| 00:00.28 | ChannelZ | Up is relative |
| 00:00.55 | phix | Indeed it is, so is time apparantly |
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| 04:13.34 | Katty | MAFIA II |
| 04:13.57 | Katty | listens to crickets |
| 04:14.02 | Katty | reads the back of the game case |
| 04:14.27 | Katty | ...this looks pretty good. |
| 04:16.08 | drmessano | lol |
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| 04:37.10 | ChannelZ | I'm more of a Worms kind of guy |
| 04:44.44 | AnonGirl | wat |
| 04:52.35 | MaliutaLap | Katty: don't listen to the cricket - watch it on channel 9! :) |
| 04:52.58 | drmessano | gouges his eyes out at "wat".. again |
| 04:53.29 | Katty | lol |
| 04:53.42 | Katty | drmessano: i'm going to bed, send me a txt. |
| 04:53.46 | Katty | night everyone! ttyl! |
| 04:53.53 | Katty | detaches from session |
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| 10:36.39 | bsv | Does Asterisk usually perform the "Client Hello" part of DTLS-SRTP handshake? |
| 10:36.53 | nunne | Anyone know about limitations in the SORT function? I'm trying to do a very basic "least recent dial" by using TIMESTAMP. But it doesn't want to sort. For example "206:1422268001,207:1422267969" till get evaluated in SORT as 206,207 instead of 207,206 as expected. Are my values in SORT to large? Anyone know the maximum number of digits SORT will take? (I can |
| 10:36.53 | nunne | probably just truncate the values in that case). |
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| 10:54.27 | nunne | Yeah, that was the problem. Truncating the TIMESTAMP to last 6 digits made wonders. I guess Weeks will be enough for least recent dial method ;) |
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| 12:58.48 | axp | hi all |
| 12:59.05 | axp | anyone here is using chan_capi? |
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| 13:01.22 | WIMPy | ~polls |
| 13:01.23 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
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| 13:03.58 | XATRIX | Em, guys, can you explain me. I'm new to VoIP. I have a 3G Dongle modem, which i use to make calls to mobile phones. It works pretty well. How can i find a context which is used to make outgoing calls ? |
| 13:04.17 | XATRIX | I'm using Elastix distro. But it should be somewhere in /etc/asterisk |
| 13:05.13 | axp | <PROTECTED> |
| 13:05.14 | axp | ? |
| 13:05.18 | WIMPy | Contexzts are used for calls commin in to Asterisk. But |
| 13:05.22 | WIMPy | ~elastix |
| 13:05.23 | infobot | extra, extra, read all about it, elastix is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
| 13:06.31 | XATRIX | axp: it has too much entries, but i can't find a needed one. I need to manually generate a call-file, for my asterisk to send a call |
| 13:09.45 | axp | XATRIX: some kind of dongle.conf ? |
| 13:09.56 | XATRIX | XATRIX: yeap. |
| 13:10.04 | bsdice | XATRIX check out https://github.com/jstasiak/asterisk-chan-dongle/ |
| 13:10.19 | axp | i know dongle.conf |
| 13:10.22 | *** join/#asterisk arc_black34 (~arc_black@116.93.58.85) |
| 13:10.26 | bsdice | beware, you need one of those http://wiki.e1550.mobi/doku.php?id=requirements |
| 13:10.30 | bsdice | with voice capability |
| 13:10.39 | bsdice | also beware, it is mighty unstable at times |
| 13:10.53 | XATRIX | No no guys, it's not exactly what i'm looking for |
| 13:10.54 | bsdice | e.g. UMTS stick frozen in "looking for network" state |
| 13:11.17 | XATRIX | My dongle works OK. I can't they the clue on how to make a call by my hands |
| 13:11.27 | *** join/#asterisk zerohalo (~zerohalo@2601:6:f80:224:9ca1:760a:3ab5:c9f5) |
| 13:11.30 | XATRIX | I need to create a call-file and put it to /var/asterisk/spool |
| 13:11.33 | bsdice | what |
| 13:11.40 | bsdice | app_originate |
| 13:11.44 | XATRIX | Which context do i have to use for the call |
| 13:12.40 | axp | XATRIX: something like: exten => s,n,Dial(Dongle/dongle0/+79139131234) |
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| 13:13.00 | XATRIX | axp: can i PM you ? |
| 13:13.14 | axp | yes |
| 13:14.01 | WIMPy | Not enough information. Each call has two ends. What's the other one? |
| 13:14.22 | bsdice | how about Context: in call file? |
| 13:14.22 | *** join/#asterisk wasanzy (~wasanzy@41-66-254-58-dedicated.4u.com.gh) |
| 13:14.27 | bsdice | http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
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| 13:17.28 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:17.59 | axp | i've got an ISDN line which is connected to asterisk via CAPIoverTCP, calling in and out works (inbound only with extension) if i try to do an inbound call asterisk does recognize the call but i can't get asterisk to take or forward the call to an SIP phone. |
| 13:18.24 | axp | <PROTECTED> |
| 13:18.49 | WIMPy | Do you have DDI on that line? |
| 13:19.14 | axp | how should my extensions.conf look like to catch all calls, even those without an extension |
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| 13:19.40 | axp | WIMPy: DID is my line, i only get the extensions passed on to the asterisk |
| 13:19.56 | WIMPy | What? |
| 13:20.10 | axp | WIMPy:? |
| 13:20.39 | WIMPy | There might be an option in the chan_capi.conf to wait for an extension. Or you can use WaitExten in your dialplan. |
| 13:20.54 | WIMPy | Your last statement didn't make any sense to me. |
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| 13:22.25 | axp | WIMPy: is there a way to take the call without an extension? |
| 13:22.40 | axp | some kind of catch all |
| 13:23.03 | WIMPy | Calls without an extension usually go to the s extension. |
| 13:23.37 | WIMPy | Just out if interest: Waht kind of device are you using? |
| 13:24.02 | axp | it is a fritzbox, with capioverTCP |
| 13:24.06 | axp | immediate=yes ;DID: immediate start of PBX with extension 's' if no digits were ; received on incoming call (no destination number yet) |
| 13:24.29 | axp | i did an s,1,DIAL(SIP/100) but had no luck |
| 13:24.54 | [TK]D-Fender | axp: Show us the call and the full dialplan |
| 13:25.04 | [TK]D-Fender | ~pb |
| 13:25.05 | 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. |
| 13:25.06 | [TK]D-Fender | ^^^ |
| 13:25.44 | WIMPy | Why don't you send the call to the phone directly from te FB? |
| 13:28.32 | axp | http://pastebin.com/iJnsnFX0 |
| 13:29.19 | axp | WIMPy: FB doesn't allow call forwarding and other things i need |
| 13:29.20 | *** join/#asterisk wasanzy (~wasanzy@41-66-254-58-dedicated.4u.com.gh) |
| 13:30.20 | axp | currently iam happy with what i got, the only problem i have is that asterisk doesn't act if i there's no extension |
| 13:30.28 | [TK]D-Fender | axp: Where's the call? |
| 13:32.00 | axp | [TK]D-Fender: if i do an inbound call without an extension asterisk doesn't pick up the call |
| 13:32.20 | [TK]D-Fender | axp: I want to see the rejection at CLI |
| 13:32.56 | axp | http://pastebin.com/vSHMrwmC |
| 13:33.49 | axp | the hangingup part is because i hung up the phone on the other side! |
| 13:33.50 | [TK]D-Fender | axp: == ISDN1#02: Incoming call '0664xxxxxx' -> '' <- call is targeting "nothing" |
| 13:34.00 | [TK]D-Fender | axp: And you don't have something that can match "nothing" |
| 13:34.16 | [TK]D-Fender | axp: "s" is NOT a catch-all |
| 13:34.24 | axp | [TK]D-Fender: i thought so |
| 13:34.42 | axp | tried everything s,i, .. |
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| 13:36.12 | [TK]D-Fender | [08:34][TK]D-Fenderaxp: "s" is NOT a catch-all <---- |
| 13:36.44 | [TK]D-Fender | axp: exten => _!,1,Goto(s,1) |
| 13:37.35 | [TK]D-Fender | axp: Actually it'd be best to leave the context entirely at that point |
| 13:37.40 | [TK]D-Fender | axp: but that should do it |
| 13:38.22 | WIMPy | Sure the FB does call forwarding. |
| 13:39.29 | *** join/#asterisk marcoAndres (~marco.dom@200-127-146-186.net.prima.net.ar) |
| 13:40.13 | WIMPy | It should go to s instead of "". Might just be a bug. Did you reload the config with immediate=yes? |
| 13:40.18 | WIMPy | Or better restart? |
| 13:40.28 | axp | WIMPy: yes you're right, the article i had seems to be very old. |
| 13:41.02 | axp | WIMPy: yes i did |
| 13:41.13 | marcoAndres | Hi all, I am trying to configure asterisk but I found difficult to find good documentation and example, can someone point me a good place to start, please? |
| 13:41.34 | WIMPy | ~primer |
| 13:41.35 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
| 13:41.39 | WIMPy | ~book |
| 13:41.39 | 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 |
| 13:41.56 | WIMPy | marcoAndres: That's two starting points. |
| 13:43.08 | WIMPy | axp: That might be a big then. OTOH I don't think FB supports DDI , so that might be part of the issue. |
| 13:43.35 | Kobaz | hey, anyone know how to push a "hard" setting to a polycom for https provisioning |
| 13:44.17 | [TK]D-Fender | Kobaz: as in? |
| 13:44.18 | Kobaz | according to the doc you're supposed to be able to do: <device device.set="1" device.prov.serverType="HTTPS" device.prov.serverName="https://foo"/> |
| 13:44.19 | Kobaz | but that's not working |
| 13:44.26 | Kobaz | to force it to be saved on the phnoe |
| 13:44.28 | *** join/#asterisk `ph8 (~ph8@unaffiliated/ph8) |
| 13:44.32 | Kobaz | rather than tftp-server-name on dhcp |
| 13:44.35 | marcoAndres | WINPy: thanks! |
| 13:45.19 | marcoAndres | ~buybook |
| 13:45.19 | infobot | You can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/0636920025894 so go buy it SERIOUSLY |
| 13:45.33 | axp | WIMPy: FB sends the extensions right(at least what i've tested), ( i can call extension 10) |
| 13:46.01 | WIMPy | axp: Yes, looks like it should be possible, even if not officially supported. |
| 13:47.46 | axp | WIMPy: the only things which isn't working is the "catch all" ;) |
| 13:48.41 | [TK]D-Fender | axp: You tried what I just gave you? |
| 13:48.49 | WIMPy | Time to debug your chan_capi or try the other one. |
| 13:50.09 | *** join/#asterisk bmurt (~brendan@8.39.115.8) |
| 13:51.29 | axp | [TK]D-Fender: just added the line as first in context, didn't work |
| 13:51.53 | [TK]D-Fender | axp: "dialplan show capi-in" |
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| 13:54.47 | axp | [TK]D-Fender: http://pastebin.com/j1qjDRkN |
| 13:55.57 | [TK]D-Fender | axp: Ok, well from the look of things it isn't even trying to hit the dialplan normally... |
| 13:56.25 | [TK]D-Fender | axp: WIMPy here is probably your best source of advice for this.... |
| 13:56.40 | WIMPy | I wouldn't expext that it's possible to match nothing. |
| 13:56.58 | [TK]D-Fender | WIMPy: It actually is... |
| 13:57.03 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:57.27 | [TK]D-Fender | WIMPy: happens on some SIP calls I've seen where they just dial the host without passing anything in the extension. |
| 13:57.33 | WIMPy | Hmm. So you think it just says it's trying to hit nothing, but actually doesn't even try? |
| 13:57.54 | *** join/#asterisk newtonr (newtonr@nat/digium/x-lbrvcaypfnqdcsmq) |
| 13:57.54 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:58.05 | [TK]D-Fender | WIMPy: educated guess from the CLI he got : == ISDN1#02: Incoming call '0664xxxxxx' -> '' |
| 13:58.12 | [TK]D-Fender | WIMPy: from -> to. |
| 13:58.30 | [TK]D-Fender | WIMPy: double quotes looks like "blank/nothing to me so it was a fair try\ |
| 13:58.37 | WIMPy | Well, as I said: Time to debug your chan_capi or try the other one. |
| 13:59.02 | WIMPy | Or use SIP instead of remote CAPI or just pend a fiver for a PCI card. |
| 13:59.38 | WIMPy | *spend |
| 14:01.57 | bsdice | Remote CAPI in 2015 dehehehe |
| 14:02.12 | bsdice | ISDN in 2015... |
| 14:02.30 | WIMPy | Do you know any replacement? |
| 14:02.44 | bsdice | IP |
| 14:03.02 | WIMPy | With? |
| 14:03.11 | axp | <PROTECTED> |
| 14:03.11 | axp | <PROTECTED> |
| 14:03.11 | axp | <PROTECTED> |
| 14:03.29 | *** join/#asterisk Qwell (north@asterisk/developer/Qwell) |
| 14:03.49 | axp | if i set capi to MSN (although i have DID (DDI) |
| 14:03.51 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 14:04.00 | bsdice | With the ears and a provider that talks SIP over IP I guess |
| 14:04.08 | wasanzy | my OBD calls are not going through, looks like there is timeout between the two servers. my call looks like this: http://pastebin.com/sjNcXSgC |
| 14:04.14 | WIMPy | no |
| 14:04.35 | axp | WIMPy: i do have an old AVM PCI Card @home but the server takes pci-x only |
| 14:05.13 | WIMPy | axp: Ok. PCI-e is expensive. But you could try to find an USB adaptor. |
| 14:05.24 | axp | bsdice: our internet connection: 300kbits down, 30kbits up, don't even think about calling via sip ;) |
| 14:06.17 | bsdice | G.729 with ptime=100 could squeeze two calls into those 30kbit/s (12kbit/s per direction) |
| 14:06.28 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
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| 14:07.02 | bsdice | not trying to be funny here, I tried it ;-) 12 kbit/s including IP and UDP and SRTP overhead |
| 14:09.43 | *** join/#asterisk wdoekes (~walter@wjd.osso.nl) |
| 14:11.38 | bsdice | at the codec's maximum ptime of 230ms, bandwidth will go as low as 9.8 kbit per direction |
| 14:12.04 | bsdice | axp so... three calls possible ;-) |
| 14:12.45 | WIMPy | "calls"? Might be ok for PTT. |
| 14:13.41 | bsdice | might even work without cutting down rest of internet traffic by employing ACK-priorization and anal htb traffic classification with voip basically getting everything of your 30kbit/s in worst case |
| 14:14.12 | bsdice | make sure you use busylevel=3 |
| 14:15.12 | XATRIX | Guys, sorry if i'm asking you twice a time, but can you help me to find out a way ? That's my dialplan http://paste.fedoraproject.org/175042/81393142/ |
| 14:15.15 | axp | bsdice: there should be a net upgrade later this year, this is why iam fiddling around with the idn thing, it will be replaced with a sip trunk to my provider |
| 14:15.49 | [TK]D-Fender | bsdice: G.723 @ 100ms, IAX2 :) |
| 14:16.03 | bsdice | axp for me, isdn using HFC or AVM (relabeled Siemens?) chips has always been a pain |
| 14:16.19 | XATRIX | That's how i'd like to make a call - http://fpaste.org/175045/22281773/ |
| 14:16.44 | XATRIX | And that's what i really have after the call made - http://fpaste.org/175026/22801501/ |
| 14:16.49 | XATRIX | What's wrong with it :( |
| 14:17.50 | [TK]D-Fender | XATRIX: Your call file / originate failed to go through |
| 14:17.58 | XATRIX | The goal is to make a call file witch ignite a call to my mobile phone, and play mp3 file |
| 14:18.04 | [TK]D-Fender | XATRIX: And nothing we see there confirms what & how it tried to dial. |
| 14:18.29 | [TK]D-Fender | XATRIX: You also just pasted the complete dilplanf rom a FreePBX system. That is 10 tons of GARBAGE that we don't support here. |
| 14:18.49 | axp | bsdice: FBox seems to do it's job, telephone, Faxing and outbound dial seems to work except inbound call without extension |
| 14:19.00 | [TK]D-Fender | XATRIXThe goal is to make a call file witch ignite a call to my mobile phone, and play mp3 file <- you just showed us a massive dialplan full of things this is NOT using and haven't shown us what you actually ARE doing. |
| 14:19.01 | XATRIX | [TK]D-Fender: yes :( but i don't know where to look for a trouble :( i'm not so experienced with |
| 14:19.02 | bsdice | can't open any pastebins, prohibited by corporate firewall because of all the Sony hacks and NSA surveillance of those sites |
| 14:19.24 | bsdice | axp what country are you in? |
| 14:19.30 | [TK]D-Fender | bsdice: use a web-proxy site.. |
| 14:19.31 | axp | [TK]D-Fender: == ISDN1#02: Pickup extension '' found. |
| 14:19.56 | axp | <PROTECTED> |
| 14:19.57 | [TK]D-Fender | axp: So it is finding a match and processing now? |
| 14:20.13 | [TK]D-Fender | axp: That's kinda vague as to why it can't... |
| 14:20.15 | XATRIX | [TK]D-Fender: the actuall things, is to make this script work http://fpaste.org/175045/22281773/ |
| 14:21.09 | WIMPy | axp: Try the other chan_capi or talk SIP to that FB. |
| 14:21.47 | WIMPy | I guess you can't have a caht-all when using SIP, however. |
| 14:22.01 | WIMPy | catch- |
| 14:22.27 | [TK]D-Fender | WIMPy: "can" |
| 14:22.45 | [TK]D-Fender | WIMPy: _! with catch blanks as well |
| 14:22.49 | [TK]D-Fender | will* |
| 14:23.13 | [TK]D-Fender | XATRIX: [2015-01-26 15:48:34] NOTICE[25874]: pbx_spool.c:392 attempt_thread: Queued call to Dongle/i:351911049927240/0638788980 expired without completion after 0 attempts |
| 14:23.18 | WIMPy | You need to create "phoenes" in the FB to match extensions. That's where the no comes from. |
| 14:23.19 | [TK]D-Fender | XATRIX: the Dongle dial is failing. |
| 14:24.21 | XATRIX | Yes, but maybe i use the wrong context ? |
| 14:24.49 | [TK]D-Fender | XATRIX: No. |
| 14:24.51 | axp | WIMPy: which other chan_capi? had troubles enough to get this running in debian wheezy. (there isn't even a package for asterisk in wheezy) |
| 14:24.56 | [TK]D-Fender | [09:24]XATRIXYes, but maybe i use the wrong context ? |
| 14:25.03 | [TK]D-Fender | [09:23][TK]D-FenderXATRIX: the Dongle dial is failing. <-- |
| 14:25.38 | *** part/#asterisk mjordan (~mjordan@75.76.55.191) |
| 14:26.03 | XATRIX | [TK]D-Fender: you mean HW problem, right ? |
| 14:26.25 | [TK]D-Fender | XATRIX: Or network refusal, or whatever.... |
| 14:26.34 | [TK]D-Fender | XATRIX: the dial attempt on the channel didn't work. |
| 14:26.51 | XATRIX | Yea, i understood |
| 14:26.52 | [TK]D-Fender | XATRIX: Go test it with a normal trunk |
| 14:27.06 | WIMPy | axp: At the bottom of http://voice.yeti.dk/Asterisk_vs_ISDN/6 |
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| 14:54.52 | axp | WIMPy: [TK]D-Fender thanks for help, i'll try further ;) |
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| 15:56.59 | axp | re again |
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| 15:57.07 | axp | [TK]D-Fender: == Spawn extension (capi-in, s, 1) exited non-zero on 'CAPI/ISDN1#02/10-4' |
| 15:57.25 | WIMPy | Cool. So you got it working? |
| 15:57.31 | [TK]D-Fender | axp: Looks like it's processing at least.... |
| 15:57.45 | WIMPy | How did you get it there? |
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| 15:58.46 | axp | WIMPy: did change some msn settings in the fbf |
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| 15:59.21 | WIMPy | So it was that. |
| 16:00.57 | WIMPy | The document I found said to put the main number as first (not) MSN, the longest number as 2nd and all others thereafter. |
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| 16:09.43 | wasanzy | What could be the possible cause of 404 not found error in sip request |
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| 16:10.16 | [TK]D-Fender | wasanzy: "not found", just like it says. |
| 16:10.26 | WIMPy | The "user" was not found. |
| 16:10.36 | [TK]D-Fender | wasanzy: It's looking for XYZ ... and you don't HAVE something to match "XYZ" where it is looking |
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| 16:49.18 | marceloamorim | guys, this is my first time trying to syncronize an ISDN but when I use dahdi_scan the type=digital-E1 |
| 16:49.28 | marceloamorim | should be another option there? |
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| 17:01.00 | marceloamorim | 2Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) |
| 17:01.00 | marceloamorim | when I do dahdi show status |
| 17:01.37 | marceloamorim | and "PRI span 1/0: Down, Active " when I use pri show spans |
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| 17:16.23 | newtonr | marceloamorim, I haven't worked with a T1/E1 in years so I won't be much help, but did you configure your system.conf and chan_dahdi.conf to the specs your telco provided you in regards to the line? |
| 17:18.30 | newtonr | After that, probably want to walk through the user guide troubleshooting section. |
| 17:24.46 | marceloamorim | yeah, I'll try in another E1 already working, and check the cables and other things |
| 17:26.54 | marceloamorim | thx for the feedback newtonr |
| 17:26.55 | *** part/#asterisk marceloamorim (~marcelo@189-90-204-85.isimples.com.br) |
| 17:41.51 | litn | I have a queue set up as rlinear. One problem I'm having is that the order of which calls are answered are not in the order that they were entered into the queue- if someone calls into the queue at this moment, they will be picked up first even if there is a call in there that's been there for 2 minutes. So what happens is that if the queue is busy, the oldest call doesn't have a chance of getting picked up. Any suggestions? |
| 17:43.36 | [TK]D-Fender | Queue should still respect caller order. The strategy should only affect member orde for ringing. |
| 17:43.51 | [TK]D-Fender | Confirm your version, show configs and calls... |
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| 18:04.10 | litn | [TK]D-Fender: here's the queue config, http://pastebin.com/yTD7KaAC |
| 18:04.28 | litn | it will also autofill to ring anyone who isn't on a call |
| 18:05.54 | [TK]D-Fender | litn: Well autofill can cause new callers to potentially get answered before oldest |
| 18:08.23 | litn | any way to prioritize it? |
| 18:08.54 | paulc | not using autofill will honour first in first out (subject to priority/weighting/penalty etc) |
| 18:10.31 | [TK]D-Fender | litn: It'll toss say 3 callers to different agents... you can't MAKE the one getting the oldest call pickup that call... |
| 18:11.02 | litn | yeah, but I mean, if you have 5 calls in the queue and 3 agents, if all 3 agents are ringing, shouldn't it be ringing with the older calls? |
| 18:11.03 | [TK]D-Fender | litn: Autofill gets you the fastest expected answering RATE, just not "in order" necessarily. |
| 18:11.37 | [TK]D-Fender | litn: So if you distibute 3 calls at a time, it SHOULD be the 3 oldest.... but #1 could keep being not-answered while newer ones sneak through. Its' luck of the draw |
| 18:12.45 | litn | I see. Ok |
| 18:13.38 | [TK]D-Fender | litn: Auto-fill bypasses enforced answer order |
| 18:15.03 | [TK]D-Fender | litn: I would recommend ring-all if you want fastest-answer in order. This is if you don't mind to possibility of certain agents taking more calls because of it. |
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| 18:22.24 | RPerre | Hi. I'm builing a asterisk application using ARI with phpari. I have a stasis application "hello-world", outside of this app I 'originate' to number A, then when it enters the stasis i originate to B. I create a bridge and add these two channels to the brigde. I have no sound, A and B don't talk to each other. Can anyone help me to understand why? |
| 18:23.07 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:35.55 | RPerre | not many ppl using ARI xD |
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| 18:41.24 | skrusty | what typeof bridge are you creating? |
| 18:41.32 | RPerre | mixing |
| 18:41.54 | skrusty | and no audio in either direction yeah? |
| 18:41.59 | RPerre | yep |
| 18:42.15 | skrusty | if you just play an audio file on the channel, i assume that works? |
| 18:42.47 | RPerre | it doesn't |
| 18:42.50 | skrusty | oh |
| 18:43.13 | skrusty | can you get audio from anything, even without ari? |
| 18:43.31 | skrusty | like just a normal dialplan playing tt-monkeys for example |
| 18:43.32 | RPerre | yea, if i playback to the channel |
| 18:43.46 | RPerre | even with ARI the sound is good |
| 18:43.51 | RPerre | if i play it to the channel |
| 18:44.04 | skrusty | oh, you said that didn't work? :) |
| 18:44.11 | RPerre | on the bridge |
| 18:44.14 | skrusty | right |
| 18:44.17 | skrusty | so audio to a channel works |
| 18:44.20 | skrusty | but no to a bridge |
| 18:44.23 | RPerre | yes |
| 18:44.23 | skrusty | not |
| 18:44.26 | skrusty | ok |
| 18:44.35 | skrusty | odd |
| 18:44.44 | skrusty | what verssion of asterisk? |
| 18:44.48 | RPerre | 12 |
| 18:44.55 | skrusty | more specific please |
| 18:45.04 | RPerre | 12.7.2 |
| 18:45.13 | skrusty | ok |
| 18:45.31 | skrusty | any errors in the asterisk console? |
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| 18:45.48 | skrusty | have you confirmed both channels are in the brdige? |
| 18:45.55 | skrusty | bridge show <id> all |
| 18:46.43 | RPerre | no erros on console, and both channels are in bridge |
| 18:47.22 | RPerre | i must have a really basic error somewhere |
| 18:47.28 | skrusty | doesn't sound like an ari issue tbh |
| 18:47.41 | skrusty | it's a very simple ari setup |
| 18:47.54 | skrusty | as long as the bridge isn't holding, it should be ok |
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| 18:48.12 | skrusty | try using proxy_media too, as the bridge type (mixing.proxy_media) |
| 18:48.27 | skrusty | type (mixing,proxy_media) |
| 18:48.29 | skrusty | even |
| 18:48.36 | RPerre | lemme try |
| 18:48.44 | skrusty | letme double check that |
| 18:49.14 | skrusty | yeah, proxy_media |
| 18:53.50 | RPerre | odd thing, i have two bridges |
| 18:53.53 | RPerre | wth |
| 18:53.54 | skrusty | erm |
| 18:54.10 | skrusty | not one bridge with two channels?:) |
| 18:54.55 | RPerre | no, two brid. |
| 18:55.03 | RPerre | this is the result of bridge_list |
| 18:55.04 | RPerre | http://pastebin.com/AsqcT9js |
| 18:55.26 | RPerre | 2 different sets of channels |
| 18:55.29 | RPerre | wth |
| 18:55.46 | skrusty | did you destory the bridge from the last call? |
| 18:55.51 | skrusty | because thoes are different channels |
| 18:55.55 | RPerre | i've restarted everything |
| 18:56.03 | skrusty | ok :) |
| 18:56.14 | RPerre | i'm going nuts O_O |
| 18:56.31 | skrusty | hehe |
| 18:56.49 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 19:00.46 | RPerre | actually |
| 19:00.48 | RPerre | i have 3 |
| 19:00.58 | RPerre | the last one is the one i've created |
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| 19:05.15 | skrusty | ok |
| 19:08.37 | skrusty | did you try proxy_media? |
| 19:12.50 | RPerre | i did same result :/ |
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| 19:13.49 | RPerre | skrusty, thanks for trying to help me bud. I have to go, i'll be back tomorrow :) |
| 19:13.57 | RPerre | cheers |
| 19:15.10 | *** join/#asterisk gravspeed (~gravspeed@66-242-174-254.ceres.bvn.net) |
| 19:15.32 | gravspeed | how do i show a list of extensions using a given codec? (g729) |
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| 19:26.50 | Penguin | gravspeed: Extensions do not have codecs, so you cannot do that. |
| 19:27.08 | bsdice | only while in call |
| 19:27.22 | Penguin | Extensions don't have codecs ever. |
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| 19:52.28 | drale_ | I have setup a SIP trunk, we used to use DAHDI PRI, i have a global variable at the top of my extensions.conf called TRUNK and find it referenced many times in extensions.conf and also replaced Dial(DAHDI with Dial(windsteam (my trunk name) |
| 19:53.16 | drale_ | between extensions.con and extensions.ael i have many things matching patterns for outgoing calls. nothing i change is affecting the call it still goes to DAHDI, altough CALLERID changes in .ael work |
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| 20:20.07 | smash` | Is there anything wrong with Polycom VVX 500 phone? |
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| 20:45.28 | drale_ | i figured it out. there was two patterns to match in extensions.ael with the same information under both |
| 20:45.52 | drale_ | i had to update the SIP in both places. did a find replace. and nows its all working yay |
| 20:47.03 | AnonGirl | drale_, SIP trunk doesn't exist |
| 20:47.52 | drale_ | are you saying thats the problem or that "sip trunk" is not correct terminology |
| 20:48.16 | AnonGirl | the latter |
| 20:53.25 | drmessano | SIP does not trunk |
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| 20:59.06 | drale_ | but, the provider trunked the service to a Edgewater device here and does all the authentication for us. we just have to do SIP to their device |
| 20:59.13 | drale_ | is that the trunk? |
| 20:59.49 | drale_ | http://www.digium.com/en/products/sip-trunk |
| 21:04.12 | voipy | Hello everyone, does anyone know about multicast paging? |
| 21:06.49 | voipy | My goal is to send an audio file via multicast page to multiple cisco SPAs |
| 21:10.07 | *** join/#asterisk pi_ (~pi@access-63-249-68-55.static.cruzio.com) |
| 21:12.15 | [TK]D-Fender | drale_: that servive has nothing to do with your provider |
| 21:12.29 | [TK]D-Fender | drale_: You need to set up a proper peer to them |
| 21:13.15 | drale_ | i know im just referencing the term "sip trunk" since its everywhere. must be marking if it doesnt exist |
| 21:13.23 | drale_ | marketing* |
| 21:13.31 | [TK]D-Fender | drale_: It's a TERM |
| 21:13.40 | [TK]D-Fender | Stop looking for it as a "thing" |
| 21:13.48 | [TK]D-Fender | that is THEIR service |
| 21:13.56 | [TK]D-Fender | You need to set up a peer to YOUR provider's equipment |
| 21:13.58 | [TK]D-Fender | ~book |
| 21:13.59 | 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 |
| 21:14.00 | drale_ | i did |
| 21:14.00 | [TK]D-Fender | ^^^ |
| 21:14.34 | [TK]D-Fender | And go read the sample config and make sure it matches what they want you to send and where they want you to send it |
| 21:15.03 | drale_ | i used a tutorial online and it worked, they didnt asked me to do anything special, except we agreed on ulaw. |
| 21:15.38 | drale_ | i have read the book and was full focused on managing this box, but that was months ago. and i forget things |
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