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| 03:13.08 | mickdubs | what hw (if any) do i need to get residential verizon fios to an asterisk box |
| 03:13.19 | mickdubs | do i need a fxo/fxs card? |
| 03:13.37 | Mango45 | You're referring to phone service from Verizon? |
| 03:13.42 | mickdubs | yes fios |
| 03:13.50 | mickdubs | right now its analog from ont |
| 03:14.05 | mickdubs | im confused? i can call and have them switch that analog to a digital line? |
| 03:14.10 | Mango45 | You could use a FXO card, or you could use an ATA with an FXO port such as an OBi110. |
| 03:14.27 | Mango45 | I am not aware of a way to connect to it directly via SIP. |
| 03:14.37 | mickdubs | what about using a cisco 2700 voice router |
| 03:14.52 | Mango45 | obediently Googles for specifications |
| 03:15.02 | mickdubs | i looked |
| 03:15.12 | Mango45 | Got a link? |
| 03:15.18 | mickdubs | theres something regarding asterisk and the cisco |
| 03:15.19 | mickdubs | yea |
| 03:16.27 | mickdubs | http://www.voip-info.org/wiki/view/Asterisk+cisco+FXO |
| 03:16.29 | mickdubs | may be bs |
| 03:16.36 | mickdubs | but i have one not being used at the office |
| 03:16.42 | mickdubs | it has 4 fxo/fxs ports |
| 03:17.13 | Mango45 | If it has an FXO port, it would do what you want. However it sounds expensive. |
| 03:17.38 | mickdubs | well its free |
| 03:17.44 | mickdubs | its sitting at my office |
| 03:17.44 | Mango45 | Good deal |
| 03:17.57 | mickdubs | but the config would prob be redic to get it working with asterisk |
| 03:17.59 | mickdubs | no? |
| 03:18.28 | Mango45 | Not sure; you're welcome to idle here and see if anyone here has experience with that hardware. |
| 03:18.39 | mickdubs | since i have free gogole voice intergration with sprint |
| 03:18.41 | mickdubs | can i just usse that |
| 03:18.59 | mickdubs | wihtout any cards/connections |
| 03:19.43 | mickdubs | or it looks like i would need a obi 110 or similiar still? |
| 03:21.00 | mickdubs | is it better to choose a pci e card or an external device like a obi |
| 03:21.23 | Mango45 | An OBi would likely be significantly cheaper. |
| 03:21.27 | Mango45 | It performs very well. |
| 03:21.41 | mickdubs | and i could use google voice and fios together? |
| 03:22.22 | Mango45 | The OBi110 has a single FXO port. |
| 03:22.38 | Mango45 | You could possibly connect your Asterisk server directly to Google Voice (without hardware). |
| 03:22.54 | mickdubs | ok i think im gonna start tthere |
| 03:23.02 | mickdubs | i got a esxi server dying for a config |
| 03:23.09 | mickdubs | i got plent of cisco phones to test with too |
| 03:23.23 | mickdubs | im sick of the 3com nbx300's at all my clients |
| 03:23.41 | mickdubs | and cisco ufc systems have proved to be unstable and buggy |
| 03:23.54 | mickdubs | hence them being discontiuned |
| 03:26.16 | Penguin | You can use Google Voice with Asterisk directly over the Internet. |
| 03:26.26 | Penguin | (for now) |
| 03:26.37 | mickdubs | nice im gonna go that route and learrn my way around asterisk |
| 03:26.46 | mickdubs | wiki has that documentation? for setup |
| 03:27.40 | Penguin | https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 03:28.11 | Penguin | But to connect a phone line to asterisk, you have to have an FXO port. |
| 03:28.50 | mickdubs | got it |
| 03:28.54 | Penguin | ATAs such as the SPA-3102 have both FXO and FXS (1 each) on the same device. |
| 03:29.00 | WIMPy | If its a POTS line. |
| 03:29.15 | mickdubs | i think fios is fxs |
| 03:29.48 | mickdubs | i have an old vonage adatper but i heard that wont work |
| 03:29.49 | Penguin | If you plug a regular telephone (analog bell telephone) into the service, it's FXS. Therefore it connects to an FXO port on your devices. |
| 03:30.07 | mickdubs | got it my acyrnms are off |
| 03:30.19 | mickdubs | im setting up pbxinaflash right now |
| 03:30.22 | Penguin | A regular phone jack that you would find on the wall is FXS. |
| 03:30.36 | Penguin | ~fxs |
| 03:30.36 | infobot | [fxs] foreign exchange station - type of port you need to connect an analog device (phone, fax machine) to a pbx. This is the type of port found in your wall jack. |
| 03:30.38 | Penguin | ~fxo |
| 03:30.38 | infobot | [fxo] foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx. This is the type of port found on phone or fax machine devices. See also: http://www.digium.com/index.php?menu=fxsvfxo |
| 03:30.48 | WIMPy | That fios thing is obviousely viop. I'd ask google how to connect directly. |
| 03:31.16 | Mango45 | At least in western Canada, the service providers don't allow it. The Ethernet link light won't even come on when you connect to it. |
| 03:31.27 | Penguin | FIOS just moves the fiber close to the house. Telcos have been using fiber for years. |
| 03:32.01 | Penguin | They're just putting the IP closer to the end point. |
| 03:32.10 | WIMPy | Wikipedia talks about SIP. |
| 03:32.32 | WIMPy | ... since 2000. |
| 03:32.43 | Penguin | The "box" that Verizon supplies likely contains an ATA. |
| 03:32.53 | WIMPy | Sorry. late 2000s. |
| 03:34.01 | WIMPy | Surely getter to hack it than converting to analog and back again. |
| 03:34.36 | Penguin | Wouldn't it be nice if you could get Verizon to give you the SIP credentials and disconnect their internal ATA, allowing you to run full IP at home? |
| 03:35.05 | WIMPy | Maybe they do. |
| 03:35.16 | Mango45 | That would definitely be nice. |
| 03:35.17 | WIMPy | If you find someone who understands the request, that is. |
| 03:35.18 | Penguin | Maybe. I don't know. |
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| 03:39.07 | mickdubs | that would be great |
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| 03:54.55 | seme | hey guys I am trying to install asterisk 13 but I am getting errors about missing packages like libpjsua.so.2 |
| 03:55.18 | seme | is there another repo I'm supposed to add |
| 04:02.26 | Penguin | I see pjproject and pjproject-devel in my asterisk-current repo. |
| 04:03.24 | WIMPy | You should ask someone knowing your distro, or preferrably the one who made the packet you installed. |
| 04:04.54 | Penguin | My asterisk-current repo is: http://packages.asterisk.org/centos/$releasever/current/$basearch/ |
| 04:09.46 | seme | ok I got it... |
| 04:09.58 | seme | I needed to have the current and asterisk-13 repos |
| 04:10.05 | seme | now it is installing :) |
| 04:11.51 | seme | oddly I get this error after the install hecking for Asterisk version..[FATAL] Asterisk 1.6, 1.8, 10 or 11 is required for this version of FreePBX. Detected version is: 13.0.1 |
| 04:11.54 | seme | weird |
| 04:14.21 | Penguin | So don't install FreePBX! |
| 04:14.24 | WIMPy | If you use FreePBX, you should ask in #freepbx. |
| 04:14.38 | WIMPy | That another option. |
| 04:14.46 | WIMPy | That's... |
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| 09:21.42 | ThoMe | hiho |
| 09:21.44 | ThoMe | anyone here? |
| 09:21.48 | ThoMe | hiho Qwell |
| 09:22.02 | ThoMe | neo.. knock knock, wake up |
| 09:22.03 | ThoMe | :-) |
| 09:22.07 | ThoMe | leo? ;) |
| 09:29.19 | ThoMe | what can I do when I have this message |
| 09:29.20 | ThoMe | [Dec 4 10:20:40] NOTICE[6336][C-000002ad]: channel.c:4301 __ast_read: Dropping incompatible voice frame on SIP/placetel-00000867 of format ulaw since our native format has changed to (alaw) |
| 09:29.24 | ThoMe | [Dec 4 10:20:40] NOTICE[6336][C-000002ad]: channel.c:4301 __ast_read: Dropping incompatible voice frame on SIP/placetel-00000867 of format alaw since our native format has changed to (ulaw) |
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| 11:19.26 | tomodachi | hmm i think i broke something |
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| 11:19.39 | tomodachi | in my asterisk log i get pbx.c: No application 'ConfBridge' for extension (confroom-internal, 1001, 1 |
| 11:19.53 | tomodachi | it does not seem to manage to load the ConfBridge application any longer |
| 11:22.22 | tomodachi | it does not seem to manage to load the ConfBridge application any longer and i cannot tab complete in the asterisk console |
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| 11:26.11 | gavimobile | currently 2 peers are registering with the same username and pass! how can I reject second registration attempts? |
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| 11:47.11 | gavimobile | type=peer? can someone confirm if this is the way to do thnis? |
| 11:47.13 | gavimobile | this* |
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| 12:11.18 | tomodachi | gavimobile: hmm why do you have multiple devices trying to auth with the same credentials? |
| 12:11.46 | tomodachi | gavimobile: easiest would just to change password in your account info , and then on the device you want working, then the second device will have the wrong password and fail |
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| 12:38.33 | alami | is that a good practice to give a user that run asterisk rights at the /etc/asterisk folder? |
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| 12:47.13 | wdoekes | alami: read rights, yes |
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| 12:48.40 | alami | wdoekes: thanks |
| 12:49.30 | alami | wdoekes: at the definitive guide sudo chown -R asteriskpbx:asteriskpbx |
| 12:49.46 | alami | wdoekes: so the ownership |
| 12:54.51 | wdoekes | ? |
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| 13:02.17 | tomodachi | does gtalk integration with voice work with asterisk still?! |
| 13:02.27 | tomodachi | considering that its hangouts now |
| 13:06.57 | *** join/#asterisk hurdman (~ygcheny@cylon.r0b0t.fr) |
| 13:09.59 | hurdman | hello, i use asterisk with chan_sip, but i can't have early media . During the call, the rtp traffic is relayed by asterisk but never after the progress. It seems that asterisk never send the progress but it pass into case 183:/* Session progress */ |
| 13:10.03 | hurdman | of chan_sip |
| 13:10.16 | hurdman | any idea ? |
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| 13:31.45 | zamba | is it possible to set up asterisk failover? |
| 13:31.52 | zamba | asterisk cluster, if you wish |
| 13:32.35 | [TK]D-Fender | To a fair degree, yes. |
| 13:33.30 | hurdman | any idea of howto make early rtp media working ? i never receive the 183 progress message |
| 13:40.07 | hurdman | is the asterisk sip early media (progress) definitively broken ? |
| 13:42.46 | zamba | confbridge is the recommended conference application now, right? |
| 13:45.06 | zamba | how do i quickly get confbridge up and running? there's too many options here.. and i have a customer that requires the conference room in a couple of hours |
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| 13:46.25 | WIMPy | Plenty of time. |
| 13:46.59 | [TK]D-Fender | zamba: Sample config is right there as is the app's instructions |
| 13:47.16 | [TK]D-Fender | zamba: "core show application confbridge" |
| 13:47.25 | [TK]D-Fender | zamba: "core show function CONFBRIDGE" |
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| 13:50.57 | zamba | oh, that was simple |
| 13:51.03 | zamba | just confbridge(<number>) |
| 13:51.05 | zamba | done |
| 13:54.35 | zamba | [TK]D-Fender: how do i change the default language of the IVR? |
| 13:54.52 | [TK]D-Fender | IVR doesn't have a language |
| 13:55.00 | [TK]D-Fender | your CHANNEL has a language |
| 13:55.07 | [TK]D-Fender | "core show function CHANNEL" |
| 13:55.09 | zamba | <PROTECTED> |
| 13:55.11 | zamba | <PROTECTED> |
| 13:55.13 | zamba | i tried that |
| 13:55.15 | zamba | but still not norwegian |
| 13:55.30 | [TK]D-Fender | zamba: And do you have norwegian prompts installed? |
| 13:55.41 | zamba | thought so |
| 13:55.49 | [TK]D-Fender | upgrade "think" |
| 13:55.58 | zamba | oh, nevermind.. those aren't packaged |
| 13:56.30 | hurdman | pffff i can't understand , i see my progress message from Client 1 => srv asterisk , i see the progress message pass into the chan_sip... but i never seen any progress message from asterisk to my client 2 .... :'( |
| 13:59.43 | zamba | i'm looking for the option of a user to say who they are when they join the conference? where is that defined in confbridge.conf? |
| 14:00.48 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/branches/11/configs/confbridge.conf.sample?revision=407857&view=markup |
| 14:00.54 | [TK]D-Fender | It's in the sample config...... |
| 14:02.04 | zamba | i see it's called intros, but i don't see how you enable it |
| 14:02.27 | zamba | sound_has_joined |
| 14:02.29 | zamba | filename |
| 14:02.31 | zamba | The sound played as a user intro, e.g. "xxxx has joined the conference." |
| 14:02.45 | [TK]D-Fender | announce_join_leave=yes ; When enabled, this option will prompt the user for a name when entering the conference. After the name is recorded, it will be played as the user enters and exists the conference. This option is off by default. |
| 14:02.59 | zamba | oh |
| 14:02.59 | [TK]D-Fender | Is that somehow unclear? |
| 14:03.00 | zamba | my bad |
| 14:03.03 | zamba | sorry, sorry |
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| 15:31.55 | skrusty | does anyone know if there's an online db/webapi type service that lists ips/phone numbers being used for voip based attacks? |
| 15:32.23 | skrusty | one that's open to review and contribute to |
| 15:32.54 | WIMPy | One of those where you have to pay to get removed? |
| 15:33.09 | skrusty | no |
| 15:33.32 | *** join/#asterisk bluesoul|work (~bluesoul@unaffiliated/bluesoul) |
| 15:33.37 | skrusty | basically, i've been collecting ip and numbers that are used in sip invite/flood attacks on our network for the last 6 months |
| 15:33.48 | skrusty | and wondered if it was worth making that data open in some way |
| 15:34.30 | WIMPy | Did you find someone using the same IP for more than a few days? |
| 15:34.52 | skrusty | we collect data on how many times the same ip was used etc yeah |
| 15:35.03 | skrusty | so that can be calculated |
| 15:35.08 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-tcwfjfcbjuadfagf) |
| 15:35.30 | skrusty | we block the IP used in the attack on a sliding window, 1 hour, 1 day, 1 week etc |
| 15:36.47 | WIMPy | I don't know why, but I hardly ever see any SIP attacks here. But H.323 started a few weeks ago. |
| 15:37.17 | skrusty | we see quite a few, we have stuff hosted in aws |
| 15:38.18 | WIMPy | Hmm. Actually I'm not even sure I ever seen anything on the hosted one. |
| 15:38.29 | skrusty | really? :) |
| 15:38.45 | WIMPy | yes |
| 15:38.56 | skrusty | hell, i've seen attacks within 24 hours of launching an instance :) |
| 15:39.00 | *** part/#asterisk XATRIX (~xatrix@77.88.209.171) |
| 15:39.20 | WIMPy | I do have boxes that are under contant attack. |
| 15:39.27 | skrusty | anyway, i just wondered if there was value in me opening that data up |
| 15:39.35 | WIMPy | Don't knwo why "they" like some and not others. |
| 15:40.01 | skrusty | the nice thing i use it for is as soon as one box contributes to my list, it's blocked on all other servers too |
| 15:41.11 | WIMPy | Just bad if you shoot yourself in the foot. Then the back door is closed as well. |
| 15:41.20 | skrusty | (whitelist) |
| 15:41.59 | WIMPy | I'm not always home. |
| 15:42.04 | skrusty | webapi manages the list, so say fail2ban picks up an attack, it sends it to an api via curl. that in tern checks it against a whitelist and then decides what to do |
| 15:42.08 | skrusty | turn |
| 15:45.54 | WIMPy | I find it quite entertaining to see where they try to call. |
| 15:47.32 | WIMPy | At the moment they seem to try to call Israel. |
| 15:53.58 | skrusty | yeah :) |
| 16:02.07 | tomodachi | has anyone hade success with jitsi asterisk and video conferencing? fail to get a picture |
| 16:02.16 | tomodachi | works with SFLphone though |
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| 16:47.04 | _omer | Is asterisk AJAM based on Ajax polling or what? |
| 16:47.47 | igcewieling | Has anyone gotten Polycom Soundpoint (recent) phones to make calls without registering. i.e. this is for outbound calls only. This is on the phone side, not the Asterisk side, so the question is a bit off-topic. I tried reg.2.server.1.register="0" in the Polycom config, but the phone is still complaining the line is not registered and the line keys have red Xs on them. |
| 16:49.11 | igcewieling | All my searches are coming up with people having problems registering, not applicable to my situation |
| 16:49.26 | igcewieling | _omer: you might need to ask a better question |
| 16:49.29 | *** join/#asterisk aruntomar (~aruntomar@114.143.169.166) |
| 16:52.35 | [TK]D-Fender | igcewieling: May be related to allowing URI Dialing. |
| 16:52.55 | [TK]D-Fender | igcewieling: As that isn't implicitly through a registered server, but rather a proxy. |
| 16:53.04 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-rrqmyagksbcojjqd) |
| 16:56.03 | _omer | I need to know how Asterisk-AJAM keeps http connection open from technical point of view. I can see that it uses prototype.js |
| 16:56.47 | igcewieling | [TK]D-Fender: I had it working before, that is what is so frustrating and the docs indicate it should work. |
| 16:56.59 | *** join/#asterisk bluesoul|work_q (~bluesoul@unaffiliated/bluesoul) |
| 17:03.56 | WIMPy | igcewieling: It wasn'y you who had the issue with lost digits won WaitExten/Background, was it? |
| 17:06.31 | igcewieling | WIMPy: I don't think so. |
| 17:07.42 | WIMPy | Damn. Then I really don't remember who I talked to about that. |
| 17:08.01 | stefan27 | Anyone recognizes this error: ? [2014-12-04 17:55:01] ERROR[6218][C-00000002]: res_rtp_asterisk.c:2040 __rtp_recvfrom: DTLS failure occurred on RTP instance '0x7fe77000cae8' due to reason 'missing verify message', terminating |
| 17:13.10 | wonderworld | hi, i ported an asterisk VM from one host to another. now asterisk crashes during startup. Asterisk crashes right after "Loading format_pcm.so." with a kernel? error message in german "ungültiger maschinenbefehl". this could be translated as "invalid machine command" or something like that. |
| 17:13.51 | wonderworld | could this be about the build options because of a different CPU? |
| 17:13.51 | WIMPy | Looks like you set up an incompatible machine. |
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| 17:14.06 | WIMPy | Definitely. |
| 17:14.19 | wonderworld | ok. so i'd edit the makefile to fix that? |
| 17:14.34 | WIMPy | Just ./configure |
| 17:14.44 | wonderworld | ok thanks, i'll try |
| 17:14.56 | WIMPy | That *should* find out, what to do. |
| 17:15.25 | WIMPy | It's said fail on some VMs, but usually it works. |
| 17:15.39 | *** join/#asterisk r00f (~r00f@av.r00f.us) |
| 17:15.44 | WIMPy | Or certain VM/Distro combinations. |
| 17:15.53 | wonderworld | i just copied the virtual disk image to the same virtualization software |
| 17:16.00 | wonderworld | just host hardware changed |
| 17:16.06 | igcewieling | WIMPy: SIP or DAHDI? |
| 17:16.32 | WIMPy | igcewieling: Dialplan |
| 17:16.51 | WIMPy | It doesn't matter where it's comming from. |
| 17:18.58 | igcewieling | that is unfortuante. If it was dahdi you could play with the gains and/or dtmf settings, if it was SIP there might be some dtmf issues, but I usally solve SIP DTMF issues on the endpoint. |
| 17:19.10 | stefan27 | a sort of complicated setup produces my error, but it doesnt happen every time! i dont know what is relevant, but: SIP peer A has encryption=no, transport=udp. SIP peer B has transport=wss and encryption=yes, avpf=yes, dtlsverify=fingerprint, dtlssetup=actpass, dtlsenable=yes. the use case is A is sitting in a holding bridge, and A is redirected to B (through AMI action) |
| 17:19.28 | stefan27 | the sip invite from asterisk to B and the response is at http://pastebin.com/i0huqrWA |
| 17:20.01 | WIMPy | igcewieling: Not it's not about the detection. Asterisk drops digits if they arrive faster than it processes them, which is hard to avoid when using e.g. DUNDi. |
| 17:20.02 | stefan27 | but asterisk is unhappy as says dtls failure -- missing verify message |
| 17:23.18 | stefan27 | It works every other time with that call-scenario and all the time with normal calls.. A calling B or B calling A, ... |
| 17:30.31 | stefan27 | setting dtlsverify=no or even force_avp=yes on B in sip.conf removes the problem |
| 17:34.03 | stefan27 | when the SIP invites and SIP OKs are as in the above pastebin, what kind of "dtls verify message" is asterisk expecting to receive and when and to which port? Im not familiar with dtls |
| 17:38.33 | wonderworld | which STUN server do you guys prefer? i'd like to setup my own to be independant off external services. |
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| 17:41.14 | Mango45 | wonderworld: The best solution is not to use a STUN server. |
| 17:41.23 | Mango45 | What are you using it for? |
| 17:41.35 | wonderworld | webrtc clients |
| 17:41.54 | wonderworld | WIMPy: rebuilding fixed the crash. thanks for the help |
| 17:42.08 | Mango45 | Oh. In that case I got nothing. :( |
| 17:42.14 | stefan27 | apparently we use the stun server contained in the fedora20-repository: resiprocate-turn-server-1.9.7-9.fc20.x86_64 |
| 17:42.18 | stefan27 | i didnt install it |
| 17:42.28 | stefan27 | looks like its both stun and turn |
| 17:42.42 | wonderworld | ok, thanks, i'll have a look |
| 17:43.27 | stefan27 | there may not exist such a package on your OS? |
| 17:43.43 | stefan27 | for your OS* |
| 17:44.14 | wonderworld | i checked, it's in the debian repo |
| 17:44.21 | wonderworld | i'll give it a try |
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| 18:46.39 | drale_ | I have a Tormenta3 four port card and using two. Tor3/0/1 is Clear (In use) RED and the light is red. |
| 18:47.22 | drale_ | I am not sure if this is a problem or when it turned red, but our long distance isnt working and AT&T says one of our circuits is down so they might be seeing that |
| 18:47.39 | drale_ | I didn't know if you have to configure how each port is used or where to troubleshoot |
| 18:48.15 | WIMPy | So what has changed since it worked? |
| 18:48.49 | [TK]D-Fender | WIMPy: Asterisk crossed its first whole-number version ;) |
| 18:49.03 | drale_ | i havent changed configurations on it in months. I stopped and started the asterisk service Tuesday night. Wednesday afternoon our CEO was talking with AT&T and said they changed something. |
| 18:49.41 | WIMPy | Nice. Did they tell what they changed? |
| 18:49.58 | drale_ | No. we are going through 3 tickets and escalations and such trying to figure it out |
| 18:50.42 | drale_ | But i know that forever we have had errors on our asterisk... so Im thinking AT&T is blaming it on that. I dont mind trying to fix it again though. I tried months ago. But everything worked anyways |
| 18:51.31 | drale_ | All our locations go to the PRI so they get considered long distance calls when forwarded lol. So no one can call us. |
| 18:53.17 | drale_ | AT&T thinks the circuit is down. I think they see that bad port. But I told them Im calling you know from it. |
| 18:53.29 | drale_ | now* |
| 18:54.03 | WIMPy | I find it a little hard to follow. You have a port that is down and they thin it's another port that is down? |
| 18:54.21 | drale_ | Sorry old on |
| 18:54.24 | drale_ | hold |
| 18:55.49 | drale_ | I dont think the long distance thing is related. So I will focus on the port issue |
| 18:56.18 | drale_ | So we have two ethernet lines from our PRI, one is green and one is red on the card. |
| 18:56.38 | drale_ | Our card, in our asterisk box |
| 18:57.05 | WIMPy | That's not ethernet. That's S2M or G.703. |
| 18:58.47 | drale_ | Do I have to tell asterisk what to use each port for? |
| 18:59.27 | WIMPy | Sure. |
| 19:01.31 | drale_ | Because I want to figure out if its bad port, bad AT&T, or bad configuration. |
| 19:02.13 | WIMPy | Pretty. Easy. Swap the two lines. |
| 19:02.42 | drale_ | Should I restart asterisk? |
| 19:02.44 | WIMPy | But if the configuration really wans't changed you should know where you don't have to search. |
| 19:03.19 | drale_ | This was built before me so I still dont know much beyond extensions.conf and configuring phones |
| 19:04.09 | drale_ | I found the span info in dahdi and found out how to display the /proc/dahdi/1 to show the status of the ports |
| 19:05.11 | drale_ | Ill have to check that line swap after hours. |
| 19:06.10 | WIMPy | Check the LEDs on the device your card is connected to. |
| 19:07.51 | drale_ | Doesnt have lights. |
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| 19:11.36 | drale_ | thanks for now.... i think the long distance issue was found in a billing error. |
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| 20:42.42 | underport | Hi guys, any of you have used the app pickup to take an incoming call without dialing the number exstension ?, only dialing "*" for example... |
| 20:43.17 | WIMPy | doesn't understand that question. |
| 20:43.50 | underport | I know there is a pickup (Tech / Exten), but I need to take a call blindly... |
| 20:44.11 | [TK]D-Fender | it's your dialplan, do it however you want |
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| 20:47.39 | underport | I know I can do as you like [TK]D-Fender, guys, but I need to know if anyone has a valid exemplified Pickup use that is not using (exten => _ **, 1, Pickup (SIP / {EXTEN: 2}.)) , understand me ?. I need to take a call blindly without knowing where extension is ringing. |
| 20:48.22 | [TK]D-Fender | Set all your devices to the same pickup group |
| 20:48.49 | WIMPy | Or use a pickupmark. |
| 20:49.23 | [TK]D-Fender | "core show application Pickup" |
| 20:49.25 | [TK]D-Fender | yup |
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| 20:50.19 | [TK]D-Fender | WIMPy: You could one-up this by setting it as a global constant :) |
| 20:50.39 | [TK]D-Fender | ALL OF THE CHANNELS! |
| 20:50.58 | WIMPy | Interesting idea. |
| 20:52.29 | underport | this is much better, to be sure that all channels can be taken with a pickup |
| 20:53.24 | underport | But always using the pickupmark [TK]D-Fender? |
| 20:53.37 | [TK]D-Fender | yes |
| 20:54.05 | [TK]D-Fender | underport: I wouldn't call it "great" ... you're just grabbing "whatever" absolutely blindly... |
| 20:54.51 | underport | exacly [TK]D-Fender :) |
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| 20:56.25 | underport | Have you any example to use the pickupmark or google is the best idea :P ? |
| 20:58.26 | WIMPy | 'core show application pickup' or wiki.asterisk.org or google or ... |
| 20:58.26 | [TK]D-Fender | underport: It's a VARIABLE. How much of an example do you need? |
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| 21:01.49 | vassilux | I try use oselec echocanceller so What is value for echocancelwhenbridged parameter |
| 21:02.14 | WIMPy | yes or no |
| 21:02.15 | vassilux | I googled but it is not clear for me |
| 21:03.12 | WIMPy | Hmmm. The parameter name already explains it. What it unclear about that? |
| 21:03.43 | WIMPy | Usually you do EC and the terminals, not inbetween. |
| 21:03.52 | underport | i know [TK]D-Fender, I do not want to dial the number of extension (** 102), I just want to dial ** and I take the call anything else, and I do not see that on this link for example: |
| 21:04.00 | underport | http://forums.digium.com/viewtopic.php?f=1&t=82908 |
| 21:04.02 | WIMPy | Or as close to the terminal as possible. |
| 21:04.14 | [TK]D-Fender | underport: Read the apps instructions |
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| 21:05.56 | underport | ok thanks :D |
| 21:09.25 | vassilux | Ok , What determines this value ? |
| 21:10.52 | [TK]D-Fender | vassilux: You |
| 21:11.06 | [TK]D-Fender | vassilux: And that's a "yes/no" |
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| 21:14.03 | vassilux | I got echo just for few called numbers ... |
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| 22:24.06 | ChannelZ-Wk | is res_fax_digium going to be built for asterisk 13? |
| 22:26.07 | malcolmd | i'd look to res_fax_spandsp if you're wanting to do faxing in asterisk 13. we're not in any hurry to do res_fax_digium for asterisk 13 right now, there's a lot of action still going on for the development team and i'm not sure when we'll be able to get a res_fax_digium for asterisk 13 out. |
| 22:36.02 | ChannelZ-Wk | OK thanks. Not desperate for it, I have a real FAX machine at the office but wanted to do some testing from home where I don't. |
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