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| 05:34.14 | taylorbyte2013 | i have ejabberd installed, i can't find any documentation on how to get asterisk users to use xmpp for chat |
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| 06:12.47 | ChannelZ | In what way are you expecting asterisk to be a part of it? |
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| 07:42.20 | taylorbyte2013 | ChannelZ: i think i have found out what i was wanting to do is be able to send instant messages between sip clients but i need asterisk 11 for that and im using 1.8 |
| 07:47.45 | ChannelZ | I guess I don't understand why you need asterisk to be involved at all if you're running an XMPP server... what client are you using? |
| 07:50.05 | ChannelZ | Yes Asterisk 11 can send/receive XMPP messages but seems like a poor way to do it just for people to be able to IM. |
| 07:51.55 | taylorbyte2013 | ChannelZ: i wanted to use the chat feature in different sip clients like jitsi and linphone so that i don't need multiple client programs for different IM methods like linphone for voip calls and pidgin for xmpp |
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| 08:52.43 | tengulre | hi,all |
| 08:53.59 | tengulre | I am beginner, how to cluster three asterisk server. |
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| 09:18.54 | Milos | google |
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| 09:47.35 | michael_work | anyone can give me an advice? i have fresh asterisk install as normal and same as other servers. But for some reasons when i start service and enter cli(verbose and debug are on) it takes about 1-2 mins till i see output that it loads modules and settings and till then most modules unloaded and if i even load them i see no changes (e.g. module load sip has no output and sip show peers would show no output as well) |
| 09:48.13 | michael_work | 11.11.0 |
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| 09:48.18 | WIMPy | DNS failure? |
| 09:48.29 | Stefan27 | is 13.0.0 really the latest version of asterisk 13? |
| 09:50.09 | michael_work | WIMPy, yeap |
| 09:50.11 | michael_work | thanks :) |
| 09:51.19 | michael_work | the computer had been setup on other network and DNS kept from there :( |
| 09:51.20 | michael_work | thanks |
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| 10:21.30 | ipalmer | Hi all, I have Asterisk 11.3 setup using realtime queues, I found a list of column names for the realtime queues table but it doesn't appear to have the reltaive-periodic-announce column, I have tried setting it in queues.conf and doing a module reload app_queue.so but the change doesn't take. Am I able to just add this column to the table or is there more work in the underlying code which would need to be changed |
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| 11:04.57 | Geek-Linux | Hello: i want to ask about the RTP probabation. when the message is thrown on the cli vebose "Probation passed - setting RTP source address to". is this related to some specific parameter that must be passed from the remote end. |
| 11:05.50 | WIMPy | NNo. It's more like statistics. |
| 11:06.21 | Penguin | No, it's just a check of the address and then a "lock." |
| 11:07.11 | Penguin | It's really nothing to worry about. |
| 11:08.04 | WIMPy | Well, not the message. But what happens is rather important. |
| 11:09.05 | Geek-Linux | Penguin: i am facing issue of One Way audio, when the bridging is done and i see this message from the peer end then i can hear voice from both ends but when the message is not thrown by asterisk i observe one way audio. |
| 11:10.29 | Penguin | Are you saying that sometimes the probation period never ends? or the probation does not pass? |
| 11:10.46 | WIMPy | Or just nothing is received? |
| 11:10.48 | Geek-Linux | Yes Exactly |
| 11:11.12 | WIMPy | How do you say yes to an or? |
| 11:12.50 | Geek-Linux | i have check the signalling packets in traces the RTP server IPs are present in the oneway audio traffic. but in that case i can see that the rtp debug shows only the sent packet but dont show the recieved packets. |
| 11:14.18 | Geek-Linux | Pengiun what you mean by probation period never ends ? is there any timer that is set ? |
| 11:14.31 | WIMPy | yes |
| 11:15.00 | Geek-Linux | WIMPy: where could i figure this out ? |
| 11:16.11 | WIMPy | YOu can only enable/disable it. But are you sure you do receive anything? |
| 11:16.33 | Geek-Linux | yes i am sure. |
| 11:16.47 | WIMPy | Disabling it is a well known security issue, so don't. |
| 11:17.16 | Geek-Linux | then what could be the solution ? |
| 11:17.31 | WIMPy | Does it com different ways or something? |
| 11:17.36 | WIMPy | come |
| 11:18.16 | Geek-Linux | the RTP traffic ? |
| 11:18.24 | WIMPy | yes |
| 11:18.53 | Geek-Linux | yes there are multiple servers at telco end for RTP and single server for signalling. |
| 11:19.12 | WIMPy | For that one call. |
| 11:20.00 | WIMPy | You can safely disable strictrtp if you don't have nat=yes. |
| 11:20.06 | Geek-Linux | no for single call they pass single RTP IP in signalling packet. but it can be any of the 4 IPs. |
| 11:20.29 | WIMPy | The signalling doesn't matter. |
| 11:20.55 | WIMPy | This is just about what really happens, i.e. required for nat support. |
| 11:21.01 | Geek-Linux | i had disabled this option and yes we dont use nat. but the issue remains the same. |
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| 12:26.08 | dan_j | Hi. I need to make a menu system that can record/update asterisk voicemail greetings. Its a customised job where the user shouldn't be able to access existing messages etc. It should be simple to use. |
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| 12:26.42 | dan_j | What method would you recommend for recording a greeting and then saving it into the correct location? I'm using odbc storage for voicemail. |
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| 12:28.07 | WIMPy | What's wrong with the built in version? Or what about it is it that you need to change? |
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| 12:29.25 | dan_j | There are currently too many mailbox options. I need to make a simple interface with 3 steps. 1) Enter mailbox number, 2) enter password, 3) record unavailable greeting. |
| 12:29.31 | dan_j | Without having to go through any other options. |
| 12:30.06 | dan_j | I also don't want to alter the actual comedian voicemail coding as that it used by other users. |
| 12:31.48 | WIMPy | And the greetings are stored in the db? |
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| 12:40.52 | s7r | my asterisk simple setup for 2 phones inside a lan will not open port 5060 for listening |
| 12:41.25 | s7r | why can this be |
| 12:41.31 | s7r | i have no firewall rules (iptables off) |
| 12:41.33 | s7r | my OS is debian |
| 12:43.24 | WIMPy | Either yiu don't have sip support enabled or you're looking for the wrong port. |
| 12:43.58 | s7r | i have bindport=5060 in sip.conf (1) |
| 12:44.02 | s7r | i am looking for port 5060 |
| 12:44.11 | s7r | what is sip support? i have just installed apt-get -y install asterisk |
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| 12:44.36 | WIMPy | What port 5060? |
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| 12:45.01 | WIMPy | And I don't know if you get a working config by installing a package. |
| 12:45.05 | s7r | port 5060 whichc is in my /etc/asterisk/sip.conf |
| 12:46.20 | WIMPy | How do you tell it's not listening? |
| 12:46.26 | s7r | netstat -nlt |
| 12:46.55 | WIMPy | That's the error. |
| 12:47.19 | WIMPy | Unless you also enable TCP, you won't see it that way. |
| 12:48.15 | s7r | oh yeah? |
| 12:48.55 | WIMPy | Usually there's no TCP involved. |
| 12:50.27 | s7r | hmm |
| 12:50.31 | s7r | my ATA does not connect to it |
| 12:50.36 | s7r | in this case |
| 12:51.00 | WIMPy | There are no connections. |
| 12:51.17 | WIMPy | Are you trying to register it? |
| 12:52.07 | s7r | I have just entered the Asterisk server ip address at proxy and the secret at password |
| 12:53.22 | WIMPy | I have no idea about the options of your ATA, but to register you usually have to set a registrar (host). |
| 12:54.23 | s7r | i have a cisco spa112 |
| 12:54.37 | s7r | i just have quick setup and it says proxy server, user id, display name, proxy and dial plan |
| 12:54.38 | s7r | :( |
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| 13:03.46 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 13:04.31 | ipalmer | <PROTECTED> |
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| 13:11.40 | s7r | anyone? |
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| 13:23.26 | dan_j | WIMPy: Did you reply to my question? Ive got a new router and it doesnt seem to be happy with irc connections. Keeps disconnecting and my irc client doesnt realise. |
| 13:23.55 | WIMPy | I asked: And the greetings are stored in the db? |
| 13:24.05 | dan_j | Yes. They are. |
| 13:24.34 | dan_j | I am correct in saying that i should use something like php and agi to record and then upload the recording to the DB? |
| 13:24.36 | WIMPy | Well, I have no clue how you'd get them there. |
| 13:25.22 | WIMPy | Not to record, but maybe to get it to the db. |
| 13:25.28 | dan_j | Same way asterisk does it surely? |
| 13:25.42 | dan_j | Or have I missed something? |
| 13:26.19 | WIMPy | Yes, but that's inside of the voicemail(main) application. |
| 13:27.03 | s7r | who can provide paid support for 20 minutes? |
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| 13:29.06 | [TK]D-Fender | ~ask |
| 13:29.06 | 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. |
| 13:29.08 | [TK]D-Fender | s7r: ^ |
| 13:29.19 | WIMPy | dan_j: YOu need a new old router. |
| 13:29.37 | [TK]D-Fender | Or an old new router |
| 13:30.50 | s7r | I am trying to connect 2 phones at my office inside a lan. I have an ATA Cisco SPA 112 with 2 phone lines/ ports. I have installed asterisk in Debian on a server on my lan. Configured in sip.conf the 2 phones (secret, host dynamic, port 5060, bindaddr, type friend, etc.) |
| 13:30.58 | dan_j | Switched to a cloud irc service for now. |
| 13:30.59 | s7r | the phones are connected to the ATA, but have no Tone. nor they can call each other |
| 13:31.46 | WIMPy | No tone as in no dialtone? Or what? |
| 13:31.57 | [TK]D-Fender | s7r: No dialtone means they have not registered to your server |
| 13:32.19 | [TK]D-Fender | s7r: "sip set debug on" <- go pastebin their registration attempts. |
| 13:32.21 | [TK]D-Fender | ~pb |
| 13:32.21 | 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. |
| 13:32.23 | [TK]D-Fender | ^^^^ |
| 13:32.26 | WIMPy | Interesting. |
| 13:32.42 | s7r | where do i sip debug on? |
| 13:32.47 | [TK]D-Fender | Asterisk CLI |
| 13:33.14 | s7r | no such command |
| 13:33.29 | [TK]D-Fender | Show us |
| 13:34.10 | WIMPy | No SIP support or Asterisk CLI not found (by user). |
| 13:34.42 | s7r | No such command 'sip debug on' |
| 13:34.58 | [TK]D-Fender | 37that is not the command I gave you |
| 13:35.03 | [TK]D-Fender | s7r: that is not the command I gave you |
| 13:35.11 | [TK]D-Fender | [08:32][TK]D-Fenders7r: "sip set debug on" <- go pastebin their registration attempts. |
| 13:39.13 | s7r | [TK]D-Fender it says debugging enabled |
| 13:39.17 | s7r | but shows nothing in the cli |
| 13:39.27 | s7r | the ata is ON |
| 13:39.48 | [TK]D-Fender | reboot it |
| 13:39.51 | Milos | did you enter the right IP in the ata |
| 13:39.52 | [TK]D-Fender | And check your firewall |
| 13:39.57 | Milos | in the proxy field |
| 13:40.00 | Milos | did you enter an IP at all |
| 13:40.02 | [TK]D-Fender | And of ccourse check your settings |
| 13:40.18 | s7r | yes |
| 13:40.20 | s7r | it's the right IP |
| 13:40.25 | s7r | in the ATA |
| 13:40.58 | Milos | in the proxy field? |
| 13:41.09 | Milos | is register set to yes? |
| 13:41.26 | [TK]D-Fender | [08:39][TK]D-Fenderreboot it <- the ATA |
| 13:42.18 | Milos | dances |
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| 13:47.01 | s7r | [TK]D-Fender http://fpaste.org/151415/62319861 |
| 13:47.34 | [TK]D-Fender | [Nov 17 13:41:05] NOTICE[2567]: chan_sip.c:25030 handle_request_register: Registration from '<sip:sender@192.168.1.25>' failed for '192.168.1.16:5060' - Wrong password |
| 13:47.39 | [TK]D-Fender | Seems pretty clear |
| 13:48.00 | s7r | the password is correct. the password is 1234 |
| 13:48.03 | s7r | :-s |
| 13:48.13 | [TK]D-Fender | s7r: something's wrong in the way you set it |
| 13:48.29 | [TK]D-Fender | show us the actual config from both sides |
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| 13:56.42 | s7r | http://fpaste.org/151420/16232581 |
| 13:57.41 | [TK]D-Fender | username=sender <-- should be "defaultuser" |
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| 13:58.13 | [TK]D-Fender | canreinvite=yes <- should be "directmedia=no" |
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| 13:58.55 | [TK]D-Fender | Those tend to cause issues across NAT (once You get to that point) |
| 13:59.04 | [TK]D-Fender | Go fix these and retest |
| 13:59.21 | [TK]D-Fender | insecure=port <- and remove this |
| 14:01.44 | s7r | ok fixed, reloaded |
| 14:01.45 | s7r | let me check |
| 14:02.58 | s7r | i have replaced username=sender with defaultuser=sender and still the same error as fist time, Wrong password |
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| 14:06.29 | [TK]D-Fender | Also verify the port and be sure to set it in each peer |
| 14:06.56 | s7r | the port ? |
| 14:06.59 | s7r | you mean 5060? |
| 14:11.48 | s7r | now the cli doesn't say anything but the ATA is tyring to connect |
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| 14:17.02 | s7r | [TK]D-Fender |
| 14:21.06 | s7r | the error is the same WRONG PASSWORD, destroying SISP dialog Method: REGISTER |
| 14:21.25 | *** join/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 14:21.27 | Zogot | Ahoy all |
| 14:28.42 | s7r | [TK]D-Fender got them to have tone |
| 14:28.52 | s7r | now can't call each other |
| 14:32.44 | s7r | if you can help me with this i will be forever greatful |
| 14:32.44 | s7r | :D |
| 14:32.52 | s7r | it's a problem in my extensions.conf file i think |
| 14:37.14 | [TK]D-Fender | Show us <---- |
| 14:40.01 | s7r | i just have 2 phones i want them to call each other. no pstn. my extensions.conf file has one lijne: exten => _X.,1,Dial(SIP/${EXTEN}) |
| 14:40.29 | s7r | but yet the phones cannot call each other. i pickup one phone, it has tone, after few seconds statrs beeping. the same if i press any key |
| 14:40.36 | davlefou | hi, i have to create an call center for receive from several on several number customer call. What i use? |
| 14:40.49 | davlefou | Queue and agent, something else? |
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| 14:43.31 | [TK]D-Fender | s7r: If if has just 1 line then it will not work |
| 14:43.36 | davlefou | Is it the good choice? |
| 14:44.10 | [TK]D-Fender | davlefou: Your wording is very hard to read. Please rephrase |
| 14:45.30 | s7r | [TK]D-Fender could you please please point me in the right way? i am on deadline to finish this and it's the first time i touch asterisk. i am reading documentation for the last 2 hours |
| 14:45.53 | [TK]D-Fender | s7r: If you have only one line.. that means you didn't even define a context |
| 14:46.23 | s7r | oh, i have context [internal] above |
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| 14:47.05 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:47.08 | [TK]D-Fender | Show us the failed call attempt |
| 14:47.49 | s7r | http://fpaste.org/151440/23545314 |
| 14:48.24 | [TK]D-Fender | that isn't a call |
| 14:48.59 | glaz | [TK]D-Fender: do you work at allstream? |
| 14:49.00 | [TK]D-Fender | If you're staring at CLI and then place a call and that is all you see then the ATA isn't even sending a call request |
| 14:49.58 | s7r | yes that is all i see |
| 14:50.05 | [TK]D-Fender | glaz: Do you work at Bell? |
| 14:50.17 | [TK]D-Fender | s7r: Then your device isn't sending the call |
| 14:50.19 | s7r | immediately after i press one key it starts beeping |
| 14:50.20 | glaz | [TK]D-Fender: used to... for 7-8 years |
| 14:50.44 | [TK]D-Fender | s7r: It has a dialplan on the device which decides when it is satisfied with what you are telling it to dial. |
| 14:50.52 | s7r | ohhh |
| 14:50.52 | s7r | yeha |
| 14:50.56 | s7r | I have dialplan line |
| 14:50.57 | s7r | but it's empty |
| 14:50.58 | s7r | :( |
| 14:51.06 | [TK]D-Fender | s7r: It does not appear to be happy. Fix the patterns |
| 14:51.26 | s7r | should i put _X.,1,Dial(SIP/${EXTEN}) ? |
| 14:51.36 | [TK]D-Fender | s7r: no, that is ASTERISK's dialplan language |
| 14:51.39 | glaz | [TK]D-Fender: iirc, you live in Montreal, or somewhere around that? |
| 14:51.52 | [TK]D-Fender | and that is both a pattern and steps to execute, etc |
| 14:51.57 | [TK]D-Fender | glaz: Yup |
| 14:52.05 | [TK]D-Fender | glaz: And no, I do not work for Allstream |
| 14:52.17 | s7r | so what dialplan should i have in order just to make one phone be able to call the other |
| 14:52.20 | s7r | no pstn |
| 14:53.03 | [TK]D-Fender | s7r: try: x.T|*x.T|#x.T |
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| 14:53.36 | [TK]D-Fender | Also PSTN exists at the point where you tell * to talk to a device that talks to the PSTN. |
| 14:54.10 | glaz | [TK]D-Fender: I privmsg'd you |
| 14:54.16 | [TK]D-Fender | What patterns you allow from you phone may matter little at the phone level. Your Asterisk dialplan decides what to do with whatever the phone sends. In this case I tend to let the phone send whatever you dial over |
| 14:54.25 | [TK]D-Fender | glaz: Here is fine. |
| 14:54.32 | glaz | it's personnal though... |
| 14:54.46 | [TK]D-Fender | sorry misread who it was from. |
| 14:55.17 | s7r | [TK]D-Fender it works |
| 14:55.18 | s7r | i love you |
| 14:55.50 | s7r | i HATE spoon feeding, i really do. i like to read. but right now i am at a deadline to make the system functional at a grain farm in the middle of nowhere so i had no choice but to be annoying |
| 14:56.01 | s7r | so thanks |
| 14:56.03 | s7r | thanks |
| 14:58.58 | [TK]D-Fender | 37You're welcome |
| 14:59.03 | [TK]D-Fender | s7r: ^ |
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| 15:11.08 | Zogot | asterisk 12 has realtime cdr support? |
| 15:11.39 | Zogot | is perhaps the schema for it not defined in alembic but somewhere else? |
| 15:14.03 | Zogot | ah, different folder, my mistake |
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| 15:32.35 | dan_j | Hi. Is there anything that would prevent me from playing back a file in /tmp, even though I can record a file to /tmp ? |
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| 15:33.10 | dan_j | Never mind. Sorted it. |
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| 15:55.13 | adeel | when using the func_ODBC module, when is the read/write portions of the function called? Write only when a SET command is issued? |
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| 17:45.21 | Katty | looks in |
| 17:48.49 | file | falls over |
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| 18:22.41 | dan_j | How do you debug php running via agi? |
| 18:23.00 | dan_j | I read somewhere that asterisk doesnt output php errors. |
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| 18:32.24 | [TK]D-Fender | dan_j: It doesn't you need to trap those in your script or in PHP's own log |
| 18:33.08 | dan_j | Would be nice to step through the PHP script. I need to check a large string is correct. Too much to verbose. |
| 18:33.45 | [TK]D-Fender | Not if you break it up. |
| 18:33.58 | [TK]D-Fender | And there are plenty of other means for checking it |
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| 19:05.01 | kippi_ | hey |
| 19:06.44 | kippi_ | what I want to do is to have a queue, you talk to one person and you press 1 you would move onto the next person. Am I better setting up a conf room for each user and then have a ivr that jumps to the next conf room or can I use some type of queue? |
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| 19:57.21 | *** join/#asterisk saint_ (~saint@198.15.119.130) |
| 19:57.31 | saint_ | hi all - in Asterisk 13 , does pjsip.conf replaces sip.conf ? |
| 19:57.49 | WIMPy | Only if you want. |
| 19:57.59 | mjordan | saint_: It isn't a replacement. It is a separate channel driver/stack with its own configuration. |
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| 19:58.28 | mjordan | you can use chan_sip with sip.conf; chan_pjsip with pjsip.conf; or technically both at the same time, so long as you're careful and don't bind things to the same ports |
| 19:59.40 | saint_ | mjordan , okay. is there any recommendation on how to do things ? or can I just keep my old sip.conf , and then experience new features with pjsip.conf ? |
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| 20:00.56 | saint_ | mjordan or can I simply get ride of the sip.conf , and put everything in pjsip.conf , since i read somewhere that the sip stack was better in this one ? |
| 20:00.57 | WIMPy | If you want the new features of pjsip, you need a new config. |
| 20:01.24 | saint_ | WIMPy okay , clear enough. i'll adventure myself in pjsip .. |
| 20:01.32 | mjordan | er |
| 20:01.42 | mjordan | saint_: have you read the documentation? |
| 20:01.49 | saint_ | mjordan i m right in it |
| 20:02.06 | saint_ | mjordan the wiki says UNDER CONSTRUCTION though .. |
| 20:02.13 | saint_ | with 2 child pages |
| 20:02.17 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip |
| 20:02.19 | mjordan | start there. |
| 20:02.28 | mjordan | you're probably in the development sections. Stay out of those :-) |
| 20:02.49 | mjordan | I'm curious, how did you find the page you were on? |
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| 20:05.14 | saint_ | mjordan i dont really remember. i went on an adventure of installing Asterisk 13 a couple of weeks ago on an esxi machine, with hard drives on a nas. once i got this running, and compile working, i left it with the wiki page i found at the time. i just went back into it today... and like i said, no clue how i ended up on that one. i'm pretty sure i was trying to follow step by step from the wiki though .. so it has to come from wiki.asterisk.org / |
| 20:05.14 | saint_ | <PROTECTED> |
| 20:06.01 | mjordan | wiki.asterisk.org contains documentation both for users and developers |
| 20:06.21 | mjordan | developers write notes on wiki pages to coordinate activities - but those notes are *not* meant as documentation when the feature is complete |
| 20:06.36 | mjordan | If you're looking in the Developer section, you have to take what you read with a grain of salt :-) |
| 20:07.13 | mjordan | anyway, the page I linked you to is the right one. |
| 20:08.09 | ipengineer | Is there a way to change the permissions that asterisk uses when creating voicemail directories? Basically asterisk is running as root and creating directories as root. I have other users that need to write to those directories that I dont want having root access. |
| 20:08.32 | ipengineer | If it could allow all users to write to them I would be good |
| 20:09.04 | WIMPy | Maybe you shouldn;t run it as root, but asterisk.conf also contains otions for permissions. |
| 20:09.31 | saint_ | mjordan thanks |
| 20:09.38 | ipengineer | I used to run it as âasteriskâ but at the end of the day I dont want asterisk running as the same user as our web server |
| 20:09.41 | newtonr | WIMPy, I don't think those options are specific to voicemail directory creation though |
| 20:09.51 | ipengineer | which needs access to write to those directories for updating vmail greetings.. |
| 20:10.16 | WIMPy | Err, why do you run your webserver as asterisk??? |
| 20:10.24 | ipengineer | WIMPy: I dont |
| 20:10.28 | WIMPy | newtonr: Ok, I haven;t tried them. |
| 20:10.57 | ipengineer | asterisk as root, php-fpm as nginx. nginx needs to be able to write into directories that Asterisk creates. |
| 20:11.06 | WIMPy | Ok, so how much do you need to do from the webserver? |
| 20:11.30 | ipengineer | Web server needs to be able to replace all voicemail greeting files |
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| 20:11.54 | WIMPy | So I doun't have to talk about groups, I guess. That would just be the same. |
| 20:12.28 | WIMPy | You could write a little application that moves a file there and checks the path that you install suid. |
| 20:13.27 | ipengineer | WIMPy: Yea. I could have a crontab job that picks up files and moves them but that just sounds so messy :) I hate stuff like that⦠Makes people down the line talk bad about me lol |
| 20:13.49 | WIMPy | Who said cron? |
| 20:14.21 | WIMPy | Just call that app instead of a normal mv/rename when you finished recording. |
| 20:15.14 | ipengineer | Ahh. Well a lot of those files get created when someone leaves a recording. Say you create an entry in voicemail.conf and ther is no directory structure when someone leaves a message it creates those directories |
| 20:15.27 | ipengineer | and default unavailable files as well I believe |
| 20:15.41 | Penguin | |
| 20:16.06 | WIMPy | Ok, so just ensure the path exists before moving. |
| 20:16.28 | WIMPy | He Penguin. You found that key again! ;-) |
| 20:16.36 | Penguin | It's not a key. |
| 20:16.52 | WIMPy | You do it with the mouse? |
| 20:17.00 | Penguin | Sometimes when I open another window on the screen, it pops out a line feed on this window. |
| 20:17.19 | Penguin | This window wasn't even in focus. |
| 20:17.25 | WIMPy | Strange stuff. |
| 20:17.28 | Penguin | very |
| 20:17.33 | Penguin | I've gotten used to it. |
| 20:17.44 | ipengineer | Hmm. Ok. I guess I could create the directory structure anytime we make a voicemail.conf entry. Asterisk will still create default unavailable greeting files but maybe I can still overwrite them if that is the case.. |
| 20:18.21 | WIMPy | Well, at least it's not as bad as hitting the mouse by accident and copying the contents of the window into itself. |
| 20:18.39 | Penguin | Luckily irssi has copy/flood protection. |
| 20:19.10 | Penguin | If you're going to paste more than X lines, you have to press Ctrl+k to permit it. I think I'm set to 5 lines now. |
| 20:19.11 | WIMPy | Depends. So does LIRC, but in that case it's bad luck. |
| 20:19.25 | WIMPy | Ah |
| 20:19.45 | WIMPy | ipengineer: I'm not suer you understood the idea. |
| 20:19.45 | Penguin | demonstrates |
| 20:20.12 | Penguin | (1419.55) -!- Irssi: Pasting 22 lines to #asterisk. Press Ctrl-K if you wish to do this or Ctrl-C to cancel. |
| 20:20.21 | WIMPy | I was suggesting to have an app to replace mv that has the permissions to overwrite files. |
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| 20:20.47 | WIMPy | Penguin: that sounds like a very good idea. |
| 20:20.53 | Penguin | It's lovely. |
| 20:21.32 | ipengineer | WIMPy: I think we are on different pages. You said âensure the path exists before movingâ. I am saying nothing is being moved. We add a new voicemail to voicemail.conf. Someone leaves a message in that mailbox and asterisk creates the appropriate directories to contain the message that was left along with the default unavailable greeting file. At that point nginx cannot write to those directories since it did not crea |
| 20:21.32 | ipengineer | them. |
| 20:21.33 | Penguin | paste_verify_line_count = 5 |
| 20:21.41 | Penguin | I could tune it if necessary. |
| 20:22.28 | WIMPy | ipengineer: It's (only) about the greetings files, isn't it? |
| 20:22.37 | ipengineer | WIMPy: yes |
| 20:22.58 | ipengineer | I need nginx to be able to replace those regardless of who/what created the original files |
| 20:23.07 | Penguin | With yum, if I remove a package, all of the dependencies are left in place. Is there a way to do a recursive removal? |
| 20:23.09 | WIMPy | Ok, so don't change anything about that and just handle the actual replacement of that file. |
| 20:23.42 | WIMPy | i.e. leave those as owned by root. |
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| 20:24.38 | ipengineer | Right. So now our PHP app tries to copy and replace the file there and it fails because NGINX does not have access. I cant think of a clean way around that. |
| 20:25.11 | WIMPy | Then read again what I wrote. It was all only about that very operation. |
| 20:28.05 | ipengineer | I see what you said but my question is how do we âhandle the actual replacement of that fileâ? Again, asterisk creates the file as root, not me. As soon as someone leaves a message asterisk does its thing I have no control over that. If I do not have something running as root I cannot modify it. |
| 20:28.59 | WIMPy | Byt writing an application that moves the file for you that has the neccessay permissions and can be called without them. |
| 20:29.18 | WIMPy | By |
| 20:29.24 | ipengineer | WIMPy: Ok. I gotcha.. |
| 20:29.44 | saint_ | so do I need chan_sip , if I am planning on using chan_pjsip ? |
| 20:29.47 | WIMPy | Just a special replacement for mv. |
| 20:29.59 | WIMPy | saint_: no |
| 20:30.21 | ipengineer | right and the php app will nened to be able to invoke the operation. |
| 20:30.25 | saint_ | WIMPy thanks. |
| 20:30.42 | WIMPy | ipengineer: Yes. hence the suid thing. |
| 20:31.51 | Penguin | I'm quickly approaching an asterisk 1.8 to asterisk 11 migration. |
| 20:32.15 | WIMPy | Asterisk 13 is out already! |
| 20:32.33 | saint_ | Penguin that s what i am working on now. from 1.8 to 13 |
| 20:32.39 | Penguin | I'm still using 1.8, so you can guess how long it will be before I get to 13. |
| 20:32.45 | ipengineer | WIMPy: Ok thanks for the help. I think that will work |
| 20:33.49 | WIMPy | ipengineer: And don't forget to validate the destination so that you don't build a security issue. |
| 20:34.42 | ipengineer | Right.. I will make the app very specific it will only write to that destination. |
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| 20:46.36 | *** join/#asterisk marlinc (~marlinc@ip1.weert.li.nl.cvo-technologies.com) |
| 20:46.39 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 20:50.49 | WIMPy | ipengineer: Sorry, that was dangerousely misleading. You need to check the source path as well, of course. |
| 20:50.51 | *** join/#asterisk kgunnIT (~kgunnIT@198-243-23-225.dia.static.qwest.net) |
| 20:50.56 | [TK]D-Fender | Penguin: 13 is an LTS release and should be about as stable as 11. You might want to use this time to save yourself another upgrade down the road for staying behind on LTS's |
| 20:51.10 | ipengineer | K will do. |
| 21:02.47 | *** join/#asterisk danjenkins_ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 21:13.21 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 21:17.50 | *** join/#asterisk lorenzo (lorenzo@nl.shell.enlab.net) |
| 21:18.05 | lorenzo | hi! is it possible to allow access to a outbound route according to the caller id? |
| 21:18.19 | lorenzo | (to the one placing the call through my pbx) |
| 21:18.25 | hardwire | no |
| 21:18.26 | hardwire | yes |
| 21:18.27 | [TK]D-Fender | Yes. That's why tehre is a field explicitly for that |
| 21:18.27 | hardwire | maybe |
| 21:18.29 | hardwire | probably. |
| 21:18.36 | hardwire | what he said. |
| 21:19.31 | hardwire | I'm confused on what enlab.net does now. |
| 21:19.56 | lorenzo | ? it's my provider |
| 21:20.04 | lorenzo | [TK]D-Fender: let me check |
| 21:21.25 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 21:22.40 | lorenzo | [TK]D-Fender: I can't seem to find it :( |
| 21:23.06 | [TK]D-Fender | lorenzo: Last field in your PATTERNS |
| 21:23.15 | [TK]D-Fender | It even SAYS "callerid" |
| 21:25.21 | lorenzo | ah, the /_ thing |
| 21:26.15 | [TK]D-Fender | There are tooltips there....I recommend reading them |
| 21:26.28 | lorenzo | where? |
| 21:26.32 | lorenzo | not on vi :p |
| 21:27.01 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 21:28.40 | [TK]D-Fender | lorenzo: Well you used the term "Outbound route" which is relatively exclusive to FreePBX. |
| 21:29.04 | [TK]D-Fender | lorenzo: If you are infact managing your own Asterisk server yourself you should not be using that term at all. |
| 21:29.16 | [TK]D-Fender | lorenzo: Because it isn't a "real" term to Asterisk |
| 21:29.48 | Penguin | Yeah, Asterisk doesn't have outbound routes, but does have peers and dialplan to utilize those peers. |
| 21:29.55 | lorenzo | yeah that's where I learned it |
| 21:30.08 | [TK]D-Fender | lorenzo: Now is a great time to forget it. |
| 21:30.19 | [TK]D-Fender | checkout time.. heading home... |
| 21:30.21 | lorenzo | okay :D |
| 21:31.43 | saint_ | [tk] me too. will be back laterz. |
| 21:34.38 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 21:39.20 | *** join/#asterisk skrusty (~skrusty@168.63.14.171) |
| 21:45.56 | *** join/#asterisk MadHatter42 (~tuwid@217.73.143.43) |
| 21:48.33 | Penguin | What's amportal? Is that a FreePBX thing? |
| 21:49.37 | malcolmd | i've no idea what all it's responsible for, but i use it to start and stop freepbx..it controls asterisk and dahdi startup among other things. yes, it's a freepbx thing. |
| 21:49.59 | Penguin | So I can safely remove it if I am not going to use FreePBX? |
| 21:50.47 | WIMPy | Will it be embedded in systemd? |
| 21:50.52 | WIMPy | scnr |
| 21:51.16 | malcolmd | if it's on your system, you're already using freepbx |
| 21:51.21 | Penguin | This is CentOS 6.5. I don't think CentOS uses systemd until 7. |
| 21:51.35 | Penguin | I've deleted the freepbx package(s) already. |
| 21:51.46 | malcolmd | if you don't want to use freepbx, i recommend installing something other than freepbx; i don't think i'd just remove amportal and try to call it a day. |
| 21:52.16 | malcolmd | but, as i'm not an expert on what amportal does, i'm not qualified :D |
| 21:52.21 | Penguin | I installed AsteriskNOW 3 and I _thought_ I selected the Asterisk 11 without FreePBX option. |
| 21:53.10 | Penguin | I've already begun configuring the system, so I don't really want to start the installer over again to be sure I select the no freepbx option... so I'm removing anything I see that is related to it. |
| 21:53.12 | malcolmd | ah..yeah, you may not want to do a new install of that. we're not shipping asterisknow 3 anymore. we moved on to letting the freepbx folk manage the distribution of asterisk now for us. |
| 21:53.24 | Penguin | I have no use for freepbx, so I'm getting it out of there.` |
| 21:53.54 | Penguin | Yeah, I found that out the other night when I got AsteriskNOW 5. I was a sad penguin. |
| 21:54.11 | Penguin | The FreePBX distro, disguised as AsteriskNOW, was a sorry surprise. |
| 21:54.16 | malcolmd | were i building a non-freepbx system, i'd start with the linux distribution with which i was most comfortable, be that a redhat-derived or a debian-derived, and i'd grab asterisk travels and do it that way |
| 21:54.30 | Penguin | It doesn't offer the no freepbx option like older AsteriskNOW had. |
| 21:54.43 | malcolmd | man...autocorrect is hammering me.... travels = source |
| 21:54.48 | malcolmd | that's correct |
| 21:55.09 | Penguin | I really liked the older one here I just select "asterisk only, no GUI" and be on my way. |
| 21:55.15 | Penguin | s/here/where/ |
| 21:55.38 | Penguin | It was trivial to upgrade that asterisk 1.4 to 1.8. |
| 21:57.07 | Penguin | I liked AsteriskNOW because it gave me the full OS ready to run with Asterisk already in it. |
| 21:57.27 | Penguin | Anyway, I'm stripping remnants of FreePBX from this one to make me happier. |
| 22:02.26 | Penguin | Oh, it's 6.6, not 6.5. |
| 22:02.29 | *** join/#asterisk danjenkins_ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 22:02.46 | Penguin | I think it was 6.5 before I did yum update and rebooted. :/ |
| 22:05.14 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 22:06.21 | *** join/#asterisk tuc0 (~tuc0@d192-24-85-250.nap.wideopenwest.com) |
| 22:07.01 | tuc0 | Hey guys, I have an extension that I'm trying to call and when I attempt it sip debug shows SIP/2.0 487 Request Terminated and the phone never rings. |
| 22:07.14 | tuc0 | it's just a local sip extension on the lan |
| 22:07.32 | tuc0 | Does anyone have any ideas for me to try? it's registered, I can dial out with it. |
| 22:07.51 | Penguin | Extensions aren't phones. What does the extension actually do? |
| 22:08.32 | tuc0 | sorry, it's not an extension, I have a SIP phone registered as 501. |
| 22:08.35 | tuc0 | I'm dialing SIP/501 |
| 22:08.51 | tuc0 | it claims it's ringing for a split second and then I get that reply. |
| 22:08.56 | Penguin | So the extension is successfully executing Dial(SIP/501)? |
| 22:08.57 | tuc0 | of: SIP/2.0 487 Request Terminated |
| 22:09.03 | tuc0 | no, I'm doing that with AMI |
| 22:09.16 | tuc0 | but it could, it can make outbound |
| 22:09.21 | Penguin | So we're not talking about extensions at all? |
| 22:10.03 | tuc0 | Well that depends on if the extension of s is the problem. |
| 22:10.22 | tuc0 | Also, maybe it's a problem with the dialplan held within *extensions*.conf |
| 22:10.29 | tuc0 | :) |
| 22:10.31 | Penguin | What is executing the Dial()? |
| 22:11.14 | tuc0 | I am running an AMI originate that connects to my originate context that takes the extension it passes and dials it. |
| 22:11.28 | tuc0 | with _X., dial(SIP/${exten}) |
| 22:12.07 | tuc0 | which works fine, and it dials SIP/501" |
| 22:12.29 | tuc0 | but then I get that error back from the sip device. I think it's the sip device that is. |
| 22:12.39 | Penguin | Okay, then I'd increase core verbose and tip on sip debug. |
| 22:12.45 | tuc0 | they are |
| 22:12.58 | tuc0 | that's how I learned about SIP/2.0 487 Request Terminated |
| 22:13.00 | Penguin | And the debug has no useful information? |
| 22:13.13 | tuc0 | I will pastebin it for you, 1 sec |
| 22:13.16 | Penguin | Does the phone have DND enabled? |
| 22:14.37 | tuc0 | http://pastebin.com/uEXThHzs |
| 22:14.56 | tuc0 | I don't believe so but I will look in the options. It is an ATA |
| 22:15.46 | tuc0 | Penguin, sending this in case you're waiting for me to ping ya |
| 22:17.47 | tuc0 | No it's not DND |
| 22:18.43 | tuc0 | I'm reading that a DND gives a 488 instead of a 487 |
| 22:19.13 | Penguin | oh |
| 22:19.50 | Penguin | Yeah, I guess it would give Busy Here rather than Request Terminated. |
| 22:20.35 | tuc0 | So I'm at a loss. I used to set this type of stuff up every day with asterisk like 10 year ago, this should be really really simple. I don't know what I'm missing. |
| 22:21.11 | tuc0 | would it tell me if there was a codec mismatch? |
| 22:21.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:21.44 | tuc0 | meh it's set to ulaw in the pap2 and in sip.conf. So I doubt it's that. |
| 22:22.09 | Penguin | If it ever gets that far, you'd see it. |
| 22:23.05 | tuc0 | Does anyone have any suggestions for me? I'm not really sure what else to try. |
| 22:23.26 | *** join/#asterisk voxter (~voxter@irc.voxter.net) |
| 22:24.45 | tuc0 | I swapped out the pap2 with a sip phone and I still get the SIP/2.0 487 Request Terminated |
| 22:29.10 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 22:29.16 | [TK]D-Fender | Stop focussing on just single messages in the overall conversation |
| 22:38.29 | *** join/#asterisk Vutral (jxy71QysoJ@mirbsd/special/Vutral) |
| 22:41.29 | Penguin | Is asterisk 11's chan_skinny good enough to replace chan-sccp-b? |
| 22:44.49 | Penguin | Oh, maybe that's not available. The asterisk-skinny.x86_64 package is actually asterisk-skinny-1.8.26.1-1.el6.x86_64 |
| 22:45.38 | Penguin | Or is it built in and I don't need an additional package? |
| 22:46.31 | Penguin | I guess it's included. I have /usr/lib64/asterisk/modules/chan_skinny.so already. |
| 22:46.34 | Penguin | DISREGARD |
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