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| 07:55.40 | ghghz | Hey, how can I use asterisk manager with asynchronous originating the calls? |
| 07:56.17 | ghghz | I'm originating calls from API using threads to originate 2-3 parallel calls |
| 07:56.27 | ghghz | but only 1 call is made |
| 07:57.56 | ghghz | I see that Manager 'admin' logged on from X.X.X.X |
| 07:58.03 | ghghz | but no originated calls only 1 |
| 07:58.37 | kaldemar | are you sure your application sends more originates? |
| 08:00.40 | ghghz | yes, because I see two 'logged on from X.X.X.X' |
| 08:01.58 | kaldemar | that's a sign of two logins. not two originate actions sent to AMI. |
| 08:02.18 | ghghz | you right :) |
| 08:02.29 | ghghz | wait, i will test to print is connection is made |
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| 08:14.23 | ghghz | kaldemar: yes, that's my fault.. |
| 08:14.25 | ghghz | :) |
| 08:14.26 | ghghz | thank you |
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| 08:32.37 | phix | hai gang |
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| 08:49.07 | arsperger | Hi does anybody know about h323 in asterisk, is it possible to register to multiple gatekeepers? |
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| 11:08.53 | davlefouAMD | hi, i try to use ilbc, i have compile it with le same version but i said me it not the same time code. I have use ubuntu with debian repositorie for install asterisk. |
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| 11:15.22 | basilic | bonjour a tous |
| 11:15.24 | basilic | hell all |
| 11:15.35 | Greenlight | Hell(o) |
| 11:15.45 | basilic | I have some question about hardware need to work this asterisk |
| 11:16.00 | basilic | I would like connect a normal phone to the VOIP |
| 11:16.18 | Greenlight | Usually you'd use an ATA |
| 11:16.50 | basilic | ATA? |
| 11:16.57 | kaldemar | ~ata |
| 11:16.58 | infobot | hmm... ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 11:17.15 | Greenlight | Ty infobot ;) |
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| 11:18.30 | basilic | ATA is a box to connect normal phone to network? we can't use directly an interface for asterisk on the asterisk server? |
| 11:18.48 | Greenlight | basilic: Yes, you could use a card |
| 11:19.15 | Greenlight | http://www.digium.com/en/products/telephony-cards/analog |
| 11:21.28 | basilic | did you have a link to a seller, I have found some but seem very expensive and lot of phone connector |
| 11:22.16 | Greenlight | http://www.digium.com/en/partners/distributors |
| 11:22.22 | Greenlight | Depends where in the world you are |
| 11:22.40 | basilic | in france |
| 11:23.05 | Greenlight | You might try ebay |
| 11:23.07 | basilic | but no really important, I would just confirme if the hardware I found is for me |
| 11:23.29 | basilic | I have found hardware to 400$ to more 3000$ |
| 11:23.40 | basilic | I'm not sure that for a personnal usage |
| 11:23.43 | Greenlight | FOr example http://www.ebay.co.uk/itm/HOT-TDM800P-with-8-FXO-FXS-ports-Asterisk-card-800P-Digium-Trixbox-4FXS-4FXS-PCI-/271063233273?pt=UK_Computing_MicrophonesPhones_RL&hash=item3f1ca0b2f9 |
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| 11:24.18 | basilic | I need only one ports :) |
| 11:24.27 | Greenlight | Get an ATA :) |
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| 11:25.35 | arsperger | hi guys |
| 11:25.42 | basilic | okai thank Greenlight |
| 11:25.50 | arsperger | does anybody know about h323 or ooh323, is it possible to register to multiple gatekeepers? |
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| 11:28.23 | *** join/#asterisk admin0 (~admin0@2a00:ec8:404:1127:10d8:f988:e80e:e3f8) |
| 11:28.44 | admin0 | hi all .. is it possible to have more than one rewrite rule per call.. like if call comes with 00, remove 00, if it coems with #, remove #, if it comes with 011, remove 011 etc |
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| 11:31.37 | madduck_ | is it possible to record a voicemail message that is notification-only, i.e. no recording of messages after playing the message to the caller? |
| 11:31.58 | madduck_ | i know i can do this with the dialplan, but I would like to be able to configure this via the voicemail menu |
| 11:32.01 | Greenlight | You mean just playback a recoridng and hangup ? |
| 11:32.26 | Greenlight | voicemail menu ? |
| 11:33.31 | madduck_ | yes, i would like to be able to configure this via standard voicemail, i.e. without access to the asterisk server |
| 11:36.47 | Greenlight | You mean inside the VoiceMailMail application? |
| 11:36.57 | madduck_ | yes, exactly |
| 11:37.22 | madduck_ | i can record a temporary greeting, but I'd really like to be able to record a temporary notice that then just hangs up instead of recording |
| 11:37.39 | Greenlight | I don't think there's a way to achieve what you're looking for via that. You'd need to customsize your dialplan to acoomodate it/ |
| 11:38.14 | Greenlight | Although there's no reason you couldn't have a mini-IVR before hitting voiemailmain, which allows configuration of a temp message |
| 11:38.53 | Greenlight | We have a similar thing where we've an extension to enable or disable an "Unforceen Circumstances" message |
| 11:42.23 | gartral | where are the default MySQL database seeds for asterisk? |
| 11:43.04 | Greenlight | database seeds ? |
| 11:44.41 | gartral | Greenlight: the "sql.db" file containing the structure of tables.. I call that a seed |
| 11:47.32 | Greenlight | Which tables ? |
| 11:48.28 | gartral | Greenlight: any. enough to get users connecting and the server to stop throwing "[Oct 25 07:39:57] WARNING[22655]: db.c:329 ast_db_put: Couldn't execute statment: SQL logic error or missing database" would be nice :) |
| 11:49.21 | Greenlight | Asterisk doesn't *need* any MySQL tables to work. |
| 11:50.28 | Greenlight | Internally it uses sqlite not mysql, and that database is generated automatically on first run |
| 11:51.20 | gartral | Greenlight: hmm.. http://paste.ubuntu.com/6300250/ |
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| 11:52.34 | gartral | Greenlight: see every time I register a phone, I get the database error.. also, why would jabber/xmpp be flooding me like that? |
| 11:54.04 | Greenlight | Are you trying to use realtime ? |
| 11:54.05 | gartral | Greenlight: nvm, wrong password in xmpp.conf! that would do it |
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| 11:54.46 | gartral | Greenlight: it is my understanding that google talk/voice functionality requires realtime, so unless I hear otherwise, yes |
| 11:55.06 | Greenlight | Um really thats new |
| 11:55.17 | Greenlight | Although I've not used google talk with asterisk |
| 11:55.52 | Greenlight | I don't see why google talk would require the usage of realtime sip peers |
| 11:56.45 | gartral | Greenlight: ok, so ignoring that, why would that db error come up? |
| 11:57.09 | Greenlight | Lets see your sip.conf |
| 11:57.49 | gartral | give me a moment to sanitize it.. |
| 11:58.20 | Greenlight | If your trying to use realtime, and have no realtime database setup, then you're going to get an error. |
| 12:00.03 | gartral | Greenlight: http://paste.ubuntu.com/6300281/ |
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| 12:01.54 | Greenlight | Ok, and are you trying to use realtime sip peers ? |
| 12:02.46 | gartral | Greenlight: i.. honestly I don't know :( I feel like an idiot staring at all these configs |
| 12:03.19 | Greenlight | Lets see extconfig.comnf |
| 12:03.41 | prasha | is it possible to make debian from asterisk src |
| 12:04.25 | gartral | prasha: i don't believe that is possible, but making asterisk from source in debian is.. |
| 12:05.14 | gartral | Greenlight: for all intents and perposes.. extconfig.conf is empty >.> |
| 12:05.56 | prasha | gartral: so the only way to install asterisk newer version is by compiling it on production machine |
| 12:06.13 | *** join/#asterisk Weezey (~ohno@i.am.weezey.com) |
| 12:06.30 | Greenlight | Very odd.. Maybe something hasn't installed correctly. |
| 12:06.36 | *** join/#asterisk dxd828 (~dxd828@195.191.107.205) |
| 12:06.42 | Greenlight | Try starting asterisk with "asterisk -cvvvv" and watch for any errrors |
| 12:06.56 | gartral | prasha: ahh, you want to make a .deb package.. yes, that should be possible, but generally it's easier to compile on the machine it's too run on |
| 12:07.07 | gartral | Greenlight: http://paste.ubuntu.com/6300306/ |
| 12:07.51 | gartral | Greenlight: I did say that for all intents and perposes, that extconfig.conf was empty, I feel that I've missed this step |
| 12:07.53 | prasha | gartral: yes I want to make .deb only. |
| 12:08.17 | Greenlight | Lunchtime... afk... |
| 12:09.02 | gartral | prasha: you may want to look at this https://wiki.debian.org/HowToPackageForDebian |
| 12:09.25 | prasha | I will give it a try .. but how to do 'make samples' and 'make config' |
| 12:09.50 | prasha | post installation |
| 12:11.08 | gartral | prasha: on that.. i'm not too sure, though I can presume that you would do all that in your chroot and package it up in one nice bundle |
| 12:13.54 | prasha | gartral: hmm. I will give it try. Thank you for help. |
| 12:16.20 | gartral | prasha: best of luck! |
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| 12:24.33 | gartral | Greenlight: please ping me when you've returne |
| 12:26.55 | admin0 | is there a way to see transcoding being done in asterisk cli or somewhere |
| 12:28.53 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 12:28.59 | [TK]D-Fender | look at your channels |
| 12:29.45 | [TK]D-Fender | When 2 that are bridged are using different codecs... there's that... |
| 12:29.49 | Greenlight | gartral: Im back, did you see any errors at startup ? |
| 12:29.59 | [TK]D-Fender | on a file basis, nothing I'm aware of |
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| 12:31.22 | gartral | Greenlight: just the same db error.. ocer and over about a minute apart |
| 12:31.56 | Greenlight | Which user are you running asterisk as ? |
| 12:32.17 | gartral | Greenlight: asterisk |
| 12:33.02 | Greenlight | Ands who owns the astdb file ? |
| 12:33.15 | Greenlight | Usually in /var/lib/asterisk |
| 12:35.19 | gartral | facepalms and mutters "root.." |
| 12:35.27 | Greenlight | lol |
| 12:35.37 | Greenlight | I think we're onto something here... |
| 12:36.14 | Greenlight | Perhaps asterisk was first started as root and that's when the file would have been created |
| 12:37.05 | gartral | Greenlight: likely.. I'm prone to doing that.. |
| 12:37.35 | Greenlight | STill best to check the ownership of the rest of the files in there and in /etc/asterisk |
| 12:38.54 | gartral | Greenlight: fixed that.. |
| 12:39.43 | Greenlight | I'd imagine that'll fix your issues. Not sure if a restart will be required or not, perhaps best to restart asterisk to be safe. |
| 12:42.42 | gartral | YAY! I can call internally now! |
| 12:43.01 | Greenlight | :) |
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| 12:46.26 | gartral | ok, never.. ever transfer a call your in too yourself.. |
| 12:46.32 | gartral | ow my ear |
| 12:46.52 | Greenlight | Did you're phone auto conference or something ? |
| 12:46.59 | Greenlight | *your |
| 12:47.10 | gartral | Greenlight: no.. feedback loop |
| 12:47.28 | Greenlight | But how if you were on the phone? |
| 12:47.34 | Greenlight | Or was it to your mobile ? |
| 12:48.06 | gartral | Greenlight: callwaiting put call 1 on hold to ring for call 2 which was transfered from call 1.. |
| 12:48.41 | gartral | I'm using softphones on different comps at the moment to test |
| 12:48.45 | Greenlight | Ahh |
| 12:49.39 | gartral | probably not the single brightest idea i've ever had, but seeing as I can't afford real sipphones.. yea.. working with what I gots |
| 12:50.15 | gartral | would love 2 of those executive-grade wifi-enabled sipphones |
| 12:51.08 | gartral | does not have money for even the basic salesbot-grade wired ones |
| 12:52.47 | Greenlight | Good old BT, they just send us a letter addressed to "Needs Updating" |
| 12:53.33 | gartral | Greenlight: BT? |
| 12:53.52 | Greenlight | British Telecom |
| 12:54.31 | Greenlight | They own and operate most of the phone network here in UK |
| 12:54.57 | gartral | Ahhh.. ok.. |
| 12:55.38 | gartral | that's better than AT&T here in the USA who stole my damn front door key >.< |
| 12:55.41 | Greenlight | Dealing with them from time to time is a neccissary evil. They've never managed to *not* to screw up an order I've placed, ever. |
| 12:56.19 | coppice | saying AT&T is better than BT is like comparing the flavour of turds |
| 12:56.19 | gartral | haha.. yea, I think I see the similarities.. |
| 12:56.51 | Greenlight | My last order I was force to play them back the call recording I'd made when originally ordering. |
| 12:58.10 | gartral | coppice: I don't think we were saying one is better than the other, it was more a comparisson of horror stories, highlighted by my ack that Greenlight was left in a slightly better position in the end |
| 12:59.02 | Greenlight | I like the turd analogy, very apt. |
| 12:59.04 | coppice | its still "my turd is stinkier than your turd" |
| 13:00.19 | Greenlight | I can't say I've ever dealt with AT&T, but as turds go, I can confirm BT are a real stinker... |
| 13:01.20 | gartral | AT&T sent me a 400USD phone bill because "they lost my billing address".. |
| 13:01.39 | gartral | they sure found it right quick.. |
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| 13:02.22 | Greenlight | Must be something about large telcos |
| 13:02.52 | gartral | Greenlight: yup. They Suck. |
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| 13:23.14 | gartral | errr... |
| 13:23.45 | gartral | the voicemail promt welcomes me to "comedian mail" Whiskry Tango Foxtrot, Over? |
| 13:25.14 | Greenlight | Some of the bundled sounds are rather amusing |
| 13:26.44 | gartral | yea but.. "Comedian Mail"? |
| 13:26.52 | gartral | really? |
| 13:27.51 | coppice | its more meaningful than the original "Meridian Mail" that is parodys |
| 13:28.26 | [TK]D-Fender | And is likely another reason Asterisk isn't taken seriously by a lot of people. |
| 13:28.54 | [TK]D-Fender | I wish they'd let the jokes go.... |
| 13:28.55 | skrusty | is there any ongoing effort to update the stock sounds in asterisk? |
| 13:29.26 | [TK]D-Fender | skrusty: "update"? They don't have "bugs".... |
| 13:30.09 | skrusty | no, but they do seem quite dated at times - maybe it's just me, but it wondered if there was a community lead effort to get standard local based sounds out for asterisk |
| 13:30.32 | skrusty | i know a few people have released their own, say for en-GB, but often they get out of date quickly |
| 13:31.09 | skrusty | if you catch my drift :) |
| 13:31.19 | coppice | dated? you mean like the one that says "Thou hast X new messages"? |
| 13:31.40 | skrusty | haha :) well, that would be odd, as it's not an English women speaking it is it! :) |
| 13:32.21 | gartral | I wouldn't mind my prompts screaming at the lusers in klingon.. now *that* would be funny! |
| 13:32.29 | skrusty | :) |
| 13:32.36 | skrusty | today is a good day to DIAL |
| 13:33.14 | gartral | must ask his mother if she would record Klingon prompts |
| 13:34.27 | *** join/#asterisk dxd828 (~dxd828@195.191.107.205) |
| 13:34.34 | [TK]D-Fender | skrusty: Digium doesn't provide en-GB IIRC... which means it's up to that packages maintainers to do whatever they have to do. |
| 13:35.11 | coppice | well, they're obviously not going to support all the obscure languages |
| 13:35.14 | Greenlight | Theres an en-GB one around but some sounds are missing, meaning you'#re left with a mix US and GB accents |
| 13:36.36 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:36.36 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:37.04 | gartral | Greenlight: you could set it to french and tell everyone to broaden their horizons and learn a new damn language! :P |
| 13:37.41 | Greenlight | Aside from Chinese, isn't French most widely spoken language? |
| 13:40.21 | skrusty | no i don't expect digium too, hence why i asked if there was a 'community' lead effort |
| 13:40.55 | skrusty | Greenlight: Mandarin, Spanish, English |
| 13:40.57 | skrusty | in that order |
| 13:41.16 | skrusty | http://en.wikipedia.org/wiki/List_of_languages_by_number_of_native_speakers |
| 13:41.30 | skrusty | French comes a long way down the list :) |
| 13:41.43 | Greenlight | Yea, so I see... I thought it was a lot higher for some reason |
| 13:41.55 | skrusty | they would just like people to think that :) |
| 13:42.01 | Greenlight | :) |
| 13:42.53 | Greenlight | Right... off for the weekend.... laters! |
| 13:43.09 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-nytcbajqmbaookgn) |
| 13:43.09 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:48.46 | eirirs | "I speak Spanish to God, Italian to Women, French to Men, and German to my Horse." |
| 13:49.51 | file | we do accept entire sound set contributions, provided that the contributor keeps them up to date when need be (they don't have to monitor, we ask when that comes up) just fyi |
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| 13:51.14 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:53.59 | *** part/#asterisk arsperger (~arsen@176-12-18-88.pon.spectrumnet.bg) |
| 13:54.45 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
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| 14:23.44 | *** join/#asterisk zafu (~pif@84-74-26-25.dclient.hispeed.ch) |
| 14:24.01 | zafu | hi, when using MixMonitor, should I place the command after Answer()? |
| 14:29.02 | pigpen | Anybody have any ideas why I discovered all my asterisk directory contents are gone? (running system for many years, not new) |
| 14:29.28 | pigpen | I went through history/last/etc, logs, auth.log, etc with no hints. |
| 14:29.33 | pigpen | disk issue? |
| 14:30.03 | pigpen | It was running 1.6.2, so it was time to upgrade anyway, but odd. |
| 14:30.27 | sruffell | hacked? |
| 14:30.31 | *** join/#asterisk opticron (~opticron@pianoben.ch) |
| 14:30.48 | pigpen | no sign of it. pretty secure. |
| 14:31.42 | pigpen | I am leaning to a raid controller issue. Box is about 8 years old. Original drives. |
| 14:31.56 | pigpen | Still, I would like to find the smoking gun. |
| 14:32.20 | *** join/#asterisk navaismo (~navaismo@189.191.200.240) |
| 14:33.40 | sruffell | raid controller is possible…but unlikely without other problems on filesystem that fsck would detect. Still most likely some user (or non-user) IMO. |
| 14:34.12 | pigpen | yeah, I agree. |
| 14:34.42 | pigpen | Waiting for my partner to get available to take a look. |
| 14:36.04 | *** join/#asterisk jygrrr (~jygrrr@201.210.249.75) |
| 14:39.10 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 14:40.59 | gartral | WOOHOO! |
| 14:41.15 | gartral | I has external calling working on my server now! |
| 14:41.56 | skrusty | :) |
| 14:43.15 | gartral | bit scratchy.. but at least I can freaking make calls |
| 14:46.24 | *** join/#asterisk dxd828 (~dxd828@195.191.107.205) |
| 14:48.03 | *** join/#asterisk dxd828 (~dxd828@195.191.107.205) |
| 14:48.28 | gartral | waits for wakerupper to call and see if incoming works |
| 14:48.44 | gartral | is lame, and has no cell phone |
| 14:51.48 | gartral | nice, incoming is working as well |
| 14:52.12 | *** join/#asterisk vlad_starkov (~vlad_star@217.26.6.70) |
| 14:52.38 | *** join/#asterisk dxd828 (~dxd828@195.191.107.205) |
| 14:53.28 | gartral | is there a way to register 2 devices to the same account without them interfering? |
| 14:53.42 | navaismo | asterisk 12 |
| 14:54.38 | gartral | phoo |
| 14:55.14 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-79-141.user.veloxzone.com.br) |
| 14:56.58 | gartral | hmm.. problematic.. I can't use * or # in a call.. |
| 15:02.16 | zafu | what recording formats are available with MixMonitor ? |
| 15:02.29 | *** join/#asterisk fprior (c9dc96b2@gateway/web/freenode/ip.201.220.150.178) |
| 15:02.40 | zafu | alaw, wav and gsm work, but I couldn't get speex to work |
| 15:02.43 | navaismo | wav gsm |
| 15:03.23 | zafu | alaw (file ext: .al) is much smaller than wav with the same quality |
| 15:03.53 | zafu | gsm sounds crappy |
| 15:10.38 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 15:11.14 | Ice_Strike | Sometime recorded conversation overlapping |
| 15:11.25 | Ice_Strike | What is causing this? |
| 15:20.15 | gartral | ok, so now I have incoming and outgoing numbers working for one user on asterisk using google.. how do I expand that so all users can use that facility? |
| 15:20.27 | gartral | or at least, more than one |
| 15:20.31 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:21.55 | newtonr | Ice_Strike, Do you mean that the resulting recording of a channel contains audio from another channel? |
| 15:21.58 | davlefouAMD | hi, how can change time code before compil asterisk? I have get same version but there not same time code. |
| 15:22.54 | *** part/#asterisk TobSnyder (~schneider@146-52-43-241-dynip.superkabel.de) |
| 15:23.10 | Penguin | davlefouamd: patch, or manually edit the code. Then compile. |
| 15:23.17 | Ice_Strike | @newtonr It seem to be an issue when a call been transferred to another line |
| 15:23.24 | Ice_Strike | then it start over-lappinf |
| 15:23.57 | Penguin | gartral: What you are asking doesn't really make any sense. Asterisk has configuration for devices. If you configured one device, configure more in the same way. |
| 15:24.00 | Penguin | ~book |
| 15:24.01 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:24.02 | Penguin | gartral: ^ |
| 15:24.37 | newtonr | Ice_Strike, what application are you using to record the call? What is the channel type? What kind of transfer are you doing(attended, blind, etc)? |
| 15:24.58 | davlefouAMD | Penguin, where i can find these information? |
| 15:35.37 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
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| 16:01.05 | Penguin | davlefouamd: I just gave you the information. So I guess you find it by reading what I typed. |
| 16:01.57 | Sythius | hi, does anyone know this error message? It happens when i do an internal call from SIP to UNISTIM. http://pastebin.com/ueXLwkkQ |
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| 17:23.23 | saxa | anonymouz666: hi, I have received the following DATACOM device from my telco. |
| 17:23.32 | saxa | http://picpaste.com/IMG_20131025_181446-L5IDukJX.jpg |
| 17:23.45 | saxa | http://picpaste.com/IMG_20131025_181302-qNVhzFd8.jpg |
| 17:24.25 | saxa | I suppose this gets connected to my TE133F card |
| 17:25.19 | [TK]D-Fender | that looks like a DATA router. |
| 17:25.22 | [TK]D-Fender | not a voice interface |
| 17:27.06 | saxa | now my question is, do I connect this in the front or on the back side ? |
| 17:27.13 | saxa | there is 2 RJ45 |
| 17:27.38 | saxa | I know I need to ask my telco , but I suppose this will work with my TE133 , hopefully |
| 17:27.46 | saxa | hi [TK]D-Fender |
| 17:28.02 | saxa | the grey cable comes in is the line |
| 17:28.09 | ChannelZ | "the line"? |
| 17:28.21 | Penguin | Your picture is too blurry to read the label on the rear port. |
| 17:28.24 | saxa | its connected to the copper wire |
| 17:28.31 | [TK]D-Fender | Your picture is blurry and we can't read the labels on them |
| 17:28.36 | saxa | Penguin: i can do a better one |
| 17:28.49 | saxa | ok but labels seem a bit strange |
| 17:28.55 | saxa | I will do a better pic |
| 17:29.10 | ChannelZ | What have you bought from the telco such that they've given you that box? |
| 17:32.12 | saxa | they given me that box |
| 17:32.25 | saxa | but to be honest seems a thing made at home :D |
| 17:34.05 | ChannelZ | yes.. but what service did you sign up for that caused them to give it to you? |
| 17:35.22 | ChannelZ | It seems like some sort of weird specialized DSL modem |
| 17:35.50 | ChannelZ | http://www.datacom.ind.br/new/files/DM991CE_Series_VI_ing.pdf |
| 17:37.13 | Penguin | It's an SHDSL modem. |
| 17:38.55 | ChannelZ | uses his expansive cuss vocabulary to imagine what the SH might stand for |
| 17:39.33 | saxa | http://picpaste.com/IMG_20131025_193123-i7cyvPqK.jpg |
| 17:39.37 | saxa | http://picpaste.com/IMG_20131025_193137-kssVZnkK.jpg |
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| 17:40.24 | *** join/#asterisk dronacid (~dronacid@vpn.openroot.de) |
| 17:40.26 | saxa | those should be better |
| 17:41.15 | saxa | ChannelZ: we asked for ISDN but they had it not, and they gave us this box to connect with them with Digitronco, a E1 line |
| 17:41.48 | Jonnys | Hey all, I have a question hoping you guys can lead me in the right direction, I have a AsteriskNow server installed with FreePBX the latest version. I cannot hear the other person on the phone. I did some research and it took me to translation times and I do have high translation times. Any ideas on what it could be. |
| 17:42.35 | coppice | saxa: the socket which is labelled 120 ohms is your E1 port |
| 17:42.36 | [TK]D-Fender | saxa: I see DSL in and an ethernet out.. I do not see a "smartjack" type out you can plug to your card |
| 17:43.04 | [TK]D-Fender | coppice: that 120ohm says "IN" though.... seems a little misleading |
| 17:44.07 | coppice | That is the E1 port. the 2 BNCs offer the same E1 as 75 ohm coax |
| 17:44.39 | saxa | coppice: ok, thx |
| 17:44.50 | saxa | l8r i check that |
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| 17:48.21 | Bkhan | hi |
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| 17:49.41 | Bkhan | I have an issue , I did codec g729 but on cli i receve -- G.729 PLC message continuously. |
| 17:50.40 | saxa | coppice: thx, I saw. The cable I put in there can be a network cable or must be an ISDN one ? |
| 17:51.09 | *** join/#asterisk dxd828 (~dxd828@94.116.252.4) |
| 17:51.10 | saxa | I mean, can make a straight eth cable ? |
| 17:55.15 | zamba | saxa: it can be a straight ethernet cable.. the pin-out is the same.. 1 to 1, 2 to 2, and so on |
| 17:55.29 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 17:55.29 | zamba | i don't think the ISDN cables are crossed in any way |
| 17:55.41 | zamba | meaning the pinout |
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| 18:09.57 | saxa | zamba: perfect, many thanks |
| 18:10.26 | saxa | zamba: i think they are not, as i tried many years ago with one and have not worked |
| 18:11.11 | saxa | anyway thanks. those days i will put together my box and will install asterisk and thank my digium card, after that i will be back with config issues for sure :D |
| 18:11.16 | saxa | thanks to all ! |
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| 19:02.59 | Jonnys | I'm still searching for info on high translation times :(, if anyone knows any fourm links or anything that might help me please let me know. |
| 19:04.10 | Qwell | Jonnys: if it's too high, you either have too much load, or you need a faster CPU |
| 19:04.17 | Qwell | There's not a whole heck of a lot you can do about it. |
| 19:08.51 | *** join/#asterisk dxd828 (~dxd828@212.183.128.231) |
| 19:08.52 | Jonnys | I see, I took that into consideration and mointoring CPU seems to be low |
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| 19:55.36 | ChannelZ | What's "high"? |
| 20:04.29 | *** join/#asterisk fprior_ (b33c66e3@gateway/web/freenode/ip.179.60.102.227) |
| 20:05.30 | fprior_ | Hi all: someone can clarify if is it possible to send a fax through asterisk, between two T.38 endpoints, using Digium Fax Module ? |
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| 20:22.30 | [TK]D-Fender | If you have 2 endpoints then you are not using Digium's Fax Module |
| 20:22.43 | [TK]D-Fender | That is for actually sending/receiving via a file |
| 20:22.51 | [TK]D-Fender | not as something that sits between endpoints |
| 20:34.34 | Jonnys | I upgraded my Dedicated server to have 4 cpu's and 5gb ram hopefully it will fix my high translation |
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| 20:36.33 | fprior_ | [TK]D-Fender so, I need to Digium's Fax Module for send/receive in Dialplan, for example |
| 20:36.52 | [TK]D-Fender | No, you can USE it. |
| 20:36.58 | [TK]D-Fender | "Need" is another matter |
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| 20:37.30 | fprior_ | [TK]D-Fender ok; so, what I need to configure in Asterisk to permit a T.38 trasmission between 2 endpoints ? |
| 20:37.41 | fprior_ | [TK]D-Fender, spandsp ? |
| 20:38.42 | [TK]D-Fender | yes |
| 20:38.45 | coppice | you don't need to add anything to pass FAXes through, unless you want to translate between audio and T.38. You just need to configure your system to allow T.38 passthrough |
| 20:38.48 | [TK]D-Fender | and no |
| 20:38.56 | [TK]D-Fender | youdon't need ANYTHING between 2 endpoints |
| 20:39.16 | [TK]D-Fender | SpanDSP is ANOTHER file send/receive option |
| 20:39.37 | coppice | you will need spandsp if you want to translate betwwen audio and T.38 |
| 20:41.45 | fprior_ | well, now it is clearer; actually I defined "t38pt_udptl = yes,maxdatagram=400" in sip.conf but I cannot send T.38 faxes from Zoiper (for example) to another T.38 Endpoint via PRI |
| 20:42.00 | [TK]D-Fender | backwards thinking |
| 20:42.01 | fprior_ | is t38pt_udptl unique option I need to set ? |
| 20:42.05 | [TK]D-Fender | there is no T.38 over PRI |
| 20:42.13 | [TK]D-Fender | You are doing T.38 to PRI |
| 20:42.25 | [TK]D-Fender | that implies you need T38 to GATEWAY to it |
| 20:42.52 | [TK]D-Fender | this means using the * 10+ function to enable T.38 termination |
| 20:43.28 | fprior_ | Zoiper <--> Asterisk 11 <--> Digium G100 Gateway <--> PRI <--> Destination |
| 20:43.41 | fprior_ | that is my environment |
| 20:43.48 | [TK]D-Fender | how do you communicate to the G100? |
| 20:43.58 | [TK]D-Fender | And you have been vague with your setup description. |
| 20:44.42 | fprior_ | Asterisk and G100 are connected via SIP trunk |
| 20:44.52 | [TK]D-Fender | that does support T.38 does it not? |
| 20:45.32 | fprior_ | it has a configuration page regarding T.38 |
| 20:45.42 | [TK]D-Fender | then it is simply T.38 pass-through |
| 20:46.00 | [TK]D-Fender | no modules or gateway mode required |
| 20:46.41 | fprior_ | let me understand: Asterisk and G100 are only T.38 pass-through; PRI support T.38 ? |
| 20:46.51 | [TK]D-Fender | forget PRI |
| 20:46.54 | [TK]D-Fender | YOU don't touch the PRI |
| 20:46.56 | [TK]D-Fender | the G100 |
| 20:46.58 | [TK]D-Fender | does |
| 20:47.25 | [TK]D-Fender | you are also vague on what that other endpoint is BEFORE asterisk and the G100 |
| 20:48.38 | fprior_ | Before Asterisk there is a Zoiper Extension for tests; but in production system there will be a Old Analog Fax connected to an Adtran |
| 20:49.44 | [TK]D-Fender | if Zoiper = T.38 and G100 = T.38 then it is simple passthrough |
| 20:51.18 | fprior_ | well, everything seems to allow my fax |
| 20:53.49 | fprior_ | regarding G100, in SIP Endpoint Configuration page, it allow change following options: Fax Mode,T.38 Error Correction,Enable Error Correction Mode,Force Local TCF Mode |
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| 22:55.06 | runfromnowhere | Does anyone know whether or not it's possible to get directional audio (i.e. only one side of the conversation) out of the Audiohook C++ API? |
| 23:01.24 | runfromnowhere | Or just about anything useful related to getting Asterisk 1.8 to record calls in stereo |
| 23:02.05 | WIMPy | You will always get both directions seperately. |
| 23:02.45 | runfromnowhere | Right now our deployment is using MixMonitor and we get a single output file with both directions mixed together |
| 23:03.15 | [TK]D-Fender | use monitor and process it yourself |
| 23:03.27 | [TK]D-Fender | "core show application monitor" |
| 23:03.40 | paulc | If you use Monitor, you get 2 audio files - one for each channel. We run it through a script to give agent audio in one channel and caller in the other.. so any "talking over each other" can be seen/heard/delved into. |
| 23:04.23 | runfromnowhere | I'm considering that - the trouble is that we have FreePBX on top and I'd have to do a lot of dialplan rewriting (and FreePBX hacking) to do that. I'm willing to put in the effort if that's the right way to do it. Also will Monitor survive transfers? There's AUDIOHOOK_INHERIT stuff in our dialplan and I'm worried about losing functionality. |
| 23:04.34 | WIMPy | That's what MixMonitor is there for, |
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| 23:07.19 | runfromnowhere | So mostly I think my question mark is whether or not Monitor is as robust as MixMonitor when it comes to inheritance - I don't want transferred calls to drop recording if possible |
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