| 00:00.50 | WIMPy | How many hours/days? |
| 00:02.04 | vlad2 | lol |
| 00:02.09 | vlad2 | it's a few lines of code |
| 00:02.12 | vlad2 | to edit the outgoing invite packet |
| 00:02.13 | vlad2 | that i know |
| 00:03.48 | WIMPy | Then do it. |
| 00:04.29 | WIMPy | I'd prepare for 1. multiple locations that need changing and 2. bad side effects. |
| 00:06.28 | vlad2 | i'm not a coder |
| 00:09.10 | WIMPy | And that's only for a diry hack. |
| 00:11.51 | iprouteth0 | sed is your friend |
| 00:17.12 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
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| 00:54.33 | *** join/#asterisk apb1963 (~apb1963@174.134.117.244) |
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| 01:06.51 | phix | I have a tdm card installed however since upgrading to 1.8 (from 1.6) I need to hold keys down longer in order for asterisk to register they were pressed. Also people are telling me they are getting echo on their end, I tried lowering the volume now the people can hardly hear me and it is still echoy |
| 01:07.02 | phix | tips? solutions? places to look / check? |
| 01:10.54 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 01:15.39 | *** join/#asterisk gnudna (~sklav@unaffiliated/sklav) |
| 01:16.01 | gnudna | hey guys maybe someone can point me in the right direction. |
| 01:16.16 | gnudna | im seeing a few of these error in notice.log |
| 01:16.33 | gnudna | chan_sip.c: Call from '' to extension '011972597135095' rejected because extension not found in context 'Default' |
| 01:16.57 | gnudna | what type of call attempt is this? |
| 01:17.11 | gnudna | since call from is blank ' ' |
| 01:19.12 | gnudna | i also see app.c: Huh....? no dial for indications? |
| 01:19.34 | gnudna | all is working but im trying to keep the system clean and 2 make sure no funny business |
| 01:22.17 | *** part/#asterisk bandroidx (~bandroidx@205.185.117.117) |
| 01:28.03 | vlad2 | fixed it |
| 01:34.29 | *** join/#asterisk Quest (~syncsys@175.110.23.215) |
| 01:45.16 | Katty | hello my asterisk does not work at all how to fix plz?? is urgent only srs answers thx. |
| 01:49.22 | apb1963 | Katty, if you watch LeXX it will explain how to fix asterisk |
| 01:50.54 | Katty | i didn't like lexx :< |
| 01:51.09 | Katty | it was all akward softcore porn |
| 01:51.17 | apb1963 | did you watch all of it, or just like the first episode? |
| 01:51.24 | Katty | first couple seasons |
| 01:51.36 | apb1963 | ok |
| 01:51.45 | apb1963 | at least you gave it a chance |
| 01:51.48 | Katty | gave it a shot. just not my thing. |
| 01:52.05 | apb1963 | I guess we just have different tastes |
| 01:52.14 | Katty | yesh, and that's ok! |
| 01:52.16 | ChannelZ | I'm re-watching Game of Thrones season 1 for my softcore porn |
| 01:52.17 | apb1963 | Hard to believe, I know. |
| 01:52.34 | Katty | yes i know. we can disagree and get along. whodathunkit?! |
| 01:52.50 | Katty | ChannelZ: gonna dig into that next, me thinks |
| 01:52.57 | Katty | ChannelZ: lot of season to catch up on >.< |
| 01:53.32 | apb1963 | well... if it weren't for the sex, I think we'd of broken up a long time ago. |
| 01:53.42 | ChannelZ | Yeah. I'm starting over becaue I can't remember who anyone is. |
| 01:54.21 | Katty | hmm, yeah. i'm not sure i have a good handle on everyone either. i know a few tho. |
| 01:54.30 | Katty | the major ones. |
| 01:55.04 | ChannelZ | Everyone has broken off into so many factions I just couldn't keep everything straight |
| 01:55.22 | apb1963 | Game of Thrones... that's the one with the military kid that save's the prince's life and has the hots for the princess right? |
| 01:55.56 | ChannelZ | Not so much, no |
| 01:55.59 | Katty | that doesn't sound like it t ome....but |
| 01:56.03 | Katty | i only watched the first season or so |
| 01:56.06 | apb1963 | no? Hmmm |
| 01:56.32 | apb1963 | so the guy didnt walk out like in front of tank and save the prince? |
| 01:56.37 | apb1963 | +a |
| 01:56.40 | Katty | no tanks. |
| 01:56.42 | ChannelZ | There are no tanks in Game of Thrones. |
| 01:56.50 | Katty | it's the one with the hot blonde and the dragons |
| 01:56.59 | Katty | and the family with all the wolves, where the little boy gets hurt |
| 01:57.03 | ChannelZ | It's more Lord of the Rings-timey |
| 01:57.18 | apb1963 | I don't think I know that one |
| 01:57.22 | ChannelZ | HBO |
| 01:57.39 | Katty | also netflix. |
| 01:57.41 | apb1963 | Humphrey Bogart's Orifice? |
| 01:57.42 | Katty | it's huge in pop culture. |
| 01:57.48 | Katty | winter is coming, and all |
| 01:58.06 | ChannelZ | Winter is still here, it's starting to snow. |
| 01:58.29 | ChannelZ | Colorado and all.. where every day is a meterological excitement |
| 01:58.46 | apb1963 | Older girl volunteers to take her younger sister's place in "the game" or something? |
| 01:58.59 | Katty | no that's hunger games. |
| 01:59.04 | ChannelZ | That's Hunger Games |
| 01:59.04 | apb1963 | right right |
| 01:59.19 | apb1963 | Have you seen Hunt for Gollum? |
| 01:59.31 | Katty | no |
| 01:59.34 | Katty | don't spoil it! |
| 01:59.44 | apb1963 | But you've heard of it |
| 01:59.44 | ChannelZ | I'm not sure what that is |
| 01:59.59 | apb1963 | <spoiler alert coming> |
| 02:00.01 | Katty | no i've not heard of it |
| 02:00.05 | apb1963 | It's where they Hunt For Gollum |
| 02:00.06 | Katty | nooo spoilers |
| 02:00.11 | Katty | <PROTECTED> |
| 02:00.14 | Katty | you're awful! |
| 02:00.18 | apb1963 | yeah |
| 02:00.19 | ChannelZ | Is this a fan thing? |
| 02:00.24 | apb1963 | it IS! |
| 02:00.37 | ChannelZ | Maybe I did see it. |
| 02:00.48 | ChannelZ | It's a few years old yes? |
| 02:00.53 | apb1963 | how about Star Trek New Voyages? |
| 02:01.16 | apb1963 | with Kirk played by a different actor? |
| 02:01.36 | iprouteth0 | word |
| 02:01.36 | apb1963 | probably. I don't pay attention to reelease dates or anything |
| 02:01.39 | *** join/#asterisk Bradada (~Bradada@59-120-137-34.HINET-IP.hinet.net) |
| 02:01.54 | iprouteth0 | apb1963: did you see JJ abrams star trek reboot? |
| 02:02.07 | apb1963 | reboot? |
| 02:02.11 | apb1963 | word to your mama |
| 02:02.27 | apb1963 | I don't know anything about "reboot" |
| 02:02.35 | ChannelZ | the last movie |
| 02:02.45 | apb1963 | not sure |
| 02:03.31 | ChannelZ | The prequil, how Kirk/Spock/et all got together |
| 02:03.36 | ChannelZ | Lots of lens flares |
| 02:03.37 | apb1963 | no! |
| 02:03.59 | ChannelZ | http://www.imdb.com/title/tt0796366/?ref_=fn_al_tt_1 |
| 02:04.02 | apb1963 | I figured they kinda met on the ship or perhaps even at star school |
| 02:04.55 | ChannelZ | A bit of both |
| 02:04.59 | apb1963 | I do not believe I've seen that one |
| 02:05.27 | apb1963 | I've been busy |
| 02:05.37 | apb1963 | with... things. |
| 02:05.40 | apb1963 | and.... stuff. |
| 02:06.41 | apb1963 | I've reviewing warehouse 13 |
| 02:07.02 | ChannelZ | Well you shoulod rent it, get caught up for the new one next month |
| 02:07.15 | apb1963 | new.... one? |
| 02:07.56 | apb1963 | they really must give me time to catch up. what with Haven and Supernatural and Once Upon a Time and others.... who has time for movies? |
| 02:08.15 | ChannelZ | Well the last one was 2009 :/ |
| 02:08.42 | apb1963 | well... I suppose that's reasonable |
| 02:08.50 | apb1963 | there's just so much |
| 02:09.03 | apb1963 | Hell... I just finished the Marx Brothers movies. |
| 02:09.21 | apb1963 | There's like 10 of those |
| 02:09.52 | ChannelZ | hehe |
| 02:10.51 | apb1963 | well, back to the grind |
| 02:10.56 | apb1963 | Have to keep my nose to the grindstone |
| 02:11.02 | apb1963 | and my shoulder to the wheel |
| 02:11.06 | apb1963 | and my ear to the ground |
| 02:11.11 | apb1963 | very uncomfortable |
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| 02:47.36 | Quest | anybody home? |
| 02:53.57 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 02:59.35 | Quest | helo.. |
| 03:03.13 | Quest | one question: to get requests on port 80 of a webserver, 80 needs to be forwarded or NAT 1:1. while asterisk server dont need NAT or port forwarding? as it has problems with NAT? |
| 03:04.41 | *** part/#asterisk gnudna (~sklav@unaffiliated/sklav) |
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| 03:14.17 | Quest | by router i mean router/switch. the server is a linux ubuntu server |
| 03:14.17 | Quest | we have 8 ip pool 5/8 usable. our office configuration DSL > router (make NAT maping for server) > server > switch > mulptiple users. The server is for monitering, prioritizing and limiting traffic. Now we also want an astersk server to also be placed (after the switch and before the the voip cisco phones). We want to moniter voip calls and bandwidth as well. I have been advised by many not to use NAT with astersk. I have read a |
| 03:14.21 | Quest | rticles on the problems. One other problem is that voip need priority. can the server set priority for viop? Provided that the phones will only be used in LAN. should I proceed or an alternative is suggested? |
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| 03:57.45 | fling | I want to somehow to play busy tones to some sip clients instead of just sending Busy(). |
| 03:58.00 | fling | How to do this without a major dialplan rewrite? :p |
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| 04:30.09 | ChannelZ | fling: PlayTones |
| 04:31.28 | [TK]D-Fender | fling: Answer the call first |
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| 06:29.31 | apurvtwr | hi |
| 06:30.41 | apurvtwr | I have an asterisk 1.8 installed on which I have a number of dial plans.. |
| 06:32.34 | apurvtwr | some of these dial plans have a SIP number with it's follow me (mobile number). Although the calls are correctly made on both the numbers, I am unable to distinguish if the call was answered on the SIP number or the follow me number |
| 06:34.06 | apurvtwr | can someone help me with that . thanks |
| 06:41.32 | *** join/#asterisk joesmoe (~joesmoe@c-76-112-110-95.hsd1.mi.comcast.net) |
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| 06:55.44 | elmargol | Receiving faxes takes long and I get this error message: app_fax.c:173 span_message: WARNING T.30 ECM carrier not found |
| 06:57.40 | elmargol | app_fax.c:432 transmit_audio: channel 'SIP/pstn-00000000' refused to negotiate T.38 |
| 06:58.56 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
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| 07:07.26 | ChannelZ | pigeon died |
| 07:13.57 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-exdfuitmrysqbjkl) |
| 07:14.26 | lanning | you sure it is not pining for the fjord? |
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| 07:28.06 | ChannelZ | heh |
| 07:34.05 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
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| 07:50.19 | apurvtwr | In case of an astertisk queue, is there way to find out if the call was answered on the SIP number or the followme? |
| 07:51.44 | ectospasm | CDR or CEL should be able to tell you that |
| 07:51.56 | ectospasm | though I don't know exactly how, off the top of my head. |
| 08:18.53 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 08:22.28 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 08:22.40 | linocisco | hi all, have a good day |
| 08:23.17 | linocisco | may I know how to autoprovision or configure sip profile on cisco IP phones from the asterisk server without touching cisco phones? |
| 08:34.22 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
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| 08:53.03 | *** join/#asterisk jacobw (~jacobw@unaffiliated/jacobw) |
| 08:53.28 | jacobw | Hi, when dialing from AGI, can I set which codec to use, or prefer? |
| 08:56.32 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 08:57.04 | linocisco | may I know how to autoprovision or configure sip profile on cisco IP phones from the asterisk server without touching cisco phones? |
| 08:59.46 | linocisco | where can I download endpoint manager for asterisk? |
| 09:03.26 | kaldemar | jacobw: set one of the SIP_CODEC variables: https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables |
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| 09:12.18 | *** join/#asterisk Schreda (~Schreda@88.116.251.174) |
| 09:12.29 | Schreda | hey |
| 09:12.34 | Schreda | guys... |
| 09:12.38 | Schreda | I have a question... |
| 09:12.41 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 09:12.53 | Schreda | I setup an asterisk 1.6 server on ubuntu... installed all stuff... |
| 09:13.00 | Schreda | but now I run into some problem... |
| 09:13.25 | Schreda | I m already able to do phone calls through ISDN and I also can receive them ... |
| 09:13.41 | Schreda | I used the [from-pstn] |
| 09:13.46 | Schreda | context for incoming calls |
| 09:13.47 | Schreda | ... |
| 09:14.13 | Schreda | but the problem is that I cant collect the extension which was dialed by the incoming call ... |
| 09:14.41 | Schreda | The s@from-pstn... doesnt give me number information... |
| 09:14.54 | Schreda | Can someone help me plz about this issue? |
| 09:15.02 | Schreda | Thanks a lot in advice |
| 09:15.04 | Schreda | :D |
| 09:19.46 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:a42:224:1dff:fecd:234c) |
| 09:25.17 | tzafrir | Schreda, what channel driver do you use for the ISDN? |
| 09:27.55 | linocisco | hi all , i am trying to configure asterisk with cisco ip phone 7942G with SIP. I would like to know how to configure soft button on phone to record memory like the frequently dialed extensions or FXO lines |
| 09:37.46 | Schreda | I use dahdi and libpri |
| 09:38.15 | Schreda | zaptl is not working on sangoma B700 card |
| 09:40.28 | tzafrir | Schreda, DAHDI will not work with Asterisk > 1.4.x , and DAHDI is the latest Zaptel |
| 09:41.04 | Schreda | but Im able to receive calls and do calls ... |
| 09:41.06 | tzafrir | The ISDN MSN should basically be the extension number |
| 09:41.14 | Schreda | into the pstn |
| 09:41.46 | tzafrir | That is, if you have in chan_dahdi.conf (in the right place) context=from-pstn ;, |
| 09:42.30 | tzafrir | and you called to number 12345678 that gets routed to your ISDN card, |
| 09:43.09 | Schreda | its working fine |
| 09:43.13 | Schreda | My problem is:+# |
| 09:43.16 | tzafrir | you will get a new channel at 12345678@from-pstn (extension: 12345678, priority: 1, context: from-pstn) |
| 09:43.38 | Schreda | e.g |
| 09:44.04 | Schreda | for my [from-pstn] |
| 09:44.05 | Schreda | conetext |
| 09:44.08 | linocisco | hi all , i am trying to configure asterisk with cisco ip phone 7942G with SIP. I would like to know how to configure soft button on phone to record memory like the frequently dialed extensions or FXO lines |
| 09:44.19 | Schreda | I defined |
| 09:44.37 | Schreda | exten=>s,1,Answer() |
| 09:44.54 | Schreda | exten=>s,n,Dial(SIP/0) |
| 09:44.54 | tzafrir | linocisco, we saw that. No need to repeat that so often. |
| 09:45.08 | Schreda | thats workijng fine |
| 09:45.23 | Schreda | so the person will be connected directly to my internal sip phone... |
| 09:45.31 | Schreda | but I need something like: |
| 09:45.49 | Schreda | exten=>s,n,Dial(SIP/CalledNumber) |
| 09:45.53 | tzafrir | needs to work on the Queue of this channel |
| 09:46.30 | Schreda | so that I can forward the call to extensions |
| 09:46.37 | tzafrir | If this is ISDN (digital), it is generally wrong of you to get to 's'. 's' is the fallback |
| 09:46.59 | Schreda | and how to fix this ? |
| 09:47.11 | tzafrir | Do something like: |
| 09:47.45 | Schreda | which files I need to change just extensions.conf or something else?= |
| 09:47.51 | tzafrir | exten => _X.,1,NoOp(Got a call to number ${EXTENSION}) |
| 09:48.05 | tzafrir | This will show you the number that was dialed |
| 09:48.23 | Schreda | I will try it |
| 09:48.24 | tzafrir | Or if you want some generic processing, use '_X.' |
| 09:48.54 | tzafrir | (my IRC client does not print the underscore character before the 'X' |
| 09:48.56 | tzafrir | ) |
| 09:49.07 | Schreda | can I I tried yetserday _X.,1,Dial(...) but I got the error that its unknown... |
| 09:49.18 | Schreda | like s@from-pstn... |
| 09:49.32 | Schreda | I need to go to a meeting |
| 09:49.35 | Schreda | 30 mins im back |
| 09:49.46 | tzafrir | What did you have in the Dial? Dial(sip/trunkname/${EXTENSION}) ? |
| 09:49.52 | Schreda | sorry |
| 09:50.02 | Schreda | Im back soon thanks already for your ghelp |
| 09:50.04 | Schreda | !! |
| 09:50.05 | tzafrir | In some cases the extension number may need some manipulations |
| 09:50.39 | wdoekes | ${EXTEN} |
| 09:50.52 | tzafrir | Oops, sorry. You're right. |
| 09:51.36 | wdoekes | (and X! makes more sense than X., my pet peeve ;) ) |
| 10:08.11 | jacobw | kaldemar: I apologise for my late response, where do I set that? |
| 10:15.08 | *** join/#asterisk vedic (~V@49.206.62.216) |
| 10:15.36 | kaldemar | jacobw: those are channel variables. set it in the agi or in your dialplan before the agi. |
| 10:17.08 | vedic | Hi Guys, I am new to AMI but have some understanding of Asterisk. I am able to login using AMI. I have a pstn line connected to server (incoming call works). I want to try calling out using AMI. But when I issue originate action, I get error saying Originate failed. How to originate a PSTN call to an outside number? |
| 10:17.33 | Greenlight | It should give a reason as to why it failed? |
| 10:17.58 | Greenlight | Can you pastebin the Origination command you're sending? |
| 10:17.59 | vedic | Greenlight: No, I just says: Response: Error\r\nMessage: Originate failed |
| 10:18.00 | ghost75 | somebody know if the cisco default ringtone can be changed by xml config? |
| 10:18.05 | Greenlight | ~pb |
| 10:18.05 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 10:18.08 | Greenlight | ^^^ |
| 10:18.30 | eirirs | like pastie.org |
| 10:19.08 | vedic | Greenlight: Action: Originate\r\nContext:from-pstn\r\nChannel:DAHDI/il/09876543210\r\nExtension:665544321\r\nPriority:1 |
| 10:19.39 | Greenlight | Ok, looks somewhat FreePBXy, but try context "from-internal" |
| 10:20.09 | vedic | Greenlight: I doubt, is this context the same as I have in chan_dahdi? or its the one I have in extensions.conf ? |
| 10:20.19 | vedic | I doubt => One doubt |
| 10:20.25 | Greenlight | Are you using FreePBX? |
| 10:20.32 | vedic | Greenlight: nope |
| 10:20.40 | vedic | Greenlight: manually installed asterisk |
| 10:20.42 | Greenlight | Ah, just when i seen "from-pstn" |
| 10:21.13 | Greenlight | In that case, the context in extensions.conf where your 665544321 destination is |
| 10:21.13 | vedic | Greenlight: 1 doubt, is this context the same as I have in chan_dahdi? or its the one I have in extensions.conf ? |
| 10:22.10 | vedic | Greenlight: I tried with that. chan_dahdi is defined inside [default] ... context=from-pstn ... and then in extensions.conf I have [from-pstn] exten=>6655321, 1, Answer() |
| 10:22.13 | Greenlight | Also, try "Exten" instead of extnesion |
| 10:22.26 | vedic | Greenlight: Tried that as well |
| 10:22.58 | Greenlight | Pastebin your dialplan, and the CLI output when you try to Originate a call |
| 10:23.34 | ghost75 | anyone using activa? |
| 10:23.40 | vedic | Greenlight: CLI doesn't give any output except saying that user so and so connected with 127.0.0.1 . When I issue any other action, there is no output on CLI |
| 10:23.44 | Greenlight | Also, maybe it was just a type, but 6655321 != 665544321 |
| 10:23.57 | Greenlight | *typo |
| 10:24.32 | vedic | Greenlight: no, thats fine. Not typo |
| 10:24.52 | Greenlight | [11:22am] <vedic> Greenlight: I tried with that. chan_dahdi is defined inside [default] ... context=from-pstn ... and then in extensions.conf I have [from-pstn] exten=>6655321, 1, Answer() |
| 10:25.01 | Greenlight | You're originating to 665544321 |
| 10:25.39 | vedic | Greenlight: Sorry!, that was typo writing here but not when I tried |
| 10:25.44 | Greenlight | Ahh ;) |
| 10:28.18 | Greenlight | It should call the DAHDI channel first, then call the other side. So I suspect that's where the problem is |
| 10:28.36 | Greenlight | Does a normal Dial() application work okay, when calling that DAHDI cahnnel ? |
| 10:28.39 | Greenlight | *channel |
| 10:52.08 | linocisco | how to install EPM on asterisk? |
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| 11:37.52 | jacobw | kaldemar: thanks |
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| 11:58.45 | elmargol | Can someone explain to me how I can get the number a fax came from? |
| 11:59.00 | ipalmer | Hi all, I've set up 2 asterisk boxes with back to back ISDN, I have an ISDN crossover cable connected between the 2, the sangoma cars are setup as pri net and pri cpe. When I try and send calls down the dahdi channel I'm getting congestion in the asterisk cli. I have confirmed that the sangoma cards are connected. Anyone have any idea why this is happening or have exeprience doing back to back config |
| 11:59.09 | WIMPy | vedic: That does not look like a valid channel name. |
| 11:59.10 | elmargol | my dedicated fax machine writes the incoming number on the paper |
| 11:59.36 | elmargol | If I use spa somehow I can't forward the callerid to sip :( |
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| 12:00.14 | WIMPy | ipalmer: What have you checked so far? |
| 12:00.45 | vedic | WIMPy: Channel: Dahdi/il/09876543210 ? |
| 12:01.40 | WIMPy | vedic: Yes. "il" does not look like a valid channel or group. |
| 12:02.15 | ipalmer | WIMPy: I've checked that the cards are connected and data packets are being sent and received, I've removed the cable and it raises alarms as expected, after that I'm a little lost where to go |
| 12:02.17 | vedic | WIMPy: When I get incoming call, my agi sets the channel like this. Thinking that, I have tried in the same fashion |
| 12:02.54 | vedic | WIMPy: In agi channel variable, I get the value as :DAHDI/il/09876543210 |
| 12:03.18 | WIMPy | ipalmer: Ok, so a pri debug sounds like a good idea. |
| 12:03.21 | vedic | Opps! It is sending Smiles |
| 12:03.23 | WIMPy | vedic |
| 12:03.25 | WIMPy | oops |
| 12:03.38 | ipalmer | WIMPy: ok thanks |
| 12:03.54 | WIMPy | vedic: The i names can not be used. And they shouldn;t be followed by a letter AFAIK. |
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| 12:04.45 | vedic | WIMPy: Does AMI depends on any other module function or app in modules.conf ? I have disabled by default to all the modules I have not been using till now. |
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| 12:05.09 | vedic | WIMPy: Just thought if that could also be the reason |
| 12:07.10 | WIMPy | vedic: Use a valig group you have defined in chan_dahdi.conf. |
| 12:08.21 | Greenlight | You can test by using Dial with the channel name you're trying to dial over |
| 12:08.54 | vedic | WIMPy: http://pastebin.com/nJeY01pZ |
| 12:10.07 | ipalmer | WIMPy: I've pasted my PRI debug file here http://pastie.org/7384525 |
| 12:10.09 | WIMPy | Is that somehow related to the channel name? |
| 12:10.54 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 12:11.01 | Greenlight | vedic: How do you dial out normally on the system, when you don't use the AMI? |
| 12:12.11 | vedic | WIMPy, Greenlight: http://pastebin.com/sADV3psU This is my chan_dahdi.conf . As of now I haven't tried Dial command. |
| 12:12.34 | Greenlight | Lets not run before you can walk then, you should get normal dialling out working first. |
| 12:13.15 | Greenlight | YOur channels are in group 0 by the looks of it |
| 12:13.42 | vedic | WIMPy, Greenlight: ok, trying out Dial. I just wanted not to disturb extensions.conf . |
| 12:14.18 | vedic | WIMPy, Greenlight: ok so Channel: DAHDI/0/09876543210 should work be right |
| 12:14.19 | Greenlight | Why do you want to distrub extensions.conf? Especially if the system doesn't even have the ability to make calls? |
| 12:14.20 | WIMPy | ipalmer: looks like some issue at a lower level. So you have timing configured so that one side is master only? |
| 12:14.34 | Greenlight | *dont |
| 12:14.35 | [TK]D-Fender | <vedic> WIMPy, Greenlight: ok so Channel: DAHDI/0/09876543210 should work be right - no.. there is no channel 0 |
| 12:14.48 | [TK]D-Fender | vedic, channel =>1-15,17-31 <-------------------------------------- |
| 12:15.10 | [TK]D-Fender | vedic, tu use them as a group you use "g0" or "G0" |
| 12:15.12 | [TK]D-Fender | to* |
| 12:15.21 | WIMPy | vedic: You want to use a group, i.e. g0 G0 r0 or R0. |
| 12:15.23 | [TK]D-Fender | (depending on order) |
| 12:15.37 | vedic | Greenlight: I have been trying agi. I shall call to server and and will dial another pstn number. |
| 12:15.46 | vedic | [TK]D-Fender, WIMPy: I see |
| 12:15.49 | ipalmer | WIMPy: yes master is on the PRI NET side |
| 12:16.37 | WIMPy | ipalmer: Same switchtype? |
| 12:17.23 | ipalmer | WIMPy: I have EuroISDN on both, if that's what you meant |
| 12:17.37 | WIMPy | yes |
| 12:17.56 | WIMPy | But they are not talking to each other at all. |
| 12:18.49 | WIMPy | Is the line up on both sides? i.e. the correct type of cable? |
| 12:19.47 | ipalmer | WIMPy: the line is up on both sides, it's a ISDN crossover cable made using the pinouts from the Sangoma site, I have used this cable before successfully for this exact scenario |
| 12:20.51 | ipalmer | WIMPy: I'm going to try and run the configure options script again in case I've done something wrong |
| 12:20.56 | [TK]D-Fender | ipalmer, what are you connecting it to? |
| 12:21.30 | ipalmer | D-Fender: 2 asterisk boxes connected via 2 Sangoma A102 cards |
| 12:23.53 | vedic | WIMPy, [TK]D-Fender, Greenlight: I worked. Thank you very much. I just used g0 and it worked. |
| 12:25.33 | [TK]D-Fender | ipalmer, PB both sides DAHDI configs in full as well as the sangoma configs |
| 12:26.19 | ipalmer | D-Fender: Ok thanks, where are the sangoma configs kept, sorry |
| 12:28.37 | [TK]D-Fender | ipalmer, /etc/wanpipe |
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| 12:52.58 | ipalmer | D-Fender: PBX 1 here, this is the customer side http://pastie.org/7385024, second to follow |
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| 12:55.13 | vlad_starkov | Question: Anyone familiar with A2Billing knows what does this error mean? https://github.com/Star2Billing/a2billing/issues/47 |
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| 13:00.33 | ipalmer | D-Fender: PBX 2 here http://pastie.org/7385119 |
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| 13:04.07 | [TK]D-Fender | ipalmer, Looks sane. What's the error you get trying to use it? |
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| 13:07.19 | ipalmer | D-Fender: http://pastie.org/7385252 the full error in the asterisk cli |
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| 13:09.55 | [TK]D-Fender | ipalmer, And on the other side? |
| 13:10.48 | ipalmer | D-Fender: nothing at all |
| 13:11.13 | ipalmer | D-Fender: I'd expect something even if it was saying it didn't know what to do with the call |
| 13:11.16 | [TK]D-Fender | ipalmer, Is looking kinda link-layer... |
| 13:11.26 | [TK]D-Fender | ipalmer, "wanrouter status" <- both ends |
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| 13:11.46 | ipalmer | D-Fender: connected |
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| 13:13.12 | [TK]D-Fender | ipalmer, PB those along with "dahdi show status", "pri show span 1" (+2) for both sides |
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| 13:19.54 | ipalmer | D-Fender: hmmm wanpipe status is now showing disconnected on one side, I know it wasn't earlier dahdi show status is ok |
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| 13:53.27 | nricciar | i have a gosub that is called when a call in a queue is answered, but i am having trouble getting the callers phone number and the person who answered the call... both ${MEMBERINTERFACE} and ${CALLERID(num)} came back as blank... is there something else I can use for the same result? |
| 13:54.38 | jmetro | are they non-blank outside of the sub |
| 13:54.41 | jmetro | if so, pass them inside. |
| 13:54.58 | nricciar | you mean before I put them in the Queue? |
| 13:56.29 | jmetro | before the gosub is called |
| 13:56.59 | nricciar | also I wouldnt have all the info before the gosub is called as part of the information i need is the person who answered the call in the queue |
| 13:58.05 | jmetro | I'm finding queues to rapidly be more trouble than they are worth in asterisk and hand-coding with a loop might be a lot less of a hassle |
| 13:59.24 | jmetro | I'm guessing memberinterface is not populated pre-call pickup |
| 13:59.37 | nricciar | thats my understanding of the documentation |
| 14:00.20 | nricciar | http://pastebin.com/Zt2wPVH9 thats the setup i have atm in case it might help my situation, but like i said it seems like when call-connected is called both CALLERID and MEMBERINTERFACE are not populated |
| 14:00.48 | jmetro | Callerid should be in there, that you should be able to pass through via arguments |
| 14:01.44 | nricciar | yea that would fix half my problem at least |
| 14:02.39 | jmetro | check the Queue application, i think there are timers that when expire, will run macros, and you could have those do what you are looking for - provided you dont need it before the call is completed |
| 14:03.02 | [TK]D-Fender | nricciar, Show us a sample call |
| 14:03.18 | nricciar | Executing [s@call-connected:1] Verbose("SIP/207-0000023a", "3,call from connected to member ") in new stack |
| 14:03.26 | [TK]D-Fender | nricciar, complete |
| 14:03.28 | [TK]D-Fender | ^ |
| 14:05.09 | ipalmer | D-Fender: have re-configured both Sangoma cards with wancfg_dahdi and all seems to be ok now, don't know what happened there, both cards were connected, then just one, then both were disconnected with alarms showing in asterisk cli. Re-configured and hey presto |
| 14:05.44 | nricciar | [TK]D-Fender, http://pastebin.com/LiCFyKWw ? |
| 14:06.15 | SuperNull | Hey guys, anyone know the SPA502G ? looking to figure out what the factory reset password is based off (we preprovision these and want to factory reset one..) |
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| 14:09.52 | [TK]D-Fender | nricciar, You need "setinterfacevar=yes" in each of your queue definitions... |
| 14:10.05 | jmetro | SuperNull: http://lmgtfy.com/?q=spa502g+default+password |
| 14:10.15 | SuperNull | thanks will look jmetro |
| 14:10.21 | SuperNull | derp. |
| 14:10.24 | SuperNull | i did that. buddy. |
| 14:10.25 | [TK]D-Fender | nricciar, As for CID, Verbose that before hitting the Queue as well.... |
| 14:10.35 | SuperNull | the default is 'reset' as #s. but its using something else.. |
| 14:10.55 | SuperNull | and im not sure where that something else is.. we attempted both admin level and user level passwords from the xml config with no success. |
| 14:10.57 | nricciar | [TK]D-Fender, ahh cool will give that a shot... figured it had to be something easy like that |
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| 14:15.20 | MLNoah | I'm working on setting up DUNDi/IAX between two peers. Peer A is running Open Source 10.7.0, Peer B is running Open Source SVN-branch-11-r373133. With IAX encryption=yes on both peers, B can call A via DUNDi/IAX, but when A tries to call B, the logs on both servers fill with "packet decrypt failed". Turning off encryption with peer A's iax.conf allows calls to complete. Any suggestions... |
| 14:15.22 | MLNoah | ...on where to look for the underlying issue? |
| 14:15.40 | nricciar | [TK]D-Fender, your suggestion worked... thanks for the help |
| 14:16.09 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 14:16.49 | [TK]D-Fender | nricciar, You're welcome. |
| 14:17.52 | vlad_starkov | Question: Could anyone recommend me billing system for Asterisk to use in production? |
| 14:17.57 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 14:18.11 | [TK]D-Fender | MLNoah, Last I checked, DUNDi doesn't use your peers to do the full dial and you wouldn't get to use all of the settings you'd have if you just used the peers like normal... |
| 14:18.21 | [TK]D-Fender | vlad_starkov, MS Excel |
| 14:19.08 | vlad_starkov | [TK]D-Fender: so how do you automate it to print invoices? :) |
| 14:19.37 | [TK]D-Fender | vlad_starkov, Plenty of books for that :) |
| 14:20.21 | vlad_starkov | [TK]D-Fender: MS Excel automation for VoIP for Dummies?) |
| 14:21.02 | [TK]D-Fender | vlad_starkov, Nothing VoIP about basic billing. That is CDR and your pricing structure. |
| 14:21.10 | [TK]D-Fender | vlad_starkov, Boring DB 101 |
| 14:21.47 | vlad_starkov | [TK]D-Fender: that is more than just calculating. LCR, auto-blocking, user payments etc |
| 14:22.11 | [TK]D-Fender | LCR isn't billing ... it's changing what it's going to cost.... |
| 14:22.28 | jmetro | Excel automation is also easy to do whether it be VBA or formula driven |
| 14:22.40 | [TK]D-Fender | Payments != billing. Payments = Accounts Receivable. |
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| 14:22.48 | [TK]D-Fender | Accpac :) |
| 14:24.54 | vlad_starkov | Ok. I just thought that "billing" is complete automated solution to make money on VoIP... |
| 14:25.18 | vlad_starkov | No I see that it's not :) |
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| 15:04.19 | itgrl | Good morning! |
| 15:04.28 | eirirs | it's a girl! |
| 15:04.45 | itgrl | hehe, last time I checked. :P |
| 15:06.06 | jmetro | not a girl, a grl. Propane or charcoal? |
| 15:08.13 | itgrl | I don't grill much, but prefer charcoal. |
| 15:08.28 | itgrl | I am running into an issue, and was hoping someone had some insight. |
| 15:08.31 | itgrl | I'm using realtime for some dynamic schedule lookups on my dnis table. I also have a script setup for some automated testing with SIPP. I have an execif to test if call is to be tested or not, and if it is tested I want to update a flag in the database. This flag now will update the wall board of stop lights to verify inbound numbers are working. (well at least hitting the dialplan) |
| 15:08.56 | itgrl | My question is, Is there a good realtime function or app these days to update the table, such as REALTIMEUPDATE ? |
| 15:09.33 | itgrl | I keep getting errors of no such app or function allowed. I've tried realtimeupdate and realtime_store in various functions. |
| 15:10.46 | itgrl | Or should I just go with fastagi at this point? |
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| 15:14.29 | willryder | How can you find out what sip provider a telephone number terminates/originates at? |
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| 15:16.35 | itgrl | Are these sip providers located on your Asterisk box? |
| 15:16.40 | willryder | no |
| 15:17.19 | willryder | In other words, if I wanted to know who was handling a specific number. |
| 15:17.56 | itgrl | Well clecs have the capability to look that information up. I haven't been able to find something completely reliable to do it outside of a clec. |
| 15:19.03 | willryder | If you wanted to find out for instance who was the phone service provider for a particular companies maine number |
| 15:19.10 | willryder | How would you go about doing that? |
| 15:20.58 | itgrl | I've found there are a few APIs out there, I'm sure a paid service. |
| 15:21.03 | itgrl | http://www.telapi.com/docs/api/rest/carrier-services/carrier-lookup/ |
| 15:23.37 | itgrl | Tropo may also be able to help with something like that. |
| 15:27.47 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 15:32.14 | *** join/#asterisk MarKsaitis (~MarKsaiti@2.217.210.210) |
| 15:34.38 | *** join/#asterisk MalMen (~MalMen@2.83.136.155) |
| 15:34.53 | MalMen | hello, status: UNREACHABLE that means that my trunk is down, right ? |
| 15:35.17 | MalMen | if i show the registry that tell me that hi is registed |
| 15:37.40 | itgrl | I often see unreachable when my register expiry is at 1 hour, and one of my agents closes eyebeam. The registration will stay until it expires, but there is no path to the destination so it is unreachable. |
| 15:39.46 | [TK]D-Fender | MalMen, Means you hit a qualify time-out |
| 15:39.58 | [TK]D-Fender | MalMen, Could mean they have a problem, could mean you have one. |
| 15:40.24 | [TK]D-Fender | MalMen, Go show us a registration attempt with SIP DEBUG enabled. |
| 15:40.30 | [TK]D-Fender | MalMen, "sip set debug on" |
| 15:40.36 | [TK]D-Fender | ~pb |
| 15:40.37 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:40.38 | [TK]D-Fender | ^^^ |
| 15:41.36 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:41.37 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 15:41.56 | itgrl | Asterisk 1.8.11-cert9 Is there currently a way to update database via realtime? I can't find any good documentation on it. |
| 15:42.20 | navaismo | define "update the database" |
| 15:43.02 | itgrl | I have a field that I want to change No to Yes for a specific entry. |
| 15:43.37 | navaismo | using MYSQL? |
| 15:43.38 | itgrl | this would be used in a test suite that will automate testing of phone numbers for DS3 cutover |
| 15:43.40 | itgrl | yes, mysql |
| 15:44.04 | itgrl | I have the realtime setup and working just fine for my dnis lookups and schedules. |
| 15:44.27 | navaismo | well you can run a Query from, the mysql cli, phpadmin, adminer.php, or the dialplan itself |
| 15:45.34 | itgrl | I wish to do it from the dialplan, for the test logic. I am moving DS3s to a Sonet ring and have 715 numbers to test. I want to verify each number actually hits dialplan logic. So we have a stop light setup to display green when calls hit dialplan. |
| 15:46.27 | itgrl | How have I overlooked the mysql cmd hmmm |
| 15:46.37 | *** join/#asterisk HolTech (~IceChat77@pool-74-107-104-2.bltmmd.fios.verizon.net) |
| 15:47.05 | HolTech | anyone have any helpful advice or links, hoping to unlock and use with * an 8x8 video phone DV326 |
| 15:49.28 | itgrl | Thank you navaismo |
| 15:49.57 | itgrl | Not sure how I overlooked the MySQL dialplan application. I could have used it in several cases instead of agi. |
| 15:50.31 | navaismo | well if i recall its deprecated the app_mysql |
| 15:51.22 | navaismo | but still working |
| 15:51.53 | itgrl | well I know cdr_mysql is depreciated but working. haven't noticed anythign on app_mysql though |
| 15:52.18 | itgrl | but if it is working, it will serve this immediate need. |
| 15:52.55 | itgrl | I don't want to test every # But if I didn't and one broke, then I'm sure I would hear about it. So I'll script it. |
| 15:53.10 | *** join/#asterisk eureka^ (~vyatta@vpnc.sfd.asgdc.net) |
| 15:53.11 | navaismo | good |
| 15:55.26 | eureka^ | trying to figure out an issue where a redirected call is silent |
| 15:56.28 | eureka^ | incoming call hits physical endpoint, endpoint says 302 moved and redirects to an external number, when the external number picks up there's no audio at all |
| 15:56.37 | eureka^ | relevant error from logs: NOTICE[2293] chan_sip.c: Disconnecting call 'SIP/den-origin-0000089e' for lack of RTP activity in 31 seconds |
| 15:56.48 | eureka^ | there is *no* NAT involved here, PBX has both private and public addresses |
| 15:57.12 | eureka^ | not sure where i should start looking |
| 15:57.37 | navaismo | at the sip debug...maybe |
| 15:58.23 | eureka^ | yeah about to check that out now, just thought i'd pose the question while i try to figure it out |
| 16:00.26 | HolTech | ok then, does anyone know a good place to find firmwares ? |
| 16:00.45 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 16:01.17 | navaismo | google |
| 16:02.07 | HolTech | that's quite astute, silence or "No, I don't know" wouldbe preferred |
| 16:02.12 | navaismo | all hail the good God Google--->GGG |
| 16:03.24 | jmetro | and Googles prophet, lmgtfy |
| 16:03.42 | navaismo | right right |
| 16:15.55 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.50) |
| 16:17.42 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.126.83) |
| 16:17.43 | *** join/#asterisk Schreda (~Schreda@88.116.251.174) |
| 16:17.52 | Schreda | Hey everybody |
| 16:18.11 | Schreda | I got some trouble with my asterisk setup v 1.6 |
| 16:18.17 | Schreda | and maybe someone can help me... |
| 16:18.39 | Schreda | I m unable to call extension from outside.. |
| 16:18.53 | Schreda | internally I have SIP |
| 16:19.12 | Schreda | and I connected the asterisk server with the ISDN via an ISDN card |
| 16:19.21 | Schreda | Im already able to place calls and receive calls |
| 16:19.27 | itgrl | does the incoming call route to an auto attendant? |
| 16:19.38 | [TK]D-Fender | Schreda, 1.6 isn't a specific branch, let alone a version. Next we'll need proper detail about the networking involved in aal parts of this picture, and debug to match |
| 16:19.44 | [TK]D-Fender | ~pb |
| 16:19.44 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:19.48 | [TK]D-Fender | Schreda, ^^^ your friend |
| 16:21.09 | Schreda | i called my context for incoming calls [from-psdn] |
| 16:21.34 | [TK]D-Fender | Schreda, Show us the complete call attempt with SIP DEBUG enabled... |
| 16:22.22 | Schreda | just one sec... plz I want to explain one more thing... |
| 16:23.43 | *** join/#asterisk din3sh (29885030@gateway/web/freenode/ip.41.136.80.48) |
| 16:24.06 | Schreda | I used the following command to route through... exten=>s,1,Dial(SIP/Number) ... in this case I would use e.g 0 for example but to do an extension call I would need something like exten=>s,1,Dial(SIP/{EXTEN:-2}) to get the last 2 dialed digits..-.. |
| 16:24.40 | Schreda | but if I use something else than the s variable Im getting errors.. |
| 16:26.56 | *** join/#asterisk navaismo (~navaismo@189.241.36.80) |
| 16:27.40 | [TK]D-Fender | Schreda, exten=>s,1,Dial(SIP/{EXTEN:-2}) <- in this, ${exten} IS "s". So using it in there is useless. It does not hold a number. It holds the letter "s" |
| 16:27.51 | [TK]D-Fender | Schreda, You need to learn your dialplan basics... |
| 16:27.52 | [TK]D-Fender | ~book |
| 16:27.52 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:28.48 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:29.45 | *** join/#asterisk Foxi352 (~Foxi352@v-172-252.access.restena.lu) |
| 16:30.49 | din3sh | hello all |
| 16:30.51 | Schreda | but all the book examples are written without Dahdi |
| 16:31.28 | Schreda | and is it enought to read only the chapter with the dial plan? |
| 16:32.36 | jmetro | Dahdi is just the connector. Dahdi phones still get a peername yes? |
| 16:32.43 | jmetro | afaik thats all that would change. |
| 16:39.53 | MalMen | where is the file with the configurations of trunk ? |
| 16:40.09 | [TK]D-Fender | malcolmd, Too vague... |
| 16:40.25 | malcolmd | indeed |
| 16:57.24 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:00.24 | MalMen | what is the best windows applications to try connect to asterisk and see the log errors ? |
| 17:00.59 | navaismo | openssh |
| 17:01.10 | navaismo | for commodity putty |
| 17:01.18 | itgrl | I use mobaxterm |
| 17:01.41 | MalMen | i want a sip client |
| 17:01.51 | itgrl | xlite? blink? |
| 17:02.14 | navaismo | sip client != application to see logs, or Is another topic |
| 17:03.12 | navaismo | anyway for windows you have a bunch of good sip clients zoiper, bria, linphone and those that itgrl said |
| 17:05.58 | [TK]D-Fender | MalMen, putty |
| 17:06.06 | [TK]D-Fender | for SSH |
| 17:06.20 | [TK]D-Fender | MalMen, Go show us the SIP DEBUG I requested |
| 17:06.56 | [TK]D-Fender | actually.. bad aim on that... |
| 17:07.03 | [TK]D-Fender | But propabably still appropriate :) |
| 17:12.20 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:12.20 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:12.33 | jmetro | navaismo 3cx softphone is really good...and free |
| 17:17.48 | Jagster` | half way through this asterisk book :> |
| 17:18.08 | jmetro | have you configured your own server yet? =p |
| 17:20.33 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 17:22.04 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:25.35 | Schreda | Well Im done with the dialplan part... |
| 17:25.41 | Schreda | that stuff I knew already |
| 17:26.20 | Schreda | but the question is how can I handle a number without using the s@from-pstn... |
| 17:26.23 | *** join/#asterisk plantseeker (~Plantseek@77.240.63.97) |
| 17:26.38 | Schreda | the problem is I only can handle incoming calls by using the s var |
| 17:26.47 | *** part/#asterisk ipalmer (~IceChat77@host81-133-140-79.in-addr.btopenworld.com) |
| 17:27.03 | jmetro | Show us your dialplan |
| 17:27.33 | [TK]D-Fender | Schreda, You haven't properly told us what is sending these calls to you and what youw ant to do with them exactly... |
| 17:28.39 | Schreda | ISDN sends them... and right now I just want to discover the last two digits of the number,.,. |
| 17:28.47 | Schreda | but the variable s doenst store numbers |
| 17:28.56 | Schreda | its the fallback variable I think |
| 17:29.04 | navaismo | Schreda, if you dont know how to handle incoming calls then you need to read again the book |
| 17:29.05 | igcewieling | Schreda: if asterisk RECEIVES a number, then it will NOT use "s" |
| 17:29.27 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:a42:224:1dff:fecd:234c) |
| 17:30.05 | igcewieling | so. pastebin the output of pri debug span x and we'll see if your "ISDN" is sending the number |
| 17:31.44 | jmetro | Schreda - Do testing, setup your server, make a call to a number that will hit your server, record what its giving you, and handle it based on that. Dont let it fall into S |
| 17:31.45 | [TK]D-Fender | Schreda, ISDN should send to a targeted exten and you should not need S |
| 17:33.00 | Schreda | the problem is its not showing a nuber becasue I get the error the number doesnt exist on my phone if I change it to something like _X.,1.Dial(...) |
| 17:33.17 | Schreda | i tried 1000 different ways |
| 17:33.20 | igcewieling | Schreda: either follow our advice or ask elsewhere. We are TRYING to help you. |
| 17:33.22 | Schreda | but it only works with s |
| 17:33.35 | igcewieling | now get us a pri debug |
| 17:33.40 | jmetro | Schreda: when the error happens, it should tell you exactly what number was not found. |
| 17:33.41 | Schreda | k |
| 17:34.07 | *** join/#asterisk plantseeker (~Plantseek@77.240.63.97) |
| 17:34.35 | igcewieling | and a pastebin of the CLI of a failed call for jmetro |
| 17:35.02 | jmetro | [2013-04-09 12:34:26] NOTICE[11481]: chan_sip.c:23999 handle_request_invite: Call from 'mypbx' (nope) to extension 'ThisWontWork' rejected because extension not found in context 'incoming'. |
| 17:35.09 | jmetro | i dialed "ThisWontWork" and thats what i see in my errors. |
| 17:35.11 | Schreda | I read it already and just now once more |
| 17:35.21 | Schreda | <PROTECTED> |
| 17:35.33 | Schreda | this is the error if I try to use it without s |
| 17:36.10 | igcewieling | now all we need is the ISDN debug |
| 17:38.17 | Schreda | where to find it = |
| 17:38.18 | Schreda | ? |
| 17:38.29 | Schreda | this should be belongs to my sangoma card right? |
| 17:39.09 | igcewieling | Schreda: the cli. As I don't have any PRI boxes handy you'll have to do like everyone else and figure it out. try pri <tab> to start with and see what options it has |
| 17:39.16 | [TK]D-Fender | Schreda, Precisely what signalling is your card using? |
| 17:39.26 | igcewieling | [TK]D-Fender: ISDN! |
| 17:39.29 | igcewieling | 8-) |
| 17:39.50 | [TK]D-Fender | :p |
| 17:40.01 | Schreda | just to make things sure :D |
| 17:43.06 | Jagster` | lol trixbox |
| 17:43.19 | Jagster` | theres like 3 sip.conf's |
| 17:43.21 | Jagster` | one for asterisk , one for freepbx and one for trixbox |
| 17:43.25 | Jagster` | and all say just use the gui -_______________- |
| 17:43.58 | Schreda | im just using console :D |
| 17:44.13 | Schreda | but cant find the damn log folder |
| 17:44.19 | Jagster` | what system? |
| 17:44.23 | Jagster` | centos? |
| 17:44.36 | Schreda | ubuntu but its not under /var/log/messages |
| 17:44.46 | Jagster` | assuming linux use "locate <faile>" |
| 17:44.50 | Jagster` | err file* |
| 17:45.34 | *** join/#asterisk MLNoah (~chatzilla@noc.metalink.net) |
| 17:45.48 | Jagster` | to find the log file which one are you looking for? |
| 17:46.30 | Jagster` | Schreda: one of the logs is under /var/log/asterisk/full |
| 17:46.38 | Jagster` | just tail -f /var/log/asterisk/full |
| 17:46.54 | jmetro | jagster: youre using the book and installed a gui? =/ |
| 17:46.56 | [TK]D-Fender | Schreda, Use live CLI, not "logs". And answer my question about the signalling you're using and PASTEBIN the configs & calls. |
| 17:47.33 | Jagster` | jmetro: its trixbox the conf files literally say "use the gui" |
| 17:47.44 | Jagster` | im in console now though |
| 17:47.52 | Jagster` | havent touched the gui yet |
| 17:48.08 | jmetro | jagster: Dont use trixbox. Honestly, you will learn more and faster by doing it CLI and no GUI |
| 17:48.30 | jmetro | jagster: Or with a GUI OS just so you can have multiple windows open that are all CLI |
| 17:48.33 | *** join/#asterisk plantseeker (~Plantseek@77.240.63.97) |
| 17:48.35 | Jagster` | jmetro unfortunately ive been tasked on fixing an existing trixbox setup |
| 17:48.55 | Jagster` | only other pbx ive touched before was a panasonic black box sooo yea |
| 17:48.59 | navaismo | Jagster`, Trixbox IS DEAD |
| 17:49.09 | Jagster` | navaismo: i understand this, but this is a production system |
| 17:49.19 | navaismo | use freepbx or asterisknow |
| 17:49.33 | jmetro | navaismo: its not his, hes just a fixer not the builder |
| 17:49.34 | navaismo | and yes you should use the GUI instead the files |
| 17:49.50 | Jagster` | right now i'm on step 1: how does it work |
| 17:49.54 | Jagster` | step 2 is troubleshoot |
| 17:49.55 | Schreda | I just use asterisk 1.6 no freepbx... |
| 17:50.06 | jmetro | upgrade to 11 <.< |
| 17:50.16 | Schreda | problem I checked the log file there is nothing strange shown |
| 17:50.17 | Jagster` | also i heaer the upgrade from 1.4* to 1.8 is not fun |
| 17:50.26 | igcewieling | Jagster`: I wish you the best of luck, but you are unlikely to get any help here on Trixbox |
| 17:50.47 | Schreda | thats why I went back to 1.6 |
| 17:50.53 | Jagster` | igcewieling: im not looking for trixbox specific help... yet :P |
| 17:51.00 | Jagster` | im reading the asterisk manual :P |
| 17:51.13 | Schreda | me too since +2 weeks |
| 17:51.27 | jmetro | I have minimal experience with Trixbox, until my professor acknowledged that I could do all the labs faster and better in CLI |
| 17:51.57 | Jagster` | if you already are comfortable in cli thats the best jmetro |
| 17:51.59 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 17:55.45 | Schreda | well I will stop for today seemse my productivity is dropping a lot almost 16 hours.. |
| 17:55.58 | Schreda | cya and thanks for you help anyway |
| 17:56.05 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:56.31 | Jagster` | gl |
| 17:58.23 | *** part/#asterisk Schreda (~Schreda@88.116.251.174) |
| 18:04.18 | igcewieling | I don't see an actual question from you Jagster` |
| 18:09.26 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 18:17.10 | *** join/#asterisk RyanTG (~Thunderbi@mailer.americassuppliers.com) |
| 18:20.57 | *** join/#asterisk jzombie (~jzombie@cpe-76-169-175-121.socal.res.rr.com) |
| 18:22.43 | jzombie | hi all. i just found out about Asterisk. I'm looking for a business solution (1-4 users) that will auto-dial my contact list, play a pre-recorded message, prompt for the person to press a button/number, then forward the call to me. does Asterisk have this capability? is their a better solution? TIA |
| 18:23.24 | leifmadsen | jzombie: asterisk can easily do that |
| 18:23.42 | navaismo | jzombie, asterisk can do that, but you need to made it or buy some solution or hire someone to do that |
| 18:23.52 | jzombie | will i need a lot of programming knowledge? |
| 18:23.56 | leifmadsen | you just need to configure it to do that... it certainly is a more complex setup than just simple extensions, but it can definitely be done |
| 18:24.07 | jmetro | You could set it up in about a weekend depending on what configuration or equipment you have jzomb |
| 18:24.43 | jzombie | is it possible to build this for a single user and run on a pc or will i need some sort of pbx telephony equipment? |
| 18:24.56 | leifmadsen | SIP connection will be fine |
| 18:25.01 | leifmadsen | e.g. virtual machine would work |
| 18:25.17 | jzombie | how about the coding? what language? |
| 18:25.46 | jzombie | thanks much for the info everyone. its been difficult finding a solution that doesnt require a massive investment |
| 18:26.42 | Joel | jzombie: google voice will get you up and running easier. |
| 18:26.44 | jmetro | coding is done in dialplan |
| 18:26.54 | Joel | all except that auto dial piece. |
| 18:28.05 | jmetro | asterisk dialplan language, i dont know its official name |
| 18:30.17 | jzombie | excellent. ill start checking out the tutorials n videos |
| 18:30.24 | jmetro | ~book |
| 18:30.24 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:30.26 | jmetro | just read that |
| 18:30.35 | jmetro | skip the quick setup |
| 18:30.45 | jmetro | by the end of chapter...6? youll have a working pbx |
| 18:30.58 | jmetro | after that, google fu |
| 18:32.54 | *** join/#asterisk MarKsaitis (~MarKsaiti@2.217.210.210) |
| 18:36.55 | igcewieling | Remember, Asterisk isn't really a PBX. Asterisk is a TOOLKIT which lets you BUILD a PBX. |
| 18:38.38 | Jagster` | is that like saying linux isnt an os? |
| 18:40.29 | Chainsaw | Jagster`: Which is true. Linux is just a kernel. |
| 18:40.57 | Jagster` | igcewieling: heres a question, my reports appear to be off (ie some days users reporting more time on calls than they were on the clock) |
| 18:41.01 | Jagster` | where would u start to troubleshoot that? |
| 18:41.15 | itgrl | The queries? |
| 18:41.18 | Jagster` | i know nothing about asterisk and have spent the first 2 days reading the manual |
| 18:41.22 | Jagster` | the queries are fine, i checked them |
| 18:41.30 | itgrl | That's usually where I find my fault. |
| 18:41.32 | Jagster` | im thinking this is a case of garbage in, garbage out |
| 18:41.50 | igcewieling | Jagster`: Asterisk does not have reports, must be a GUI thing. |
| 18:42.02 | Jagster` | thinking when the calls are transferred they dont get tagged right |
| 18:42.15 | itgrl | Then possibly do to transfers. User A transfers to USer B. User B gets full credit for call. Which = query wrong. |
| 18:42.22 | itgrl | that's what I've found in many cases |
| 18:42.48 | Jagster` | is there good reading on query best practices somewhere? |
| 18:43.20 | itgrl | I've also had to insert specific CDR tags to help build better queries. |
| 18:43.35 | igcewieling | Jagster`: not really, since it is beyond the scope of Asterisk, and each company has differnet needs. |
| 18:43.48 | Jagster` | yeah hopefully it doesnt mean i have to rebuild this whole system, until last week i had onlyheard of asterisk, neveer touched it :o |
| 18:44.25 | itgrl | I like building phone systems. :) |
| 18:44.35 | Jagster` | i like turtles |
| 18:44.36 | Joel | Jagster`: you're using trixbox, which was based off of freepbx, and I can tell you right now good luck in getting accurate cdr's out of it. |
| 18:44.37 | igcewieling | FreePBX puts the CDRs in asteriskcdrdb database and cdr table. don't know where Trixbox puts it |
| 18:44.51 | Jagster` | Joel: its buggy i presume? |
| 18:44.59 | Joel | Jagster`: ... |
| 18:45.06 | igcewieling | Jagster`: apparently you like impossible tasks. |
| 18:45.11 | Joel | ^ |
| 18:45.20 | Jagster` | igcewieling: this was the task handed to me |
| 18:45.29 | Jagster` | gotta do due diligence before i can say this is unpossible |
| 18:45.40 | igcewieling | and drag us along with you? |
| 18:45.41 | Joel | Jagster`: this isn't an easy pop on irc and have someone tell me how to fix it. |
| 18:46.00 | Jagster` | igcewieling: not at all, i was just asking to see if someone had tried this before and point me in the right direction |
| 18:46.06 | Jagster` | hand holding is the last thing i'd request from anyone |
| 18:46.13 | Joel | you'll need to think like a computer person, start by reproducing the call that causes problems, then start pawing through all the bits and pieces until you find where things go wrong. |
| 18:46.22 | igcewieling | This is more of spending weeks and weeks learning that most of your problems are related to the GUI and how the GUI configures Asterisk, and how old the GUI is. |
| 18:46.40 | Jagster` | sounds about right |
| 18:46.54 | itgrl | I agree, I've found most of my problems is the GUI and the way the queries are structured. |
| 18:47.27 | itgrl | I have been undertaking the task of cleaning up and rebuilding a 1000 seat call center and it is a rabit hole of bad code. |
| 18:48.24 | Jagster` | heh i supported a software with over a million lines of java |
| 18:48.25 | Jagster` | <PROTECTED> |
| 18:49.01 | itgrl | eek |
| 18:49.07 | igcewieling | Jagster`: now imagine doing that when "you first came across Java last week" and you'll get an idea of your task ahead |
| 18:49.29 | Jagster` | thats exactly what i did :< learned java through stack traces lol |
| 18:49.41 | Jagster` | itgrl: good idea though i hadnt considered rewriting the queries to optimize them |
| 18:49.44 | igcewieling | Jagster`: also read the first edition of the Asteisk book, not the other editions since they are for versions of asterisk far far newer than the one you have |
| 18:50.00 | Jagster` | igcewieling: good point |
| 18:50.27 | igcewieling | IIRC Trixbox is in violation of the GPL as well. |
| 18:51.03 | Jagster` | so what you're saying is i landed on a giant pile of poo |
| 18:51.28 | *** join/#asterisk mchou_ (~quassel@unaffiliated/mchou) |
| 18:51.37 | jmetro | 1000 seat call center should be the same amount of code as a 10 seat call center |
| 18:51.41 | igcewieling | ask on #FreePBX for more information, as they are the ones Trixbox stole from |
| 18:51.41 | jmetro | dynamic code is ftw |
| 18:52.35 | igcewieling | jmetro: exactly. We use a combo of extensions.conf, AEL, AGI, and database for everything. |
| 18:52.56 | igcewieling | with a nice nifty GUI to manage what needs to be managed. |
| 18:52.57 | jmetro | i use asterisk dialplan, PHP, and database |
| 18:55.22 | jzombie | do i need extensive knowledge of linux to create an autodialing system with Asterisk? |
| 18:55.36 | *** join/#asterisk cedr (cedr@unaffiliated/cedr) |
| 18:55.57 | igcewieling | jzombie: you need a good knowledge of linux in order to manage Asterisk regardless of what you want to do with Asterisk |
| 18:56.20 | igcewieling | you also need a good knowledge of telephony, networking, SIP and RTP. |
| 18:56.43 | jzombie | thx |
| 18:56.47 | jmetro | not the greatest knowledge, as long as all your stuff works |
| 19:02.42 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 19:12.15 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.smb.curriegrad2004.ca) |
| 19:12.44 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.smb.curriegrad2004.ca) |
| 19:20.30 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 19:23.36 | Joel | igcewieling: trixbox didn't steel freepbx, nor did it violate the gpl, just a fyi. |
| 19:23.45 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.smb.curriegrad2004.ca) |
| 19:24.02 | igcewieling | So they release the source code? |
| 19:24.07 | Joel | igcewieling: yes. |
| 19:24.54 | *** join/#asterisk BrokenArrow (~BrokenArr@unaffiliated/brokenarrow) |
| 19:25.38 | Joel | one would think you would know the answer to that before making such a claim |
| 19:27.05 | Joel | you also might note that trixbox ce has been eol'd for a bit now |
| 19:27.45 | Joel | but it's much more fun to light torches on fire and grab pitch forks :) |
| 19:28.28 | igcewieling | the EOL was noted multiple times |
| 19:28.37 | jmetro | like 2 hours ago <.< |
| 19:28.48 | igcewieling | I must have been thinking of Fonality |
| 19:31.06 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 19:31.23 | eXcAliBuR | i'm trying to get my incomming calls to ring my sip extension 300 |
| 19:31.26 | eXcAliBuR | and i'm having trouble |
| 19:31.34 | eXcAliBuR | it goes stright to voicemail |
| 19:31.36 | jmetro | show the configs |
| 19:31.44 | jmetro | show the cli of the call |
| 19:31.59 | eXcAliBuR | do i use the answer or dial command? |
| 19:32.12 | *** join/#asterisk edwin_quijada (~macaruchi@190.167.161.58) |
| 19:32.14 | edwin_quijada | Hi! |
| 19:32.39 | edwin_quijada | There is any problem to do a dialplan just using perl or pho as AGI |
| 19:32.56 | eXcAliBuR | the cli doesn't show anything |
| 19:33.04 | eXcAliBuR | hmmm |
| 19:33.09 | edwin_quijada | I have a complex dialplan and I wanna do it with AGI |
| 19:33.11 | jmetro | excalibur core set verbose 1000 |
| 19:33.14 | jmetro | core set debug 1000 |
| 19:33.16 | jmetro | make the call |
| 19:33.18 | jmetro | pastebin it |
| 19:33.21 | itgrl | edwin_quijada> I've been working with Adhearsion which is a Ruby framework. |
| 19:33.47 | igcewieling | "sip set debug on" might be ore useful |
| 19:33.50 | edwin_quijada | itgrl: Ok, explain me about that , can i programming in Ruby |
| 19:34.12 | itgrl | http://adhearsion.com |
| 19:34.27 | eXcAliBuR | ill pastebin it |
| 19:34.47 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 19:34.47 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 19:34.53 | pabelanger | jmetro: really, 1000? |
| 19:35.19 | jmetro | *shrug* why not |
| 19:36.01 | edwin_quijada | but it is like phpagi or perlagi? |
| 19:36.12 | navaismo | verbose 5 its enough...even 3 |
| 19:36.13 | eXcAliBuR | http://pastebin.com/S0JkZzGw |
| 19:36.21 | eXcAliBuR | i put in a fake number so no one calls me |
| 19:36.22 | eXcAliBuR | :) |
| 19:36.42 | eXcAliBuR | although it does show in the cli |
| 19:36.43 | eXcAliBuR | :/ |
| 19:37.17 | edwin_quijada | it is the same as php or perl but for Ruby |
| 19:37.31 | itgrl | <edwin_quijada> It is fastagi, yes. You can do similar with perl or PHP |
| 19:37.44 | itgrl | There are PHP and Perl frameworks as well that you could use. |
| 19:37.46 | jmetro | edwin_quijada: it looks like you answer the incoming call, and then pipe it into voicemail. Ringing a phone uses the dial() application |
| 19:38.04 | igcewieling | eXcAliBuR: it is doing exactly what you programmed it to do. |
| 19:38.09 | eXcAliBuR | which is wrong |
| 19:38.11 | navaismo | eXcAliBuR, wrong dialplan |
| 19:38.15 | igcewieling | then change it. |
| 19:38.21 | eXcAliBuR | i don't know what i did wrong |
| 19:38.26 | jmetro | eXcAliBuR: it looks like you answer the incoming call, and then pipe it into voicemail. Ringing a phone uses the dial() application |
| 19:38.32 | navaismo | answer(300,15,rt) |
| 19:38.49 | edwin_quijada | My question is advantages and disvantage about use AGI to do a dialplan |
| 19:38.52 | igcewieling | eXcAliBuR: perhaps you should read the Asterisk book, which covers such basic stuff. |
| 19:39.00 | eXcAliBuR | i have it open beside me |
| 19:39.14 | navaismo | then you need glasses |
| 19:39.19 | eXcAliBuR | i do |
| 19:39.20 | eXcAliBuR | :[ |
| 19:39.23 | eXcAliBuR | i can't afford glasses |
| 19:39.51 | navaismo | I know that stuff is expensive :'( |
| 19:40.02 | navaismo | havent change mine in two years |
| 19:40.09 | igcewieling | "A Basic Dialplan to Test Your Devices" |
| 19:40.16 | eXcAliBuR | ok, so I change answer to Dial, and move my exten for 300 into that context |
| 19:40.19 | eXcAliBuR | lets see |
| 19:40.28 | edwin_quijada | igcewieling: it is a book? |
| 19:40.34 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 19:40.57 | igcewieling | I believe the n00b level has exceeded my maximum tolerances. |
| 19:40.59 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 19:41.03 | itgrl | ~book |
| 19:41.03 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:41.03 | navaismo | indeed |
| 19:41.24 | jmetro | ^ the asteriskdocs link is a free online one |
| 19:42.00 | navaismo | wondering if igcewieling throw his desktop |
| 19:42.37 | eXcAliBuR | errr |
| 19:42.45 | eXcAliBuR | I forgot the bloody Sip |
| 19:42.48 | eXcAliBuR | before 300 |
| 19:43.40 | navaismo | and use the correct application |
| 19:44.57 | eXcAliBuR | i did it all wrong |
| 19:44.58 | eXcAliBuR | >:( |
| 19:45.02 | *** join/#asterisk MarKsaitis (~MarKsaiti@2.217.210.210) |
| 19:46.24 | jmetro | Its all good excalibur. Just remember to read that chapter 6 when you need to.. or google fu sometimes. |
| 19:59.18 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 20:01.05 | *** join/#asterisk tangledhelix (~dan@99-59-105-172.lightspeed.bcvloh.sbcglobal.net) |
| 20:19.42 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 20:21.35 | eXcAliBuR | ok so it says in the CLI called Sip/300 |
| 20:21.41 | eXcAliBuR | then nobody picked up in time... |
| 20:21.44 | eXcAliBuR | but my phone don't ring |
| 20:22.05 | itgrl | Is your phone registered with Asterisk? |
| 20:22.14 | eXcAliBuR | let me make sure |
| 20:22.41 | eXcAliBuR | yes |
| 20:22.47 | eXcAliBuR | but |
| 20:22.52 | eXcAliBuR | my host is external |
| 20:22.59 | eXcAliBuR | and might have firewall problems |
| 20:23.03 | eXcAliBuR | hmmm |
| 20:23.34 | eXcAliBuR | anything coming in on port5060 won't know where to go |
| 20:23.42 | eXcAliBuR | >:( |
| 20:24.05 | itgrl | ahh, yeah nat and firewalls can be a problem sometime |
| 20:24.43 | eXcAliBuR | i'm so excited |
| 20:24.44 | eXcAliBuR | :] |
| 20:24.47 | eXcAliBuR | i bought a slice |
| 20:25.00 | eXcAliBuR | to host my asterisk |
| 20:39.23 | jmetro | a slice of raspberry pi? |
| 20:42.56 | drmessano | The Raspberry Pi will impress me when it supports FuFme |
| 20:43.05 | drmessano | Then we can call it complete |
| 20:45.23 | navaismo | FUFME? |
| 20:49.58 | eXcAliBuR | THANK YOU ASTERISK |
| 20:50.03 | eXcAliBuR | just wanted to get that out there |
| 20:50.04 | eXcAliBuR | :] |
| 20:51.23 | eXcAliBuR | now how to tie caller id into an out going call |
| 20:51.24 | eXcAliBuR | :) |
| 20:51.35 | eXcAliBuR | maybe i should look in the book first |
| 20:53.05 | *** join/#asterisk Spengler1 (~Spengler1@pool-98-117-215-165.bltmmd.fios.verizon.net) |
| 20:54.05 | itgrl[A] | is now away - Reason : Auto-Away after 30 minutes |
| 20:54.32 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 21:01.56 | eXcAliBuR | i tried what i found on the interweb, and i get errror no application 'SetCallerID' |
| 21:02.40 | navaismo | deprecated |
| 21:02.51 | WIMPy | That was a long time ago. |
| 21:02.53 | navaismo | SET(CALLERID(num/name/all)) |
| 21:02.57 | WIMPy | Set CALLERID() |
| 21:03.14 | navaismo | s/\//|/ |
| 21:03.37 | eXcAliBuR | omg |
| 21:03.50 | eXcAliBuR | all the /\/ hurts my head |
| 21:04.38 | eXcAliBuR | whats the all for? |
| 21:04.43 | WIMPy | /.-./.-./.-./.com |
| 21:05.07 | WIMPy | num/name |
| 21:05.28 | navaismo | supossed to replace some string but.. or I did wrong or the bot is dead |
| 21:06.17 | WIMPy | It doesn't understand escapes. Just plain 3/ |
| 21:06.56 | navaismo | hmm |
| 21:09.12 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 21:11.52 | eXcAliBuR | i need to see an example please |
| 21:12.18 | eXcAliBuR | SET(CALLERID(name)=Xcal) |
| 21:12.21 | eXcAliBuR | so far I got that |
| 21:13.10 | navaismo | and... |
| 21:13.13 | navaismo | did you tried |
| 21:13.39 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 21:14.46 | eXcAliBuR | well i dont have a number in yet |
| 21:14.57 | eXcAliBuR | can i have 2 sets on the same line? |
| 21:14.58 | navaismo | not need a number for try that |
| 21:15.22 | navaismo | you can try it with normal devices |
| 21:15.38 | navaismo | eXcAliBuR, ~book |
| 21:15.45 | navaismo | ~book |
| 21:15.45 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:16.05 | eXcAliBuR | I HAVE THE BOOK |
| 21:16.15 | eXcAliBuR | i looked in the index |
| 21:16.17 | navaismo | THE READ IT! |
| 21:16.19 | eXcAliBuR | and it doesn't help for callerid |
| 21:16.21 | navaismo | THEN* |
| 21:16.49 | eXcAliBuR | it shows how to set caller id on incomming calls |
| 21:16.52 | eXcAliBuR | but not out going |
| 21:16.56 | navaismo | if the bible hasn't the answer then ask the LORD a.k.a Google |
| 21:17.08 | navaismo | ¬¬ same thing |
| 21:17.28 | WIMPy | It is always outgoing. |
| 21:20.03 | *** join/#asterisk MarKsaitis (~MarKsaiti@2.217.210.210) |
| 21:24.13 | eXcAliBuR | i got it working |
| 21:24.14 | eXcAliBuR | i'm so happy |
| 21:24.18 | eXcAliBuR | i found this website: http://www.voip-info.org/wiki/view/Setting+Callerid |
| 21:24.20 | eXcAliBuR | >:) |
| 21:24.23 | eXcAliBuR | YAY! |
| 21:24.26 | eXcAliBuR | giggles |
| 21:24.29 | *** join/#asterisk matrix1233 (~matrix123@41.228.40.10) |
| 21:24.32 | jmetro | How can i make a peer that is not an endpoint? |
| 21:24.35 | eXcAliBuR | something so simple makes me so happy |
| 21:25.52 | WIMPy | Better try |
| 21:26.00 | WIMPy | https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information |
| 21:28.20 | eXcAliBuR | is it hard to setup a queue ? |
| 21:28.36 | eXcAliBuR | simple yes or no will do |
| 21:29.18 | eXcAliBuR | oh god, it has a whole chapter |
| 21:29.40 | eXcAliBuR | 40 pages |
| 21:29.52 | jmetro | if you have a queue with a lot of needs, make your own and dont bother |
| 21:30.08 | jmetro | otherwise you can use the queue() app |
| 21:31.19 | *** join/#asterisk heffer (felix@fedora/heffer) |
| 21:31.47 | matrix1233 | hello why a2billig must work on realtime asterisk |
| 21:32.53 | *** join/#asterisk drmessano^ (~nonya@pdpc/supporter/active/drmessano) |
| 21:33.49 | *** join/#asterisk mwally (~mwally@mwally.opencnam.com) |
| 21:34.25 | malcolmd | because that's how the a2billing people like to do things? |
| 21:37.03 | *** join/#asterisk willryder (~tedryder@nc-184-3-98-65.dhcp.embarqhsd.net) |
| 21:38.18 | matrix1233 | here an example of installation of freepbx (so its means not a realtime asterisk) and a2billing |
| 21:38.21 | matrix1233 | http://sysadminman.net/blog/2009/integrating-freepbx-with-a2billing-621 |
| 21:39.53 | matrix1233 | i try to chose the best solution am configuring a new asterisk with a2billig where i will send 1000 simultanous call |
| 21:40.11 | *** join/#asterisk slidesinger (~slidesing@c-69-141-208-250.hsd1.nj.comcast.net) |
| 21:40.54 | malcolmd | if your goal is to provide a2billing as a solution, then the best people to talk to about using a2billing and getting the most out of it are the a2billing people. people here are good at asterisk questions, but the level of a2billing knowledge here is going to be much less than at asterisk2billing.org |
| 21:42.04 | *** join/#asterisk clopez_ (~tau@neutrino.es) |
| 21:43.05 | matrix1233 | ok an asterisk question .. we can install asterisk realtime with freepbx |
| 21:43.07 | matrix1233 | ? |
| 21:43.11 | *** join/#asterisk bmg505 (~leon@196-209-7-250.dynamic.isadsl.co.za) |
| 21:43.32 | WIMPy | That's not an Asterisk question, either. That's for #freepbx. |
| 21:44.14 | navaismo | and I guess the answer is: no freepbx has different tables |
| 21:45.01 | *** join/#asterisk k611 (~K610@cable-78.29.241.186.coditel.net) |
| 21:45.29 | jmetro | or: no, its like trying to upgrade a mac |
| 21:46.35 | matrix1233 | it's exist a good interface for asterisk realtime |
| 21:46.46 | matrix1233 | ? |
| 21:47.11 | navaismo | i dont know one, |
| 21:47.27 | navaismo | only couple of self build GUIs |
| 21:48.38 | navaismo | I use one with realtime static its based on charisma template but the dialplan still using plain file |
| 21:48.51 | navaismo | i can pm the url if you want |
| 21:49.28 | matrix1233 | ok thx |
| 21:49.42 | matrix1233 | send me plz |
| 21:52.16 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 21:52.24 | igcewieling | Are they gone? |
| 21:52.37 | drmessano | Isn't the whole point of realtime for dymanic DB storage which is often interfaced with another system? |
| 21:53.01 | drmessano | I dont see the point of an Asterisk specific GUI AND realtime |
| 21:53.07 | igcewieling | I found Realtime to be disapointing in ways |
| 21:55.03 | navaismo | ah you back igcewieling im concerned about your desk and pc XD |
| 21:55.30 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 21:55.30 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 21:55.44 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:57.31 | igcewieling | navaismo: as long as you are not concerned about Trixbox, FreePBX, PIAF, A2Billing, or setting up asterisk without learning anything, I think we are good. |
| 21:59.12 | *** join/#asterisk matrix1233 (~matrix123@41.228.40.133) |
| 21:59.30 | jmetro | igcewieling: I'm trying to setup A2Billing on my Trixbox distro of FreePBX, but my phones arent registering. |
| 21:59.50 | igcewieling | nukes jmetro from orbit. |
| 21:59.55 | matrix1233 | sorry i was disconnected |
| 21:59.58 | eureka^ | LOIC incoming |
| 22:00.14 | matrix1233 | so a2billing say that asterisk realtime is necessery |
| 22:00.29 | igcewieling | oh god |
| 22:00.32 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 22:00.38 | navaismo | XD |
| 22:00.39 | matrix1233 | and freepbx don't work with asterisk realtime and i wanna use it |
| 22:01.12 | [TK]D-Fender | jmetro: You mean the one you had running on AsteriskWin32 in a VM and using Cisco phones with SCCP behinda remote NAT? |
| 22:01.39 | WIMPy | ouch |
| 22:02.11 | [TK]D-Fender | That's * 1.2 of course... |
| 22:02.53 | eureka^ | well, I just puked in my mouth a bit |
| 22:04.30 | drmessano | Since when do you need Realtime for A2B?> |
| 22:04.40 | drmessano | It has documented FreePBX integration |
| 22:04.43 | drmessano | Derp |
| 22:05.25 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 22:08.45 | navaismo | Asterisk 11.3 locks when I try to: restart dahdi or unload the module or load the module |
| 22:08.54 | navaismo | here is the core show locks output http://pastebin.com/XqzDsFrE |
| 22:08.57 | navaismo | any hints? |
| 22:09.41 | drmessano | pthread. Blah |
| 22:09.58 | WIMPy | Maybe you can disable the restarts. |
| 22:10.56 | navaismo | drmessano, do i neded to load dahdi intead pthread? |
| 22:11.07 | navaismo | WIMPy, dont follow |
| 22:11.27 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 22:11.36 | WIMPy | If you have dahdi you don't need other timers. |
| 22:14.08 | mjordan | navaismo: that actually doesn't show a deadlock. Just that two locks are held by threads. That doesn't mean that things aren't locked up - but you don't have a circular waiting condition. It also doesn't mean that it's being caused by res_timing_pthread - pthread_mutex_lock is something completely different, and nothing in that core show locks output points to the timers being involved |
| 22:14.11 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:14.12 | mjordan | there is an open issue for this |
| 22:14.31 | mjordan | but it - so far - only happens when a reload is issued through the FreePBX module, so we aren't sure (yet) what's going on |
| 22:15.19 | navaismo | hmmm, im not using freepbx, vanilla asterisk |
| 22:16.01 | navaismo | but i see some strabnge behavior in the dahdi/tdm card side, if i disconnect the cable they dont report alarm until I restart dahdi |
| 22:16.13 | navaismo | so not sure if is the card or dahdi |
| 22:17.09 | mjordan | hm. |
| 22:17.33 | mjordan | in addition to getting a core show locks output, a gdb backtrace would help |
| 22:17.46 | navaismo | ok |
| 22:18.16 | mjordan | when it isn't a deadlock but things aren't responding, it could be that something has just gone out to lunch. The gdb backtrace helps with those kinds of issues |
| 22:19.10 | mjordan | ASTERISK-21099 was the issue I was thinking of. If it looks like it's the same symptoms as what you're seeing, you might be able to help out on debugging the issue there |
| 22:19.11 | LieutPants | [ASTERISK-21099] [Status: Waiting for Feedback] [Assigned: jasb] Reload makes dahdi not work - https://issues.asterisk.org/jira/browse/ASTERISK-21099 |
| 22:20.19 | navaismo | looking at it |
| 22:27.36 | navaismo | trying to get the gdb output |
| 22:27.53 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-kirasjyeerslejvp) |
| 22:29.39 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-lhxdxihefturxdxn) |
| 22:32.01 | navaismo | argh asterisk is not crashing just became unresponsive no "core" file is created |
| 22:34.45 | wdoekes | navaismo: you can kill it with kill -6 and generate a core |
| 22:34.54 | wdoekes | or attach to the running process and get the output from there |
| 22:34.57 | *** join/#asterisk vlad2 (vlad@91.223.89.101) |
| 22:35.13 | wdoekes | gdb -p `pidof asterisk` |
| 22:36.24 | navaismo | ok let me try that |
| 22:36.51 | vlad2 | hi my calls from my asterisk are dropping after 20 seconds exactly every single time to more than 2 different termination poitns my provider has given me,ive looked up on google and tried 3 different solutions including pedantic=no im using asterisk 11.3.0 and my provider is using cisco ios |
| 22:40.42 | [TK]D-Fender | vlad2: And more than any other loose description you have to show us THE CALLS |
| 22:41.55 | vlad2 | ? |
| 22:50.41 | [TK]D-Fender | vlad2: SIP DEBUG. |
| 22:50.53 | [TK]D-Fender | vlad2: You know... things like evidence as to what's actually happening. |
| 22:51.04 | [TK]D-Fender | vlad2: You want an autopsy, show us the body |
| 22:57.54 | vlad2 | where shall i put the debug |
| 22:59.09 | navaismo | ok got that files anyone can help me in check it or ... should i attach that in the ticket |
| 22:59.38 | navaismo | i use tis command to get the gdb file gdb -ex "bt full" -ex "thread apply all bt" --batch -p 28057 | tee backtrace.txt |
| 23:01.41 | elguero | navaismo: I would attach it to the issue |
| 23:04.23 | navaismo | doesnt matter if the original ticket is with freepbx? I just go and say: Hey i have similar issue these are my files? |
| 23:05.16 | elguero | navaismo: It would be helfpul to say that you are having the same issue with vanilla asterisk... based on the comments it owuld appear they thought it was just a FreePBX issue |
| 23:06.21 | [TK]D-Fender | ~pb |
| 23:06.22 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 23:06.25 | [TK]D-Fender | vlad2: ^^^ |
| 23:06.43 | navaismo | ok thanks elguero |
| 23:06.47 | vlad2 | http://pastebin.com/7BWtTTX9 |
| 23:07.21 | [TK]D-Fender | [18:15]navaismohmmm, im not using freepbx, vanilla asterisk <--- elguero |
| 23:07.31 | [TK]D-Fender | elguero: Look pretty explicit to me.... |
| 23:07.57 | elguero | [TK]D-Fender: I was referring to putting a comment in the issue tracker |
| 23:08.29 | elguero | [TK]D-Fender: I already saw that he said it was vanilla asterisk, hence why I asked him to put that in the issue tracker |
| 23:08.33 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 23:11.32 | [TK]D-Fender | vlad2: 101 0110 86 Call having the requested call identity has been cleared |
| 23:11.32 | vlad2 | [TK]D-Fender i pasted it |
| 23:11.55 | vlad2 | yes |
| 23:12.03 | [TK]D-Fender | vlad2: I see it constanting trying to ACK the 200 OK on INVITE which leads me to suspect a routing mismatch. |
| 23:12.08 | vlad2 | the connection drops 20 seconds |
| 23:12.09 | [TK]D-Fender | constantly* |
| 23:12.28 | vlad2 | which ip tries to constantly ACK it? |
| 23:12.45 | [TK]D-Fender | vlad2: Do both * & your IOS switch have direct public IP's on them with no ALG, etc between them? |
| 23:13.00 | vlad2 | both have public ip's |
| 23:13.04 | vlad2 | my * i control |
| 23:13.07 | vlad2 | the IOs my provider controls |
| 23:13.13 | vlad2 | but he has many clients with no complaints |
| 23:13.19 | vlad2 | and this is not the first IOS i have this problem with from him |
| 23:13.27 | vlad2 | the asterisk has no ALG etc |
| 23:13.50 | [TK]D-Fender | vlad2: Lines 357 407, 507 for starters... |
| 23:14.58 | vlad2 | right |
| 23:15.03 | vlad2 | what could i do to stop this from happening? |
| 23:15.22 | vlad2 | the connection is made.. the call is perfect but it just drops after 20 seconds |
| 23:15.28 | vlad2 | which is not good obviously |
| 23:17.12 | [TK]D-Fender | vlad2: It's never getting ack'd properly |
| 23:17.49 | vlad2 | the 80.50.x.x ? |
| 23:18.05 | [TK]D-Fender | vlad2: You can see them.... |
| 23:19.22 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 23:19.23 | vlad2 | hmmm right |
| 23:19.30 | vlad2 | so how can i fix this issue? |
| 23:19.30 | vlad2 | even temporarily |
| 23:19.47 | vlad2 | which server is not receiving the ack properly? i really dont see it |
| 23:19.50 | vlad2 | there is many |
| 23:24.06 | [TK]D-Fender | First I can't tell if there any descrepencies in the IP's you've masked |
| 23:24.34 | vlad2 | there is no descrepencies as i did find and replace on them all |
| 23:24.36 | vlad2 | but ok |
| 23:24.47 | vlad2 | what do you suggest i should do please |
| 23:26.39 | [TK]D-Fender | Second, it's all the I'd suggest you show the unmasked debug to your IOS guy and have him explain why his side isn't accepting *'s ACK's for their 200 OK on INVITE |
| 23:27.09 | elguero | something is not right you in your setup... there is a BYE coming in from 80.50.xxx.xxx on port 62468 and not port 5060... look at line 607 |
| 23:29.16 | slav3_kitten | !@#>!@ i m going to chew out some bitches at my ISP tomorrow |
| 23:29.30 | vlad2 | elguero i see that |
| 23:30.17 | vlad2 | and [TK]D-Fender the 'masked' debug you reffer to, are you talking about the last 2 octets of each ip? |
| 23:30.20 | vlad2 | that's all that is masked |
| 23:30.26 | vlad2 | so i don't think that would make a big difference |
| 23:30.42 | vlad2 | Are there any * options for configuration to help this? |
| 23:30.58 | vlad2 | maybe it has NAT, or so, but i couldn't work it out |
| 23:31.05 | elguero | so, like [TK]D-Fender said, something is not right and * ack's are not being accepted |
| 23:37.13 | *** join/#asterisk mchou_ (~quassel@unaffiliated/mchou) |
| 23:38.15 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 23:38.16 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 23:41.01 | vlad2 | elguero yes ok |
| 23:41.07 | vlad2 | so something is not right wit hthe provider? |
| 23:41.08 | vlad2 | or with my isde |
| 23:41.14 | vlad2 | side** |
| 23:43.37 | vlad2 | how about instead of sending ACK to the IP which does not reply |
| 23:43.44 | vlad2 | i could send ACK to the orignial IP i invited |
| 23:43.46 | vlad2 | would that work |
| 23:57.31 | *** join/#asterisk mmlj4 (1000@ip68-11-55-215.no.no.cox.net) |
| 23:59.28 | mmlj4 | I have 2 servers, one running 1.8.20, and one running 11.3.0... with an identical sip.conf, I can place calls via voicepulse on the older server, but the one running 11.3.0 gives me a 513 message too big error.... thoughts? |
| 23:59.42 | vlad2 | mmlj4 too many codecs on 11.3.0 |
| 23:59.45 | vlad2 | i just installed it so i know |
| 23:59.47 | vlad2 | disable some of them |
| 23:59.49 | vlad2 | your sip message is too big |
| 23:59.56 | vlad2 | sip.conf disallow=somecodecs |