| 00:06.12 | *** join/#asterisk ruben23 (~OpenDIAL@112.198.77.58) |
| 00:06.18 | slackie | unignore 1 |
| 00:06.32 | ruben23 | hi |
| 00:07.16 | ruben23 | guys any idea on this error when i dial-out ---> Got SIP response 603 "Declined" back from xxx.xxx.xxx - congested. |
| 00:08.56 | *** join/#asterisk NoobSaibot (~NoobSaibo@CPE-24-208-40-235.new.res.rr.com) |
| 00:14.31 | [TK]D-Fender | The other end turned you down. |
| 00:14.49 | [TK]D-Fender | As to why ... could be many reasons |
| 00:15.50 | ruben23 | [TK]D-Fender: can you site some of it..? |
| 00:16.01 | [TK]D-Fender | What's the point of guessing? |
| 00:16.08 | [TK]D-Fender | How about you show us the entire actual call... |
| 00:16.34 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 00:20.39 | ruben23 | ok |
| 00:22.27 | [TK]D-Fender | Time's up on my side... |
| 00:22.32 | [TK]D-Fender | heads out for the evening... |
| 00:23.16 | ruben23 | http://pastebin.com/Epp0N3iE |
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| 00:49.38 | TazzNZ | ruben23: my best guess is - you have exceeded the number of calls allowed |
| 00:49.50 | TazzNZ | or they themself don't know how to get to that number |
| 00:51.06 | ruben23 | TazzNZ: im just dialing oen at a time |
| 00:51.11 | ruben23 | one* |
| 00:51.28 | TazzNZ | then the second sugestion is more likely |
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| 06:17.30 | D30 | hi all, Im experiencing a weird issue with my asterisk particularly with its IVR, during a call to a local pstn number, i can see that asterisk plays the IVR but still on the other end its still ringing. |
| 06:18.27 | *** join/#asterisk kalz (~kalz@2602:fff6:f:29c9::2c33) |
| 06:18.32 | D30 | somebody with the same issue with me, can you tell how did you able to sort out your issue :) |
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| 07:03.25 | ChannelZ | What exactly is the call path? |
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| 08:37.31 | _omer | what is the right way to run asterisk? just "asterisk" command on linux shell or "service asterisk start" ? |
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| 08:48.29 | bitwize | Hello! I need to configure multiple IP-adresses on a peer in sip.conf (incoming calls comes from multiple ip's) |
| 08:48.42 | bitwize | is this possible or do I need to define multiple peers for each ip? |
| 08:59.15 | *** join/#asterisk _omer (~omer@116.71.186.191) |
| 08:59.26 | _omer | what is the right way to run asterisk? just "asterisk" command on linux shell or "service asterisk start" ? |
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| 09:07.07 | phpboy | _omer: anyway that makes you happy |
| 09:07.13 | phpboy | your own happiness is what counts |
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| 09:25.34 | alami | hello, is there any GUI tool that show me the number of participant in a conference? |
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| 09:36.41 | KNERD | Loading DAHDI hardware modules: |
| 09:36.43 | KNERD | <PROTECTED> |
| 09:37.06 | KNERD | How can I dereference this from DAHDI? |
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| 09:44.13 | KNERD | alami: flash operator panel |
| 09:44.24 | stevePearPear | hi |
| 09:44.31 | phpboy | oh hi stevePearPear |
| 09:44.51 | stevePearPear | i just started with asterisk today and been reading the 4th edition, really informative :) |
| 09:44.57 | stevePearPear | i have a query though |
| 09:45.35 | stevePearPear | i managed to connect to my sip trunk provider by writing down the host, defaultuser and secret in sip.conf |
| 09:46.26 | stevePearPear | is there anyway that I could dynamically use different defaultuser/secret? For example one user can pass me the credential and I will create a call to the sip trunk provider based on the credential |
| 09:50.19 | TazzNZ | stevePearPear: no, you need a "peer" definition |
| 09:50.53 | TazzNZ | unless you use something like Asterisk Realtime |
| 09:52.04 | stevePearPear | I saw a recommendation for Asterisk Realtime too, stating that it can allow me to load dyanmic config from database |
| 09:53.11 | stevePearPear | I am checking if there’s any other alternatives Just want to check if I would be able to extrapolate the userid and password that the user has passed me and create a peer instead |
| 09:54.50 | TazzNZ | stevePearPear: not that I am aware of - unless you have something that creates the file for you, and reloads asterisk |
| 09:55.24 | WIMPy | stevePearPear: What's your idea? |
| 10:01.36 | stevePearPear | I have a group of users who have sip trunk accounts with a provider |
| 10:01.52 | stevePearPear | however the provider doesn’t support websocket and I am creating an application |
| 10:02.43 | stevePearPear | i would like my user to key in their sip credentials (through JSSIP) and communicate to Asterisk where I could be a relay between the provider and the users and potentially adding other features such as call recording |
| 10:05.39 | *** join/#asterisk kayatwork (~kayfox@orca.zerda.net) |
| 10:05.45 | WIMPy | I don;t see any way other than the user "dialling" all the information every time. |
| 10:06.37 | stevePearPear | can you help elaborate what is it by user dialing all the information everytime? |
| 10:07.53 | WIMPy | They will have to put hostname, username, password and the destination all together in the destination field. |
| 10:08.12 | WIMPy | Woudn't be very userfirendly. |
| 10:10.02 | stevePearPear | ic, that can be resolve as I could use JSSIP to append those information |
| 10:10.31 | stevePearPear | however doing that, it would mean that my user wouldn’t be able to recieve a call since there’s no registration is it? |
| 10:11.34 | stevePearPear | what you said is actually this? Dial(technology/user[:password]@remote_host[:port][/remote_extension]) |
| 10:12.02 | WIMPy | You need spme way to register them to you and then passing on both the registration and calls. |
| 10:14.08 | stevePearPear | yeah I was researching about something called outbound proxy, which could relay all messages (something like Kamailio) |
| 10:14.12 | stevePearPear | can Asterisk simply relay? |
| 10:14.29 | WIMPy | no |
| 10:14.45 | WIMPy | Asterisk is a |
| 10:14.50 | WIMPy | ~b2bua |
| 10:14.50 | infobot | hmm... b2bua is a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 10:15.20 | stevePearPear | icic |
| 10:15.42 | stevePearPear | when you mentioned passing on registration and calls, can you give some direction (libraries) that I could look into to perform it? |
| 10:16.12 | WIMPy | To do it what way? |
| 10:16.40 | WIMPy | You might want to do it in another way, another channel. |
| 10:16.48 | stevePearPear | when the user registers with me, how could i pass on the registration |
| 10:17.33 | WIMPy | Maybe you should do that part via http? |
| 10:20.52 | stevePearPear | something along the line on my asterisk forward the registration by a http request |
| 10:20.53 | stevePearPear | ? |
| 10:21.04 | *** join/#asterisk CeBe1 (~CeBe@wlan-141-23-95-31.tubit.tu-berlin.de) |
| 10:21.21 | WIMPy | Set up the user via http |
| 10:21.48 | WIMPy | Then use whatever to configure Asterisk to use that information. |
| 10:23.12 | stevePearPear | ic, I am actually flipping Asterisk book furiously now :) |
| 10:23.20 | stevePearPear | is it something to do with AMI? |
| 10:23.28 | stevePearPear | I read on how we can talk to Asterisk using http |
| 10:23.31 | WIMPy | No |
| 10:24.09 | WIMPy | It's something that you have to write an application for. |
| 10:25.06 | stevePearPear | is it something like the application that I’m writing will store the user into a database where Asterisk could use that information? |
| 10:25.24 | WIMPy | For example. |
| 10:32.17 | stevePearPear | yeah? ;) |
| 10:34.38 | *** join/#asterisk timahvo1 (~rogue@197.237.131.169) |
| 10:39.42 | cusco | hi |
| 10:39.50 | cusco | I'm looking at the options of CDR() |
| 10:40.11 | cusco | option r .. I'm a bit dubious as of what it means |
| 10:40.19 | cusco | suposed to be recursive ? |
| 10:41.03 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 10:42.14 | cusco | so if I Set(CDR(customField,r)=something); |
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| 10:52.22 | snmp | core show function CDR |
| 10:52.25 | snmp | > r: Searches the entire stack of CDRs on the channel. |
| 10:53.24 | snmp | isnt clean,pardn me |
| 10:54.05 | snmp | need examples? |
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| 11:00.59 | cusco | snmp: yes, when setting, not retrieving |
| 11:01.17 | cusco | will it set it on sub sequent cdr records? |
| 11:31.46 | cusco | and.. another question... originate places a call in queue (local channel) and goes to a context, exten when answered.. that exten is basically a dial ... but before the dial we're setting some cdr values... they're not being stored |
| 11:32.21 | cusco | I'm reading in https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CDR that values cannot be modified after a call being bridged? so we should modify them before dial() |
| 11:33.22 | cusco | so... can't I set a cdr value there ? |
| 11:36.10 | alami | KNERD: ok thanks |
| 11:38.32 | cargill | how can I subscribe to a different mailbox than my own? when trying to subscribe to a different mailbox, asterisk sends me my mailbox info, not the one for my mailbox |
| 11:38.54 | cargill | *not the one for the mailbox I subscribed to |
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| 12:00.19 | *** join/#asterisk sted (~sted@ppp-94-68-80-60.home.otenet.gr) |
| 12:00.25 | sted | hey guys |
| 12:02.34 | sted | do you know how can select a context before a call in ip phones? |
| 12:02.44 | sted | using programmable buttons? |
| 12:05.44 | *** join/#asterisk iulhk (iulhk@39.34.137.112) |
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| 12:16.57 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 12:45.48 | malcolmd | a context? you can't select an asterisk diaplan context from the phone; the "context" is just an asterisk concept for organizing endpoints into groups that have common dialing capabilities. the context is defined by the channel driver, whether it be chan_sip or chan_pjsip |
| 12:47.15 | sted | ok then to select trunk |
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| 13:14.35 | *** join/#asterisk MKEbrew (4581b2fe@gateway/web/freenode/ip.69.129.178.254) |
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| 13:28.59 | MKEbrew | heya, anyone around? |
| 13:29.06 | [TK]D-Fender | possibly |
| 13:29.09 | MKEbrew | sweet |
| 13:29.37 | MKEbrew | wondering if anyone can talk to me about sending DTMF during early media. having a hard time finding what I'm looking for via Google |
| 13:30.36 | [TK]D-Fender | Any audio set during early media would make it an actual call... |
| 13:31.43 | MKEbrew | hmm.. what I'm trying to do is send a 4 digit access code for long distance calls. the carrier prompts us for a PIN before completing the call. |
| 13:32.11 | [TK]D-Fender | Not sure since I've never had to mess with it myself |
| 13:33.02 | MKEbrew | alright, no worries. I'll hang around for a bit. maybe someone can point me in the right direction. |
| 13:34.03 | cusco | hi |
| 13:34.33 | cusco | so I'm using ForkCDR() to be able to set some values on cdr |
| 13:34.44 | cusco | can I make the previous cdr not be written? |
| 13:35.07 | cusco | can I use NoCDR() and just after, ForkCDR |
| 13:35.08 | cusco | ? |
| 13:42.07 | michael_work | MKEbrew, did you try to use DTMF in Dial() ? or it is sent only after call is connected? |
| 13:45.32 | MKEbrew | I have tried somethign like Dial(Local/${EXTEN},,D(www1234)) but I still am prompted for the tone |
| 13:45.58 | MKEbrew | I think becuase it has not connected yet - or I'm just doing it wrong |
| 13:47.29 | MKEbrew | I also tried putting a SendDTMF() command on the next line in the dialplan, but I'm guessing it doesn't get to that until it's connected? and it doesn't connect until the PIN is dialed |
| 14:01.08 | r00f | you could connect call to some extension, which executes SendDTMF before DIAL. so its sends DTMF and after that calls your SIP. but you will have to find a way to ask * to originate that call. so it is only as a last option, i guess |
| 14:01.28 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 14:01.54 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-mrgqwuhztmguvhav) |
| 14:01.54 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:03.05 | michael_work | as early media is not connected yet it's little bit confusing. As you could use marco or sub on connection or to use G(newdualplan^${EXTEN},1) |
| 14:03.15 | MKEbrew | that's interesting r00f. We are using a PRI and the reason I am trying to automate this is for our fax lines that are connected to * via analog card |
| 14:04.02 | Katty | hello my asterisk does not work at all how to fix pls is urgent thx |
| 14:04.40 | r00f | Katty paste your /etc/passwd to somewhere and don't show us, it could help |
| 14:05.36 | Katty | what is etcpasswd pls??? |
| 14:05.44 | Katty | Nugget: holy shoes batman. |
| 14:05.55 | Katty | Nugget: size 5 by chance? ;> |
| 14:06.18 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
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| 14:07.19 | fullstop | Hi. Silly question here, but how many 729 licenses do I need to have to record VM? |
| 14:07.32 | fullstop | The incoming call is in g729 |
| 14:08.18 | fullstop | I have one license today, but my voicemails all seem to be 0s long |
| 14:09.14 | r00f | MKEbrew you could achieve that with a dirty hack in call routing . I don't know how to do that with early media |
| 14:09.51 | MKEbrew | Thanks for the suggestions, r00f |
| 14:14.09 | r00f | considering you are using fax, it could be as such: fax calls some extension, it answers and reads dtmf. fax enters the long distance number it wants to dial, and it hangs up. then it calls that number, sends the dtmf needed, and after that calls the fax back and connects them |
| 14:14.50 | r00f | or, as it is fax, even easier - extension receives fax, and then just sends it with SendFAX. no need for callback |
| 14:15.25 | r00f | i'd fall for the second option, as callback may confuse oldschool users |
| 14:16.14 | jameswf | They updated fax... its now called email :-/ |
| 14:16.38 | r00f | i know. personally i hate faxes. but people somehow still use them |
| 14:16.42 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-fjoaiegsmdcxmwnr) |
| 14:16.51 | r00f | shrugs |
| 14:17.38 | r00f | for my users, i created dynamic feature - press 9 during call and fax received will get delivered to your mailbox |
| 14:17.43 | r00f | first they hated me |
| 14:17.49 | r00f | but now they love me |
| 14:17.55 | r00f | :-p |
| 14:17.59 | r00f | no moar faxes |
| 14:21.20 | MKEbrew | interesting. we are in the insurance industry and faxing is still as regular today as it was years ago. We don't use Asterisk fax, we are using a fax server that integrates with our line of business application. |
| 14:22.04 | [TK]D-Fender | [10:08]fullstopI have one license today, but my voicemails all seem to be 0s long <- codec isn't why otherwise your call would have dropped |
| 14:22.33 | fullstop | I'll look at the logs, then. It all started when I switched to g729, so I likely broke something else. |
| 14:22.52 | alami | any one know some thing like MeetMe-Web-Control, this one is old and don't work with new Asterisk version |
| 14:23.14 | anonymouz666 | GS is moving toward android land |
| 14:23.17 | file | falls into existence |
| 14:23.28 | anonymouz666 | GXV3240 it seems to be nice equipament |
| 14:23.53 | anonymouz666 | android 4.2 |
| 14:24.31 | anonymouz666 | you can use also lync, skype, hangouts, etc. |
| 14:24.31 | MKEbrew | r00f, even using SendFAX whow/when would I send the dtmf as the call doesn't "connect" ubntil dtmf is sent. |
| 14:24.51 | r00f | MKEbrew just wondering why is fax better than "email as pdf" option. there are lots of free pdf printers, and manager could as well print to pdf from crm, instead of creating hard copy and then faxing it. it would save alot of trees |
| 14:25.12 | MKEbrew | I almost need to bridge the call and send the dtmf that way or something. |
| 14:25.30 | MKEbrew | fax vs email is an industry thing. trust me, I have been banging my head on it for years now |
| 14:25.59 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 14:26.07 | MKEbrew | certain things are required to be faxed, other thigns can be emailed, then other thigns need to be emailed securely |
| 14:26.15 | r00f | i c |
| 14:26.51 | MKEbrew | also, even thogu hwe are "faxing" it is a paperless process on our end and very likely on most recieving ends as well |
| 14:27.36 | MKEbrew | print to fax server print driver (same as print to pdf) - receiving side probably emails same way ours does |
| 14:27.37 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 14:28.00 | r00f | so basically you transfer email without internet ) |
| 14:28.34 | MKEbrew | lol, basically, yes |
| 14:29.14 | MKEbrew | like I said, it's an industry thing and the insurance industry moves slow. |
| 14:30.33 | r00f | so, imagine that you have some exten, which answers and reads dtmf. after that it hangs up the call and runs some script, which crafts the call file. that file calls the remote party and connects it with some context, which can have something like this |
| 14:30.36 | r00f | exten => _X.,1,Answer() |
| 14:30.36 | r00f | exten => _X.,2,SendDTMF(1234) |
| 14:30.36 | r00f | exten => _X.,3,Dial(DAHDI/faxnumber) |
| 14:31.15 | r00f | so it will send dtmf and then call your fax. i have this working good for me (without dtmf, for other purpose) |
| 14:32.02 | MKEbrew | I think I understand that. I will give it a try |
| 14:34.22 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:34.22 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:36.39 | *** join/#asterisk wasanzy (~wasanzy@197.159.129.10) |
| 14:36.54 | wasanzy | Hello |
| 14:37.00 | *** join/#asterisk timahvo1 (~rogue@197.237.131.169) |
| 14:37.16 | r00f | sup |
| 14:37.18 | *** join/#asterisk zerick (~eocrospom@190.118.43.113) |
| 14:37.49 | wasanzy | I have asterisk setup alright but when I make a call with mobile phone, am getting this error from tcpdump: SIP/2.0 401 Unauthorized |
| 14:38.48 | wasanzy | amazingly, asterisk is not producing the debugging logs for me to see even though I put asterisk in debug. So I had to use tcpdump before I see the error |
| 14:39.12 | Katty | looks in |
| 14:39.44 | wasanzy | We have a media gateway which connects to MNO's ss7 network so media gateway sends the request to asterisk, the same error is on media gateway |
| 14:40.14 | r00f | so maybe it is not authorized? |
| 14:40.35 | Katty | tugs on [TK]D-Fender's sleeve |
| 14:41.38 | wasanzy | r00f: the username and password is correct so how else should I authorize? |
| 14:42.51 | MKEbrew | Katty, what specifically is the problem you are having? |
| 14:44.25 | r00f | if it is peer, try adding insecure = port, invite |
| 14:46.05 | Katty | MKEbrew: well... |
| 14:46.21 | Katty | MKEbrew: i got my hair re-dyed yesterday and they kind of got it everywhere |
| 14:46.31 | wasanzy | r00f: it is not |
| 14:46.38 | Katty | MKEbrew: that's clearly a problem. |
| 14:46.47 | wasanzy | type=friend |
| 14:47.00 | Katty | hmm what else. |
| 14:47.18 | wasanzy | should I past my sip.conf? |
| 14:47.19 | Katty | MKEbrew: oh! my drink is nearly empty. |
| 14:48.06 | Katty | MKEbrew: and it's not friday. that's a problem too. |
| 14:48.34 | r00f | lucky me. i have weekend on friday and saturday :-p |
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| 14:51.15 | MKEbrew | makes sense |
| 14:54.43 | Katty | feels like friday |
| 14:56.02 | r00f | cya, gtg |
| 14:58.16 | wasanzy | anymore help? |
| 14:59.21 | *** join/#asterisk timahvo1 (~rogue@197.237.131.169) |
| 15:00.30 | Katty | wasanzy: well i'm not a therapist, but i'll listen |
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| 15:12.32 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:12.44 | wasanzy | hmmm |
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| 15:29.26 | *** join/#asterisk matthew-moretalk (~matthew@88.96.27.150) |
| 15:30.40 | matthew-moretalk | Hi Guys strange one for you all. I have a rasberry pi machine as an SSH CLIENT and I ssh into my pbx and run asterisk -r but the ssh session completly locks up every time it hits a Goto in the verbose??? |
| 15:30.42 | matthew-moretalk | any ideas?? |
| 15:32.33 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 15:35.23 | Katty | looks in |
| 15:36.03 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 15:36.10 | jameswf | What goto |
| 15:36.19 | Katty | hi jameswf |
| 15:36.36 | jameswf | waves |
| 15:37.10 | matthew-moretalk | Hi Jameswf the pbx runs the GoTo statement absolutly fine but the output crashes the ssh connection which I just dont understand |
| 15:37.42 | jameswf | What is the goto |
| 15:38.53 | matthew-moretalk | <PROTECTED> |
| 15:38.54 | matthew-moretalk | <PROTECTED> |
| 15:38.54 | matthew-moretalk | <PROTECTED> |
| 15:40.02 | matthew-moretalk | output runs absolutly fine in Putty on my windows 7 box and the server excutes the command and other stuff fine |
| 15:40.08 | jameswf | matthew-moretalk: what do you see from a real computer |
| 15:40.33 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 15:41.34 | matthew-moretalk | <PROTECTED> |
| 15:41.34 | matthew-moretalk | <PROTECTED> |
| 15:41.34 | matthew-moretalk | <PROTECTED> |
| 15:41.34 | matthew-moretalk | <PROTECTED> |
| 15:41.34 | matthew-moretalk | <PROTECTED> |
| 15:41.34 | matthew-moretalk | <PROTECTED> |
| 15:41.34 | matthew-moretalk | <PROTECTED> |
| 15:41.35 | matthew-moretalk | <PROTECTED> |
| 15:41.35 | matthew-moretalk | <PROTECTED> |
| 15:41.36 | matthew-moretalk | <PROTECTED> |
| 15:41.36 | matthew-moretalk | <PROTECTED> |
| 15:41.37 | matthew-moretalk | <PROTECTED> |
| 15:41.40 | matthew-moretalk | etc etc |
| 15:44.30 | ChannelZ | ~pb |
| 15:44.31 | 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. |
| 15:44.35 | ChannelZ | Don't flood the channel like that |
| 15:45.46 | *** join/#asterisk timahvo1 (~rogue@197.237.131.169) |
| 15:46.26 | cusco | I've been trying to set CDR values on the second call leg from originate ... |
| 15:47.18 | *** join/#asterisk navaismo (~navaismo@187-178-254-98.dynamic.axtel.net) |
| 15:47.44 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 15:47.56 | matthew-moretalk | a pastebin of my issue http://pastebin.com/Y1MDfGRX |
| 15:48.30 | cusco | I'm not understanding how I should overcome this |
| 15:50.34 | navaismo | if this a ssh connection to the raspberry? Maybe packet lost, what happen if you attach a monitor to the pi and see the cli still happen? |
| 15:51.01 | matthew-moretalk | this is with a monitor to the PI |
| 15:51.19 | matthew-moretalk | the Pi is a SSH Client to an asterisk Virtual Machine |
| 15:51.30 | navaismo | ok, happen on all ttys? |
| 15:51.35 | *** join/#asterisk jaflong (5bec7504@gateway/web/freenode/ip.91.236.117.4) |
| 15:52.22 | matthew-moretalk | what do you mean by ttys? sorry my linux knowledge is a little rough |
| 15:52.27 | cusco | here I'm testing with a .call file, and test dialplan ... http://paste.debian.net/103579/ |
| 15:52.45 | cusco | I can't get it to write the 'bar' value, on CDR |
| 15:52.53 | cusco | how can I overcome this? |
| 15:53.21 | navaismo | matthew-moretalk, if you open another terminal by pressing ctrl+alt+f2(or f3 or f4) and then connect to the asterisk still happen? |
| 15:53.38 | jaflong | hi, is it possible to share a variable containg a channel name ascross contexts. Tried to use group but cant find a solution so far |
| 15:53.56 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 15:54.07 | navaismo | jaflong, didi you tried with double underscore: __MYVAR |
| 15:54.22 | *** join/#asterisk roentgen (~none@openvpn/community/support/roentgen) |
| 15:55.10 | jaflong | goes that add to sip header? |
| 15:55.14 | *** join/#asterisk danjenkins_ (~dan@nat/digium/x-hzdgiufcjfeggocb) |
| 15:55.33 | davlefou | hi, i have to use ami to made clicktocall? |
| 15:55.36 | navaismo | uh? sorry i think you only need to pass variables not sip headers |
| 15:55.59 | navaismo | davlefou, ~lmgtf |
| 15:56.06 | navaismo | davlefou, ~lmgtfy |
| 15:56.55 | jaflong | no I dnot need sip header I want to avoid it. I am looking for most efficent way |
| 15:57.07 | davlefou | navaismo, have allready look google, all seems use ami to made clicktocall, |
| 15:57.26 | navaismo | no the hype now is webrtc LOL |
| 15:58.39 | navaismo | matthew-moretalk, dont pm me stay in the channel please, sounds like a shell issue what distro are you using? |
| 15:59.18 | navaismo | jaflong, http://www.voip-info.org/wiki/view/Asterisk+variables |
| 15:59.19 | matthew-moretalk | happens on all ttys |
| 15:59.46 | matthew-moretalk | raspbian |
| 16:00.20 | navaismo | weird, I not use that but never experimented that on fedora for pi so who knows |
| 16:00.46 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 16:00.47 | matthew-moretalk | intrestingly if I do it from full console mode it kills the whole pi and I have to hard reboot it. if I do it from a terminal window from desktop it locks that terminal |
| 16:00.47 | navaismo | but i dont think this is an asterisk issue |
| 16:01.03 | matthew-moretalk | could it be the syntax hightlighting in asterisk CLI? |
| 16:01.37 | *** join/#asterisk ulogic (421e6b4f@gateway/web/freenode/ip.66.30.107.79) |
| 16:02.00 | navaismo | dont think so but you can try to disable and see if fix it |
| 16:04.03 | navaismo | cusco, that values inside the CDR function are in asterisk 12? in asterisk 11 i cant see it or is like odbc magic, or patched cdr? |
| 16:04.26 | navaismo | but anyway you can enable the debug and see what happen when it try to query the db |
| 16:04.50 | *** join/#asterisk imcdona (~imcdona@75-151-101-98-Washington.hfc.comcastbusiness.net) |
| 16:07.58 | cusco | navaismo: asterisk 11 |
| 16:08.16 | cusco | 11.10.0 |
| 16:08.57 | cusco | navaismo: I'm querying the db, the queue exten record has the value set |
| 16:09.02 | cusco | but the dial, doens't |
| 16:09.19 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
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| 16:09.31 | cusco | I also read in https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CDR that values cannot be modified after a call being bridged? so we should modify them before dial() |
| 16:09.44 | cusco | maybe because of a originate or .call file this call is bridged? |
| 16:09.48 | cusco | I'm not understanding |
| 16:10.09 | navaismo | neither do i |
| 16:10.11 | navaismo | :D |
| 16:10.12 | cusco | I was also trying to play around with NoCDR() and ForkCDR |
| 16:10.25 | cusco | :/ |
| 16:10.58 | *** join/#asterisk petris (~petris@2607:5300:60:5475:a9d7:583:99b:a901) |
| 16:11.03 | cusco | in the verbose I see the Set() and the query does not have the value |
| 16:17.03 | cusco | navaismo: what did you mean by odbc magic vs patched cdr? |
| 16:18.23 | navaismo | i usually use a pacthed version of the cdr adding new variables so I just can call the set(CDR(mynewVAR)=foo) and that write using crd_mysql, but the line is to use odbc |
| 16:18.29 | navaismo | since mysql is deprecated |
| 16:20.11 | cusco | navaismo: at the moment we're using both :p, but never had to patch cdr |
| 16:20.20 | cusco | mysql can also add custom values |
| 16:20.37 | cusco | bue yes, I'm looking at odbc (mssql) only |
| 16:20.45 | cusco | let me see in mysql |
| 16:21.15 | TazzNZ | I'd say odbc can |
| 16:21.39 | TazzNZ | the whole reason behind odbc is that it takes away the specific's of a DB - like MySQL or MSSql |
| 16:21.41 | TazzNZ | etc |
| 16:22.02 | TazzNZ | and I doubt that Digium would allow a feature like that to be dropped from the source |
| 16:22.13 | cusco | anyway, I'm troubled trying to find a solution to my issue :/ |
| 16:24.41 | TazzNZ | cusco: only with the .call file ? |
| 16:24.49 | cusco | also with originate |
| 16:24.55 | cusco | I just set this call file for testing |
| 16:25.06 | TazzNZ | what are you going to use in production ? |
| 16:25.52 | cusco | originate |
| 16:26.03 | cusco | its already in production |
| 16:26.09 | cusco | I was trying to add a custom cdr field |
| 16:26.15 | malcolmd | yeah, ODBC support isn't going away, it's the preferred way |
| 16:26.16 | cusco | so it writes on the first call leg |
| 16:26.19 | cusco | but not on the second |
| 16:27.52 | cusco | just tested it |
| 16:27.56 | cusco | on mysql the same happens |
| 16:28.00 | cusco | foo get written |
| 16:28.01 | cusco | but not bar |
| 16:28.26 | TazzNZ | cusco: what does the cli show when you do that call ? |
| 16:28.28 | cusco | malcolmd: care to take a look? :P http://paste.debian.net/103579/ |
| 16:28.57 | cusco | TazzNZ: the verbose shows the Set(), but the query does not contain the column/value to be added |
| 16:29.00 | cusco | :/ |
| 16:30.14 | TazzNZ | cusco: and turning up debug ? |
| 16:30.21 | cusco | er ... |
| 16:30.30 | cusco | ok... let me see.. |
| 16:31.00 | *** join/#asterisk glphvgacs (~glphvgacs@unaffiliated/glphvgacs) |
| 16:31.27 | glphvgacs | is sms defined in sip protocol? |
| 16:32.53 | TazzNZ | glphvgacs: as in messaging or SMS - Short Messaging service ? |
| 16:33.46 | TazzNZ | http://www.voip-info.org/wiki/view/Asterisk+cmd+SendText vs http://en.wikipedia.org/wiki/Short_Message_Service |
| 16:37.06 | glphvgacs | http://www.ietf.org/rfc/rfc3428.txt |
| 16:37.44 | cusco | TazzNZ: I don't see anything relevant, shows setting the variable to 'bar' but cdr query does not have it |
| 16:37.53 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 16:38.42 | cusco | let me look better |
| 16:39.14 | glphvgacs | http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms |
| 16:39.49 | *** join/#asterisk cmendes0101| (~cmendes01@office.phone.com) |
| 16:40.42 | TazzNZ | glphvgacs: using that - to answer your question, no, it's not |
| 16:43.52 | wasanzy | Hello |
| 16:43.57 | wasanzy | I still have the problem |
| 16:44.03 | wasanzy | what can I do? |
| 16:44.16 | wasanzy | SIP/2.0 401 Unauthorized |
| 16:48.19 | TazzNZ | wasanzy: authorize to the peer ? |
| 16:49.39 | wasanzy | is friend not peer |
| 16:50.11 | cusco | the debug log: http://paste.debian.net/103588/ has: Result of 'CUSTOMID' is NULL |
| 16:50.16 | cusco | line 277 on the paste |
| 16:51.09 | cusco | ah no |
| 16:51.12 | cusco | that is another thing |
| 16:51.13 | cusco | :/ |
| 16:56.11 | cusco | so I was wondering if I could forkCDR() to achieve disired effect |
| 17:05.32 | TazzNZ | wasanzy: ok then - authorize to the friend... |
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| 17:15.41 | wasanzy | TazzNZ: Yes |
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| 17:49.50 | matthew-moretalk | just so people know it was console colors that were crashing the console on rasberry pi! |
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| 18:04.02 | navaismo | hmm check your versions since you are the first person with that "bug" |
| 18:11.38 | *** join/#asterisk BooeyOH (~Bryan@rrcs-24-106-233-62.central.biz.rr.com) |
| 18:12.06 | BooeyOH | I couldn't find a recommended hardware list for AsteriskNOW. I am creating it as a VM and am wondering what sort of cpu/memory/hd I should use, any suggestions? |
| 18:15.18 | Qwell | BooeyOH: It's Linux. It'll use practically anything you can throw at it. |
| 18:15.43 | BooeyOH | so 2cpu/4gbmemory/100gb would work? |
| 18:15.48 | Qwell | sure |
| 18:16.23 | Qwell | If you need to expand later, it's a VM, so that's trivial. |
| 18:17.48 | *** join/#asterisk dxd828__ (~dxd828@dsl-dynamic-77-44-45-38.interdsl.co.uk) |
| 18:18.28 | BooeyOH | good point |
| 18:18.30 | BooeyOH | thanks! |
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| 18:46.19 | leifmadsen | BooeyOH: start smaller and expand if required |
| 18:46.24 | leifmadsen | usually you can make bigger, but not smaller |
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| 19:38.37 | TazzNZ | BooeyOH: also note - that unless you are doing direct media, you will run into timing issues with a VM |
| 19:38.52 | BooeyOH | TazzNZ, what do you mean? |
| 19:39.55 | TazzNZ | well - a VM is never 100% assured it will get CPU resources when needed. This leads to the VM not always being around when it needs to to flip the RTP packets |
| 19:40.12 | BooeyOH | TazzNZ, what if you reserve cpu usage for it? |
| 19:40.21 | TazzNZ | same issue |
| 19:40.31 | BooeyOH | interesting |
| 19:40.33 | BooeyOH | ok, thanks |
| 19:40.37 | TazzNZ | we have tried it with almost any of the VM stuff out there |
| 19:40.47 | TazzNZ | and we found KVM on Redhat the "best" |
| 19:41.07 | TazzNZ | it still has the issue when the host is under load |
| 19:41.17 | TazzNZ | imho, a VM for testing/dev is fine |
| 19:41.37 | Katty | hello my asterisk does not work at all how to fix pls? is urgent thx |
| 19:41.48 | TazzNZ | a VM where directmedia=yes is set will be ok for prod |
| 19:41.59 | TazzNZ | other than that, I would get physical boxes |
| 19:42.06 | TazzNZ | Katty: we need *way* more info |
| 19:42.18 | Katty | ok what info do you need?? |
| 19:42.30 | TazzNZ | well - to start with - what is broken ? |
| 19:42.36 | TazzNZ | or what doesn't work |
| 19:42.42 | Katty | the lock button on my iphone is broken. |
| 19:42.53 | TazzNZ | ... |
| 19:43.09 | Katty | leifmadsen: how did the zoo go? |
| 19:43.25 | leifmadsen | very zooery |
| 19:44.06 | Katty | leifmadsen: he's gettin so BIG! |
| 19:44.25 | leifmadsen | he's huge now |
| 19:44.33 | Katty | 20lbs? |
| 19:44.47 | leifmadsen | he was 20lbs like 3 months ago :) |
| 19:44.52 | leifmadsen | getting closer to 30 lbs now I think |
| 19:45.00 | Katty | gosh, what are you feeding him?! rhinos?! |
| 19:45.12 | TazzNZ | that is why he was at the zoo right ? |
| 19:45.19 | leifmadsen | mostly kittens |
| 19:45.19 | TazzNZ | :D |
| 19:45.22 | Katty | mmm kitten ears, my favorite. |
| 19:48.03 | Katty | leifmadsen: we're taking the miniture hooman to the zoo next weekend, me thinks |
| 19:48.13 | leifmadsen | nice! |
| 19:48.20 | Katty | he likes the penguins. |
| 19:48.54 | leifmadsen | mine wasn't really that interested in the animals, he just wanted to run around |
| 19:49.11 | Katty | an admirable goal! |
| 19:49.25 | Katty | not too many bumps and bruises i hope? |
| 19:50.17 | leifmadsen | sooooo many |
| 19:50.25 | Katty | poor thing |
| 19:50.29 | leifmadsen | cuts, scrapes, bruises! |
| 19:50.31 | leifmadsen | all self inflected |
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| 21:06.27 | pjensen00 | When using PJSIP, none of the sip headers seem to be populated on the called channel when I do a "Dial". I'm executing the "U" option to do a Gosub on the called channel and trying to see the "To" and "From" headers but none seem to be there |
| 21:06.55 | pjensen00 | I definitely see them in the pjsip debug on the asterisk console, but I'm trying to access them in a dialplan to no avail |
| 21:07.50 | pjensen00 | Anyone got any ideas? |
| 21:11.18 | whizzi | Does anyone have an explanation why a Reg. Contact/Def. Username and IP-address would suddenly change to something completely random (other username) while in sip.conf the defaultuser is set to the username and the connecting phone hasn’t changed a bit ? |
| 21:11.45 | whizzi | this problem’s been bugging me a full week now and I just can’t seem to figure it out |
| 21:25.32 | pjensen00 | if it helps, the line that I"m seeing when I can't access any headers.res_pjsip_header_funcs.c:269 read_header: There was no datastore from which to read headers. |
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| 22:05.33 | whizzi | How can I empty the Def. Username cache in Asterisk 11 |
| 22:06.09 | whizzi | reloading chan_sip doesn’t help, changing defaultuser doesn’t help, reloading sip doesn’t work.. rebooting server doesn’t work… |
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| 23:25.05 | pjensen00 | I never see any PJSIP headers attached to an outbound leg unless I manually set them. Even the "From" and "To" are missing. Is this expected behavior? |
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| 23:40.14 | Greek-Boy | Anybody here have an idea why my Polycom SoundPoint IP335's can not accept 2 ringtones? I've followed howtos to the tee and am able to get the phone to show one ringtone in the menu but when I try to provision more than 2 ringtones then it will still only show the first one. |
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