| 00:15.03 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 00:16.37 | dan_j | Hi, Anyone seen this before? |
| 00:16.44 | dan_j | <PROTECTED> |
| 00:22.17 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
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| 00:52.32 | *** join/#asterisk ovoshlook (55158cfd@gateway/web/freenode/ip.85.21.140.253) |
| 00:53.33 | ovoshlook | Hello .Cann ot add variable to ARI application. Asterisk 13.1.0 |
| 00:53.43 | JaniceKitten | wat |
| 00:53.57 | ovoshlook | One moment |
| 00:54.08 | ovoshlook | Take some log |
| 00:55.50 | ovoshlook | http://pastebin.com/index/8mfTMGzp |
| 00:57.46 | ovoshlook | you can see that channel is created, but when I add variable to it ARI get error status 409 |
| 00:57.56 | ovoshlook | how itcan be? |
| 00:58.13 | ovoshlook | If channel already at the stasis app? |
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| 03:24.28 | dijib | Penguin: still sleeping? |
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| 03:46.55 | JaniceKitten | I need someone to crank call |
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| 04:16.13 | dralezero | big day tomorrow switching isp and going from PRI to SIP. had some experience testing out trunks between two asterisk and google voice |
| 04:17.00 | dralezero | looks like i just got to define my trunk in sip.conf, and reference it in my extensions.conf, got to replace all my literal uses of Dial(DAHDI/G2.....) |
| 04:17.51 | dralezero | also using extensions.ael to match exten pattern to determine the public caller id on outgoing. also literally referencing DAHDI =] |
| 04:18.18 | dralezero | probably should make a global if im going to find/replace |
| 04:19.14 | dralezero | windstream is giving us a edgewater to hand off the sip on private network, their device does the authenticating |
| 04:24.45 | phix | any ideas? |
| 04:25.17 | phix | JaniceKitten: 911 |
| 04:25.51 | phix | Although I don't recommend it |
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| 05:09.44 | ChannelZ | I have many ideas |
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| 06:41.48 | xochilpili | hi all |
| 06:42.15 | ChannelZ | ahoyhoy |
| 06:44.20 | xochilpili | i have a SPA-3102, i have a server IBM with 2 eth cards, i set up the eth0 to my internet, and in SPA i put wan in another segment: 172.16.0.1, lan: 192.168.0.1 (for configurations) with no dhcp, in eth1 of ibm server, static ipaddr: 172.16.0.2, trying to ping -I eth1 172.16.0.1 and response Unreachable |
| 06:44.32 | xochilpili | what am i doing wrong? |
| 06:45.53 | xochilpili | the server IBM is installed with vmware esxi (IBM), with guest centos 6.5 (updated) |
| 06:47.01 | xochilpili | from wan port of SPA-3102 connect to eth1 of that ibm server, and lan port of SPA-3102 to a laptop with static ip just for configure the SPA |
| 06:47.06 | ChannelZ | I don't understand the 192.168.0.1 vs 172.16.0.1 |
| 06:47.53 | ChannelZ | 2 ethernet cards on the server; One (eth0) is your WAN with some external IP; The other (eth1) is presumably your LAN. What IP does it have? |
| 06:48.31 | xochilpili | eth1 172.16.0.2 (connected directly to SPA-3102) |
| 06:49.25 | *** join/#asterisk zblk (~andrey@ip-91-219-66-102.opnet.spb.ru) |
| 06:49.36 | [TK]D-Fender | First the WAN port on the SPA is meant to be internet-facing |
| 06:49.46 | [TK]D-Fender | Don't go assuming it will respond to pings at all |
| 06:50.01 | [TK]D-Fender | That isn't a necessarily meaningful test |
| 06:50.20 | [TK]D-Fender | In the configuration you can open up the WAN IP for web configuration |
| 06:50.22 | [TK]D-Fender | Go prove that |
| 06:50.28 | [TK]D-Fender | (responds) |
| 06:50.29 | *** join/#asterisk jhlavacek (~jirka@84.19.95.180) |
| 06:50.34 | ChannelZ | Oh I see, he was talking about the 'WAN' and 'LAN' sides of the 3102 its self |
| 06:50.37 | [TK]D-Fender | and check if you can actually talk SIP to it |
| 06:50.42 | [TK]D-Fender | Yes |
| 06:50.49 | xochilpili | yes |
| 06:51.36 | xochilpili | doing what? |
| 06:52.26 | xochilpili | i have no configuration for SIP already, im trying to be sure that i have contact to 3102 from server |
| 06:53.00 | xochilpili | in the server i have no gui so i cant open web configuration |
| 06:53.29 | [TK]D-Fender | You don't need a gui |
| 06:53.35 | [TK]D-Fender | links works fine from CLI |
| 06:53.42 | [TK]D-Fender | or you can PORT SCAN just to see if it's there |
| 06:53.53 | [TK]D-Fender | Or you could tunnel it out and see it from your worstation |
| 06:54.06 | [TK]D-Fender | Or any of a few dozen otehr different ways |
| 06:56.11 | xochilpili | wait!, now i have some internet issues, i have no contact to internet when eth0 and eth1 are up |
| 06:58.50 | phix | Sounds like a good for #linux |
| 06:59.02 | phix | s/good/job/ |
| 06:59.18 | phix | I have no idea how autocomplete fucked that up |
| 06:59.47 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 07:04.24 | xochilpili | [TK]D-Fender, lynx cant access to 172.16.0.1 (3102) |
| 07:04.54 | *** join/#asterisk mokmeister (~quassel@95.45.40.109) |
| 07:06.38 | [TK]D-Fender | xochilpili, I told you you had to ENABLE this from the LAN side first.... |
| 07:08.06 | xochilpili | i had it enabled... Enable WAN Web server=yes |
| 07:09.11 | xochilpili | if i connect the same wire to another computer(lap) over WAN port and setup static ip on laptop i can see 172.16.0.1 (configuration) i did this before asking... |
| 07:11.13 | [TK]D-Fender | telnet to that port. |
| 07:11.18 | [TK]D-Fender | Go scan it... |
| 07:13.27 | xochilpili | telnet: connect to address 172.16.0.1: No route to host -> port 80 |
| 07:15.16 | xochilpili | please answer this, why when i connect the laptop to a wan port of 3102 i can see it? |
| 07:15.22 | *** join/#asterisk pouledodue (~textual@modemcable082.140-131-66.mc.videotron.ca) |
| 07:15.39 | xochilpili | ...and not over eth1 (centos) |
| 07:16.18 | [TK]D-Fender | I wonder about that route & your interface setup at this point... |
| 07:16.23 | [TK]D-Fender | but it's late. |
| 07:16.25 | [TK]D-Fender | Keep at it |
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| 07:49.50 | xochilpili | someone? |
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| 07:59.22 | bsv | xochilpili: this is #openbsd |
| 07:59.35 | bsv | you'r asking questions about linux as I see it. |
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| 08:07.07 | MaliutaLap | bsv: this is #asterisk |
| 08:08.52 | eirirs | hmm nothing in #* |
| 08:13.41 | xochilpili | no answer for those channels :( |
| 08:13.59 | xochilpili | i have issues configuring both eth different segments |
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| 08:20.18 | bsv | HA |
| 08:20.25 | bsv | better get some coffeeeee :) |
| 08:20.43 | bsv | y bad |
| 08:20.48 | bsv | my* |
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| 08:41.29 | skrusty | morning |
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| 09:27.39 | xochilpili | any hand here? |
| 09:27.46 | xochilpili | hopeless right now |
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| 10:38.25 | wasanzy | hellooo |
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| 10:41.27 | wasanzy | am trying to do an outbound call using call file but no success. here is the call: http://pastebin.com/jXZ9zpK6 and my call file" http://pastebin.com/iYbgY1fe |
| 10:41.49 | wasanzy | what could be wrong? |
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| 11:41.43 | wasanzy | <PROTECTED> |
| 11:44.15 | xochilpili | wasanzy, headers of the call? |
| 11:51.25 | *** join/#asterisk infina (~infina@unaffiliated/infina) |
| 11:53.51 | wasanzy | xochilpili: Hearders how? how can I get the headers? |
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| 12:07.01 | xochilpili | wasanzy, asterisk -rvvvvvv |
| 12:07.07 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 12:07.24 | wasanzy | what is the call header I sent |
| 12:07.27 | *** join/#asterisk sky8 (~sky8@93.81.242.188) |
| 12:07.31 | xochilpili | se the help in asterisk to check sip debug (on | off) |
| 12:07.50 | xochilpili | SIP headers are like HTTP headers |
| 12:09.15 | sky8 | Hello guys. Is there any good tutorial on how to collect call information through AMI? |
| 12:10.12 | xochilpili | sky8, http://www.voip-info.org/wiki/view/Asterisk+manager+API |
| 12:12.34 | sky8 | Yeah, I did read this. But there is no information on how do channels operate with each other. For example, when call is initiated there is two channels created, how do I know which channel is belong to call initiator? How can I determine call type - outbound/inbound? |
| 12:14.16 | xochilpili | sky8, there's also a doc in asterisk.org did you see that? |
| 12:14.17 | sky8 | Basically, I need to collect Cdr + record filename, but before that I need to determine on which channel to issue Monitor() action |
| 12:14.33 | sky8 | Yeah, I read it all |
| 12:17.14 | sky8 | I tried to catch Newchannel events, followed by Newstate with ChannelState 6 (Up), but it this case there is two of them, outbound and inbound channels, but I can't figure how to make them out. |
| 12:20.44 | xochilpili | sky8, im not an expert, maybe you can help me with some things :D |
| 12:21.17 | xochilpili | sky8, im trying to set up a SPA-3102 to my own asterisk with no success |
| 12:21.22 | sky8 | I'm not expert either, just forced to integrate CRM with Asterisk, so trying to figure things out :) |
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| 13:00.02 | sky8 | So, anyone has experience in parsing AMI events? |
| 13:00.51 | skrusty | a bit |
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| 13:03.09 | sky8 | You used Monitor() AMI action? I need to start recording after call been picked up, but event Newstate with ChannelState 6 (Up) is unrealiable because there is two channels for each call |
| 13:04.20 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 13:04.29 | skrusty | maybe newstate isn't the best event to look for |
| 13:04.38 | skrusty | maybe you need to look for the call being bridged |
| 13:05.19 | skrusty | the bridge event contains both channels |
| 13:06.32 | sky8 | Yeah, I thought so, but bridging changed in new versions, instead of plait Bridge event there is many BridgeCreate, BridgeEnter, etc. |
| 13:06.41 | sky8 | plain* |
| 13:07.47 | skrusty | checks |
| 13:07.53 | skrusty | what version of * you running? |
| 13:07.54 | *** join/#asterisk blee (~blee@50-88-4-82.res.bhn.net) |
| 13:08.23 | sky8 | There is several versions, 1.8 and 13. |
| 13:08.50 | sky8 | Maybe more will be added in future. Isn't there universal solution? |
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| 13:11.28 | skrusty | hmm |
| 13:12.39 | skrusty | i would say, if you want to record the call on answer, just track the BridgeEnter event |
| 13:13.12 | skrusty | that wont happen if the call isn't being bridged, and it'll be bridged if it's answered |
| 13:13.20 | skrusty | (unless there's some edge case im not aware of) |
| 13:14.09 | sky8 | Yeah, but BridgeEnter happens for both channels just like Newstate does |
| 13:14.14 | skrusty | right? |
| 13:14.18 | skrusty | what are you trying to record? |
| 13:14.24 | skrusty | the callee or the caller |
| 13:14.38 | skrusty | ('s channel) |
| 13:15.05 | sky8 | Both, into one audio file, so I issue AMI action Monitor with Mix option set to 1 |
| 13:16.00 | skrusty | you only monitor one channel, you don't need to monitor both |
| 13:16.29 | sky8 | Sure, problem is that I can't determine which one to monitor |
| 13:16.55 | sky8 | And I need to tie this record with CDR information somehoq |
| 13:16.59 | sky8 | somehow* |
| 13:17.03 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:17.09 | skrusty | ok, so you monitor the caller then |
| 13:17.37 | sky8 | And how do you determine which channel belongs to caller? |
| 13:17.43 | skrusty | model it |
| 13:17.48 | skrusty | follow events |
| 13:18.09 | skrusty | see which channels fire dial events |
| 13:18.15 | skrusty | track those channels into the bridge etc |
| 13:18.39 | sky8 | Dial event is always at start of call? |
| 13:18.43 | skrusty | ami is, in my opinion, mostly about modeling call flow |
| 13:18.59 | skrusty | if the caller dial's a callee, then yes |
| 13:19.12 | skrusty | it really depends on your setup |
| 13:19.23 | skrusty | if we're talking about queues and agents |
| 13:19.25 | skrusty | then it's different |
| 13:19.56 | sky8 | Problem is that I have to integrate with remote asterisks that belong to clients, so I guess setup can differ |
| 13:26.26 | *** join/#asterisk andremar (5f5d8424@gateway/web/freenode/ip.95.93.132.36) |
| 13:30.34 | skrusty | you need to be able to have some sort of rules you can match to see which calls you want to record |
| 13:30.49 | skrusty | AMI just gives you the info, you decide what to do with it :) |
| 13:34.07 | sky8 | rules are pretty simple, issue monitor() action when call is answered, and collect cdr event on that call. CDR is enabled on all asterisks |
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| 14:58.30 | alexises | Hi |
| 14:58.42 | alexises | I have some issue with bri lines in france |
| 14:58.54 | alexises | someone have similar conf and could help me ? |
| 14:59.05 | WIMPy | ~ask |
| 14:59.05 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:59.38 | alexises | WIMPy: yes, I know, it's common from mmultiple chan |
| 15:00.14 | alexises | I can't configure properly my BRI line, and at this stage I don't know where to search |
| 15:01.04 | WIMPy | What hardware? What drivers do you try to use? |
| 15:01.42 | skrusty | alexises: http://lmgtfy.com/?q=asterisk+bri+config+france |
| 15:01.52 | WIMPy | Or try that for some genaral info: http://voice.yeti.dk/Asterisk_vs_ISDN/1 |
| 15:02.36 | alexises | skrusty: we are blocked me and 2 others peoples on this bug since 3 ou 4 days |
| 15:02.52 | alexises | hardware : 03:08.0 Ethernet controller: Digium, Inc. Hx8 Series 8-port Base Card (PCI-Express) (rev 11) |
| 15:02.59 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 15:03.20 | alexises | kernel module wctdm24xxp |
| 15:04.00 | WIMPy | That does not look like a BRI card, does it? |
| 15:04.07 | sekil | that's not BRI at all |
| 15:04.10 | [TK]D-Fender | perhaps the driver module is shared |
| 15:04.15 | sekil | but an analog one |
| 15:04.45 | alexises | it's a bri cand if you plug the correct module in ;) |
| 15:04.53 | alexises | that the case |
| 15:04.55 | [TK]D-Fender | alexises: Go physically look at your card and confirm the exact model # |
| 15:05.07 | WIMPy | Yes. I see. That thing looks ugly. |
| 15:05.16 | alexises | it's the correct one, seel by digium |
| 15:05.20 | alexises | *seel |
| 15:05.32 | [TK]D-Fender | http://www.digium.com/en/products/telephony-cards/hybrid |
| 15:05.40 | [TK]D-Fender | looks like this is it |
| 15:05.41 | alexises | the ally interesting reference is Hx8 ^^ |
| 15:05.51 | alexises | sorry, I'm not clear |
| 15:06.10 | sekil | hah |
| 15:06.12 | sekil | interesting |
| 15:07.06 | [TK]D-Fender | alexises: your card comes with installation support direct from Digium. |
| 15:07.34 | alexises | [TK]D-Fender: already call the support |
| 15:07.36 | [TK]D-Fender | alexises: And documention at the bottom of that page |
| 15:07.59 | *** join/#asterisk PaybackTony (~Adium@2601:7:700:7fa:c90d:4451:38fd:2b0a) |
| 15:08.11 | alexises | the documentation would anly sell us a realy expensive full integreated solution based on asterisk |
| 15:08.14 | alexises | :'( |
| 15:08.22 | alexises | *only |
| 15:08.31 | WIMPy | And we still have no idea what it's all about. How farr did you get? What works? What doesn't? |
| 15:09.04 | alexises | OK, I have 4 bri lines |
| 15:09.09 | WIMPy | What's the question? |
| 15:09.16 | alexises | yesteday, only 2 works |
| 15:09.33 | alexises | today with another configuration, we can have unstable communication |
| 15:10.00 | alexises | If some people could check it for error |
| 15:10.12 | WIMPy | Is the wiring terminated crrectly? |
| 15:10.36 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN/7 |
| 15:10.46 | alexises | wiring is correct, we have an old PBX connected to theses line that work properly |
| 15:10.55 | alexises | we have double check it :'( |
| 15:11.25 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 15:11.33 | WIMPy | checks if the Digium documentation is correct. It used to be wrong in that respect. |
| 15:11.50 | [TK]D-Fender | alexises: You have asked the wrong things on support if they are trying to upsell you on it. |
| 15:12.04 | [TK]D-Fender | alexises: Make sure to ask for INSTALLATION SUPPORT. |
| 15:14.54 | alexises | http://fpaste.org/173052/14219396/ some config file |
| 15:15.02 | WIMPy | Looks like Digium corrected the information on termination. Good. |
| 15:16.36 | WIMPy | Config looks good. But we still don't know much about what issue you actually have. |
| 15:17.26 | *** join/#asterisk newtonr (newtonr@nat/digium/x-wilteuicijwhigad) |
| 15:17.27 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:19.37 | alexises | okey, I go to my colleague, It's seens that today when we pass call to the outside, sometime it's work sometime not |
| 15:20.16 | WIMPy | We need some more information than "it doesn't print - err - work". |
| 15:20.47 | alexises | asterisk is unable to mount the line, lets me search the error code on logfile |
| 15:21.39 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |
| 15:25.50 | alexises | [2015-01-22 10:51:27] ERROR[31468] chan_dahdi.c: PRI Span: 1 Unable to receive TEI from network in state 3(Establish awaiting TEI)! |
| 15:26.15 | alexises | voila, I found |
| 15:26.16 | WIMPy | Are you sure it's a ptmp line? |
| 15:26.55 | WIMPy | That's what happens if you configured ptmp, but have ptp. |
| 15:27.29 | WIMPy | And unless you want to connect another device in parallel, it's safe to always configure ptp. |
| 15:27.52 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |
| 15:28.22 | sekil | I thought ptp is in Europe mostly |
| 15:29.09 | WIMPy | It depends on what you ordered. Nothing else. |
| 15:29.16 | alexises | we are in France |
| 15:29.26 | sekil | also with ptmp TEIs are dynamically thrown from the network |
| 15:29.33 | alexises | with Numeris line |
| 15:29.36 | *** join/#asterisk mjordan (mjordan@nat/digium/x-sfraoywnvhkbmywz) |
| 15:29.36 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:29.45 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 15:29.58 | sekil | where as with ptp it's static and set to 0 usually |
| 15:30.37 | alexises | I'm not sure to understend, it's can explan why it's work randomly ? |
| 15:31.07 | WIMPy | No. |
| 15:31.14 | WIMPy | Are all lines the same? |
| 15:32.03 | *** join/#asterisk bgmCoder (~Instantbi@70.43.111.18) |
| 15:32.05 | sekil | alexises: if it's a ptp than set bri_cpe |
| 15:32.09 | bgmCoder | How can I originate a call from a label in a macro? http://stackoverflow.com/questions/28091772/originate-from-a-macro |
| 15:32.37 | alexises | yes, all the same |
| 15:33.27 | alexises | on logne I have also : |
| 15:33.30 | alexises | PRI got event: HDLC Abort (6) on D-channel of span 2 |
| 15:33.48 | WIMPy | That's bad. |
| 15:34.28 | WIMPy | Tht is usually an indication of receiving garbage, i.e. a physically bad connection. |
| 15:35.25 | alexises | ok |
| 15:35.42 | alexises | It's could explain something |
| 15:36.13 | alexises | peraps my analog pbx is more tolerent about it |
| 15:36.46 | sekil | you didn't have bri ports on the pbx? |
| 15:37.56 | alexises | I don't know the old pbx hardware |
| 15:38.42 | sekil | was it connected to S port or AB one of the NT? |
| 15:38.44 | alexises | When we make our test, we disconnect the old one from the 4 bri line and connect the new one |
| 15:39.53 | alexises | it's response to your question sekil ? |
| 15:40.12 | alexises | for more information, I have this line : |
| 15:40.15 | alexises | [2015-01-22 10:38:43] VERBOSE[31343][C-0000021f] sig_pri.c: -- Span 3: Channel 0/1 got hangup, cause 21 |
| 15:41.26 | *** join/#asterisk Akuma (~Akuma@apn-37-248-12-89.dynamic.gprs.plus.pl) |
| 15:42.31 | WIMPy | That looks like it worked correctly in that case. |
| 15:43.19 | alexises | if you have no ideas, we try in 3 hours to change pymp yo ptp and see ;) |
| 15:43.19 | WIMPy | Just that someone didn't want to accept that call. |
| 15:44.12 | WIMPy | Well, you must have obtained a TEI to get that far. |
| 15:44.31 | WIMPy | So it's still a very unclear situation. |
| 15:44.44 | WIMPy | Are you really sure all lines are configured the same? |
| 15:44.48 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:248:8219:34ff:fecf:17f0) |
| 15:45.49 | alexises | yes |
| 15:46.49 | *** join/#asterisk superscrat (asanders@nat/digium/x-yplnafqbzcxryyiu) |
| 15:50.26 | bgmCoder | [TK]D-Fender: Could you take a look at this, please? http://stackoverflow.com/questions/28091772/originate-from-a-label-within-a-macro |
| 15:50.50 | WIMPy | Marcos are deprecated. |
| 15:52.12 | alexises | we try to go to ptp mode ;) thanks |
| 15:52.15 | *** join/#asterisk jaflong (5bec7504@gateway/web/freenode/ip.91.236.117.4) |
| 15:53.14 | jaflong | why does asterisk need gtk and how do I disable it - configure error: No package 'gtk+-2.0' found |
| 15:54.28 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-vfjatwwysamgytqp) |
| 15:54.31 | *** join/#asterisk wiske57_ (~kevin@184-155-211-24.cpe.cableone.net) |
| 15:54.38 | file | it does not require it |
| 15:54.38 | pabelanger | jaflong, make menuselect |
| 15:55.57 | jaflong | soory, I meant this comes up when I do make menuselect - asterisk version 13.1.0 |
| 15:56.57 | [TK]D-Fender | WIMPy: Polo :) |
| 15:57.18 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 15:57.33 | WIMPy | oops |
| 15:58.16 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 16:02.18 | *** join/#asterisk sgriepentrog (sgriepentr@nat/digium/x-mbcuaoxvxhqzigpr) |
| 16:02.37 | [TK]D-Fender | bgmCoder: I've read it and you are using macro variables there referring to stuff we don't see.... |
| 16:03.13 | *** join/#asterisk last1 (~last1@109.103.111.89) |
| 16:03.31 | bgmCoder | [TK]D-Fender: the only thing missing is the extension to call the macro. |
| 16:03.38 | bgmCoder | I don't have the macro syntax right |
| 16:04.11 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:04.12 | [TK]D-Fender | bgmCoder: I fail to see the point of this even being a macro. You never go back and you aren't passing args |
| 16:04.23 | bgmCoder | exten => 23333,1,Macro(Paginator,angelusbell,1);Angelus Bell |
| 16:04.41 | [TK]D-Fender | bgmCoder: Oops... strike that... you seem to be passing one... which I'm sure you don't even need to |
| 16:05.19 | bgmCoder | I'm passing two arguments: arg1 is the name of the soundfile, arg2 is the dtmf tone |
| 16:05.37 | [TK]D-Fender | is the tone ever going to be DIFFERENT? |
| 16:05.49 | bgmCoder | yes |
| 16:06.39 | [TK]D-Fender | what else does that Viking do with them? |
| 16:06.40 | bgmCoder | There are three possible values; I was trying to prevent from triplicating the same macro just to have one different value |
| 16:06.59 | bgmCoder | The Viking uses that tone to know what zones to play over |
| 16:07.14 | bgmCoder | If you don't send the tone, it won't play anything over the speakers. |
| 16:07.20 | [TK]D-Fender | makes sense then |
| 16:07.24 | bgmCoder | That's all it does though |
| 16:07.35 | [TK]D-Fender | bgmCoderexten => 23333,1,Macro(Paginator,angelusbell,1);Angelus Bell <- missing code for that though |
| 16:08.07 | bgmCoder | It is in [from-internal-custom] |
| 16:08.21 | bgmCoder | it gets dialed by a user |
| 16:09.15 | [TK]D-Fender | We don't see it. |
| 16:09.18 | bgmCoder | I can't figure out how to get the dial command into the originate part of the system macro |
| 16:09.23 | [TK]D-Fender | So what we don't see ... we don't trust even exists :) |
| 16:09.35 | bgmCoder | what do you need to see that you don't? |
| 16:09.41 | [TK]D-Fender | exten => s,n,(callviking),Dial(SIP/100,10,D(${ARG2})) <--- comma between priority & label |
| 16:09.42 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:248:8219:34ff:fecf:17f0) |
| 16:10.22 | [TK]D-Fender | And not even sure why you have a label.... |
| 16:10.58 | bgmCoder | because the originate call needs to do the last two lines of the macro and return to hangup |
| 16:11.00 | *** join/#asterisk jiuweigui (~jiuweigui@unaffiliated/jiuweigui) |
| 16:12.05 | [TK]D-Fender | There is no "return" |
| 16:14.21 | [TK]D-Fender | exten => s,n,System(asterisk -rx "channel originate Local/callviking@${MACRO_CONTEXT}/n extension ${ARG1}@soundeffects") <--- this has nothing to do with touching that priority label. and is a logical error |
| 16:15.08 | [TK]D-Fender | ${MACRO_CONTEXT} <- reminder that this refers to the context that called the macro..... |
| 16:15.16 | *** join/#asterisk drale_ (~drale@unaffiliated/drale) |
| 16:16.23 | drale_ | hello. our sip provider has given a device to use that they do the authentication over the internet, then hand it off to us to use on our asterisk |
| 16:17.01 | bgmCoder | [TK]D-Fender: I showed an alternate method that works if I don't use a macro: http://stackoverflow.com/questions/28091772/originate-from-a-label-within-a-macro |
| 16:17.18 | drale_ | their device (edgewater device) is not expecting a username and password, we should just be able to send calls and receive, but our asterisk is trying to authenticate sip:anonymous |
| 16:17.37 | drale_ | how do i let calls passthrough without authentication |
| 16:17.55 | last1 | anyone tried pjsip ( pjsua ) to send out calls through Asterisk ? |
| 16:19.08 | [TK]D-Fender | last1: Since it was introduced a branch ago? Tons of people |
| 16:19.43 | last1 | well, I have pjsip installed on another server and I try to send a call through Asterisk. it doesn't work |
| 16:19.50 | [TK]D-Fender | drale_: Don't use a secret. |
| 16:20.12 | last1 | I see this error on the pjsip server: 11:13:54.566 pjsua_media.c ......pjsua_aud_channel_update() failed for call_id 0 media 0: Invalid operation (PJ_EINVALIDOP) |
| 16:20.26 | [TK]D-Fender | drale_: and make sure whatever channel initiates a dial-out using that peer doesn't have any privacy flags set |
| 16:20.27 | last1 | 11:13:54.566 pjsua_call.c .....Unable to create media session: No active media stream after negotiation (PJMEDIA_SDPNEG_ENOMEDIA) [status=220048] |
| 16:20.27 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 16:20.52 | file | pjsua is used by many things and it has been used to send calls into Asterisk |
| 16:21.01 | last1 | I have no idea what that means and Google isn't very helpful |
| 16:21.19 | jaflong | anyone know why when I do make menuselect it requires gtk? |
| 16:21.58 | bgmCoder | [TK]D-Fender: How can I use the last two lines of the macro in the originate call - I though it would work the same way as the example context I gave (which you helped me write, by the way.... :) ) |
| 16:23.40 | [TK]D-Fender | bgmCoder: Orignate has targets... jumping into the middle of a macro isn't one of them |
| 16:23.46 | [TK]D-Fender | bgmCoder: it clearly does not belong in there. |
| 16:23.47 | last1 | @file: do you have an example of how you invoke pjsua in order to make a call ? |
| 16:23.54 | [TK]D-Fender | bgmCoder: Put it somewhere else |
| 16:23.57 | bgmCoder | [TK]D-Fender: I know its wrong. |
| 16:24.15 | [TK]D-Fender | bgmCoder: it need's it's own extension. |
| 16:24.15 | Penguin | [TK]D-Fender: its |
| 16:24.23 | file | last1, nope but your problem looks pjsua audio specific |
| 16:25.38 | bgmCoder | [TK]D-Fender: What is "it"? |
| 16:26.26 | bgmCoder | If I change the callviking line in the macro and give it an extension, then can I still use arg2? |
| 16:27.10 | [TK]D-Fender | bgmCoder: Nope. Pick another way |
| 16:27.41 | [TK]D-Fender | bgmCoder: suggestion : EXTENSION |
| 16:27.54 | bgmCoder | Right. Got that. Its the only other way. |
| 16:28.47 | [TK]D-Fender | No, several other ways |
| 16:28.56 | [TK]D-Fender | This one I would consider the easiers |
| 16:29.05 | [TK]D-Fender | easiest * |
| 16:29.24 | bgmCoder | Do I have to put the callviking in its own context? If I do that then I can't use arg2 |
| 16:30.11 | [TK]D-Fender | ARG2 wouldn't exist in a new call anyway.... |
| 16:30.23 | [TK]D-Fender | you aren't thinking about where these things come from... |
| 16:30.57 | [TK]D-Fender | a NEW call that just happens to call into there (by some miracle) Still wouldn't inherit a VALUE that you passed that generated an Originate... |
| 16:31.30 | drale_ | [TK]D-Fender, thanks, i got pulled away from the chat but i did just that. the provider told me my username was our main phone number, so i just didnt use a secret register => 2223334444@192.168.0.237 , the rest of the sip.conf is just a default tutorial i found |
| 16:31.41 | drale_ | our incoming call on new number works |
| 16:32.00 | drale_ | while still communicating our old numbers on the PRI at the same time, nice |
| 16:32.08 | [TK]D-Fender | drale_: glad to hear it's working. |
| 16:33.14 | bgmCoder | [TK]D-Fender: When some miracle happens and the macro is called into, the macro is supposed to hang up on the caller and then go about with the originate |
| 16:33.30 | drale_ | the next step is outgoing, we are using extensions.ael matching a dial pattern for 91XXXXXXXXX, and also matching what your extension is, say 8101, sets callerid to display, and to Dial(DAHDI/G2 |
| 16:33.41 | bgmCoder | It doesn't need anything from the caller except the parameters |
| 16:33.50 | drale_ | would i change that to Dial(SIP/foobar) ? |
| 16:33.53 | [TK]D-Fender | bgmCoder: You have 2 lines further in the macro you'll never reach |
| 16:34.05 | bgmCoder | I know that! |
| 16:34.39 | bgmCoder | That's why we used the system call before - look at my link to see the example of what we did before. |
| 16:35.05 | bgmCoder | The progress doesn't get to those two lines, but the lines are used in the context. |
| 16:36.36 | bgmCoder | I don't need the ${Macro_Context}, I think |
| 16:40.45 | bgmCoder | It should be just ${context} |
| 16:41.13 | [TK]D-Fender | What is the point of the value there even being variable? |
| 16:41.45 | bgmCoder | Well, it was just self-referencing; I guess I could write out macro-Paginator |
| 16:42.22 | [TK]D-Fender | also won't work. You don't have multiple extensions in a macro... |
| 16:42.36 | *** join/#asterisk sgriepentrog (sgriepentr@nat/digium/x-cdkqzqiyfhxpjxbp) |
| 16:43.25 | bgmCoder | And I can't originate from a goto statement either... |
| 16:43.51 | [TK]D-Fender | bgmCoder: No, you can jump to other dialplan ... that does it |
| 16:44.12 | [TK]D-Fender | Goto ... is a gogo. Macro is a macro. Macro isn't an Originate, so why are we talking about this? |
| 16:44.31 | [TK]D-Fender | Macro & Goto are different mechanisms for simply changing where you are in the dilplan. |
| 16:51.16 | bgmCoder | [TK]D-Fender: I can't see any way to do this without creating three separate macros |
| 16:51.59 | [TK]D-Fender | You don't need any macro's at all. |
| 16:52.56 | last1 | anyone using pjsip has any idea what the following error is: http://pastebin.com/nMNwBpiC ? |
| 16:55.20 | [TK]D-Fender | http://makeameme.org/meme/stop-trying-to-0kjd1l |
| 17:00.02 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:05.05 | *** part/#asterisk spditner (~simon@108.175.226.200) |
| 17:05.43 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
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| 17:13.01 | newtonr | last1, what is the behavior you are seeing? |
| 17:13.23 | newtonr | it is hard to tell what is going on without a full pcap/trace |
| 17:14.51 | last1 | well, no call gets made |
| 17:15.01 | last1 | I see a call attempt and then a cancellation |
| 17:22.59 | file | newtonr, last1 is referring to PJSUA - not chan_pjsip |
| 17:26.11 | *** join/#asterisk Synthase_ (uid63346@gateway/web/irccloud.com/x-mluqpgyxjqksetiy) |
| 17:28.43 | last1 | I never thought sending a call through could be so complicated |
| 17:28.56 | last1 | from *nix cli |
| 17:29.01 | *** join/#asterisk rock1979 (~rock@46.252.100.15) |
| 17:29.08 | rock1979 | hello |
| 17:29.52 | newtonr | file, ah |
| 17:30.15 | rock1979 | is the following sip configuration correct "allow=ulaw,alaw,g729:10,g729:20 |
| 17:30.31 | bgmCoder | [TK]D-Fender: This is what I came up with: http://stackoverflow.com/a/28094755/1038866 |
| 17:30.52 | rock1979 | is it possible to define g729 codec with two different ptime values |
| 17:31.49 | rock1979 | any idea ? |
| 17:32.26 | *** join/#asterisk infina (~infina@unaffiliated/infina) |
| 17:32.48 | WIMPy | No idea, but G.720 with 10ms frames doesn't make sense. |
| 17:33.02 | WIMPy | Isn't G.729 even in 20ms increments? |
| 17:34.24 | rock1979 | yes but in invite message remote side is sending g729 with 10 ms ptime |
| 17:36.14 | rock1979 | no g729 increment is 10ms in asterisk |
| 17:38.29 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 17:39.28 | [TK]D-Fender | bgmCoder: Why is ${CONTEXT} even variable there? |
| 17:40.02 | bgmCoder | Isn't that the same as writing "Paginator"? |
| 17:41.34 | rock1979 | any idea ? |
| 17:41.37 | [TK]D-Fender | Don't use variables where you don't need variables. |
| 17:41.53 | [TK]D-Fender | You'll use the code elsewhere and forget that where you are matters and things will break |
| 17:41.57 | [TK]D-Fender | It's asking for trouble |
| 17:42.42 | bgmCoder | But ${context} would always be "Paginator" in this case. If I ever change the name of the context it would save me from remembering to change those.... |
| 17:42.54 | bgmCoder | That's the exact reason I did that, actually. |
| 17:42.56 | *** join/#asterisk drale_ (~drale@unaffiliated/drale) |
| 17:44.59 | *** join/#asterisk pouledodue (~textual@modemcable082.140-131-66.mc.videotron.ca) |
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| 18:04.25 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 18:05.02 | drale__ | is this syntax correct to capture the last 4 digits and send the call off to the exten matching the last 4 digits exten => _270365XXXX,1,Dial(Local/${EXTEN}@incoming) |
| 18:07.08 | *** join/#asterisk TazzNZ (~TazzNZ@office-nat1.ytechops.co.nz) |
| 18:07.30 | TazzNZ | w00t - I am back ! : |
| 18:07.33 | TazzNZ | morning all |
| 18:09.35 | TazzNZ | so I am running Asterisk 11.15.0 with snom phones and I am having issues with BLF. I have logged a ticket to snom regarding this and they are saying that the issue is caused by Asterisk sending the "terminate" state to the phone (using SIP notify). |
| 18:09.48 | TazzNZ | So I was wondering, is there any way to change that message ? |
| 18:10.05 | *** join/#asterisk CeBe (~CeBe@port-92-200-9-49.dynamic.qsc.de) |
| 18:12.29 | [TK]D-Fender | drale__is this syntax correct to capture the last 4 digits and send the call off to the exten matching the last 4 digits exten => _270365XXXX,1,Dial(Local/${EXTEN}@incoming) <- it doesn't "capture" the last 4 digits. It matches something dialed with the first 6 being those fixed digits, and any 4 extra digits. It then goes on to dial that number in its entirety into a local channel. Why... |
| 18:12.30 | [TK]D-Fender | ...you would use Dial instead of a Goto there I don't know.... |
| 18:13.53 | drale__ | [TK]D-Fender, i found it ${EXTEN:-4:4} you have to say how many digits from the end to start counting, count 4 digits and send those to Local/ |
| 18:15.11 | drale__ | I may clean this all up someday but for now its to get us back online, our old PRI use to send us the last 4 digits and thats all our extens were setup for. so i am putting this at the top since new provider sends us the full number, extract it, then dial to our preexisting exten |
| 18:15.34 | drale__ | its working =] only added one exten line at the top |
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| 18:25.45 | necronian | I'm building a voicemail system for an analog key phone system, I have an ivr system that lets people dial voicemail boxes to leave messages. However after the caller hangs up I would like to dial up the local extension and dial 2 to turn on the handset message light. |
| 18:26.53 | necronian | all I really need to do is pickup the sip/112 channel SendDTMF(<extension>2) and then hangup |
| 18:27.35 | Katty | there is an extension for hangup that you can put in the call routing |
| 18:27.56 | necronian | hangup_handler_push you mean? |
| 18:28.25 | Katty | that might work too |
| 18:28.32 | necronian | My problem is I don't understand how to call out without connecting 2 lines? |
| 18:28.53 | Katty | Local? |
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| 18:29.47 | necronian | Katty: No? I want it to pick up the line on my ata |
| 18:30.15 | Katty | necronian: something like originate Local/100@outgoing extension 123456@Outgoing |
| 18:30.42 | necronian | Katty: ah okay, I think that helps me! Thank you |
| 18:30.48 | Katty | necronian: that would connect a local extension, but you could do whatever you want instead of a local |
| 18:30.54 | Katty | necronian: like SIP or....you know |
| 18:32.16 | ChannelZ-Wk | Weird MWI system :) |
| 18:32.47 | Katty | i'll WMI your system in a minute. |
| 18:32.47 | ChannelZ-Wk | Or is it a custom device with like a giant light-up sign and a klaxon that goes off? |
| 18:32.59 | ChannelZ-Wk | That'd be fun |
| 18:33.11 | Katty | i recommend more cowbell. |
| 18:33.25 | ChannelZ-Wk | And pyrotechnics, etc |
| 18:33.30 | ChannelZ-Wk | Voicemail has never been so fun |
| 18:34.02 | Katty | this sounds like a job for IFTTT |
| 18:34.28 | bsv | ILM or WETA |
| 18:34.43 | Katty | i firmly believe that i could AT LEAST get a lamp to turn off or not |
| 18:34.49 | Katty | based on a phone call |
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| 18:45.14 | linagee | where does telcodata get data from? Is it feasible to collect it myself like they do, or do telcos go directly to them to update stuff? |
| 18:45.44 | linagee | disregard, this is their first FAQ question. :) |
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| 18:46.04 | ChannelZ-Wk | hit & run |
| 18:46.11 | Katty | those are the BEST. |
| 18:52.52 | necronian | Guys I'm sorry, can someone hold my hand and speak to me like I'm a small child? |
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| 18:54.18 | necronian | http://pastebin.com/7BDbrf7c I can drop a call file in the outgoing queue and call a desk phone. and I can call the voicemail extension to leave a voicemail |
| 18:54.45 | necronian | But I can't figure out how to combine the two. |
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| 18:56.15 | paulc | necronian: You want to dial to a phone, then drop it in voicemail? (like your 1 extension in demo-menu?) |
| 18:56.53 | skrusty | ok, quick question. how do i match (in diaplan) a number with or without a '+' at the start? e.g. +44123456789 and 44123456789 |
| 18:57.23 | WIMPy | skrusty: With two extensions. |
| 18:57.32 | necronian | paulc: yes. but then after the voicemail I want to hang up, pick up the same channel and then dial an internal extension in order to turn on the message lamp on a desk phone |
| 18:57.34 | skrusty | bah :) |
| 18:57.42 | ChannelZ-Wk | Just match one with + and Goto it to your one without stripping it off |
| 18:57.42 | skrusty | i was hoping there was a nicer way to do it |
| 18:57.46 | ChannelZ-Wk | (or vice-versa) |
| 18:58.16 | ChannelZ-Wk | Hmm. That would have made more sense with some commas. Sorry. |
| 18:59.20 | paulc | necronian: ok.. if you have a SIP phone associated with a mailbox, the message waiting indicator will take care of itself - once the message is left, it'll light up |
| 18:59.34 | ChannelZ-Wk | maybe |
| 18:59.53 | ChannelZ-Wk | (if it's configured right :) |
| 19:00.15 | necronian | paulc: the problem is that I am only using astrisk for the voicemail. the phone system is actually an analog key phone system Panasonic DBS-842 if anyone cares |
| 19:00.37 | paulc | necronian: in your call file, you have the channel.. who you want to call.. SIP/112 or SIP/2125551234@itsp (for calling externally). Then just change the context/extension/priority |
| 19:00.45 | paulc | ah.. ok - external PBX for MWI.. |
| 19:00.54 | necronian | I have a single ata Grandstream HT-502 hooked up to an analog port on the phone system |
| 19:01.00 | paulc | nods |
| 19:01.31 | paulc | ok.. there's a setting in sip.conf that yuo can run a script for message waiting.. I think what you want to do there is pass into that script the mailbox number, then generate another call file to dial out on that channel the relevant digits to activate MWI |
| 19:01.50 | ChannelZ-Wk | Make a piece of dialplan that does the SendDTMF; Then originate a channel to the SIP device and have it connect to the extension you made |
| 19:01.51 | skrusty | cheers anyway WIMPy |
| 19:01.59 | paulc | only thing I wonder about there is how you switch it off afterwards - if there's something in Asterisk voicemail to do that - it's been a while since I've had a use case like yours.. |
| 19:02.01 | skrusty | (and ChannelZ-Wk) |
| 19:02.38 | ChannelZ-Wk | skrusty, it's not that messy. Only need 1 extra line in the dialplan |
| 19:02.56 | necronian | paulc: well it switches off if you dial the extension that turned on the light, so dialing the voicemail extension to retrieve the message will turn it off |
| 19:03.19 | skrusty | yeah |
| 19:04.02 | WIMPy | ChannelZ-Wk: Depends... If you only want to match certain extensions, having a catch-all will give you other errors on your phone. |
| 19:04.30 | WIMPy | You can use a loopback switch to avoid that. |
| 19:05.03 | ChannelZ-Wk | Well presumably he has a pattern matching what he wants already, and the wildcard is the + on the front.. so it's basically the same pattern, not necessarily a catch-all a-la _X. or whatever |
| 19:05.57 | WIMPy | It is a catach-all for extensions starting with +. They will be accepted even if the Goto will fail. |
| 19:06.45 | ChannelZ-Wk | exten => 12345 and exten => +12345 |
| 19:07.09 | WIMPy | Yes. That works without side effects. |
| 19:07.20 | skrusty | :) |
| 19:10.10 | ChannelZ-Wk | 12345 does all the busywork; +12345 just gotos to 12345 via ${EXTEN:1}. Even if 12345 was a pattern (_NNNNN or whatever) I don't see how there's much difference. |
| 19:10.33 | WIMPy | There is a difference. |
| 19:10.43 | ChannelZ-Wk | If they dialed something invalid, they dialed something invalid either way. |
| 19:10.59 | WIMPy | If the extension doesn't exist, Asterisk will (by default) reject the call. |
| 19:11.30 | WIMPy | If it matches the pattern it will accept the call and the drop it later. |
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| 19:12.11 | WIMPy | then |
| 19:13.09 | WIMPy | But back to the MWI thing... |
| 19:13.57 | WIMPy | Does anyone have an idea, how to externally trigger an LED on a button on an OpenScape / Hipath thing? |
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| 19:46.10 | paulc | WIMPy: Hipath like.. Siemens? or.. no, maybe I'm thinking Hicom |
| 19:47.22 | cusco | hi |
| 19:47.29 | cusco | so allo me to re-ask |
| 19:47.34 | cusco | I'm trying to setup a custom feature applicationmap, instead of regular atxfer, I would like to press say.. *123 and perform my own atxfer to a internal queue |
| 19:47.45 | cusco | I tried: expressCovilha => *508,peer/callee,Dial(Local/508@expressTransfer) |
| 19:47.53 | cusco | but this way, the caller goes trough the queue, and the calee gets only silence. Callee call hangs when caller hangs |
| 19:48.01 | cusco | I was hopping to find a way to set 4 application maps with different dtmf sequences that allowed me atxfer without going trough the 'Transfer....' ivr |
| 19:48.22 | cusco | is there such possibility? |
| 19:48.59 | WIMPy | paulc: Unify. Siemens doesn't want to be named in that context any more. |
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| 20:01.58 | paulc | WIMPy those phones are ugggggglyyyyyy eh |
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| 20:20.53 | kemmler | What's the best way to spawn a call from php? Last way I used was .call files but that was a few years ago. |
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| 20:25.07 | [TK]D-Fender | kemmler: Call files are fine, the next comparable option is AMI |
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| 20:25.44 | kemmler | sweet, just wanted to make sure I wasn't using depreciated stuffs |
| 20:25.49 | [TK]D-Fender | kemmler: AMI's advantage is that you can issue it remotely without access to the filesystem |
| 20:26.09 | [TK]D-Fender | kemmler: But if you're local then it's just more work really. |
| 20:26.22 | kemmler | the AMI being more work? |
| 20:26.46 | [TK]D-Fender | Yes. Making a text file = easy. AMI = sockect connection with login. |
| 20:27.09 | [TK]D-Fender | less of a deal if you use a ready-made library like PHP-AGI for it |
| 20:27.15 | [TK]D-Fender | (has an AMI object) |
| 20:27.35 | kemmler | gotcha. I don't think I'll mess with the AMI unless I run into an issue with asterisk not picking up the call files or something |
| 20:27.50 | kemmler | but it's worked well in the past |
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| 20:35.34 | gravspeed | hello everybody... |
| 20:37.41 | gravspeed | so an amsterdam ip has been grinding away on one of my clients asterisk boxes running on a hosted vm, i want to restrict connections to their static ip, i see that i can do that pretty easily on the extension configuration, however they have about 30 extensions and i don't want to have to make the change to every one (and new ones in the future) |
| 20:37.58 | gravspeed | where should i put the deny and permit statements to apply to all extensions? |
| 20:38.21 | gravspeed | freepbx btw |
| 20:39.23 | WIMPy | ~freepbx |
| 20:39.23 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 20:39.23 | [TK]D-Fender | Wrong channel BTW :) |
| 20:39.58 | gravspeed | i thought of that about halfway through typing that actually, thanks for he heads up, i will go ask there. |
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| 22:26.44 | Katty | infobot: ~thebook |
| 22:26.51 | [TK]D-Fender | ~book |
| 22:26.51 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:26.52 | Katty | infobot: book? |
| 22:26.52 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:26.58 | Katty | infobot: good bot |
| 22:26.58 | infobot | thanks, Katty |
| 22:27.01 | Katty | infobot: botsnack |
| 22:27.01 | infobot | aw, gee, Katty |
| 22:27.10 | [TK]D-Fender | ~areyouadog |
| 22:27.10 | infobot | Bark! Bark! |
| 22:27.17 | [TK]D-Fender | infobot, good boy! |
| 22:27.17 | infobot | [TK]D-Fender: thanks |
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| 22:51.26 | linagee | does RBOC = landline? (Or can/do those carriers reuse the numbers for wireless?) |
| 22:51.34 | linagee | (assuming no LNP) |
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