| 00:04.20 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 00:05.26 | sawgood | Is there a 'term' to describe when another side of a connection (not Asterisk) is sending your box a "+" symbol and 11 digits |
| 00:05.52 | sawgood | I do not have a + in myside of the dialplan, so incoming calls are being dropped |
| 00:06.16 | sawgood | I do not know the correct term to tell 'the other side' to stop sending a + symbol |
| 00:06.24 | [TK]D-Fender | so fix your dialplan |
| 00:06.48 | sawgood | I could fix it ... but all other connections send me 11 digits just fine |
| 00:07.02 | sawgood | A standard has been set so to speak (send me 11 digits) |
| 00:07.10 | [TK]D-Fender | Well these guys dson't |
| 00:07.23 | WIMPy | exten => _+.,1,Goto(${EXTEN:1},1) |
| 00:08.36 | sawgood | point being ... why do other connections send a + in their SIP messages and by default Asterisk does not? |
| 00:09.13 | sawgood | could be me though ... I slammed my finger into a wall while skating ... |
| 00:09.19 | [TK]D-Fender | "by default Asterisk does not" <- Asterisk sends whatever you tell it to |
| 00:09.20 | sawgood | nasty hit |
| 00:09.52 | [TK]D-Fender | Stop the insanity </powter> |
| 00:10.11 | WIMPy | Yes, nno such thing as a default. |
| 00:11.41 | sawgood | WIMPy: thank you for the shortcut (nice work) |
| 00:12.22 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176145181.dsl.bell.ca) |
| 00:12.48 | *** join/#asterisk Russ (~russ@206.29.182.170) |
| 00:12.54 | *** join/#asterisk coppice (~chatzilla@host86-156-235-178.range86-156.btcentralplus.com) |
| 00:22.05 | WIMPy | Yes, that direction is the easy one. |
| 00:22.48 | sawgood | I like how it stays in the same context and picks up where it was last at |
| 00:24.10 | *** join/#asterisk F2Knight (~ben@c-24-22-60-186.hsd1.or.comcast.net) |
| 00:24.14 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 00:24.43 | F2Knight | Q: Using ChanIsAvail , does anyone know what ${AVAILSTATUS} == 21 means? |
| 00:41.18 | *** join/#asterisk mindCrime_ (~chatzilla@24.106.207.82) |
| 00:48.15 | *** join/#asterisk dwmw2 (~dwmw2@twosheds.infradead.org) |
| 00:53.01 | bacon4leif | F2Knight: what version are you using? |
| 00:54.17 | bacon4leif | F2Knight: regardless, the answer is likely found at the top of the include/asterisk/causes.h file |
| 00:56.08 | carrar | F2Knight, did you look in the source? |
| 00:57.13 | [TK]D-Fender | F2Knight: Who shot J.R.? |
| 00:57.32 | [TK]D-Fender | F2Knight: What is the average airspeed velocity of an unladen swallow? |
| 00:57.53 | [TK]D-Fender | F2Knight: How much wood could a woodchuck chuck if a woodchuck could chuck wood? |
| 00:58.59 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-gjveqcczjswgokxd) |
| 01:03.02 | F2Knight | carrar, no did not look , didn't know where to look but now I do. |
| 01:03.42 | carrar | devicestate.c |
| 01:04.01 | carrar | however I don't see it in there |
| 01:04.36 | F2Knight | bacon4leif, thanks, located /usr/src/asterisk/include/asterisk/causes.h |
| 01:04.47 | bacon4leif | yes, as I stated :) |
| 01:05.03 | *** join/#asterisk moy (~moy@12.238.42.3) |
| 01:06.25 | *** join/#asterisk dwmw2 (~dwmw2@twosheds.infradead.org) |
| 01:06.43 | carrar | yeah its in that file |
| 01:07.03 | F2Knight | interesting .. the code says it is for a _CALL_REJECTED.. but the extension is really not on line. |
| 01:07.12 | carrar | a hangup cause |
| 01:08.06 | F2Knight | no, it was is the AVIAILSTATUS as returned from ChanIsAvail |
| 01:08.35 | F2Knight | I know the extension is 'offline' but I would have expected perhaps an unavailable rather then a rejected. |
| 01:09.23 | bacon4leif | ah, AVAILSTATUS may not be what links to causes.h then |
| 01:09.43 | F2Knight | ah.. |
| 01:09.51 | bacon4leif | that is likely what AVAILCAUSECODE goes to |
| 01:09.52 | F2Knight | the love hate relationship of asterisk |
| 01:10.13 | F2Knight | the availcausecode actually returns nothing. |
| 01:10.30 | bacon4leif | right, so you're looking for a different cause -- someone mentioned devicestate.c |
| 01:10.56 | bacon4leif | runs away from the computer |
| 01:11.57 | F2Knight | -- Executing [~~s~~@ael-std-exten:10] Goto("SIP/sip2.didx.net-00000019", "sw_160_21,10") in new stack |
| 01:11.57 | F2Knight | [Nov 21 16:58:41] -- Goto (ael-std-exten,sw_160_21,10) |
| 01:11.57 | F2Knight | [Nov 21 16:58:41] -- Executing [sw_160_21@ael-std-exten:10] NoOp("SIP/sip2.didx.net-00000019", "AVAILSTATUS: 21, []") in new stack |
| 01:11.57 | F2Knight | [Nov 21 16:58:41] -- Executing [sw_160_21@ael-std-exten:11] Goto("SIP/sip2.didx.net-00000019", "~~s~~,11") in new stack |
| 01:14.05 | *** join/#asterisk Diffen (~diffen@c-a27ce555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 01:14.52 | Diffen | Evning. Is it possible to setup my asterisk as a sip-trunk provider to another asterisk? If so, where can I read more about it? |
| 01:18.59 | F2Knight | Diffen, sure is., first quesiton is are the boxes with static IP addresses or dynamic |
| 01:19.28 | F2Knight | bacon4leif, looked couldn't find it anywhere.. suppose its not a big issue at this point. |
| 01:20.10 | Diffen | F2Knight it will be static IP-addresses |
| 01:33.06 | F2Knight | do you want to connect over IAX or SIP? |
| 01:33.53 | Diffen | F2Knight over SIP. |
| 01:34.37 | F2Knight | if you are doing it over sip and with static IP's you can forgo authenticaion, and just use the host= with the ip of the destination |
| 01:36.30 | Diffen | Ok, so the only thing I need to do is to let the Asterisk server 2 connect to Asterisk server 1 and then the outgoing calls from Asterisk 2 will reach PSTN via Asterisk 1 without i need to do anything on Asterisk 1? |
| 01:37.07 | Diffen | Inbound I need to route the incoming traffic on certain numbers on to the SIP-trunk to Asterisk 2. That shouldnt be a problem. |
| 01:39.04 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 02:02.58 | *** join/#asterisk celord (~cesar@celord.ice.co.cr) |
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| 02:41.53 | *** part/#asterisk cbwest (~cbwest@nat/cisco/x-jzowdjvmbymhziof) |
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| 03:03.38 | *** part/#asterisk hovel (~hovel@unaffiliated/hovel) |
| 03:14.30 | justdave | do I need to do anything special in a SIP config for a trunk if I know the RTP is going to be coming from a different IP address than the SIP signalling? |
| 03:14.44 | justdave | I imagine I need an allow line or something |
| 03:15.15 | bacon4leif | that should just be configured in the SIP signalling |
| 03:15.19 | pabelanger | justdave: no, it should just work. |
| 03:15.20 | bacon4leif | no configuration required |
| 03:15.28 | justdave | ok. |
| 03:15.36 | bacon4leif | pabelanger: what are you doing here?! |
| 03:15.51 | pabelanger | bacon4leif: I know right |
| 03:15.53 | justdave | I know the sip signalling reports the IP to talk to, but I was thinking it might balk at it being different by default or something. |
| 03:16.00 | bacon4leif | nope |
| 03:16.02 | bacon4leif | that's just how it works |
| 03:16.06 | justdave | then again, if it did care that much, NAT would probably work better :) |
| 03:16.07 | F2Knight | does anyone have audio files for call forwarding setups? |
| 03:16.22 | bacon4leif | SIP signalling and RTP media are separate protocols anyways, so you have to say what the IP is regardless |
| 03:16.41 | bacon4leif | I rarely have problems with NAT |
| 03:16.47 | bacon4leif | not sure why people find it such a big deal |
| 03:17.03 | WIMPy | F2Knight: Look at your sounds directory. |
| 03:17.10 | [TK]D-Fender | justdavedo I need to do anything special in a SIP config for a trunk if I know the RTP is going to be coming from a different IP address than the SIP signalling? <- nat = no |
| 03:17.32 | justdave | 90% of the times I've had people who had problems with NAT it turned out to be they had a firewall that was trying to alter the SIP packets and screwing them up instead |
| 03:17.56 | bacon4leif | justdave: yes... that |
| 03:18.03 | justdave | having them disable sip handling on their firewall usually fixed it |
| 03:18.07 | bacon4leif | don't let your router mangle packets |
| 03:18.17 | bacon4leif | because it almost does it wrong |
| 03:20.50 | F2Knight | WIMPy, yea i have just didnt like any the way i put them together... was wondering if someone else had a better chain was all |
| 03:21.55 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-zrppvncasjmzxjcc) |
| 03:22.06 | WIMPy | What's wrong with them? |
| 03:41.40 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 03:45.35 | *** join/#asterisk mindCrime_ (~chatzilla@cpe-076-182-089-009.nc.res.rr.com) |
| 03:45.46 | *** join/#asterisk dandate2 (~dan@112.206.78.17) |
| 03:46.45 | dandate2 | so i have an extention using eyebeam whose calls automatically drop after about 20 seconds. and his callerID reads oddly; in the format of <did@ip:5060> |
| 03:47.12 | dandate2 | whereas functioning softphones just see <did> on their c aller id |
| 03:47.23 | *** join/#asterisk master_of_master (~master_of@p57B55A84.dip.t-dialin.net) |
| 03:49.24 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-vhyxlsaljwgwuuru) |
| 04:00.12 | F2Knight | dandate2, can you try a different softphone? |
| 04:00.39 | dandate2 | actually he got it working by switching internet providers |
| 04:00.43 | dandate2 | strange times |
| 04:01.04 | *** join/#asterisk spotter (~spotter@24.42.114.21) |
| 04:01.43 | spotter | is it possible to have an ivr/auto attendant system while a user is on hold in a queue? |
| 04:02.03 | spotter | ex: to extract more information from caller if agent isn't available |
| 04:02.30 | spotter | but on the flip side, not delay the user if an agent is ready (or becomes ready) |
| 04:03.08 | dandate2 | everything is possilble |
| 04:03.23 | dandate2 | you just need the right berzerk programmer to set it up |
| 04:03.29 | spotter | let me rephrase, without hacking Queue |
| 04:03.45 | spotter | I can modify Queue if need be, but wondering if its possible to leverage it and have this functionalit |
| 04:04.09 | dandate2 | gunna need to hack that queue actually |
| 04:06.19 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 04:18.07 | spotter | dandate2, the other Q is, what does the AGI functionality of Queue do |
| 04:20.01 | dandate2 | i dont think theres any preset options like that |
| 04:20.19 | dandate2 | to change what happens when you put people on hold other than hearing default moh; gunna need to rescript asterisk |
| 04:20.41 | dandate2 | but certainly possible |
| 04:22.10 | spotter | is it just forking a copy of queue and playing around with that? |
| 04:22.33 | spotter | wondering what the agi option in queue accomplishes |
| 04:25.11 | spotter | I guess I can play with it and see what happens |
| 04:25.42 | F2Knight | spotter, from what I recall, the Queue is pertty much isolated from the rest of the way asterisk works.. a walled garden so to speak. |
| 04:26.25 | F2Knight | you may be able to impliment the same type of features as a queue using a custom AGI script by its self, then still have the ability to keep additional informaiton. |
| 04:26.44 | F2Knight | Remember AGI is a way for asterisk to run 'external' programs. |
| 04:27.03 | spotter | F2Knight, my Q is, can the AGI basically replace moh |
| 04:27.04 | [TK]D-Fender | [23:01]spotteris it possible to have an ivr/auto attendant system while a user is on hold in a queue? <- "context=" in your queue definition |
| 04:27.25 | F2Knight | it would be trivial to make a user hear a music loop with out using moh |
| 04:27.29 | spotter | [TK]D-Fender, can you explain? |
| 04:27.29 | [TK]D-Fender | [23:26]F2KnightRemember AGI is a way for asterisk to run 'external' programs. <- AGI is an external program |
| 04:27.51 | [TK]D-Fender | System() and SHELL() are ways to directly run a simple outside command |
| 04:27.53 | spotter | F2Knight, right, so agi can then ask user Qs and do the ivr itself |
| 04:28.05 | spotter | if agent picks up, agi presumambly ends |
| 04:28.29 | [TK]D-Fender | It's advisable to use AGI only when you need to do more processing than is reasonable in the dialplan by other means |
| 04:28.44 | [TK]D-Fender | [23:28]spotterif agent picks up, agi presumambly ends <- this cannot be done. |
| 04:28.57 | [TK]D-Fender | (without serious trickery) |
| 04:29.05 | spotter | what's the agi option to queue then? |
| 04:29.12 | F2Knight | spotter, no AGI for the most part stays active the entire time of the call |
| 04:29.47 | [TK]D-Fender | in queues you have an "exit " IVR which maps out what keys a user can use to exit the queue. Single digit exten you can advertise in a prompt like "to leave a Vm for us to call you back you can press 4 at any time", etc |
| 04:30.17 | spotter | interesting |
| 04:30.20 | spotter | so I can fake it out |
| 04:30.34 | spotter | enter and exit the queue as needed |
| 04:30.43 | spotter | suck if lose position |
| 04:30.46 | F2Knight | spotter, no... you would record your custom audio prompts to include what you wanted to tell them. |
| 04:30.47 | spotter | so would need to figure that out |
| 04:31.07 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:31.08 | spotter | F2Knight, yes, and I can have them "exit" the queue as a response |
| 04:31.11 | [TK]D-Fender | "AGI Will setup an AGI script to be executed on the calling party's channel once they are connected to a queue member." <- The instruction it gives you tells you right up front. |
| 04:31.16 | F2Knight | if they pressed one of the digits (DTMF tones) they would exit the Queue and run said AGI script |
| 04:31.16 | spotter | and make note of that in my dialplan |
| 04:31.34 | [TK]D-Fender | "once connected". not "in the background while they wait" |
| 04:31.46 | spotter | [TK]D-Fender, ok that makes sense |
| 04:32.26 | F2Knight | you might be better trying to see if the agent is available first and then if not ask additional quesitons, |
| 04:32.39 | spotter | F2Knight, I agree |
| 04:32.53 | spotter | but then basically delay if agent becomes available |
| 04:32.59 | F2Knight | pretty much. |
| 04:33.12 | F2Knight | sounds ugly and stupid but ... well sometimes things are. |
| 04:33.26 | spotter | options |
| 04:33.32 | spotter | just trying to learn what's possible |
| 04:33.56 | spotter | maybe I'll make a project of trying to add dialplan like actions to queue |
| 04:34.08 | [TK]D-Fender | spotter: Now there is a "dirty" way of making this possible : instead of dumping into a queue directly your caller will instead Originate() a new call passing the currrent channel name on in a variable. That new local channel will sit in queue for you. on connect have THAT one use the AGI parm, taking the original channel name and BRIDGE-ing them in. |
| 04:34.25 | [TK]D-Fender | Effectively hijacking the guy who merrily went along filling out IVR options, etc |
| 04:34.52 | F2Knight | [TK]D-Fender, that is 'dirty' |
| 04:34.56 | [TK]D-Fender | Yup |
| 04:34.57 | spotter | [TK]D-Fender, can I pick your brain on this later when I understand more asterisk? |
| 04:35.02 | [TK]D-Fender | sure |
| 04:35.27 | F2Knight | I can only see the nightmare now of trying to debug why 'queues' arent working with that |
| 04:35.42 | [TK]D-Fender | Chan_local is the most incredibly useful piece of Asterisk. |
| 04:35.59 | F2Knight | so long as you remember to pass /n to the channel |
| 04:36.11 | spotter | I just got the book on sat, read through a lot of it |
| 04:36.12 | [TK]D-Fender | Never rebridge <- |
| 04:36.15 | spotter | but still trying to learn |
| 04:36.16 | [TK]D-Fender | <PROTECTED> |
| 04:36.43 | F2Knight | the lack of the /n has perplexed me too many times. |
| 04:37.34 | F2Knight | [TK]D-Fender, I am writing a new dialplan full of fun little macros.. What is a good way to 'dynamicly' create a failover ? |
| 04:37.38 | F2Knight | for the trunks that is. |
| 04:37.51 | [TK]D-Fender | define "dynamic" |
| 04:38.20 | ChannelZ | jazz hands |
| 04:38.34 | F2Knight | well lets say I have a database (astdb,mysql,sqlite, flatfile, what ever) of providers. |
| 04:39.36 | F2Knight | in said list I perform some type of matching ... say for example ... I have a list of numbers I might perfer to send out one trunk over another, or maybe route based on cost. |
| 04:39.49 | F2Knight | but that list may be anywhere from 2 providers to 100 providers. |
| 04:40.46 | F2Knight | I can create an AGI script to do the SQL lookup and loop through the list with out a problem, then hit each dial from there... but that will keep the AGI open the whole time... |
| 04:40.49 | [TK]D-Fender | Everything is vague right now. Multiple criteria and no means of prioritizing yet |
| 04:41.09 | F2Knight | I see no other way though looping through a 'list' of providers |
| 04:41.16 | F2Knight | at leat not cleanly. |
| 04:41.38 | [TK]D-Fender | Hit AGI. make your choice in there. Set VARIABLES for the results of your processing. exit AGI back to the dialplan and use those vars to do your dialout <- |
| 04:41.52 | [TK]D-Fender | Simply don't do the Dial in the AGI itself |
| 04:42.15 | [TK]D-Fender | Make the choice in there, but no the dial and you won't leave those processes sitting open for no good reason |
| 04:42.17 | F2Knight | right but ... how would you 'loop' it? |
| 04:42.29 | [TK]D-Fender | Variables <- |
| 04:42.42 | [TK]D-Fender | So the AGI knows where it left off. |
| 04:42.45 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-dxbriiflesfdefdy) |
| 04:42.53 | F2Knight | oh and keep recalling the agi? |
| 04:42.55 | [TK]D-Fender | yup |
| 04:43.01 | F2Knight | umm that seems a drag. |
| 04:43.04 | [TK]D-Fender | only the smallest processing to pick the next step |
| 04:43.24 | [TK]D-Fender | a drag is having a ton of calls all leaving agi's open all over the place |
| 04:43.40 | F2Knight | but I guess it would only be hitting 1 anyways omst of the time.. only the second or more hit if they failed. |
| 04:43.51 | [TK]D-Fender | And there you have it |
| 04:44.14 | F2Knight | thnx, didn't even thing about just looping the call on the AGI part, |
| 04:44.39 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 04:46.08 | F2Knight | dont even know why i didn't think of that I have a chat-line that does the exact same thing... looks for an available agent in the agi and sends the dial back to asterisk |
| 04:46.21 | [TK]D-Fender | Yup |
| 04:48.32 | [TK]D-Fender | There are a few other ways you could do this as well. Like building ALL of the dial strings in order of applicability and storing them in a SQL with the channel name as key. Single DB pull for the next record, etc. |
| 04:48.48 | [TK]D-Fender | So only 1 AGI call, and 1 DB pull for the loop. |
| 04:49.22 | [TK]D-Fender | If you build the full list and just execute in order you only call AGI once. You'd simply wat to clean up the DB at the end |
| 04:49.39 | [TK]D-Fender | But the processing load of that would be very petty |
| 04:54.49 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 04:59.42 | *** join/#asterisk the_5th_wheel (~edd@tcs-gw.bulwer.thusa.net) |
| 05:00.37 | the_5th_wheel | Hi All. Has anyone had issues with soxmix not mixing entire asterisk mixed tracks together? Im having a strange issue where the mixed recording stops at about 20% into the call. Are there perhaps any alternatives to soxmix? |
| 05:08.31 | *** join/#asterisk radic (~radic@dslb-178-002-231-152.pools.arcor-ip.net) |
| 05:21.50 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176002135.dsl.bell.ca) |
| 05:22.03 | dijib | honey honey yall |
| 05:22.07 | dijib | hows everyone? |
| 05:22.29 | *** join/#asterisk gogasca (~Adium@nat/cisco/x-jfxdypacxppgnsef) |
| 05:31.19 | SeRi | dijib, |
| 05:31.26 | SeRi | waz up |
| 05:33.05 | SeRi | took my final. today. man it was hard but I think I did ok :) |
| 05:37.25 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 05:39.41 | ChannelZ | Did you have to use the N-word in a sentence? |
| 05:40.20 | SeRi | Yes. a few times. It was awkward... :/ |
| 05:40.30 | SeRi | lol j/k :P |
| 05:52.27 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 06:01.50 | F2Knight | can anyone tell me the difference between asterisk functions and asterisk applications? example func_db.so app_db.so |
| 06:02.24 | SeRi | ~functions |
| 06:02.32 | SeRi | damn boot. |
| 06:02.34 | [TK]D-Fender | You call functions in applications to get potentially variable results |
| 06:02.54 | [TK]D-Fender | Some are used to get & set certain properties, etc |
| 06:02.57 | F2Knight | <PROTECTED> |
| 06:03.40 | F2Knight | same error if i just do DB(RCID/${EXTEN}) |
| 06:03.48 | [TK]D-Fender | Because it is a function, not an application |
| 06:03.58 | [TK]D-Fender | (both) |
| 06:04.41 | F2Knight | I am calling it from the dialplan. |
| 06:04.56 | [TK]D-Fender | You are attempting to call it as an applicaiton |
| 06:05.15 | [TK]D-Fender | instead of as a function within some other application |
| 06:05.16 | F2Knight | okay .. well lets define the differnce |
| 06:05.28 | F2Knight | in my context ael-inbound |
| 06:05.29 | [TK]D-Fender | I already did. |
| 06:05.35 | F2Knight | well macro actually |
| 06:05.45 | F2Knight | I am calling it from a macro |
| 06:05.46 | [TK]D-Fender | Functions return values.. values you wmay be using in an application call. |
| 06:06.16 | kaldemar | exten => s,1,Application(${FUNCTION(value)}) |
| 06:06.25 | SeRi | Like GotoIf |
| 06:06.35 | F2Knight | running it in an ael macro |
| 06:07.10 | [TK]D-Fender | macro = jsut a context. Has no specific relation to functions |
| 06:07.10 | F2Knight | Set(DB(RCID/${exten})=${CALLERID(num)}); works fine |
| 06:07.27 | F2Knight | but does not when trying to ready it. |
| 06:07.47 | [TK]D-Fender | ]F2Knightbut does not when trying to ready it. <- huh? |
| 06:07.48 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 06:08.56 | F2Knight | read. |
| 06:09.15 | [TK]D-Fender | Then you're reading it wrong |
| 06:13.22 | kaldemar | F2Knight: think about this for a moment: ${DB(RCID/${exten})} |
| 06:23.37 | *** join/#asterisk Defraz (~tim@70.36.76.167) |
| 06:24.15 | Defraz | Hey all I am having trouble with a cisco 7960 phone. Everything is great except when I se the call forwarding softkey, my logs of asterisk say app_dial.c: Not accepting call completion offers from call-forward recipient |
| 06:24.20 | Defraz | Has anyone seen this? |
| 06:30.44 | Defraz | exit |
| 06:45.59 | *** join/#asterisk irroot (~gregory@196-215-57-105.dynamic.isadsl.co.za) |
| 06:51.08 | justdave | who would I need to ping to get a symlink for 5Server pointing at 5 in http://packages.asterisk.org/centos/ ? |
| 06:52.16 | justdave | that fun asterisknow-version package they pushed in the last couple days to switch people over to the new repos automatically fails on RHEL because of that |
| 06:52.37 | justdave | (and breaks yum until you manually fix the repo files) |
| 07:07.03 | *** join/#asterisk IsUp (5b8e8e8f@gateway/web/freenode/ip.91.142.142.143) |
| 07:08.39 | *** join/#asterisk crioto (~crioto@92-245-121-119.mega.kg) |
| 07:09.38 | crioto | Hi everyone. Does TIMEOUT(response)=20 will include my Background playback time or it will wait for 20 seconds after Background is finished? |
| 07:13.17 | IsUp | crioto: 'core show function TIMEOUT' may help |
| 07:17.30 | IsUp | i am debugging my PRI with 'intense' debug, also i am using Sangoma's debug tool, it creates pcap files. |
| 07:18.03 | crioto | I've read it already, but i don't think i'm getting it - when timeout starts? after background is finished or right when my extension starts |
| 07:18.10 | IsUp | how can i understand if i have a problem on my span? i see some "[Malformed Packets]" in pcap files. |
| 07:18.35 | IsUp | crioto: maybe you can test? :p |
| 07:18.51 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 07:19.16 | WIMPy | IsUp: Do you see any disturbing messages in the pri debug? |
| 07:19.39 | *** join/#asterisk irroot (~gregory@196-215-57-105.dynamic.isadsl.co.za) |
| 07:19.47 | WIMPy | And is the a reason you think there might be a problem? |
| 07:20.01 | IsUp | WIMPy: actually its all disturbing :P well actually yes, i suspect that my telcos timing is corrupted. let me explain |
| 07:20.37 | WIMPy | Do you see HDLC aborts or something? |
| 07:20.43 | IsUp | WIMPy: my system was stable 3 days ago. i have telco PRI on span 1, also i have 3 GSM gateways on span 2, 3, 4 - my gsm gateways are using span 1 clock (ref clock) |
| 07:21.15 | IsUp | WIMPy: i have robotic voices, strange sounds, buzzers. my telco link was gone 3 days ago, just for 5 minutes. but i think thats when the problem started. |
| 07:22.11 | WIMPy | Have you tried to restart the interface? Either by software or by replugging the line? |
| 07:22.33 | kaldemar | crioto: "after falling through a series of priorities" |
| 07:22.57 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 07:23.55 | IsUp | WIMPy: yes i did all. ive checked physical layer, software and everything. to be clear, some ports are working fine on GSM gateways, no sound problems or anything |
| 07:24.28 | WIMPy | Have you also tried to reset the nt? |
| 07:24.34 | IsUp | WIMPy: also my problem is not sound at all, i have a problem with specific channels, DAHDI 38,39...,48 |
| 07:24.51 | IsUp | WIMPy: yes, also i already talk to my telco for reset their side but nothing changed |
| 07:25.20 | IsUp | WIMPy: "Stopping T203 counter since we got an ACK", "Restarting T203 timer" |
| 07:25.27 | WIMPy | Err, where exactely do you have bad voice? |
| 07:26.20 | WIMPy | Nothing interesting about that. Timers get started, reset and stopped all the time. It's only when they expire, you should have a closer look. |
| 07:26.31 | IsUp | WIMPy: all DAHDI channels which connected to my GSM gateways. (not fxo or fxs) |
| 07:27.02 | WIMPy | But only some of them? |
| 07:27.34 | IsUp | WIMPy: also i am sure theres no problem with my GSM gateways, lets says my first GSM gateway is using channels between 32-62, but i only have problem on 38-48 |
| 07:27.57 | IsUp | WIMPy: ive never changed any configuration, even dialplan. system was stable and fine |
| 07:28.29 | IsUp | WIMPy: but as i said, our telco link gone 3 days ago, just for 5 mins. when link comes back, problems started |
| 07:28.59 | WIMPy | If you only have issues on some channels of an interface, that sounds like some driver fuckup to me. |
| 07:29.20 | WIMPy | Did you do the master reset? |
| 07:29.39 | WIMPy | AKA "Have you tried turning it off and on again?" |
| 07:30.09 | IsUp | WIMPy: yes i did reboot, but i am really sure this is not a driver problem |
| 07:30.30 | IsUp | WIMPy: also this is my production server, i cant reboot and test thing everytime |
| 07:30.56 | WIMPy | Just reboot or power cycle? |
| 07:31.00 | irroot | WIMPy IsUp GSM E1 channel banks are E1 upto the system internally they relay on a MUX to the GSM module and are affected by |
| 07:31.02 | irroot | many things ... some of these MUX also have management software one i know of runs a gui |
| 07:31.20 | WIMPy | And did you do it to your Asterisk only or to the gateway as well? |
| 07:32.10 | IsUp | WIMPy: i did power on/off, also i did it on my broadband modem (telco line is connected to it), and all gsm gateways |
| 07:32.42 | WIMPy | Sounds evil. |
| 07:32.54 | WIMPy | Especially if you can't test. |
| 07:33.12 | IsUp | WIMPy: also i have an interface for my GSM gateways, it allows layer 1 2 3 tracing. ive compared calls fine call and bad call. |
| 07:33.27 | WIMPy | I'd try to disconnect the telco to see if that affects the communication to the GWs. |
| 07:33.34 | IsUp | WIMPy: i dont know much about ISDN but at least i can see ACK, and everything seems normal |
| 07:34.09 | IsUp | WIMPy: my pbx is providing clock to gsm gateways, and my pbx is getting clock from telco |
| 07:34.18 | IsUp | WIMPy: so its very complicated |
| 07:34.30 | WIMPy | Bad timing most likely results in the famous HDLC aborts. |
| 07:35.06 | WIMPy | Can you generate timing for the interfaces to your GWs instead of passing it on from the 1st interface? |
| 07:35.54 | IsUp | WIMPy: i can generate but not with hardware, i think i have to use dahdidummy right? |
| 07:36.38 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 07:36.39 | WIMPy | No that would be in the card configuration. But I don't know the Sangomas. |
| 07:37.02 | IsUp | WIMPy: yes i can generate timing |
| 07:37.37 | WIMPy | Then you could try to run the ports asynchronously. |
| 07:37.58 | IsUp | WIMPy: also i can send pcap files and pri debug output if needed, i can provide SSH too if you want |
| 07:39.27 | WIMPy | As I said: I'm not familiar with the Sangoma hardware and drivers. |
| 07:39.59 | WIMPy | But bad voice only on vertain channels is pretty strange. |
| 07:40.24 | IsUp | WIMPy: do you think that bad timing cause it? |
| 07:40.31 | IsUp | WIMPy: and is there any way to test timing? |
| 07:40.49 | WIMPy | Are teh GWs configured correctly to accept clock from your Asterisk? |
| 07:40.49 | F2Knight | Q: I got a macro I am working on .. goes like this.. inbound-did call macro, macro does it things to check how to deliver call, but ... it keeps looping. like the macros end, and then get called again. |
| 07:41.18 | IsUp | WIMPy: yes and consider that system was stable 3 days ago, not even a single line changed |
| 07:41.49 | IsUp | WIMPy: we were using this configuration well, Sangoma techs helped me on setup too, it was stable since 4 months |
| 07:41.59 | WIMPy | Well you wouldn;t be the first inhere fo whom it worked most of the time. |
| 07:42.30 | WIMPy | Do you get bad voice in both directions? |
| 07:43.02 | WIMPy | Bad typing :-( |
| 07:43.13 | IsUp | WIMPy: i think bad voice is only from GW side, i did dahdi_monitor 40 -vvv for example, ive checked RX/TX levels |
| 07:44.32 | WIMPy | You have 3 GWs? |
| 07:44.47 | WIMPy | Do you have some bad channels on all of them? |
| 07:46.04 | IsUp | WIMPy: yes i have 3x 2N Stargate GWs |
| 07:47.14 | IsUp | WIMPy: no all channels are ok, i mean hw/sw |
| 07:47.32 | WIMPy | channels with bad voice, |
| 07:48.01 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:48.11 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:48.39 | IsUp | WIMPy: I only have bad voice on GW 1 and GW 3, GW 2 seems ok |
| 07:51.24 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 07:51.50 | IsUp | WIMPy: 1 sec, ill post pcap file |
| 07:52.24 | WIMPy | I think the only way forward would be a software loopback. But I have no clue what Sangoma provides or what the 2N can do. |
| 07:53.39 | IsUp | WIMPy: http://imageshack.us/photo/my-images/268/pcap.png/ |
| 07:54.28 | *** join/#asterisk pietro (~pietro@88-149-226-183.dynamic.ngi.it) |
| 07:55.11 | WIMPy | That doe not look good. But I have no idea, how reliable pcap capture or wireshark are for that purpose. Anything similar in the intense debug? |
| 07:56.11 | WIMPy | The interesting part is that these malformed packets don't even have a direction. |
| 07:59.13 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:06.07 | *** join/#asterisk polysics (~polysics@host210-142-static.228-95-b.business.telecomitalia.it) |
| 08:06.11 | polysics | hello |
| 08:06.20 | polysics | at it again with in-call DTMF stuff |
| 08:06.29 | WIMPy | Damn. Have to get up in 90 Minutes and haven't slept yet. Try to get a quick nap. |
| 08:06.37 | polysics | there was an "info" setting someone mentioned i didn't' get the chance to try |
| 08:06.47 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 08:06.52 | schmidts | good morning |
| 08:08.15 | polysics | do i need to set the dtmf mode using SIPDtmf_Mode application? |
| 08:08.42 | polysics | or can i set it in the sip peer config? |
| 08:11.57 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 08:14.11 | *** join/#asterisk Nasga (~Nasga@78.117.113.110) |
| 08:19.42 | *** join/#asterisk kriegerod (~krieger@chaf.tenet.odessa.ua) |
| 08:20.50 | Nasga | Hello, does anyone know a nice way to block some caller_id |
| 08:21.05 | Nasga | i would like to manage them in a text file |
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| 08:28.51 | ChannelZ | Nasga: Usually best to just write an AGI to do it |
| 08:29.43 | Nasga | ChannelZ: agi is better than a bash script ? |
| 08:30.18 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:31.12 | ChannelZ | well, an AGI can be a bash script. |
| 08:32.09 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 08:32.16 | ChannelZ | It's a simple 'protocol', when Asterisk calls the AGI it spits out a bunch of variables to its stdin, and you can do what you want and spit some things back to Asterisk if needed via stdout. |
| 08:32.33 | irroot | AGI can also be a socket connex to a bash script even |
| 08:33.26 | Nasga | i was thinking about using "System" in my dialplan |
| 08:33.32 | Nasga | AGI is far better ^^ |
| 08:33.36 | Nasga | thanks for the help |
| 08:35.38 | *** join/#asterisk polysics (~polysics@host210-142-static.228-95-b.business.telecomitalia.it) |
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| 08:42.08 | polysics | ok |
| 08:42.15 | polysics | how do i get DTMF events during a call? |
| 08:42.31 | polysics | i don't really care about features running a macro for me |
| 08:42.45 | polysics | i would just like to see DTMF pressed by users during a call |
| 08:47.30 | ChannelZ | polysics: Read() |
| 08:47.54 | ChannelZ | or are you meaning on a debug level? |
| 08:48.31 | polysics | let me present the full use case |
| 08:48.36 | polysics | user A calls user B |
| 08:48.45 | polysics | then user A or user B want to bring user C into the call |
| 08:48.55 | polysics | they both only have cell phones |
| 08:49.01 | polysics | so it has to be done by DTMF |
| 08:49.13 | polysics | i am stuck at the "detect DTMF in call" step |
| 08:49.28 | polysics | as i am not seeing the DTMF tones anywhere |
| 08:49.36 | polysics | DTMF works when in the IVR |
| 08:49.45 | polysics | so something might be wrong at a fundamental level |
| 08:50.03 | polysics | i then need to figure out how to join the three in a conversation but that's another thing :-) |
| 08:52.30 | ChannelZ | hmm |
| 08:53.40 | polysics | is it even feasible? |
| 08:53.50 | polysics | i suppose so, it doesn't look too complicated |
| 08:54.10 | polysics | what i know for sure is that pressing DTMF on calls right now results in nothing |
| 08:54.43 | ChannelZ | yea when a call is bridged nothing is really happening that isn't built into Dial |
| 08:56.00 | polysics | i suppose i need some features.conf magic |
| 08:56.08 | polysics | although i have no idea WHICH magic :-) |
| 08:56.31 | ChannelZ | I'm not sure if you can get DTMF events via AMI which would be the only way I can think of to make your own arbitrary commands |
| 08:57.43 | ChannelZ | actually there are 'dynamic features' which I've never used |
| 08:58.09 | polysics | the book is very clear about no DTMF events in bridged calls |
| 08:58.12 | ChannelZ | see the [applicationmap] section of the sample features.conf |
| 08:59.24 | ChannelZ | though what you can do is a bit limited probably for what you'd need to do. But I'm going to bed. |
| 09:00.04 | polysics | i could call a macro going into AGI |
| 09:00.12 | polysics | cumbersome but might be the only way |
| 09:00.19 | irroot | ChannelZ extern ivr ?? |
| 09:00.28 | ChannelZ | well but I don't think you can call a macro |
| 09:00.29 | irroot | have a good down time |
| 09:01.44 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 09:01.48 | polysics | you can, looking at the Cookbook |
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| 09:20.35 | polysics | hmm, dynamic features are not working for me |
| 09:20.48 | polysics | i tried a very simple example just doing playback |
| 09:23.21 | polysics | umm , think i found what's wrong |
| 09:23.43 | polysics | no DTMF can work in a bridged call if Asterisk is not in the media flow |
| 09:24.08 | polysics | but that would break our carefully crafted tunnel setup |
| 09:24.11 | polysics | argh |
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| 10:19.54 | *** join/#asterisk kashu (7ab3a0c8@gateway/web/freenode/ip.122.179.160.200) |
| 10:20.21 | kashu | hello |
| 10:21.01 | kashu | please give me details to video call |
| 10:23.14 | kashu | how to config confbridge.conf |
| 10:25.39 | *** join/#asterisk crioto (~crioto@109.201.168.183) |
| 10:25.45 | kashu | please how to video call possible asterisk 1.10 |
| 10:26.19 | crioto | I have an extension, e.g. for button 1, but when i press it on my mobile - nothing happens. On sip phone everything works well |
| 10:26.34 | kashu | hello crioto |
| 10:26.51 | crioto | kashu, hello, sorry |
| 10:26.55 | crioto | Hi everyone! |
| 10:28.07 | kashu | hey check dial plane extension.conf |
| 10:29.59 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 10:31.06 | crioto | Everything works well with soft phones, but not with real mobile cellphones |
| 10:31.30 | irroot | crioto real mobile phones ?? dect / wifi / ??? |
| 10:31.54 | kashu | wifi |
| 10:31.58 | crioto | irroot, not a wifi. GSM phone |
| 10:32.57 | kashu | video conference is possible asterisk1.10? |
| 10:33.44 | crioto | Is the key presses related to DTMF or i misunderstood something? |
| 10:37.53 | *** part/#asterisk the_5th_wheel (~edd@tcs-gw.bulwer.thusa.net) |
| 10:38.09 | irroot | crioto ok with you but now you calling into the system from mobile ie you dialing the number "outside" you not inside on a sip phone any more |
| 10:38.19 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 10:38.30 | irroot | kashu yes limited video conf support is in ConfBridge |
| 10:39.02 | as001 | Hi is it possible in Asterisk to transfer call from Queue to other extension (for play back and recording purposes) and then to transfer it back to the same Agent in Queue? |
| 10:39.38 | irroot | as001 explain play back / recording |
| 10:39.42 | crioto | irrot what should i do then? I'm making simple IVR system and i really need to deal with all the phone types. |
| 10:40.31 | irroot | crioto the connection you coming in on is not configured right for DTMF |
| 10:40.57 | as001 | call comes in queue and agent gets it. Agent talk and at the end of call agent press button and call is transfer to other extension where i plan to playback message "Do you agree on ...." and then Record what client will say then playback other message then record again etc... and at the end to retransfer call to agent who transfered it |
| 10:42.29 | irroot | as001 i see not as simple as it seems |
| 10:42.49 | as001 | yes that is not simple at all it seems impossible to me. |
| 10:44.46 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 10:45.27 | kashu | please help about bridge.conf how to configure |
| 10:45.36 | irroot | as001 not impossible there a couple things to take into account the agent transfering call needs to be on a "wrapuptime" so as not to get a call in this gap |
| 10:46.19 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:154c:5fcc:8c3c:5867) |
| 10:46.26 | as001 | ok I will pause agent from manger when it press that button and i will also transfer call to other extension but how to bring that call back to that agent after ? |
| 10:46.31 | *** part/#asterisk as001 (~uros@82.117.198.142) |
| 10:47.12 | irroot | variables are set on the channel by app_queue |
| 10:47.43 | irroot | the agent if they do a attended transfer there info will be registered in the dialplan |
| 10:48.32 | irroot | then when complete use the "Transfer" to get them back |
| 10:52.39 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 10:52.57 | as001 | sorry i crashed.. |
| 10:53.15 | irroot | variables are set on the channel by app_queue |
| 10:53.17 | irroot | the agent if they do a attended transfer there info will be registered in the dialplan 12:47:42 |
| 10:53.19 | irroot | then when complete use the "Transfer" to get them back |
| 10:53.44 | as001 | ok thanks |
| 10:53.46 | *** part/#asterisk as001 (~uros@82.117.198.142) |
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| 11:11.02 | kashu | how to config confbridge.conf |
| 11:12.59 | *** join/#asterisk qakhan (~qakhan@182.185.249.74) |
| 11:13.03 | qakhan | hi all |
| 11:14.28 | qakhan | i have 4 queues and 6 agents every queue. i want to pause an agent in all queues. plz help me in this regard |
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| 11:31.17 | qakhan | anyone can help me? |
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| 12:18.33 | kashu | asterisk 1.10 conbridge.conf how to config? |
| 12:22.14 | *** join/#asterisk ccesario (~ccesario@189.29.62.245) |
| 12:22.36 | pabelanger | kashu: check out the sample, it will have examples and documentation |
| 12:27.14 | qakhan | i have 4 queues and 6 agents every queue. i want to pause an agent in all queues. plz help me in this regard |
| 12:29.28 | pabelanger | qakhan: *CLI> core show application PauseQueueMember |
| 12:32.44 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 12:33.23 | nunne | Anyone have any experience with asterisk 1.4 and misdn? I cant quite get it to work (trying to connect it to a panasonic pbx via BRI) |
| 12:34.10 | nunne | trying to configure it as NT-PTP i cant get it up. But if i configure misdn to be NT-PMP and the PBX to PMP I can get L1Link: UP, but L2Link: DOWN |
| 12:34.33 | irroot | nunne what kernel ?? |
| 12:34.40 | qakhan | <@pabelanger> i want to pause agent thourgh my application |
| 12:35.04 | irroot | qakhan hi there you can do it via AMI |
| 12:35.04 | nunne | 2.6.28.10 |
| 12:35.21 | nunne | is an embeded system (thats why im running asterisk 1.4) |
| 12:35.34 | nunne | and misdn isnt really the newest version as well, but i know it supposed to work :( |
| 12:35.40 | qakhan | irroot can u give me more detail how? |
| 12:35.42 | irroot | nunne you need to have mISDN v1 compiled into the kernel |
| 12:36.11 | irroot | it changed at some point in the 2.6.2X |
| 12:36.27 | nunne | irroot: i have it in kerne |
| 12:36.28 | nunne | l |
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| 12:37.55 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:38.41 | nunne | irroot: or no, i dont have it IN kernel. actually as modules.. but its based on switchfin latest stable revision with BRI support. |
| 12:38.48 | nunne | so misdn |
| 12:38.55 | nunne | *should* work :) |
| 12:39.15 | nunne | :Q |
| 12:39.16 | qakhan | <PROTECTED> |
| 12:39.18 | irroot | nunne but only mISDN v1 will work with chan_misdn |
| 12:39.43 | irroot | http://www.voip-info.org/wiki/view/Asterisk+manager+API qakhan |
| 12:40.09 | irroot | http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+QueuePause |
| 12:41.15 | nunne | irroot: im not on the computer where i have the sources. but im pretty sure it's the v1-branch.. for example i have the old configuration file etc. not the new xml-based one |
| 12:42.02 | nunne | <6>Modular ISDN Stack core version (1_2_0) revision ($Revision: 1.40 $) |
| 12:43.38 | irroot | ok so it should be ok nunne now what type of interface USB/*PCI |
| 12:43.46 | irroot | lspci / lsusb |
| 12:44.08 | irroot | and make sure its loaded right with right options |
| 12:44.15 | nunne | its embeded, its via SPI bus |
| 12:44.48 | nunne | http://pastebin.com/unaTfSk8 |
| 12:44.53 | irroot | nunne and mISDN supports it ? |
| 12:45.13 | nunne | I have gotten it to work once before.. But Im not sure why im not getting it to work now :( |
| 12:45.26 | nunne | that is from dmesg |
| 12:45.36 | pabelanger | qakhan: then use the manager interface |
| 12:48.13 | irroot | nunne you have it |
| 12:48.30 | irroot | now to set up NT mode |
| 12:48.45 | irroot | should have misportinfo ? |
| 12:50.14 | nunne | http://pastebin.com/nJ7q3yu5 |
| 12:50.30 | nunne | (im only trying to use port 1-2) |
| 12:51.44 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 12:51.55 | nunne | hmm, now i got it to go up on PMP.. i tested removing some jumpers.. but last time i thought i had to put the jumpers on.. but obviously i must have been misstaken. |
| 12:52.01 | irroot | nunne looks like asterisk is running and all good |
| 12:52.17 | nunne | but would be nice to know why it doesnt want to work in PTP.. since it's PTP PBX usually want to work in |
| 12:52.24 | *** join/#asterisk slidesinger-lt (~jtatum@c-174-57-5-70.hsd1.nj.comcast.net) |
| 12:52.29 | irroot | nunne need to terminate on 100ohm resistors and use cross over cable |
| 12:52.46 | nunne | irroot: problem has been that misdn show stacks show it as offline |
| 12:52.56 | irroot | nunne its PMP PTP is a leased line |
| 12:53.13 | nunne | crossover i am using.. and the 100ohm is built in.. but obviously i should have removed the jumpers to use the 100ohm.. i was thinking in reverse it seems! :( |
| 12:53.41 | nunne | irroot: i have put one of the PBX bri lines to PTP. would be nice to just see that its working :P |
| 12:55.07 | irroot | its offical i dont like users.conf BLEGH |
| 12:56.15 | nunne | irroot: why even use users.conf? :) |
| 12:57.39 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 12:57.53 | irroot | nunne just came across a site using it first time i have been exposed to iy |
| 12:58.33 | mandla | d |
| 12:58.47 | irroot | mandla you know who is guilty :P |
| 13:00.14 | nunne | i dont like things that create mailboxes and what not on the fly for you.. like users.conf tend to do :P |
| 13:00.48 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 13:02.07 | irroot | nunne or extensions in a context bit messy .... |
| 13:03.54 | qakhan | irroot i am using Astrisk .net library to integrate my application with asterisk. i have a queue and 6 agents in that queue. i want that agent can transfer call to other agent or to supervisor. calls can be transfer and CLI number also. but caller data is not transfering to other agent or supervisor |
| 13:04.08 | qakhan | plz help |
| 13:13.11 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 13:13.47 | kashu | hello all of you |
| 13:14.51 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:15.04 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:15.35 | kashu | video call with asterisk 1.10 how to work? |
| 13:21.16 | *** join/#asterisk SeRi (~seriosly@c-76-31-169-54.hsd1.tx.comcast.net) |
| 13:21.37 | qakhan | irroot u there? |
| 13:22.12 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 13:23.07 | [TK]D-Fender | kashu, http://www.voip-info.org/wiki/view/Asterisk+video |
| 13:29.29 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 13:29.42 | fprior | Hi all: how to offer to my customer the possibility to add new extension, or manage other littles changes in * pbx ? |
| 13:29.53 | *** join/#asterisk mandla (~quassel@168.167.180.161) |
| 13:30.54 | *** join/#asterisk bchia (~Adium@nat/digium/x-yvyilbomdhemrwgg) |
| 13:31.35 | mandla | irroot: you there?? |
| 13:31.56 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-nyuybjmesrslpyrt) |
| 13:32.53 | [TK]D-Fender | fprior, Go make them a GUI to bluid what limited things you wnt them to be able to build.... or install one of the bolt-on ones that already exist. |
| 13:33.20 | [TK]D-Fender | fprior, However you will be having to redo your configurations pretty much from scratchw hen you do. |
| 13:34.37 | [TK]D-Fender | fprior, FreePBX is the most popular free one. IIRC there is a limited release of Switchvox available, and ther is AsteriskGUI, but that one Is still not really "mature" and has a very small user & support base |
| 13:40.19 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:40.36 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 13:40.39 | irroot | fprior [TK]D-Fender realtime with a simple php or similar interface is quite quick and easy |
| 13:41.34 | [TK]D-Fender | Realtime is just an "end result dump option" really... Could generate flat configs like the others. Depends if you care about using a DB for this. |
| 13:43.10 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 13:45.16 | fprior | [TK]D-Fender, in your opinion, freePbx dirty the dialplan with many other extensions ? is it safe ? |
| 13:45.52 | [TK]D-Fender | fprior, No, you won't be keeping your dialplan with FreePBX. You will have to restart using their concepts. |
| 13:46.49 | irroot | fprior you will be boxed into what there "skin" offers and forget about doing something custom its a nightmare |
| 13:48.05 | [TK]D-Fender | well ... lets not blanket all "custom" as being a nightmare. We'd have to see what you want to be able to do in addition to what their interface already allows you to do. |
| 13:50.22 | fprior | [TK]D-Fender, irroot, my doubt is which is the correct/sane/professional strategy to work with asterisk dialplan, "manual" dialplan or freePbx ? |
| 13:55.11 | [TK]D-Fender | fprior, Either. |
| 13:55.22 | [TK]D-Fender | fprior, Just not a lot of both combined |
| 13:55.54 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:58.34 | *** join/#asterisk bchia (~Adium@nat/digium/x-yfrqdjioalqlalas) |
| 14:00.00 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 14:04.35 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-auqxzfmmgwaxvdxu) |
| 14:06.10 | fprior | [TK]D-Fender, thanks. And howto install freepbx ? is better manually (installing tarball, apache, php, mysql, etc) o using freePbx distro ? [FreePbx distro == Trixbox & Co ?] |
| 14:06.41 | [TK]D-Fender | fprior, Depends what you're comfortable with and what else you server will be doing for you, etc |
| 14:07.04 | [TK]D-Fender | Either works fine |
| 14:09.13 | fprior | [TK]D-Fender, thank you |
| 14:10.55 | fprior | someone here will go to 4kconf.com ? |
| 14:13.26 | *** join/#asterisk mocker (~mocker@72.165.148.230) |
| 14:14.47 | carrar | hahah |
| 14:14.48 | carrar | http://www.amazon.com/Defense-Technology-56895-Stream-Pepper/dp/B0058EOAUE/ref=cm_cmu_pg_t |
| 14:14.51 | carrar | read the reviews |
| 14:17.38 | Kobaz | haha, someone walked by the cube i'm sitting at and said "do you have enough phones?" |
| 14:17.56 | carrar | one can never ave enough phones |
| 14:18.00 | carrar | have |
| 14:18.21 | carrar | It's best if they are all different models too |
| 14:18.29 | carrar | one for every mood |
| 14:19.11 | Kobaz | yeah |
| 14:19.26 | Kobaz | i've got two 331s two 330s, a 650, a 650 with a sidecar |
| 14:19.51 | [TK]D-Fender | carrar, GOLD |
| 14:19.52 | Kobaz | developing some stuff, sitting at a customer site |
| 14:19.52 | carrar | no 4xx series? |
| 14:19.54 | Kobaz | always fun |
| 14:20.11 | Kobaz | i try to avoid that, but... i'm here, i have some phones |
| 14:20.53 | [TK]D-Fender | I've got about a12+ extra Polycom phones in 2 boxes behind me due to downsizing.... |
| 14:21.02 | [TK]D-Fender | Mostl IP600's |
| 14:21.04 | carrar | PICS!! |
| 14:21.34 | carrar | wanna sel em? |
| 14:21.37 | carrar | sell |
| 14:21.46 | [TK]D-Fender | Hardly worth it.... |
| 14:22.07 | carrar | 601's? |
| 14:22.11 | carrar | or 600 |
| 14:22.19 | [TK]D-Fender | Used market price might be 100$ for them. And well.. I don't really care that much. |
| 14:22.24 | [TK]D-Fender | 600's, no 601's |
| 14:22.28 | carrar | oh |
| 14:22.51 | [TK]D-Fender | 2-3 IP 430's, and a 320 |
| 14:23.11 | [TK]D-Fender | I also have 2 Uniden UIP-200's which I'm glad I pulled first. |
| 14:23.18 | [TK]D-Fender | Those were crap |
| 14:26.58 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 14:30.14 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 14:37.26 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 14:38.51 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-dulwsatsnldsvufg) |
| 14:39.18 | Kobaz | so that seems bad |
| 14:40.04 | *** join/#asterisk rgsteele (~rgsteele@173-15-180-105-BusName-Philadelphia.hfc.comcastbusiness.net) |
| 14:42.11 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 14:42.18 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 14:43.31 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 14:45.58 | *** join/#asterisk dubcl (~dub@190.196.69.194) |
| 14:49.05 | dubcl | hi, i try to redirect a no answer call from a specific extension (ex. from 8199 to 8150), i search info for that but i dont find anything, any idea/clue? |
| 14:50.02 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 14:50.39 | [TK]D-Fender | dubcl, Please rephrase your question... |
| 14:50.40 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.132.191) |
| 14:55.59 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 14:56.22 | *** join/#asterisk MrTelephone (~MrTelepho@h697179-171.picriverisp.net) |
| 14:58.51 | MrTelephone | has anyone ever had crosstalk on their digital t1 card? I'm getting the odd report about it and I experienced it myself a couple times. We are talking 100-500ms of conversation from a random timeslot other than the one that is being opened. Could this be inconsistencies with the pci bus/t1 card transmissions? |
| 14:59.25 | nunne | irroot: now my misdn is *working* but i hear a stutter/scramble on the line maybe 3-4 times per second. you have any idea what this might be? I have tried enabling echocancel and jitterbuffer to no effect :( |
| 15:00.22 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 15:00.34 | FlashDeluxe | hi! does anybody know how i can compile a old chan_capi (e.g. .0.6.3)? |
| 15:00.44 | irroot | nunne sorry have not seen this before maybe io load ?? |
| 15:01.13 | irroot | FlashDeluxe with a compiler :P will need the right version possibly 1.4/1.2 |
| 15:01.34 | nunne | its the "only" call on the pbx :( hmm.. i have to play some more with this i guess :P |
| 15:02.03 | nunne | or could it possibly be my kernel timing? but then i guess misdn wouldnt work at all? |
| 15:03.06 | MrTelephone | You have to be an electrical engineer to figure out any of this stuff. |
| 15:03.17 | irroot | nunne what is your "HZ" setting in kernel i remember time past if it is not high it causes strange problesm |
| 15:03.21 | MrTelephone | When you pay 1400 for a dual port t1 card you expect it to work flawlessly though |
| 15:04.01 | MrTelephone | And when you spend less on your server than your t1 card that can probably be the root cause. lol |
| 15:04.18 | MrTelephone | is away ordering a new server |
| 15:04.18 | carrar | haha |
| 15:04.24 | irroot | MrTelephone i use them all the time installed 2x4 + 2x2 port cards at a university last week |
| 15:04.43 | irroot | aint seen "crosstalk" |
| 15:05.17 | irroot | unless its a mux you connect to with analogue/GSM on the other side but that is not likely |
| 15:05.20 | carrar | Get a supermicro server or a HP |
| 15:05.20 | MrTelephone | it's not really crosstalk. I think the card is messed up or the pci bus is messed up. |
| 15:05.46 | MrTelephone | It's sampling a few slices of audio from the wrong "memory address" or something |
| 15:05.57 | nunne | irroot: i have to check, i dont remember actually. and it resides on a differnet computer than i am at now :P but thanks for the tip! |
| 15:06.08 | irroot | carrar LOL if HP is inbussiness soon after the hit apothetker commited |
| 15:06.19 | MrTelephone | I'm an amatuer program but during call setup it's like its reading audio from a buffer that wasn't emptied that was used for another channel? |
| 15:06.33 | carrar | older HP's work great |
| 15:06.40 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:06.42 | MrTelephone | what about newer stuff? |
| 15:06.43 | carrar | we have a lot older machines working flawlessly |
| 15:06.59 | carrar | newer stuff is over priced for HP |
| 15:07.01 | carrar | or IBM |
| 15:07.05 | MrTelephone | I like dual xeons. Could that be a problem? |
| 15:07.07 | carrar | go supermicro |
| 15:07.14 | MrTelephone | I like asus |
| 15:07.14 | carrar | DP Xeon series |
| 15:07.46 | carrar | Because you know you need 24 cores in your asterisk server :) |
| 15:07.53 | MrTelephone | I had some riser cards in some of these servers before and always had issues |
| 15:08.44 | MrTelephone | I think my network cards are stressing out more than the cpu though |
| 15:09.53 | MrTelephone | I haven't heard of anyone else having crosstalk on digital cards. I have to attribute this to bad pci bus driver or something |
| 15:10.08 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 15:10.32 | MrTelephone | Hey does anyone use arris tmg502/602 modems? |
| 15:12.02 | FlashDeluxe | hi! are there any chan_capi people here? i got a problem using asterisk 1.6.2 with a gerdes card and chan_capi trunk. If i want to load chan_capi.so i get a segmention fault. looks like chan_capi sends a correct LISTEN_REQ on controller1 but after that it says that the controller number is 0 and not 1. Can somebody help me please? |
| 15:13.51 | MrTelephone | Arris modems have a feature *65 that is called "calleridpermdisable". There is no supported feature called "calleridpermenable". The only way to display your callerid is to reboot the device. Is that odd? |
| 15:14.20 | *** join/#asterisk ccesario (~ccesario@189.29.62.245) |
| 15:15.06 | r0m|u | MrTelephone, is that a ata/modem? |
| 15:17.40 | MrTelephone | yeah |
| 15:17.45 | MrTelephone | modem |
| 15:18.27 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:18.27 | puzzled | FlashDeluxe: I don't think you need chan_capi trunk for 1.6. Have you tried the latest regular release? |
| 15:18.29 | leifmadsen | for anyone interested, I just updated the asteriskdocs.org site to show the latest 3rd edition of Asterisk: The Definitive. Should be a bit faster than the OFPS links. |
| 15:19.23 | MrTelephone | Too bad my boss was an asshole and didn't want me to goto astricon |
| 15:19.48 | MrTelephone | I'm going on strike next week |
| 15:20.17 | leifmadsen | s/Definitive/Definitive Guide/ |
| 15:21.37 | puzzled | leifmadsen: thanks, loads pretty fast |
| 15:25.48 | tompaw | Hey guys, I finally managed to build dahdi from source (non-stock kernel). With 1.8, what do I need to do now to use MeetMe? Do I have to modprobe dahdi-something? |
| 15:28.05 | kaldemar | tompaw: install it and modprobe dahdi. the core module has the timer (dummy) in it nowadays. |
| 15:28.24 | FlashDeluxe | puzzled: yes, that doesn`t work either |
| 15:28.44 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 15:28.46 | tompaw | kaldemar: thanks. |
| 15:29.14 | FlashDeluxe | puzzled: i guess there is a general problem between the card driver from gerdes and chan_capi |
| 15:31.08 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:32.40 | dubcl | [TK]D-Fender, how to redirect a no answer call from one extension to a specific extension? |
| 15:34.29 | puzzled | FlashDeluxe: so it seems. iirc there is a mailing list on the chan_capi site. maybe ask there |
| 15:41.33 | *** join/#asterisk AmirBehzad (~behzad@86.57.4.72) |
| 15:44.27 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.132.191) |
| 15:46.25 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 15:46.56 | Katty | you know what i hate. |
| 15:47.01 | Katty | when your irssi windows are out of order. |
| 15:47.21 | Katty | this room is supposed to be alt 4, not alt 3 >.< |
| 15:48.34 | Katty | also, hi :> |
| 15:49.22 | Qwell | yes |
| 15:49.31 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:49.31 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:49.31 | Qwell | xchat is stupid. it gets worse when you have like 14 windows |
| 15:49.54 | Katty | hai Qwell! |
| 15:49.57 | Katty | hugs Qwell to bits. |
| 15:50.05 | Katty | sweeps up bits, puts Qwell back together. |
| 15:50.08 | Qwell | my bits! |
| 15:50.17 | Katty | 4 bits in a byte. |
| 15:50.23 | Katty | or was it 4 bits is a nibble |
| 15:50.25 | Katty | and 4 nibbles is a byte |
| 15:50.38 | Kobaz | 4 bits in a nibble |
| 15:50.58 | sysreq | hi everyone. has anyone played with distributed device states via xmpp using tigase? i got it working, but i was wondering if anyone had been able to disable event publishing to the originating node. server1 sends out a change of state to tigase, then tigase publishes that change to both server1 and server2. |
| 15:51.02 | Katty | it is soooo adorable that 4 bits are in a nibble. |
| 15:52.06 | sysreq | i don't feel it's very efficient.. server1 obviously disregards these state changes, but still has to process them (i keep getting these notices: "res_jabber.c:3259 aji_handle_pubsub_event: Returning here, eid of incoming event matches ours!") |
| 15:54.40 | Katty | what's the little symbol for delete |
| 15:55.29 | [TK]D-Fender | <dubcl> [TK]D-Fender, how to redirect a no answer call from one extension to a specific extension? <- its your dialplan. If they don't answer, Dial something else |
| 15:56.21 | Katty | it's like.. ^D or something |
| 15:56.34 | Katty | when you hit delete or backspace |
| 15:56.36 | Katty | but it doesn't do it |
| 15:56.42 | Katty | it sticks those characters in |
| 15:56.52 | Katty | ...yes, i realize i sound like a complete looney |
| 15:57.15 | dubcl | [TK]D-Fender, but i need redirect the incoming calls of one extension only, on DND and no connect extension |
| 15:57.54 | [TK]D-Fender | dubcl, Its your dialplan. just dial the failover after trying the main |
| 16:01.53 | Katty | Qwell: Qwell |
| 16:02.03 | Qwell | Katty: Katty: Katty |
| 16:02.04 | Katty | Qwell: what's the symbol the terminal throws back at you when you hit delete |
| 16:02.09 | Katty | Qwell: but you can't |
| 16:02.09 | Qwell | ^H? |
| 16:02.13 | Katty | Qwell: it's like ^something |
| 16:02.13 | Katty | ty |
| 16:03.17 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:03.23 | Katty | i can always depend on Qwell for the smrts. |
| 16:03.29 | Katty | ^- and refreshing my memory |
| 16:03.33 | Qwell | on other stuffs |
| 16:03.43 | Qwell | if you know what I mean. I mean cookies. |
| 16:03.57 | Katty | i make a mean cookie. |
| 16:04.00 | Katty | but meaner cupcakes. |
| 16:04.36 | Qwell | err, s/on/or/ |
| 16:05.51 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 16:06.11 | wcselby | o/ |
| 16:07.13 | Katty | it's a wcselbyyy!!!!! |
| 16:10.51 | *** join/#asterisk celord (~celord@201.195.243.194) |
| 16:11.01 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 16:11.11 | *** join/#asterisk casix (~casix@xenpbxedifici.adamvozip.es) |
| 16:11.18 | casix | hello |
| 16:11.50 | tompaw | Guys, every 2-3 hours my 1.8 gets "stuck", it stops responding to SIP requests, when I do 'core stop now' it's just happily ignoring my request... |
| 16:11.58 | tompaw | what's funny is that "reload" still works |
| 16:12.06 | tompaw | wtf is going on? |
| 16:12.45 | tompaw | I thought it might be grsec problem, but I disabled grsec and its happening again :/ |
| 16:13.28 | casix | I have a problem with a hangup. When I make hangup(1) asterisk makes a "503 Service Unavailable" response and not "404 Not found" as it says in the documentation. How can I produce a 404 not found response? |
| 16:14.24 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 16:15.02 | tompaw | It's 1.8.7.1... latest version, not much I can do there... |
| 16:15.52 | *** join/#asterisk godmachine-x6 (~poorelanc@funtoo/user/godmachine-x6) |
| 16:16.32 | casix | I have tested this with an asterisk 1.4.42 and 1.4.23 |
| 16:18.46 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:18.46 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 16:18.51 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:20.58 | wcselby | o/ Katty |
| 16:21.18 | Katty | hugs on wcselby |
| 16:21.51 | sysreq | tompaw: looks like a deadlock.. you'd need the output of a 'core show locks' (which you can only get if you compile Asterisk with DEBUG_THREADS). |
| 16:25.43 | wcselby | casix we need to see a SIP debug of the hangup message that's being sent, along with the CLI output of the call |
| 16:25.45 | wcselby | use pastebin |
| 16:25.46 | wcselby | ~pb |
| 16:25.47 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 16:26.10 | wcselby | so Katty - my wife is getting ready to have a baby, any day now |
| 16:26.22 | wcselby | she called me on my way in to work to say she'd lost her plug, which means we're down to less than two days |
| 16:26.38 | [TK]D-Fender | <casix> I have a problem with a hangup. When I make hangup(1) asterisk makes a "503 Service Unavailable" response and not "404 Not found" as it says in the documentation. How can I produce a 404 not found response? <- you can't |
| 16:26.41 | sysreq | tompaw: i would also get a backtrace of the running process when it occurs, using gdb as such: gdb -ex "bt" -ex "bt full" -ex "thread apply all bt" --batch /usr/sbin/asterisk `pidof asterisk` > /tmp/backtrace.txt |
| 16:29.51 | casix | [TK]D-Fender: do you mean I can't choose witch response makes asterisk in a hangup? |
| 16:30.18 | casix | It allways response 503? |
| 16:36.11 | [TK]D-Fender | casix, 503 is what happens if it hits the dialplan and never gets an answer |
| 16:37.54 | *** join/#asterisk zyphlar (~zyphlar@wsip-68-14-229-148.ph.ph.cox.net) |
| 16:37.58 | r0m|u | waz up wcselby.... Raining down there? |
| 16:38.17 | r0m|u | wcselby, congrats! |
| 16:38.40 | wcselby | thank r0m|u |
| 16:38.53 | wcselby | my wife said it's pouring down in friendswood, out here near katy it's just dark and grey |
| 16:39.20 | Qwell | friendswood sounds like a happenin' place |
| 16:39.55 | r0m|u | I commute in to downtown. I got pored on... sucks.... |
| 16:43.30 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 16:44.23 | wcselby | believe me, friendswood (and the rest of Houston / Texas in general) needs the rain |
| 16:44.26 | casix | [TK]D-Fender: then this information in voip-info.org is false? 'It is possible to send different reply errors ("404 Not found", "484 Address incomplete" etc.) by setting <causecode> to one of the values defined by RFC 3398 - page 24.'. How can I read the X-Asterisk-HangupCause and X-Asterisk-HangupCauseCode from the 503 message? |
| 16:44.40 | wcselby | we've had a drought since last October, record low rainfall levels, etc etc |
| 16:45.03 | wcselby | casix a lot of info on that site is old or outdated or just plain incorrect |
| 16:45.23 | wcselby | casix if you could provide the requested information, we may be able to help you more |
| 16:45.30 | casix | ok |
| 16:45.33 | casix | one moment |
| 16:45.45 | Katty | wcselby: oh boy!!!! |
| 16:45.50 | Katty | wcselby: or...girl? |
| 16:45.52 | wcselby | boy |
| 16:45.56 | wcselby | any day now |
| 16:46.03 | wcselby | we shoudl probably pick a name |
| 16:46.39 | Katty | :> |
| 16:46.41 | Katty | OH BOY! |
| 16:46.46 | Katty | yes, yes a name would be good |
| 16:47.11 | wcselby | I'm a fan of the name "Inigo Montoya", myself, but my wife just won't go for it |
| 16:47.15 | Qwell | wcselby: I've got a recommendation for one. |
| 16:47.18 | Qwell | <-- |
| 16:47.19 | wcselby | it really flows into my last name too |
| 16:47.25 | Katty | Inigo montoya is a lovely name. |
| 16:47.31 | Katty | but i wouldn't want anyone to kill you |
| 16:47.35 | Katty | else he may have to seek revenge. |
| 16:47.36 | wcselby | lol |
| 16:47.52 | Katty | pick a masculine sexy name. |
| 16:47.55 | Katty | to bring sexy back |
| 16:47.57 | Qwell | <-- |
| 16:48.04 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 16:48.06 | Katty | that way he has all the girls in college. |
| 16:48.30 | wcselby | like i said, inigo montoya |
| 16:48.44 | wcselby | could you imagine all the play he'd get with that name |
| 16:48.56 | wcselby | "Hello ladies, my name is Inigo Montoya......" |
| 16:49.07 | wcselby | and I'm sure there's a catchy ending to that, but I can't think of it right now |
| 16:49.08 | r0m|u | lmao |
| 16:51.30 | tompaw | sysreq: does enabling DEBUG THREADS have serious impact on performance? |
| 16:51.47 | tompaw | I am considering switching to stock kernel and trying MeetMe instead of ConfBridge. |
| 16:52.48 | [TK]D-Fender | casix, "When call is hang up, Asterisk sends the extra SIP header "X-Asterisk-HangupCauseCode" in in the BYE message." <- this is not a SIP 404 packet |
| 16:53.50 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:54.33 | casix | [TK]D-Fender: yes I know. If I cannot make asterisk response 404 maybe I can read the header to know why the call is not working if is because the number does not exist or because the destinations is busy.. |
| 16:55.13 | sysreq | tompaw: yes, it kind of does. |
| 16:57.04 | [TK]D-Fender | casix, Read where? |
| 16:57.22 | kaldemar | casix: if you hangup with cause 2, asterisk will send a 404. |
| 16:57.42 | kaldemar | casix: Hangup(2) that is. |
| 16:58.15 | *** join/#asterisk Mimmus (~cg05947@ext.pitagora.it) |
| 16:58.27 | *** part/#asterisk Mimmus (~cg05947@ext.pitagora.it) |
| 16:58.53 | wcselby | ulaw call in on one provider, ulaw call out on another provider, both legs talking to each other but all traffic flowing through asterisk = approx 128 kbit of bandwidth, yeah? |
| 17:00.30 | kaldemar | casix: you'll see other supported causes in hangup_cause2sip in chan_sip. numerical values for the causes are in include/asterisk/causes.h. |
| 17:01.14 | singler | wcselby: nop, ~100k (with overhead) up and down to provider1, and ~100k up and down to provider2, totaling ~400k |
| 17:01.29 | Qwell | 200 + 200 != 400 |
| 17:01.36 | Qwell | 200 + 200 = 200 |
| 17:01.51 | wcselby | hmmm |
| 17:02.04 | Qwell | it's more like 80k * 2 |
| 17:02.13 | wcselby | inbound leg = 200k ?, outbound leg = 200k? |
| 17:02.21 | Qwell | ~160, but yes |
| 17:02.24 | wcselby | inbound leg = 160k? |
| 17:02.25 | wcselby | wow |
| 17:02.32 | wcselby | hmmmmm |
| 17:02.36 | singler | http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml |
| 17:02.48 | Qwell | TIL: People believe Cisco |
| 17:03.00 | wcselby | TIL: Qwell is a redditor |
| 17:03.11 | Qwell | But, that 87k is about right. (still a far cry from 100k) |
| 17:03.38 | singler | it is almost the same as you telling 80k |
| 17:04.19 | *** join/#asterisk Mimmus (~cg05947@ext.pitagora.it) |
| 17:04.21 | casix | kaldemar: I will look at that |
| 17:04.26 | *** part/#asterisk Mimmus (~cg05947@ext.pitagora.it) |
| 17:05.23 | *** join/#asterisk luke-jr (~luke-jr@2001:470:5:265:222:4dff:fe50:4c49) |
| 17:06.25 | *** join/#asterisk troyt (~troyt@c-24-10-222-127.hsd1.ut.comcast.net) |
| 17:07.00 | wcselby | so along those same lines, what is the bandwidth cost of g729 legs? |
| 17:07.17 | wcselby | 10k per leg? |
| 17:08.11 | [TK]D-Fender | 9.6 + 21 |
| 17:08.26 | [TK]D-Fender | under RTp |
| 17:10.01 | *** join/#asterisk irroot (~gregory@197.174.183.71) |
| 17:10.42 | *** join/#asterisk Tim_Toady (~fuzzy@188.4.14.167.dsl.dyn.forthnet.gr) |
| 17:24.00 | *** join/#asterisk hardwire (~spencersr@cl-36.anc-01.us.sixxs.net) |
| 17:26.40 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
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| 17:29.11 | *** join/#asterisk irroot (~gregory@197.108.15.45) |
| 17:30.46 | *** join/#asterisk godmachine-x6 (~poorelanc@funtoo/user/godmachine-x6) |
| 17:30.57 | *** join/#asterisk TSM (~the_softw@78-105-6-158.zone3.bethere.co.uk) |
| 17:32.18 | *** join/#asterisk PoWeRKiLL (~powerkill@IGLD-84-229-217-233.inter.net.il) |
| 17:36.17 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 17:37.53 | p3nguin | Why won't variables parse from astdb? http://pastebin.com/HieT4A4w |
| 17:37.58 | *** join/#asterisk PoWeRKiLL (~powerkill@IGLD-84-229-217-233.inter.net.il) |
| 17:39.16 | p3nguin | If I put ${myVar} in the database and then refer to it from dialplan, it shows ${myVar} instead of the value of the variable. |
| 17:39.37 | [TK]D-Fender | p3nguin, it doesn't double-parse |
| 17:39.54 | p3nguin | I also tried ${${myVar}} in the database, and it returns ${${myVar}}. |
| 17:39.55 | [TK]D-Fender | p3nguin, ${EVAL(${DB(testing/variable)})} |
| 17:40.10 | p3nguin | Let me try. |
| 17:41.21 | leifmadsen | yes, that's the purpose of EVAL() |
| 17:42.04 | p3nguin | I had never had an occasion to use it before. |
| 17:42.11 | p3nguin | But it does what I need! Thanks! |
| 17:42.12 | leifmadsen | now you do :) |
| 17:42.22 | leifmadsen | yep, EVAL() was created precisely for what you're doing |
| 17:42.23 | [TK]D-Fender | p3nguin, You're welcome |
| 17:42.46 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 17:43.33 | p3nguin | Is it safe to use EVAL() around all variables which are comprised of the DB() function? |
| 17:43.51 | hardwire | yeh.. you just have to sanity check what you're putting into the DB |
| 17:44.25 | hardwire | I have on occasion gotten back corrupt (looking in to that) astdb values.. Evalling it probably would do nothing.. but still. |
| 17:44.36 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176002135.dsl.bell.ca) |
| 17:44.46 | hardwire | I don't think little bobby tables lives on your phone system tho. |
| 17:44.58 | p3nguin | I think for now I'll do it on a case by case basis just to be sure. Perhaps later I will make it more uniform and wrap them all. |
| 17:45.04 | wcselby | hardwire lol @ bobby tables |
| 17:45.16 | *** join/#asterisk PoWeRKiLL (~powerkill@IGLD-84-229-217-233.inter.net.il) |
| 17:46.22 | hardwire | p3nguin: I use 'replace' since I'm used to specifying template contexts. |
| 17:46.31 | hardwire | or however asterisk does substring replacement. |
| 17:46.40 | hardwire | it's a bit saner. |
| 17:46.40 | *** join/#asterisk irroot (~gregory@197.170.62.211) |
| 17:48.58 | p3nguin | Okay, I used EVAL() on my actual case, and it works perfectly. |
| 17:52.29 | p3nguin | Set(DEVICE=${EVAL(${DB(phones/${EXTEN}/device)})}); Now I can put variable references, such as ${ringer}, in the device data and ${DEVICE} will include it, rather than using ${DEVICE/ringer=${ringer} in dial plan. |
| 17:52.51 | r0m|u | p3nguin, waz up d00d. |
| 17:53.12 | p3nguin | ${DEVICE}/ringer=${ringer}, that is. |
| 17:53.22 | p3nguin | just workin' |
| 17:53.25 | wcselby | um, wife just texted me, her contractions are like 6 minutes apart |
| 17:53.32 | wcselby | i think it's time to leave, since I'm like an hour away |
| 17:53.36 | r0m|u | wcselby, you better go home |
| 17:53.41 | wcselby | adios, #asterisk ! |
| 17:53.46 | *** join/#asterisk IsUp (5b8e8e8f@gateway/web/freenode/ip.91.142.142.143) |
| 17:53.48 | r0m|u | vongrats and good lcuk! |
| 17:54.02 | hardwire | p3nguin: I'm gonna argue just a bit more about using REPLACE instead |
| 17:54.05 | hardwire | and I'm done |
| 17:54.06 | hardwire | that was my bit. |
| 17:54.08 | r0m|u | p3nguin, I see. |
| 17:54.33 | r0m|u | p3nguin, Android 2.3 has a native sip client. works very well. |
| 17:55.05 | r0m|u | is listening to Nero - Guilt |
| 17:55.25 | p3nguin | hardwire: I'm not quite sure how to use it for this exact case. |
| 17:55.51 | Qwell | r0m|u: I keep hearing that, but I've not seen it. How does one get to it? |
| 17:56.34 | r0m|u | Qwell, What android version you on? What Carrier? rooted or stock? |
| 17:56.40 | *** join/#asterisk TimeRider (steve@host81-136-216-215.in-addr.btopenworld.com) |
| 17:56.53 | Qwell | r0m|u: 2.3, t-mobile, rooted |
| 17:57.35 | hardwire | p3nguin: if you know what variables you will eventually want to replace (because you planned out what you're doing) then doing string replacement in a series via a macro may be faster than evaluating it and safer. |
| 17:57.55 | r0m|u | Qwell, Settings ----- Call Settings ------ Scroll all the way to the bottom "Call Settings" |
| 17:58.05 | r0m|u | Hit Accounts |
| 17:58.16 | Qwell | ooo |
| 17:58.18 | hardwire | p3nguin: so I just use __variablename__ then replace it with $variablename later via replace. |
| 17:58.30 | r0m|u | Qwell, "Internet Call Settings" |
| 17:58.53 | Qwell | r0m|u: How do I actually call once I've set it up? I don't have a SIP account to play with ATM |
| 17:59.35 | Qwell | does it change up the dialer somehow? |
| 17:59.42 | r0m|u | Qwell, once it regsiter you can set it up to ether receive calls on sip, make calls on sip, or propt you to use "internet calls" |
| 17:59.46 | hardwire | Qwell: ekiga.net! |
| 17:59.48 | hardwire | haha |
| 17:59.53 | hardwire | sigh. |
| 17:59.59 | r0m|u | Qwell, no. Is native |
| 18:00.02 | r0m|u | so no changes |
| 18:00.22 | Qwell | r0m|u: so then how do I choose whether a call goes out via SIP or cell? |
| 18:00.34 | r0m|u | I have it set where for every calls it ask me what do I want to use "cell" or internet |
| 18:00.39 | Qwell | I see |
| 18:01.00 | Qwell | neat |
| 18:01.06 | r0m|u | indeed |
| 18:01.19 | r0m|u | quality is excellent compare to none native apps |
| 18:01.41 | r0m|u | it blows tmobile's wifi callings out of the water |
| 18:02.06 | r0m|u | I am CM7.0.1 Firmware |
| 18:02.20 | r0m|u | I am CM7.1.0.1 Firmware* |
| 18:02.46 | r0m|u | Samsung Vibrant with FFC modd |
| 18:04.00 | *** join/#asterisk Tim_Toady (~fuzzy@188.4.14.167.dsl.dyn.forthnet.gr) |
| 18:07.32 | leifmadsen | hmmm darn.... I don't see that on my Samsung S1 w/ android 2.3.5 |
| 18:07.46 | *** join/#asterisk kotis_ (~kotis@ext-dip-171.hnl.cdsinc.com) |
| 18:08.06 | r0m|u | leifmadsen, you need to let it grow some roots :P |
| 18:08.44 | leifmadsen | r0m|u: sounds like it! |
| 18:08.53 | r0m|u | leifmadsen, not all carriers allow it.... tmobile allows sip so it includes it on there 2.3 firmware. |
| 18:09.05 | leifmadsen | gotcha, this is on Telus, so probably not allowed |
| 18:09.13 | leifmadsen | searches root samsung galaxy s |
| 18:09.35 | hardwire | leifmadsen: oooh |
| 18:09.43 | r0m|u | leifmadsen, even though I am on tmobile I am using cyanogenmod.... |
| 18:09.43 | hardwire | has a rooted samsung galaxy captivate |
| 18:10.00 | r0m|u | hates samsungs bloated firmware |
| 18:10.56 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 18:10.58 | leifmadsen | oh snap, thought it was 2.3.5 but it's only 2.3.4 |
| 18:11.16 | *** join/#asterisk DelphiWorld (~VoCloud@openvpn/user/DelphiWorld) |
| 18:11.18 | DelphiWorld | hey all |
| 18:11.44 | r0m|u | leifmadsen, if you need help let me know. and in caser of er on your cell I have a jig and can recover your cell from hard or soft brick. |
| 18:11.54 | r0m|u | leifmadsen, ouch.... didnt know that :/ |
| 18:12.00 | DelphiWorld | hello folks, please see my dialplan at http://dpaste.com/660381/ |
| 18:12.07 | DelphiWorld | i want to add "+" to the Caller Id |
| 18:12.09 | DelphiWorld | but i don't know how |
| 18:12.26 | hardwire | if you like it then you should have put a + on it. |
| 18:12.30 | hardwire | erm. |
| 18:12.42 | leifmadsen | DelphiWorld: just put a + in front |
| 18:12.44 | hardwire | yeh |
| 18:12.51 | DelphiWorld | leifmadsen: front of what ? |
| 18:12.57 | leifmadsen | of the CALLERID() function |
| 18:13.15 | leifmadsen | sorry... of the value passed to CALLERID() |
| 18:13.24 | hardwire | well.. CALLERID(number) |
| 18:13.35 | leifmadsen | Set(CALLERID(number)=+5551212) |
| 18:13.43 | leifmadsen | this is asterisk 101 |
| 18:13.43 | IsUp | Set,CALLERID(number)=+${DB(${CALLERID(number)} |
| 18:13.54 | p3nguin | fail |
| 18:13.59 | r0m|u | epic |
| 18:14.38 | r0m|u | :P |
| 18:15.00 | DelphiWorld | leifmadsen: i only have this set: exten => s,1,Set(dialuri=${CUT(ARG1,\;,1)}) |
| 18:15.15 | leifmadsen | that means nothing to me |
| 18:15.30 | DelphiWorld | oh |
| 18:15.38 | DelphiWorld | leifmadsen: see that: |
| 18:16.10 | DelphiWorld | exten => s,n,ExecIf($["${DB(${CALLERID(number)}/user_sipname)}" != ""],Set,CALLERID(number)=+${DB(${CALLERID(number)}/user_sipname)}) |
| 18:16.15 | DelphiWorld | leifmadsen: right? |
| 18:16.24 | r0m|u | leifmadsen, 2.3.5 haz sip |
| 18:17.18 | leifmadsen | r0m|u: cool I'll check kies2 again and see what's up, although just checked it a couple weeks ago |
| 18:18.11 | DelphiWorld | leifmadsen: mine right ? |
| 18:18.15 | DelphiWorld | exten => s,n,ExecIf($["${DB(${CALLERID(number)}/user_sipname)}" != ""],Set,CALLERID(number)=+${DB(${CALLERID(number)}/user_sipname)}) |
| 18:18.19 | r0m|u | leifmadsen, good luck. I am ure your carrier striped out the function. If the did root it and go get a custom firmware.... But from the android dev release 2.3.5 does have sip. |
| 18:18.30 | r0m|u | if they did* |
| 18:18.40 | leifmadsen | r0m|u: makes sense, might do that because I found going to 2.3 from 2.2 was really slow |
| 18:18.44 | leifmadsen | (made the phone slow rather) |
| 18:18.53 | leifmadsen | DelphiWorld: asked and answered |
| 18:19.00 | DelphiWorld | leifmadsen: didn't see it :P |
| 18:19.08 | leifmadsen | DelphiWorld: how about trying first |
| 18:19.19 | DelphiWorld | leifmadsen: LOL don't want to breick it up :) |
| 18:19.48 | leifmadsen | DelphiWorld: that's what development machines are for |
| 18:19.55 | DelphiWorld | leifmadsen: buy me one:) |
| 18:19.56 | r0m|u | leifmadsen, Yes. A lot of people are complaining about stock carriers firmware slowing down there phones specially 2.3.X Thats why I moved to CM7.1 |
| 18:20.11 | r0m|u | lunch time |
| 18:20.17 | r0m|u | bbl |
| 18:20.18 | pabelanger | if you cannot afford a development box, you are doing it wrong |
| 18:20.22 | leifmadsen | pabelanger: +1 |
| 18:20.31 | DelphiWorld | pabelanger: leifmadsen -1 |
| 18:20.36 | DelphiWorld | :) |
| 18:20.38 | DelphiWorld | LOL |
| 18:21.43 | luke-jr | using a Local channel for a call file… but Dial() seems to abort at failure; any way to have it go on instead? |
| 18:21.47 | luke-jr | so it can try dialing another numb? |
| 18:23.14 | wdoekes2 | luke-jr: Dial will continue to the next extension after failure |
| 18:23.22 | wdoekes2 | *next priority |
| 18:23.52 | luke-jr | wdoekes2: it doesn't in this case :/ |
| 18:25.28 | wdoekes2 | are you sure it failed then? |
| 18:25.51 | wdoekes2 | which technology are you dealing with? |
| 18:26.00 | luke-jr | well, it goes to OutgoingSpoolFailed |
| 18:26.11 | luke-jr | wdoekes2: Local/ext@context/n |
| 18:26.16 | tzanger | morning |
| 18:26.26 | luke-jr | ext@context is where the Dial isn't going forward |
| 18:29.55 | DelphiWorld | thx leifmadsen |
| 18:30.58 | IsUp | fail |
| 18:30.59 | IsUp | epic |
| 18:31.01 | *** join/#asterisk troyt (~troyt@PariahZero.broker.freenet6.net) |
| 18:31.39 | [TK]D-Fender | luke-jr, Show us |
| 18:33.00 | luke-jr | [TK]D-Fender: translated to .conf, or is AEL ok? |
| 18:33.12 | [TK]D-Fender | conf |
| 18:33.23 | [TK]D-Fender | luke-jr, and the actual call. |
| 18:33.30 | [TK]D-Fender | (and call-file) |
| 18:35.58 | *** join/#asterisk pietro (~pietro@88-149-226-183.dynamic.ngi.it) |
| 18:36.41 | luke-jr | http://pastebin.com/2j7CbzU7 |
| 18:37.52 | [TK]D-Fender | luke-jr, Your 2 dial's have no timeout <- |
| 18:38.04 | [TK]D-Fender | luke-jr, If the 1st one doesn't "die" you'll never get to the 2nd |
| 18:38.22 | luke-jr | 1st one does die, in the log? O.o |
| 18:38.28 | [TK]D-Fender | luke-jr, And you've limited the waittime to 30s which is pretty bad |
| 18:38.41 | luke-jr | is it? |
| 18:40.50 | [TK]D-Fender | luke-jr, You've overall limited the length of the attempt and not limited the individual dials. |
| 18:41.00 | [TK]D-Fender | luke-jr, And we don't see that out-call being answered |
| 18:41.10 | luke-jr | hmm |
| 18:41.11 | luke-jr | I see |
| 18:41.16 | luke-jr | thanks |
| 18:41.28 | [TK]D-Fender | indefinite (or 30s ring) = fail |
| 18:44.20 | *** part/#asterisk DelphiWorld (~VoCloud@openvpn/user/DelphiWorld) |
| 18:46.11 | *** join/#asterisk KryptoKnight (~MDalby@cpc16-stkp7-2-0-cust163.10-2.cable.virginmedia.com) |
| 18:47.06 | KryptoKnight | Hey guys, Bad practice aside is it practical to run 20 PRI interfaces on a single box using Asterisk1.6, there will be around 300 concurrent calls split amongst the 20 at any one time |
| 18:48.50 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-xuwrlketeozlzusu) |
| 19:00.37 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-qstsdmbasqwlbtgp) |
| 19:03.07 | *** join/#asterisk hudony (~chatzilla@modemcable078.217-70-69.static.videotron.ca) |
| 19:03.51 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 19:04.12 | hudony | Hi, I have a pbx configured with 1 DID. I ordered 2 other dids, do I have to register them too or by some magic, only define extensions? |
| 19:04.31 | r0m|u | no magic here |
| 19:04.41 | hudony | ok thank you |
| 19:05.05 | r0m|u | You will have to define the registration just as you did with the ones before |
| 19:05.17 | r0m|u | define your peer and create your context |
| 19:05.18 | [TK]D-Fender | KryptoKnight, Every guideline I've seen has stated no more than 2 interface cards per system which on an 8-port Sangoma would only offer 16 ports. However it's been a while since the last written doc and Sangoma was always very good at resource management anyway. |
| 19:05.50 | [TK]D-Fender | <hudony> Hi, I have a pbx configured with 1 DID. I ordered 2 other dids, do I have to register them too or by some magic, only define extensions? <- depends on your provider |
| 19:06.06 | p3nguin | hudony: Within your context for that peer, define all the DIDs as extensions. |
| 19:06.08 | [TK]D-Fender | You might need a registration for each. You might not. |
| 19:06.20 | IsUp | KryptoKnight: i agree with D-Fender, i have Sangoma on my all servers |
| 19:06.39 | hudony | ok cause I tried with the register method but I can't get it to work |
| 19:06.40 | [TK]D-Fender | KryptoKnight, I'd suspec that with any halfway decent box you should be fine |
| 19:06.43 | r0m|u | Actually rereading I answer wrong :/ |
| 19:06.59 | r0m|u | hudony, [TK]D-Fender is right. if is from the same privider |
| 19:07.14 | hudony | yes it is |
| 19:07.21 | r0m|u | Over looked that. sorry |
| 19:07.32 | hudony | no problem! |
| 19:07.49 | p3nguin | Does your register statement require your extension in it? |
| 19:07.59 | KryptoKnight | Ah, I was just looking at the 8 port Digium cards but from what I'm reading Sangoma looks like the preferred option |
| 19:08.24 | r0m|u | than yes just create a context for each new did if you want them to act independently.... |
| 19:08.30 | r0m|u | hudony, ^^ |
| 19:08.37 | KryptoKnight | I guess the only way to know really is to try, i have dual 6 core AMD CPU's and 16GB RAM |
| 19:08.38 | hudony | ok |
| 19:08.41 | KryptoKnight | so im hoping it will wotk |
| 19:08.43 | KryptoKnight | work |
| 19:08.49 | KryptoKnight | Cheers for your advice guys |
| 19:08.50 | hudony | p3nguin: yes |
| 19:08.55 | p3nguin | You don't need a context for each DID; you need a context for each peer. |
| 19:09.01 | p3nguin | And you need an extension for each DID. |
| 19:09.13 | p3nguin | One provider, one peer, one context. |
| 19:09.46 | r0m|u | p3nguin, voip.ms requires each did to be define in the context for incoming no? |
| 19:10.08 | p3nguin | Not exactly, no. |
| 19:10.21 | r0m|u | ah I see. |
| 19:10.30 | p3nguin | They just send calls to your extensions. Whatever you want to do with the calls, that's up to you. |
| 19:10.42 | p3nguin | They can't require you to do anything with them. |
| 19:10.43 | r0m|u | I base my self from that... I see that is wrong now. |
| 19:11.21 | p3nguin | But what you said was, "just create a context for each new did," which is not necessary. |
| 19:11.25 | r0m|u | p3nguin, Thanks for the info. |
| 19:11.34 | p3nguin | (1309.13) <p3nguin> One provider, one peer, one context. |
| 19:11.44 | [TK]D-Fender | <p3nguin> You don't need a context for each DID; you need a context for each peer. And you need an extension for each DID. One provider, one peer, one context. <- not really... how you want to split stuff up "depends" |
| 19:11.54 | r0m|u | ^^ |
| 19:11.57 | hudony | ... |
| 19:11.59 | hudony | :S |
| 19:11.59 | r0m|u | Thats what I was refering too |
| 19:12.16 | r0m|u | <r0m|u> than yes just create a context for each new did if you want them to act independently.... |
| 19:12.19 | p3nguin | If you have one provider with one account... |
| 19:12.26 | p3nguin | You can't have more than one context for it. |
| 19:12.39 | p3nguin | You get one context per peer entry. |
| 19:12.59 | r0m|u | Mhhhh O see what you mean. |
| 19:13.01 | [TK]D-Fender | Some providers send to a single (and sometimes "empty") extension requiring some extra ugly parsing. As for contexts... depends again. Multiple peers can still share the same inbound context. Depends on the contexts, etc |
| 19:13.03 | r0m|u | I* |
| 19:13.27 | [TK]D-Fender | contents* |
| 19:13.41 | p3nguin | Multiple peer entries can share a single context, but a single peer entry cannot have more than one assigned context. |
| 19:13.47 | *** part/#asterisk pietro (~pietro@88-149-226-183.dynamic.ngi.it) |
| 19:14.09 | r0m|u | p3nguin, for incoming? |
| 19:14.14 | p3nguin | period |
| 19:14.16 | *** part/#asterisk AmirBehzad (~behzad@86.57.4.72) |
| 19:14.28 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 19:14.28 | IsUp | lol |
| 19:14.36 | p3nguin | All calls to extensions are incoming from asterisk's perspective. |
| 19:14.49 | r0m|u | Ah! yes! I see. |
| 19:17.09 | p3nguin | And always remember, there is an exception to almost every rule. |
| 19:18.20 | luke-jr | whee, found another bug in Asterisk |
| 19:18.59 | *** join/#asterisk francisvgarcia (~francis.g@190.80.239.124) |
| 19:19.01 | luke-jr | checks that it's still in 10.0 |
| 19:19.04 | *** join/#asterisk happylife (~happylife@46.251.83.126) |
| 19:19.05 | p3nguin | It could be a feature. |
| 19:19.21 | r0m|u | p3nguin, I can do this under a context for incoming right? http://pastebin.com/QmF8Gf5N |
| 19:20.05 | luke-jr | p3nguin: infinite loop is a feature? |
| 19:20.17 | p3nguin | r0m|u: I don't understand your question. |
| 19:20.17 | luke-jr | try this pattern: _[\|] |
| 19:20.45 | [TK]D-Fender | luke-jr, * 10 isnt released yet. |
| 19:21.21 | r0m|u | p3nguin, If I have two DID's and I want them to do something different under a peer I can have it all under the same context. |
| 19:21.55 | p3nguin | r0m|u: Okay. Continue. |
| 19:21.57 | [TK]D-Fender | r0m|u, Wasteful duplication in there |
| 19:22.08 | p3nguin | That was all I saw. |
| 19:22.16 | p3nguin | (just a duplication) |
| 19:22.27 | r0m|u | I know it just an example |
| 19:22.29 | luke-jr | [TK]D-Fender: rc is |
| 19:23.49 | r0m|u | dont be so "all ways correct" I am just trying to see if "yes you can have two did's do independent functions" under the same context. or "not" :) |
| 19:24.11 | r0m|u | I know that my paste was a duplication. |
| 19:24.12 | r0m|u | :) |
| 19:24.16 | p3nguin | That's what extensions are for. |
| 19:24.41 | p3nguin | Several extensions don't always have to do the exact same thing. |
| 19:24.59 | p3nguin | See my example dial plan. |
| 19:25.27 | [TK]D-Fender | luke-jr, RC = Candidate. |
| 19:25.33 | luke-jr | I'm aware. |
| 19:25.42 | r0m|u | p3nguin, ok. so I use the wrong word. Thanks for the clarification. not context but extension. |
| 19:25.52 | r0m|u | :) |
| 19:25.53 | [TK]D-Fender | luke-jr, Means don't bitch about bugs in hre, do that in -dev ;) |
| 19:26.15 | p3nguin | In my example, I have three DIDs (extensions) defined in my incoming cotext. |
| 19:26.26 | p3nguin | Each does something different. |
| 19:26.30 | r0m|u | yes sr. that is correct. |
| 19:27.12 | r0m|u | I need to get my terms right. |
| 19:27.22 | luke-jr | [TK]D-Fender: I'm doing it on Jira :p |
| 19:27.39 | [TK]D-Fender | luke-jr, I'm sure your contribution is appreciated :) |
| 19:28.08 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 19:28.27 | IsUp | perfection is a disease |
| 19:28.47 | r0m|u | not around here :P |
| 19:28.57 | r0m|u | You have p3nguin and [TK]D-Fender perfection at best |
| 19:30.49 | luke-jr | https://issues.asterisk.org/jira/browse/ASTERISK-18909 fwiw |
| 19:31.24 | r0m|u | If is not correct you will get corrected and bitch slap all in the same sentence.... With so fines and swiftness that you wont even notice. |
| 19:32.45 | luke-jr | r0m|u: it is. :p |
| 19:32.46 | pabelanger | luke-jr: attach a simple dialplan that reproduces it |
| 19:32.57 | [TK]D-Fender | r0m|u, If you've been bitch-slapped ... and "won't even notice" .... then what are you talking about? It never happened (to you. Or DID it?) </philosoraptor> |
| 19:33.10 | r0m|u | The best part is the results. You will do as told despite of anything.... :) it is called skills. |
| 19:33.20 | Qwell | luke-jr: like this? exten => _[a\bc],1,NoOp(${EXTEN}) |
| 19:33.39 | luke-jr | Qwell: right |
| 19:33.42 | Qwell | works on my box |
| 19:33.46 | luke-jr | oh? |
| 19:34.06 | *** join/#asterisk libryder (~david@209.33.214.243) |
| 19:34.14 | luke-jr | what version? |
| 19:34.28 | luke-jr | I only tested 1.6.2.9 (Debian stable) and reviewed the code in 10.0-rc2 |
| 19:34.36 | libryder | I'm getting an error on incoming calls: Call from 'Tollfreefwd-USA1' to extension 'device' rejected because extension not found in context and I'm wondering why it's reading "device" as the extension |
| 19:34.39 | Qwell | 1.6.2 isn't supported. |
| 19:34.58 | r0m|u | [TK]D-Fender, LOL. not to me I am just joking :) felt like saying something... :P |
| 19:35.02 | luke-jr | regardless, that's where I found it initially. |
| 19:35.12 | luke-jr | and I didn't see any fix in 10.0-rc2 |
| 19:35.24 | Qwell | So you didn't bother testing it in 10? |
| 19:35.29 | luke-jr | try _![\|]! |
| 19:35.33 | Qwell | Even though your report says you did exactly that? |
| 19:35.58 | p3nguin | libryder: Perhaps your register statement includes /device on it. |
| 19:36.10 | Qwell | luke-jr: That isn't a valid pattern. |
| 19:36.31 | luke-jr | Qwell: perhaps, but an infinite loop is still bad :p |
| 19:36.46 | luke-jr | reviewing 10.0-rc2 code again, I see no possible way this works |
| 19:36.47 | r0m|u | see's a body slam on the making.... |
| 19:36.58 | luke-jr | s1 never gets incremented |
| 19:37.16 | Qwell | It's never even getting to the range with that pattern. |
| 19:37.18 | luke-jr | perhaps the .conf parser filters it somehow |
| 19:38.05 | r0m|u | [TK]D-Fender, I meant no harm by the way. You guys have been awesome! You and p3nguin :) |
| 19:38.46 | luke-jr | Qwell: try via AEL |
| 19:39.11 | *** join/#asterisk voipeng (~voipeng45@75-150-128-2-Philadelphia.hfc.comcastbusiness.net) |
| 19:39.36 | Qwell | Give examples on the issue of the failure, and steps to reproduce. |
| 19:39.44 | luke-jr | I gave an example |
| 19:39.55 | luke-jr | context testcase { _![\|]! => NoOP(); }; |
| 19:40.11 | Qwell | On the issue. |
| 19:40.32 | r0m|u | p3nguin, on my final test they had you use its and it's it reminded me of you... I busted out laughing on the middle of the test. |
| 19:40.37 | *** join/#asterisk vpopov (~happylife@46.251.83.126) |
| 19:40.47 | p3nguin | heh |
| 19:41.00 | p3nguin | But did you get it right? |
| 19:41.07 | voipeng | not sure what the proper etiquite is for these channels, but I am having a problem with the zaptel driver on my pbxes, I started a thread at http://forums.digium.com/viewtopic.php?f=1&t=80702&sid=74fab22d77333362e638331ac639ee31 |
| 19:41.10 | r0m|u | YES SR! O DID I! LOL |
| 19:41.24 | p3nguin | Then I have been successful. |
| 19:41.29 | voipeng | anyone available to assist? |
| 19:41.31 | *** join/#asterisk vinhdizzo (~vinh@vqn-routerpbx.ics.uci.edu) |
| 19:42.04 | r0m|u | p3nguin, rofl! exactly why I was laughing! It engraved in my brain. |
| 19:43.01 | hudony | ok, got it working : only 1 register, 1 context, 3 extensions |
| 19:43.27 | hudony | AS one said previously, when one call my second did, it is like the first one was forwarding the call explicitely to the second |
| 19:43.33 | hudony | From what I can see from the console |
| 19:43.37 | [TK]D-Fender | voipeng, Zaptel was replaced by DAHDI years ago and is no longer being developed under that name. Your card is also discontinued. |
| 19:44.13 | r0m|u | p3nguin, your kung fu methods have been known to work and be effective. no doubt about that :P |
| 19:44.19 | r0m|u | afk |
| 19:44.52 | voipeng | d-fender, im not actually using any hardware resources |
| 19:44.55 | p3nguin | hudony: Still having that problem, or it is all fixed now? |
| 19:45.27 | [TK]D-Fender | voipeng, And you also haven't even mentioned what versions you're running |
| 19:45.38 | hudony | fixed |
| 19:46.06 | voipeng | d-fender, one sec ill post them as well here and on the forum |
| 19:46.14 | [TK]D-Fender | voipeng, If you have a card that is what it will use for timing, never dummy first |
| 19:46.41 | voipeng | i never had a card, i shouldnt have posted that link... i wasnt familar with the resource so i posted it as a reference] |
| 19:48.07 | voipeng | posted on the forum my related modinfo |
| 19:48.10 | voipeng | here it is as well |
| 19:48.10 | voipeng | PBX11 who is not experiencing the problem seems to be running the same version as the problem pbx14. |
| 19:48.10 | voipeng | Looks like our working pbx11 is running this: |
| 19:48.10 | voipeng | [tfiore@fs11(pbx11 primary) ~]$ sudo /sbin/modinfo /lib/modules/2.6.18-194.8.1.el5/kernel/misc/zaptel.ko |
| 19:48.10 | voipeng | filename: /lib/modules/2.6.18-194.8.1.el5/kernel/misc/zaptel.ko |
| 19:48.11 | voipeng | version: 1.4.9.2 |
| 19:48.12 | voipeng | license: GPL |
| 19:48.14 | voipeng | description: Zapata Telephony Interface |
| 19:48.17 | voipeng | author: Mark Spencer <markster@digium.com> |
| 19:48.18 | voipeng | srcversion: 09F9962E84B1D28F6C7CD09 |
| 19:48.21 | voipeng | depends: crc-ccitt |
| 19:48.22 | voipeng | vermagic: 2.6.18-194.8.1.el5 SMP mod_unload 686 REGPARM 4KSTACKS gcc-4.1 |
| 19:48.24 | [TK]D-Fender | .. |
| 19:48.24 | voipeng | parm: debug:int |
| 19:48.26 | [TK]D-Fender | PASTEBIN |
| 19:48.27 | voipeng | parm: deftaps:int |
| 19:48.30 | voipeng | [tfiore@fs11(pbx11 primary) ~]$ |
| 19:48.33 | voipeng | [tfiore@fs16(primary) ~]$ sudo /sbin/modinfo /lib/modules/2.6.18-274.7.1.el5/kernel/misc/zaptel.ko |
| 19:48.34 | voipeng | filename: /lib/modules/2.6.18-274.7.1.el5/kernel/misc/zaptel.ko |
| 19:48.36 | [TK]D-Fender | ~pb |
| 19:48.37 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 19:48.37 | voipeng | version: 1.4.9.2 |
| 19:48.38 | voipeng | license: GPL |
| 19:48.40 | voipeng | description: Zapata Telephony Interface |
| 19:48.43 | voipeng | author: Mark Spencer <markster@digium.com> |
| 19:48.44 | voipeng | srcversion: 09F9962E84B1D28F6C7CD09 |
| 19:48.45 | navaismo | stop |
| 19:48.46 | voipeng | depends: crc-ccitt |
| 19:48.48 | voipeng | vermagic: 2.6.18-274.7.1.el5 SMP mod_unload 686 REGPARM 4KSTACKS gcc-4.1 |
| 19:48.50 | [TK]D-Fender | ops? |
| 19:48.50 | voipeng | parm: debug:int |
| 19:48.52 | voipeng | parm: deftaps:int |
| 19:48.54 | voipeng | [tfiore@fs16(primary) ~]$ |
| 19:48.56 | voipeng | [tfiore@fs14(fs14) ~]$ sudo /sbin/modinfo /lib/modules/2.6.18-194.32.1.el5/kernel/misc/zaptel.ko |
| 19:49.00 | voipeng | filename: /lib/modules/2.6.18-194.32.1.el5/kernel/misc/zaptel.ko |
| 19:49.03 | voipeng | version: 1.4.9.2 |
| 19:49.03 | tzanger | voipeng: don't dump everything to the channel |
| 19:49.04 | voipeng | license: GPL |
| 19:49.06 | voipeng | description: Zapata Telephony Interface |
| 19:49.08 | voipeng | author: Mark Spencer <markster@digium.com> |
| 19:49.10 | voipeng | srcversion: 09F9962E84B1D28F6C7CD09 |
| 19:49.12 | voipeng | depends: crc-ccitt |
| 19:49.14 | voipeng | vermagic: 2.6.18-194.32.1.el5 SMP mod_unload 686 REGPARM 4KSTACKS gcc-4.1 |
| 19:49.16 | voipeng | parm: debug:int |
| 19:49.18 | [TK]D-Fender | Qwell <- |
| 19:49.18 | voipeng | parm: deftaps:int |
| 19:49.20 | navaismo | stooooop |
| 19:49.20 | voipeng | [tfiore@fs14(fs14) ~]$ |
| 19:49.23 | voipeng | ? |
| 19:49.24 | voipeng | the file? |
| 19:49.26 | voipeng | gotcha sorry |
| 19:49.30 | tzanger | we need a few more people with op status here |
| 19:49.31 | voipeng | SORRY |
| 19:49.32 | voipeng | IVE NEVER USED IRC BEFORE |
| 19:49.34 | voipeng | jesus |
| 19:49.36 | voipeng | lol really? |
| 19:49.57 | r0m|u | ~pb |
| 19:49.58 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 19:50.55 | voipeng | <voipeng> PBX11 who is not experiencing the problem seems to be running the same version as the problem pbx14. |
| 19:50.55 | voipeng | <voipeng> Looks like our working pbx11 is running this: |
| 19:50.55 | voipeng | <voipeng> [tfiore@fs11(pbx11 primary) ~]$ sudo /sbin/modinfo /lib/modules/2.6.18-194.8.1.el5/kernel/misc/zaptel.ko |
| 19:50.55 | voipeng | <voipeng> filename: /lib/modules/2.6.18-194.8.1.el5/kernel/misc/zaptel.ko |
| 19:50.56 | voipeng | <voipeng> version: 1.4.9.2 |
| 19:50.58 | voipeng | <voipeng> license: GPL |
| 19:51.02 | voipeng | <voipeng> description: Zapata Telephony Interface |
| 19:51.04 | voipeng | <voipeng> author: Mark Spencer <markster@digium.com> |
| 19:51.06 | voipeng | <voipeng> srcversion: 09F9962E84B1D28F6C7CD09 |
| 19:51.08 | voipeng | <voipeng> depends: crc-ccitt |
| 19:51.09 | [TK]D-Fender | asddasasd |
| 19:51.10 | voipeng | <voipeng> vermagic: 2.6.18-194.8.1.el5 SMP mod_unload 686 REGPARM 4KSTACKS gcc-4.1 |
| 19:51.12 | voipeng | <voipeng> parm: debug:int |
| 19:51.14 | voipeng | <voipeng> parm: deftaps:int |
| 19:51.14 | [TK]D-Fender | Qwell ? |
| 19:51.16 | voipeng | <voipeng> [tfiore@fs11(pbx11 primary) ~]$ |
| 19:51.18 | voipeng | <voipeng> [tfiore@fs16(primary) ~]$ sudo /sbin/modinfo /lib/modules/2.6.18-274.7.1.el5/kernel/misc/zaptel.ko |
| 19:51.18 | tzanger | sighs |
| 19:51.21 | voipeng | <voipeng> filename: /lib/modules/2.6.18-274.7.1.el5/kernel/misc/zaptel.ko |
| 19:51.21 | *** kick/#asterisk [voipeng!~north@pdpc/sponsor/digium/Qwell] by Qwell (go away) |
| 19:51.23 | navaismo | facepalms |
| 19:51.28 | [TK]D-Fender | Qwell, thanks... |
| 19:51.32 | tzanger | heh |
| 19:52.09 | r0m|u | iiinnnnn ttthhhhaaaa ffffaaacccceeee |
| 19:52.33 | francisvgarcia | wtf? |
| 19:52.40 | *** join/#asterisk voipeng (~voipeng45@75-150-128-2-Philadelphia.hfc.comcastbusiness.net) |
| 19:52.46 | voipeng | thanks... here is my pb |
| 19:52.51 | voipeng | http://pastebin.com/4gB977Zr |
| 19:53.19 | Qwell | Upgrade. Next? |
| 19:53.29 | voipeng | upgrade what to what? |
| 19:53.35 | Qwell | everything, to not 1999 |
| 19:53.49 | voipeng | ... ok? |
| 19:53.56 | voipeng | well it works on one of the 4 pbxes |
| 19:54.29 | hudony | Thx for your help all of you. My 3 did are working just fine |
| 19:54.37 | navaismo | voipeng: same load, same chipset, same processor, same ram? |
| 19:55.07 | navaismo | voipeng: upgrade your versions |
| 19:55.43 | voipeng | ok, is there a stable version I should try and update to? |
| 19:55.50 | voipeng | we attempted to update to the latest on pbx16 |
| 19:55.52 | r0m|u | voipeng, you didn't get the memo? |
| 19:56.08 | Qwell | latest? You're running zaptel. |
| 19:57.06 | voipeng | ok, so how can i tranisition to dahdi or upgrade to the latest stable zaptel version? |
| 19:57.20 | voipeng | it seems like they are both active on my system, zaptel and dahdi |
| 19:57.20 | Qwell | Install dahdi, upgrade Asterisk. |
| 19:57.31 | voipeng | anyway i can verify that? |
| 19:57.50 | voipeng | i do see directories/files when i do a locate for dahdi, and some of the core commands use the dahdi commands instead |
| 20:00.09 | voipeng | any links you would recommend that outline this process? |
| 20:00.23 | voipeng | either zaptel upgrade or dahdi install and asterisk upgrade |
| 20:00.39 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 20:00.43 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v028-178.mobile.uci.edu) |
| 20:03.08 | navaismo | voipeng: if you are using tarballs from the beginning download the latest version and compile it, if you are using distro packages ask Qwell |
| 20:05.13 | voipeng | navaismo: is this for upgrading the zaptel driver or converting to dahdi? |
| 20:05.57 | navaismo | both, if you compiled the source code |
| 20:06.31 | voipeng | typically just use yum, thats what i used to upgrade the zaptel version |
| 20:07.05 | navaismo | use that method, mine is only if you compiled the packages |
| 20:07.40 | voipeng | ok so i do a yum update for dadhi first? then i would look into upgrading the asterisk version? |
| 20:09.27 | navaismo | i dont know i dont use distro packages |
| 20:09.33 | navaismo | but some else can helo you |
| 20:09.36 | navaismo | help* |
| 20:09.52 | *** part/#asterisk IsUp (5b8e8e8f@gateway/web/freenode/ip.91.142.142.143) |
| 20:10.02 | voipeng | this is what i get when i run a yum update http://pastebin.com/8q0wUy6B |
| 20:10.15 | voipeng | i dont see dahdi listed at all, i guess its not installed? |
| 20:11.35 | [TK]D-Fender | voiceaxis <-------- go ask these people.. you're running off their repo which we don't support |
| 20:12.31 | voipeng | they state its not using voiceaxis |
| 20:12.35 | voipeng | its an asterisk related issue |
| 20:13.13 | voipeng | well they say it is, i obviously am unsure |
| 20:14.05 | *** join/#asterisk celord (~celord@201.198.102.2) |
| 20:17.30 | [TK]D-Fender | Yes and you've installed it from their resources so they should provide you with newer packages |
| 20:19.30 | voipeng | ive brought the problem to their attention and they stated they are running the same zaptel verison and it works |
| 20:19.39 | voipeng | which i believe since one of our pbx'es work on that version as well |
| 20:20.17 | [TK]D-Fender | voipeng, Well if you want to say "same software works elsewhere" then the problem is hardware. |
| 20:20.20 | voipeng | instead of focusing on what i am using, if i run the zttest and i see responses less than 90 something its definetly a timing issue right? |
| 20:20.35 | [TK]D-Fender | zttest = timing test. |
| 20:21.18 | voipeng | right, so the fact that voice quality/moh sounds bad when i see a zttest with a low percent it has to be timing related as well |
| 20:21.22 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-efqrxdmozlxevyfa) |
| 20:22.06 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-hfbsnrexlndxpuut) |
| 20:22.27 | voipeng | sorry if your repeating yourself, I come from a CUCM background... this is a lot different |
| 20:23.45 | voipeng | as for the hardware, the test server has virtually no load on it, so im doubting it is hardware |
| 20:27.26 | *** join/#asterisk GameGamer43 (~GameGamer@li119-60.members.linode.com) |
| 20:28.42 | *** join/#asterisk GameGamer43 (~GameGamer@li119-60.members.linode.com) |
| 20:35.21 | voipeng | i just contacted the manager of the voiceaxis support and they stated its outside of their scope... any additional help you guys could provide would be greatly appreciated. |
| 20:39.33 | *** join/#asterisk [Outcast] (~outcast@westford-nat.juniper.net) |
| 20:40.00 | [Outcast] | does meetme support video conferencing |
| 20:41.05 | pabelanger | [Outcast]: no |
| 20:41.13 | pabelanger | confbridge in asterisk 10 does |
| 20:41.20 | [Outcast] | aah |
| 20:41.56 | *** join/#asterisk Eitan (~Eitan@12.192.84.98) |
| 20:42.09 | [Outcast] | is it good? |
| 20:42.46 | pabelanger | it's pretty bad ass |
| 20:43.09 | [Outcast] | so does it show all the uses or just the active speaker? |
| 20:43.14 | [Outcast] | *uers |
| 20:43.19 | [Outcast] | errr........users |
| 20:43.37 | [Outcast] | my fingers tripped over my keyboard |
| 20:45.09 | pabelanger | no transcoding, so just the active speaker |
| 20:45.16 | [Outcast] | ok that is cool |
| 20:47.48 | leifmadsen | pabelanger: I can only get this far :) |
| 20:48.13 | leifmadsen | pabelanger: forget it... logout time |
| 20:49.25 | voipeng | can anyone suggest how to tweak zaptel drivers? I am getting negative percentages sometimes... |
| 20:49.25 | voipeng | --- Results after 166 passes --- |
| 20:49.26 | voipeng | Best: 99.153 -- Worst: -315.872 -- Average: 96.286203, Difference: 103.713797 |
| 20:50.08 | p3nguin | Zaptel? There is no supported asterisk version using zaptel currently. |
| 20:51.42 | *** join/#asterisk vinhdizzo (~vinh@vqn-routerpbx.ics.uci.edu) |
| 20:51.48 | voipeng | gotcha, my repo is locked where i have to use it... so i guess keep searching? |
| 20:54.03 | navaismo | voipeng: yes with your repo-maintener its possible if you use another you can broke your system |
| 20:54.33 | voipeng | gotcha, i did an yum update and obtained a newer zaptel module |
| 20:54.43 | voipeng | but still same results in the test |
| 20:56.21 | *** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:59.36 | *** part/#asterisk libryder (~david@209.33.214.243) |
| 21:03.44 | [TK]D-Fender | <p3nguin> Zaptel? There is no supported asterisk version using zaptel currently. |
| 21:04.56 | voipeng | yea i get that, but im limited to using it |
| 21:06.05 | pabelanger | download dahdi, compile and install |
| 21:07.19 | jaytee | last time I saw a system with zaptel it was 1.4.10 or something |
| 21:07.30 | p3nguin | It was in use up to 1.4.21, I think. |
| 21:07.45 | p3nguin | Regardless, it is antiquated. |
| 21:08.00 | jaytee | yep |
| 21:08.04 | voipeng | yea beleive me if i could change it i would |
| 21:08.17 | voipeng | just a tech trying to fix a problem heh |
| 21:08.53 | jaytee | I think you can still download the zaptel source and compile it |
| 21:08.55 | voipeng | Im on 1.4.29 btw |
| 21:09.04 | voipeng | I downloaded the one stamped good for my repo |
| 21:09.06 | jaytee | that version would need dahdi |
| 21:09.12 | jaytee | not zaptel |
| 21:10.39 | voipeng | i have dahdi commands from the cli |
| 21:10.46 | voipeng | but i dont see it when i do a yum update |
| 21:13.23 | voipeng | vmfs01a*CLI> dahdi show status |
| 21:13.23 | voipeng | Description Alarms IRQ bpviol CRC4 |
| 21:13.23 | voipeng | ZTDUMMY/1 (source: Linux26) 1 UNCONFIGUR 0 0 0 |
| 21:13.23 | voipeng | vmfs01a*CLI> |
| 21:13.37 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 21:13.47 | voipeng | on my pbx that works it says its using source: RTC not source:Linux26 |
| 21:13.52 | voipeng | would that make a difference? |
| 21:14.58 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 21:17.01 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:17.03 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 21:19.27 | voipeng | so does that mean i have some type of dahdi version installed? |
| 21:19.51 | navaismo | no |
| 21:20.03 | voipeng | k |
| 21:21.18 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:29.34 | [TK]D-Fender | checkout time, BBIAB |
| 21:31.15 | voipeng | so i am trying to install dahdi, is it possible to do through yum or do i need to manually pull down the files and scp them over ? |
| 21:31.58 | voipeng | not sure which to download from http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/ |
| 21:33.49 | *** join/#asterisk pietro (~pietro@88-149-226-183.dynamic.ngi.it) |
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| 21:41.09 | navaismo | latest |
| 21:41.21 | navaismo | but you need to update your asterisk too |
| 21:42.17 | voipeng | ah i need to update asterisk first? |
| 21:42.27 | voipeng | I extracted the latest dahdi complete on the server |
| 21:42.36 | voipeng | but theres no make command? |
| 21:43.18 | r0m|u | voipeng, you have bigger issues.... |
| 21:43.21 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v028-178.mobile.uci.edu) |
| 21:43.32 | r0m|u | no "make command" |
| 21:43.51 | r0m|u | bbl...... is time to switch home's |
| 21:44.21 | voipeng | [t@vmfs01a(vmfs01b) dahdi-linux-complete-2.6.0-rc1+2.6.0-rc1]$ ./configure |
| 21:44.21 | voipeng | -bash: ./configure: No such file or directory |
| 21:44.21 | voipeng | [t@vmfs01a(vmfs01b) dahdi-linux-complete-2.6.0-rc1+2.6.0-rc1]$ make |
| 21:44.21 | voipeng | -bash: make: command not found |
| 21:44.21 | voipeng | [t@vmfs01a(vmfs01b) dahdi-linux-complete-2.6.0-rc1+2.6.0-rc1]$ |
| 21:46.36 | *** join/#asterisk libryder (~david@209.33.214.243) |
| 21:47.33 | libryder | is it possible that i have a configuration problem if an incoming number from a SIP trunk is showing in asterisk as an extension "device" ? |
| 21:48.18 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:48.57 | p3nguin | Did you ever paste your register statement and peer entry for that provider? |
| 21:48.59 | pabelanger | voipeng: there is no configure script, and you are likely missing development tools |
| 21:49.03 | pabelanger | EG: gcc, make |
| 21:51.06 | *** join/#asterisk pietro (~pietro@88-149-226-183.dynamic.ngi.it) |
| 21:51.50 | voipeng | gotcha, i did try through yum and got the following output... looks like i need to uninstall the old asterisk version first?http://pastebin.com/hp74jHCj |
| 21:54.27 | voipeng | im pretty sure gcc is installed |
| 21:54.37 | p3nguin | "which gcc" will tell you. |
| 21:55.08 | voipeng | heh guess not |
| 21:55.11 | *** join/#asterisk ulogic (4a59e7fc@gateway/web/freenode/ip.74.89.231.252) |
| 21:55.29 | voipeng | which: no gcc in (/usr/bin:/bin) |
| 21:55.44 | p3nguin | Should be found at /usr/bin/gcc. |
| 21:56.04 | _Corey_ | voipeng: If you're on Centos/Fedora you can do 'yum groupinstall "Development Tools" ' or something like that |
| 21:56.09 | p3nguin | On an rpm system, you can use rpm to see everything installed. |
| 21:56.25 | tompaw | Guys, I rebuilt the kernel, dahdi, modprob'd dahdi, have it loaded - what else do I have to do to use MeetMe? |
| 21:56.46 | p3nguin | "rpm -qa gcc" would show you what gcc is installed. |
| 21:56.57 | p3nguin | gcc-4.1.2-48.el5, for example. |
| 21:57.42 | voipeng | when i run that command no output |
| 21:57.57 | voipeng | i tried the development tools got this output |
| 21:57.57 | voipeng | http://pastebin.com/Znqgmi1B |
| 21:58.06 | p3nguin | Right, because you already determined gcc was not installed. |
| 21:58.13 | voipeng | mm ok |
| 21:59.01 | libryder | p3nguin: http://pastebin.com/zB8bf1xP |
| 21:59.20 | libryder | not sure where the register statement is |
| 21:59.31 | p3nguin | It should be in the general section. |
| 22:00.29 | ulogic | To use MeetMe, make sure to rerun ./configure in the asterisk source directory, then it should come up as an option when you run "make menuselect" |
| 22:00.48 | tompaw | ulogic: at this very second I realized I had to rebuild it :-) Thanks mate. |
| 22:01.09 | ulogic | However, app_meetme is being deprecated and is being replaced by app_confbridge |
| 22:01.31 | KavanS | why is meetme being deprecated? |
| 22:01.40 | KavanS | is just wondering |
| 22:01.58 | p3nguin | *being* would be the key word, since ConfBridge on the only currenly released/supported Asterisk version isn't very great. |
| 22:02.21 | p3nguin | So until 10 is released, MeetMe is superior. |
| 22:02.41 | tompaw | ulogic: there is no way to control confbridge in 1.8 (list/record confs) and I feel it's killing my * every 2-3 hours |
| 22:02.57 | tompaw | going to try meetme now and see if dreadlocks are gone |
| 22:05.09 | libryder | p3nguin: i don't see a register statement; it's all commented out |
| 22:05.22 | ulogic | MeetMe depends on DAHDI whereas ConfBridge has no dependencies. You will see this when you "make menuselect" in version 10. |
| 22:05.38 | p3nguin | libryder: How do you tell your ITSP where to send your calls? |
| 22:06.54 | tompaw | ulogic: correct. I rebuilt * and meetme works great, thanks :-) |
| 22:07.30 | *** join/#asterisk pietro (~pietro@88-149-226-183.dynamic.ngi.it) |
| 22:08.25 | libryder | p3nguin: we are a level3 peer co-located with one of their servers so we have a direct ip connection |
| 22:09.12 | p3nguin | So what? That doesn't magically make them authenticated to deliver calls to your system. |
| 22:09.46 | p3nguin | There has to be some way to tell the other system where to send calls, and your system has to be told to accept them. |
| 22:10.37 | p3nguin | If you are not telling the other side, via register statement, where to send calls, then the other side is responsible for this problem. |
| 22:10.44 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 22:11.13 | p3nguin | And from what I have seen so far, you aren't telling them anything. |
| 22:13.48 | tompaw | what do I need to build chan_dahdi.so? |
| 22:13.54 | tompaw | it's disabled in my menuconfig |
| 22:14.57 | p3nguin | Now I'm confused. |
| 22:15.00 | p3nguin | (1606.54) <tompaw> ulogic: correct. I rebuilt * and meetme works great, thanks :-) |
| 22:15.11 | p3nguin | If you run "dahdi show channels" in your asterisk CLI, what happens? |
| 22:15.36 | tompaw | p3nguin: meetme itself works, modprobe dahdi works, but: |
| 22:15.38 | tompaw | [Nov 22 23:12:01] WARNING[22764]: app_meetme.c:4073 find_conf: No DAHDI channel available for conference, conference recording disabled (is chan_dahdi loaded?) |
| 22:15.49 | tompaw | there is no chan_dahdi.so and I cannot enable it in menuconfig |
| 22:15.52 | p3nguin | Oh, I see. |
| 22:16.05 | tompaw | no such command "dahdi" |
| 22:16.11 | tompaw | looks like it's... partially installed? |
| 22:16.11 | ulogic | tompaw: Each time you meet a dependency, you have to rerun ./configure in the asterisk source directory |
| 22:16.26 | p3nguin | You installed dahdi and then reconfigured asterisk? |
| 22:16.31 | tompaw | yes |
| 22:17.02 | p3nguin | Does "module load chan_dahdi.so" work? |
| 22:17.07 | ulogic | tompaw: Also, you need to define some channels in /etc/asterisk/chan_dahdi.conf before the chan_dahdi.so module will load. |
| 22:17.15 | tompaw | After I installed dahdi, app_meetme became available (before it was XXX). |
| 22:17.17 | p3nguin | That's not true. |
| 22:17.24 | tompaw | But chan_dahdi it's still XXX. |
| 22:17.40 | tompaw | ulogic: I don't think so, because the module does not exist. It's not being built during the build. |
| 22:18.14 | tompaw | p3nguin: no such file or directory. |
| 22:18.19 | p3nguin | You don't have to define channels in chan_dahdi.conf when you have no hardware channels to define. |
| 22:18.25 | libryder | p3nguin: the numbers i'm actually having problems with are coming from tollfreeforwarding.com and according to their super basic instructions, we just need to open a specific set of ip addresses, which is what i was attempting to do with that sip conf i pasted earlier |
| 22:18.30 | ulogic | tompaw: which version of asterisk are you building? |
| 22:18.40 | p3nguin | No such file or directory? That's not an asterisk error message. |
| 22:18.41 | tompaw | p3nguin: could it be that I only installed dahdi-linux and not dahdi-tools? |
| 22:18.45 | p3nguin | no |
| 22:18.47 | tompaw | ulogic: 1.8.7.1 |
| 22:19.09 | tompaw | And 2.5.0.2 for dahdi. |
| 22:19.21 | p3nguin | (1618.39) <p3nguin> No such file or directory? That's not an asterisk error message. |
| 22:19.24 | p3nguin | ^ |
| 22:19.33 | p3nguin | Try again. |
| 22:19.35 | ulogic | make menuselect says chan_dahdi also depends on tonezone |
| 22:19.38 | tompaw | Well, asterisk surely somehow "detects" dahdi, because it let me build app_meetme. |
| 22:19.54 | tompaw | p3nguin: [Nov 22 23:18:03] WARNING[22042]: loader.c:387 load_dynamic_module: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: cannot open shared object file: No such file or directory |
| 22:20.04 | p3nguin | That's more like it. |
| 22:20.19 | tompaw | last part says no such file or directory :P |
| 22:20.27 | p3nguin | I see. |
| 22:21.03 | tompaw | don't I have to provide path to dahdi when I ./configure asterisk? |
| 22:21.13 | p3nguin | I find it interesting that you solved the dahdi dependency and app_meetme became available, yet dahdi doesn't exist. |
| 22:21.28 | p3nguin | Not usually, no. |
| 22:21.51 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-nyuybjmesrslpyrt) |
| 22:22.45 | tompaw | It might be true I'm missing tonezone... |
| 22:22.54 | tompaw | whatever that is |
| 22:22.57 | libryder | p3nguin: http://www.fonality.com/trixbox/forums/trixbox-forums/trunks/tollfreeforwarding-inbound-sip-800-number-setup |
| 22:23.02 | libryder | that's pretty much the same email i got |
| 22:23.26 | ulogic | tompaw: try installing dahdi-tools, then after you install it, i believe you need to run make config |
| 22:23.45 | p3nguin | libryder: Okay, so you can configure your destination on their system. It looks like they have a "ring-to" field. |
| 22:24.25 | p3nguin | libryder: Whatever you put in the field is what you should get. If you put "device@ipaddress" expect the call to arrive at extension "device". |
| 22:24.42 | libryder | omg |
| 22:24.58 | p3nguin | If you want it to arrive at your phone number, which is what I would do, I would use 13145551212@myipaddress in the ring-to field. |
| 22:25.19 | p3nguin | or just 3145551212@myipaddress |
| 22:26.03 | p3nguin | Then I would configure extension 3145551212 in my system, in the context calls from that provider go into. |
| 22:26.05 | tompaw | TIL: dahdi-tools is required in order to build chan_dahdi ;-) |
| 22:26.17 | p3nguin | It is? |
| 22:26.21 | p3nguin | Since when? |
| 22:27.58 | tompaw | Since 5 mintues ago. As soon as I installed dahdi-tools, chan_dahdi became available in *'s menuconfig (after ./configure of course). |
| 22:28.13 | libryder | p3nguin: you rock man |
| 22:28.28 | p3nguin | I don't recall ever having dahdi tools installed. |
| 22:29.39 | tompaw | p3nguin: maybe you install dahdi-complete |
| 22:29.58 | p3nguin | Nope. |
| 22:30.29 | p3nguin | I'll investigate later. |
| 22:30.58 | tompaw | p3nguin: I can provide you with my version numbers if you need them later |
| 22:31.18 | p3nguin | I'll assume you are using the current versions of all software involved. |
| 22:31.43 | *** part/#asterisk pietro (~pietro@88-149-226-183.dynamic.ngi.it) |
| 22:35.33 | tompaw | p3nguin: yes plus custom kernel (since the stock version doesn't support my 10Gbps network card) |
| 22:35.45 | tompaw | not sure if that's relevant. |
| 22:52.40 | tompaw | ] |
| 22:53.24 | *** join/#asterisk vinhdizzo (~vinh@vqn-routerpbx.ics.uci.edu) |
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| 22:57.10 | *** join/#asterisk celord (~celord@201.198.102.2) |
| 23:08.44 | *** part/#asterisk ulogic (4a59e7fc@gateway/web/freenode/ip.74.89.231.252) |
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| 23:18.35 | SeRi | waz up p3nguin |
| 23:19.05 | p3nguin | Trying to figure out how to unlock this damn iPhone 3G. |
| 23:19.09 | *** join/#asterisk Greenlight (~wluke@cpc4-dund11-2-0-cust378.sgyl.cable.virginmedia.com) |
| 23:19.42 | SeRi | :/ well after the last incident I have not been able to go to the PO over by my house.... |
| 23:19.57 | SeRi | :( sorry. |
| 23:20.11 | p3nguin | I can't even figure out what search terms to use for this case. |
| 23:20.27 | SeRi | I am still puzzled about not been able to unlock it without a sim... |
| 23:20.48 | p3nguin | It is considered to be not activated. iTunes wants to activate the phone. |
| 23:20.52 | SeRi | p3nguin, well you will have to jail brake it first |
| 23:21.07 | p3nguin | I don't know if I can do that, even. |
| 23:21.21 | p3nguin | I don't have any way to find out what iOS version is on it. |
| 23:21.43 | SeRi | p3nguin, you should be able to since you dont eve need the phone to be communicating with itunes |
| 23:22.02 | SeRi | now thats an issue :/ |
| 23:22.05 | p3nguin | What jailbreak app do you suggest I try first? |
| 23:22.28 | SeRi | I have been very succesfull with pawnage |
| 23:22.40 | p3nguin | pwnage tool? |
| 23:22.58 | SeRi | http://blog.iphone-dev.org/post/4332841631/three-years-of-pwnage-tool |
| 23:23.06 | SeRi | yes that :) |
| 23:23.55 | SeRi | There is another one that's very good.... let me check my history one sec. |
| 23:24.50 | p3nguin | What about redsn0w? |
| 23:26.47 | SeRi | redsn0w is load it after its been jailbroken |
| 23:27.26 | Greenlight | Evening folks, am getting a strange issue when I'm using the AMI to bridge two channels. The bridge works correct and the channels are connected and can speak, but when either hangup the other starts to ring again and in the CLI I get the message "putting chan <CHANNEL> back into PBX again" |
| 23:27.33 | Greenlight | Any ideas why this is happening? |
| 23:28.18 | p3nguin | seri: No, that's ultrasn0w. |
| 23:28.20 | SeRi | p3nguin, I would really give pwnage tool a try... |
| 23:28.26 | SeRi | ah yes. sorry thats true |
| 23:28.29 | p3nguin | redsn0w is a jailbreaking app. |
| 23:28.34 | SeRi | true. |
| 23:28.39 | SeRi | some times I get them mix. |
| 23:29.05 | SeRi | p3nguin, http://www.redsn0w.us/2011/04/preserve-iphone-4-432-baseband-unlock.html |
| 23:29.07 | p3nguin | I'm trying to figure out which version I should use of pwnagetool and/or redsn0w. |
| 23:29.25 | *** part/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 23:30.04 | SeRi | never used redsn0w here... but I dont see why it wouldnt work. |
| 23:30.09 | Greenlight | Is there a setting somewhere that controls if bridge channels are put back into the PBX, or something like that? |
| 23:30.58 | p3nguin | The first page you linked me to said that redsn0w is the easier-to-use incarnation of pwnagetool. |
| 23:32.05 | SeRi | Yes i was just reading that. Give it a try. |
| 23:32.20 | SeRi | Yours is a 3G or 3GS? |
| 23:34.41 | SeRi | p3nguin, ^^ |
| 23:34.45 | *** join/#asterisk jmwpc (~jmwpc@c-24-5-58-60.hsd1.ca.comcast.net) |
| 23:34.59 | p3nguin | 3G |
| 23:35.41 | SeRi | Mhhhhhhhh...... I think you are still in 3.x |
| 23:35.47 | p3nguin | But I still have no way to know what iOS version I have. |
| 23:35.47 | SeRi | I doubt you are in 4.0 |
| 23:35.52 | p3nguin | Maybe. |
| 23:36.10 | p3nguin | If I am in 3, I have five choices. |
| 23:36.46 | SeRi | p3nguin, the worst that it can help is that it will fail. It will not downgrade unless you are jail broken first so it will fail. |
| 23:37.01 | SeRi | I think you are safe to give it a try |
| 23:37.10 | SeRi | I would try 3.x firsth sthough |
| 23:41.36 | SeRi | is loving Androids 2.3 Native SIP :D |
| 23:43.00 | *** part/#asterisk libryder (~david@209.33.214.243) |
| 23:44.54 | SeRi | p3nguin, you like dubstep, trance, or chill? |
| 23:47.02 | SeRi | p3nguin, http://www.iphonehacks.com/jailbreak_iphone <---- good site |
| 23:55.23 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:56.12 | Greenlight | Can't for the life of me work out why a channel would start ringing again when the other side has hungup, anyone got any ideas at all as to why this would happen? |
| 23:56.51 | *** join/#asterisk Dovid (42570475@gateway/web/freenode/ip.66.87.4.117) |
| 23:58.23 | SeRi | Greenlight, that happens to me when I use a specific sip client over 3G connecting directly to Asterisk.... I found that the sip client is loosing so many packets that Asterisk never got the "HANGUP" |
| 23:58.56 | SeRi | I for got the actual name of the update it sends... :/ |
| 23:59.39 | Greenlight | It's getting the hangup okay, can see that in the CLI, but it's like for some weird reason it thinks it should send the other side of the call back to start ringing somewhere |
