| 00:03.24 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
| 00:03.30 | p3nguin | There is an option for MixMonitor to only record when the call is bridged. Make sure you are not using it. |
| 00:03.56 | eZz | I have no b there |
| 00:05.23 | ChannelZ | note that you're not going to get any ringing sounds in your recording over SIP of the ringing is not actually being transmitted/received as audio. Which SIP typically doesn't do. |
| 00:06.24 | ChannelZ | Call progress is like metadata, and the endpoint generally makes the ringing sound for the user locally. |
| 00:07.25 | [TK]D-Fender | I mentioned to use :r: but does not appear to have been done... |
| 00:07.34 | ChannelZ | yah |
| 00:07.51 | WIMPy | Ode doesen't exclude the other |
| 00:08.15 | WIMPy | Where is the beginning of that? |
| 00:08.16 | eZz | I tried 'r', nothing was changed |
| 00:08.30 | [TK]D-Fender | I'm not seeing dialplan to match, or a call |
| 00:13.00 | eZz | http://pastebin.com/iTNFUxYN |
| 00:15.18 | [TK]D-Fender | -- Executing [s@default:3] Hangup("Local/123@from-dialer-a57c;1", "") in new stack |
| 00:15.21 | [TK]D-Fender | Instant hangup |
| 00:15.25 | [TK]D-Fender | You're not looking at what you're doing |
| 00:15.49 | eZz | no |
| 00:15.51 | eZz | Wait("Local/123@from-dialer-a57c;1", "1") in new stack |
| 00:15.56 | eZz | after 1 second |
| 00:16.15 | [TK]D-Fender | Yeah, not enough to catch the rings let alone an entire call |
| 00:16.22 | [TK]D-Fender | 1 s may as well be instant |
| 00:16.43 | [TK]D-Fender | What is the point of a 1s call that hasn't had a chance to do anything? |
| 00:16.53 | [TK]D-Fender | Re-examine your approach |
| 00:17.02 | eZz | this is not I've expected. dialout-2 should be executed WHEN user WILL ANSWER |
| 00:17.16 | [TK]D-Fender | Answer <-- You answer |
| 00:17.24 | [TK]D-Fender | don't go thinking that "Dial" is everything |
| 00:17.28 | eZz | yes but w/o answer - I can't record a call |
| 00:17.33 | [TK]D-Fender | It exectues if ANYTHING answers |
| 00:17.43 | [TK]D-Fender | Change your approach |
| 00:17.48 | eZz | :-) |
| 00:18.24 | eZz | ok I need a full recording from Dial() moment to Hangup(). Not when a call will be bridged |
| 00:18.32 | [TK]D-Fender | Tip : you're recording the wrong leg of the call |
| 00:19.26 | eZz | there is 2:20AM to me, maybe I'm too tired to find an error in my brain :-D |
| 00:19.43 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 00:19.52 | WIMPy | That's normal. |
| 00:19.55 | [TK]D-Fender | Good odds |
| 00:20.10 | WIMPy | If you've got an error in your brain, you can't trust it. |
| 00:41.52 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 00:44.45 | dandate2 | so we're trying to transfer callers by using ## then a speed dial. will we see faster transfer performance by setting up speed dial in the ATA or the * server |
| 00:51.07 | eZz | oops, I got a brain code dump, need goto bed :( |
| 00:51.14 | eZz | *core |
| 00:51.22 | eZz | good night |
| 01:01.22 | [TK]D-Fender | "##" = Asterisk transfer. This means your call is always in prgress at the ATA and you can't do an ATA speed-dial no matter what |
| 02:02.47 | *** join/#asterisk kl4m (~klamontag@dsl-69-172-67-201.acanac.net) |
| 02:23.39 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 02:28.35 | weinerk | Hi. Please help: PSTN->DID->SIP->ASTERISK vs. SIP Client->Asterisk - rings until answer. |
| 02:28.36 | weinerk | PSTN takes at least 3 rings until the call hits the PBX (SIP Cli - instant) |
| 02:28.36 | weinerk | How can I make it instant/faster on PSTN? |
| 02:29.44 | Vicksters | honestly this looks like it just might be your provider's fault. |
| 02:29.51 | WIMPy | You need to tell us how your asterisk is connected to the PSTN first. |
| 02:31.19 | *** join/#asterisk GameGamer43 (~GameGamer@li119-60.members.linode.com) |
| 02:33.00 | weinerk | Vicksters: thanks, is there any way to check. Is there anything I can do about it? |
| 02:33.31 | [TK]D-Fender | weinerk: You need to answer WIMPy's question |
| 02:33.56 | weinerk | WIMPy: thanks, the setup: i bought DID that does an anonymous SIP call to pbx |
| 02:34.14 | weinerk | [TK]D-Fender: thanks for helping |
| 02:34.15 | *** join/#asterisk kl4m (~klamontag@dsl-69-172-67-201.acanac.net) |
| 02:35.59 | WIMPy | I have no clu how they might do it, but I go with Vicksters. |
| 02:37.05 | [TK]D-Fender | weinerk: Show us the call |
| 02:37.46 | [TK]D-Fender | weinerk: Well if it rings before your PBX gets the call, then the delay is on their side and you have to ask what they can do about it. |
| 02:38.38 | p3nguin | Or the PBX could be causing the delay and ringing. |
| 02:39.15 | weinerk | [TK]D-Fender: on the pbx I did asterisk -rvvvvvvvvvvvv |
| 02:39.15 | weinerk | then I dial from a pstn - it rings approx 3 rings 10 seconds - |
| 02:39.15 | weinerk | then I see the call hit the pbx. |
| 02:39.15 | weinerk | But for example if you dial ATT 1800-callatt - you dont have to wait for three rings - it answers right away |
| 02:39.16 | weinerk | <PROTECTED> |
| 02:39.25 | WIMPy | Only if the description was wrong. |
| 02:39.58 | weinerk | If I have a SIP client registered with pbx - no rings - instant answer |
| 02:41.49 | weinerk | Thanks guys for your help. So guess the concensus is that I have to fix the provider or get a better provider? |
| 02:41.51 | [TK]D-Fender | weinerk: the delay is starting at your provider then. See what they can do about it |
| 02:42.04 | weinerk | thanks! |
| 02:42.09 | p3nguin | What happens if your PBX is offline and you call the DID from a mobile phone? |
| 02:42.45 | weinerk | the provider allows up to 90 seconds to connect the call |
| 02:42.49 | p3nguin | Three rings and then temp file? |
| 02:42.52 | p3nguin | s/file/fail/ |
| 02:44.14 | weinerk | I am not sure what happens if pbx does not answer within 90 seconds |
| 02:44.39 | weinerk | until 90 seconds that they allow - it will ring (approx 30 rings i guess) |
| 02:45.08 | weinerk | then probably go to busy. |
| 02:45.39 | WIMPy | doesn't think that assumptions will be of any help. |
| 02:46.28 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 02:46.53 | weinerk | I got it - sounds like the only options it to look to provider or get someone else who does it faster (PSTN->SIP) |
| 02:49.07 | [TK]D-Fender | weinerk: I've never seen a real provider with any delay in handing off a call |
| 02:49.25 | p3nguin | Anyone know of something like a music emporium where I could call them up and ask them about a particular song to try to find out what album it is on? The interweb has failed to come through for me. |
| 02:50.11 | p3nguin | It seems to be a very obscure song. |
| 02:51.48 | weinerk | [TK]D-Fender: I am using callwithus.com - I am trying to follow up with them? |
| 02:52.26 | [TK]D-Fender | p3nguin: www.midomi.com |
| 02:52.53 | [TK]D-Fender | p3nguin: Or do you already know what it is and jsut need the album name? |
| 02:53.15 | [TK]D-Fender | p3nguin: Normally CCDB has that for that vast majority of registered discs |
| 02:54.25 | p3nguin | I found one single reference with the lyrics listed on google, and there is a potential title and artist. I have searched for the lyrics alone, the lyrics with the artist name, the lyrics with the song name... only to find that one single post again. |
| 02:54.41 | [TK]D-Fender | What fragment do you have? |
| 02:54.57 | p3nguin | http://www.rapworlds.com/forums/archive/index.php/t-152664.html |
| 02:55.30 | p3nguin | Allegedly by Ja Rule, allegedly named Broadcasting Communications. |
| 02:56.20 | p3nguin | I've tried new searches with lyric fragments, with quoted bits, etc, just to find nothing more than this one post in a forum. |
| 02:58.05 | p3nguin | oh, "Broadcasting Communication" without the s. My mistake. |
| 02:58.19 | [TK]D-Fender | p3nguin: So that post really is the song itself right? More than enough to match your memory? |
| 03:01.37 | p3nguin | That's kind of a problem; I personally do not remember any of it. An electronic note with some of the lyrics was written when the song was playing. I expected that note to be permanent, but after I was not able to find the song by the original name I was given, I asked to read the note. Apparently the note was deleted for some unknown reason, so I don't have much to go on right now. |
| 03:02.20 | p3nguin | This is a case of "If I wanted it done right, I should have done it myself." |
| 03:05.06 | carrar | So much hate in that song |
| 03:05.13 | *** join/#asterisk kl4m (~klamontag@dsl-69-172-67-201.acanac.net) |
| 03:06.35 | [TK]D-Fender | p3nguin: Nothing coming up indeed... |
| 03:07.43 | p3nguin | It is so obscure that I finally got to the point where I still want to hear that song even if it isn't the one I'm supposed to be trying to find info on. |
| 03:09.31 | *** join/#asterisk bbourdage (~bbourdage@75-146-129-225-SouthBend.hfc.comcastbusiness.net) |
| 03:09.43 | p3nguin | I think I just need to call an emporium where they are well-educated in hip-hop and talk to someone about it. |
| 03:11.13 | [TK]D-Fender | Well You don't seem to have anything to go on as you doubt that the only thing you did find was even it... |
| 03:11.22 | [TK]D-Fender | Where did you hear it? Noone you can ask there? |
| 03:12.37 | p3nguin | It was at a club, and I might be able to call up the DJ there to see if he can give me any info. |
| 03:13.18 | [TK]D-Fender | Yup, best odds. |
| 03:22.45 | ChannelZ | "Do you know which hip-hop song it is that has 'mother fucker' and 'nigga' and cop killing and stuff in the lyrics?" |
| 03:23.10 | p3nguin | "Uhm, yes... all of them." |
| 03:23.52 | ChannelZ | Exactly |
| 03:27.20 | p3nguin | I intend to take a different approach than that, hoping to get a better result than, well, that. |
| 03:29.38 | ChannelZ | "Oh, the artist doesn't know the difference between a clip and a magazine either." |
| 03:29.45 | ChannelZ | no, wait.. that doesn't narrow it down |
| 03:36.48 | [TK]D-Fender | Yes, Top-class gangstas actually know how to handle a gun... |
| 03:39.18 | cstachris | does asterisk-1.8 raise the callgroup/pickupgroup limit from 64? |
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| 04:54.48 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 04:59.35 | autofsckk | night everyone, i have to recover some info from a ntfs partition, i did a backup but it didnt work, and i find out after i already format and installed win7 on that partition, is there a way to recover some very important info from there? |
| 05:02.42 | [TK]D-Fender | autofsckk: http://www.google.ca/#hl=en&cp=21&gs_id=26&xhr=t&q=recover+data+from+formatted+drive&pf=p&sclient=psy-ab&source=hp&pbx=1&oq=recover+data+from+for&aq=0&aqi=g4&aql=&gs_sm=&gs_upl=&bav=on.2,or.r_gc.r_pw.r_cp.,cf.osb&fp=6985811cc01f9710&biw=1920&bih=1114 |
| 05:05.54 | autofsckk | thanks |
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| 06:23.06 | *** join/#asterisk prasanthprabalan (~root@218.248.24.19) |
| 06:23.15 | prasanthprabalan | Hi all |
| 06:23.47 | prasanthprabalan | How can i forward a call to another line if called number is busy |
| 06:24.16 | prasanthprabalan | I have done like this: |
| 06:24.21 | prasanthprabalan | exten => 5002,n,Dial(SIP/prasanth,20) |
| 06:24.21 | prasanthprabalan | exten => 5002,n,Dial(SIP/sai,10) |
| 06:24.41 | prasanthprabalan | but this will forward the call after 20 seconds only |
| 06:25.07 | prasanthprabalan | Is it possible to transfer at the instance if line is busy |
| 06:26.02 | prasanthprabalan | Hi plz help |
| 06:35.13 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 06:39.04 | prasanthprabalan | hi any help plz |
| 06:39.51 | cstachris | use ${DEVSTATE} to find out the status of the device you want to call |
| 06:40.31 | cstachris | http://www.voip-info.org/wiki/view/Asterisk+func+device_State |
| 06:42.41 | prasanthprabalan | Ok let me try that |
| 06:42.46 | prasanthprabalan | Thanks for the reply |
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| 08:26.44 | jzaw | morning |
| 08:43.16 | jacc0 | morning |
| 08:53.53 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:54.08 | jzaw | hi |
| 08:54.21 | jzaw | any one use gtalk/jabber/jingle ? |
| 08:54.42 | *** join/#asterisk Nasga (~Nasga@239.212.10.93.rev.sfr.net) |
| 08:54.50 | jzaw | i can easily get my gmail jid to accept calls and dial out via gtalk |
| 08:55.29 | jzaw | and it should be similar for my xmpp jid over gtalk or jingle ... but it just wont work |
| 09:03.40 | ChannelZ | I use google talk |
| 09:04.58 | ChannelZ | But not really sure what you're asking, what the differentiation is |
| 09:06.08 | *** join/#asterisk coppice (~chatzilla@globbits.tripleone.co.uk) |
| 09:13.48 | jzaw | ChannelZ: hi |
| 09:13.53 | jzaw | sorry for the delay |
| 09:14.05 | jzaw | i was preparing some pastebin stuff to show you |
| 09:14.44 | jzaw | i have a user ..... jzaw@dzki.co.uk and i run my own xmpp server ... ejabberd |
| 09:15.50 | jzaw | jzaw@dzki.co.uk can receive calls fine from any other jabber or gtalk user (at least my friends can call fine |
| 09:16.10 | jzaw | my user at gmail.com ... can receive calls too |
| 09:16.14 | jzaw | and can make calls |
| 09:16.26 | jzaw | <PROTECTED> |
| 09:16.47 | jzaw | i can pick up a phone dial 601 and i speak to someuser |
| 09:17.04 | jzaw | but i cant do similar with jzaw@dzki.co.uk |
| 09:17.14 | jzaw | <PROTECTED> |
| 09:17.16 | jzaw | doesnt work |
| 09:18.19 | jzaw | where asterisk_jingle = jingle/jzaw_dzki which is what ive called both the context and the connection in gtalk.conf |
| 09:19.22 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:21.01 | jzaw | ChannelZ: |
| 09:21.02 | jzaw | http://pastebin.com/6Qa3veA4 |
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| 09:24.07 | jzaw | ive also tried gtalk/jzaw_dzki in the DIAL() |
| 09:25.44 | *** join/#asterisk s[X] (~s_x_@ppp59-167-157-96.static.internode.on.net) |
| 09:25.48 | jzaw | ive also tried putting the jzaw_dzki in its own jingle.conf |
| 09:27.54 | jacc0 | test if your Cisco phone/ata cazn be remote controled without the need of authentication : www.securitysource.eu |
| 09:28.15 | jacc0 | I've added a war-dialer and a factory defaults button :) |
| 09:28.29 | jacc0 | have fun!!\ |
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| 10:52.57 | qakhan | i setup directory on my asterisk and its working very well |
| 10:54.27 | qakhan | but when ivr specks person name and number, it speacks alphabets and digit. |
| 10:55.11 | qakhan | like j-a-m-e-s ext 1-0-0-4 |
| 10:56.02 | qakhan | i want to setup. it speck person name in words, like james and ext 1004 |
| 10:56.17 | qakhan | anyone can tell me how i can do this |
| 10:58.23 | jzaw | qakhan: take a look at text to speech |
| 10:59.13 | jzaw | http://www.google.co.uk/search?q=asterisk+text+to+speech |
| 10:59.34 | kaldemar | qakhan: you need a third party text to speech application such as festival, flite, espeak or swift. asterisk will not do that for you. |
| 11:00.10 | qakhan | are these open source? |
| 11:00.49 | ketas | imo all |
| 11:00.57 | jzaw | or record your own .... how many names are in the directory? |
| 11:01.16 | qakhan | around 100 user |
| 11:01.44 | ketas | having tts phonebook sounds bad |
| 11:02.14 | jzaw | id haev to agree tbh |
| 11:02.16 | jzaw | have |
| 11:02.32 | qakhan | kaldemar are these softwares supports directory application? |
| 11:08.08 | ketas | those are plain tts solutions, you need to put something together to get it working |
| 11:08.10 | kaldemar | qakhan: they have nothing to do with directories. they are only text-to-speech. how they bind with your directory application is up to you to implement. |
| 11:10.19 | qakhan | kaldemar i dont know about it. can u please guide me how to do that |
| 11:18.06 | Faustov | any idea where I could find a 100% coherent dialplan format for covering all landlines and mobiles in spain? |
| 11:29.27 | kaldemar | qakhan: without knowing what your directory is, no. |
| 11:30.57 | qakhan | i have person name and their exts |
| 11:33.11 | kaldemar | you have those in dialplan? feed them to a TTS application. those forementioned TTS engines bind to asterisk with a dialplan application. |
| 11:34.50 | *** join/#asterisk iMelnik (~melnik@217.147.17.74) |
| 11:37.07 | iMelnik | Hello. I need to connect 4 E1 ports to asterisk box. 2 are PRI_NET and 2 PRI_CTE. Should I use 2 Digium TE220 or 1 TE420? Which is more stable? |
| 11:41.24 | qakhan | kaldemar i have directory application in dialplan exten => 1,1,Directory(abd,internal,ef) |
| 11:41.55 | qakhan | how these TTS application access directory |
| 11:45.28 | kaldemar | oh, that directory. they can't access it. |
| 11:45.52 | qakhan | so what is the sulotion |
| 11:46.00 | qakhan | solution* |
| 11:49.20 | kaldemar | something else as a directory or modify play_mailbox_owner() in app_directory. |
| 11:51.09 | qakhan | where app_directory place? |
| 11:53.31 | Faustov | what could be the cause of a loud buzzing sound on a call between two hardphones, each behind it's own asterisk server, reproducible always? |
| 11:54.32 | Faustov | version 1.8.7.2 |
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| 11:55.17 | kaldemar | qakhan: app_directory.c in the source package. |
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| 12:13.46 | Faustov | I've ruled out networking problems, got a lot of bandwidth |
| 12:15.37 | Faustov | ok got it, broken handset |
| 12:15.40 | Faustov | heh |
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| 12:23.15 | Faustov | ok, a different question - iirc call-limit=X in sip.conf was supposed to disappear in 1.8, but it still works and there is even no warning - change of plans? |
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| 12:30.42 | MarKsaitis | Network A has 1 WAN IP address and loads of internal computers/ips under NAT. We want to use a service which requires incoming port 4444 open. That service needs to be used by multiple computers in the network. As I understand, port 4444 can only be forwarded to only 1 computer inside the network at any one time. Is that correct thinking? |
| 12:31.17 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 12:31.51 | Faustov | MarKsaitis: yes, you can only forward ports to a single destination:port |
| 12:32.33 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 12:32.34 | Faustov | MarKsaitis: note you DON'T need port forwarding (for sip, iax or rtp) if your asterisk is facing both interfaces, LAN and WAN |
| 12:35.40 | *** join/#asterisk moy (~moy@173.239.155.74) |
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| 12:40.12 | *** join/#asterisk reber (~reber@cl-157.dub-01.ie.sixxs.net) |
| 12:40.15 | reber | hi |
| 12:40.26 | reber | what is FXS port please ? |
| 12:41.40 | tully` | reber, http://www.3cx.com/PBX/FXS-FXO.html |
| 12:43.59 | reber | tully`, interesting thanks. And then i have a spa3102, and i don't know what i could do of this FXS port. Can i plug it on the wall jack to receive phone calls from the wall jack ? |
| 12:45.19 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 12:47.49 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
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| 13:15.15 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:18.09 | jzaw | ping ChannelZ |
| 13:18.48 | *** part/#asterisk iMelnik (~melnik@217.147.17.74) |
| 13:20.37 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:20.58 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 13:21.56 | *** join/#asterisk Squeeb (~meow@host81-149-117-179.in-addr.btopenworld.com) |
| 13:22.45 | Squeeb | Hi, I'm trying to install the Digium B410 quad-BRI card, I followed the manual closely, configured the spans correctly for my country but when I run dahdi_cfg -vv I get the following error: |
| 13:22.52 | Squeeb | DAHDI_SPANCONFIG failed on span 1: No such device or address (6) |
| 13:23.06 | Squeeb | I've checked that the wcb4xxp module is loaded, which it is |
| 13:23.22 | Squeeb | I've searched google for the error but just keep finding the same post over and over again |
| 13:23.35 | Squeeb | which is just some guy asking the same question |
| 13:28.59 | leifmadsen | Squeeb: when you purchase Digium hardware you do get support with it |
| 13:29.14 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:29.16 | Squeeb | For ever? |
| 13:29.19 | Squeeb | this card is a few years old |
| 13:30.46 | *** join/#asterisk DarkRift (~dark@unaffiliated/darkrift) |
| 13:36.56 | kaldemar | Squeeb: do you see the card with lspci? d161:b410 |
| 13:37.44 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 13:39.33 | Squeeb | yes |
| 13:42.06 | kaldemar | Squeeb: pastebin your system.conf |
| 13:44.36 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:44.53 | tzafrir_laptop | Squeeb, the config dahdi_genconf generates for it should be good enough (for dahdi_cfg to be satisfied) |
| 13:45.20 | tzafrir_laptop | But do you actually have span 1? What's the output of lsdahdi ? |
| 13:47.26 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.117.41) |
| 13:51.10 | Squeeb | it just says device cannot be found again |
| 13:51.16 | Squeeb | well No such device or address |
| 13:51.36 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 13:51.43 | Squeeb | I'm starting again with the config using CentOS instead of Debian now |
| 13:51.45 | elliot98 | politely enteres |
| 13:51.53 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 13:52.10 | elliot98 | how is it possible to change indications (ring-tones) per channel? |
| 13:54.02 | leifmadsen | elliot98: 'r' option |
| 13:54.04 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 13:54.23 | leifmadsen | elliot98: 1.6.2 and later lets you indicate which tones to play while ringing |
| 13:54.28 | elliot98 | leifmadsen: will that also work if the line is busy? |
| 13:54.56 | elliot98 | leifmadsen: there may be tones other than ring that need to be changed. |
| 13:55.17 | leifmadsen | shrugs |
| 13:55.19 | leifmadsen | you'll have to try |
| 13:55.35 | elliot98 | wondering if the indications is per channel or global |
| 13:58.32 | [TK]D-Fender | it should be channel selectable, though I don't recall ever actually seeing it... |
| 14:04.57 | *** join/#asterisk gaetronik (~gaetan@ks370400.kimsufi.com) |
| 14:05.22 | *** join/#asterisk serafie (~erin@nat/digium/x-eeytfguggbsnwvuh) |
| 14:05.57 | elliot98 | [TK]D-Fender: interesting, perhaps a setindication function can be implemented, similar to settings the language |
| 14:06.21 | [TK]D-Fender | elliot98, It should be a parm for CHANNEL() |
| 14:06.37 | gaetronik | !trunk |
| 14:06.47 | gaetronik | Hi |
| 14:06.53 | gaetronik | ~trunk |
| 14:06.53 | infobot | trunk is probably a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
| 14:07.15 | gaetronik | ~help |
| 14:07.20 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-xvvmriybcuimqfta) |
| 14:07.30 | leifmadsen | gaetronik: you can also msg infobot directly |
| 14:07.33 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:07.41 | leifmadsen | mjordan: o/ |
| 14:08.06 | gaetronik | leifmadsen: it's what i'm doing |
| 14:08.23 | leifmadsen | gaetronik: not quite.... /msg infobot <command> |
| 14:08.26 | gaetronik | but i did not remember the name of the bot |
| 14:08.55 | gaetronik | and ~help was a wrong chan error |
| 14:09.20 | gaetronik | sorry for disturbing |
| 14:09.27 | leifmadsen | I didn't say you were |
| 14:09.55 | mjordan | leifmadsen: 'ello |
| 14:10.10 | leifmadsen | mjordan: another day, another doll hair |
| 14:10.17 | mjordan | ew |
| 14:11.12 | *** join/#asterisk moobius (~mobius@static062038158036.dsl.hol.gr) |
| 14:11.18 | Squeeb | does dahdi_genconf need anything before running it? |
| 14:11.31 | Squeeb | do I have to plug the cables into the BRI card? |
| 14:11.56 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 14:13.25 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 14:13.35 | WIMPy | Squeeb: no |
| 14:13.47 | Squeeb | cool |
| 14:13.56 | Squeeb | how does it know about the spans? |
| 14:14.55 | WIMPy | It wo'nt knwo everything. |
| 14:15.05 | Squeeb | ? |
| 14:16.04 | WIMPy | It will generate a valid config that may or may not fit your situation. |
| 14:17.16 | WIMPy | IIRC newer version do detect TE or NT mode but eveything else has to be configured manually. |
| 14:17.43 | Squeeb | so I still need to define the bchans,dchans myself? |
| 14:17.54 | WIMPy | no |
| 14:18.06 | Squeeb | oh |
| 14:18.09 | Squeeb | so it'll pick those up? |
| 14:18.23 | *** join/#asterisk hudony (~chatzilla@modemcable078.217-70-69.static.videotron.ca) |
| 14:18.23 | Squeeb | There's really not much to this B410 manaul :/ |
| 14:18.25 | Squeeb | Manual * |
| 14:18.47 | WIMPy | Yes, but it won't go further than that. |
| 14:18.59 | Squeeb | heh. well that'd be a nice start I guess : ) |
| 14:19.03 | Squeeb | 03:01.0 0204: d161:b410 (rev 01) |
| 14:19.09 | Squeeb | this is in, so I guess it *should work ( |
| 14:20.16 | hudony | hi, I have a very weird issue: I have about 15 phones connected to an asterisk box and few days ago, 3 phones decided that one of their line could not register anymore. I've double checked username and passwords and its all good. It just stopped working... |
| 14:20.27 | hudony | using spa504g |
| 14:20.32 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 14:20.43 | jacc0 | lol @spa504g |
| 14:21.01 | hudony | sip show peers display UNKOWN for the specific line I'm trying to troubleshoot |
| 14:21.03 | jacc0 | it has a secury hole so big that there fits an airplane in there |
| 14:21.03 | Squeeb | hudony: pastebin your 'sip show registry' 'sip show peers' and whatno |
| 14:21.05 | Squeeb | oh |
| 14:21.18 | Squeeb | can you ping the phones from the asterisk server? :} |
| 14:21.29 | hudony | sure...the phone has 4 lines |
| 14:21.32 | hudony | 3 of them are working fine |
| 14:21.39 | jacc0 | check your if your SPA phone has this security hole using the tool I made: www.securitysource.eu |
| 14:21.42 | hudony | jlabonte-1 (Unspecified) D N 0 UNKNOWN |
| 14:21.44 | hudony | jlabonte-2/jlabonte-2 192.168.2.185 D N 5061 OK (8 ms) |
| 14:21.45 | hudony | jlabonte-3/jlabonte-3 192.168.2.185 D N 5062 OK (8 ms) |
| 14:21.47 | hudony | jlabonte-4/jlabonte-4 192.168.2.185 D N 5063 OK (8 ms) |
| 14:21.53 | hudony | jacc0: I will |
| 14:22.08 | WIMPy | ~pb |
| 14:22.08 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 14:22.11 | hudony | The only thing I can think of are the credentials but they are good |
| 14:22.13 | jacc0 | please tell me if the exploit works; I haven't tested it agains SPA504gb |
| 14:22.50 | hudony | Sorry for the flood..tough max was 5 :S |
| 14:22.51 | gaetronik | i'm looking for a decent portuguese voip provider |
| 14:23.09 | anonymouz666 | PORTUGUESE? |
| 14:23.10 | Squeeb | gaetronik: google.com |
| 14:23.17 | anonymouz666 | did i read this right? |
| 14:23.36 | gaetronik | Squeeb: for the feedback it's not the better option |
| 14:23.36 | Squeeb | hudony: might be worth recording the sip debug for a bit |
| 14:23.53 | gaetronik | anonymouz666: did i mispell? |
| 14:24.03 | Squeeb | http://www.voipproviderslist.com/country/voip-portugal/voip-providers-portugal/ |
| 14:24.06 | Squeeb | enjoy |
| 14:24.08 | hudony | Well...I've looked into it and didn't see a clear explanation for the issue :( |
| 14:24.31 | gaetronik | Squeeb: all dead in the list |
| 14:24.43 | Squeeb | gaetronik: might have to use a european breakout service then |
| 14:24.44 | gaetronik | sorry |
| 14:24.49 | Squeeb | sipgate may have portugese numbers |
| 14:24.53 | Squeeb | worth a check anyway |
| 14:25.00 | gaetronik | i check the wrong url |
| 14:25.04 | gaetronik | yours seems fine |
| 14:25.15 | Katty | drags in |
| 14:25.46 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 14:25.55 | gaetronik | thnks Squeeb |
| 14:27.20 | *** join/#asterisk nighty^ (~nighty@static-68-179-124-161.ptr.terago.net) |
| 14:27.42 | anonymouz666 | gaetronik: I only know brazillian voip provider, but it is always good to see portuguese people |
| 14:28.00 | gaetronik | anonymouz666: i'm not portuguese |
| 14:28.14 | gaetronik | but the company i work in as an office in Lisboa |
| 14:28.27 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 14:28.38 | hudony | What can stop a line from registering to asterisk (except wrong username/password) if 3 can and only 1 can't on the same phone? |
| 14:28.40 | hudony | :S |
| 14:28.50 | hudony | it's not a network issue |
| 14:29.00 | hudony | i'm lost |
| 14:29.52 | hudony | I did a vimdiff on 2 xml and only credentials are differents and I can tell that they match what is in sip.conf so the phone configuration seems just fine |
| 14:29.54 | Squeeb | core set verbose 100 |
| 14:29.59 | Squeeb | try that and reset your phone |
| 14:30.01 | Squeeb | see what's going on |
| 14:30.02 | hudony | ok |
| 14:30.16 | Squeeb | also there's a lines = option in the sip conf |
| 14:30.20 | Squeeb | i can't remember what that's for |
| 14:31.50 | hudony | phone rebooted : Saved useragent "Cisco/SPA504G-7.4.3a" for peer jlabonte-2 |
| 14:31.54 | hudony | also got this for line 3 and 4 |
| 14:31.58 | hudony | nothing for line 1 |
| 14:32.00 | hudony | :S |
| 14:33.29 | [TK]D-Fender | hudony, You aren't showing something real to advise to on. |
| 14:33.46 | [TK]D-Fender | you* |
| 14:34.11 | hudony | which means? |
| 14:34.14 | hudony | What do you need |
| 14:35.10 | [TK]D-Fender | hudony, You haven't shown us your phone trying to conect * in reference to that 1st "line" |
| 14:35.21 | [TK]D-Fender | hudony, So there is no "failure" to examine. |
| 14:35.38 | hudony | well... there is not 1st line at all |
| 14:35.45 | hudony | there was 3 lines |
| 14:35.55 | [TK]D-Fender | hudony, We don't see your phone configs to determine if we think they're right either |
| 14:36.16 | hudony | hold on |
| 14:36.16 | [TK]D-Fender | hudony, Where's the problem? |
| 14:36.54 | hudony | sip show peers tells me the phone cannot register but when I boot it with verbosity set to max...there is nothing about the first line failling to register |
| 14:37.44 | [TK]D-Fender | "sip show peers" does't tell you the phone can't register. |
| 14:38.17 | *** join/#asterisk DennisG (~dennisg@541AFD1E.cm-5-3d.dynamic.ziggo.nl) |
| 14:38.25 | [TK]D-Fender | Who said it was failing? Where is the failure? |
| 14:39.00 | hudony | Well.. I assume it was the case from what I could find by googling |
| 14:39.05 | hudony | assumed* |
| 14:39.23 | [TK]D-Fender | Google has no clue what your phone is doing. * does. You should be looking at the SIP debug |
| 14:39.48 | hudony | http://pastebin.com/NueknxBD |
| 14:39.55 | hudony | The phone configuration in sip.conf |
| 14:39.57 | hudony | ok |
| 14:40.00 | [TK]D-Fender | Forget configs |
| 14:40.16 | [TK]D-Fender | Go prove what your phone is asking * for in the first place |
| 14:40.35 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 14:40.38 | [TK]D-Fender | Then we can see if there is an issue with what you believe it should match |
| 14:40.42 | hudony | ok |
| 14:43.39 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 14:47.54 | hudony | ok |
| 14:47.58 | hudony | got the sip debug log |
| 14:48.27 | hudony | if i search for jlabonte-1 (which is my "faulty" line), I find nothing for today, only few days ago when the line was working |
| 14:48.35 | hudony | Do you want me to show you a pastebin? |
| 14:48.56 | hudony | I see however a lot of stuff for the other 3 lines |
| 14:49.11 | [TK]D-Fender | Show me the actual failure. Live. From now. |
| 14:50.25 | hudony | sorry for looking retarded but I'm not sure what you *exactly* want me to do |
| 14:50.55 | [TK]D-Fender | hudony, Show me the precise registration attempts for this line that is supposedly geting rejected. |
| 14:52.44 | hudony | Here is what I done : modified logger.conf to log to messages everything. I then launch asterisk console and core set verbose 100 + sip set debug on. I then reloaded asterisk. Then I rebooted the phone itself. I waited till it was up then stopped the logging. I opened the file with vim, went to the end with shift+G then use the backward find like this : ?jlabonte-1 |
| 14:52.56 | hudony | The first thing I met was when the line was alive jan 20th |
| 14:53.13 | hudony | So I see nothing for the registration attempt |
| 14:53.23 | hudony | So I guess no registration attempt took place |
| 14:53.56 | hudony | I however see all other lines trying and succeeding registration : jlabonte-2, jlabonte-3 and jlabonte-4 |
| 14:53.59 | hudony | Hope it helps |
| 14:54.10 | *** join/#asterisk eZz (~ez@195.114.6.134) |
| 14:54.11 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 14:55.29 | *** join/#asterisk fhmiv (~fhmiv@c-24-23-87-52.hsd1.ga.comcast.net) |
| 14:55.44 | [TK]D-Fender | hudony, It doesn't. You're talking about the past from a log instead of "Live" and "From now" like I explicitly asked for. Nor are you actually showing me anything. You are giving me your interpreted sumamry. |
| 14:55.48 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 14:55.49 | [TK]D-Fender | summary |
| 14:56.33 | [TK]D-Fender | * SIP debug. Live. No logs. No ancient history. Only what the phone is actually asking *. |
| 14:57.51 | hudony | I actually logged it to be able to search trough it since console doesn't allow it |
| 14:58.01 | hudony | now, i have sip set debug on on console |
| 14:58.47 | [TK]D-Fender | Search after. |
| 14:59.12 | [TK]D-Fender | restart phone. Wait. Copy entire buffer. Paste and look. |
| 14:59.27 | hudony | ok |
| 15:01.32 | jsjc | Hello, I got my debuggins and I am getting some isseus in between to SIP users because they do not hear each others voice… http://pastebin.com/Zt7g1JSj |
| 15:01.39 | gaetronik | Is anyone yet thought about a log anonymiser |
| 15:02.00 | gaetronik | to get off public ip, username, phone numbers? |
| 15:02.34 | jsjc | gaetronik: that will benice |
| 15:02.44 | jsjc | beccause I was now lazy of taking all away hehe. |
| 15:03.28 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.104.126) |
| 15:05.09 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 15:06.30 | hudony | http://pastebin.com/MQN3Fp8v |
| 15:06.36 | hudony | I'm looking through it |
| 15:06.58 | hudony | but I don't see anything |
| 15:07.03 | hudony | usefull |
| 15:08.27 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:08.27 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:11.13 | [TK]D-Fender | hudony, No, the complete lack thereof is useful. It says your phone isn't even trying. Fix your phone |
| 15:12.44 | hudony | ok so according to you, I have 3 faulty phone |
| 15:13.17 | [TK]D-Fender | hudony, You mentioned 1 single registration. |
| 15:13.30 | hudony | yes...indeed but I have 3 phones with the same problem |
| 15:13.36 | hudony | 1 of their 4 line isnt registering |
| 15:13.40 | [TK]D-Fender | 3 are "ok", one you first claimed "failed" and from SIP debug looks more like "never tried" |
| 15:14.11 | [TK]D-Fender | hudony, And I never said the phone was faulty. It just isn't asking what you think it shuold. |
| 15:14.35 | [TK]D-Fender | hudony, Naturally the first thing that comes to mind is PEBKAC |
| 15:14.35 | hudony | oh ok |
| 15:14.55 | hudony | :S |
| 15:15.09 | *** join/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 15:16.08 | hudony | The only thing I can see right now is that line1 could be disabled since the option exists on the phone |
| 15:16.34 | hudony | But when I compared 2 xml config files (one for a working one and one with my "problem"), they are identical |
| 15:16.38 | hudony | except for the credentials |
| 15:16.51 | hudony | So I guess I'll have to digg deeper |
| 15:17.12 | hudony | But to be honest, I have no clue where to search |
| 15:18.02 | jsjc | what could be the reason of no voice on a call between two sip phones connected to asterisk? |
| 15:19.10 | [TK]D-Fender | jsjc, improper NAT setup, lack of ports being open /forwarded. Other generic networking. SIP ALG that twists addresses wrong. Reinvites where they can't be supported. And most importantly, sun-spots |
| 15:20.04 | jsjc | Thanks [TK]D-Fender this is going to be a nightmare for debugging it then… Thanks. |
| 15:20.40 | hudony | Well [TK]D-Fender thank you for your help and time |
| 15:20.59 | hudony | I'll try my best to find out why about the issue |
| 15:21.18 | hudony | At least.. I now know for sure they the line isn't trying to register |
| 15:21.24 | hudony | I guess it's a start |
| 15:21.53 | hudony | Have a good day |
| 15:24.44 | [TK]D-Fender | jsjc, That to say you have lots of hostile routers in your path? |
| 15:24.57 | jsjc | I have… just 1 router. |
| 15:25.15 | jsjc | SIP phone — ASterisk — Internet — Router — Sip Phone |
| 15:25.28 | jsjc | between the inside SIP phones on the left side of the internet not problems |
| 15:25.39 | jsjc | from a dahdi phone on the left to the right SIP phone no problem |
| 15:26.02 | [TK]D-Fender | jsjc, what router? what services is it doing for you? Show us what you have forwarded, dump the firewall on our server itself |
| 15:26.35 | jsjc | the problem comes when sip one from the other…. |
| 15:28.06 | jsjc | the router on the right side it is just the Internet/modem/router of the client on the left there is an asterisk connected to the internet trough a router (no filtering anything) |
| 15:28.06 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 15:28.35 | jsjc | could it be possible that the firewall on the same machine as asterisk is filtering something? |
| 15:28.44 | [TK]D-Fender | yes |
| 15:28.50 | [TK]D-Fender | Your * is public? |
| 15:29.30 | [TK]D-Fender | If a remote phone is behind NAT then you don't normally need to forward anything. What are they using? |
| 15:36.13 | *** join/#asterisk Vicksters (Vicksters@modemcable142.35-177-173.mc.videotron.ca) |
| 15:41.37 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 15:48.12 | jsjc | Remote phone it is behind NAT… they are using a normal router form their internet provider. |
| 15:52.20 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:54.28 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:55.25 | Squeeb | Any reason why "dialplan reload" doesn't seem to take an effect |
| 15:55.29 | Squeeb | Asterisk 1.8.8.2 |
| 15:55.54 | ChannelZ | "doesn't seem to"? |
| 15:55.59 | Squeeb | well, doesn't |
| 15:56.04 | ChannelZ | Does it look like it's reloading, or it says nothing, or..? |
| 15:56.05 | Squeeb | I edit extensions.ael |
| 15:56.09 | Squeeb | I do dialplan reload |
| 15:56.14 | Squeeb | and my old extensions still exist |
| 15:56.18 | ChannelZ | oh.. you want ael reload |
| 15:56.21 | ChannelZ | (I think?) |
| 15:56.23 | Squeeb | aah |
| 15:56.29 | Squeeb | perfect, ta |
| 15:56.33 | Squeeb | first time using 1.8 |
| 15:57.54 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 15:58.00 | *** join/#asterisk serafie (~erin@nat/digium/x-bnacqvjbidaqqrsq) |
| 15:58.28 | Ice_Strike | I want to develop a Call Center website so the agents can use the system on the website. Which PHP Libabry I should use? |
| 15:58.31 | jsjc | [TK]D-Fender: In case the remote phone it is not behind NAT what I would need to redirect? |
| 15:58.48 | [TK]D-Fender | Ice_Strike, ... |
| 15:58.52 | Squeeb | lolwut? |
| 15:58.53 | [TK]D-Fender | ~toywy |
| 15:58.53 | infobot | hmm... toywy is The one you write yourself. |
| 15:59.08 | [TK]D-Fender | jsjc, Nothing. I asked 3 other things you didn't asnwer. |
| 16:00.28 | Ice_Strike | [TK]D-Fender yes? |
| 16:01.38 | Squeeb | anybody seen this? |
| 16:01.38 | Squeeb | [Jan 23 16:10:08] ERROR[873]: chan_dahdi.c:13991 dahdi_pri_error: PRI Span: 1 Unable to receive TEI from network in state 3(Establish awaiting TEI)! |
| 16:01.52 | Squeeb | Not sure what it means, google doesn't seem to either |
| 16:01.59 | [TK]D-Fender | <infobot> hmm... toywy is The one you write yourself. <-------- |
| 16:05.03 | jsjc | [TK]D-Fender: I am sorry didnt saw your question. The asterisk it is public. The remote client it is behing an internet router froma internet provider. There is a firewall in the asterisk machine but it is not blocking anything related to asterisk... |
| 16:06.02 | [TK]D-Fender | jsjc, Make sure canreinvite/directmedia = no for both, nat=yes in general isn't a bad idea (except for ITSP entries) |
| 16:06.42 | jsjc | ok let me check those reinvite. |
| 16:09.51 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:10.04 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 16:13.13 | Squeeb | what does :T303 timed out. mean? |
| 16:19.21 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:22.10 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 16:22.43 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v000-101.mobile.uci.edu) |
| 16:24.46 | Squeeb | My /etc/dahdi/system.conf keeps reverting back to us whenever I run dahdi_genconf |
| 16:24.52 | Squeeb | how can I keep it as uk? |
| 16:26.00 | [TK]D-Fender | lStop running genconf |
| 16:26.04 | jsjc | [TK]D-Fender You got it right ;) canreinvite=no on the outside sip phone and all working ;) |
| 16:26.18 | [TK]D-Fender | jsjc, You're welcome |
| 16:26.33 | jsjc | Thanks heaps! |
| 16:28.20 | Squeeb | I can't get my BRI to assign with the exchange |
| 16:28.21 | Squeeb | PRI Span: 1 Changing from state 2(Assign awaiting TEI) to 1(TEI unassigned) |
| 16:28.27 | WIMPy | Squeeb: Unless you plan to connect phones to the card the setting has no effect anyway (AFAIK). |
| 16:28.28 | Squeeb | and I'm completely out of my depth |
| 16:28.32 | Squeeb | oh |
| 16:29.03 | WIMPy | Squeeb: Are you sure, you've got a ptmp line? |
| 16:29.26 | Squeeb | not really, we have ISDN2e walljacks |
| 16:29.28 | Squeeb | provided by BT |
| 16:29.30 | Squeeb | 3 of them |
| 16:30.05 | WIMPy | So what did you order? Or what was confirmed? |
| 16:30.15 | Squeeb | order? confirmed? |
| 16:30.20 | Squeeb | this is an existing line, it's worked for years |
| 16:30.24 | WIMPy | Type of line. |
| 16:30.28 | Squeeb | but the box blew up so we're re-installing |
| 16:30.34 | WIMPy | With what? |
| 16:30.37 | Squeeb | asterisk |
| 16:31.17 | WIMPy | Do the lines share a common set of numbers of does each line have their own set of numbers. |
| 16:32.02 | Squeeb | common set |
| 16:32.22 | WIMPy | Then it will surely be ptp. |
| 16:32.35 | Squeeb | ptmp ? |
| 16:32.56 | WIMPy | no. |
| 16:33.00 | Squeeb | oh |
| 16:33.10 | WIMPy | You've got it configured for ptmp, but it's most probably ptp. |
| 16:33.11 | Squeeb | well the old pbx was using zaptel and it was set to ptmp |
| 16:33.27 | WIMPy | Err, ok. |
| 16:33.58 | Squeeb | bri_cpe_ptp then? |
| 16:34.17 | WIMPy | That's what I'd assume. |
| 16:34.53 | Squeeb | ok |
| 16:34.55 | Squeeb | set them to that |
| 16:35.06 | Ice_Strike | Which is better to use? parsed the Asterisk results or use AGI command? |
| 16:35.08 | *** join/#asterisk navaismo (~navaismo@189.146.54.165) |
| 16:35.09 | Squeeb | PRI Span: 1 > TEI: 0 State 7(Multi-frame established) |
| 16:35.10 | Squeeb | oh right |
| 16:35.26 | Ice_Strike | like phpagi |
| 16:35.37 | WIMPy | But if you uses ptmp before, that may be wrong. |
| 16:35.51 | WIMPy | used |
| 16:36.01 | Squeeb | well it's doing more than it did before |
| 16:36.06 | Squeeb | this is the latest libpri btw |
| 16:36.43 | Squeeb | now there's all sorts of weirdness happening |
| 16:37.03 | Squeeb | Ext: 1 Cause: Invalid call reference value (81), class = Invalid message (e.g. parameter out of range) (5) ] |
| 16:37.59 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.138.70) |
| 16:38.39 | WIMPy | Bad, but better than nothing . |
| 16:38.58 | *** join/#asterisk timahvo1 (~rogue@41.80.124.1) |
| 16:39.02 | WIMPy | 'pri set debug 2 span 1' and dump the output to a pastebin. |
| 16:39.11 | Squeeb | ok hang on |
| 16:40.25 | *** join/#asterisk rdegges (u4891@gateway/web/irccloud.com/x-zngotuwvgiyinljz) |
| 16:40.41 | [TK]D-Fender | <Ice_Strike> Which is better to use? parsed the Asterisk results or use AGI command? <- for what? |
| 16:40.45 | Squeeb | http://pastebin.com/1jN5q554 |
| 16:40.58 | [TK]D-Fender | Ice_Strike, Where? When? |
| 16:41.25 | Squeeb | that's just it idling btw |
| 16:41.27 | Squeeb | no calls |
| 16:41.41 | WIMPy | We need a call. |
| 16:41.46 | Squeeb | ok hng on |
| 16:41.54 | Squeeb | incoming? |
| 16:42.32 | WIMPy | I'll take both directions, but outgoing is preferred. |
| 16:42.41 | Squeeb | http://pastebin.com/3jVnTUzh |
| 16:42.43 | Squeeb | oh |
| 16:42.45 | Squeeb | this is incoming |
| 16:42.47 | Squeeb | call faioled |
| 16:42.50 | Squeeb | failed * |
| 16:43.32 | WIMPy | That actually looks ok. |
| 16:43.55 | Squeeb | what's line 83 about? |
| 16:44.01 | WIMPy | Like just missing the appropriate extension. |
| 16:44.23 | Squeeb | oh |
| 16:44.27 | Squeeb | well, ok .. let me try outgoing |
| 16:44.34 | WIMPy | It doesn't find the called extension in yourt dialplan and rejects the call. |
| 16:44.46 | Squeeb | well that's interesting |
| 16:45.03 | Squeeb | Oh fuck! |
| 16:45.05 | Squeeb | outbound call works |
| 16:45.11 | Squeeb | ok well, .. that's a start |
| 16:46.00 | Squeeb | right so we're getting somewhere |
| 16:46.02 | WIMPy | So everythin is ok. |
| 16:46.07 | Squeeb | that's cpe btw |
| 16:46.12 | WIMPy | Just needs a dialplan. |
| 16:46.16 | Squeeb | ok |
| 16:46.25 | Squeeb | i'll start core set verbosing :) |
| 16:46.29 | Squeeb | this bit I should be ok with |
| 16:46.36 | Squeeb | oh it's an american ringing noise btw :/ |
| 16:46.58 | WIMPy | So you have a phone on the 4th port or what? |
| 16:47.09 | Squeeb | SIP phone connected |
| 16:47.11 | Squeeb | softphone |
| 16:47.29 | WIMPy | That does it's own stuff. |
| 16:47.35 | Squeeb | ah fair enough |
| 16:47.41 | Squeeb | I thought the ringing noise came from the other end |
| 16:47.53 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 16:47.56 | WIMPy | It can, but it need not. |
| 16:48.07 | Squeeb | i see |
| 16:48.28 | Squeeb | ISDN is one of those dark magic things :/ |
| 16:48.33 | Squeeb | I wish there was a decent online resource for it |
| 16:48.58 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN/ is my try on that. |
| 16:49.01 | Squeeb | Span 1: Extension 425762@from-pstn does not exist. :D yay |
| 16:49.09 | Squeeb | so I know it's coming to the right place now |
| 16:49.43 | Squeeb | thanks |
| 16:49.57 | Squeeb | maybe chuck it on voip-info site? |
| 16:50.58 | WIMPy | I haven't been there for ages. |
| 16:52.15 | *** join/#asterisk rjvvliet (~rjvvliet@217.21.249.170) |
| 16:52.53 | Squeeb | right, best go and install this emergency PBX then :) |
| 16:52.54 | Squeeb | thanks |
| 16:53.01 | Squeeb | for all your help, really got me out of some trouble |
| 16:53.04 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 17:00.19 | Ice_Strike | [TK]D-Fender Well I dont know, just reading some documentation and trying to fig it out what do I need and how to use it. |
| 17:00.38 | Ice_Strike | [TK]D-Fender I would like agents to use the website - the website connect to the asterisk server. |
| 17:00.47 | Ice_Strike | Agent can manually dail a number on the system |
| 17:00.55 | [TK]D-Fender | Ice_Strike, First thing is that none of terms you have used to date mean anything. There is no fixed definition therefor there is no way to address them. |
| 17:01.22 | [TK]D-Fender | Ice_Strike, what is an "agent"? this can mean MANY things. Dial? Dial on what? What is done with the number dialed? |
| 17:01.51 | [TK]D-Fender | Ice_Strike, "Agents to use the webesite" <- says nothing specific at all. |
| 17:02.08 | [TK]D-Fender | "connect to Asterisk server" <- ditto. |
| 17:02.24 | Ice_Strike | [TK]D-Fender I meant operator/user. I would like to develop a call manager using PHP. Somehow I need to find a way to connect to Asterisk server. |
| 17:03.11 | [TK]D-Fender | Ice_Strike, clarify "call manager" |
| 17:03.15 | Ice_Strike | When user logged onto the website - it will automaically dail a number (read from the database) - When they have finish the call, it will go to next call. |
| 17:03.32 | [TK]D-Fender | Ice_Strike, You want to make a predictive dialier then? |
| 17:03.36 | [TK]D-Fender | dialer* |
| 17:04.05 | [TK]D-Fender | Ice_Strike, There are solutions already made for this. Have you looked at them? |
| 17:04.48 | Ice_Strike | [TK]D-Fender I have not, which one? |
| 17:04.58 | [TK]D-Fender | Ice_Strike, http://www.google.ca/#hl=en&cp=14&gs_id=2e&xhr=t&q=asterisk+predictive+dialer&pf=p&sclient=psy-ab&site=&source=hp&pbx=1&oq=asterisk+predi&aq=0&aqi=g2g-v2&aql=&gs_sm=&gs_upl=&bav=on.2,or.r_gc.r_pw.r_cp.,cf.osb&fp=cfbde561d013d8ec&biw=1600&bih=927 |
| 17:05.09 | [TK]D-Fender | Ice_Strike, Go look around before reinventing the wheel |
| 17:05.44 | Ice_Strike | [TK]D-Fender Thanks! I will look into it. |
| 17:06.33 | Ice_Strike | [TK]D-Fender How can it be done if I create my own? |
| 17:06.43 | Ice_Strike | Using AGI? |
| 17:09.20 | [TK]D-Fender | Ice_Strike, You seem to lack a grasp of what AGI really is. I recommend a thorough reading of ... THE BOOK. |
| 17:09.21 | [TK]D-Fender | ~book |
| 17:09.21 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:10.12 | Ice_Strike | From what I have read, I can use any language. AGI can execute php script. |
| 17:13.31 | [TK]D-Fender | Ice_Strike, You missed the point of when AGI gets actually called |
| 17:14.00 | [TK]D-Fender | And AGI doesn't "executes" scripts.. it IS an external script in whatever language you write it in |
| 17:14.53 | _Corey_ | Ice_Strike: From reading what you guys have been talking about so far, I think it's important to mention that anything AGI will be a small part of any predictive dialer solution... |
| 17:15.28 | _Corey_ | building any kind of a call center package (predictive, etc.) with Asterisk is one of the most complex and challenging projects you could possibly undertake |
| 17:15.48 | WIMPy | Is it? |
| 17:15.51 | [TK]D-Fender | No, there are far worse :) |
| 17:16.04 | *** join/#asterisk alexscott (~alexscott@2a01:e35:8b11:2e40:223:6cff:fe84:b66c) |
| 17:16.09 | paulc | *nods in agreement* You'll need some solid database design, and some good code to keep it all running efficiently |
| 17:16.10 | WIMPy | I find using it as a replacement for a normal PBX more challenging. |
| 17:16.48 | paulc | I was going through an old box the other day and found some dialplan I'd written for PBX functionality replacement.. do not disturb, system speed dial, personal speed dial.. made me smile |
| 17:16.54 | [TK]D-Fender | High-availability, realtime everything in * with distributed hints, etc... taht would do it... oh and supporting 50,000 phones and 20,000 simultaneous calls while recording and transcoding ;) |
| 17:17.32 | [TK]D-Fender | A predictive dialer could be comparatively petty crap :) |
| 17:18.24 | _Corey_ | depends on how low you're aiming I guess... Developing something that even remotely compares with one of the many commercial dialer platforms out there is a big project |
| 17:20.14 | [TK]D-Fender | _Corey_, Aim low.. you'll miss less ;) |
| 17:20.27 | Ice_Strike | There are a few things I need: Create a number of groups, for example: Sales Group, Technical Group. When operator finish their call - it will then go to next call automaically. Operator should also dail a number manually if they need to. After the call they need to classify what has been done before go to next call. |
| 17:20.28 | _Corey_ | well, just go with Vicidial then ;) |
| 17:20.47 | [TK]D-Fender | _Corey_, FFS show SOME standards :p |
| 17:20.52 | _Corey_ | hahaha |
| 17:21.16 | Ice_Strike | How reliable is Vicidial? |
| 17:21.29 | *** join/#asterisk chasing`Sol (~cS@197.132.32.149) |
| 17:21.56 | _Corey_ | Ice_Strike: Well, they have customers... |
| 17:22.15 | [TK]D-Fender | Ice_Strike, There are many ways do control the process you're looking for. Go see what solutions are out there already and see if one suits your needs. Then if not, go read the book and understand how *'s bit work to see where YOU would implement it |
| 17:23.43 | Ice_Strike | Thanks for suggestion [TK]D-Fender |
| 17:24.18 | Ice_Strike | Vicidial interface is horrible :/ |
| 17:24.31 | paulc | You have the source to make it prettier ;-) |
| 17:25.19 | Ice_Strike | :) |
| 17:28.28 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 17:36.50 | [TK]D-Fender | Ice_Strike, Did you hear that? |
| 17:37.59 | Ice_Strike | [TK]D-Fender Did I hear what? |
| 17:38.02 | [TK]D-Fender | Ice_Strike, It's the sound of no-one disagreeing with you ;) |
| 17:38.44 | Ice_Strike | Ha :) |
| 17:43.08 | Ice_Strike | goautodial.com look good |
| 17:43.11 | Ice_Strike | anyone tried that? |
| 17:44.53 | leifmadsen | Ice_Strike: just looks like vicidial platform |
| 18:01.04 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 18:09.19 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 18:13.02 | *** join/#asterisk Keisuke (~Keisuke@132.208.12.66) |
| 18:21.10 | *** join/#asterisk nighty^ (~nighty@static-68-179-124-161.ptr.terago.net) |
| 18:23.53 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 18:39.38 | paulc | Has anyone experienced issues where you've got an IVR in Asterisk dialplan, it works great, then mid call the DTMFs stop being recognised? (possibly something to do with RTP packet sequence numbers? rolling over?) This is Asterisk 1.6, I'm wondering if it's worth writing a ping-pong type task to test/prove (and test against Asterisk 10) |
| 18:49.59 | *** join/#asterisk min3r (1000@173.81.252.114) |
| 18:54.39 | *** join/#asterisk Linux4Eric (~change@cpe-71-72-172-65.woh.res.rr.com) |
| 19:12.33 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 19:17.46 | *** join/#asterisk logicwrath_work (~no@74-94-239-197-Michigan.hfc.comcastbusiness.net) |
| 19:26.19 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:31.23 | navaismo | hi im having this using asterisk 1.8.8.1 Received response: "Forbidden" from '"asterisk" <sip:XXXXXXX@siptrunk.com>;tag=as2921a5b0' |
| 19:31.35 | navaismo | but ins asterisk 1.6.2.1 i can call with no problems |
| 19:31.44 | navaismo | what im missing? |
| 19:33.19 | navaismo | sorry asterisk version with failed attemp its 1.8.7.2 |
| 19:35.11 | *** join/#asterisk sereal-work (~sereal@unaffiliated/sereal) |
| 19:35.29 | sereal-work | Are there any working skype gateways for asterisk or is that long dead now? |
| 19:38.20 | leifmadsen | sereal-work: pretty much dead at this point from what I can tell; the chan_skype channel driver is no longer being sold |
| 19:38.42 | leifmadsen | everything else I've seen has been some sort of hack through a windows box, macro scripts and sound card interfaces |
| 19:38.54 | sereal-work | yeah that's what I heard too |
| 19:38.54 | leifmadsen | (that being said, I never really researched it that far) |
| 19:39.05 | *** join/#asterisk ks3 (~ks3@74.115.41.6) |
| 19:39.16 | sereal-work | I'm not sure why a windows box would be need though, skype runs on linux now . |
| 19:40.02 | *** join/#asterisk kombi (~kombi@port-92-198-15-96.static.qsc.de) |
| 19:40.03 | sereal-work | I understand people are using VM's that run windows or something that has skype, then plugging into those audio interfaces. What i'm more curious about is how they are managing dialing out |
| 19:40.50 | leifmadsen | sereal-work: well, it might not always be windows, but basically it's a single connection / client running via some windowing interface and being controlled via a macro or something like that |
| 19:41.23 | sereal-work | humm anyone have any guides? |
| 19:44.14 | kombi | compiling chan-sccp against asterisk-10.0.1 it shouts "found unsupported version", chan-sccp.org claims it supported. Am I missing something? |
| 19:46.48 | eZz | i have had Xfb to launch X sessions of skype to get it worked |
| 19:46.55 | eZz | even on host w/o X |
| 19:50.55 | _Corey_ | sereal-work: I got the impression that you could just buy a Skype Connect account and set up a SIP config... have you tried this? |
| 19:52.15 | sereal-work | _Corey_, no, I did look at that, but it seems a bit silly to pay for something that should be free anyways. If I used a straight skype client this wouldn't be a issue. The fact that I need to pay to essentially use a different headset and microphone is just stupid. |
| 19:53.12 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 19:55.03 | _Corey_ | sereal-work: Well, Skype Connect is a business service. (From Skype's perspective, individual consumers aren't going to be connecting PBXs to Skype) |
| 19:55.34 | _Corey_ | For $6/mo it seems like more than a bargain to me if your goal is to connect a PBX to Skype and not have some half-assed solution |
| 19:55.54 | sereal-work | $6 a month is pretty steep for using something that is free |
| 19:56.37 | _Corey_ | nothing's really "free" ;) |
| 19:56.46 | sereal-work | well their client is free. |
| 19:57.04 | sereal-work | In my case I would really be paying 6$ a month to use my special microphone |
| 19:58.21 | Katty | ohai |
| 19:58.21 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 19:58.38 | [TK]D-Fender | Katty, Mew. |
| 19:58.52 | Katty | how're mew |
| 19:59.29 | [TK]D-Fender | Katty, The mewsual |
| 20:00.34 | kombi | I' bad at c... Those includes at the top of an header file, how do I point them to the right places (such as the asterisk dir)? |
| 20:00.48 | Katty | :> |
| 20:01.12 | chuckf | sereal-work: the client is free as it is a gateway to more features. The pbx addon costs because you are not likely to use the fee based services provided by skype |
| 20:01.43 | chuckf | sereal-work: 'more features' == 'paid features' |
| 20:02.05 | *** join/#asterisk CorvetteZR1 (~scratchi@195.34.234.216.sta.connection.ca) |
| 20:02.20 | Katty | i want a corvette |
| 20:02.41 | chuckf | Katty: a little red one? |
| 20:02.48 | Katty | no |
| 20:02.51 | Katty | a black one. |
| 20:03.40 | kombi | must the asterisk source dir be in my path to compile against it? |
| 20:03.53 | sereal-work | I don't quite understand how using a different microphone is a feature. |
| 20:04.07 | sereal-work | We don't have this issue with google talk and asterisk |
| 20:06.10 | Katty | what are we complaining about? i want in |
| 20:07.13 | _Corey_ | Katty: I've given up |
| 20:07.20 | Katty | given up what |
| 20:07.41 | chuckf | Katty: paying to tie skype into your asterisk pbx |
| 20:07.47 | *** join/#asterisk timahvo1 (~rogue@41.90.129.49) |
| 20:08.28 | Katty | ah right |
| 20:08.44 | Katty | i guess their employees have to paid too huh |
| 20:08.54 | Katty | if only they worked for peanuts!!! *shakes fist* |
| 20:09.56 | _Corey_ | Katty: Their revenue model has always been mysterious... one of our former guys has worked there for a few years now though and seems to like it a lot (despite the msft takeover) |
| 20:10.27 | Katty | maybe they have a laundry mat on site |
| 20:10.32 | Katty | that'd make me wanna stay |
| 20:11.04 | Katty | day care services. |
| 20:11.12 | Katty | ATTACHED STARBUCKS CAFE :> |
| 20:11.40 | Katty | or maybe if they let you take your pets to work, that'd be totally awesome |
| 20:11.58 | _Corey_ | I'll have to ask him.. he had been in Estonia for a while and just relocated to Palo Alto a couple weeks ago |
| 20:12.20 | _Corey_ | I guess they're keeping things pretty autonomous for now |
| 20:13.00 | Katty | that's ok |
| 20:13.07 | Katty | we don't need to be all up in their business anyway |
| 20:13.10 | Katty | that's impolite. |
| 20:13.18 | chuckf | now that skype is owned by MS they don't need to make money |
| 20:13.45 | Katty | i think they still need to pay their employees! |
| 20:13.58 | Katty | even if they are owned by microsoft |
| 20:14.25 | chuckf | that's where MS comes in, they just fire all the employees and let skype die off |
| 20:14.32 | chuckf | :) |
| 20:14.46 | _Corey_ | chuckf: yeah, no indication of that thus far ... |
| 20:14.54 | leifmadsen | I've not seen anything like that happen |
| 20:14.58 | _Corey_ | who knows what will happen though... :) |
| 20:15.11 | Katty | maybe they will all get a raise! yay! |
| 20:15.33 | *** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:17.28 | chuckf | short of my last sarcastic remark, them charging for things like PBX integration is a revenue model to pay for development and such things |
| 20:18.18 | Katty | yes. |
| 20:18.20 | Katty | most likely. |
| 20:18.29 | Katty | and for caffeines, like coffee, to appease their developers. |
| 20:18.32 | Katty | possibly cheetos too. |
| 20:18.45 | Katty | cheetos are very important for development. |
| 20:23.15 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 20:32.18 | *** join/#asterisk McBoingBo (~McBoingBo@mail.hrsg.ca) |
| 20:33.39 | McBoingBo | getting a strange request, wondering if this is feasable, having folks making calls from their home but they need to somehow gain our companies name on their ID when calling out, they are not using or connecting to our asterisk server in any way |
| 20:34.43 | n3hxs | Cheetos cause orange letters on keyboards. |
| 20:35.35 | n3hxs | McBoingBo, put an asterisk server in the caller's home. |
| 20:35.44 | McBoingBo | n3hxs, or spoof caller id! |
| 20:36.31 | *** join/#asterisk timahvo1 (~rogue@41.80.215.41) |
| 20:36.52 | chuckf | McBoingBo: allowing something like that is bad on so many levels |
| 20:37.48 | chuckf | McBoingBo: once you get over all the technical hurdles to implement it |
| 20:38.06 | RyuunoAelia | hi, I have this setup : http://nopaste.info/3b07b3264d.html <- and everybody registers ok, but the calls doesn't get from localphone to world or world to localphone, what is wrong here, localphone answers with a "SIP/2.0 404 Not Found" when a call is incoming from world and I am quite clueless with asterisk config :p |
| 20:39.00 | *** join/#asterisk Goldwing (~Goldwing@84.245.46.83) |
| 20:40.35 | Goldwing | anyone here have experience with Time Interval is Asterisk? |
| 20:41.54 | Goldwing | i'm trying to make asterisk pick up the phone, and say goodmorning/afternoon/evening, but without luck |
| 20:45.03 | kombi | can't shutdown asterisk, tried killall, kill and kill -9. Are there more drastic measures? (besides reboot) |
| 20:46.56 | RyuunoAelia | if kill -9 doesn't work... except reboot I don't think there is anything.. the only case (that I know) when kill -9 doesn't kill an application is when it is stuck in kernel mode... |
| 20:47.22 | [TK]D-Fender | Goldwing, "core show application gotoiftime" |
| 20:47.37 | kombi | RyuunoAelia: So I thought, back after reboot then..;) |
| 20:48.25 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 20:48.45 | Goldwing | thx |
| 21:05.02 | *** join/#asterisk ccesario (~ccesario@189-46-93-199.dsl.telesp.net.br) |
| 21:08.37 | *** join/#asterisk d_preston215 (~chatzilla@50-73-214-237-philadelpia.hfc.comcastbusiness.net) |
| 21:08.55 | d_preston215 | Anyone ever had issues with chanspy channels not responding? |
| 21:12.58 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:15.07 | McBoingBo | anyone here ever use any spoofing card services like spoofcard.com? |
| 21:15.41 | *** join/#asterisk seijirou (477f3132@gateway/web/freenode/ip.71.127.49.50) |
| 21:16.27 | seijirou | Hi. I have 2 Polycom soundpoint 335 registered. When I call a phone there is no audible ringtone. Calls work, but they do not ring audibly. Is this a phone problem or a config problem? The phones do ring when registered with a VOIP provider like aptela. |
| 21:23.39 | *** join/#asterisk rjvvliet (~rjvvliet@178-85-122-187.dynamic.upc.nl) |
| 21:29.14 | *** join/#asterisk dandate2 (~dan@180.190.185.183) |
| 21:30.32 | dandate2 | when I call comcast from my pap2 phone, their IVR can't recognize my dialing attempts. How to fix? |
| 21:38.47 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:40.42 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 21:41.26 | FLeiXiuS | Whats the best way to receive the status of a call through CLI and remotely? I've been looking at AMI; although the Status event isnt in real-time |
| 21:42.11 | eZz | you can get it from DIALSTATUS |
| 21:44.22 | FLeiXiuS | eZz, Can you get dial status through AMI? |
| 21:45.00 | eZz | sure, manager sents an event when setting a variable |
| 21:45.09 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 21:50.48 | seijirou | Anybody know how to configure ringtones for registering phones? |
| 21:54.57 | FLeiXiuS | eZz, http://pastie.org/3239627 DIALSTATUS is always blank. |
| 21:55.09 | eZz | Note: To obtain useful DIALSTATUS information when dialing a peer the peer's definition must contain qualify=yes (e.g., in sip.conf or iax.conf). |
| 21:55.27 | FLeiXiuS | eZz, Even for local extensions? |
| 21:56.03 | eZz | no should work with local |
| 21:56.40 | eZz | i don't know why it's a blank, to me it's working for years |
| 21:56.48 | FLeiXiuS | Check that pastie again eZz I included the console. |
| 21:57.28 | eZz | hah |
| 21:57.39 | eZz | in this case you shouldn't get this |
| 21:57.45 | eZz | because a channel was answered |
| 21:58.16 | eZz | put h,1,NoOp(Dial Status: ${DIALSTATUS}) |
| 21:58.23 | eZz | and you'll get it |
| 21:59.56 | FLeiXiuS | Interesting..but h happens after hangup. I need to know precisely when the user hangsup |
| 22:00.00 | FLeiXiuS | Bah - answers* |
| 22:00.54 | *** join/#asterisk jsjc (~Adium@181.Red-83-35-52.dynamicIP.rima-tde.net) |
| 22:01.02 | eZz | you'll know it inline when hangup/congestion but not on answer |
| 22:02.06 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-jhjhgnceuylthrei) |
| 22:03.43 | FLeiXiuS | I guess I could use M() for some of this. |
| 22:04.06 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 22:05.11 | FLeiXiuS | Next question, in cellular networks, do immediate hangups/rejected calls show on up the bill? |
| 22:05.16 | FLeiXiuS | on * |
| 22:05.22 | Dovid | they **should** |
| 22:05.48 | FLeiXiuS | Dovid, Even if it was never answered? |
| 22:06.43 | Dovid | oh thought u meant billed for. r they on bills? depends if ur an end user ur lucky if u get cdr's at all. most likely not. if ur a carrier then u get everything |
| 22:06.49 | eZz | this is a good question, i have the same problem too |
| 22:07.05 | eZz | my isp is not sending any sit's or other info when is never answered |
| 22:07.16 | eZz | just request timeout |
| 22:11.08 | Dovid | ur ISP is not sending any what? |
| 22:12.03 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:12.05 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:12.12 | *** join/#asterisk s[X] (~s_x_@eth589.qld.adsl.internode.on.net) |
| 22:13.09 | eZz | Dovid: Special Information Tones (SITs) |
| 22:13.12 | eZz | a series of |
| 22:13.12 | eZz | three precise, sequential audio tones, which indicate that the callee cannot |
| 22:13.13 | eZz | be reached |
| 22:13.20 | eZz | sorry for flood |
| 22:17.44 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 22:20.06 | *** join/#asterisk Tamo (~tamo@irc.tactest.net) |
| 22:20.10 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 22:20.25 | *** join/#asterisk rotten777 (~matthew@fl-76-3-160-158.dhcp.embarqhsd.net) |
| 22:22.15 | rotten777 | can anyone help? I'm running into a weird issue where asterisk doesn't play my voicemail greeting when calling in and the messages are left in empty wav files and emailed to me |
| 22:22.22 | rotten777 | i'm thinking something wrong with the playback |
| 22:22.28 | rotten777 | i have the format=wav already |
| 22:22.54 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 22:25.03 | *** join/#asterisk Nasga (~Nasga@AAmiens-157-1-137-63.w90-58.abo.wanadoo.fr) |
| 22:32.07 | seijirou | Anybody know how to configure ringtones for registering phones? My phone rings, but it's not audible. |
| 22:32.34 | WIMPy | Look in to your phones manual. |
| 22:34.08 | seijirou | It's an asterisk configuration issue. the phone rings fine when registering to a voip service provider. |
| 22:35.05 | WIMPy | Asterisk doesn't tell a phone how to ring unless you manually tweak headers. |
| 22:35.43 | [TK]D-Fender | And even then it isn't Asterisk directly. The phone has to permit some means of signalling it to behave differently |
| 22:35.54 | seijirou | I haven't done anything that I'm aware of, but the phone rings with a different provider, it does not ring on asterisk. I'm not sure where to look. |
| 22:35.55 | WIMPy | yes |
| 22:36.08 | [TK]D-Fender | in the PHONE |
| 22:36.36 | WIMPy | Maybe I should have said... |
| 22:36.44 | seijirou | Nothing has changed on the phone except what the tftp server IP address is. |
| 22:36.46 | WIMPy | Asterisk doesn't tell a phone how to ring unless you manually tweak some phone specific headers. |
| 22:37.38 | seijirou | What are those headers? Maybe they're tweaked and i don't know it. |
| 22:37.57 | [TK]D-Fender | seijirou: And the TFPT server is what configures the phone. If that also allow some sort of header then you'll have to read up on that too |
| 22:38.37 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 22:38.56 | seijirou | And the tftp server is asterisk. So my conclusion is there's something amiss with my asterisk config. But I'm not sure where to look. That's what I'm looking for a clue on. |
| 22:39.07 | FLeiXiuS | Whats the best way to hangup a call after being answered an established for 10 seconds? |
| 22:39.25 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-zslxruxygpcujawu) |
| 22:39.46 | WIMPy | FLeiXiuS: 'core show application Dial' Look for L. |
| 22:40.23 | FLeiXiuS | WIMPy, Ahh thats easier than defining a macro. |
| 22:42.48 | [TK]D-Fender | seijirou: It isn't "what's amiss" about *. *'s default behaviour is NORMAL. Whatever else is going on is what is not normal. |
| 22:44.13 | seijirou | [TK]D-Fender: I'm willing to blame anything, but I don't know what to blame? |
| 22:44.25 | [TK]D-Fender | seijirouAnd the tftp server is asterisk. <- No... it may be on the same box as Asterisk, but AAsterisk in to a tftp server |
| 22:44.41 | [TK]D-Fender | is not* |
| 22:44.52 | seijirou | Okay, I'm sure it's tftpd. |
| 22:47.17 | [TK]D-Fender | Fine. Still not * |
| 22:47.40 | seijirou | Alright. It's a default config, it doesn't work. Still not *. Roger. |
| 22:47.48 | navaismo | in fact not the tftp server but the configuration files |
| 22:47.49 | seijirou | Thanks for the help. o7 |
| 22:52.41 | *** join/#asterisk grkblood (~grkblood@c-71-226-163-238.hsd1.sc.comcast.net) |
| 22:52.45 | grkblood | is the rumor true that voip is a reseller of vitelity |
| 22:53.13 | [TK]D-Fender | ... |
| 22:53.15 | [TK]D-Fender | huh? |
| 22:53.37 | Chainsaw | The game is up Fender. We'd better admit. |
| 22:53.42 | Chainsaw | We all just resell Vitelity. |
| 22:53.56 | Chainsaw | bows head in shame |
| 22:55.19 | [TK]D-Fender | lops it clean off |
| 22:55.23 | *** join/#asterisk serafie (~erin@nat/digium/x-fzuoobbpsmwcgbxu) |
| 22:55.25 | [TK]D-Fender | DEATH BEFORE DISHONOUR! |
| 22:55.28 | grkblood | ive heard voip is a reseller of vitelity but voip offers several PoPs while vitelity only offers one. if theyre a reseller I would think vitelity would have numerous PoPs in the same locations as voip |
| 22:55.40 | Qwell | "voip"? |
| 22:55.44 | grkblood | voip.ms |
| 22:55.50 | [TK]D-Fender | grkblood: I think you're missing some clear proper nouns in there... |
| 22:56.06 | FLeiXiuS | Is there a more organized way to store Dial options? |
| 22:56.15 | FLeiXiuS | I have a dial line a mile long.. |
| 22:56.30 | grkblood | i meant voip.ms the company, not voip the acronym |
| 22:57.10 | [TK]D-Fender | "voip" is not really an acceptable abbreviation for them if you want us to know what youre talking about |
| 22:57.40 | [TK]D-Fender | FLeiXiuS: Dialplan is what it is. Use variables if you want tobreak it down a little... |
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| 23:25.42 | bbourdage | ? |
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