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| 00:20.33 | oli-work | hey guys/gals, is it possible to do a CID lookup on internal calls? ie. someone dials an internal extension 7119 and the phone displays the name of the extension? |
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| 00:55.52 | oli-work | i suppose internal reverse CID lookup is what i'm after |
| 00:56.30 | oli-work | it is possible in a broadsoft implementation |
| 00:58.34 | oli-work | the lookup might have been local to the phone though.. |
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| 01:23.13 | [TK]D-Fender | oli-work, "core show function CONNECTEDLINE" |
| 01:25.11 | Katty | FENDERBENDER. |
| 01:25.24 | oli-work | i suppose that means enabling and trusting rpid |
| 01:25.30 | oli-work | thanks [TK]D-Fender |
| 01:29.19 | [TK]D-Fender | oli-work, You're welcome |
| 01:29.22 | [TK]D-Fender | Katty, Mew. |
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| 01:33.29 | oli-work | got it working, thanks again [TK]D-Fender |
| 01:33.59 | oli-work | ...although "core show function CONNECTLINE" was a lot less helpful than voip-info article on it ;p |
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| 02:50.37 | chandoo | hi |
| 02:50.42 | chandoo | i am installing freepbx |
| 02:50.47 | chandoo | i am getting error |
| 02:50.50 | WIMPy | lo |
| 02:51.14 | WIMPy | Ask in #freepbx |
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| 03:47.23 | natu | hello, i try to use webrtc (using the sipML5 client) to place a video call into the Echo() application. Audio works with vp8 and the rtp packets seem to reach the server but I cannot see anthing in sipML4. With h264 neither audio nor video is working. I tested the latest chrome and firefox. Do you have any ideas what I am doing wrong? |
| 03:48.08 | natu | i am using the asterisk version 13.4.0 |
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| 08:08.33 | ldc | hi! |
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| 08:08.37 | ldc | is anyone experienced with endpoint manager? |
| 08:08.45 | ldc | (if that's an asterisk feature at all) |
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| 12:18.29 | [TK]D-Fender | ldc: It isn't. That is a FreePBX module and is supported either by the author based on whether you're using OSS vs commercial |
| 12:24.56 | ldc | ok thanks |
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| 14:02.43 | etherealmktg1 | hello room |
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| 14:03.24 | etherealmktg1 | is there any cloud hosts good for freePBX or similar distribution? |
| 14:03.42 | etherealmktg1 | i cant find a sinlge link or article that isnt over a year old on the topic |
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| 14:05.43 | qakhan | i am using asterisk 11.8, it crashing after 5 mins |
| 14:05.57 | [TK]D-Fender | etherealmktg1: Because a host is a host is a host |
| 14:06.02 | qakhan | some time less then 1 mins |
| 14:06.10 | [TK]D-Fender | qakhan: Upgrade |
| 14:06.14 | [TK]D-Fender | qakhan: NOW |
| 14:06.23 | [TK]D-Fender | <PROTECTED> |
| 14:06.24 | [TK]D-Fender | ^ |
| 14:06.36 | qakhan | currently can we do some thing |
| 14:06.45 | qakhan | for time being |
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| 14:07.13 | etherealmktg1 | i imagine anything you "can" do is unsupported/dangerous |
| 14:08.21 | etherealmktg1 | so just install freepbx from scratch? is there anything I should know before i go/ |
| 14:08.23 | etherealmktg1 | ? |
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| 14:08.53 | babak | Hi,I installed DAHDI latest, libPRI latest and Asterisk 11.18 on debian 7.8 from source code, now led is green, dahdi_genconf generated correct file , lsdahdi output is ok, but in asterisk in pri show channels or Dahdi show channels I see nothing |
| 14:11.28 | [TK]D-Fender | [10:07]etherealmktg1i imagine anything you "can" do is unsupported/dangerous <- It's a virtual server. It's yours. You should know how to maintain a server in all aspects. |
| 14:11.33 | WIMPy | Did you configure chan_dahdi.conf? |
| 14:11.57 | [TK]D-Fender | babak: Show us all of your dahdi configs |
| 14:11.59 | [TK]D-Fender | ~pb |
| 14:12.00 | 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. |
| 14:12.01 | [TK]D-Fender | ^^^ |
| 14:12.26 | [TK]D-Fender | qakhan: You've shown us NOTHING. And you are not on the latest. |
| 14:12.42 | [TK]D-Fender | qakhan: Don't waste time. upgrade now. You are 10 releases behind. |
| 14:12.56 | etherealmktg1 | eek |
| 14:13.18 | flujan | hello. I have a sip trunk between two asterisk boxes, when I enable tcpenable=yes an transport=tcp,udp on sip.conf [general] I got a netsock2.c: getaddrinfo error. |
| 14:13.33 | flujan | Here is a pastie with sip.conf and the error: http://pastie.org/10231419 |
| 14:13.53 | ldc | etherealmktg1: I can tell you a cloud host in query if you want since here it would be unsolicited |
| 14:14.08 | ldc | but bear in mind I don't recommend it |
| 14:14.11 | ldc | :) |
| 14:14.21 | flujan | I have the host IP specified, how asterisk could require a dns lookup? |
| 14:14.38 | babak | http://paste.lisp.org/display/149452 |
| 14:15.13 | etherealmktg1 | brb afk |
| 14:15.17 | qakhan | [TK]D-Fender it is asterisk 11.10, it is generating Core-Asterisk.XX file in /tmp folder |
| 14:15.31 | qakhan | do you want me to show you Core file? |
| 14:15.39 | [TK]D-Fender | <PROTECTED> |
| 14:15.50 | [TK]D-Fender | qakhan: NO. Older relases are NOT supported. Upgrade NOW. |
| 14:16.44 | [TK]D-Fender | babak: Nothing tells me that SIP is the reason for that lookup. |
| 14:16.53 | [TK]D-Fender | babak: You aren' showing enough of what's going on. |
| 14:17.01 | wdoekes | flujan: when? incoming call? outgoing call? |
| 14:17.16 | [TK]D-Fender | flujan: rather |
| 14:17.19 | qakhan | ok is there any command which update asterisk like "update asterisk" on centOS 6.5 |
| 14:17.26 | [TK]D-Fender | 's aim is just a tad off this morning |
| 14:17.38 | [TK]D-Fender | qakhan: How did you install it? |
| 14:17.53 | qakhan | .tar |
| 14:18.04 | qakhan | digium website |
| 14:18.05 | flujan | both. |
| 14:18.16 | flujan | wdoekes: both |
| 14:19.29 | [TK]D-Fender | qakhan: Then that';s how you upgrade |
| 14:19.55 | wdoekes | flujan: next task; enable more output to show what it does when it wants to do that lookup |
| 14:20.14 | wdoekes | core set debug 10 |
| 14:20.51 | wdoekes | the debug statements leading up to that error could provide some clues (sometimes the ones after) |
| 14:20.54 | babak | [TK]D-Fender: WIMPy : it seems ahdi_genconf generate system.conf and dahdi_channels.conf but don't include it in chan_dahdi.conf |
| 14:21.08 | babak | ahdi/dahdi |
| 14:21.16 | [TK]D-Fender | babak: Because that's your job. |
| 14:21.24 | [TK]D-Fender | babak: So go fix it so it's in the right place |
| 14:21.35 | babak | thx ok |
| 14:22.53 | wdoekes | I bet tcpbindaddr isn't set or something, causing the source ip to be null or something |
| 14:23.33 | [TK]D-Fender | babak: I recommend using genconf only ONE and then building your own chan_dahdi 100% independent of it so it doesn't get ripped out from under you after |
| 14:25.25 | [TK]D-Fender | once* |
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| 14:39.11 | litn | Hello. Going to ask this again today and see if there's anyone else that might point me in the right direction for debugging, |
| 14:39.42 | litn | I have an AMI script that when a Newchannel event comes, I look up the number in my database and if it's a certain sort of customer, I send a Setvar and change MOHCLASS to a different hold music for them. |
| 14:40.18 | litn | if I send the Setvar instantly, e.g. not looking it up, it works. If I wait a moment (and I tested it by intentionally waiting before sending it, it seems if I wait 0.003 seconds it will work, anything longer it will not), then the default hold music plays. |
| 14:40.35 | litn | if I look up the actual channel variables in the asterisk CLI, it shows MOH to be set to the new hold music in both scenarios ! |
| 14:43.17 | [TK]D-Fender | Why aren't you just doing it in the dialplan normally? |
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| 14:48.29 | CptBurger | Any have trouble detecting DTMF from Sprint phones? I seem to have trouble detecting DTMF only from Sprint phones. I'm using Digium 4-port analog PCI cards. I've tried relax DTMF and it doesnt seem to help. |
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| 15:18.48 | litn | [TK]D-Fender: I need to do some logic on the various conditions for the hold music |
| 15:18.55 | litn | I figured writing it in a script would be easiest |
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| 15:20.11 | Geek-Linux | Hi. i am facing strange issue. I dont know what am i missing. I am using asterisk box behind firewall over sip tls and RTP. when users call connects and sends hangup request(BYE) it is not received at the server end and calls dont hangs on the server. |
| 15:20.35 | Geek-Linux | I am using Simple ASA firewall. |
| 15:21.03 | Geek-Linux | every thing is working fine except hangup. |
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| 15:33.31 | CptBurger | Geek-Linux: Sounds like you and your SIP are just too impatient... |
| 15:34.36 | [TK]D-Fender | litn: Dialplan = script. Also where you can set all this stuff WITHIN the call instead of trying to invade on it from the outside |
| 15:39.22 | litn | [TK]D-Fender: man, I already have all this code written. Do you know if my issue is unresolvable and I must do it from dialplan? |
| 15:39.48 | litn | the other thing that's nice about the script is that if the database is down for some reason the script passes over it, I don't know how to go about handling that in the dialplan |
| 15:39.53 | [TK]D-Fender | litn: It should always have been done there... |
| 15:39.56 | litn | or even how to access ODBC and run SQL queries from a dialplan |
| 15:40.11 | [TK]D-Fender | AGI <- |
| 15:40.41 | [TK]D-Fender | Should be easy to adapt |
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| 15:53.04 | CptBurger | litn: You may not even need AGI. Depending on your Asterisk verision, you can access ODBC from the dialplan. http://astbook.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/getting_funky.html |
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| 16:00.40 | [TK]D-Fender | Possible, but less control over the rest of the process |
| 16:00.48 | [TK]D-Fender | And he's already written code |
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| 16:18.26 | davlefouAMD | hi, how i change default message on voicemail? |
| 16:19.08 | CptBurger | davlefouAMD: which message? |
| 16:19.24 | [TK]D-Fender | davlefouRECORD one. |
| 16:19.29 | CptBurger | vm-login is the initial "Comedian Mail. Mailbox?" |
| 16:19.36 | [TK]D-Fender | That's what VoicemailMain is for |
| 16:20.44 | CptBurger | But maybe he means the Comedian Mail part. I've editted that out of my vm-login because clients have thought it "improper" /turns up nose. |
| 16:21.15 | CptBurger | "Comedian mail, you say? What is this? Some fly-by-night phone system>?!" |
| 16:22.27 | davlefouAMD | when have an call i play that : Playing 'vm-intro.slin' (language 'fr'), i would like to have my own message for each user. |
| 16:22.43 | coppice | but comedian mail is still here, while the original meridian mail has long since gone |
| 16:22.49 | [TK]D-Fender | go RECORD one |
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| 16:23.28 | [TK]D-Fender | [12:19][TK]D-FenderThat's what VoicemailMain is for |
| 16:23.51 | [TK]D-Fender | coppice: And people wonder why many exec's might not be taking Asterisk seriously... |
| 16:25.27 | davlefouAMD | RECORD one? |
| 16:26.36 | *** join/#asterisk kcormier (185b0fab@gateway/web/freenode/ip.24.91.15.171) |
| 16:26.39 | CptBurger | davlefouAMD: vm-intro plays unless the user has recorded their own greeting. |
| 16:26.41 | [TK]D-Fender | [12:23][TK]D-Fender[12:19][TK]D-FenderThat's what VoicemailMain is for |
| 16:27.07 | [TK]D-Fender | Want a personal message? RECORD one. Go into voicemailmain and RECORD A GREETING |
| 16:27.14 | kcormier | Hi All. I've noticed that the certified branch of asterisk 13 has been released for a bit now, but there are no centos 5 or 6 builds in the yum repo. Is there any plan to get those or can someone direct me to where I can help contribute to those? |
| 16:28.03 | *** join/#asterisk aross42 (~aross@dsl-eclipsefnd.storm.ca) |
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| 17:06.17 | *** join/#asterisk robmal (r@wporzo.pl) |
| 17:09.09 | davlefouAMD | [TK]D-Fender, thanks |
| 17:09.42 | flujan | <PROTECTED> |
| 17:10.34 | flujan | my customer network is dropping UDP invites, he is using a sonicwall⦠Google tells me this: https://support.software.dell.com/kb/sw10958 |
| 17:11.15 | [TK]D-Fender | Google didn't tell you that. |
| 17:11.18 | [TK]D-Fender | DELL told you that. |
| 17:15.51 | *** join/#asterisk niek (niek@unaffiliated/niek) |
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| 17:26.18 | flujan | [TK]D-Fender: yeah⦠google points that link⦠so⦠I am doing it right with tcpenable and transport=udp, tcp I wanna to make it compatible with udp sip trunks. |
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| 17:55.35 | jpsharp | Is there an irc channel or other support for chan_dongle or should I just break out the candles and pray a lot? |
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| 18:04.07 | [TK]D-Fender | jpsharp: I doubt they have an actual IRC channel. For such a small project it'd be pretty damn small. |
| 18:04.16 | [TK]D-Fender | jpsharp: I'd start by asking your QUESTION |
| 18:06.07 | jpsharp | Okay. I have a E1550 dongle. I did some usb_modeswitch stuff on it. I only get one of the /dev/ttyUSB entries for it, rather than one for voice and one for data. |
| 18:06.29 | jpsharp | I probably did some wrong usb_modeswitch stuff on it and put it in the wrong mode. |
| 18:07.04 | jpsharp | I'm trying to figure out how to get it back in a chan_dongle friendly mode. |
| 18:07.10 | jpsharp | or into, rather than back into. |
| 18:08.51 | robmal | Plug it into a windows box. |
| 18:09.49 | monsterco | How can I fix this line to send Callerid Name as one variable? s,1,System(php /var/lib/asterisk/agi-bin/blackhole.php ${CALLERID(dnid)} ${CALLERID(num)} ${CALLERID(name)} ) |
| 18:12.50 | jpsharp | robmal: I figure that was the answer. The dongle is 5000+ miles away. |
| 18:15.01 | [TK]D-Fender | monsterco: Put it in quotes |
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| 19:13.50 | jpsharp | Seems that I was completely chasing my tail. The E1550 was properly initialized. The newly added E303 wasn't configured at all. it's talking now, but the SIM isn't active on the mobile carrier. |
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| 23:25.29 | karlfife | When Windows clients, or network-attached scanners are lookign for so-called 'network fax servers', is there a standard protocol that's typically used? Not likely T-38, but is it a standard thing? |
| 23:25.39 | karlfife | Is asterisk usable in this role? |
| 23:31.01 | *** join/#asterisk italorossi (~Adium@179.211.186.236) |
| 23:31.44 | [TK]D-Fender | as what? a "scanner", or as a "network fax server"? |
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| 23:46.09 | *** join/#asterisk leftleg (~JBurke@nsseofirewall.nsseo.org) |
| 23:46.16 | leftleg | hello all |
| 23:47.42 | leftleg | i have a simplex 5100 series pagin system from 2002. i'm using a cisoc spa 3102 to trunk to the paging system. has anyone used aligator clips on damaged 66 blocks? if you have do you have a link to where i could purchase aligator clips that are designed to connect to 66 blocks? |
| 23:48.14 | leftleg | i have a bug set that has these special clips, i don't know what they are called. |
| 23:48.19 | leftleg | bud set*** |
| 23:48.50 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 23:57.31 | karlfife | [TK]D-Fender: as a network fax server. We're already using Asterisk as a network fax server via AMI (&FFA). If it's straightforward to adopt another protocol, it would be useful. |
| 23:57.59 | [TK]D-Fender | Sounds like a "windows" thing. |
| 23:58.20 | [TK]D-Fender | there is no "use another computer as a fax server" protocol as a normal standard |
| 23:58.54 | karlfife | Windows servers can function in the rold of fax server, but it appears to be by way of hardware FXO modems. |
| 23:59.05 | [TK]D-Fender | karlfife, that's largely it |
| 23:59.07 | *** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com) |
| 23:59.41 | [TK]D-Fender | if you want something to put on the client side then look at an actual fax server solution like Hylafax |