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| 01:12.58 | Katty | so much quiet |
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| 01:43.48 | newtonr | Katty, sound! |
| 01:44.05 | Katty | ^_^ |
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| 01:50.41 | JaniceKitten | hi |
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| 02:51.33 | kemmler | I'm probably doing this wrong but I've got mysql cdr set up. I'm trying to toss some custom values in there and I see the command being ran in the asterisk log. I'm using this exten => h,n,set(CDR(username)=${username}) |
| 02:52.40 | kemmler | i'm passing ${username} to the dialplan externally, just need to figure out why it's not getting written |
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| 03:47.28 | hariom | Hi, I have shared my remote directory over ssh (sshfs) which contains prompts. But streamfile gives following output 200 result=-1 endpos=0 |
| 03:47.58 | hariom | What is the meaning of 200 result=-1 endpos=0 ? |
| 04:03.53 | mjordan | kemmler: what is the value of endbeforehexten in cdr.conf? |
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| 04:23.08 | kemmler | mjordan, setting it to no fixed the inability to pass the vars but broke billsec and duration |
| 04:45.06 | mjordan | which version are you using? |
| 04:45.22 | mjordan | btw, "broke billsec and duration" isn't specific enough for someone to understand what you're running into |
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| 04:56.31 | kemmler | I was retarded. I set the CDR stuff before the dial and didn't need to use the hangup exten |
| 04:56.38 | kemmler | so it's all working now |
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| 05:20.16 | hariom | I am getting streamfile result as: 200 result=-1 endpos=0 |
| 05:20.19 | hariom | What could be the reason? |
| 05:20.56 | hariom | It means failure but why is it failing? File is accessible on the path provided to steamfile. But that directory is mount from remote system |
| 05:36.54 | ChannelZ | Bad file? What codec? |
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| 05:37.26 | ChannelZ | And remote system.. NFS? |
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| 05:45.10 | hariom | ChannelZ: sshfs. File is .gsm |
| 05:45.34 | hariom | ChannelZ: same file I am able to play with streamfile if provided from another directory which is not mount. |
| 05:45.55 | hariom | ChannelZ: not sure why it is not able to read file from the mounted directory. I am see the files in the dir |
| 05:46.41 | ChannelZ | Barring anything dubious about sshfs (which I've never used) does your asterisk run as root or some other user? If the latter, can that user actually read the file? |
| 05:46.43 | hariom | ChannelZ: That directory is own by "nobody:myuser" and asterisk runs as another user "asterisk". But I have given 'r' permission to user group and others |
| 05:51.50 | hariom | ChannelZ: btw, if NFS is used, how do you secure it? I am on LAN but can't say it can not be misused. Its in cloud |
| 05:54.59 | hariom | ChannelZ: sshfs seems to have permission problem. Without mount, asterisk is able to read from that directory |
| 05:56.52 | ChannelZ | I just installed sshfs and tested playing a sound and it worked here |
| 05:59.26 | ChannelZ | And sorry I don't have any experience with NFS over WAN |
| 05:59.56 | hariom | ChannelZ: Was the user different? On local system I have "myuser" and "asterisk" as users. Asterisk is running as "asterisk" but the sshfs is mounted in the home dir of "myuser" which has 777 permision. Remote server has completely different users |
| 06:00.25 | hariom | "asterisk" won't be able to enter into directory after it is mounted by "myuser" |
| 06:03.30 | ChannelZ | Well should be readable by everyone. Does the user you're sshing to the remote system as have access though? |
| 06:04.11 | hariom | yea |
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| 06:29.54 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.1.0 (2014/12/15), 11.15.0 (2014/12/15), 1.8.32.1 (2014/11/20); Standard: 12.8.0 (2014/12/15); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 06:37.46 | kemmler | Can someone elaborate on the Realtime_Field function? I'm trying to do "Select minutes from users where name=$username" this is the code i'm using exten => 30,1,Set(mins=${REALTIME_FIELD(users,name,${username},minutes)}) |
| 06:38.17 | kemmler | i dont see any errors and mins = nothing in the asterisk cli log |
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| 06:56.04 | Penguin | You don't have to mount your sshfs target with your local user's login name. You can use it just like you'd use ssh with non-matching user names. |
| 06:56.52 | Penguin | sshfs otheruser@remotehost:somedir localmountpoint |
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| 07:17.04 | xochilpili | hi all |
| 07:17.32 | xochilpili | who does dial patterns work? |
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| 07:46.07 | wdoekes | *how do dial patterns work? -- https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
| 07:46.11 | wdoekes | weak, he left |
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| 07:52.50 | skrusty | morning |
| 07:53.42 | wdoekes | morning indeed |
| 08:04.28 | alexises | Hi |
| 08:05.39 | alexises | about my yesteday problem, My collegue had make some test yesteday, |
| 08:06.27 | alexises | We have a numeris line that use cpe_ptmp and not ptp, in France to make ptp we should have a numerise+ line that can make boths |
| 08:07.16 | alexises | my problems seens to come from a biavior from the local telecom provider that shut down bri line when not used to save energy according to the https://issues.asterisk.org/jira/browse/ASTERISK-13176 |
| 08:07.49 | alexises | thanks to the digium support that don't help us in this case by providing us wrong config ^^ |
| 08:08.35 | alexises | to deal with it, we should set the layer1_presence = ignore on chan_dahdi.conf |
| 08:08.42 | alexises | if it's could help someone ^^ |
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| 08:09.17 | alexises | we expect all the config is ok :) |
| 08:11.10 | wdoekes | not sure what you're saying. are you saying digium support didn't help you adequately? are you saying the sample config in the configs/ folder is wrong? |
| 08:13.23 | alexises | we have bought the analog card from digium that provide a free installation support |
| 08:20.30 | wdoekes | if digium support didn't help you adequately according to your expectations, it's best to write them a mail explaining so |
| 08:20.56 | wdoekes | the chance that your particular bri issue is picked up by someone with the same problem on irc, is about 0% |
| 08:21.17 | wdoekes | if you want it to help someone else, filing a bug report is better |
| 08:21.59 | wdoekes | perhaps a request for documentation enhancement |
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| 08:53.56 | WIMPy | Oh. Does that mean, dahdi still can't sensibly handle power saving? Like in the ooooold mISDN1 days? |
| 08:54.54 | WIMPy | alexises: What's the situation you got then? Incomming calls working and outgoing only if you were fast enough? |
| 08:55.06 | WIMPy | How did they fail, when they failed? |
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| 09:26.59 | linocisco | hi all |
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| 09:37.55 | alexises | WIMPy: it's work with the proper option ;) |
| 09:39.00 | alexises | it's why the option whan added ;) https://issues.asterisk.org/jira/browse/ASTERISK-13176 |
| 09:41.17 | linocisco | https://bugs.gentoo.org/show_bug.cgi?id=530056 is REAL? |
| 09:44.44 | WIMPy | alexises: Yes, but what's the characteristics of the issue when that option is required? |
| 09:46.25 | WIMPy | I gave my test box a nw kernel recently so I can't test DADHI at the moment. |
| 09:46.25 | alexises | It's related to french telecom provider, you should have a numeris installation (I'm not sure if it's also related to numeris+) |
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| 09:47.08 | WIMPy | It's pretty standard to have power saving on ptmp lines. |
| 09:47.53 | alexises | hum, it's also related to another contry |
| 09:47.56 | WIMPy | I'd like to add this to the usual issues section, but I need the if.. part. |
| 09:48.13 | alexises | to detect it, you have warning about D channel not available |
| 09:48.20 | WIMPy | It's definitely the same in Germany. And probably elsewhere as well. |
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| 09:48.27 | alexises | and when you pass a call all line are busy |
| 09:48.38 | alexises | okey |
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| 09:49.45 | alexises | I think the lack of information on the internet is due to telecom provider that won't to comunicate them to purpose some additional maintenance services provided by her proffesionnal division |
| 09:51.18 | WIMPy | Before Asterisk you would have trouble to find equipment that cared. |
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| 10:23.58 | q_a_z_steve | What's the easiest way to test necessary functions on say a RasPBX before going out and putting an ATA on the production phone? |
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| 11:48.48 | necronian | I know I keep harping on the same question in here, but I still can't get it working the way I want. I have 1 ata, so only one line in and out of asterisk. After someone leaves a voicemail I want asterisk to pick up the line and senddtmf to the analog phone system to notify the user there is a message. |
| 11:50.08 | necronian | Yesterday I was told that originate can help me, but it doesn't work because the channel I'm in and the channel I want to originate on are the same... so it's busy |
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| 11:52.23 | necronian | I have a stupid solution using system to execute a shell script I wrote on hangup, the script waits 5 seconds and then puts a call file into the outgoing queue |
| 11:52.48 | necronian | sometimes it works, sometimes the sip channel is still busy so nothing happens |
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| 16:02.17 | BarthezZ | Hi Guys, I'm having a weird problem on Asterisk 1.8.latest.. A call is delivered on a peer with sendrpi=no and trustrpid=no, globally send- and trustrpid's PAI/yes... When a call comes in with the PAI header, and goes out to a trunk... the "original" PAI-header is still there |
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| 16:20.50 | Jacoby6000 | is there an asterisk dialplan pattern for something like exten=>_([ub])|manage,1,app(appdata)? |
| 16:21.05 | Jacoby6000 | where the string will match on u, b, or manage? |
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| 16:29.09 | [TK]D-Fender | Jacoby6000: No, you'll require multiple extens |
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| 17:03.21 | vomit | Hello. Some of my phone system users are experiencing issues when calling out some Cell phone numbers. Instead of voicemail they get a fast busy signal and this is happening intermittently. |
| 17:03.25 | vomit | I have performed a troubleshooting, didn't find any issues, call is passed to the service provider. Provider's records show that the call was passed to their other provider and call ended with a NOANSWER state, which is correct as the caller got a fast busy signal and no voicemail. Re trying usualy allows to reach voicemail of the other party, the call is marked as ANSWERED then. We observerd this behavior with two providers: Voip.ms and Twilio. |
| 17:03.31 | vomit | Did anybody experienced anything similiar or have any clues what may be causing this? |
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| 17:37.44 | Penguin | jacoby6000: While you can't do the | for OR, the [ub] part does work for u or b. So you'll need two extensions for that configuration. |
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| 17:47.04 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 17:50.45 | *** join/#asterisk imox (~imox@p54843281.dip0.t-ipconnect.de) |
| 18:03.14 | Penguin | Can I not use $[${ISNULL(varA)} & ${ISNULL(varB)}] to check that both variables must be null for the expression to be true? |
| 18:04.46 | Penguin | And I actually mean $[${ISNULL(${varA})} & ${ISNULL(${varB})}] |
| 18:04.47 | [TK]D-Fender | && IIRC |
| 18:06.10 | Penguin | *shrug* I use a single | for an OR in the expression, so I would have expected a single & for an AND. |
| 18:06.45 | Penguin | Oh... but I wasn't using ISNULL() before. |
| 18:06.54 | Penguin | I was comparing a string to a null string. |
| 18:07.00 | Penguin | Those could very well be completely different. |
| 18:07.09 | [TK]D-Fender | Didn't match your sample here? |
| 18:07.11 | [TK]D-Fender | That would be silly |
| 18:07.17 | [TK]D-Fender | And it does look like & not && |
| 18:08.52 | Penguin | I've changed the ISNULL()s to string comparisons to see if that makes the difference. |
| 18:10.21 | Penguin | I guess I don't really care if I have to use string comparisons for this rare occasion. I've mostly given it up for ISNULL() though. |
| 18:12.08 | Penguin | Now I'll just wait for the next call that hits that extension and we'll see what it does. |
| 18:18.19 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 18:19.43 | *** join/#asterisk ryan_turner (~ryan_turn@eth0.irc-bouncer.ret.memhamwan.net) |
| 18:21.03 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 18:21.03 | *** mode/#asterisk [+o file] by ChanServ |
| 18:21.26 | ryan_turner | Hey, trying to troubleshoot an issue connecting two asterisk servers via IAX; right now, if there are multiple entries in iax.conf on one of the servers, that server rejects calls |
| 18:21.38 | ryan_turner | When we remove the other entries, so that single IAX trunk is the only one in the file, it works fine. |
| 18:21.53 | ryan_turner | In both situations, the one server can call the other fine |
| 18:22.04 | Penguin | Show us the file with the multiple entries. |
| 18:22.33 | *** join/#asterisk d1gital (~d1gital@fsf/member/d1gital) |
| 18:22.47 | ryan_turner | Penguin, here you go! https://gist.github.com/ryanturner/1653641a1e1092509ed7 |
| 18:23.11 | Penguin | There I went! |
| 18:23.19 | ryan_turner | Troubleshooting between this server and w6kwf; we can call w6kwf fine. When we comment out hamwanpsdr and allstar, they can call us. |
| 18:23.25 | [TK]D-Fender | ryan_turner: Type=friend = FAIL |
| 18:23.43 | ryan_turner | ? |
| 18:23.43 | [TK]D-Fender | friend can auth on username alone |
| 18:23.56 | [TK]D-Fender | if ou have IP's .. stick with that mand make all type=peer |
| 18:23.59 | [TK]D-Fender | make* |
| 18:24.23 | ryan_turner | Ok, so both ends need to be changed to type=peer? Would that actually potentially resolve my problem, or just solve a security risk? |
| 18:25.49 | Penguin | user= ? |
| 18:25.59 | Penguin | user isn't a valid setting to my knowledge. |
| 18:26.30 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 18:26.50 | *** join/#asterisk MarcoZink (~marcozink@189.249.148.191) |
| 18:27.01 | ryan_turner | Ah, so user needs to be changed to username |
| 18:28.01 | Penguin | I also notice that you are allowing a codec on a couple of those peers, but you haven't disallowed anything prior to that. On the other peer, you aren't disallowing or allowing any. |
| 18:28.21 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 18:29.30 | ryan_turner | Ok, so I just updated the gist with the username and allow changes |
| 18:30.03 | Penguin | I don't see that the disallow has been added anywhere. |
| 18:30.34 | ryan_turner | Ok, do I need to put disallow somewhere? |
| 18:30.58 | Penguin | In order to utilize only a single codec, you must first disallow all codecs and then allow only the codec you want. |
| 18:31.16 | Penguin | The disallow can be done in general or in each peer. |
| 18:31.59 | ryan_turner | Ok, updated again to include that -- https://gist.github.com/ryanturner/1653641a1e1092509ed7 |
| 18:32.22 | ryan_turner | Still encountering the same odd issue -- I can call them, they cant call me until I comment out hamwanpsdr and allstar |
| 18:33.05 | Penguin | Now that the config has been improved, let's take a look at some call detail to see what is happening. |
| 18:33.30 | ryan_turner | What would you like to see? When they attempt to call, CLI doesnt show anything with verbosity at 5 |
| 18:33.42 | ryan_turner | the only troubleshooting so far thats given us hints have been packet dumps from their end |
| 18:33.53 | Penguin | Can you turn on iax debug? |
| 18:34.15 | Penguin | It's been quite some time since I used IAX, so I don't remember the exact commands to do it. |
| 18:34.41 | Penguin | Probably iax2 set debug on, I'd guess. |
| 18:35.00 | ryan_turner | Enabled, Ill have him call. |
| 18:36.17 | ryan_turner | https://gist.github.com/ryanturner/1653641a1e1092509ed7 |
| 18:36.20 | ryan_turner | its in the comment |
| 18:36.35 | ryan_turner | oh damnit |
| 18:37.00 | ryan_turner | yay password... |
| 18:37.32 | ryan_turner | So, its trying to user username hamwanpsdr instead of w6kwf :/ |
| 18:38.31 | *** join/#asterisk nigelvh (~nigel@c-50-132-67-209.hsd1.wa.comcast.net) |
| 18:38.56 | *** join/#asterisk PhirePhly (~PhirePhly@99-46-142-3.lightspeed.sntcca.sbcglobal.net) |
| 18:40.52 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 18:41.16 | Penguin | So the [memphis] peer entry on the other side has a username of w6kwf? |
| 18:41.55 | ryan_turner | PhirePhly can confirm, probably would be easiest for him to share his conf |
| 18:42.29 | PhirePhly | https://gist.github.com/PhirePhly/b98e2f30e44b191f5618 |
| 18:42.34 | Penguin | Is that Ken? |
| 18:42.42 | ryan_turner | Hah! |
| 18:42.45 | PhirePhly | Kenneth |
| 18:42.51 | Penguin | Sorry, Kenneth. |
| 18:43.56 | Penguin | It does have that username specified. |
| 18:44.17 | Penguin | but the comment for it is misleading. |
| 18:47.47 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:48.14 | PhirePhly | ok, so getting everyone correctly set to username fixed it |
| 18:49.13 | *** join/#asterisk moke (~moke@unaffiliated/moke) |
| 18:49.35 | PhirePhly | gosh it would be nice if Asterisk had a conf parser |
| 18:50.03 | Penguin | What do you expect it to tell you? |
| 18:50.14 | PhirePhly | "user" unknown parameter |
| 18:50.34 | PhirePhly | Penguin: but seriously, thank you. Been driving us up the wall |
| 18:50.34 | Penguin | I would have thought it would tell you that when you load the iax module. |
| 18:50.39 | PhirePhly | nope |
| 18:51.07 | PhirePhly | <PROTECTED> |
| 18:51.08 | PhirePhly | [Jan 23 10:50:58] NOTICE[18100]: iax2-provision.c:558 iax_provision_reload: No IAX provisioning configuration found, IAX provisioning disabled. |
| 18:51.15 | PhirePhly | thats it |
| 18:51.54 | Penguin | Was that for a reload or a cold load after having unloaded (such as would be with a fresh restart of asterisk)? |
| 18:51.58 | PhirePhly | reload |
| 18:52.18 | PhirePhly | iax2 reload |
| 18:52.48 | Penguin | I quit using IAX2 a few years ago, so I've forgotten a few of the things it does or doesn't do. |
| 18:52.57 | ryan_turner | Penguin, how have you replaced it? |
| 18:53.01 | Penguin | SIP |
| 18:53.01 | PhirePhly | ^ |
| 18:53.04 | PhirePhly | ah |
| 18:53.10 | PhirePhly | :-/ |
| 18:53.19 | ryan_turner | Dont some view IAX2 as superior to SIP for trunking? |
| 18:53.26 | ryan_turner | (I am a noob) |
| 18:53.37 | nigelvh | IAX2 is significantly better in terms of firewall friendliness |
| 18:53.38 | Penguin | Considering SIP does not trunk, I would imagine they do. |
| 18:53.45 | ryan_turner | Penguin, :) |
| 18:53.52 | PhirePhly | well that's one thing to consider for our system... we won't have any firewalls. |
| 18:53.56 | [TK]D-Fender | [13:53]ryan_turnerDont some view IAX2 as superior to SIP for trunking? <- I see dead people.... |
| 18:54.15 | ryan_turner | [TK]D-Fender, I'm such a noob |
| 18:55.57 | PhirePhly | so IAX is firewall friendly, SIP exists, is there anything else we should consider for PBX to PBX interchange? |
| 18:56.10 | Penguin | I wouldn't. |
| 18:56.30 | PhirePhly | and when firewalls arent an issue, youd go with SIP? |
| 18:56.34 | Penguin | If you can make IAX2 work, use it. |
| 18:56.36 | Penguin | If you can't, use SIP. |
| 18:56.49 | PhirePhly | copy |
| 18:56.50 | Penguin | I don't have problems with SIP and firewalls. |
| 18:57.08 | ryan_turner | Penguin, thanks much for your help. |
| 18:59.01 | *** join/#asterisk MarcoZink (~marcozink@189.249.148.191) |
| 19:00.08 | *** join/#asterisk chatran (~chatran_@179.183.199.121) |
| 19:00.35 | chatran | hi, can someone give me an idea on how to make a call if the serveer receive an e-mail? |
| 19:00.56 | chatran | i need make a call if asterisk server receive e-mail from my nobreak |
| 19:01.50 | [TK]D-Fender | chatran: To make Asterisk call out look up "AMI Originate" and "call files" |
| 19:01.54 | rrittgarn | getting the email to execute a script would be step 1. Once you have that you could just write a script that creates a call file |
| 19:02.09 | rrittgarn | Fender wins for speed and accuracy |
| 19:02.13 | [TK]D-Fender | ChaNow getting it to do that based on an e-mail condition is scripting YOU will have to do. This is not *'s job |
| 19:02.24 | [TK]D-Fender | chatran: Now getting it to do that based on an e-mail condition is scripting YOU will have to do. This is not *'s job |
| 19:04.49 | JaniceKitten | [TK]D-Fender, you see living people :^) |
| 19:05.13 | [TK]D-Fender | JaniceKitten: Highly transient |
| 19:05.19 | JaniceKitten | views IAX2 as superior for the normal user |
| 19:06.57 | JaniceKitten | because the normal user is usually behind NAT upon NAT upon NAT nowadays |
| 19:07.17 | [TK]D-Fender | Hardly the norm. |
| 19:07.34 | [TK]D-Fender | A few idiot doing tests behind triple NAT thinking they have a clue... sure. |
| 19:07.36 | PhirePhly | is one of those users behind NAT-NAT |
| 19:07.56 | JaniceKitten | I'm behind one NAT, and I suspect many people East of the Atlantic are behind 2 |
| 19:08.02 | [TK]D-Fender | Also a few tragic victims of circumstance... |
| 19:08.06 | JaniceKitten | [TK]D-Fender, That wasn't to be taken literally... |
| 19:09.39 | PhirePhly | I'm hoping our final deployment doesnt involve NAT, but that depends on getting everyone set up with OSPF/BGP |
| 19:10.40 | PhirePhly | soooo yeah, I need to plan on dealing with NAT |
| 19:10.47 | Penguin | My asterisk is behind a NAT, my clients are behind NATs... I use SIP. |
| 19:11.21 | gusto | lol |
| 19:11.37 | gusto | Penguin, I ll break your mirrors for it |
| 19:12.24 | JaniceKitten | My asterisk is behind one 1:1 NAT, I'm behind another, and my friend is behind a third NAT. All of these NATs are one-level. I use IAX2 in the c2s direction. |
| 19:14.48 | chatran | [TK]D-Fender yes my job write the script i just want know if this can be made, and a way to do. thanks a lot ! |
| 19:17.05 | ChannelZ-Wk | Hi-larious! business.comcast.com is down |
| 19:25.35 | Penguin | I guess they host that site themselves. |
| 19:26.25 | *** part/#asterisk PhirePhly (~PhirePhly@99-46-142-3.lightspeed.sntcca.sbcglobal.net) |
| 19:27.35 | ChannelZ-Wk | Yeah. And it's even more ironical since I'm trying to hit it using their service. |
| 19:28.03 | ChannelZ-Wk | I think someone let some air pressure out of the inter-tubes. |
| 19:28.14 | ChannelZ-Wk | (topical NFL rimshot) |
| 19:31.13 | mbowie | Looks like it's back now. |
| 19:31.19 | mbowie | After a reboot. |
| 19:32.03 | ChannelZ-Wk | whowhat? |
| 19:32.35 | *** join/#asterisk s7r (~s7r@openvpn/user/s7r) |
| 19:38.05 | mbowie | Exactly, |
| 19:41.34 | *** join/#asterisk slackology_ (~slackolog@unaffiliated/quintux) |
| 19:45.27 | *** join/#asterisk CeBe (~CeBe@port-92-200-2-36.dynamic.qsc.de) |
| 19:58.08 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 20:01.12 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 20:05.31 | *** join/#asterisk frek818 (~frek818@172.56.14.90) |
| 20:07.51 | *** part/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 20:16.14 | *** join/#asterisk JeffC_NN (32cad19e@gateway/web/freenode/ip.50.202.209.158) |
| 20:17.14 | JeffC_NN | If I wanted to store a call's jitter/dropped packets/etc to my db using an AGI, how could I access this? I'm thinking I could modify my AGI that runs on hangup, but I don't know if the stats are still set on the channel by then, or how I'd get them... |
| 20:26.15 | JeffC_NN | Looks like ${RTPAUDIOQOS} might be a clue... |
| 20:29.29 | JeffC_NN | this blog looks like it has the answer if you're interested: http://qos.wawit.pl/2010/06/asterisk-rtpaudioqos/ |
| 20:47.17 | *** join/#asterisk frek818 (~frek818@172.56.14.90) |
| 20:48.57 | jeffspeff | i'm looking to setup something that will not only foward a voicemail to a users e-mail, but also provide a transcript of the voicemail as well. any suggestions? |
| 20:51.48 | JeffC_NN | None that I know about. Might be worth checking this page out: http://en.wikipedia.org/wiki/Speech_recognition_software_for_Linux |
| 20:55.38 | *** join/#asterisk jhlavacek (~jirka@84.19.95.180) |
| 20:57.53 | *** join/#asterisk jiuweigui (~jiuweigui@unaffiliated/jiuweigui) |
| 20:59.24 | JeffC_NN | @jeffspeff Here's a good link I just found http://cmusphinx.sourceforge.net/wiki/asteriskdetails |
| 21:00.20 | JeffC_NN | (though that may be for real-time conversion) |
| 21:00.44 | jeffspeff | thanks, i'll check it out. adding real-time conversion would make my boss happy too. |
| 21:01.08 | jeffspeff | it's almost the end of our fiscal year, which means review time. the happier the boss is, the happier my bank account is. |
| 21:28.26 | *** join/#asterisk jiuweigui (~jiuweigui@unaffiliated/jiuweigui) |
| 21:33.56 | *** part/#asterisk q_a_z_steve (~q_a_z_ste@unaffiliated/q-a-z-steve/x-0522206) |
| 21:47.56 | *** join/#asterisk CeBe (~CeBe@92.200.2.36) |
| 21:53.32 | JaniceKitten | Anyone here I can talk to? |
| 21:54.14 | Synthase_ | All 204 of us. |
| 21:54.34 | Qwell | ~ask |
| 21:54.34 | 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. |
| 21:54.39 | WIMPy | Simultameousely! |
| 21:54.59 | JaniceKitten | I'm not asking to ask. I'm asking to have someone call my conference service so I have someone to talk to. |
| 21:55.04 | Qwell | WIMPy: we prefer they go down the channel list, asking each person individually. That way, file gets to answer everything. |
| 21:55.22 | file | harsh |
| 21:55.25 | WIMPy | Qwell: Handy |
| 21:55.39 | JaniceKitten | file, are you able to make SIP calls at this time? |
| 21:55.45 | Qwell | WIMPy: see? |
| 21:56.01 | WIMPy | Jepp |
| 21:56.20 | JaniceKitten | also... i used to be known as j4jackj... |
| 21:57.43 | JaniceKitten | Qwell, hi. |
| 21:58.08 | drmessano | OMG A WOMAN |
| 21:58.13 | drmessano | Sorry... |
| 21:58.17 | JaniceKitten | stares at drmessano |
| 21:58.30 | file | I could originate a call to echo, does that count? |
| 21:58.35 | JaniceKitten | file, oh pls |
| 21:58.47 | JaniceKitten | IRL people still see me as a male... :/ |
| 21:59.04 | JaniceKitten | shouldn't be dragging transition drama here :/ |
| 21:59.16 | Qwell | s/transition // |
| 21:59.22 | JaniceKitten | Qwell, good point |
| 21:59.33 | JaniceKitten | Qwell, are you able to make sip calls at this time? |
| 21:59.34 | drmessano | s/Qwell// |
| 21:59.43 | JaniceKitten | drmessano, pls |
| 21:59.53 | mjordan | tt-monkey all the things |
| 21:59.59 | JaniceKitten | mjordan, wat |
| 22:00.16 | mjordan | what is the SIP URI? I'm curious if this thing I have works. |
| 22:00.17 | JaniceKitten | if anyone wants to call me, crank or otherwise, i'm janicez@sip.umbrellix.tk |
| 22:01.59 | drmessano | Under my Umbrellix-ellix-ellix, eh, eh |
| 22:01.59 | JaniceKitten | drmessano, wat |
| 22:01.59 | *** part/#asterisk JaniceKitten (janice@need.sleep.caffeinet.uk.to) |
| 22:01.59 | *** join/#asterisk JaniceKitten (janice@need.sleep.caffeinet.uk.to) |
| 22:01.59 | JaniceKitten | and unpoof |
| 22:01.59 | drmessano | You missed all the fun |
| 22:02.17 | JaniceKitten | I was that weirdo who used to run a PBX from her home connection. :p |
| 22:02.31 | JaniceKitten | Oh how fucked up I was then, and still am now |
| 22:02.59 | Qwell | yeah, none of us would ever run a PBX at home... that would be totally weird... |
| 22:03.03 | drmessano | lol |
| 22:03.18 | drmessano | or put Asterisk on a Pi and have a PBX in our car |
| 22:03.23 | drmessano | O.o |
| 22:03.36 | JaniceKitten | can someone call, crank or otherwise, my SIP URI that I pasted before I parted |
| 22:03.41 | Qwell | drmessano: sadly, I could name at least 2 other people that have done that. |
| 22:04.15 | drmessano | Qwell, but did they transfer calls from the front seat to back seat? |
| 22:04.42 | Qwell | drmessano: worse, it was probably the passenger that they transferred. |
| 22:04.44 | drmessano | I also did that all-page |
| 22:04.48 | JaniceKitten | Qwell, wat |
| 22:05.04 | *** join/#asterisk funtriaco (~funtriaco@mail.brickellmotors.com) |
| 22:05.17 | drmessano | s/wat/what/ |
| 22:05.29 | drmessano | s/ICQ/IRC/ |
| 22:06.30 | funtriaco | Any suggestion on how to retain the Original Caller ID when transfering a call? Caller ID is 786-222-3333 and call is answer by extension 100, then its transfer to extension 101... How can i get to the original caller id? |
| 22:06.47 | drmessano | writes down funtriaco's phone number |
| 22:07.00 | funtriaco | after the transfer the caller id becomes 100 |
| 22:07.07 | funtriaco | drmessano: you can call anytime :) |
| 22:07.17 | JaniceKitten | pokes drmessano |
| 22:07.40 | JaniceKitten | drmessano, what are your preferred pronouns? |
| 22:08.17 | drmessano | "What" makes my eyes bleed less than "wat" and "wut", that is all |
| 22:09.05 | mjordan | funtriaco: what version of Asterisk are you using? |
| 22:09.11 | drmessano | But wateva |
| 22:09.12 | funtriaco | reads his question looking for 'wat' and 'wut' |
| 22:09.42 | funtriaco | mjordan: Asterisk 1.8.18.0 |
| 22:09.58 | drmessano | Ooooh Retro |
| 22:10.13 | funtriaco | mjordan: im so close from finishing my popup project.. lol and this is killing me as it requires the oringinal caller id to show the pop of the caller |
| 22:10.18 | mjordan | you most likely need to use connected line. |
| 22:10.29 | funtriaco | connected line? |
| 22:10.34 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information |
| 22:10.39 | JaniceKitten | drmessano, don't make jokes based on my choice of domain name. It's a very old choice I hardly remember, and I'm thinking of phasing it out some day. |
| 22:11.15 | mjordan | Either that, or just cache off the CALLERID in the AstDB and restore it when it is transferred |
| 22:12.17 | drmessano | JaniceKitten, Syntax Error in process 'drmessano' error code 0x1000069 DOES_WHAT_IT_WANTS |
| 22:12.21 | funtriaco | mjordan: hmmm |
| 22:12.25 | JaniceKitten | drmessano, so? |
| 22:12.40 | JaniceKitten | drmessano, I don't want people to make jokes... :/ |
| 22:12.52 | JaniceKitten | drmessano, can you call me on that thing I pasted that you joked on? |
| 22:13.06 | drmessano | I'm unable to at this time |
| 22:13.08 | JaniceKitten | ok |
| 22:13.20 | WIMPy | funtriaco: Maybeit wasn't like that in 1.8, but at least in 11+ it happens automagically. At least in chan_sip. |
| 22:13.25 | JaniceKitten | WIMPy, hai |
| 22:16.40 | JaniceKitten | can someone call janicez@sip.umbrellix.tk ? I'm desperately in need of someone to talk to. |
| 22:18.08 | JaniceKitten | waits for the flood of crank calls |
| 22:18.34 | JeffC_NN | Call from '1003' to extension 'janicez' rejected because extension not found in context 'DLPN_Default'. |
| 22:18.38 | JeffC_NN | Heh, apparently not |
| 22:18.43 | JaniceKitten | JeffC_NN, umm |
| 22:18.57 | Qwell | JeffC_NN: ...that would be your end |
| 22:19.07 | JeffC_NN | yep :D |
| 22:19.22 | JaniceKitten | JeffC_NN, so like, can you create an extension "janicez" that points to Dial(SIP/janicez@sip.umbrellix.tk) |
| 22:19.45 | JaniceKitten | (or something, I'm still not familiar with Asterisk having only used it a few years back) |
| 22:20.17 | JeffC_NN | exten = janicez,1,Dial(SIP/janicez@sip.umbrellix.tk) |
| 22:20.41 | JaniceKitten | um |
| 22:21.15 | *** join/#asterisk frek818 (~frek818@172.56.14.90) |
| 22:21.22 | JaniceKitten | JeffC_NN, try again, but don't make two calls at once |
| 22:21.25 | JaniceKitten | I use pcmu |
| 22:21.37 | JeffC_NN | my office has a crazy firewall. we don't usually do sip externally |
| 22:21.44 | *** join/#asterisk sgriepentrog (~sgriepent@2602:30a:2ef2:840:20c:29ff:fe4f:cac8) |
| 22:21.57 | JeffC_NN | I take it you could hear me then. :) |
| 22:22.00 | JaniceKitten | JeffC_NN, yep |
| 22:22.19 | JaniceKitten | JeffC_NN, do you have your own PBX you can setup IAX using? |
| 22:22.29 | JeffC_NN | if the fw would let me out, yeah, lol |
| 22:22.32 | JaniceKitten | I apologise for my desperate micwhoring |
| 22:22.45 | mjordan | connected line should be in Asterisk 1.8. |
| 22:24.44 | WIMPy | mjordan, funtriaco: Yes, I guess it was like that in 1.8 already. Probably just the used phone not supporting it. |
| 22:28.08 | *** join/#asterisk u0m3_ (~u0m3@109.96.139.134) |
| 22:28.11 | funtriaco | WIMPy: i'm trying to pull the Original CID in the extension file |
| 22:28.45 | funtriaco | Im trying to run a script before dealing every extension and the script requires the original caller id |
| 22:29.01 | funtriaco | The script is a pop up. |
| 22:29.12 | funtriaco | that shows the callers information. |
| 22:29.30 | funtriaco | mjordan: unless im looking at the wrong thing, i'm not having much luck with connected line |
| 22:29.49 | funtriaco | btw, this is a FreePBX system.. if that matters |
| 22:30.17 | file | is it an attended transfer? |
| 22:30.42 | funtriaco | i just tried a blind transfer and didnt work either. |
| 22:31.07 | funtriaco | phones are Cisco SPA504G not sure if that matters as well. |
| 22:34.03 | JeffC_NN | @funtriaco have you tried ${BRIDGEPEER} |
| 22:34.09 | JaniceKitten | JeffC_NN, hi |
| 22:34.31 | JeffC_NN | JaniceKitten, hello |
| 22:35.06 | JaniceKitten | JeffC_NN, how crazy is your office's firewall? Does it allow UDP 4569 through? |
| 22:35.18 | funtriaco | JeffC_NN: no... let me try that. |
| 22:36.21 | JeffC_NN | it may not be exactly what you want. If your sip peer is ABC, it will be BRIDGEPEER=SIP/ABC-00002202 |
| 22:36.54 | funtriaco | JeffC_NN: it was empty |
| 22:37.06 | *** join/#asterisk pouledodue (~textual@modemcable082.140-131-66.mc.videotron.ca) |
| 22:37.10 | JeffC_NN | oh. I'm on a different version. Sorry |
| 22:37.19 | funtriaco | so far what i'm trying |
| 22:37.20 | funtriaco | exten => 297,1,NoOp("BRIDGEPEER: ". ${BRIDGEPEER} ."" ${CONNECTEDLINE(num)} ."". ${CALLERID(dnid)} ."". ${CALLERID(num)}) |
| 22:38.06 | JeffC_NN | Looks like BRIDGEPEER was introduced in Asterisk 12 |
| 22:39.30 | JeffC_NN | I just did a "core show channel xxx" after the transfer and looked at what vars were available |
| 22:39.36 | lvlinux | no I'm pretty sure BRIDGEPEER has been around for a good while now |
| 22:40.02 | lvlinux | at least since 11 |
| 22:41.07 | lvlinux | actually I believe i used it on v1.8 |
| 22:41.58 | JeffC_NN | Oh, I was basing my statement on this, https://wiki.asterisk.org/wiki/display/AST/New+in+12 which shows new behavior of BRIDGEPEER, but doesn't explicitly call it new. |
| 22:44.06 | lvlinux | yes they made it be a comma separated list with multi-party stuff---used to it was "the other end of the line" :-) |
| 22:45.54 | JaniceKitten | JeffC_NN, does your Asterisk PBX have guest access? |
| 22:48.12 | *** join/#asterisk jhlavacek (~jirka@84.19.95.180) |
| 22:51.01 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:51.12 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 22:58.43 | JeffC_NN | JaniceKitten, nope. We're a call center with not much of a "fun side" |
| 22:58.58 | *** join/#asterisk s7r (~s7r@openvpn/user/s7r) |
| 23:00.15 | JaniceKitten | JeffC_NN, try changing that janicez extension to Dial(IAX2/guest@sip.umbrellix.tk/janicez) |
| 23:01.46 | JeffC_NN | ah, we have IAX2 disabled ever since we reported https://issues.asterisk.org/jira/browse/ASTERISK-24600 (which was fixed a few days ago) |
| 23:01.59 | phix | People still use IAX2? |
| 23:02.06 | phix | I thought it was all SIP now |
| 23:02.18 | JeffC_NN | IAX2 is fantastic for firewalls |
| 23:02.41 | phix | so I have heard, but none of the VoIP providers I use support it |
| 23:02.47 | phix | It is either SIP or skinny |
| 23:02.49 | JaniceKitten | phix, I use IAX2... :P |
| 23:03.15 | phix | JaniceKitten: Can I connect to you via IAX2? :) |
| 23:03.25 | JaniceKitten | phix, if you want to contact me, make an extension whose application is Dial(IAX2/guest@sip.umbrellix.tk/janicez) |
| 23:03.29 | JaniceKitten | phix, why? |
| 23:04.10 | phix | Well making a connect to you would be my only use for IAX2 |
| 23:04.18 | phix | Nowhere else I know of uses it |
| 23:04.52 | JaniceKitten | phix, call me via IAX2/guest@sip.umbrellix.tk/janicez and we'll talk. |
| 23:05.06 | phix | tk, you from NZ? |
| 23:05.08 | JeffC_NN | (so he can get free trunking in Tokelau, hehe) |
| 23:05.14 | phix | :) |
| 23:05.14 | JaniceKitten | JeffC_NN, lol |
| 23:05.27 | JaniceKitten | phix, no... I'm in Canada. .tk is one of those freedomain stupidities |
| 23:05.33 | phix | ah |
| 23:05.36 | phix | like .nu |
| 23:05.38 | JaniceKitten | sorta. |
| 23:05.40 | JaniceKitten | maybe. |
| 23:05.51 | phix | which is also owned by NZ |
| 23:06.02 | phix | well, not really owned but you know what I mean :) |
| 23:07.40 | *** join/#asterisk technoid_ (~Technoid@74.84.14.161) |
| 23:08.56 | *** join/#asterisk [44] (~slackolog@unaffiliated/quintux) |
| 23:09.35 | *** join/#asterisk CeBe (~CeBe@port-92-200-2-36.dynamic.qsc.de) |
| 23:11.15 | *** join/#asterisk jiuweigui (~jiuweigui@unaffiliated/jiuweigui) |
| 23:12.55 | technoid_ | Would I be able to run a single voip phone (2 channel) asterisk on a VPs with 256mb? |
| 23:14.03 | JaniceKitten | wut |
| 23:15.20 | *** join/#asterisk frek818 (~frek818@172.56.14.90) |
| 23:15.29 | technoid_ | I would like to setup asterisk for a single single voip phone starting out, but was wondering what the minumum requires would be |
| 23:23.39 | newtonr | technoid_, https://wiki.asterisk.org/wiki/display/AST/Hello+World |
| 23:24.06 | newtonr | oh you are asking about the system hardware requirements? |
| 23:24.28 | newtonr | to run a single call you don't need much of anything.. whatever you are trying to run it on will probably run it |
| 23:25.14 | [TK]D-Fender | technoid_, My watch can handle that ... and it's analog... |
| 23:25.26 | technoid_ | newtonr: Thanks. I have ran asterisk before (years ago though) and just trying to figure out the size of the VPS I need to host it. |
| 23:26.07 | technoid_ | [TK]D-Fender: must be an expensive watch |
| 23:26.49 | technoid_ | Instead of encrusted diamonds yours have network ports? |
| 23:55.42 | *** join/#asterisk blee (~blee@72-189-166-247.res.bhn.net) |
| 23:57.40 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 23:57.43 | *** join/#asterisk Draecos (~Draecos@106-68-101-185.dyn.iinet.net.au) |