38 | [TK]D-Fender | MrNaz: place some calls |
| 01:57.12 | digitalirony | MrNaz: Digium Tech support tier 1 here msg me for help |
| 01:58.15 | jeev | Fender, the replacement WIP330 works pretty damn good. |
| 01:58.46 | MrNaz | [TK]D-Fender to be honest, i dont even know how to do that... should i download a softphone? do i need to create accounts first? |
| 01:59.13 | [TK]D-Fender | MrNaz: No I'm quite sure you can download a softphone without even having heard about *. |
| 01:59.27 | [TK]D-Fender | jeev: Congratulations |
| 01:59.38 | MrNaz | [TK]D-Fender no i mean do i need to create accounts on my asterisk server first? |
| 01:59.48 | [TK]D-Fender | MrNaz: Before what? |
| 02:04.26 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 02:07.48 | *** join/#asterisk brendan_ (n=brendan@72.15.28.7) |
| 02:08.38 | brendan_ | hello, i'm trying to get a custom subroutine to run on in incoming call so i can set the cid for certain numbers |
| 02:09.31 | brendan_ | the [ext-did] section there is a include => ext-did-custom |
| 02:09.54 | brendan_ | so i put my GoSub in the [ext-did-custom] section in my extensions_custom.conf |
| 02:10.28 | brendan_ | but the gosub never runs, if i put the exact same line in the [ext-did] section, it works properly |
| 02:18.12 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
| 02:18.44 | jaytee | extensions_custom.conf? that's not a standard Asterisk config file. Are you running trixbox or AsteriskNOW? |
| 02:19.03 | Qwell | ~freepbx |
| 02:19.05 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 02:30.30 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 02:35.40 | *** join/#asterisk BeeBuu (n=beebuu@218.13.82.138) |
| 02:36.09 | BeeBuu | is meetingroom same as confenrence? |
| 02:36.24 | [TK]D-Fender | BeeBuu: Where aer you seeing this term? |
| 02:36.46 | BeeBuu | somewhere |
| 02:37.13 | [TK]D-Fender | BeeBuu: "somewhere"? You'll have to do better than that. |
| 02:37.21 | BeeBuu | [TK]D-Fender: no idea about that |
| 02:37.26 | JT | indeed |
| 02:37.36 | JT | i was going to ask where he was seeing "meetingroom" |
| 02:38.15 | BeeBuu | trixbox or else,i forgot |
| 02:39.09 | Entranced | web-meetme? |
| 02:39.49 | BeeBuu | can i 3-ways talk by confenrence? |
| 02:39.58 | [TK]D-Fender | BeeBuu: Feel free to ask again when you know what it is you're referring to |
| 02:40.13 | Entranced | you can 69-way if you want |
| 02:40.32 | drmessano | Oh, it's BeeBee |
| 02:40.35 | [TK]D-Fender | BeeBuu: what "conference" are you referring to? What do you define "3-way" as exactly as well? |
| 02:40.36 | BeeBuu | [TK]D-Fender: A call Bï¼and A want C join ,how to do that? |
| 02:40.41 | Entranced | if your box can handle it |
| 02:40.50 | BeeBuu | drmessano: hello.nice to meet you. |
| 02:40.55 | [TK]D-Fender | BeeBuu: Qhat phone is "a" using? |
| 02:41.02 | [TK]D-Fender | What* |
| 02:41.20 | BeeBuu | what phone? |
| 02:41.34 | JT | yes, the thing you talk into and listen to |
| 02:41.36 | BeeBuu | any options? |
| 02:41.43 | [TK]D-Fender | BeeBuu: This was not a complex question. What kind of phone is person "A" using? |
| 02:41.58 | Entranced | is it rotary ? |
| 02:42.03 | [TK]D-Fender | ... |
| 02:42.24 | LiNeTuX|Home | click click click click |
| 02:42.24 | [TK]D-Fender | stabs Entranced in the eye with a rusty spork |
| 02:42.32 | Entranced | hehe ..ouch! |
| 02:43.29 | jaytee | a rusty spork? I've never seen a metal spork. All the ones I've ever seen are plastic. |
| 02:44.00 | LiNeTuX|Home | jaytee: You've never been camping, eh? |
| 02:44.12 | Entranced | he made it when spending time in the penitentiary |
| 02:44.34 | jaytee | of course I've been camping, I've just never seen a metal spork though |
| 02:46.18 | LiNeTuX|Home | jaytee: metal rusty sporks are what you make out of spoons when you're bored on the trip |
| 02:47.20 | jaytee | I'll have to remember that. Usually when I think of sporks I think of all the times the assclown at the drive thru at Taco Bell forgets to put on in my bag for my enchirito. |
| 02:47.25 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 02:48.22 | LiNeTuX|Home | heh |
| 02:48.30 | Entranced | jaytee, mmmm enchirito! |
| 02:49.29 | JT | jaytee: i've seen a few metal sporks |
| 02:49.32 | LiNeTuX|Home | (_._) :o) |
| 02:49.40 | JT | they were stainless steel and not rusty though |
| 02:49.44 | jaytee | I miss the original enchirto. the enchilada sauce was better and it had black olives on it but they stopped putting those on in the 80's. Then they did away with it completely for awhile but brought them back finally. |
| 02:50.35 | jaytee | it wasn't like I didn't believe that metal sporks existed. I'm sure you can get one at any Eddie Bauer but I'd just never seen a metal one. no big deal. |
| 02:51.15 | *** join/#asterisk javb (n=valdezba@adsl-246-122.tricom.net) |
| 02:51.19 | Entranced | I think that we should continue to explore the non-metalic sporks a bit more |
| 02:51.28 | LiNeTuX|Home | jaytee: you haven't seen a real one until you've seen the drunk evil clown fliling away at one |
| 02:51.53 | javb | hi, i set logger.conf, without verbose and debug, but im still getting this in console, any help? http://pastebin.com/m60a99c28 |
| 02:52.19 | jaytee | man I wish I could find more shows or pictures of Marjorie Monaghan. She is soooooo fine! |
| 02:53.20 | LiNeTuX|Home | http://images.google.com/images?hl=en&q=%22Marjorie%20Monaghan%22&um=1&ie=UTF-8&sa=N&tab=wi |
| 02:53.26 | Entranced | salmonella tomatoes ! |
| 02:53.49 | jaytee | Oh!!!! Bless you, my son!!!! |
| 02:54.40 | Entranced | fap fap |
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| 03:42.19 | *** join/#asterisk xpot (n=xpot@c-24-10-234-202.hsd1.ut.comcast.net) |
| 03:49.54 | *** join/#asterisk rootlogin (n=root@saturn2.franken.de) |
| 03:55.06 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:03.20 | xpot | anyone what this error is: res_musiconhold.c:71: | dahdi_compat.h:27:24: error : dahdi/user.h: No such file or directory (tonzone.h either) |
| 04:03.20 | xpot | I can use pastebin with actual if this doesn't help |
| 04:08.13 | *** part/#asterisk resin0008 (n=resin000@7.218.204.68.cfl.res.rr.com) |
| 04:10.39 | mosty | dahdi is what zaptel was renamed to |
| 04:10.54 | mosty | do you have dahdi installed? |
| 04:11.31 | jsmith | xpot: It's a known issue, that should hopefully be fixed in the morning |
| 04:12.23 | JT | worst name ever :P |
| 04:12.57 | jsmith | I don't particularly care for the name either... but it is what it is |
| 04:13.06 | JT | heh |
| 04:14.07 | *** join/#asterisk MrNaz (n=naz@ppp121-44-202-83.lns3.mel4.internode.on.net) |
| 04:14.19 | *** join/#asterisk logi4023 (n=logi4021@76-10-149-62.dsl.teksavvy.com) |
| 04:14.56 | logi4023 | how do I the following: ' I want * to do callback using the callerid number of the caller' |
| 04:15.03 | jblack | seriously? |
| 04:15.13 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 04:15.28 | logi4023 | yes |
| 04:15.34 | *** join/#asterisk bkruse (n=bkruse@69.73.127.92) |
| 04:15.34 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 04:15.34 | jblack | logi4023: Setup up an agi that makes a callback file. You'll find agi files in the book on page.. |
| 04:15.47 | xpot | jsmith: thank you... I will try again tomarrow -=0) |
| 04:16.10 | jblack | 207 on, and call files on 306. |
| 04:16.34 | jblack | Anyways, zaptel is seriously named dahdi now? |
| 04:16.43 | jsmith | jblack: Yes |
| 04:16.57 | xpot | mosty: no, I do not have dahdi installed... don't have zaptel cards |
| 04:17.23 | jblack | hopes no one decides to rename asterisk to mahmi. |
| 04:17.36 | jsmith | jblack: To make a long story short, the zaptel name was trademarked by a guy selling telephone calling cards. He was *very* patient with us while we scrambled to find a new name that wasn't trademarked, the web sites were available, etc. |
| 04:17.57 | jblack | yeah, that sounds close enough for trademark protection. |
| 04:18.02 | jblack | And it's a good reason for a change. |
| 04:18.17 | jsmith | jblack: But it boils down to the fact that we respect other people's copyrights and trademarks, and hope others would extend us the same courtesy |
| 04:18.52 | jblack | No project changes a project name without a good reason. I understand. |
| 04:20.07 | jblack | Has anyone submitted a bugfix yet with "who's your dahdi" signature? |
| 04:20.25 | jsmith | jblack: Yup... the joke is getting really old |
| 04:20.39 | nick125 | But it's still kind of funny :P |
| 04:21.00 | jsmith | cries when he thinks of how much documentation he'll have to re-do |
| 04:21.13 | nick125 | jsmith: sed |
| 04:21.29 | jsmith | nick125: It's not that easy... |
| 04:22.30 | jsmith | nick125: Especially on things like training presentations, etc. |
| 04:23.01 | nick125 | Ah, yeah. That really has to suck. |
| 04:25.11 | jblack | It'll get older, and older, and older. |
| 04:26.22 | logi4023 | how do i 'issue a callback based on callerid of callerid using dialplan only' |
| 04:26.36 | jblack | logi4023: You don't. |
| 04:26.47 | jblack | You use the method I explained to you something like 10 minutes ago. |
| 04:27.35 | jsmith | logi4023: You *can* do it from the dialplan only, but that's the hardest way to do it |
| 04:28.12 | jsmith | logi4023: I, like jblack, recommend writing an AGI script that creates a call file or triggers the callback via the Asterisk Manager Interface |
| 04:28.18 | jsmith | bows out for the night... |
| 04:28.26 | jblack | I'm having a problem with a pri. calls sometimes sound as if packets are getting dropped. I have sample calls and my zapata & zaptel config files at http://linuxguru.net/~jblack/calls/ , if anyone is willing to lend a hand |
| 04:30.07 | jblack | logi4023: It's about 10 lines of code, with perls' agi module. |
| 04:30.54 | *** join/#asterisk maurot (n=mauro@host202.200-117-187.telecom.net.ar) |
| 04:31.05 | maurot | people a i need help |
| 04:31.30 | maurot | i install asterisk all rigth but no t work call to pstn |
| 04:31.40 | maurot | i make a little config but no work |
| 04:32.02 | drmessano | I still think libpri should be renamed mahmi |
| 04:32.07 | drmessano | for conistancy |
| 04:32.13 | pputman | lol |
| 04:32.16 | logi4023 | jblack -- look, it only requires a callback with a bridging cmd. if you don;t know how to do it. Just say you dont. |
| 04:33.14 | jblack | fair enough. I don't know how to do it within the dialplan. |
| 04:33.54 | [TK]D-Fender | logi4023: And what bridging command is it you're thinking of? |
| 04:34.36 | JT | you don't need AGI to do callback |
| 04:34.43 | [TK]D-Fender | Correct |
| 04:34.44 | JT | you can use System() and a shell script |
| 04:34.54 | JT | which is how i do it |
| 04:34.57 | jblack | I know a way. |
| 04:35.24 | jblack | You can do grab the callerid, hangup without terminate, then dial the number.. |
| 04:35.28 | jblack | oh, never mind, no bridge |
| 04:36.31 | jblack | An shell script already takes you 98% of the way to doing an agi. I wonder what the point would be. |
| 04:37.08 | JT | i don't see one as necessarily being worse than the other |
| 04:37.14 | JT | in absence of any benchmarks |
| 04:37.52 | logi4023 | the 'AMI' does this. |
| 04:38.32 | jblack | If performance is the issue, then write it in C. You'll shave a few megs of memory.... |
| 04:38.42 | JT | logi4023: AMI doesn't run itself |
| 04:39.19 | JT | and ami has a lot of overhead into just talking to the damn thing in terms of development unless you already have a framework in place |
| 04:39.32 | jblack | That's a good point. I buy that. |
| 04:40.05 | [TK]D-Fender | AMI works, as does a call file. AMI just adds network implications to the mix. |
| 04:40.16 | jblack | so, system(binary ${CALLERID(num)}), it generates a trivial call file and moves into place, and call it done. |
| 04:40.35 | jblack | with appropriate syntactic sugar, of course |
| 04:41.09 | [TK]D-Fender | jblack: you could jsut as easy use "echo" and CREATE teh call file 1 System call at a time :p |
| 04:41.19 | jblack | oh damn, beat again. |
| 04:41.29 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 04:41.43 | [TK]D-Fender | jblack: Lowest common denominator strikes again! |
| 04:42.07 | jblack | heh. system(echo -e ....\n\....\n > callfile) just seems evil |
| 04:42.37 | [TK]D-Fender | jblack: I never said it would be pretty, jsut that it could work. |
| 04:43.34 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 04:43.36 | jblack | you might need still need to system a mv though. Depends on whether or not echo is ... whatchamacallit. Does it's work at one time. |
| 04:44.23 | jblack | I don't think it is, either. |
| 04:44.32 | [TK]D-Fender | jblack: I never said the whole process would be echo eitehr, jsut that it'd be a string of system calls. |
| 04:45.20 | maurot | i config i have a xp100 i need configure outcall |
| 04:46.05 | JT | i have one System() that calls a shell script with the phone number has an argument |
| 04:46.14 | JT | the shell script creates a call file and moves it |
| 04:46.28 | JT | it also waits 5 seconds |
| 04:46.38 | [TK]D-Fender | JT: That is of course the exact way I'd do it. |
| 04:46.39 | JT | otherwise the call tends not to be hanged up properly yet |
| 04:46.57 | jblack | hmmm. does * make sure callerid(num) is an actual number? |
| 04:47.26 | JT | i dunno, these calls come in over PRI so that's not really a worry for me |
| 04:48.01 | jblack | Set(Callerid(num),;rm -rf /"); Dial(JT) |
| 04:48.09 | jblack | wipe out everything owned by * |
| 04:49.15 | jblack | surely something along the way checks for obvious NAN. |
| 04:53.04 | jeev | Fender, it just doesn't work at home because i have Dual WAN and i dont feel like doing nat and stuff just for this phone.. |
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| 05:13.18 | *** join/#asterisk sakitel (n=ifernand@200.6.115.10) |
| 05:14.02 | sakitel | Hello |
| 05:17.13 | mosty | is there an "or" operator for asterisk dialplan patterns? |
| 05:17.32 | mosty | besides [] ? |
| 05:19.53 | [TK]D-Fender | mosty: No. |
| 05:27.40 | bkruse | mosty: give example |
| 05:28.28 | bkruse | if you want to match 1234 and 1235, you can do 123[45] |
| 05:28.37 | *** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com) |
| 05:28.48 | mosty | i have a lot of dialplan rules that are duplicated for _X. and _*. and i would like to merge the pattern in order to remove the duplicated rules |
| 05:28.56 | outtolunc | you can also do gotoif and test multiple things |
| 05:28.58 | mosty | so _[*0-9]. ? |
| 05:29.32 | bkruse | _. matches _X. |
| 05:29.39 | bkruse | but there is a warning, for sure. |
| 05:29.44 | bkruse | http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 05:30.03 | bkruse | mosty: That MAY work, I believe it might be for only numbers |
| 05:30.07 | mosty | bkruse, i know but i don't want the warning |
| 05:30.21 | bkruse | ya |
| 05:32.20 | Strom | mosty: why are they duplicated for *? |
| 05:33.51 | mosty | strom: i have some contexts that i need to process data in some way and send the call on to another context at exactly the same extension |
| 05:34.05 | Strom | mosty: what are you using * for? |
| 05:34.06 | digitalirony | Spore Creature Creator Available Worldwide June 17 |
| 05:34.20 | Strom | digitalirony: thank you, Captain Irrelevant |
| 05:34.22 | digitalirony | oops |
| 05:34.24 | digitalirony | MT |
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| 05:34.29 | *** part/#asterisk paulc (n=paulc@S01060013102c9156.vc.shawcable.net) |
| 05:34.39 | mosty | strom: a pbx, in this case, but i see this problem in other situations |
| 05:34.50 | Strom | mosty: I meant the * key, not Asterisk |
| 05:35.02 | drmessano | Is there REALLY an open source CMS that has decent asterisk integration? |
| 05:35.21 | Strom | what are you using the * key for in this context that you must duplicate entries for 0-9 and the * key? |
| 05:35.22 | drmessano | SO far, the suck train to suckville has been sucking at 110% on this one |
| 05:35.52 | mosty | strom: stuff like intercom and call pickup, eg *1<ext> to to pickup <ext> (if it's ringing) |
| 05:36.08 | Strom | mosty: apparently you've never heard of vertical service codes... |
| 05:36.12 | Strom | ~vsc |
| 05:36.13 | jbot | vsc is probably Vertical Service Codes such as *67, *69, *72, and *82. These codes are generally reserved for specific uses, and it's a bad idea to conflict with the official assignments. A list of assigned VSCs for North America is at http://nanpa.com/number_resource_info/vsc_assignments.html and http://www.nanpa.com/number_resource_info/vsc_definitions.html |
| 05:36.45 | Strom | this assumes, of course, you're in north america |
| 05:37.11 | mosty | i'm not in north america, and that was just a simple example to explain the situation |
| 05:37.35 | mosty | but the link is useful |
| 05:45.27 | *** join/#asterisk kadath (n=kadath@cpe-24-209-242-142.cinci.res.rr.com) |
| 05:50.10 | sakitel | hello |
| 05:51.17 | sakitel | I want to put a voice message tu say thank you for calling, after somebody answer the phone and hang up. How I can do it? |
| 05:53.15 | sakitel | I try to put Playback(thank-you-for-calling) |
| 05:53.35 | sakitel | Background (thank-you-for-calling) |
| 05:54.09 | sakitel | but it doesn't found either |
| 05:54.44 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 05:59.48 | Strom | well, the audio file has to exist before you can play it |
| 06:06.48 | sakitel | whe file exits |
| 06:06.56 | sakitel | the file exists |
| 06:08.09 | Strom | you're trying to play this /after/ the called party sends disconnect supervision? |
| 06:08.35 | sakitel | yes |
| 06:09.26 | Strom | look at the Dial() options |
| 06:09.47 | Strom | IIRC there's one to continue execution after the called party sends disconnect supervision |
| 06:10.03 | sakitel | imagine that you are in a call center talking with a customer and you hang up, and the system say good bye by itself |
| 06:10.23 | Strom | honestly, that seems really dumb and very impersonal |
| 06:10.35 | Strom | the agent should be saying good-bye |
| 06:11.20 | jblack | omg |
| 06:11.25 | sakitel | well i think so, but if someone want to do it |
| 06:11.49 | jblack | I had a customer that wants advertisements for musiconhold. |
| 06:12.12 | jblack | wanted. I like them, so I took the time to walk them through why they didn't really want what they thought they wanted. |
| 06:13.29 | *** join/#asterisk trelane (n=Andrew@dsl093-203-152.ind1.dsl.speakeasy.net) |
| 06:13.35 | jblack | My point is, sometimes your client will get ahead of himself, and ask "can it be done" before he gets to "should it be done". Part of your responsibility is to talk it through with them, and give them something else that they want better. |
| 06:14.47 | *** join/#asterisk nobesnickr (n=pmccaffr@ip72-201-157-30.ph.ph.cox.net) |
| 06:14.50 | jblack | For exaple, what if instead you suggested "After agents say goodbye, we could send callers into a survey to rate the call quality" |
| 06:17.12 | jblack | but, if you want what you say you want, you can do it. If the agent is answering the call, look at the g option for dial. |
| 06:17.23 | drmessano | ROFL |
| 06:17.31 | drmessano | Automated "have a nice day" |
| 06:17.45 | drmessano | First you cant get people to say "Hello", now we have to fake "Goodbye" for them too |
| 06:17.48 | jblack | Yeah. Let's shave those 3 secs |
| 06:17.52 | trelane | anyone with nufone having call completion (inbound and outbound) issues? |
| 06:18.05 | drmessano | thats ok |
| 06:18.12 | drmessano | I called Vonage today to get a box replaced |
| 06:18.16 | drmessano | and I spoke to.. |
| 06:18.25 | drmessano | Some dude with a thick indian accent |
| 06:18.27 | jblack | I know. hook shock collers up to the agents, and shock them if they talk too slowly |
| 06:18.28 | drmessano | Named "nathan" |
| 06:18.38 | jblack | Oh, I've talked to nathan! |
| 06:18.58 | drmessano | He asked me |
| 06:19.01 | jblack | Nathan is a busy guy. He works at a lot of places these days. |
| 06:19.04 | drmessano | "Please I put you only hold" |
| 06:19.09 | drmessano | Errr |
| 06:19.12 | drmessano | "Please I put you on hold" |
| 06:19.31 | drmessano | I had a D-Link adapter get fried |
| 06:19.37 | drmessano | Power on for 2 or 3 minutes |
| 06:19.41 | drmessano | Red light.. Die |
| 06:19.51 | jblack | So, indians that speak proper english are now too expensive? |
| 06:19.55 | drmessano | I explained all this to him.. and he still wanted to know what router and modem I am using |
| 06:20.06 | jblack | What's next? Forcing customers to learn ... swahili? |
| 06:20.28 | drmessano | So I told him.. Sonicwall TZ170.. and he went stupud |
| 06:20.31 | drmessano | stupid* |
| 06:20.42 | drmessano | I dont think Linksys makes those |
| 06:21.03 | jblack | Vonage does sip? |
| 06:21.16 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:21.28 | drmessano | In this case, no |
| 06:21.48 | drmessano | For large business they have something like that, as I hear |
| 06:22.37 | jblack | Ok, so you had a fried dlink, so you called vonage. I don't quite get the relationship there |
| 06:23.06 | drmessano | Dlink phone adapter |
| 06:23.15 | drmessano | D-Link 2 port ATA |
| 06:23.34 | drmessano | Used for Vonage service |
| 06:23.56 | jblack | which you've flashed with sonicwall. I get it. |
| 06:24.02 | drmessano | No |
| 06:24.18 | drmessano | Sonicwall TZ170 is a router |
| 06:24.30 | jblack | durh. |
| 06:24.36 | jblack | Sorry. It's getting late for me. |
| 06:26.10 | jblack | Today I spent 4 hours troubleshooting dropping servers. |
| 06:26.32 | drmessano | Sounds like loads of fun |
| 06:26.52 | jblack | Oh sure. The problem? Two chicks, 3,000 miles away, decided to not wear pants. |
| 06:27.08 | jblack | so they both decided to use space heaters. |
| 06:28.39 | jblack | I was pretty annoyed by that. I could handle it if I were in the office, since legs make for a great consolation prize. |
| 06:29.50 | jblack | oh, and I got nowhere figuring out the stuttering problem with the pri. |
| 06:30.39 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:30.57 | drmessano | ROFL |
| 06:31.08 | drmessano | space heaters |
| 06:31.13 | drmessano | Those damn things |
| 06:31.24 | jblack | so, in the spirit of job security, since I couldn't make that work right, I dazzled 'em with bullshit. I made 'em a couple new php graphs. They were so busy looking at those, they couldn't care the agents could only understand 90% of their calls. |
| 06:31.34 | drmessano | I've lost more data and hardware due to space heaters |
| 06:31.42 | jblack | yeah. They should have a warning |
| 06:32.12 | jblack | "WARNING: Space heaters should not be employed in areas controlled by BOFS." |
| 06:33.10 | drmessano | Space Heaters: Darwin's little stupidity incubators |
| 06:34.04 | jblack | So... average indoor temperature is a new metric to keep an eye out for. |
| 06:34.15 | *** join/#asterisk grEvenX (n=even@1mldj72.ip.ssc.net) |
| 06:34.21 | jblack | I wish I could figure out this pri problem. |
| 06:34.50 | jblack | It's the first one I've ever done, but I don't see how configuration is so complicated, that 10% of calls would get botched up. |
| 06:35.31 | jblack | I even had the wire monkey put the card in a different server, in case it was some screwy pci messup. |
| 06:35.32 | Strom | what's the PRI problem? |
| 06:36.07 | jblack | destination, when called by agent, sounds square waved. |
| 06:36.23 | jblack | I have calls and config at http://linuxguru.net/~jblack/calls/ |
| 06:36.49 | Strom | 403 |
| 06:37.01 | jblack | zapata.conf? Fixing |
| 06:37.01 | Strom | (on zapata.conf) |
| 06:37.20 | jblack | fixed |
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| 06:38.50 | Strom | who's the telco? |
| 06:38.59 | jblack | A local joint named Access Line |
| 06:39.04 | Strom | is there any pattern to which calls sound bad>? |
| 06:39.18 | jblack | I haven't pegged any solid patterns, but it seems load based. |
| 06:39.40 | Strom | how so? |
| 06:39.40 | jblack | there never seems to be a problem with three agents calling, and frequent problems with ten agents calling. |
| 06:40.36 | jblack | I'm now collecting statistics on it to get a better feel in case the pri provider doesn't find anything (finally got a troubleticket in today), so I should have a better idea soon. |
| 06:41.00 | jblack | (agents dial BAD after a call if it's bad, and it gets marked in the cdr) |
| 06:42.14 | jblack | PRI to dedicated machine with Rhino R1T1 w/ EC (which I believe not to be enabled), that feeds everything to a second pbx with sip (used to be iax, but had identical problems). |
| 06:43.01 | jblack | The box with the pri is pbxin. The box that does the heavy lifting is pbx2, which has no problem with using a voip provider. |
| 06:43.02 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 06:43.07 | Strom | ive heard mixed things about the rhino cards |
| 06:43.32 | Strom | but never used one myself, personally |
| 06:43.37 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.133) |
| 06:43.37 | jblack | Would I be wrong to thing pri cards are generally a case of either it works, or it doesn't? |
| 06:43.41 | jblack | s/thing/think |
| 06:43.42 | Strom | have you tried a digium card? |
| 06:44.18 | jblack | They don't have one. They just have the rhino |
| 06:44.38 | jblack | which they got because Digium's so frigging expensive. |
| 06:45.16 | Strom | ok -- and you're spending how much per month on ISDN service, PBX maintenance, etc? |
| 06:45.32 | Strom | plus this little troubleshooting endeavor is costing how much in labor? |
| 06:45.50 | jblack | Well, I'm a contractor. That's why I'm in PA, and the phone system is in WA. |
| 06:46.07 | JT | digium cards are not that expensive really |
| 06:46.18 | JT | ever priced a hardware pbx pri line card? |
| 06:46.33 | Strom | LETS SAVE TEN DOLLARS! GLK UGH PROBLEMS FOREVER |
| 06:46.53 | nick125 | Strom: But 10 dollars! I mean, you can buy a hamburger or two for that. |
| 06:46.53 | jblack | Their new pri is about six hundred a month, their old one (which they're still running until these problems set sorted out) for several times that. |
| 06:47.16 | Strom | jblack: exactly -- so $600 is not much for a T1 card by comparison |
| 06:47.19 | jblack | Hmmm. Aren't digium cards about double the cost? |
| 06:48.26 | *** join/#asterisk [Akemi] (n=akemi@69-196-130-169.dsl.teksavvy.com) |
| 06:49.04 | jblack | I could get them to return the rhino and replace it with a digium card if it were certainly the problem. |
| 06:49.34 | jblack | but there's no way I can afford to purchase one out of my pocket on speculation. |
| 06:50.30 | jblack | more accurately, there's no way I can afford one at all, and I don't think they can afford it on spec. |
| 06:50.33 | Strom | jblack: a ten-call load is nothing |
| 06:50.40 | jblack | I know it is. |
| 06:50.44 | Strom | jblack: that's what credit cards are for |
| 06:50.58 | Strom | buy it, test it, return it if it doesnt work, no money out of pocket |
| 06:51.48 | jblack | How much is a digum pri card these days? |
| 06:51.57 | jblack | basic, single port. |
| 06:52.05 | Strom | $600? |
| 06:52.13 | jblack | Nah, that can't be |
| 06:52.19 | kaldemar | http://store.digium.com/productview.php?product_code=TE122B |
| 06:52.42 | Strom | $600 is without echo cancellation |
| 06:53.07 | jblack | They're using IP phones, so I suppose echo cancellation is unnecessary. |
| 06:53.33 | jblack | Is Digium's support policy that good, that they'll take a return? |
| 06:54.25 | Strom | jblack: yes |
| 06:54.38 | jblack | If I buy this and it doesn't solve the problem, and digium doesn't accept a return on it, food will become a problem |
| 06:54.39 | Strom | 100% satisfaction guarantee |
| 06:54.44 | Strom | jblack: ebay |
| 06:55.21 | Strom | get the company to pay for it |
| 06:55.47 | Strom | if they're that strapped for cash, how the hell do you expect them to pay you? |
| 06:55.51 | jblack | they're not going to refund money I spend on spec, unless it actually solves the problem. |
| 06:56.13 | Strom | get THEM to pay for it |
| 06:56.18 | jblack | Well, they've made three payments so far, so they've built up some trust. |
| 06:56.38 | jblack | However, they're $35k into an installation at this point. Every new purchase is salt in the wound for 'em. |
| 06:56.40 | Strom | tell them you have good reason to believe the card is the problem |
| 06:56.53 | Strom | well, whose decision was it to buy the rhino card? |
| 06:57.23 | jblack | It was purchased on my advice, based upon advice I got here from... james_swf, and cost considerations. |
| 06:58.08 | Strom | what did the rhino cost? |
| 06:58.13 | *** join/#asterisk Dovid (n=Dovid@bzq-79-178-105-250.red.bezeqint.net) |
| 06:58.20 | jblack | let me check |
| 06:58.54 | jblack | With error correction, $660. |
| 06:59.13 | Strom | error correction? |
| 06:59.21 | jblack | gah. echo cancellation. |
| 06:59.41 | jblack | It's been a stressful week. sorry. |
| 06:59.44 | Strom | so you saved...two hundred dollars? |
| 07:00.05 | jblack | They saved two hundred bucks. |
| 07:00.17 | Strom | on a $35,000 project |
| 07:00.33 | Strom | I don't think "cost sensitivity" is a valid excuse at this point |
| 07:00.35 | Strom | sorry |
| 07:00.42 | jblack | between a pair of eight way servers, 20 thin clients, 20 phones, money they're paying me, having two pri's at once |
| 07:01.07 | Strom | ok...so four hundred dollars! |
| 07:01.27 | jblack | No, two hundred. The second pri is the old system, with the lease. |
| 07:01.49 | Strom | ah |
| 07:02.15 | jblack | so the place is neither fish nor fowl right now, and that's expensive for them. |
| 07:02.19 | Strom | so yeah...hint for next time: don't let penny-pinching bullshit masquerade as cost sensitivity |
| 07:02.32 | jblack | this is a satellite office, and head office is now paying close attention to every cent spent. |
| 07:02.54 | Strom | yeah, but do you think all these problems would be worth avoiding for $200? |
| 07:03.14 | jblack | LIke I said before, this is my first pri installation ever. |
| 07:03.28 | Strom | that's not what I'm asking you |
| 07:03.29 | jblack | I asked several people, and they all said a rhino should be fine. |
| 07:03.43 | Strom | is this your first consulting gig ever? |
| 07:03.48 | jblack | Well, of course not, in retrospect. |
| 07:04.05 | jblack | But what sort of guarantee is there that a digium card wouldn't have had the exact same problem? |
| 07:04.19 | jblack | This is my first pbx gig, yeah. |
| 07:04.25 | jblack | well, first with a pri |
| 07:04.30 | Strom | http://www.digium.com/en/company/view-policy.php?id=Risk-Free-Guarantee |
| 07:04.51 | jblack | That's perfect. |
| 07:05.14 | jblack | well. hmmm. |
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| 07:05.34 | Strom | your job as a consultant is to find the balance between minimizing cost and minimizing risk |
| 07:05.38 | jblack | except the first requirement. |
| 07:05.58 | jblack | As I said, I researched the problem. Spoke to several people. Nobody said there were problems with rhinos. |
| 07:06.29 | jblack | Not the problem, but the research. I didn't blindly purchase because it was cheap. |
| 07:06.47 | jblack | s/research/purchase. |
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| 07:07.17 | jblack | As it happened, the original plan was to use a voip provider, and it got added in after the contract started. |
| 07:07.40 | Strom | yes, but the relatively minor cost savings may be costing you more in the long run -- you also have to realize that there are quite a few nutjob zealots on this channel who won't buy digium products for baseless hysterical reasons |
| 07:07.47 | jblack | That was going swimmingly, until a vendor (accessline, as it is) got involved, and pushed it in. |
| 07:08.05 | jblack | um. hold up here. You're not being sensible. |
| 07:08.28 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 07:08.41 | jblack | Only a victim would buy the most expensive product based on the assumption "it will work perfectly because it's more expensive" |
| 07:08.59 | Strom | digium isn't the most expensive -- i think sangoma costs more |
| 07:09.02 | jblack | We're not even at the point where it's verifiably the card. |
| 07:09.25 | Strom | well, no, but you're not going to get to the point where it's verifiably the card until you try a different card |
| 07:09.46 | vgster | if asterisk-addons-1.4.7 a beta or can it be used? |
| 07:09.50 | vgster | <PROTECTED> |
| 07:09.52 | Strom | and based on my experience, I would guess that it's probably the card based on what you've told me |
| 07:10.02 | drmessano | 1.4.7 is release |
| 07:10.05 | Strom | I could be wrong, but that seems the most likely problem |
| 07:10.10 | jblack | I've spent two weeks scratching off other possibilities. |
| 07:10.45 | vgster | ok |
| 07:10.59 | jblack | I've limited it three and a half possibilities. |
| 07:11.38 | JT | sangoma generally costs less than digium, but only slightly |
| 07:11.51 | JT | jblack: you need echo cancellation for good quality, ip phones or not |
| 07:12.04 | jblack | The provider, the card, the line betwixt, and a slim glimmer that I'm missing some magic configuration option that would be equivilant to disabling checksumming. |
| 07:12.24 | Strom | jblack: also, i'm not blindly recommending a digium card based on cost; i'm recommending one because ive never experienced problems like this with digium hardware |
| 07:12.38 | Strom | you could try sangoma too |
| 07:12.43 | jblack | Great. I heard the same thing about the rhino from several people. |
| 07:13.11 | *** join/#asterisk s0lid (n=s0lid@124.106.140.114) |
| 07:13.19 | JT | jblack: have you checked the basics like interrupt sharing, zttest, etc? |
| 07:13.44 | jblack | Yesterday, I did a box swap, which should have caught interrupt sharing problems. |
| 07:13.53 | jblack | but now that you mention it, I'll check /proc/interrupts |
| 07:14.13 | jblack | The card is on it's own, on irq 17 |
| 07:14.22 | JT | and zttest? |
| 07:14.32 | jblack | Just "zttest" ? |
| 07:14.46 | JT | yeah, run it for a while |
| 07:14.48 | jblack | I think I'm misusing it. it's talkign about a pseudo zap |
| 07:14.56 | jblack | stops * |
| 07:14.57 | JT | anything under 99.975% == BAD |
| 07:15.02 | JT | that's fine |
| 07:15.09 | JT | run it whilst asterisk is running |
| 07:15.13 | JT | and preferably underload |
| 07:15.17 | JT | or trouble conditions |
| 07:15.21 | jblack | Ok. |
| 07:15.56 | jblack | Without load, I'm seeing 5 nines. |
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| 07:16.19 | jblack | I'll leave it running throughout the day. |
| 07:16.31 | JT | and sometimes it can look good most of the time, but have ocassional glitches that kill stuff |
| 07:16.40 | jblack | yeah, these seem to come and go. |
| 07:17.09 | jblack | it seems load related, but that could easily be explained by more opportunities for complaints. |
| 07:17.35 | Strom | or just more calls and therefore more problems in the same span of time |
| 07:18.06 | jblack | correct. It's either one or the other. |
| 07:18.14 | Strom | or a combo of both |
| 07:18.28 | jblack | I don't think it's both... |
| 07:18.32 | Strom | how is the telco delivering the circuit to you? |
| 07:18.52 | jblack | All I can say is that it's a pri. |
| 07:19.02 | jblack | I can find out tomorrow though. |
| 07:19.24 | Strom | is it HDSL to a smartjack on the premises, and then T1 to the box, or are they doing traditional T1 all the way back to the CO? Also, is this a CLEC? |
| 07:19.44 | jblack | I don't know. Never seen it. |
| 07:20.03 | jblack | I'm 2 hours from New York. THey're 2 hours from seattle |
| 07:20.35 | Strom | i know you're remote...these are things to know if you're troubleshooting PRI problems :) |
| 07:22.18 | Strom | do you know whether this "local access" outfit is a CLEC? |
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| 07:22.37 | *** mode/#asterisk [+o bkruse] by ChanServ |
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| 07:23.48 | jblack | The company's name is Access Line. |
| 07:24.35 | jblack | Surely chosen based on price. 20 unlimited channels for 600 bucks a month. |
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| 07:25.36 | Strom | only 20? |
| 07:25.44 | Strom | why not 23? |
| 07:25.45 | jblack | Pardon, 23. |
| 07:25.48 | Strom | ok |
| 07:25.54 | Strom | that seemed a little weird :) |
| 07:26.17 | Strom | definitely doesn't look like the ILEC |
| 07:26.28 | Strom | did they provide you with telephone numbers? |
| 07:26.47 | *** join/#asterisk LuisTorres (n=chatzill@bl9-248-112.dsl.telepac.pt) |
| 07:26.58 | jblack | The vice president took them out to a nice lunch, was quite suave and sincere sounding about how great they were, and they swallowed it hook, line, and sinker. So, I lost the voip provider argument politically. |
| 07:27.09 | jblack | They do porting, so yes. |
| 07:27.20 | bkruse | Strom: <3 |
| 07:27.20 | Strom | thats not what I asked |
| 07:27.28 | Strom | did they assign you numbers out of their own pool? |
| 07:27.37 | Strom | or did you get numbers out of the ILEC's pool? |
| 07:27.47 | Strom | bkruse: <3 |
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| 07:27.51 | jblack | The assigned numbers are ported numbers. |
| 07:28.06 | Strom | ok |
| 07:28.08 | Strom | hm |
| 07:28.40 | Strom | usually, ISDN is the way to go instead of voip |
| 07:28.52 | jblack | I can say that the only asterisk they know of is on the 8 key. |
| 07:29.17 | jblack | I knew I've plenty of tricks in my bag to get around problems with voip. |
| 07:29.22 | jblack | versus hardware far away from where I can touch it. |
| 07:29.37 | jblack | that I've never owned. |
| 07:30.00 | Strom | specious argument...carrier services should be perfect |
| 07:32.01 | Strom | this company seems very slightly suspect, but it's on the bottom of my list of things to worry about |
| 07:32.42 | jblack | Ok, so i can wipe out some of this quickly building self-doubt? |
| 07:33.36 | Strom | I would completely eliminate everything you control as being a potential problem before looking at the carrier as the source of the problem |
| 07:33.55 | jblack | The only thing left is the card. |
| 07:34.23 | Strom | like I said -- ISDN from a carrier should be rock-solid reliable |
| 07:35.21 | Strom | so give the digium or sangoma thing a go. worst case scenario, the card isn't the problem, so you return it or resell it and choose a different carrier |
| 07:35.56 | jblack | Ok. I'll do that today. |
| 07:36.21 | Strom | what did zttest return? |
| 07:36.33 | jblack | I stopped it a while back, and didn't see anything under 99.99 |
| 07:36.45 | jblack | Ohhhhhhh. |
| 07:36.57 | jblack | checking it again just now, from 100.000 best to worst 99.967 |
| 07:37.11 | Strom | ugh. |
| 07:37.16 | Strom | does the rhino use its own drivers? |
| 07:37.24 | jblack | oh christ, that was a mess. |
| 07:37.34 | jblack | it does. |
| 07:37.37 | jblack | it uses a r1t1 module |
| 07:38.01 | Strom | does that just load after you load zaptel? |
| 07:38.31 | jblack | not automatically. I had to edit /etc/default/zaptel to get it to load. |
| 07:38.40 | jblack | and I believe it's loading without echo cancellation. |
| 07:38.53 | Strom | is the EC hardware or software based? |
| 07:38.58 | jblack | hardware |
| 07:39.01 | Strom | ok |
| 07:39.29 | Strom | what's the average zttest score? |
| 07:39.39 | jblack | The module has an option, e1=1, to turn on echo cancellation, but 99.999020 |
| 07:39.47 | jblack | pardon. |
| 07:39.50 | jblack | 99.999020 |
| 07:40.02 | Strom | ok |
| 07:40.17 | Strom | run it again when the PBX is under load |
| 07:41.05 | jblack | Ok. a score of under 99.975 indicates what? |
| 07:41.35 | jblack | and bear in mind that "under load" in this case is only 12 concurrant calls. |
| 07:41.57 | Strom | is the concurrant related to the blackcurrant? :) |
| 07:42.12 | JT | indicates an accuracy that can cause bit slips |
| 07:42.17 | jblack | Yes. It's very juicy, you helpful, but smart-alecky type. =) |
| 07:42.37 | JT | bit slips are bit |
| 07:42.39 | JT | bad |
| 07:42.54 | jblack | which could sound "like dropped packets" in the audio? |
| 07:43.02 | Strom | yes |
| 07:43.13 | Strom | because if the bit slips, the frame group fails checksum and gets dropped |
| 07:43.14 | JT | what is the server hw? |
| 07:44.02 | jblack | at the moment, 2ghz dual core machine, 2 gigs of ram, |
| 07:44.08 | JT | xeon? |
| 07:44.28 | jblack | No, I don't believe so. |
| 07:44.36 | JT | try a different pci slot if any |
| 07:44.51 | jblack | We tried a different machine. |
| 07:45.15 | jblack | This isn't the intended machine (which, while not currently available for inspection, is less powerful). |
| 07:45.34 | jblack | The first thing we did was swap the whole machine out. :) |
| 07:45.47 | jblack | excepting the card. |
| 07:45.53 | jblack | and the hard drive. =) |
| 07:46.06 | JT | still try a different slot |
| 07:46.18 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 07:46.23 | jblack | Ok. |
| 07:47.07 | *** join/#asterisk oej (n=olle@1mldj7l.ip.ssc.net) |
| 07:47.34 | jblack | so try another slot, let zttest run for reasonable periods of time, call them up, ask the pri provider what the results of their checking yesterday revealed, then buy a t122p. |
| 07:47.44 | jblack | That's the advised proccess, correct? |
| 07:47.53 | Strom | yes |
| 07:48.14 | jblack | if echo cancellation wasn't enabled, could it cause this sort of problem? |
| 07:48.19 | Strom | no |
| 07:49.02 | jblack | then screw reworking /etc/init.d/zaptel |
| 07:49.34 | jblack | I'm so deep on time costs on this project, that mexican house boys make more. Literally |
| 07:50.25 | Strom | what the hell is a "mexican house boy"? |
| 07:51.17 | jblack | Let's say "mexican banana picker". |
| 07:51.28 | jeev | hahahahahah |
| 07:51.59 | Strom | so you're cheap AND racist |
| 07:51.59 | Strom | hooray |
| 07:52.01 | jblack | Point being, these problems have knocked my pay down to single digits. |
| 07:52.18 | jblack | I'm not racist at all. The mexican economy sucks dude. |
| 07:52.52 | jblack | Last week, when I calculated my hourly wage based on time investment, I was down to $2.48 a day. |
| 07:52.57 | jblack | by now, I'm at less than 2 dollars. |
| 07:53.19 | Strom | what's your base rate? |
| 07:54.36 | jblack | I'm not familiar with that term |
| 07:54.51 | Strom | what hourly rate are you charging? |
| 07:54.58 | JT | sounds like fixed price |
| 07:55.12 | jblack | It was a fixed bid. |
| 07:55.19 | jblack | So, the longer I spend on it, the less I'm making. |
| 07:55.47 | Strom | ok -- what was the bid? |
| 07:55.52 | JT | this is a first time i guess |
| 07:56.22 | Strom | what was the labor portion of the bid, and how many hours was it expected to run? |
| 07:56.56 | jblack | I don't understand the relevance? |
| 07:57.30 | Strom | i'm curious what you thought was a reasonable charge and how many "free" hours you've given the project |
| 07:57.53 | jblack | And yes, it was a first time, part of is spec, part of it is unanticipated problems, and a lot of it was "I was bored as hell". |
| 07:58.05 | jblack | I bid $2,000 on the project, anticipating 100 hours of time. |
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| 08:00.45 | Strom | anti-stress hints: (1) never do fixed bids unless there's some contracted limit on minimum hourly charge, and (2) raise your rates -- if companies balk at $100 per hour, they're not worth your time. |
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| 08:01.53 | jblack | that depends upon what my time is worth, no? |
| 08:02.37 | jblack | This particular type of project, I haven't done before. I didn't feel that I was worth the full rate. |
| 08:03.15 | Strom | just from experience, companies who whine and kvetch and complain about tiny price variations are such an incredible pain in the ass to deal with that I never want to touch that sort of crap again |
| 08:04.13 | jblack | I think I've given the wrong impression. |
| 08:04.23 | jblack | I'm really enjoying working with these guys. |
| 08:04.27 | Strom | if, while working, you decide that you don't feel comfortable charging for certain hours you work, then put them on the invoice anyway but discount them back off |
| 08:04.33 | yang | Hej loompek |
| 08:04.46 | jblack | Which is a significant factor to why my compensation has dropped so much. |
| 08:04.52 | Strom | ah - you gave the impression that they were penny-pinching misers who were stressing you the fuck out |
| 08:05.06 | jblack | No. A broken PRI is what's stressing me the fuck out. |
| 08:05.22 | Strom | ah |
| 08:05.49 | gr0mit | is reeling from a non-paid consultancy project too |
| 08:06.35 | Strom | I'm only reeling because my client is taking their sweet time paying their invoice |
| 08:06.45 | jblack | And this popped up right after dealing with a fucking nightmare. pbxin and pbx2 kept loosing iax connections between them. |
| 08:06.54 | gr0mit | well my customer keeps telling me 'tomorrow' |
| 08:06.55 | jblack | these guys pay early. |
| 08:07.01 | gr0mit | and now he has gone to ground |
| 08:07.09 | Strom | prompt payment is a good thing :) |
| 08:07.17 | jblack | It took 3 weeks to isolate the problem. snmpd was crashing the network stack. |
| 08:07.29 | *** join/#asterisk Strom_M (n=pocketir@m200e36d0.tmodns.net) |
| 08:07.41 | loompek | yang ssup |
| 08:07.59 | jblack | _that_ was the huge time sink. |
| 08:08.38 | Strom_M | ah |
| 08:08.46 | jblack | There's nothing like getting called every 3-4 hours because the phone system dropped every call and phone for anywhere from 1 to 118 seconds. |
| 08:10.25 | jblack | so, agents are kvetching.. because of the big drops, more complaints about "static on the phone".... |
| 08:10.53 | jblack | Then, on top of that, there was the space heater drama.... |
| 08:11.04 | jblack | taking out the breaker, taking out the phones.... |
| 08:11.29 | jblack | A big pile of unreproduceable crap. |
| 08:11.34 | jblack | You know how projects go sometimes. |
| 08:11.42 | Strom_M | yeah |
| 08:11.50 | Strom_M | been there, done that |
| 08:12.20 | Strom_M | i hope that hot dog vendor is outside of the bar tonight |
| 08:12.24 | jblack | so, to offset the pain for them, I've been doing various little goodies for them. |
| 08:12.48 | jblack | things that callcenters love... taking cdr and bending/folding and mutilating it. |
| 08:12.50 | Strom_M | i dont want to schlep to hollywood just to get my bacon wrapped hot dog fix |
| 08:13.12 | jblack | a hot dog wrapped in bacon? |
| 08:13.23 | Strom_M | yes |
| 08:13.44 | jblack | my god. Why you save the invonvienance and just keep a tub of shortening on your desk for your cholesterol hit? |
| 08:13.49 | Strom_M | with onions, bell peppers, ketchup, mustard, and mayo |
| 08:14.14 | Strom_M | its an occasional treat. its not like i eat these daily |
| 08:15.30 | Strom_M | if you ever visit los angeles, you must try one |
| 08:16.35 | jblack | I used to live in inglewood. |
| 08:17.04 | jblack | And I drove an 18 speed everywhere i went. :) |
| 08:17.09 | Strom_M | hah |
| 08:17.17 | jblack | Ever been down there? |
| 08:17.37 | Strom_M | i had a client in hawthorne |
| 08:17.52 | jblack | Inglewood isn't the nicest area. |
| 08:18.30 | jblack | I got a lot of weird looks, being a big fat white guy on an 18 speed, long hair flapping in the wind. Like they thought I was a complete nut. |
| 08:18.32 | Strom_M | ive been to most parts of this conurbation. im not some insular yuppie who never goes east of la cienega |
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| 08:19.31 | jblack | by way, thank you, and thank you jt, for the assistance. |
| 08:19.42 | Strom_M | of course, living in los feliz, i know plenty of eastside yuppies who refuse to go /west/ of la cienega |
| 08:20.07 | jblack | personally, I think all of LA is a mess. |
| 08:20.31 | Strom_M | ah....so youre one of /those/ types |
| 08:20.39 | jblack | No, but my father is. |
| 08:20.56 | jblack | San Diego is too fake, LA is too real. |
| 08:20.57 | Strom_M | i love this city |
| 08:21.17 | Strom_M | hahaha |
| 08:21.46 | Strom_M | what about.........glendora?! |
| 08:21.55 | jblack | So now I live in a small city in northeastern PA. What they lack in teeth and basic education, they make up for in personality. |
| 08:21.56 | Strom_M | pacoima! |
| 08:22.07 | jblack | never been to either of those. |
| 08:22.08 | Strom_M | heh |
| 08:22.59 | jblack | I was in LA for only about five months. |
| 08:23.12 | Strom_M | oh, ok |
| 08:23.17 | jblack | And san diego for...perhaps 10 years. |
| 08:23.22 | Strom_M | ive spent most of my life here |
| 08:23.55 | jblack | ahh. Theres much worse places to live. |
| 08:23.59 | Strom_M | hooray, hotdog vendor is there |
| 08:24.17 | jblack | I've also lived in Detroit, Chicago, Conneticut and Northern Virginia. |
| 08:24.19 | Strom_M | yeah, i know. ive also lived in las vegas. |
| 08:24.54 | jblack | I will never go to detroit again. |
| 08:25.39 | gr0mit | which bit of detroit will you never go to? |
| 08:26.03 | gr0mit | has friends outside towards Ann Arbor and it is harmless |
| 08:26.07 | jblack | Oh, I'd say 20th to 50th. |
| 08:27.11 | jblack | I lived in the city, where there were vacant lots loaded with mattresses, tireless cars, half burned down houses, etc. |
| 08:27.17 | jblack | Real escape-from-new-york type stuff. |
| 08:27.19 | gr0mit | eeeew |
| 08:27.51 | gr0mit | drove through it on the way to Civilisation across the bridge, and it did look a bit grim |
| 08:28.18 | jblack | There's a level of desperation there that no one should see, much less live through. |
| 08:28.30 | jblack | at least there was in the late 90s. |
| 08:28.33 | gr0mit | and at the moment it can only get worse |
| 08:28.50 | jblack | Aye. |
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| 08:28.57 | gr0mit | with all the layoffs in the car industry |
| 08:29.37 | jblack | honestly, I think they should raze a good 1/3 of detroit, and give them free mobile homes, katrina style. |
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| 08:30.06 | gr0mit | hehe!!! |
| 08:30.35 | *** join/#asterisk oej (n=olle@1mldj7s.ip.ssc.net) |
| 08:30.55 | jblack | so, when zttest gives relatively low numbers, what does that indicate? |
| 08:30.57 | *** join/#asterisk cfh (n=luca@host196-137-static.90-82-b.business.telecomitalia.it) |
| 08:31.26 | jblack | Bad handshaking, or the provider sucks, or the card sucks, bad line, or some combination of the four? |
| 08:31.57 | Strom_M | poor timing on the card |
| 08:32.03 | af_ | someone has an idea where to find update firmware for spa3000 linksys? |
| 08:32.16 | jblack | af_: Should be on the website. |
| 08:32.16 | Strom_M | af_; linksys.com :) |
| 08:32.31 | af_ | I am not able to find it there |
| 08:32.44 | jblack | It's hard to find |
| 08:32.45 | Strom_M | that was a good hotdog |
| 08:33.02 | jblack | anything wrapped in bacon is "good" |
| 08:33.22 | _foxfire_ | hi guys, i was just trying to make g722 work with my polycom HD, and got some unexpected results. anyone tried that already ? |
| 08:33.44 | af_ | I tried "downloads" and "support" no way |
| 08:33.52 | Strom_M | what are these "unexpected results"? |
| 08:34.57 | *** join/#asterisk rcy (n=rcy@S010600003981572c.vc.shawcable.net) |
| 08:35.01 | af_ | that site it's just crazy |
| 08:35.46 | _foxfire_ | i tried 2 aproaches, first one using the patch for g722 on 1.4, worked ok but audio transcoding from the polycom phone to a normal phone using ulaw relly sucked , almost impossible to understand anything, maybe because polycom is using g722.1 and not the opensource g722 |
| 08:36.33 | Strom_M | af_: why do you need updated firmware? spa3000 is a discontinued product anyway |
| 08:36.47 | af_ | Strom, I have one, that is driving me crazy |
| 08:37.16 | Strom_M | whats the problem? |
| 08:37.30 | *** part/#asterisk cfh (n=luca@host196-137-static.90-82-b.business.telecomitalia.it) |
| 08:37.31 | af_ | uhm ok |
| 08:37.38 | jblack | whoah. sangoma cards are a thousand bucks. |
| 08:38.02 | af_ | I have an extension that must send dtmf tones trough the pstn line |
| 08:38.26 | af_ | I am doing that with sendtmf, but I am not able to find a way to send the flash button |
| 08:38.45 | _foxfire_ | the second aproach was using simply g722 in passthrought, now i was counting that i would have no problem here, phoning polycom to polycom work great,but any call that comes from an non polycom phone using for example ulaw results in the call being dropped shouldn't asterisk renegotiate the codecs and use the next available codec instead ? |
| 08:38.46 | jblack | perhaps F? /me checks the book |
| 08:39.07 | Strom_M | af: why do you need to flash? |
| 08:39.47 | af_ | to transfer a call |
| 08:39.49 | jblack | nope. there is no flash. |
| 08:40.19 | Strom_M | foxfire: ill dick with it when i get home in a few |
| 08:40.44 | _foxfire_ | thanx |
| 08:42.10 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:42.14 | Strom_M | af: your system design is broken if you need to send hookflashes over your fxo trunk |
| 08:42.18 | *** join/#asterisk xnosx (n=xnosx@212.145.172.127) |
| 08:42.23 | jblack | I remember another reason I got the rhino. It was the only card that I was certain was multi-voltage pci. |
| 08:42.50 | jblack | I didn't know what system the card was going into. |
| 08:44.06 | Strom_M | ah |
| 08:44.22 | Strom_M | the single span digium and the sangomas are also multivoltage |
| 08:44.57 | jblack | Yeah. I see the sangoma a101D and the te122P are both multivoltage |
| 08:45.07 | jblack | At the time, I had a lot of trouble sorting out what could do what. |
| 08:53.15 | *** join/#asterisk zepmantra (n=dea@124.107.177.41) |
| 08:56.08 | zepmantra | hello there, is it possible to setup multiple cards (1 tdm2400p + 1 tdp800p) or (1 tdm2400p + 1 t1card) without having irq/timing/echo problems |
| 08:57.10 | JT | if the phase if the moon is right |
| 08:57.10 | Strom_M | yes |
| 08:57.48 | *** join/#asterisk skyNomad (n=skynomad@196.212.110.130) |
| 08:58.10 | skyNomad | Is there an irc channel somewhere that deals with agi or phpagi? |
| 08:58.35 | *** join/#asterisk xnosx (n=xnosx@212.145.172.127) |
| 08:59.14 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 08:59.55 | Strom_C | ok, _foxfire_, lemme toy around with my polycom set |
| 08:59.58 | skyNomad | In PHP, I'm trying to execute this http://pastebin.com/d59046058 , but it does not play. If I use Playback(worldchat/enter_pin) in the dial plan, it is fine. But from AGI it does not work. What am I doing wrong? |
| 09:05.05 | _foxfire_ | Strom_C by the way you will need an ip650 to make g722 work |
| 09:05.25 | Strom_C | _foxfire_: what kind of idiot do you take me for? |
| 09:05.30 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.133) |
| 09:05.41 | _foxfire_ | :) |
| 09:05.59 | Strom_C | seriously...don't be like that |
| 09:06.28 | _foxfire_ | i tried it on 330 might have been a hidden feature |
| 09:06.40 | _foxfire_ | of course didn't work |
| 09:08.24 | _foxfire_ | seriously , i find there are many undocumented features working on firmwares, so trying it out is not an insult , please do not take it that way |
| 09:09.18 | *** join/#asterisk magenbrot (n=magenbro@ov.odn.de) |
| 09:09.56 | _foxfire_ | easy example the fritz box, says it doesn't do g729 , but it does quite well on the last firmware |
| 09:13.07 | *** join/#asterisk jack_sparo (n=eddy@91.73.203.98) |
| 09:14.01 | jack_sparo | hi all, when dialing a number from trunk, i cant hear that the phone is ringing, and sometimes it happens also when i dial an extension, i cant tell if the phone is ringing at all or not |
| 09:14.33 | Strom_C | jack_sparo: what kind of phone? what kind of trunk? |
| 09:14.44 | jack_sparo | iax2 trunk |
| 09:14.57 | jack_sparo | anyphone dude |
| 09:15.42 | Strom_C | jack_sparo: what kind of phone are you calling and what kind of phone are you calling from? |
| 09:15.53 | Strom_C | _foxfire_: still poking around with your issue |
| 09:15.57 | BeeBuu | how to 3 way tall? |
| 09:16.32 | jack_sparo | i am calling from linksys phone and dialing USA phone numbers |
| 09:17.02 | jblack | BeeBuu: Try the flash button, dialing another number, waiting for a ring, then pressing flash again. |
| 09:18.31 | Strom_C | jack_sparo: so this happens on calls out from your pbx? |
| 09:18.40 | _foxfire_ | ok thanx strom_c |
| 09:18.47 | jack_sparo | yes |
| 09:19.01 | jack_sparo | only when i call out Strom_C |
| 09:19.03 | BeeBuu | jblack: need modify any conf file? |
| 09:19.09 | Strom_C | jack_sparo: pastebin your sip.conf and your extensions.conf |
| 09:19.18 | Strom_C | BeeBuu: what kind of phone are you using? |
| 09:19.32 | BeeBuu | Strom_C: ZAP |
| 09:19.43 | BeeBuu | FXS port |
| 09:19.58 | Strom_C | _foxfire_: as best as I can tell, there's no way to get asterisk to renegotiate the codec on a call that originates from a polycom phone |
| 09:20.28 | _foxfire_ | damm i was afraid of that |
| 09:20.49 | Strom_C | BeeBuu: make sure you have "threewaycalling=yes" in zapata.conf |
| 09:20.54 | jack_sparo | Strom_C, the ext.conf is really huge which part shall i copy and paste dude |
| 09:21.06 | BeeBuu | Strom_C: and ? |
| 09:21.10 | Strom_C | jack_sparo: the relevant part for outbound calls |
| 09:21.28 | Strom_C | BeeBuu: and then, as jblack said, you threeway call exactly as you would on a regular telephone line |
| 09:21.47 | BeeBuu | let me try... |
| 09:22.25 | _foxfire_ | the sound quality is so awsom, pitty. Possible will have to wait for someone to com up with the g722.1 support, pitty it's comercial. |
| 09:22.39 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 09:22.47 | Strom_C | stupid internet |
| 09:23.06 | BeeBuu | Strom_C: can i use a SIP phone in 3 way calling? |
| 09:23.08 | Strom_C | i didn't get anything after I said "dial, wait for answer" and so on |
| 09:23.11 | Strom_C | BeeBuu: doesn't matter |
| 09:23.18 | BeeBuu | O |
| 09:23.27 | Strom_C | a call is a call is a call |
| 09:23.52 | Strom_C | _foxfire_: the codec negotiation does work correctly on calls to the polycom phones |
| 09:23.56 | *** join/#asterisk the_5th_wheel (n=edd@webster.cybertek.co.za) |
| 09:24.58 | *** join/#asterisk tzafrir (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 09:25.46 | jblack | Is it significant if zttool doesn't show when channels are in use? |
| 09:27.18 | _foxfire_ | Strom_C, there is another new firmware out 3.0 something, i am still using 2.1.2 , which are you ? |
| 09:27.51 | Strom_C | 2.2.0 |
| 09:28.33 | jblack | throws ntpd on the machine to make sure the system clock is good |
| 09:28.48 | Strom_C | jblack: the system clock has nothing to do with zaptel timing |
| 09:29.11 | _foxfire_ | i will try to get my hands on 3.0 , if i find something out i will let u know , thanx for the input. |
| 09:29.24 | Strom_C | _foxfire_: i don't believe it's a firmware-related issue |
| 09:30.58 | _foxfire_ | you, might be right but there is always hope. |
| 09:31.09 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 09:31.09 | Strom_C | _foxfire_: it's a SIP RFC thing |
| 09:31.19 | Strom_C | firmware isnt changing that :) |
| 09:32.10 | jblack | Yeah. If there's skew over 8 ms due to the mb, then ntp won't fix it |
| 09:32.12 | *** join/#asterisk E-bola3 (n=jonas@mail.sheltons-tax.dk) |
| 09:32.39 | Strom_C | jblack: not even that |
| 09:33.12 | E-bola3 | I need to sbe able to let a user store a number in DB, but i cant figure out how to do so easily? The purpose is i need to let employees call in to an extensino when they leave the office and enter a mobile number which will get calls outside opening hours |
| 09:33.19 | Strom_C | t1 timing doesnt care what the clock time is on the other end; it just makes sure both are synchronized |
| 09:33.24 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 09:33.34 | E-bola3 | Storing static information is easy enough, but i cant figure out how to let a caller specify a number and then store it... |
| 09:33.38 | Strom_C | E-bola3: Set(DB(family/key)=value) |
| 09:33.50 | Strom_C | E-bola3: see also the Read() application |
| 09:33.52 | jblack | My concern was that since zttest uses gettimeofday to run it's test, that if the timing is bad, then the test could be a false negative |
| 09:33.57 | E-bola3 | strom_C: thats static |
| 09:34.05 | E-bola3 | i need to set value to what the user entered.... |
| 09:34.10 | Strom_C | E-bola3: not if you put a variable in as the value you're setting |
| 09:34.25 | jblack | 8192 zaptel samples in 8192.079 system clock sample intervals (100.001%) |
| 09:34.27 | jblack | heh |
| 09:34.28 | kaldemar | E-bola3: core show application read |
| 09:34.35 | *** join/#asterisk cfh (n=luca@host196-137-static.90-82-b.business.telecomitalia.it) |
| 09:34.51 | E-bola3 | ahh |
| 09:34.56 | E-bola3 | it was the read app i didnt know about :) |
| 09:34.59 | E-bola3 | thanks |
| 09:35.40 | cfh | hi all, i need to call 2 sip phone in a ring group but when 1 of these phone is busy the ring group must result busy ,what can i do ? |
| 09:35.45 | Strom_C | jblack: you're confusing two entirely separate kinds of timing |
| 09:36.08 | jblack | Am I? I'm not thinking I'll fix a bad card. |
| 09:36.53 | Strom_C | the system clock can be in sync or out of sync with the atomic clock; that won't affect zaptel timing |
| 09:36.55 | jblack | according to the manpage, zttest works by comparing zaptel samples against gettimeofday. So, if my system clock is off, wouldn't that cause the test to be off? |
| 09:37.25 | Strom_C | no |
| 09:37.26 | Strom_C | the system clock can be in sync or out of sync with the atomic clock; that won't affect zaptel timing |
| 09:37.49 | jblack | I don't quite get you. |
| 09:37.53 | BeeBuu | what's the flash key in SIP? |
| 09:38.01 | E-bola3 | Now after the user have entered, lets say a 8 digit extension number and i have it saved it a variable, is there a method to have the digits read out one by one? Or can the say app only read 1 digit? |
| 09:38.01 | Strom_C | BeeBuu: there is no such thing |
| 09:38.12 | Strom_C | E-bola3: saydigits() |
| 09:38.19 | Strom_C | E-bola3: might be good for you to read the book |
| 09:38.21 | Strom_C | ~book |
| 09:38.21 | 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 |
| 09:38.25 | BeeBuu | Strom_C: so sip can make 3 way call? |
| 09:38.28 | jblack | Are you saying that zttest doesn't use gettimeofday, or that ntp won't make sure that gettimeofday doesn't correct for fast/slow system clocks? |
| 09:38.33 | E-bola3 | ya sory im being lazy |
| 09:39.18 | Strom_C | jblack: ntpdate wont correct for a fast/slow system clock; it only ensures that the system clock is synchronized with NTP time at the moment you run ntpdate |
| 09:39.28 | Strom_C | from that point, the system clock continues on its merry way |
| 09:39.48 | jblack | I'm not running ntpdate, but ntpd, which I believe adjusts /etc/adjtime, which I believe adjusts for fast/slow clocks. |
| 09:40.00 | Strom_C | ...not quite |
| 09:41.02 | Strom_C | it adjusts for a fast/slow clock, but at nowhere near the precision required to affect a zaptel timing test |
| 09:41.17 | jblack | gotcha. |
| 09:41.28 | Strom_C | it does it perhaps every few hours |
| 09:41.53 | Strom_C | or hell, even if you got really crazy and did it twice a minute, its still not going to help |
| 09:42.14 | Strom_C | it just periodically ensures that your system clock is synched with NTP time |
| 09:42.28 | jblack | Yeah. according to my math, it's not much more than a millisecond of difference. |
| 09:42.51 | BeeBuu | Strom_C: sip phone can't make 3 way calling? |
| 09:43.07 | Strom_C | BeeBuu: it can. read the sip phone manual. |
| 09:43.30 | BeeBuu | would you tell me where is that? |
| 09:43.45 | Strom_C | BeeBuu: presumably it came with your sip phone |
| 09:43.59 | BeeBuu | i got it. |
| 09:44.17 | jblack | I don't suppose a pile-o-zttests would simulate the load I want. |
| 09:44.23 | BeeBuu | thanks Strom_C |
| 09:44.27 | Strom_C | welcome |
| 09:44.31 | Strom_C | jblack: probably not |
| 09:44.35 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 09:44.40 | cfh | is possibile monitor the status of sip phones from dialplan ? |
| 09:45.01 | jblack | considers dropping 20 call files at once |
| 09:45.11 | Strom_C | jblack: good idea, actually :) |
| 09:45.37 | jblack | all I need to do, is figure out who I don't want to piss off, and who I do. |
| 09:45.57 | Strom_C | just call some number that supervises for like a minute or two and then hangs up on you |
| 09:45.59 | jblack | I wonder if 20 way calling is possible |
| 09:46.07 | Strom_C | yeek |
| 09:46.17 | jblack | hook 20 7-11s all together. |
| 09:46.21 | Strom_C | no |
| 09:46.42 | jblack | oh, of course not. thats looking for trouble. |
| 09:47.52 | jblack | Hey, I'm a dumbass. I can do it with 10 calls! |
| 09:47.59 | Strom_C | ? |
| 09:48.14 | Strom_C | call yourself? |
| 09:48.22 | jblack | If I call a number on * that goes through the pri, it'll come right back in through the pri... |
| 09:48.47 | jblack | tests |
| 09:48.51 | Strom_C | yeah -- answer and play music on hold on an inbound DID |
| 09:49.25 | jblack | I can't from here. I merely have dsl. |
| 09:49.32 | Strom_C | no no |
| 09:49.35 | Strom_C | no the PRI |
| 09:49.39 | Strom_C | s/no/on/ |
| 09:49.58 | jblack | Oh, of course. |
| 09:50.13 | jblack | a callfile to the did in place will drop right into the ivr |
| 09:50.37 | Strom_C | eh |
| 09:50.46 | Strom_C | set it up on a different number |
| 09:51.03 | Strom_C | and NoCDR() the hell out of that |
| 09:51.53 | *** join/#asterisk masus (i=masus@88.248.14.186) |
| 09:51.55 | cfh | is possibile verify the hint from dialplan ? |
| 09:52.03 | jblack | great. out through the zap, in through the zap |
| 09:52.16 | jblack | plops in a MusicOnHold |
| 09:54.31 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:55.05 | jblack | might help if I put it before the ivr, not after. |
| 09:55.18 | jblack | think I can drop all 10 at once, or should I stagger them? |
| 09:56.37 | Strom | you should be fine doing 10 at once |
| 09:58.17 | Strom | http://www.jerkcity.com/jerkcity2531.html |
| 09:59.28 | *** join/#asterisk xnosx (n=xnosx@212.145.172.127) |
| 10:03.12 | E-bola3 | hmm |
| 10:03.22 | E-bola3 | wonders where he's gonna get a wave file saying "Connecting..." |
| 10:03.39 | E-bola3 | hmm vm-dialout could do |
| 10:04.58 | Strom | why do you need a wav file of that? |
| 10:05.16 | E-bola3 | So i can play it for a caller |
| 10:05.43 | Strom | that's a non-answer |
| 10:05.51 | Strom | why do you need to play it for a caller? |
| 10:06.52 | E-bola3 | Well the dialplan goes something like this: If the phones are put into "nite mode" then the caller has a choice of either leaving a message or getting connected to an on call person. IF the caller chooses to be connected to the on call guy, I want them to hear "Connecting" after having pressed that option, while it dials the mobile phone |
| 10:07.22 | Strom | how about just the file that says "one moment please"? |
| 10:08.29 | E-bola3 | Could work also, i think i prefer the "please wait while i connect your call" from vm-dialout |
| 10:10.11 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 10:12.06 | jblack | that's odd. they aren't lasting long enough |
| 10:12.56 | jblack | 20 active channels, 10 active calls |
| 10:13.17 | Strom | jblack: are you answering before playing music on hold? |
| 10:13.43 | jblack | actually, they're in the ivr, giving up after about 10 sec. |
| 10:13.47 | jblack | I'm going to replace the ivr with moh |
| 10:14.02 | jblack | so that joan osborne is singing to joan osborne |
| 10:14.06 | jblack | 20 times |
| 10:14.15 | Strom | how romantic |
| 10:16.08 | jblack | oh shit. |
| 10:16.34 | Strom | ? |
| 10:16.56 | jblack | somehow, I fell through to 511, so my wire monkey just got called at 3am. 20 times. on his cell. |
| 10:17.09 | Strom | ...not good |
| 10:17.17 | Strom | this is why you test with a single phone call first |
| 10:17.19 | jblack | that's ok. he deserves it for being a cheapskate. |
| 10:17.21 | jblack | I did! |
| 10:17.25 | Strom | oh |
| 10:18.48 | jblack | <PROTECTED> |
| 10:19.46 | Strom | just have it loop spam.wav indefinitely :) |
| 10:19.49 | jblack | oh, it's staring me in the face. |
| 10:21.59 | jblack | well, this is interesting. |
| 10:22.18 | jblack | i'm getting congestion on call 6 on. |
| 10:23.30 | Strom | is there a known limit on concurrent calls? |
| 10:23.37 | jblack | on a PRI? |
| 10:23.39 | Strom | who's sending congestion? |
| 10:23.54 | jblack | not sure. I need to bump up verbosity |
| 10:25.50 | jblack | I'm dumping the files on pbx2 |
| 10:25.59 | jblack | so they're sipping over to pbxin, then coming back, supposedly. |
| 10:26.10 | jblack | pbxin is giving congestion |
| 10:26.32 | jblack | and that's because pbxin is getting congestion. |
| 10:27.12 | jblack | I can't tell tell who's hanging up on whom. |
| 10:27.18 | Strom | dont make it complex |
| 10:27.21 | Strom | do it all on pbxin |
| 10:27.33 | *** part/#asterisk BeeBuu (n=beebuu@218.13.82.138) |
| 10:28.11 | Strom | http://www.jerkcity.com/jerkcity1247.html |
| 10:29.48 | jblack | Yeah, I'll do that. |
| 10:30.02 | jblack | I asn't quite sure how to form a callfile for zap callfiles |
| 10:30.36 | Strom | same deal |
| 10:30.57 | Strom | zap, sip, doesnt matter |
| 10:31.12 | jblack | I'm gonna have to do a pile of playbacks. no moh on pbxin |
| 10:31.56 | Strom | yea, like i said...just have it loop playbacks infinitely |
| 10:32.07 | *** join/#asterisk Trifixxx (n=Mike@c-67-169-21-227.hsd1.ca.comcast.net) |
| 10:32.20 | *** part/#asterisk Trifixxx (n=Mike@c-67-169-21-227.hsd1.ca.comcast.net) |
| 10:34.17 | masus | does anyone know an free opensource predictive dialer script? Thanks |
| 10:36.34 | Strom | how about kill_yourself_and_fuck_the_body.pl |
| 10:39.07 | jblack | definitely weird. |
| 10:39.26 | jblack | 4 calls worked. |
| 10:40.06 | jblack | 5 worked, 5 are stuck on ringing |
| 10:40.19 | jblack | could there be a limit on outgoing ? |
| 10:40.38 | Strom | that will depend on how stupid your provider is |
| 10:40.52 | jblack | sorry. I mean callfiles. |
| 10:41.05 | jblack | not concurrent calls. |
| 10:42.08 | jblack | big piles of .991s and .992s |
| 10:42.44 | *** join/#asterisk codestr0m (n=asura@76.74.174.194) |
| 10:42.52 | jblack | I'll sleep a second between starts |
| 10:43.19 | jblack | I think they're limisiting to 5 concurrent calls. |
| 10:44.13 | Strom | lame on a srick |
| 10:44.18 | Strom | s/sri/sti/ |
| 10:44.29 | jblack | That's what a 600 dollar a month pri gets ya |
| 10:45.07 | jblack | actually, 10 concurrent calls. |
| 10:45.09 | jblack | 5 in, 5 out |
| 10:45.32 | Strom | what fucking good is that? it's 23 channels |
| 10:48.23 | jblack | $600 a month, unlimited long distance. |
| 10:48.36 | *** join/#asterisk lzhang (n=lzhang@24-155-240-48.dyn.grandenetworks.net) |
| 10:48.55 | Strom | the phrase "dongtacular" comes to mind |
| 10:48.55 | jblack | Unless somewhere I somehow capped calls, but I don't remember ever doing that. The dialplan for pbxin is dead-simple |
| 10:49.57 | jblack | Do you still feel sure that it's the rhino? |
| 10:50.14 | jblack | I suppose I could have him put it on loopback in a few hours, and zttest it for awhile to verify, no? |
| 10:50.43 | raz | anyone know a fax-over-voip *client* for linux? |
| 10:50.48 | Strom | where are those config files again? |
| 10:50.54 | jblack | raz: It's complicated. |
| 10:51.00 | Strom | raz: i hope you mean t.38 |
| 10:51.01 | jblack | http://linuxguru.net/~jblack/calls/ |
| 10:51.09 | jblack | oh yeah. look at that. |
| 10:51.19 | raz | Strom, whatever works, i just want to send faxes over my SIP line :) |
| 10:51.37 | Strom | raz: does your provider support t.38? |
| 10:51.44 | jblack | trying to start 100 calls, 1 a second , I instantly drop into .998, .996.... |
| 10:52.26 | Strom | because fax over voice over IP is very much like jamming white hot railroad spikes six feet into each one of your orifices |
| 10:52.32 | raz | Strom, looking |
| 10:52.42 | *** join/#asterisk rootlogin (n=root@saturn2.franken.de) |
| 10:52.49 | jblack | Best: 100.000 -- Worst: 99.993 -- Average: 99.998045, Difference: 100.000062 |
| 10:52.53 | jblack | How's that look. :( |
| 10:53.02 | Strom | looks fine |
| 10:53.08 | Strom | it may not be a timing thing |
| 10:53.19 | jblack | I thought you said < .9975 was bad? |
| 10:53.23 | raz | Strom, hmm.. they only talk about inbound fax (and that works, i've tested it) |
| 10:53.46 | jblack | remember; the agent's main problem is that calls that go through sound packet-droppy |
| 10:54.00 | Strom | i thought you said that they sounded square-wavey |
| 10:54.14 | jblack | well, you listen to 'em. |
| 10:54.21 | Strom | ok |
| 10:54.22 | Strom | hang |
| 10:55.15 | raz | ah... found it, they say "fax sending is not possible" |
| 10:55.18 | raz | now that sucks :\ |
| 10:55.34 | jblack | it's possible. Just an incredible, nearly impossible, pain in the ass. |
| 10:55.36 | raz | guess i'll have to find some internet fax service |
| 10:55.46 | Strom | dear linux media player applications: all of you go die in a fire now |
| 10:55.48 | Strom | thanks |
| 10:55.49 | Strom | love, |
| 10:55.50 | Strom | strom |
| 10:55.53 | jblack | mpg123 will work |
| 10:55.55 | Strom | <3 <3 <3 xoxoxoxoxxoxo |
| 10:56.07 | Strom | no, NOTHING works |
| 10:56.11 | Strom | i have to restart x |
| 10:56.14 | Strom | fucking shit |
| 10:56.15 | raz | jblack, i hate fax anyways, it's a relic from the past and a waste of paper. but so many people want faxes with signatures... |
| 10:56.30 | jblack | that's because they're legal documents. |
| 10:56.51 | raz | jblack, as if there was any difference between printing them and sending them as pdf... |
| 10:56.52 | Strom | brb |
| 10:57.13 | jblack | that's the difference. Faxes are legal documents. pdfs aren't. |
| 10:57.47 | *** join/#asterisk d-k-t-2 (n=dt@125.120.129.131) |
| 10:57.58 | raz | jblack, yea and that's stupid. i can print my pdf and you'll never be able to tell whether it was faxed or mailed. |
| 10:58.03 | jblack | This is what you get when your peers vote in a congress that puts a man that thinks the internet is "a series of tubes" in charge of the internet. |
| 10:58.10 | raz | lol |
| 10:58.11 | *** join/#asterisk Strom (n=strom@208.127.172.112) |
| 10:58.12 | Strom | ok |
| 10:58.16 | Strom | lets try this again |
| 10:58.22 | jblack | crosses his fingers |
| 10:58.32 | Strom | BONERS AND DONGS (and it took pepperidge farm to bring them together) |
| 10:59.06 | jblack | Is there some way I can make it easier for you? |
| 10:59.37 | Strom | no, now it works |
| 10:59.55 | Strom | give me that URL again though |
| 10:59.57 | jblack | until you turn your back |
| 11:00.25 | jblack | I imagine you had a paused youtube somewhere locking oss |
| 11:00.35 | Strom | nope |
| 11:00.37 | jblack | or alsa for that matter. I don't think alsa mixes |
| 11:00.38 | Strom | no youtube |
| 11:00.46 | Strom | whatever -- it's all total shit |
| 11:00.51 | jblack | I'm speaking metaphorically |
| 11:01.23 | Strom | URL please |
| 11:01.31 | jblack | http://linuxguru.net/~jblack/calls |
| 11:03.06 | Strom | listening |
| 11:03.50 | raz | bbl |
| 11:03.51 | *** part/#asterisk raz (n=y@unaffiliated/raz) |
| 11:04.18 | Strom | what does it sound like from the called party's end? |
| 11:04.36 | jblack | unknown |
| 11:04.42 | jblack | they seem to act oblivious |
| 11:05.03 | Strom | well, let's try this -- let me call a recording |
| 11:05.37 | jblack | this happens for about 5% of calls, and only seems to happen when there's 4 or more calls going on |
| 11:06.00 | jblack | but that's not much more than a guess. |
| 11:06.30 | Strom | hm |
| 11:06.31 | jblack | Agents don't complain where only 3 are doing phones. agents frequently complain when 10 are calling |
| 11:07.08 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 11:07.13 | Strom | id be interested to know what kind of equipment is on the carrier's end before you go buying more hardware |
| 11:07.34 | jblack | my suspicion is tinfoil and duct tape. Lots of duct tape |
| 11:07.45 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 11:07.53 | jblack | mind if I repair extensions.conf before I forget? |
| 11:07.59 | Strom | not at all |
| 11:09.16 | jblack | I'm starting to get the impression that this isn't a case of incomptence on my part. |
| 11:09.44 | Strom | maybe, maybe not |
| 11:09.58 | jblack | we're not certain they limit to 10 concurrent calls. It may be 10 concurrent calls to 1 number. |
| 11:10.16 | jblack | though that would seem like an odd guard. |
| 11:11.39 | *** join/#asterisk xnosx (n=xnosx@212.145.175.26) |
| 11:11.51 | *** part/#asterisk cfh (n=luca@host196-137-static.90-82-b.business.telecomitalia.it) |
| 11:13.56 | jblack | What sort of questions would you like me to ask them? |
| 11:14.29 | jblack | We started a trouble ticket yesterday. They spent about an hour on the machine according to status indicator on the pri |
| 11:15.05 | Strom | where is their switch? what's between the switch and their terminal? |
| 11:15.25 | jblack | Don't know. |
| 11:15.26 | razu | can someone tell me what does this alert mean : [Jun 13 14:14:22] ERROR[31606]: chan_zap.c:8248 zt_pri_error: !! Got reject for frame 8, but we only have others! |
| 11:15.31 | jblack | I can get a picture for you. |
| 11:15.53 | Strom | sure |
| 11:16.02 | jblack | Not at the moment. It's 4 am there |
| 11:16.46 | Strom | i'm on the west coast too |
| 11:16.48 | Strom | i know the time :) |
| 11:17.28 | jblack | Ahh. Here, there weird chirping things outside, and a great ball of fire is up over the horizon. |
| 11:18.35 | jblack | each day, I hope the great ball of fire, which burns my eyes, will consume the chirpy things, just so I can have some STFU time... but no such luck |
| 11:18.48 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:20.24 | jblack | I could really increate the productive hours with an automatic bb gun and a huge fire extinguisher. |
| 11:20.49 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583881.dsl.bell.ca) |
| 11:25.03 | *** part/#asterisk codestr0m (n=asura@76.74.174.194) |
| 11:31.07 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:33.18 | *** join/#asterisk ManxPower (n=manxpowe@75.sub-75-201-141.myvzw.com) |
| 11:36.10 | *** join/#asterisk BBHoss (n=hoss@c-68-62-175-86.hsd1.al.comcast.net) |
| 11:37.02 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-87-136.vif.net) |
| 11:41.44 | razu | does anyone have list of explanations for isdn state codes (not cause codes) ? |
| 11:44.22 | Strom | doesnt q.931 cover that? |
| 11:45.48 | jsmith | Yeah... see q931.c in libpri |
| 11:46.11 | Strom | or just download q.931 from ITU |
| 11:46.13 | Strom | ~itu |
| 11:46.14 | jbot | extra, extra, read all about it, itu is the International Telecommunication Union. Current versions of ITU-T recommendations (Q,931, T.38, V.32, et cetera) are available for free in PDF format from their website: http://www.itu.int/rec/T-REC/e |
| 11:46.15 | jsmith | But I'm pretty sure they all come from the q.931 specification |
| 11:46.16 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:46.16 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 11:46.41 | razu | I have freaky situation ... when I call into E1 and hangup my mobile, call still stay up ... only solution to kill it is via asterisk cli |
| 11:46.43 | razu | :( |
| 11:48.47 | yang | I am unable to set CALLERID, the number always comes out as 059209580 (I should get 059209590) - here are my settings and error output http://www.pastebin.sk/en/6988/ |
| 11:49.15 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:49.48 | lmadsen | yang: what technology? |
| 11:50.11 | yang | lmadsen: asterisk 1.4 |
| 11:50.23 | lmadsen | yang: I mean, what is the end point? ITSP? |
| 11:50.33 | lmadsen | via SIP, etc.. ? |
| 11:50.34 | yang | the technician on the other side told me that he is getting some sort of a string before my CALLERID, i wonder how is tht possible |
| 11:50.46 | yang | via SIP |
| 11:51.25 | yang | ok i received also error fromhis side, uploading that |
| 11:51.50 | *** join/#asterisk xnosx (n=xnosx@212.145.55.118) |
| 11:52.17 | lmadsen | what does the SIP debug look like? Are you sending it via Remote-Party-ID? |
| 11:52.24 | lmadsen | sendrpid=yes in sip.conf |
| 11:53.25 | yang | http://www.pastebin.sk/en/6989/ |
| 11:53.36 | *** join/#asterisk christophocles (n=christop@cpe-68-201-114-229.gt.res.rr.com) |
| 11:54.01 | yang | there is a SIP debug from operators side, if you check |
| 11:55.35 | christophocles | hi, i am trying to follow the asterisk book to set up my first PBX, and I hit a major problem... my PBX will not recognize any DTMF tones! it will play sound back to me but if i try to dial an extension, absolutely nothing happens (even in the logs) and it eventually just times out and goes to exten => t,1,Playback(vm-goodbye) |
| 11:55.39 | yang | ;sendrpid = yes is commented |
| 11:55.42 | s0ck | anyone using the cutglass ivr prompts? |
| 11:55.46 | christophocles | anybody have a clue as to what the problem is? |
| 11:58.44 | lmadsen | yang: you should be sending RPID for the CallerID, especially if you are modifying it and it isn't matching what your username is |
| 11:58.47 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 11:58.59 | lmadsen | sendrpid=yes should be uncommented |
| 11:59.02 | yang | lmadsen: ok I will try to uncomment |
| 11:59.12 | lmadsen | don't try to uncomment it... actually do it :) |
| 12:00.45 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:01.30 | yang | lmadsen: still same error |
| 12:02.06 | lmadsen | then your ITSP is parsing on the From: or Contact: header, and unless you change your username, there isn't anything you can do |
| 12:03.14 | yang | From: "90" <sip:338606057@slo.detel.eu>;tag=as39a6ec8e |
| 12:03.18 | yang | Should it be |
| 12:03.31 | yang | From: "05920959090" <sip:338606057@slo.detel.eu>;tag=as39a6ec8e |
| 12:03.39 | yang | From: "059209590" <sip:338606057@slo.detel.eu>;tag=as39a6ec8e |
| 12:04.19 | lmadsen | right -- it's parsing on the sip:338606057@.... it sounds like |
| 12:04.53 | yang | The technician told me that I should loose the 338606057 and place the number there (somehow) |
| 12:04.57 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 12:05.13 | loompek | yang like i told you |
| 12:05.29 | *** join/#asterisk jack_sparo (n=eddy@91.73.203.98) |
| 12:05.29 | loompek | Set(Callerid(num)=059209590) |
| 12:05.31 | jack_sparo | where are music on hold files saved? |
| 12:05.37 | loompek | err.. CALLERID(num) |
| 12:05.46 | lmadsen | loompek: he did that |
| 12:05.51 | yang | i have done it |
| 12:06.02 | yang | in several ways |
| 12:06.35 | *** join/#asterisk ManxPower (n=manxpowe@75.sub-75-201-141.myvzw.com) |
| 12:08.04 | loompek | umm |
| 12:08.16 | loompek | your outgoing number for detel should be WITHOUT 0 prefix |
| 12:08.16 | loompek | so |
| 12:08.21 | loompek | 59209590 |
| 12:08.30 | yang | same thing happens |
| 12:08.30 | loompek | i'm willing to bet on it |
| 12:08.38 | loompek | i just checked with my friend's detel config |
| 12:09.06 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:09.44 | loompek | afterwards you changed the config.. did you do a dialplan reload? |
| 12:09.49 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:09.49 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:10.35 | yang | loompek: extensions reload |
| 12:10.53 | christophocles | can anybody help me understand why my asterisk pbx won't register any dtmf tones? |
| 12:11.05 | ManxPower | Leading 1, 0, and 00 are not port of CallerID info. |
| 12:11.55 | ManxPower | christophocles: chances are the phone is sending DTMF in a format different from what type of DTMF Asterisk is configured to expect. Generally you want the phone and Asterisk to be set to RFC2833 (aka AVT) |
| 12:13.09 | christophocles | manxpower, i am dialing in using my POTS to an ipkall number that connects to my asterisk box at home |
| 12:13.28 | christophocles | i cannot change anything on my phone or my ipkall service, only within asterisk |
| 12:13.32 | *** join/#asterisk sack (n=sack@249.Red-81-32-160.dynamicIP.rima-tde.net) |
| 12:13.37 | ManxPower | christophocles: That is MUCH more complicated. |
| 12:13.43 | christophocles | why is that more complicated? |
| 12:13.48 | ManxPower | Why can't you change the setup on the phone? |
| 12:13.58 | christophocles | what would i change? its just a regular phone |
| 12:14.02 | ManxPower | christophocles: because you do not have control over all the devices involved. |
| 12:14.10 | ManxPower | christophocles: what is the phone connected to? |
| 12:14.20 | christophocles | an att phone line i guess |
| 12:14.23 | christophocles | just a regular landline |
| 12:14.41 | ManxPower | A phone and a phone line are different. Which is it? |
| 12:15.16 | ManxPower | So you have analog phone -> analog line -> IPKall -> Internet -> Asterisk? |
| 12:15.17 | christophocles | im not understanding the question? I am paying for 'regular phone service' and i have a cord coming from a jack in the wall and it is plugged into a 'regular telephone' |
| 12:15.21 | christophocles | not an ip phone |
| 12:15.25 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 12:15.31 | christophocles | yes that is correct |
| 12:15.31 | ManxPower | so that line is not plugged into asterisk in any way? |
| 12:15.40 | christophocles | no, it has nothing to do with asterisk |
| 12:15.50 | christophocles | i can use my cell phone to dial in... actually i will try that |
| 12:16.00 | ManxPower | Contact IPKall and ask them what DTMF format they expect you to send. Set Asterisk to that format. |
| 12:16.36 | christophocles | ok, i'll read up on their forums but for some reason the people who ask questions similar to mine have no replies and the threads are closed... ?? |
| 12:16.42 | lmadsen | or you could try setting dtmfmode=auto (which defaults to rfc2833 if nothing is offered) if using Asterisk 1.4 |
| 12:16.51 | lmadsen | there is only like... 3 settings you could try... :) |
| 12:17.01 | ManxPower | lmadsen: I never trust =auto 8-) |
| 12:17.04 | christophocles | i tried all three already |
| 12:17.17 | lmadsen | ManxPower: I don't either :) |
| 12:17.34 | ManxPower | christophocles: then I guess IPKall is not compatable with Asterisk DTMF. |
| 12:17.56 | lmadsen | would IPKall maybe be using asterisk 1.2 to deliver calls? if so, there is rfc2833compensate=yes you could try |
| 12:18.03 | ManxPower | You tried all three DTMFmode options and none of them worked. There is not much else you can do. |
| 12:18.20 | christophocles | jeez, that makes ipkall completely worthless, lol |
| 12:18.23 | christophocles | i'll try this rfc2833compensate=yes |
| 12:18.37 | lmadsen | I'd like to know what version of asterisk you're running |
| 12:18.40 | ManxPower | christophocles: perhaps you did something wrong with setting the DTMFmode? |
| 12:19.19 | christophocles | [general] dtmfmode=rfc2833 |
| 12:19.25 | christophocles | on 2 lines of course |
| 12:19.26 | ManxPower | christophocles: I've never actually seen an ITSP that would just not work with Asterisk DTMF, but you just said none of the three options work, so IPKall must be the first one. |
| 12:19.47 | christophocles | it just seems strange that nobody would answer those questions on the forum |
| 12:19.52 | ManxPower | christophocles: I don't know of dtmfmode is allowed in [general] What does sip.conf.sample say about it? |
| 12:20.09 | lmadsen | ManxPower: you can set it there as a default option |
| 12:20.14 | lmadsen | but it doesn't override the peer |
| 12:20.20 | ManxPower | christophocles: those forums are the blind leading the deaf to fined the mute. |
| 12:20.31 | ManxPower | lmadsen: thanks for the info |
| 12:20.55 | lmadsen | given that -- I never set it in [general] and always set it in the peer |
| 12:21.02 | lmadsen | gym time! |
| 12:21.19 | ManxPower | lmadsen: me too, which is why I didn't know if you could put it in [general] |
| 12:23.43 | christophocles | ok theres rfc2833, auto, and inbound right? |
| 12:23.50 | christophocles | err, whats the third one? |
| 12:24.13 | christophocles | inband, i got it |
| 12:24.21 | christophocles | its still not working :( |
| 12:30.50 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 12:33.01 | *** join/#asterisk allankardec (n=root@20150187219.user.veloxzone.com.br) |
| 12:35.53 | loompek | yang who's the man... who's the man! |
| 12:36.07 | yang | you re the man |
| 12:41.15 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:44.03 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:45.41 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 12:49.23 | viraptor | is there a way to tell if asterisk was compiled with ztdummy support? |
| 12:50.49 | [TK]D-Fender | viraptor: Try loading the module. And Ztdummy is part of Zaptel, not Asterisk |
| 12:51.07 | ManxPower | viperdude: Asterisk is NEVER compiled with ztdummy support. It is compiled with Zaptel support,. |
| 12:51.28 | *** join/#asterisk delparnel (n=delparne@KTNRON06-1168103470.sdsl.bell.ca) |
| 12:51.47 | viraptor | ok, so if I have a working asterisk and ztdummy loaded, how do I tell if asterisk is using it? |
| 12:51.52 | ManxPower | viperdude: Was zaptel installed when you install Asterisk |
| 12:52.17 | ManxPower | viperdude: Why do you have ztdummy loaded? |
| 12:52.56 | viraptor | yes, but I think it didn't find correct zaptel/*.h files and I'm not sure how to check |
| 12:53.27 | viraptor | ManxPower: I want to run it for conferences - got choppy sound sometimes |
| 12:53.28 | ManxPower | viperdude: if chan_zap.so was built then asterisk detected zaptel when you built Asterisk. WHY are you wanting to use ztdummy? |
| 12:53.49 | viraptor | afaik that's the way to make it run more smooth, right? |
| 12:53.54 | ManxPower | viperdude: MeetMe won't work at all unless Asterisk detects a Zaptel timing source like ztdummy |
| 12:54.28 | [TK]D-Fender | viraptor: So if you've got no zaptel hardware and Meetme works at all, then yes, its clearly loaded |
| 12:54.30 | ManxPower | viperdude: in the Asterisk CLI, do core zap show channels |
| 12:55.28 | ManxPower | what does it return? |
| 12:55.55 | russellb | minor clarification ... MeetMe does not use zaptel for timing, it uses it for conference mixing (which internally to zaptel, requires timing) |
| 12:56.01 | ManxPower | I'm not going to sit here all day and wait for your response. |
| 12:56.21 | [TK]D-Fender | ManxPower: And fix your aim a bit ;) |
| 12:57.16 | viraptor | there's no "core zap..." + ManxPower take it easy -> I don't need that !@# - it takes time to check everything properly, ok... you don't have to help me :/ |
| 12:57.28 | ManxPower | viperdude: if you are on 1.4 and there is no "core zap" then Asterisk is not compiled with zaptel support. |
| 12:57.37 | _foxfire_ | viraptor what version of asterisk are u using ? |
| 12:57.38 | [TK]D-Fender | viraptor: Either way, you have no hardware and Meetme works at all, right? |
| 12:57.38 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:57.45 | ManxPower | viperdude: OK. Someone else can help you then. We are here UNPAID. |
| 12:57.59 | ManxPower | [TK]D-Fender: I think he's using app_conference, but is not telling us. |
| 12:58.07 | ManxPower | But he's your problem now. |
| 12:58.22 | [TK]D-Fender | ManxPower: You are projecting all sorts this morning... go caffeinate. |
| 12:58.33 | [TK]D-Fender | viraptor: Well? |
| 12:58.35 | *** join/#asterisk RoyK (n=roy@ip-153-17-149-91.dialup.ice.no) |
| 12:58.39 | ManxPower | [TK]D-Fender: alreadt cafineated |
| 12:58.50 | [TK]D-Fender | ManxPower: Not properly. Go adjust. |
| 12:59.03 | ManxPower | I guess it has not reached my fingers yet 8-) |
| 13:00.22 | viraptor | ehh... broken routing... I'm not sure that conference lands where I think it does :/ anyways - I've found an answer to checking if zap is loaded - I'll get back here if there are still some problems |
| 13:00.24 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:00.29 | viraptor | thanks for answers :) |
| 13:00.43 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:00.51 | [TK]D-Fender | viraptor: Conference... LANDS? |
| 13:01.11 | viraptor | (where is it handled) |
| 13:01.34 | ManxPower | mutters something that sounds like "Golly, Beave, is it OK to play with routing?" "It sure is, Wally, as long as you do it in private and wash your hands after!" |
| 13:01.38 | viraptor | we've got many asterisks... too many ;) |
| 13:02.53 | russellb | takes a nap and drifts off to conference land |
| 13:05.05 | [TK]D-Fender | I just tt-monkey paged my marketing dept. Hillarity. |
| 13:05.31 | [TK]D-Fender | viraptor: Where what is "handled"? Its 1 dialplan app. there is no "handling". |
| 13:06.01 | ManxPower | Have a *great* day, [TK]D-Fender 8-) |
| 13:06.18 | [TK]D-Fender | ManxPower: 8) |
| 13:06.55 | ManxPower | *grumble* Another trip to Lowes today. 8-( |
| 13:06.57 | awk | Please can somebody tell me why on bristuff install on my cdr table i'm getting this as a unique id, | 2008-06-13 14:13:40 | asterisk-1213359216.275 | |
| 13:07.06 | awk | why is it adding the prefix asterisk to the front? |
| 13:08.22 | awk | I can't see this anywhere inside cdr_addon_mysql.c |
| 13:08.32 | awk | please any help would be appricated |
| 13:08.48 | ManxPower | I'll bet your hostname is "asterisk" |
| 13:09.31 | [TK]D-Fender | awk: thats part of a "shared SQL" setup so that you can separate CDR pooled from multiple servers on 1 database table |
| 13:10.10 | *** join/#asterisk BBHoss (n=hoss@c-68-62-175-86.hsd1.al.comcast.net) |
| 13:10.54 | awk | [TK]D-Fender: how can I make it vanish? |
| 13:10.56 | awk | :) |
| 13:11.16 | awk | or else i'm going to ahve to change the billing engine to some how ignore the first part when matching uniqueid's... |
| 13:11.18 | [TK]D-Fender | awk: Don't know specifically, but I'm sure its pretty quick to find. |
| 13:12.06 | awk | must be prefix asterisk with a calldate() or something |
| 13:12.15 | awk | ok, let me go look some more |
| 13:13.54 | jsmith | awk: Do you have "systemname=asterisk" in asterisk.conf by chance? |
| 13:13.57 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:14.55 | awk | jsmith: no def not |
| 13:16.03 | awk | mysql> select calldate,uniqueid from cdr order by calldate limit 9000,400; |
| 13:16.07 | jsmith | awk: Ok, just curious |
| 13:16.10 | awk | | 2008-06-13 14:07:16 | 9344 | |
| 13:16.16 | awk | ok, thats my entries I added |
| 13:16.28 | awk | now this is asterisk adding it | 2008-06-13 14:17:09 | asterisk-1213359425.283 | |
| 13:24.06 | jaytee | wow! awk is here! is sed around too today? :-) |
| 13:24.18 | *** part/#asterisk mags2 (n=egray@ampulex.whoi.edu) |
| 13:29.29 | *** part/#asterisk jsmith (n=jsmith@72.21.36.138) |
| 13:30.53 | *** join/#asterisk coppice (n=chatzill@240.166.17.210.dyn.pacific.net.hk) |
| 13:31.39 | *** join/#asterisk orionr (n=arosen@c-76-26-221-76.hsd1.sc.comcast.net) |
| 13:33.05 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-df7065eab36ae082) |
| 13:33.25 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 13:34.26 | awk | jaytee on another network yes... |
| 13:34.27 | awk | :P |
| 13:40.34 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:40.51 | *** join/#asterisk kink0 (n=xchat@212.170.176.86) |
| 13:40.53 | kink0 | hello |
| 13:40.56 | dandre | hello |
| 13:41.30 | kink0 | can someone give me a little help ? |
| 13:42.04 | kink0 | I have this scenario: SIP/H323 g729/g723 to Asterisk and then ASterisk pass-through to a Cisco media gateway, |
| 13:42.13 | kink0 | but I got several : Asked to transmit frame type 256, while native formats is 1 |
| 13:42.28 | dandre | How can I set a default callerid for an inbound zap channel? I use callerid=asreceived but I want to set it to a more significant value if not frovided by the party |
| 13:42.37 | kink0 | I guess is due because client -> Asterisk choose i.e. g723 , while Asterisk-Cisco leg did g729 |
| 13:42.49 | *** join/#asterisk af_ (n=getsmart@88-149-230-57.dynamic.ngi.it) |
| 13:42.51 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 13:42.58 | [TK]D-Fender | kink0: And * can't transcode. Yes, that would be an issue |
| 13:43.07 | Zeeek | TGIF!!! |
| 13:43.08 | *** join/#asterisk hyegeek (n=hakimian@rw.aha.com) |
| 13:43.15 | [TK]D-Fender | dandre: Set it in your dialplan. |
| 13:43.43 | kink0 | [TK]D-Fender, I pretend no do transcoding here, just to find a way to choose same codec in both legs |
| 13:43.55 | dandre | ok |
| 13:44.17 | [TK]D-Fender | kink0: its doing it clearly. Go review your settings |
| 13:44.35 | kink0 | I did transcoding with some g729 licensed from digium and g723 from IPP , but I want avoid transcoding or so, due to CPU utilization |
| 13:44.53 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 13:44.59 | kink0 | [TK]D-Fender, but how to choose the same codec in both legs when one use g723 and the other one g729 |
| 13:45.07 | kink0 | the scenario is like this: |
| 13:45.31 | kink0 | client ( g723) -> Asterisk ( g723, g729) -> Cisco ( g729, g723 ) |
| 13:45.32 | [TK]D-Fender | kink0: set ONE codec for your peers, and make them the same. |
| 13:45.52 | kink0 | yeahh.. but this peer can route calls with both codecs :( |
| 13:46.39 | *** part/#asterisk hyegeek (n=hakimian@rw.aha.com) |
| 13:47.02 | x86 | pick which one you want to use then |
| 13:47.19 | kink0 | I was trying to figure if some channel variable like ${CODEC} then do an if and send to one or other priority, based on codec, adding prefixing and forwarding it to one of two dial-peers on Cisco |
| 13:47.31 | kink0 | where each dial-peers uses g729 or g723 |
| 13:47.49 | kink0 | but no way to get the codec in use on the client->Asterisk leg |
| 13:47.52 | *** join/#asterisk ZaVoid (n=zavoid@75.147.121.177) |
| 13:47.55 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 13:48.06 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 13:48.07 | hsv-al | . |
| 13:48.18 | af_ | is it possible to send flash via a sip ata ? I am trying to speak with a legacy pbx with an fxo ata |
| 13:48.27 | [TK]D-Fender | hsv-al: Wow, you DO have a point! |
| 13:48.43 | af_ | * <-> sip ata <-> fxo <-> pbx |
| 13:48.45 | kink0 | [TK]D-Fender, any way to proxy all RTP direct to media gateway in ASterisk ? |
| 13:48.47 | [TK]D-Fender | af_: Clearly this depends on your ATA |
| 13:48.47 | *** join/#asterisk ZaVoid (n=zavoid@75.147.121.177) |
| 13:48.59 | [TK]D-Fender | kink0: No, * is not a proxy |
| 13:49.10 | af_ | [TK]D-Fender, like the choice of ata brand? what model I could be sure of that? |
| 13:49.14 | dandre | where can I set the default callerid information to something else than asterisk. For instance 'My PBX' |
| 13:49.26 | [TK]D-Fender | af_: I never said I knew of one that could. |
| 13:49.26 | af_ | or better, [TK]D-Fender what sta support that? |
| 13:49.30 | kink0 | yap... :( may be freeswitch then, but I know better Asterisk than woomera and so |
| 13:49.34 | *** join/#asterisk chendy (n=chatzill@58.251.115.51) |
| 13:49.36 | [TK]D-Fender | dandre: In the source. |
| 13:49.40 | Zeeek | stained himself black with leaky inket and can't be seen |
| 13:49.41 | hsv-al | d-fender after googling for ages yesterday |
| 13:49.48 | hsv-al | i came across a group of people who once worked at rim |
| 13:49.57 | hsv-al | who are working on SIP/IAX SOft clients for blackberry models(modern ones) |
| 13:50.00 | af_ | [TK]D-Fender, i miss the dependency thing then |
| 13:50.05 | dandre | ok |
| 13:50.58 | Zeeek | hsv-al these are people that had rim-jobs, then? |
| 13:52.12 | *** join/#asterisk xnixan_ (n=xnixan@unaffiliated/xnixan) |
| 13:53.17 | Zeeek | Free SMS alerts through asterisk using Twitter: http://www.uk-experience.com/2008/06/13/asterisk-twitter-call-monitor/ |
| 13:53.20 | *** join/#asterisk s0lid (n=s0lid@122.53.69.11) |
| 13:53.23 | Zeeek | nice if twitter is up |
| 13:53.35 | Zeeek | (uses curl) |
| 13:54.18 | hsv-al | zeeek dont know, some of them like to hit balls during lunchtime |
| 13:54.27 | hsv-al | :) - . . . . . at the golf course/driving range |
| 13:55.11 | Zeeek | hsv-al nice comeback :) |
| 13:58.03 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 14:00.15 | Zeeek | In exactly 2 hours you will be asked to tune to... http://VoipUsersConference.org for information on how to join us. Please do. |
| 14:01.20 | Zeeek | I love when they ask you build a "dynamic" site and then want to demo it without Internet access. |
| 14:02.13 | cpm | heh |
| 14:03.20 | *** join/#asterisk xnosx (n=xnosx@212.145.55.118) |
| 14:04.32 | Zeeek | I suppose I could instal LAMP on the laptop and have them pay? |
| 14:05.39 | *** join/#asterisk CVirus (n=GoD@41.233.138.197) |
| 14:09.27 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 14:10.38 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:14.37 | jaytee | Is this sytax valid for setting CALLERID to Unavailable if a call comes in with caller ID blocked? exten => 123,1,Set(CALLERID(NUM)=${IF(ISNULL(${CALLERID})?Unavailable:${CALLERID})) |
| 14:16.24 | *** join/#asterisk xnosx (n=xnosx@212.145.55.118) |
| 14:17.00 | [TK]D-Fender | jaytee: No. Why are you referencing some sort of variable instead of the function? Uniformity <- |
| 14:17.25 | [TK]D-Fender | jaytee: And yeah, your braces are mashed in there |
| 14:18.51 | jaytee | [TK]D-Fender, darn :-( back to the drawing board |
| 14:19.24 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 14:23.57 | jaytee | [TK]D-Fender, how about this one? exten => 123,1,Set(CALLERID(num)=${IF(ISNULL(CALLERID(num))?Unavailable:CALLERID(num)}) |
| 14:24.39 | jets | Wow these feels a lot like perl |
| 14:24.52 | xpot | anyone know if the musiconhold error is fixed? |
| 14:25.30 | jets | jaytee, are you trying to approach it like it was perl? |
| 14:25.40 | jaytee | no |
| 14:27.12 | jaytee | I'm just trying to figure out the right command and syntax to check if the callerid is null or not and if it is set it to "unavailable" and otherwise leave it alone and proceed with the call. |
| 14:28.26 | *** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi) |
| 14:29.02 | ZenBSDi | Whats a good pastebin site these days for asterisk related examples? |
| 14:29.09 | ZenBSDi | pastebin.ca seems to be dead these days =p |
| 14:32.17 | ZenBSDi | Oh and does anyone know of any other companies like callwithus.com that has better rates maybe? |
| 14:32.36 | [TK]D-Fender | jaytee: Another mess. Fix your function references. |
| 14:32.53 | Zeeek | does anyone here use cartmanager for web payment? |
| 14:32.56 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:33.08 | [TK]D-Fender | ZenBSDi: .com |
| 14:35.27 | *** join/#asterisk e2e5 (n=chatzill@gw-Paramon-chel.suttk.ru) |
| 14:35.36 | Zeeek | [TK]D-Fender I finally found out why the Polycom boot log was always zero bytes after a reboot |
| 14:36.15 | [TK]D-Fender | Zeeek: namely? |
| 14:36.20 | Zeeek | Because the ftp user account was over quota, it wrote to the directory but couldn't write the file. I should have figured that out months ago! |
| 14:36.30 | [TK]D-Fender | Zeeek: SMRT |
| 14:36.32 | Zeeek | I am a fool. |
| 14:36.45 | Zeeek | I wish to be publicly whipped |
| 14:36.50 | *** part/#asterisk mintee (n=mintone@75.150.132.150) |
| 14:36.51 | Zeeek | Please proceed. |
| 14:37.00 | Zeeek | slouches on the bloch |
| 14:38.10 | Zeeek | it's the wait that hurts the most |
| 14:39.40 | Zeeek | I'm sure the Digium (tm) guys want to whip me. |
| 14:40.48 | ZenBSDi | [TK]D-Fender, problem with pastebin.com is .. they don't have a search like pastebin.ca |
| 14:40.53 | ZenBSDi | shrugs |
| 14:41.06 | Zeeek | what about a google search on the site? |
| 14:41.34 | hsv-al | -- Call accepted by 216.207.245.8 (format gsm) |
| 14:41.34 | hsv-al | <PROTECTED> |
| 14:41.34 | hsv-al | <PROTECTED> |
| 14:41.36 | hsv-al | jeez |
| 14:41.41 | hsv-al | clarity is horrible |
| 14:41.56 | *** join/#asterisk grEvenX (n=even@193.71.255.75) |
| 14:42.01 | raytruz` | If i want to ask a caller to enter a 10 digit phone, but not go to an extension (using it to look up their information) do I still have to make an extension to take the 10 digits they entered? |
| 14:42.04 | hsv-al | I havent learned yet how to get out of gsm, altering codecs, but is there a way to choose a better codec to negotiate? |
| 14:42.59 | Zeeek | raytruz` I am unclear as to the meaning of that sentence? |
| 14:43.17 | raytruz` | I want to ask caller for their 10 digit phone. |
| 14:43.26 | raytruz` | And store it. |
| 14:43.36 | ZenBSDi | raytruz Read() |
| 14:43.59 | Zeeek | I hate when they do that! Then the person answers and asks you again! |
| 14:44.04 | ZenBSDi | read at astDB and setting and storing variables |
| 14:44.15 | raytruz` | LOL |
| 14:44.19 | ZenBSDi | or else setup asterisk to a database like postgre or mysql and store the variables |
| 14:44.20 | ZenBSDi | =p |
| 14:44.22 | raytruz` | Well, its for when customer service is closed |
| 14:44.23 | Zeeek | but no you do not need an extension |
| 14:44.36 | raytruz` | if customer service is open, they are sent there automatically anyway :-) |
| 14:44.47 | Zeeek | "Please enter your...." and then live operator "What is your...?" |
| 14:44.51 | raytruz` | thanks ZenBSDi, Zeeek |
| 14:44.57 | raytruz` | LOL yeah |
| 14:44.57 | ZenBSDi | raytruz, just setup an afterhours box and forward them to it |
| 14:44.59 | raytruz` | I hate that too |
| 14:45.00 | Zeeek | says erm I just entered that" |
| 14:45.03 | raytruz` | like why waste my time |
| 14:45.06 | hsv-al | my lord the clarity is grainy |
| 14:45.22 | Zeeek | GSM sucks. Get over it and choose another codec |
| 14:45.24 | hsv-al | im using my deskphone at work to call my * at home via pstn, then i press 5 for it to do an iax2 to misery |
| 14:45.30 | hsv-al | and the clarity is so grainy, is that due to gsm? |
| 14:45.35 | raytruz` | ZenBSDi: thats pretty much what i'm doing, except it hits the ivr first because there are a couple other options that go to a different number :-) |
| 14:45.38 | Zeeek | Use ulaw |
| 14:45.51 | hsv-al | zeeek, i havent learned how to alter codec preferences yet |
| 14:45.58 | hsv-al | im going straight through the book, but is it tough for now? |
| 14:46.00 | hsv-al | im in chap 6 |
| 14:46.04 | raytruz` | Gsm isn't THAT bad |
| 14:46.06 | Zeeek | of course you have. I've seen you here for at least 2 months |
| 14:46.12 | raytruz` | all the voice prompts i recorded are in gsm |
| 14:46.41 | Zeeek | prompts are ok but calls are less |
| 14:47.10 | hsv-al | zeek, i honestly havent had came across in the book yet where it teaches how to fiddle |
| 14:47.14 | hsv-al | with codec preferences |
| 14:47.52 | Zeeek | hsv-al disallow=all |
| 14:48.00 | Zeeek | allow=ulaw |
| 14:48.03 | Zeeek | etc |
| 14:48.10 | hsv-al | well thats good if its that easy |
| 14:48.14 | Zeeek | stick it in sip.conf and iax.conf |
| 14:48.59 | hsv-al | what section |
| 14:49.04 | hsv-al | [general] ? |
| 14:49.10 | Zeeek | no in each peer entry |
| 14:49.17 | Zeeek | or first in default |
| 14:49.25 | hsv-al | my iax.conf is extremely basic |
| 14:49.32 | hsv-al | [general] only has autokill=yes |
| 14:49.32 | [TK]D-Fender | hsv-al: there has been compile issues with GCC 4.1 & GSM previously. Not sure if this is still current. |
| 14:49.44 | hsv-al | [idefisk] entry in iax.conf |
| 14:49.45 | Zeeek | you need to read a book or look it up, this tipic is well covered everywhere |
| 14:49.46 | hsv-al | from book example |
| 14:49.47 | xpot | question: is it "branches" or "trunk" that is the dev environment? I thought branches was the stable on and I am getting a musiconhold.c error on make |
| 14:50.10 | hsv-al | type=friend host=dynamic context=phones |
| 14:50.15 | hsv-al | thats all my iax.conf is for now |
| 14:50.33 | [TK]D-Fender | hsv-al: Where is your phone relative to your server? |
| 14:50.33 | hsv-al | from previous iax soft client examples , ie: idefisk for linux |
| 14:50.44 | hsv-al | pstn/* = 15 miles away |
| 14:51.17 | hsv-al | ill just add that allow=ulaw and disallow=all |
| 14:51.22 | hsv-al | in iax.conf, restart * |
| 14:51.50 | Zeeek | hsv-al have you ever read this article? http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 14:52.20 | hsv-al | no, im just plowing through the book |
| 14:52.26 | hsv-al | im in the pattern matching section now, is all |
| 14:52.35 | Zeeek | take a look, it's John Todd |
| 14:52.50 | Zeeek | It is a beginner's article that will jump start you |
| 14:53.49 | Zeeek | look at this: http://www.oreillynet.com//cs/user/view/cs_msg/23739 |
| 14:54.01 | Zeeek | ^^^^^^^^^ might clear things up a little ^^^^^^ |
| 14:54.36 | Zeeek | what about trillian? |
| 14:54.42 | Zeeek | oops |
| 14:54.49 | Zeeek | this is nice: http://blog.voipsupply.com/voip-commentary/women-in-voip-vibrant-vanderhorst-larson |
| 14:54.53 | hsv-al | stop now |
| 14:54.56 | hsv-al | reload chan_iax2.so |
| 14:54.58 | hsv-al | wtf, wrong win |
| 14:55.11 | hsv-al | one sec |
| 14:55.24 | Zeeek | stop NOW. I said STOP NOW. I said, son, stop now... |
| 14:55.39 | Zeeek | <foghorn leghorn> |
| 14:56.29 | hsv-al | heh, putting those allow/disallow statements in [general] section of iax.conf |
| 14:56.32 | hsv-al | gave some weird errors |
| 14:56.45 | ThoMe | Hat wer dazu nen TIP: http://paste.keks.be/41 <Geht um Asterisk+ISDNKARTE und daran *normale* Telefonanlage (german) |
| 14:57.05 | Zeeek | hsv-al [TK]D-Fender will be happy to guess the errors |
| 14:57.27 | Zeeek | Ich spreche kein Deutsch |
| 14:57.32 | hsv-al | http://pastebin.com/m4293c1d1 |
| 14:57.35 | ThoMe | Zeeek: Ist ok |
| 14:57.43 | Zeeek | ya |
| 14:58.02 | hsv-al | if i remove those allow/disallow statements, it'll work again |
| 14:58.06 | hsv-al | in gsm |
| 14:58.27 | Zeeek | hsv-al we'd have to see your whole conf file |
| 14:58.44 | hsv-al | just the iax.conf? |
| 14:59.13 | Zeeek | well if you are using iax, that would be logical |
| 14:59.24 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 14:59.24 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:59.26 | Zeeek | just the parts that apply |
| 14:59.42 | Zeeek | are you all gonna be naked? |
| 14:59.48 | hsv-al | http://pastebin.com/m232e87c5 |
| 14:59.52 | Zeeek | wrong window again :) |
| 15:00.11 | Zeeek | put the disallow FIRST |
| 15:00.17 | Zeeek | disallow=all |
| 15:00.27 | Zeeek | then allow, one per line other codecs |
| 15:00.42 | Zeeek | disallow=all |
| 15:00.45 | Zeeek | allow=ulaw |
| 15:00.50 | Zeeek | allow=alaw |
| 15:00.52 | Zeeek | ETC |
| 15:01.19 | esaym | how does asterisk connect to it's console? |
| 15:01.25 | hsv-al | -r |
| 15:01.26 | esaym | I can't connect to the console |
| 15:01.28 | esaym | yea |
| 15:01.47 | esaym | I have 2 asterisk servers on the same box |
| 15:01.49 | hsv-al | zeek, dig doesnt allow gsm it seems |
| 15:01.52 | esaym | I am wondering if that is why? |
| 15:01.58 | hsv-al | <PROTECTED> |
| 15:01.58 | hsv-al | [Jun 13 09:01:29] WARNING[20544]: chan_iax2.c:7736 socket_process: Call rejected by 216.207.245.8: Unable to negotiate codec |
| 15:01.58 | hsv-al | <PROTECTED> |
| 15:01.58 | hsv-al | <PROTECTED> |
| 15:02.22 | hsv-al | ulaw rather |
| 15:02.28 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 15:03.05 | fogo | anyone running 1.4.20.1 willing to test for a bug in chanspy? |
| 15:08.01 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:08.03 | Zeeek | I would like to formally state that the http://VoipUsersConference.org loves Digium(tm) and willbe talking about this in one hour |
| 15:10.51 | SuPrSluG | ~phones |
| 15:10.51 | jbot | hmm... phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 15:11.01 | hsv-al | my god |
| 15:11.03 | hsv-al | http://forums.digium.com/viewforum.php?f=6&sid=50df3b0416632109a04b17826fd10a23 |
| 15:11.16 | hsv-al | look at those posts, heh |
| 15:11.26 | Qwell | hsv-al: are you offering to moderate it? :p |
| 15:11.36 | hsv-al | heh |
| 15:11.39 | Zeeek | Grandstream phones work great |
| 15:11.45 | Qwell | but yeah...need to do something about that |
| 15:12.27 | Zeeek | update: customer called requesting we re-date a post on his news page to remove the Friday 13th reference. |
| 15:12.40 | Zeeek | MUAHAHAHA |
| 15:13.22 | Qwell | Zeeek: what is the topic for today? |
| 15:13.30 | Zeeek | The new Asterisk.... |
| 15:13.36 | russellb | Qwell: sales pitch for a new asterisk appliance |
| 15:13.38 | Zeeek | appliance from Amanda |
| 15:13.39 | Qwell | pass |
| 15:13.44 | russellb | Qwell: same :-p |
| 15:13.47 | Zeeek | NOO PLEASE |
| 15:13.52 | JT | what a loser, i think i should make press releases on friday the 13th |
| 15:14.03 | Zeeek | we may have serious questions about 1.6!!! |
| 15:14.09 | russellb | i'll call in if you do |
| 15:14.12 | Zeeek | ty |
| 15:14.19 | Zeeek | bows |
| 15:14.28 | Qwell | "Can TAA's appliance run 1.6?" |
| 15:14.34 | Qwell | is going to be the question >.< |
| 15:14.36 | JT | headdesks |
| 15:14.40 | Zeeek | are you kidding? It's running 2.0 |
| 15:14.51 | JT | just discovered a customer of mine in my co-location space |
| 15:15.04 | Zeeek | which by the way has a few bugs :) |
| 15:15.07 | JT | had their smtp server set to relay for /8 off their ip |
| 15:15.14 | JT | no wonder it was sending spam |
| 15:16.01 | davevg-btwtech | qwell, i'd moderate it for blatent spam posts if no one wants to delete them. i reported them to webmaster and Shelly responded rt84304 but she never handled them |
| 15:16.16 | Zeeek | with my new Siemens DECT/SIP phone I barely need any appliance or server at all |
| 15:16.48 | Zeeek | it connects to one POTS and six SIP servers |
| 15:17.12 | Zeeek | Only thing that bothers me is the missing Allison SMith voice! |
| 15:18.24 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:19.40 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
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| 15:20.02 | *** part/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
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| 15:25.01 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 15:26.05 | Zeeek | got it? |
| 15:26.20 | *** join/#asterisk kannan (n=kannan@123.201.60.110) |
| 15:27.00 | kannan | hello all |
| 15:28.26 | *** join/#asterisk whye (n=whye@unaffiliated/whye) |
| 15:31.39 | Zeeek | http://VoipUsersConference.org pre-conference is live |
| 15:32.15 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:36.00 | Zeeek | music playing |
| 15:38.08 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 15:39.17 | kannan | hello, i configured 2 plycom IP 301 phones with Asterisk. The calls are very clear. However, i have 2 problems. (1) If i set call fwd in the phone, Asterisk sends calls to the phone to unavl VM box and also (2) If I pich the handset and dial a 11 digit number it takes the first 10 , straightaway, which matches another extension pattern of 10 digits. However, if i dial the number on the keypad and then use the dial key on the phone all is fine. This is |
| 15:39.17 | kannan | not a prob on other phones like cisco 7960 or grandstream or x-lite on the same pbx. Any ideas how to resolve these two issues? |
| 15:39.47 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 15:40.27 | *** join/#asterisk javb (n=javb@190.166.114.161) |
| 15:40.49 | [TK]D-Fender | kannan: Go fix the dialplan on the phone itself. |
| 15:41.07 | [TK]D-Fender | kannan: And pastebin a sample call with this forwarding issue including sip debug. |
| 15:41.12 | javb | in the dial cmd, in asterisk, Dial(SIP/XXX,40) --> "40" means the number of seconds it will be ringing, if i just dont put anything there, what will be ? |
| 15:41.36 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-149.hsi.ish.de) |
| 15:41.39 | [TK]D-Fender | javb: Until the originating channel hangs up or the other side tells * to stop (which might be never) |
| 15:41.57 | javb | Perfect, so it will be forever. |
| 15:42.02 | javb | hehehe |
| 15:42.23 | kannan | [TK]D-Fender , ok thanks for that. I will do it some other time, as I cannot acces the box now, due to a network problem there |
| 15:42.34 | Zeeek | randulo |
| 15:48.19 | *** join/#asterisk makkksimal (n=makkksim@e177215112.adsl.alicedsl.de) |
| 15:48.44 | jack_sparo | ~book |
| 15:48.45 | 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 |
| 15:49.03 | jack_sparo | ~buybook |
| 15:49.03 | jbot | You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 15:49.46 | lmadsen | and sales are down |
| 15:50.28 | Corydon76-dig | and if sales aren't back up, there's a risk that the 3rd edition will continue to be delayed |
| 15:51.06 | coppice | huh? why would he produce a third edition if the second is earning well? |
| 15:51.40 | Corydon76-dig | coppice: you mean, not earning well? |
| 15:51.51 | De_Mon | why produce a sequal to a book that isn't doing well |
| 15:52.05 | De_Mon | thats what the publishers are asking |
| 15:52.05 | coppice | no, I mean its only falling sales that will inspire a refresh |
| 15:52.24 | Corydon76-dig | Because information changes, as Asterisk progresses |
| 15:53.06 | Corydon76-dig | coppice: if only the authors made enough off the book to survive independently... |
| 15:53.59 | Corydon76-dig | I personally think it was a mistake to have the PDF released concurrently with the book. |
| 15:54.30 | Corydon76-dig | but not really my place to say, as I'm not one of the 3 authors |
| 15:54.46 | coppice | Is there a cheap student edition available in china? that's how I generally get books these days :-) |
| 15:55.21 | *** join/#asterisk JoZu (i=asdf@84.120.223.83.dyn.user.ono.com) |
| 15:55.24 | Corydon76-dig | Why would China publish a book that available in PDF form for free? |
| 15:55.42 | raytruz` | sales are probably down due to the rising cost of gas |
| 15:55.44 | coppice | dunno. ask pearsons |
| 15:55.51 | raytruz` | Just like everything else :-) |
| 15:55.58 | mog | raytruz`, hilarious |
| 15:56.35 | Zeeek | http://VoipUsersConference.org is about to begin. Please join us |
| 15:56.51 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
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| 15:59.53 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.120) |
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| 16:06.27 | *** join/#asterisk Marquel (n=Marquel@port-232.pppoe.wtnet.de) |
| 16:06.31 | Marquel | morning... |
| 16:06.42 | raytruz` | o/ |
| 16:07.55 | outtolunc | morn'n |
| 16:08.18 | Marquel | how do i keep Dial() from failing b/c of a network error w/ one of the called targets while all other targets don't fail? |
| 16:08.24 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 16:08.44 | jameswf-home | anyone seen Asterisk rash caused by a pri hangup? |
| 16:08.54 | jameswf-home | s/rash/crash/ |
| 16:10.33 | *** join/#asterisk sp00kz (i=ilubj00@our.government.is.in.the.dark.bz) |
| 16:11.04 | sp00kz | Anyone know why I might be getting an FTP Error on phone boot trying to grab sip.ld? Error is: Response: 426 Data connection: Illegal seek. |
| 16:11.13 | sp00kz | Polycom 650 |
| 16:11.17 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:11.44 | sp00kz | soundpoint ip 4000, and 550 |
| 16:12.02 | coppice | If you have an Asterisk rash, I think you should consult a doctor |
| 16:12.23 | jameswf-home | http://pastebin.com/m678d12b2 <<~~ message on the asterisk tombstone b4 it dies |
| 16:14.39 | *** join/#asterisk MrNaz (n=naz@ppp59-167-157-26.lns4.mel6.internode.on.net) |
| 16:14.54 | makkksimal | coppice: :D |
| 16:15.12 | makkksimal | i do get it sometimes.. |
| 16:16.41 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 16:17.14 | jameswf-home | is his own best friend |
| 16:18.28 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
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| 16:30.17 | javb | when i call a sip agent, inside my network, it will ring for 30 seconds, and then voicemail, perfect, but if call outside network, that extn, using IAX, it will ring and after the same quantity of seconds, it will spawn, and busy tone. any idea? |
| 16:30.24 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 16:31.46 | seanbright | anyone know of and/or recommend good wireless sip devices? our sales managers want to have something they can just clip to their hips and walk around the floor with instead of carrying a tablet pc with a soft phone on it. |
| 16:32.01 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:32.03 | Strom | javb: pastebin the section of extensions.conf that's handling your outbound calls |
| 16:32.17 | fogo | seanbright: after looking all over on voip-info we bought some Linksys WIP330s |
| 16:32.18 | Strom | seanbright: IIRC, UTStarcom makes some |
| 16:32.31 | fogo | seanbright: sound quality is great, and they worked out of the box |
| 16:32.42 | seanbright | fogo, Strom: thanks. |
| 16:33.11 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:33.16 | *** join/#asterisk mihr (n=albert@netsys.bts.corp.amdatex.net) |
| 16:34.28 | *** join/#asterisk xnosx (n=xnosx@212.145.55.118) |
| 16:34.45 | seanbright | fogo: do the 330s support headsets? |
| 16:34.56 | seanbright | heh... they call them 'iphone's |
| 16:35.17 | Strom | seanbright: linksys owned that trademark long before apple ever used it :) |
| 16:35.22 | seanbright | ahh |
| 16:35.40 | javb | iax.conf, of the originating call, --> http://pastebin.com/m7b4b9024 ; dialplan of the receiving call: http://pastebin.com/m6af13772 Strom |
| 16:36.01 | *** join/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 16:36.59 | Strom | javb: so the problem occurs when you're calling /from/ the outside, or is this occurring when you're calling /to/ the outside? |
| 16:37.00 | seanbright | standard headphone jack... no bluetooth. |
| 16:37.44 | javb | from the outside |
| 16:37.50 | javb | from-pstn |
| 16:37.58 | Titanous | I'm trying to use realtimje, and I've got ODBC setup to connect, but 'odbc show' doesn't do anything |
| 16:38.11 | Titanous | s/realtimje/realtime/ |
| 16:38.28 | mvanbaak | Titanous: odbc show all |
| 16:38.34 | javb | first one, dials using iax, then the receiving, receive it very well, execute the first line, but the, if nobody picks it up, it should go to the second, which is voicemail, just busytone |
| 16:38.46 | Titanous | mvanbaak: nothing |
| 16:39.08 | Strom | javb: pastebin the CLI output of one of these calls -- set verbosity to 10 first |
| 16:39.23 | mvanbaak | you have a correct res_odbc.conf ? |
| 16:41.13 | Titanous | mvanbaak: http://pastebin.com/d6dd14655 |
| 16:42.20 | javb | strom, there u have: http://pastebin.com/m5d51da67 |
| 16:42.45 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 16:42.52 | sp00kz | Anyone know why I might be getting an GSSFTP Error on any Polycom phone when it boots trying to grab sip.ld? Error is: Response: 426 Data connection: Illegal seek. |
| 16:43.25 | Strom | javb: the extension that the CLI output is referencing doesn't appear in your pastebins. Will you please pastebin your ENTIRE extensions.conf? |
| 16:43.34 | javb | Strom, if i sustitute, the dial, with the voicemail app, it will get me rigth to the voicemail. |
| 16:43.42 | *** join/#asterisk gaetronik (n=gaetan@eficaz2.manquehue.net) |
| 16:43.47 | javb | Strom, it DOES appear |
| 16:43.54 | gaetronik | Hi all |
| 16:44.00 | Strom | == Spawn extension (to-everywhere-with-pass, 3211, 1) exited non-zero on 'IAX2/delgado-4585' |
| 16:44.05 | javb | Exten => _32[01456789]X,1,Dial(SIP/${EXTEN},30) <---- THERE IS 3211 |
| 16:44.18 | Strom | javb: I don't see a context called "to-everywhere-with-pass" anywhere here |
| 16:44.35 | Strom | and don't yell at me if you want me to help. |
| 16:44.40 | mvanbaak | Titanous: that's not a correct /etc/asterisk/res_odbc.conf |
| 16:45.02 | javb | thats a context which has "include=>from-internal" |
| 16:45.14 | gaetronik | reseller question |
| 16:45.21 | Strom | javb: pastebin the entire extensions.conf so I can help you, please. |
| 16:45.21 | javb | the iax2 peer calling is in the context "to-everywhere-with-pass" |
| 16:45.34 | gaetronik | between voip-supply and telphonydepot which one? |
| 16:45.41 | Strom | gaetronik: telephonydepot |
| 16:46.14 | Titanous | mvanbaak: oops |
| 16:46.18 | mvanbaak | Titanous: this is mine: |
| 16:46.22 | mvanbaak | http://pastebin.com/d68f77917 |
| 16:46.26 | Titanous | mvanbaak: got it working |
| 16:47.08 | mvanbaak | :) |
| 16:47.13 | javb | Strom --> http://pastebin.com/m32a57bba |
| 16:48.10 | gaetronik | Strom, why? |
| 16:48.47 | gaetronik | whatever thanks |
| 16:48.54 | Strom | javb: this doesn't appear to be the full file either. Will you please paste the ENTIRE extensions.conf from line 1 to the last line? |
| 16:48.58 | Strom | gaetronik: jeez, so patient |
| 16:49.18 | Strom | gaetronik: they're cheap and reliable and i've been happy with their service |
| 16:49.25 | gaetronik | ok thanks |
| 16:49.37 | gaetronik | i think i said anything wrong |
| 16:50.07 | gaetronik | thanks a lot Strom |
| 16:50.07 | javb | Strom, http://pastebin.com/m5cc46f79 |
| 16:52.10 | Strom | javb: ok -- now run "extensions reload" at the asterisk CLI to ensure we're debugging the code that asterisk has in memory, run another call, and pastebin its CLI output |
| 16:52.14 | Strom | set verbosity to 10 |
| 16:52.49 | *** join/#asterisk JHilgeman (n=jh@c-69-143-43-248.hsd1.va.comcast.net) |
| 16:53.28 | sp00kz | activate photon torpedos |
| 16:54.47 | jaytee | make sure you check the interlink channels for distortion on the Heisenberg Uncertainty Compensator before beaming up |
| 16:55.13 | javb | Strom, extensions reload ---> http://pastebin.com/m34f7badc --- Call again ... http://pastebin.com/m1e835c92 |
| 16:55.24 | coppice | You'll need the Interocitor working, too. |
| 16:55.31 | Strom | i didnt need the output of extensions reload |
| 16:55.32 | jaytee | hahahaha |
| 16:55.40 | jaytee | I remember that movie |
| 16:56.25 | fogo | Has anyone used chan-sccp to get Cisco SCCP devices working? |
| 16:58.47 | Strom | javb: what are you calling from? |
| 16:59.13 | *** join/#asterisk BBHoss (n=hoss@c-68-62-175-86.hsd1.al.comcast.net) |
| 16:59.24 | javb | pstn, to a pstn-gate, which has an iax2 trunk with the asterisk. |
| 16:59.39 | *** join/#asterisk deeperror (n=deeperro@adsl-76-226-146-19.dsl.sfldmi.sbcglobal.net) |
| 17:00.47 | Strom | javb: try this just for debugging purposes -- add a priority to your 3211 extension so it runs Answer() before calling the sip phone |
| 17:01.00 | Strom | leave the voicemail and the dial() as they are |
| 17:01.21 | javb | Strom, of course, and set the Dial to "n" priotity/ |
| 17:01.22 | javb | ? |
| 17:01.28 | Strom | yes |
| 17:04.05 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 17:06.15 | javb | Strom, IT WORKED, with the Answer, before ... any idea why? |
| 17:06.23 | sp00kz | Strom = Digium Tier 1 Tech Support * 1000 |
| 17:06.45 | Strom | javb: as I thought -- your PSTN gateway is cutting your calls off because you're not answering quickly enough |
| 17:07.16 | Strom | if it's your PSTN gateway, I'd reconfigure it...otherwise, find a new ITSP |
| 17:07.29 | Strom | sp00kz: heh, thanks |
| 17:07.31 | *** join/#asterisk ManxPower (n=manxpowe@111.sub-70-223-146.myvzw.com) |
| 17:07.39 | javb | Strom, what do you mean with reconfigure |
| 17:08.04 | Strom | well, configure it not to have a limit on calls that haven't answered yet |
| 17:08.20 | javb | Strom, im sorry didnt get that. |
| 17:08.46 | javb | Strom, it doesnt have a limit, when it does DIAL, it has no time for it. |
| 17:08.58 | Strom | it's your PSTN gateway? |
| 17:10.16 | *** join/#asterisk nirz (n=nir@bzq-79-181-149-183.red.bezeqint.net) |
| 17:10.51 | javb | yes |
| 17:10.53 | javb | Strom |
| 17:11.05 | javb | it just take the call from the PSTN and give to me via IAX2 |
| 17:11.09 | javb | JUST THAT. |
| 17:11.28 | javb | with Dial(IAX2/"my pbx"/{EXTEN}) |
| 17:11.30 | Strom | javb: ok...pastebin CLI output from that box on the same call |
| 17:15.58 | truent | i got asterisknow installed on a laptop.. but its more just because it was a painless install with no bulk.. i configured the dialplan and sip etc manually.. but how can i record voicemenu's etc through the phone? any direction would be helpful |
| 17:16.16 | Strom | truent: #asterisknow |
| 17:16.21 | truent | oh geez |
| 17:16.22 | mog | isnt that an option in the gui truent |
| 17:16.23 | ManxPower | truent: Please use the correct channel |
| 17:16.28 | truent | screw the gui |
| 17:16.28 | mog | Strom! |
| 17:16.31 | *** join/#asterisk whye (n=whye@unaffiliated/whye) |
| 17:16.32 | mog | how you been |
| 17:16.36 | truent | im not using the gui |
| 17:16.37 | Strom | mog: been good! |
| 17:16.48 | ManxPower | truent: in order to "screw the gui" simply delete all the config files in /etc/asterisk |
| 17:16.49 | russellb | truent: you would use the Record() application |
| 17:16.56 | mog | truent show application record |
| 17:16.57 | truent | thank you |
| 17:16.59 | truent | Record |
| 17:17.15 | truent | why is that no where in the Future of telephony book ? ;p |
| 17:17.22 | truent | im sure its in the appendix |
| 17:17.24 | russellb | it's in the application appendix |
| 17:17.27 | mog | im sure its i nthere |
| 17:17.32 | truent | but not a proper mention |
| 17:17.42 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 17:17.45 | ManxPower | truent: "core show applications" in the CLI shows you all the applications install for YOUR Asterisk |
| 17:17.50 | truent | gotcha |
| 17:17.58 | truent | thanks for the actual help |
| 17:18.06 | ManxPower | Best of luck trying to edit your config files. |
| 17:18.07 | truent | #asterisknow references arent very useful :P |
| 17:18.10 | javb | Strom, http://pastebin.com/m126420d5 |
| 17:18.16 | truent | manxpower, i just started fresh |
| 17:18.16 | ManxPower | truent: then why are you using it if the support is so bad? |
| 17:18.26 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 17:18.33 | jm|laptop | . |
| 17:18.39 | ManxPower | We are NOT 2nd tier support for #asterisknow |
| 17:18.48 | sp00kz | Anyone know why I might be getting an GSSFTP Error on any Polycom phone when it boots trying to grab sip.ld? Error is: Response: 426 Data connection: Illegal seek. |
| 17:19.05 | ManxPower | sp00kz: what FTP server? |
| 17:19.31 | truent | i just wanted a quick install with no gui etc to mess around with, twas a pretty good option for now |
| 17:19.34 | sp00kz | GSSFtp, the one built into xinetd |
| 17:19.42 | Strom | javb: what kind of entrance facilities is this gateway using? |
| 17:19.43 | sp00kz | but have tried vsftp too |
| 17:19.48 | truent | i'll do something different when i actually deploy |
| 17:19.53 | truent | thanks for the help |
| 17:19.56 | truent | brb |
| 17:19.56 | *** join/#asterisk JHilgeman (n=jh@c-69-143-43-248.hsd1.va.comcast.net) |
| 17:19.58 | javb | Strom, T1 |
| 17:19.59 | ManxPower | sp00kz: looks like it does not support the features Polycom requires. I have used vsftpd with Polycom, I think. |
| 17:20.12 | Strom | javb: channelized T1 or ISDN PRI? |
| 17:20.21 | javb | Strom: channelized |
| 17:20.36 | sp00kz | ManxPower: well it sorta just started.... and gssftp does do everything, i use it on ~40 pbx's :\ |
| 17:20.43 | Strom | javb: looks like your provider is cutting you off, then |
| 17:21.07 | sp00kz | ManxPower: just one is having issues, stopped firewall, replaced ftp, but the poly phones keep doing that when trying to dl binaries |
| 17:21.13 | JHilgeman | can someone help me with a problem? I'm having trouble dialing out - the dialplan seems to be there and executing a macro called trunkdial, but when it tries to Dial, it gets a hangup request from the channel with a cause of 1. I have no idea what that means or what to do. |
| 17:21.50 | Strom | JHilgeman: you want trixbox/freepbx support as pointed out in the topic |
| 17:22.19 | deeperror | deano? |
| 17:22.40 | JHilgeman | not using freepbx/trixbox, tho |
| 17:22.57 | JHilgeman | using AsteriskNOW, but I asked the question in those channels, and nobody's really answering |
| 17:23.03 | Strom | er, yeah |
| 17:23.29 | JHilgeman | hoping someone can help me from here |
| 17:23.31 | Strom | well, still, this isnt exactly the asterisknow support channel either |
| 17:23.49 | JHilgeman | yeah but i figure the base is still the same... |
| 17:24.06 | JHilgeman | it's still executing the same basic asterisk commands |
| 17:24.23 | Strom | this is much like asking a Nortel technician to fix a Cisco system because they're both phone systems |
| 17:24.46 | JHilgeman | i was under the impression that *NOW was just Asterisk with a GUI on top of it |
| 17:24.52 | [TK]D-Fender | JHilgeman: paste the dial command that it called |
| 17:25.42 | javb | problem solved, the answer fun in the pstn gate way |
| 17:25.57 | Strom | javb: that's not a solution |
| 17:26.07 | Strom | JHilgeman: is it an ISDN PRI circuit? |
| 17:26.15 | javb | Stom,what is it then |
| 17:26.33 | JHilgeman | Executing [s@macro-trunkdial:2] Dial("SIP/1109-00751950", "Zap/g2/17034539120") in new stack |
| 17:26.43 | JHilgeman | It's a T1 PRI |
| 17:26.55 | Strom | JHilgeman: you should know your Q.931 cause codes |
| 17:27.01 | Strom | cause code 1 means unassigned number |
| 17:27.22 | Strom | ~itu |
| 17:27.23 | jbot | methinks itu is the International Telecommunication Union. Current versions of ITU-T recommendations (Q,931, T.38, V.32, et cetera) are available for free in PDF format from their website: http://www.itu.int/rec/T-REC/e |
| 17:27.44 | Strom | javb: find a telco that doesn't blow? |
| 17:27.50 | [TK]D-Fender | JHilgeman: ok, PASTEBIN your zapata.conf , zaptel.conf, and the output of "pri debug span 1" |
| 17:27.52 | [TK]D-Fender | ~pb |
| 17:27.53 | 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 |
| 17:27.54 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 17:28.45 | outtolunc | that is assuming his 'g2' is span 1 <G> |
| 17:28.50 | javb | Strom, ok... thanks for you help dude . |
| 17:32.07 | *** join/#asterisk shido6 (n=shido6@74-130-224-188.dhcp.insightbb.com) |
| 17:32.33 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 17:32.39 | JHilgeman | Ok - here's the pastebin url |
| 17:32.53 | JHilgeman | http://pastebin.com/d4696e3a7 |
| 17:33.06 | JHilgeman | seems like the config files were separated out a bit |
| 17:33.19 | JHilgeman | so i put in a couple that seemed relevant (users.conf and extensions.conf) |
| 17:33.47 | Strom | JHilgeman: did you catch what I said about ISDN cause codes? |
| 17:33.50 | JHilgeman | I manually added the two lines to the zapata.conf file in an attempt to fix it |
| 17:33.59 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 17:34.15 | JHilgeman | yes - outtolunc sent me a page about cause codes - I'm not sure I understand how it knows that it is unallocated, though? |
| 17:34.36 | Strom | JHilgeman: the PSTN is telling you that the number is unassigned |
| 17:34.38 | JHilgeman | does that mean it's trying to dial internally and it can't find it there? |
| 17:34.43 | Strom | no no no. |
| 17:34.59 | JHilgeman | i've tried several different phone numbers that are all valid |
| 17:35.18 | JHilgeman | they all have the same issue |
| 17:35.25 | Strom | which area code are you located in? |
| 17:35.51 | JHilgeman | the T1 is a 571 area code |
| 17:36.18 | Strom | ok, thats an overlay complex with 703 |
| 17:36.18 | [TK]D-Fender | JHilgeman: pastebin "zap show channels" and "span = 1,0,0,esf,b8zs" should be 1,1,0 if you're connected to the telco |
| 17:36.41 | Strom | is the 703 number in your example local to the rate center your PRI is in? |
| 17:36.44 | JHilgeman | I've tried dialing other 571 numbers w/o an area code (i.e. 91231234), and other completely different area codes (919491231234) |
| 17:37.02 | [TK]D-Fender | JHilgeman: and indeed : Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] <- the # you dialed isn't valid. |
| 17:37.48 | JHilgeman | hmm, i believe so |
| 17:38.04 | JHilgeman | i'm still new to the general area, but I believe it's in the same county |
| 17:38.05 | Strom | JHilgeman: the telco wants ten digits on local calls and eleven digits on toll calls |
| 17:38.35 | Strom | http://nanpa.com/nas/public/npa_query_step2.do;nanpaid=nG88LSvW6gzL7T3gmv9h2vvKz4ZpSJjgrppYSym1T5MDRh2m3vKC!-287911660?method=displayNpa |
| 17:38.35 | JHilgeman | Fender - so span = 1,1,0,esf,b8zs? |
| 17:40.29 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 17:40.37 | [TK]D-Fender | JHilgeman: Yes |
| 17:40.49 | [TK]D-Fender | JHilgeman: You should be taking timing from your telco, not the other way around |
| 17:41.19 | JHilgeman | Pastebin from zap show channels: http://pastebin.com/d6a6f549a |
| 17:41.48 | JHilgeman | making that change to the span - just a sec |
| 17:42.09 | [TK]D-Fender | JHilgeman: Ok, well everything else checks out, it is indeed an improper # you are passing, and the telco is telling you directly taht ti doesn't like it. |
| 17:42.14 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 17:42.22 | *** join/#asterisk whye_ (n=whye@unaffiliated/whye) |
| 17:42.28 | [TK]D-Fender | JHilgeman: And you'll have to reload chan_zap, or restart * completely for the timing change. |
| 17:42.45 | JHilgeman | okay - i'll reboot it |
| 17:42.57 | *** join/#asterisk MrNaz (n=naz@ppp59-167-130-84.lns3.mel6.internode.on.net) |
| 17:43.44 | JHilgeman | Strom - regarding the digits - I've tried dialing a number out in California using any combination i could think of |
| 17:43.54 | JHilgeman | none of the combinations work |
| 17:44.20 | Strom | was it an assigned number? |
| 17:44.32 | JHilgeman | cell phone # |
| 17:44.39 | JHilgeman | yeah - it's active |
| 17:44.43 | Strom | ok |
| 17:44.45 | JHilgeman | i can call it with any other phone and it works |
| 17:45.00 | JHilgeman | (phone not connected to this box, of course) |
| 17:45.53 | JHilgeman | could the timing be causing a problem with the number getting passed to the telco properly? |
| 17:45.58 | Strom | perhaps |
| 17:46.03 | Strom | fix that and try again |
| 17:46.20 | JHilgeman | k - still waiting for it to finish booting |
| 17:46.27 | Strom | ...booting? |
| 17:46.36 | Strom | reads up |
| 17:46.40 | Strom | um, ok |
| 17:46.48 | outtolunc | why don't you try originating a call without using the dialplan? |
| 17:46.48 | fucstik | i am getting this error when i configure asterisk-1.4.21 configure: error: *** termcap support not found |
| 17:46.57 | Qwell | fucstik: libncurses5-dev |
| 17:47.04 | fucstik | thanks |
| 17:47.16 | *** join/#asterisk tmjb (n=tmjb@mail.bigapple.co.yu) |
| 17:48.05 | tmjb | hello do i need "Cisco CallManager Express License For Single" for asterisk |
| 17:48.17 | Qwell | tmjb: elaborate |
| 17:48.33 | Qwell | why would you need a license for a proprietary PBX, when you're using an open source one? |
| 17:48.54 | JHilgeman | problem's still there |
| 17:49.21 | kannan | tmjb , if you get the Cisco phone with SIP firmware, it work on Asterisk |
| 17:49.27 | JHilgeman | k - let me try the originating sugg. from out |
| 17:51.29 | outtolunc | your zapata.conf looks rather bare |
| 17:51.48 | ManxPower | You would, of course, need a license for the SIP firmware for the phone. |
| 17:56.47 | s0ck | anyone use snom360s? |
| 17:56.57 | s0ck | how do you get into 'admin' mode on em |
| 17:58.31 | jaytee | I can't figure out the proper way to express something in * with the correct syntax. Basically what I'm trying to do in plain English terms is this: if the callerid on the incoming call is blank then assign it the value "Unavailable" otherwise if the callerid has a value other than null continue to the next priority in the context. Can anyone point me to a resource that would help me other than the book because it's function reference isn't enough |
| 17:58.31 | jaytee | for me to figure this out. |
| 18:00.10 | s0ck | ok got it, default pass is 0000 |
| 18:00.53 | Strom | jaytee: look at GotoIf() |
| 18:01.01 | [TK]D-Fender | jaytee: 52nd try's the charm.... |
| 18:01.35 | jaytee | Strom, so I have to do a conditional branch? I can't just test for null and set it on one line? |
| 18:01.49 | Strom | jaytee: correct |
| 18:01.53 | jaytee | [TK]D-Fender, more like the 62nd but whose counting? |
| 18:02.05 | [TK]D-Fender | jaytee: </irony> |
| 18:02.15 | Titanous | Do I have to contact Digium to change what maching my g729 license is installed? |
| 18:02.23 | Titanous | s/maching/machine/ |
| 18:02.25 | jaytee | yeah, irony is pretty ironic sometimes :-) |
| 18:02.30 | [TK]D-Fender | Titanous: yes. |
| 18:02.33 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 18:02.35 | Titanous | k |
| 18:04.32 | Marquel | nobody? |
| 18:04.55 | Strom | EVERYBODY! |
| 18:05.15 | Marquel | *g* |
| 18:05.50 | *** join/#asterisk JHilgeman (n=jh@c-69-143-43-248.hsd1.va.comcast.net) |
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| 18:06.05 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 18:06.07 | JHilgeman | the irc server keeps disconnecting me |
| 18:06.11 | JHilgeman | anyways |
| 18:06.40 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 18:06.48 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 18:06.54 | JHilgeman | Fender - after the server came back up, there weren't any available channels - doing a zap show channels resulted in nothing but a line with a channel called "pseudo" |
| 18:07.22 | Marquel | how do i keep Dial() from failing b/c of a network error w/ one of the called targets while all other targets don't fail? |
| 18:08.39 | Strom | Marquel: pastebin the CLI output of one such failed call |
| 18:08.56 | *** join/#asterisk jbroome (n=jbroome@unaffiliated/jbroome) |
| 18:09.23 | Marquel | Strom: which level of verbosity do you like? :) |
| 18:10.05 | Strom | 10 |
| 18:10.14 | tmjb | tnx guys very much Qwell and kannan my dealer just sended me this in the offer and alywas worked with Linksys phones which have SIP . |
| 18:10.55 | Marquel | Strom: okay, but it may cause me to disconnect b/c of a ping timeout - don't bother ;) |
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| 18:11.05 | *** part/#asterisk jbroome (n=jbroome@unaffiliated/jbroome) |
| 18:11.21 | Strom | Marquel: um, ok? |
| 18:12.01 | Marquel | Strom: my laptop is the sip-phone in question and i need to disconnect the wire ;) |
| 18:12.12 | JHilgeman | hmm |
| 18:12.18 | Strom | but why are you telling me "don't bother"? |
| 18:12.43 | Marquel | don't think i left w/o coming back ;) |
| 18:12.53 | Strom | that's not what "don't bother" means |
| 18:13.09 | Strom | perhaps you meant "don't worry" |
| 18:13.23 | Marquel | perhaps :) |
| 18:14.03 | zeeesh | how to troubleshoot if getting noice in voice.. i m receiving calls at E1 then coming to asterisk then sending these calls to my route via E1 ? |
| 18:14.16 | *** join/#asterisk dlynes (n=daniel@S01060016b683f535.ok.shawcable.net) |
| 18:14.48 | *** join/#asterisk kannan (n=kannan@123.201.60.110) |
| 18:15.38 | dlynes | Is there a way so that when asterisk decides it's going to interfere in teh call path that I can tell it not to interfere with the caller id information? It's overwriting my caller id number with the peer name of the connection between the two asterisk boxes; the caller id name information remains unchanged |
| 18:16.37 | Marquel | *narf* :( |
| 18:18.05 | Marquel | seems to be not so easy to create that behavior w/o long time disconnect :( |
| 18:18.57 | Strom | dlynes: that shouldnt be happening anyway |
| 18:19.19 | Strom | is this SIP? |
| 18:20.14 | jaytee | Strom, how should I properly nest the ISNULL function in a GotoIf to test if CALLERID is null or not? |
| 18:21.02 | Strom | ${ISNULL(${CALLERID(num)})} perhaps? |
| 18:21.09 | [TK]D-Fender | jaytee: Yes, and I told you tor formatting was wrong and that you weren't REFERENCING your functions properly |
| 18:21.20 | [TK]D-Fender | Strom : correct |
| 18:21.38 | Strom | do I win a prize now?! |
| 18:21.43 | *** join/#asterisk ipstatic (n=ipstatic@24.106.202.78) |
| 18:22.30 | jaytee | [TK]D-Fender, I know and rather than pester you with 18 to a 100 questions I kept trying to figure out what you told me but the book is scant on examples. |
| 18:22.48 | Marquel | Strom: i have a log notice from this morning saying "app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination)" - that doesn't help w/o the rest i guess? |
| 18:23.02 | ipstatic | hello all, Is there anyone who is using the dial plan to dynamically add agents (like the sample queues-with-callback-members.txt in the docs)? |
| 18:23.09 | Strom | Marquel: did you observe it actually failing? |
| 18:23.19 | Strom | Marquel: or are you just extrapolating from the logs? |
| 18:23.19 | [TK]D-Fender | jaytee: its full of the only thing you did wrong : forgetting how to get the VALUE of a function. |
| 18:23.33 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 18:24.51 | Marquel | Strom: at the time this message appeared i observed it failing. i was calling in, it rang a few seconds (the list of missed calls of those phones working (ZAP-channel) stating an unanswered incoming call at this time) but after two or three times of ringing asterisk hung up on me. |
| 18:25.17 | Strom | Marquel: that shouldnt happen. |
| 18:25.22 | jaytee | [TK]D-Fender, I swear I'll get better at this but doing this project is the most "coding" I've had to do in a decade and my neurons have a nice patina of rust all over them. |
| 18:25.42 | [TK]D-Fender | hands jaytee some CLR |
| 18:26.00 | jaytee | plus the coffee is kinda weak today :-) |
| 18:26.05 | Damin | Here is something I was asked today.. I thought this would be an easy answer, but I'm not finding the answer readily available.. |
| 18:26.22 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 18:26.51 | Damin | Let's say I define a queue and want it to always ring in the same order as listed in the queues.conf. What strategy should I use? I thought there was some sort of fixed order option, but I can't find it. |
| 18:27.24 | [TK]D-Fender | Damin: "roundrobin" |
| 18:28.08 | [TK]D-Fender | Damin: there was a varient in 1.2/1.4 for RRMEMORY which would do roundrobin and remember who answered last and pick up where that left off. |
| 18:28.29 | [TK]D-Fender | Damin: I think they merged the 2 though. So either wway, roundrobin is the best you'll get. |
| 18:28.31 | Marquel | Strom: precise config is: exten => s,1,Dial(SIP/laptop&ZAP/1&ZAP/2,60) - "SIP/laptop" is a softphone on my laptop and all works well as long as my laptop's connected to the lan (either direct or via vpn). but if my laptop's been disconnected for a while w/o logging off the softphone, inbound calls are failing. |
| 18:28.39 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 18:29.05 | Strom | Marquel: perhaps you should try adding "qualify=yes" to the laptop's entry in sip.conf |
| 18:29.15 | Damin | [TK]D-Fender: Thanks.. the Voip-Info page has some more detail on it, but it's not entirely clear.. |
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| 18:30.13 | *** part/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 18:30.32 | Marquel | Strom: which causes what? |
| 18:30.53 | JHilgeman | sorry for taking so long, but I'm back - still the same problem, even with the corrected timing. I tried to manually originate a couple calls from the CLI using different numbers, both local and long-distance, and i still get a cause code of 1 |
| 18:31.07 | Strom | JHilgeman: call your telco. |
| 18:31.08 | JHilgeman | could the T1 router be issuing that cause code? |
| 18:31.22 | Strom | JHilgeman: call your telco. |
| 18:31.36 | JHilgeman | alright - i'll give that a shot. |
| 18:31.40 | Strom | JHilgeman: T1 is just the transport. |
| 18:31.49 | Strom | T1 has nothing to do with cause codes |
| 18:31.53 | Strom | that's ISDN |
| 18:37.17 | Marquel | Strom: (okay, read the docs, sorry) - so if my laptop is regarded as unreachable Dial() will not fail b/c it doesn't try SIP then? |
| 18:37.39 | Strom | yep |
| 18:40.20 | Marquel | that'll solve it then, thx :) - i'll try on sunday, there'll be enough time of disconnect then. how often is that ping sent? |
| 18:40.35 | Strom | every few seconds? |
| 18:41.27 | Marquel | good enough. |
| 18:42.40 | Marquel | and the default timeout for asterisk-1.2 is? |
| 18:44.59 | Strom | for qualify? |
| 18:45.05 | Marquel | yep |
| 18:46.11 | jaytee | ~pb |
| 18:46.11 | 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 |
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| 18:47.47 | JHilgeman | alright |
| 18:47.48 | jaytee | Strom, [TK]D-Fender ? how about this? http://rafb.net/p/cVCz2m22.html |
| 18:48.03 | JHilgeman | apparently, the telco hadn't enabled outbound calling (smacking forehead) |
| 18:48.11 | JHilgeman | they turned it up and all is fine now |
| 18:48.12 | Strom | JHilgeman: wow. |
| 18:48.26 | Strom | jaytee: no |
| 18:48.43 | JHilgeman | I appreciate the help from all of you guys |
| 18:48.57 | JHilgeman | sorry that it was just me being a dumbass |
| 18:49.07 | [TK]D-Fender | jaytee: Looks valid |
| 18:49.40 | JHilgeman | ciao |
| 18:49.53 | [TK]D-Fender | JHilgeman: No, not a dumbass, just unaware that your telco felt like holding stuff back for no good reason. |
| 18:50.00 | Strom | too late !!! |
| 18:50.09 | [TK]D-Fender | Strom : oh well |
| 18:52.01 | *** join/#asterisk yxa (n=lonari@bb116-14-242-160.singnet.com.sg) |
| 18:52.31 | yxa | any knows how long i can wire a FXS module to a analog phone? |
| 18:52.35 | jaytee | Strom, [TK]D-Fender thinks it looks valid, why do you think it's not? |
| 18:52.53 | *** join/#asterisk deStone (n=deStone@unaffiliated/destone) |
| 18:52.58 | Strom | because you're not setting caller ID correctly on the next line |
| 18:53.37 | yxa | would 1km or 3280ft be a problem? |
| 18:53.50 | keith4 | yxa: how many phones attached to it? |
| 18:53.52 | Qwell | yxa: that far, probably.. |
| 18:53.55 | Qwell | that's...long |
| 18:53.59 | keith4 | holy crap. 1km? |
| 18:54.00 | yxa | only 1 phone |
| 18:54.57 | Qwell | I don't think 1km cables even exist |
| 18:55.14 | Qwell | Strom: ? |
| 18:55.32 | keith4 | yxa: that long of a distance screams SIP to me |
| 18:55.57 | Strom | Qwell: of course such cable exists. what do you think outside plant is? |
| 18:56.09 | Qwell | lots and lots of splicing? :p |
| 18:56.27 | yxa | whats the furthest you guys have tried? |
| 18:56.50 | Strom | well, yeah, but i'm assuming yxa means a total loop length of 1km |
| 18:56.59 | Strom | yxa: why such a long analog loop? |
| 18:57.02 | Qwell | still, that's really long |
| 18:57.03 | *** join/#asterisk fogo (n=fogo@72.8.104.15) |
| 18:57.20 | jaytee | Strom, is it the (all) that's wrong or the quotes for the string value "Unknown"? |
| 18:57.22 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 18:57.25 | sp00kz | probably too long |
| 18:57.36 | yxa | just a theoretical question as of now |
| 18:57.38 | Strom | jaytee: (all( |
| 18:57.43 | Strom | s/(/)/ |
| 18:57.46 | Strom | hah |
| 18:57.49 | keith4 | won't you have impedance problems with a 1km run? |
| 18:57.49 | Strom | fails |
| 18:58.09 | yxa | yeah would imagine the voltage would drop significantly |
| 18:58.23 | keith4 | not to mention weird signal attenuation |
| 18:58.32 | Strom | you all seem to be forgetting that analog telephone sets are designed to compensate for this |
| 18:58.34 | keith4 | like, the high freq dropping off more than low freq |
| 18:59.00 | sp00kz | yxa: I have successfully run a phone cable for ~1000ft |
| 18:59.29 | jaytee | Strom, ok. I wasn't sure since when a call comes in with proper callerid it shows up my Exchange UM voicemail as the number but when callerid is blocked it shows up as call from Asterisk. |
| 18:59.31 | yxa | sp00kz digium fxs module? |
| 19:00.11 | sp00kz | yxa: no a different use completly |
| 19:00.28 | sp00kz | but phone worked over it |
| 19:00.30 | jaytee | so I'm thinking I should use (name) instead of (all) or (num) |
| 19:01.04 | Strom | Qwell: what's the deal with Asterisk rewriting null caller ID as 'asterisk' anyway? that seems really dumb. |
| 19:01.12 | Qwell | dunno |
| 19:01.29 | jaytee | it's a default in the source code somewhere |
| 19:02.27 | jaytee | I haven't found anyplace else to set a default for it. I thought I remembered reading about a way but it was over a year ago and I couldn't find the reference recently now that I need it. |
| 19:02.54 | keith4 | Strom: subtle advertising? |
| 19:03.04 | jaytee | hehe |
| 19:03.23 | Strom | Qwell: what Act of God would have to occur for that to change? |
| 19:03.36 | Strom | (Act of God used here in the legal sense only) |
| 19:05.35 | Qwell | (is there a legal definition of that?) |
| 19:07.51 | Strom | http://en.wikipedia.org/wiki/Act_of_God |
| 19:10.12 | Qwell | http://en.wikipedia.org/wiki/The_Man_Who_Sued_God |
| 19:10.14 | Qwell | that sounds hilarious |
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| 19:11.00 | Strom | haha |
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| 19:25.28 | Cassetrop | hi there, anyone can tell me how to use this variables again in extensions.conf ? ${DNID} * Dialed Number Identifier (Deprecated; use ${CALLERID(dnid)}) |
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| 19:28.52 | jaytee | Strom, [TK]D-Fender, thank you both very much for your assistance. It worked! |
| 19:30.11 | [TK]D-Fender | Cassetrop: You just wrote the instruction on what you should be replacing it with.... |
| 19:30.21 | [TK]D-Fender | jaytee: you're welcome. |
| 19:30.24 | gaetronik | Cassetrop, what do you want do to do with this variable? |
| 19:30.43 | [TK]D-Fender | Cassetrop: Got any more questions you feel like answering yourself? :) |
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| 19:31.17 | Marquel | Strom: thx for your help :) |
| 19:31.33 | Strom | jaytee and Marquel: you're welcome |
| 19:31.49 | *** part/#asterisk Marquel (n=Marquel@port-232.pppoe.wtnet.de) |
| 19:35.54 | *** join/#asterisk qdk_ (n=qdk@195.242.194.41) |
| 19:37.07 | Cassetrop | [TK]D-Fender haha i can't figure out where to insert it. |
| 19:37.23 | [TK]D-Fender | Cassetrop: the exact place you would normally call that variable |
| 19:37.42 | Cassetrop | [TK]D-Fender i'm a nooooooooooooob with this. |
| 19:37.58 | Cassetrop | jaytee i need to know what number the customer dialed when he called us |
| 19:38.03 | [TK]D-Fender | Cassetrop: See where you want to use ${DNID} ? Use ${CALLERID(dnid)} instead. |
| 19:38.20 | [TK]D-Fender | Cassetrop: What is your call coming in on? |
| 19:39.42 | Cassetrop | http://www.cassetrop.com/extensions.JPG |
| 19:39.48 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 19:39.48 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:40.42 | [TK]D-Fender | Cassetrop: You already know what # it comes in on... those are the extens which all dump you to your menu |
| 19:41.12 | ManxPower | Cassetrop: usually what the customer dialed is ${EXTEN} |
| 19:41.13 | [TK]D-Fender | Cassetrop: before doing the GOTO, you should save the current ${EXTEN} into another variable so you can compare it later. |
| 19:41.55 | Cassetrop | at the [from-pstn] |
| 19:42.13 | [TK]D-Fender | Cassetrop: and just a warning : the timeout application you are using in your phrase menu are removed in 1.4 |
| 19:42.45 | [TK]D-Fender | Cassetrop: the extens in that context ARE the number that it comes in on. the fact that it MATCHE them shows that you know what # they dialed. |
| 19:43.47 | Cassetrop | they don't show on the phones |
| 19:43.55 | gaetronik | one question, why use 1.2 asterisk? |
| 19:45.14 | [TK]D-Fender | Cassetrop: Showing on the phones is YOUR job. You might do this by manipulating the callerID for example. |
| 19:45.22 | [TK]D-Fender | Cassetrop: like by adding a prefix, etc. |
| 19:46.06 | [TK]D-Fender | Cassetrop: so if the CID used to say "Jean-Francois", you would add something in front so it would look like "Sales:Jean-Francois" so you'd know he called from the "sales" line. |
| 19:47.56 | [TK]D-Fender | Cassetrop: and because i'm feeling generous I'll show you an example : exten => 5143161312,1,Set(CALLERID(name)=Sales:${CALLERID(name)}) |
| 19:48.26 | gaetronik | why Jean François? |
| 19:48.29 | Cassetrop | haha |
| 19:48.30 | [TK]D-Fender | Cassetrop: Make that the first priority in taht exten for your inbound calls and when that is the # they dialed it will add "sales:" in front of their name so you see that on your phone. |
| 19:48.42 | Cassetrop | I see |
| 19:48.48 | [TK]D-Fender | gaetronik: Just picking a name, esp as he's french. |
| 19:49.00 | [TK]D-Fender | gaetronik: the personal touch. |
| 19:49.07 | gaetronik | ok |
| 19:50.00 | [TK]D-Fender | gaetronik: and as we say... Si ca vous derange, crisse-toi dehors tabarnac! |
| 19:50.07 | Cassetrop | hahaha |
| 19:50.12 | Strom | gardez-vous au refregeradeur |
| 19:50.27 | Strom | je ne me souviens sexe du chat |
| 19:50.29 | gaetronik | [TK]D-Fender, oh my god "du quebecois" |
| 19:50.47 | [TK]D-Fender | gaetronik: "Franglais" ;) |
| 19:51.11 | gaetronik | [TK]D-Fender, I'm best in "franspagnol" |
| 19:51.19 | [TK]D-Fender | gaetronik: lol |
| 19:52.27 | gaetronik | "quebecois" is not a word i use frequently in english |
| 19:52.55 | Cassetrop | [TK]D-Fender thanks for you help i'll try that |
| 19:52.58 | Cassetrop | merci le gros! |
| 19:53.40 | gaetronik | tu es le bienvenue? |
| 19:53.46 | Cassetrop | oui |
| 19:54.00 | Strom | couche-tard |
| 19:54.29 | gaetronik | why i can't understand this kind of french |
| 19:55.38 | [TK]D-Fender | Cassetrop: and rather than hosting images use : |
| 19:55.40 | [TK]D-Fender | ~pb |
| 19:55.42 | 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 |
| 19:55.44 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 19:56.36 | Cassetrop | it works |
| 19:56.39 | Cassetrop | you rock! |
| 19:56.41 | Cassetrop | thanks |
| 19:59.09 | [TK]D-Fender | Cassetrop: np |
| 20:10.34 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 20:10.34 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 20:18.32 | *** join/#asterisk bkruse (n=bkruse@216.207.245.1) |
| 20:18.32 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 20:18.39 | bkruse | /j #asterisk-dev |
| 20:18.41 | bkruse | ouch |
| 20:18.54 | bkruse | ctrl+x + ctrl+v ftl :/ |
| 20:20.31 | *** join/#asterisk bkruse (n=bkruse@216.207.245.1) |
| 20:20.31 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 20:22.04 | s0ck | anyone use call park? |
| 20:22.26 | s0ck | doesn't always say where it's parked the call, i want to force it to a specific park extension, if possible |
| 20:23.07 | sp00kz | I use call parking, try inserting a 1/4s wait before it speaks the parked # |
| 20:23.07 | [TK]D-Fender | BBL |
| 20:23.17 | keith4 | I think when I use it, it speaks the parked spot over zap channels, and flashes the number on the screen of SIP hardphones |
| 20:23.36 | keith4 | at least, last time I parked a call on a snom 300 series, I think it did that |
| 20:24.41 | s0ck | yeh, it flashed me the number once just now |
| 20:24.48 | s0ck | the second time i tried it, it didn't happen :s |
| 20:31.42 | s0ck | snom blf still doesn't work as of 6.5.17 then... |
| 20:39.27 | unpaidbill | Read can't play multiple audio files? |
| 20:39.45 | unpaidbill | i tried using & with no luck.. that's how background does it :/ |
| 20:40.48 | *** join/#asterisk giorgiolapietra (n=lapietra@pc-110-151-104-200.cm.vtr.net) |
| 20:41.11 | giorgiolapietra | Hi, can somebody helpme? |
| 20:41.43 | giorgiolapietra | i have a some strange problem whith my agi. |
| 20:42.32 | giorgiolapietra | WARNING[16364]: res_agi.c:1779 parse_args: Too many |
| 20:42.32 | giorgiolapietra | arguments, truncating |
| 20:42.49 | giorgiolapietra | it works in 1.4.10, but no in 1.4.19 |
| 20:45.27 | unpaidbill | 01798 if (x >= MAX_ARGS -1) { |
| 20:45.27 | unpaidbill | 01799 ast_log(LOG_WARNING, "Too many arguments, truncating\n"); |
| 20:45.27 | unpaidbill | 01800 break; |
| 20:45.27 | unpaidbill | 01801 } |
| 20:45.44 | unpaidbill | it's set from.... |
| 20:45.56 | unpaidbill | a const |
| 20:46.03 | unpaidbill | 00067 #define MAX_ARGS 128 |
| 20:46.12 | unpaidbill | you'll probably have t orecompile 1.4.10 and modify that variable |
| 20:46.18 | unpaidbill | unless there's a ./configure command for it |
| 20:46.23 | jaytee | time to head home, bbl |
| 20:46.25 | unpaidbill | either way you have t orecompile |
| 20:46.34 | unpaidbill | it's in res_agi.c |
| 20:47.09 | giorgiolapietra | the proble is in 1.4.19 |
| 20:47.14 | giorgiolapietra | but no in 4.10 |
| 20:47.36 | unpaidbill | maybe they lowered the argument count in .19? check res_agi.c |
| 20:48.05 | giorgiolapietra | ok thaks... |
| 20:51.17 | unpaidbill | #define MAX_ARGS 128 |
| 20:51.23 | unpaidbill | it's 128 in .20 |
| 20:51.40 | unpaidbill | also i think .19 has IAX issues, you may want to upgrade to .20 |
| 20:51.59 | giorgiolapietra | ok... i understand... |
| 20:52.14 | giorgiolapietra | y gona change te onstan for .20 |
| 20:52.54 | seanbright | has a parse error |
| 20:53.00 | Corydon76-dig | Uh, why do you have more than 127 arguments? |
| 20:53.10 | Corydon76-dig | Isn't that a little excessive? |
| 20:56.48 | ac1djazz | hey can someone give me a pretty brief look at the optimal type of box/hardware setup for an asterisk server? |
| 20:58.01 | sp00kz | runs |
| 20:58.01 | sp00kz | ;o |
| 20:58.01 | ac1djazz | im mainly asking cuz this is gonna be ran at like maximum capacity |
| 20:58.01 | ac1djazz | so i need to get the max amount of calls at once |
| 20:58.01 | ac1djazz | from it |
| 20:58.05 | sp00kz | most of my pbx's are dual cores w/ 1-2gb ram and 2x scsi disks raid1 |
| 20:58.10 | sp00kz | ibm servers is what we use mainly |
| 20:58.43 | unpaidbill | it isnt excessive if the agi is married |
| 20:58.57 | Strom_C | ac1djazz: perhaps you should do some traffic engineering before you ask for blanket things like "maximum capacity" |
| 20:58.58 | unpaidbill | instantrimshot.com |
| 20:59.43 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:01.49 | *** join/#asterisk makkksimal (n=makkksim@e177215112.adsl.alicedsl.de) |
| 21:02.25 | *** join/#asterisk jks (i=jks@193.189.93.254) |
| 21:04.37 | jks | I have a problem where Asterisk out of the blue starts using a lot of CPU-time, audio gets choppy and it stops responding to IAX for a short while... apparently it spawns a lot of IAX dynamic threads, and then it restores itself to normal... anyone have an idea why this happens? (asterisk 1.4) |
| 21:05.35 | ac1djazz | ok to be more specific i mean 'maximum capacity' at a level of how many calls i can have simultaneousely |
| 21:06.02 | [TK]D-Fender | ac1djazz: depends on easily a dozen or more parameters |
| 21:06.13 | [TK]D-Fender | ac1djazz: What load will * be under? |
| 21:06.18 | bbryant | jks: some iax fixes have been made to the latest releases of 1.4, upgrading might help your problem |
| 21:06.25 | ac1djazz | what do you mean what load? |
| 21:06.33 | [TK]D-Fender | ac1djazz: Call recording? Transcoding? Media proxy? SWEC? |
| 21:07.08 | jks | bbryant, I have upgraded, though not to the version from yesterday... I have looked the changelogs through, do you have any more specific information that could help determine if this is the cause of the problems? |
| 21:07.36 | ac1djazz | making calls and playing a sound for like 2-3 minutes each call |
| 21:07.38 | ac1djazz | thats it |
| 21:07.39 | jks | it's not easily reproducable but seems to happen when the system is most used... (although that is still a quite "light" load from my point of view) |
| 21:08.48 | jks | bbryant, when it happens the threads are all in the socket_process() function... I didn't see that mentioned in the change log |
| 21:09.17 | bbryant | jks: what's the output of "core show locks" |
| 21:09.35 | jks | bbryant, during the problem or in general? |
| 21:09.40 | bbryant | during the problem |
| 21:09.59 | jks | bbryant, sadly I don't know, as I haven't run that... I cannot reproduce the problem myself :-( |
| 21:10.20 | jks | bbryant, hmm, there doesn't seem to be a core show locks command? |
| 21:10.33 | bbryant | ah, then you have to recompile with DEBUG_LOCKS enabled |
| 21:10.35 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 21:11.10 | spokra | ac1djazz: how many simultaneous calls do you want up playing files at one time! |
| 21:11.27 | ac1djazz | as many as i can possible :) |
| 21:11.29 | [TK]D-Fender | ac1djazz: that says nothing for the points I specifically asked for. |
| 21:11.38 | jks | bbryant, it would be really great if I could reproduce it... do you have any specific knowledge regarding a deadlock bug that could give me some information to go on in terms of reproducing the problem? |
| 21:11.39 | ac1djazz | this is already going to be a series of asterisk boxes |
| 21:11.47 | *** join/#asterisk cabbiepete (n=cabbiepe@host86-140-164-84.range86-140.btcentralplus.com) |
| 21:12.10 | bbryant | jks: no, i don't know anything about the iax deadlocks |
| 21:12.25 | spokra | then get the biggest machine possible.. more CPU's.. are the calls coming in via SIP or PRI POTS etc |
| 21:12.28 | jks | on a related note: anyone know of a sample sipp scenario that handles both register and call setup? |
| 21:13.15 | jks | bbryant, okay, it's probably something like it... it looks like a definite bug... goes instantly from almost no load to massive load and 100 dynamic threads (the limit I have set), and then drops down to 0 again after 15 seconds |
| 21:13.42 | spokra | I've heard there is a 500 call limit bacause on context switching.. would help if you had a number of calls type of hardware the calls are coming in on .. |
| 21:14.52 | ac1djazz | spokra SIP im guessing so far |
| 21:15.05 | ac1djazz | spokra yea thatd be nice |
| 21:15.06 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:16.13 | spokra | I think that is what people need here to help you./. and what was ment be 10+ different variables.. |
| 21:16.45 | ac1djazz | so we are a content distribution media company |
| 21:16.49 | ac1djazz | and we want to do this on phones |
| 21:16.58 | ac1djazz | 2-3min calls of like sports scores or whatever teh client wants |
| 21:17.06 | ac1djazz | w/ customers of like lets say 10k |
| 21:17.14 | ac1djazz | and so i need to make 10k calls in an 8hr period |
| 21:17.18 | ac1djazz | each call being 2-3 minutes |
| 21:17.26 | ac1djazz | so 30k minutes of talk time |
| 21:17.35 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 21:17.40 | spokra | ok are these prerecorded or TTS that makes a differance.. |
| 21:17.46 | ac1djazz | how many servers and what kind would i need to get this done in a fairly short amt of time |
| 21:17.49 | ac1djazz | yes |
| 21:17.51 | ac1djazz | prerecorded |
| 21:17.53 | ac1djazz | what is TTS? |
| 21:18.11 | spokra | text to speech |
| 21:18.29 | spokra | TTS would take more CPU |
| 21:18.40 | ac1djazz | nono this is pre-recorded |
| 21:18.52 | ac1djazz | so 10k pre-recorded calls in like an hour or two would be awesome |
| 21:18.59 | ac1djazz | what kinda server and how many :) |
| 21:19.32 | spokra | fast disk if you have many differnt streams going at one time.. think of it like a database system with random reads.. many disks in a raid gets fast read times |
| 21:20.07 | spokra | that depends on the hardwar and your load. best to build the bigest and find out what it can do. |
| 21:21.52 | ac1djazz | ohya? |
| 21:21.55 | ac1djazz | so maybe a solid state disk? |
| 21:22.09 | spokra | at 10,000 3 minute calls and 2 hours you would keep 250 calls up.. |
| 21:22.17 | ac1djazz | 240 at a time |
| 21:22.19 | ac1djazz | 250* |
| 21:22.30 | [TK]D-Fender | ac1djazz: 30,000 minues in a span of 720 minutes (8hrs) = 41.1 simultaneous calls assuming it never runs over. that'd require 2 PRI's or a whole pile of bandwidth and an ITSP. If you kept to a matching codec (as I'm sure you would), you could do this off a single big server. |
| 21:23.16 | ac1djazz | i cant do 8hrs |
| 21:23.22 | ac1djazz | i need to do like 1-3hrs |
| 21:23.23 | ac1djazz | 1-2hrs |
| 21:23.39 | ac1djazz | i just said 8 originally beacuse thats like the general range people are up and accepting calls |
| 21:23.50 | [TK]D-Fender | ac1djazz: then maybe 4 servers, and connectivity to match |
| 21:24.06 | ac1djazz | what kind of server? |
| 21:24.14 | [TK]D-Fender | ac1djazz: I'm ight just say 2.... |
| 21:24.20 | spokra | so do you get free porn for helping.. ROFLOL |
| 21:24.25 | [TK]D-Fender | ac1djazz: decent core2 + 4 gig ram |
| 21:25.30 | ac1djazz | 2 thats it eh ? |
| 21:25.38 | ac1djazz | each doing like 125 calls at a time? |
| 21:26.04 | [TK]D-Fender | ac1djazz: look at the math I jsut gave you for 1 server for 8 hours.... do the math |
| 21:26.21 | x86 | http://blog.wired.com/gadgets/2008/06/snake-oil-alert.html |
| 21:26.26 | x86 | LAWLZ! |
| 21:26.29 | ac1djazz | oh 8hrs |
| 21:26.50 | ac1djazz | what about for 1hr, what about 250 simultaneous calls |
| 21:27.15 | jks | anyone know of a site with sample scenarios for sipp? (besides the 4 samples on the sipp wiki) |
| 21:27.27 | x86 | sipp? |
| 21:27.41 | [TK]D-Fender | ac1djazz: Go take grade-school math over again... |
| 21:27.57 | jks | x86, it's a trafic generator for SIP |
| 21:28.37 | ac1djazz | or how about this, how many simultaneous asterisk calls do you think a core2 machine w/ 4gigs of ram could take? |
| 21:28.52 | jks | ac1djazz, you need to be way more specific for anyone to give you a hard figure |
| 21:29.00 | jks | ac1djazz, you will have to do some measurements yourself |
| 21:29.05 | [TK]D-Fender | ac1djazz: for the kind you'd do, 1-2 hundred easily. |
| 21:29.15 | ac1djazz | really? sweet |
| 21:29.28 | ac1djazz | so to be safe maybe ill limit each box to 150? |
| 21:29.42 | [TK]D-Fender | ac1djazz: go TRY. |
| 21:29.47 | ac1djazz | what version would be best for this? 1.4 or 1.6? |
| 21:29.54 | ac1djazz | i will, but im still buying the hardware right now :) |
| 21:30.25 | [TK]D-Fender | ac1djazz: what do YOU think? |
| 21:30.26 | Strom | ac1djazz: I strongly recommend you do actual traffic analysis and run the erlang formulae BEFORE you spend any money on hardware |
| 21:30.39 | jks | mmm, erlang |
| 21:32.27 | bkruse | <3's erlang |
| 21:32.27 | ac1djazz | hmm ok |
| 21:33.03 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 21:33.09 | *** join/#asterisk FuriousGeorge (n=brian@ool-457f216e.dyn.optonline.net) |
| 21:33.20 | FuriousGeorge | hey all |
| 21:35.07 | x86 | FuriousGeorge: http://blog.wired.com/gadgets/2008/06/snake-oil-alert.html |
| 21:35.41 | x86 | no one is taking a break here to appreciate the hilarity of the $500 1.5 meter cat5 cable |
| 21:35.45 | ac1djazz | erlang unit eh? |
| 21:37.01 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
| 21:41.44 | FuriousGeorge | ive never been able to get asterisk to run reliably. its 80% chan_zap, 10% chan_sip, and 10% occasional crash |
| 21:41.59 | FuriousGeorge | its very difficult to get a bug report out of this |
| 21:42.18 | FuriousGeorge | been using it since 1.1.x, but im not a professional C developer, so i think that is the problem |
| 21:43.34 | [TK]D-Fender | FuriousGeorge: I've been running * for over 4 years, it pretty much never crashes on me, and I use SIP / Zap, etc |
| 21:43.59 | [TK]D-Fender | FuriousGeorge: And I don't code either, or even patch the release-only tarballs I install |
| 21:44.08 | jks | anyone know of a channel where discussions acout SIPp takes place? |
| 21:45.08 | FuriousGeorge | [TK]D-Fender: i never get crashes either |
| 21:45.51 | FuriousGeorge | but i do get bad hangup detection, on one server its SIP (strangely between two specific extensions), and on another its zap on incoming calls |
| 21:47.10 | FuriousGeorge | well, i wouldnt say never. i got a core dump in april, and my bug report then got a shoutout in the release notes |
| 21:47.13 | *** join/#asterisk fetcher (n=jnh@ip-72-55-165-168.static.privatedns.com) |
| 21:47.22 | FuriousGeorge | and I see that I had another core dump in late may |
| 21:48.08 | FuriousGeorge | [TK]D-Fender: anyway, the point is not to bitch about it, but to instead spam for my new forum post, as I'm sick of the restart nightly and cross-fingers approach. http://forums.digium.com/viewtopic.php?p=72332#72332 |
| 21:48.48 | fetcher | Is there a trick to making voicemail files world-readable (0644 access mode instead of 0600)? |
| 21:48.52 | ac1djazz | whats the most reliable version of asterisk out there? |
| 21:49.19 | Qwell | The latest one. |
| 21:49.19 | fetcher | changing VOICEMAIL_FILE_MODE and VOICEMAIL_DIR_MODE in apps/app_voicemail.c doesn't seem to have any effect |
| 21:51.25 | x86 | http://i17.tinypic.com/5xqrlsg.jpg |
| 21:51.26 | x86 | LOL |
| 21:52.25 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:59.32 | *** part/#asterisk MikeJ (n=MikeJ@33.203.64.208.static.accentrainc.com) |
| 22:03.31 | FuriousGeorge | how can i file a useful bug report for something like hangup detection, when there is no core dump? |
| 22:04.17 | FuriousGeorge | or, on another server, two sip extensions that randomly after a few weeks will have 20 active channels between them, but no one is on the phone. |
| 22:04.34 | FuriousGeorge | the first problem is by far my biggest, but i'd be happy with a suggestion for either :) |
| 22:05.15 | *** part/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 22:09.54 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 22:11.54 | *** join/#asterisk Segnale007 (n=Segnale0@cheetah-tiv-ve3000-127-0064.bmts.com) |
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| 22:53.04 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 22:53.42 | CanWood | Hey folks. I have a call that won't register as having ended. Is there a way, from the asterisk command line, to force it to end? |
| 22:53.55 | CanWood | the guy hung up and can dial out. It's odd |
| 22:54.38 | *** join/#asterisk maurot (n=usuario@host209.201-252-132.telecom.net.ar) |
| 22:55.03 | maurot | i need help i configured asterisk but i listening very slow the voice |
| 22:55.07 | maurot | any idea? |
| 22:58.34 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 23:01.30 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 23:05.58 | CanWood | soft hangup [the channel listing] didn't end it. Neither did an unplug and plug in of the phone |
| 23:07.23 | Strom_C | what kind of channel is hung here again? |
| 23:08.15 | CanWood | SIP |
| 23:08.36 | CanWood | show channels shows he was checking voicemail and the call won't end |
| 23:09.14 | CanWood | I've even tried a reload to no avail |
| 23:10.05 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:10.13 | Strom_C | restart asterisk? |
| 23:10.20 | Strom_C | (assuming you can) |
| 23:10.59 | CanWood | the other extensions are in calls |
| 23:11.16 | CanWood | so I'm trying everything I can think of before doing that] |
| 23:11.55 | CanWood | I'm going to try unloading and reloading the voicemail app since that's what he was connected to |
| 23:12.13 | Strom_C | what kind of phone is it, out of curiosity? |
| 23:12.30 | CanWood | gxp-2000, but I've even unplugged it and plugged it back in |
| 23:12.32 | CanWood | he can dial out |
| 23:13.47 | Strom_C | ah, grandstream |
| 23:13.49 | Strom_C | how did I know |
| 23:14.58 | CanWood | "show channels" returns |
| 23:14.59 | CanWood | SIP/20##-081e3a58 *97@from-internal:10 Up VoiceMailMain(20##@default) |
| 23:16.08 | Strom_C | you know, you don't need to mask the extension number |
| 23:16.28 | Qwell | Strom_C: but, he doesn't want anybody to call him |
| 23:19.57 | Strom_C | CanWood: which version of asterisk? |
| 23:20.09 | [TK]D-Fender | <PROTECTED> |
| 23:20.12 | drako | grandstream phone and ATAs sucks.... |
| 23:20.21 | Strom_C | Qwell: isn't there some "forcibly tear down the channel" command like there is with zaptel? |
| 23:20.50 | Strom_C | sip destroy channel SIP/2368 |
| 23:20.55 | [TK]D-Fender | CanWood: And when you try to soft hangup it, what exactly does it say? |
| 23:20.56 | Strom_C | sip rape channel SIP/2368 |
| 23:21.16 | Strom_C | etc |
| 23:26.12 | CanWood | sorry there |
| 23:26.25 | CanWood | stepped off to go explain to my manager why I took the phone system down :) |
| 23:27.48 | CanWood | to answer, it's 1.4.19, and I don't remember what the console said with the soft hangup command |
| 23:29.48 | Strom_C | for future reference, CanWood, a single hung channel is not exactly a reason to take the entire system down during business hours |
| 23:30.59 | CanWood | small shop, and he gets the irate customers |
| 23:31.29 | Strom_C | was it preventing him from receiving calls? |
| 23:31.35 | CanWood | when no one can call the VP, I do what I need to. Yes, it was |
| 23:31.44 | Strom_C | ah, ok |
| 23:31.49 | Strom_C | you didn't make that clear. |
| 23:33.59 | CanWood | well, that was fun. thanks for the tips folks |
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| 23:47.22 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
