| 00:03.21 | mchou | atan: where you you like your DID to be? |
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| 00:14.47 | atan | mchou, Nova Scotia, outside of Halifax |
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| 01:23.00 | TedNJ41 | Hi guys. Can anyone let me know what package I can add to my box to enable Caller ID Lookup for US Phone Numbers? |
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| 01:32.30 | TedNJ41 | Can someone help me please? |
| 01:32.41 | cj | depends ;) |
| 01:33.02 | TedNJ41 | I'd like to implement Caller ID in my box. Do you know what I can install? |
| 01:33.16 | TedNJ41 | To lookup numbers from the US. |
| 01:33.32 | cj | google, perhaps |
| 01:34.05 | TedNJ41 | I did that and it shows Caller ID Superfecta. |
| 01:34.12 | TedNJ41 | Have you tried it? |
| 01:34.18 | cj | trying it now |
| 01:34.46 | cj | not very useful, eh? |
| 01:34.49 | cj | tried whitepages.com? |
| 01:35.03 | TedNJ41 | Not yet. |
| 01:35.07 | TedNJ41 | I am checking. |
| 01:35.13 | cj | if you are willing to pay, I can get them to service you |
| 01:35.34 | TedNJ41 | No, I am looking for a free lookup service. |
| 01:35.58 | cj | hurm... check with the FCC for US numbers |
| 01:36.06 | cj | ask brian aker. I be he knows... |
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| 02:08.48 | lanning | http://www.phonenumber.com/search/ReversePhone?phone=847-286-2500 |
| 02:09.33 | lanning | something should be able to be built off of that interface. |
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| 02:12.43 | [TK]D-Fender | lanning: And he left half an hour ago.... |
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| 03:12.53 | jeremy_g | yo |
| 03:14.00 | jeremy_g | Is there a way to make System (/a.sh) return to dial plan instead of waiting for a.sh to finish |
| 03:14.37 | jeremy_g | i just want the System app to start a script a.sh and then just return to the dialplan execution. Don't wait for a.sh to hit the exit |
| 03:15.18 | *** join/#asterisk dandate2 (~gtejkgjke@58.69.25.48) |
| 03:15.32 | dandate2 | so i'm trying to decide whether to go with ulaw or g729. what % of americans still use landlines? |
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| 03:21.46 | [TK]D-Fender | jeremy_g: Run it as a daemon like any other process |
| 03:22.04 | [TK]D-Fender | dandate2: 99% |
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| 03:24.48 | NEEDINGHELP123 | what does mediawaitforconnect do exactly in chan_ooh323 |
| 03:27.01 | jeremy_g | [TK]D-Fender: you mean if a.sh contains only sleep 40; i should make it sleep 40 & |
| 03:27.46 | jeremy_g | [TK]D-Fender:as far as I understand, then system will wait for the daemon to exit... |
| 03:27.50 | jeremy_g | let me man system |
| 03:28.04 | jeremy_g | its the same call that * uses i guess |
| 03:28.46 | [TK]D-Fender | jeremy_g: Run the SCRIPT in the background, not jsut each COMMAND in it |
| 03:28.55 | [TK]D-Fender | jeremy_g: script.sh & |
| 03:29.52 | jeremy_g | [TK]D-Fender:i did that, it didnt help |
| 03:30.03 | jeremy_g | um thinking about "nohup $program < /dev/null > /dev/null 2>&1 &". |
| 03:30.22 | jeremy_g | sorry |
| 03:30.30 | jeremy_g | my bad, didnt reload asterisk |
| 03:30.42 | jeremy_g | just realized, its 5:30 AM here |
| 03:33.35 | jeremy_g | [TK]D-Fender:it worked, funny enough i tried & before coming to this channel but forgot to reload. |
| 03:33.42 | jeremy_g | [TK]D-Fender:thanks |
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| 03:40.21 | jeremy_g | NEEDINGHELP123:google, and show application is also nice |
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| 03:41.43 | jeremy_g | i gotta sleep guys, bye. |
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| 03:53.30 | NEEDINGHELP123 | why is t |
| 03:53.52 | NEEDINGHELP123 | why is it that in chan_ooh323, as soon as i dial to my box, for any call failed or good or whatever, it is ringing instantly with fake ringtone |
| 04:01.47 | dandate2 | fender: u mean 99% of americans are calling on cell phones heh |
| 04:02.40 | dandate2 | even if they are sitting at their desk with a landline in front of them prolly still calling on a cell heh |
| 04:04.50 | [TK]D-Fender | dandate2: and your last 2 questions have little or nothing to do with each other |
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| 04:17.33 | dandate2 | well i figure why spend for the extra bandwidth to do ulaw when g729 is not too off from the GSM sound anyway |
| 04:18.26 | [TK]D-Fender | dandate2: you clearly don't see the big picture |
| 04:19.48 | dandate2 | does a conversation from voip to cell sound better with the pbx handling ulaw? |
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| 04:23.20 | dandate2 | i mean the way i'm looking at it. if someone calls in on a cell phone to voip it sounds like GSM to both parties |
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| 04:25.14 | NEEDINGHELP123 | [TK]D-Fender help me out please. |
| 04:27.24 | *** join/#asterisk Besticles (~Besticles@ip68-104-111-21.lv.lv.cox.net) |
| 04:28.37 | Besticles | I'm having issues with Festival Text2Wave via AGI. I am sending to asterisk EXEC SYSTEM "echo 'Hi Everyone' | text2wave -o /tmp/sound.wav -otype -ulaw -" And it does not create the file, yet if i type in the command via Terminal letter for letter, it generates the file. I'm confused why it's not working. |
| 04:28.48 | Besticles | I tried doing Festival Server, but I couldn't get that to work either. |
| 04:31.26 | ChannelZ | why not run the program in whatever language your AGI is in, rather than telling Asterisk to execute it? |
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| 04:31.40 | Besticles | Program is running under windows. |
| 04:32.01 | ChannelZ | What is the AGI? |
| 04:32.11 | Besticles | FastAGI |
| 04:32.38 | ChannelZ | but what is it? A windows binary? A script? what does it do? |
| 04:33.22 | Besticles | .NET Framework 2.0. Can't call Framework 100% binaary. :/ |
| 04:33.38 | ChannelZ | sighs |
| 04:33.48 | Besticles | It controls every aspect of the call. |
| 04:34.15 | ChannelZ | Yet it's not able to run another binary? Ooooook... |
| 04:35.41 | ChannelZ | or are you saying you're running asterisk on one platform and the AGI on another? |
| 04:35.46 | Besticles | Yes, that's it. |
| 04:36.04 | Besticles | Asterisk is running on CentOS, and I'm running on Windows 7 |
| 04:36.11 | Besticles | Rather FastAGI is |
| 04:37.25 | Besticles | Sorry if I seem difficult. I'm pass that point of frustration with Festival lol. I tried doing Festival --Server, I've done the command I listed above in Terminal, it works. I added the command to the dialplan that works, I just cant get the command to work in AGI. |
| 04:37.26 | Besticles | FastAGI. |
| 04:37.28 | ChannelZ | well I'm not sure System supports piping since that's really a shell construct |
| 04:38.03 | ChannelZ | does it work if you supply a static piece of text to text2wav rather than telling it to read stdin? |
| 04:39.57 | Besticles | I'll try right now. |
| 04:40.54 | [TK]D-Fender | [00:23]<dandate2>i mean the way i'm looking at it. if someone calls in on a cell phone to voip it sounds like GSM to both parties <- You're looking wrong |
| 04:42.16 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 04:42.21 | [TK]D-Fender | dandate2: If you call a cell at G.711 then you are both dragged to CELL quality because he is the common lowest denominator. If you call G.729, then it's already degraded going to the ITSP. They upconvert to G.711 to hit the PSTN and then reach the cell where your already degraded audio get pushed through the cheese-grater AGAIN in going to GSM. DOUBLE CONVERSION |
| 04:42.26 | Besticles | Nah, that didn't work either. I basically got the command from: http://www.voip-info.org/wiki/view/Asterisk+cmd+Festival |
| 04:42.30 | Besticles | The PHP Portion of it. |
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| 04:44.12 | ChannelZ | well then you're doing something else wrong.. probably 'text2wav' isn't really being run because you're supplying no path and * can't find it |
| 04:45.19 | Besticles | god, i hope thats it. |
| 04:47.35 | dandate2 | hmm double conversion no good huh |
| 04:49.14 | dandate2 | i guess ill have to stick with the US datacenter heh |
| 04:52.12 | ChannelZ | hmm I guess text2wav has to read from a file. So it could be one of a few different things causing it not to work. |
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| 05:05.22 | Besticles | I actually got it to start making the file. |
| 05:05.31 | Besticles | For some reason it's adding a ? at the end of the file extension. |
| 05:07.19 | Besticles | I changed the location of the destination, and it started working somewhat. |
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| 06:38.52 | dandate2 | fender: i spoke with my sysadmin about this and he says "telco end codec coversion is not same as software endconversion" |
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| 06:49.44 | sawgood | Hi everyone! |
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| 07:32.15 | *** join/#asterisk _buck (~webp@121.242.23.197) |
| 07:33.43 | _buck | Hi I am following this tutorial to integrate kamailio with asterisk. kamailio is working fine and regstering the phones. But asterisk is not loading users from the tables . Can anyone please help.? http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb |
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| 07:45.52 | _buck | titter, can u help ? |
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| 08:30.22 | jblack | five gazillion years and I finally get around to setting up irssi to auto-nickserv |
| 08:30.49 | ChannelZ | nickserv in x-chat hasn't been working well for me lately |
| 08:31.07 | ChannelZ | or rather it works but it won't join the channel afterwards |
| 08:31.50 | tzafrir_laptop | jblack, actually freenode should support simple server authentication |
| 08:32.45 | jblack | I set it up two ways, with /network, and with an autocmd. |
| 08:33.33 | jblack | pardon, with /server, and with /network. |
| 08:34.48 | ChannelZ | some other networks' services would let me turn off forced Nickserv auth if your hostmask matched |
| 08:40.23 | ChannelZ | course now I am just now seeing that my identd hasn't been working for... an unknown length of time |
| 08:42.44 | ChannelZ | Oh. Hmm.. |
| 08:42.49 | *** part/#asterisk ChannelZ (~bobm@burner.com) |
| 08:44.18 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 08:44.31 | ChannelZ | I think it has to be services here have gotten really laggy |
| 08:44.56 | WIMPy | Not only services. |
| 08:45.57 | WIMPy | My Lagometer is often almost completely filled here and that's 2s. |
| 08:46.13 | ChannelZ | Hmm. I wonder if I turned mine off cuz it's 0 |
| 08:47.20 | ChannelZ | huh no, I don't even see where I could turn it off |
| 08:47.51 | ChannelZ | (without them still being in the interface) |
| 08:48.28 | ChannelZ | oh well, it's barf that this chan requires you be identified in Nickserv |
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| 09:24.36 | elliot98 | is there a way get the name of the sip device of the channel? |
| 09:24.50 | elliot98 | I could parse the the {CHANNEL} variable, but wondering if there is a better way |
| 09:25.27 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 09:42.26 | *** join/#asterisk dandate2 (~gtejkgjke@112.206.130.149) |
| 09:42.30 | dandate2 | <[TK]D-Fender> dandate2: If you call a cell at G.711 then you are both dragged to CELL quality because he is the common lowest denominator. If you call G.729, then it's already degraded going to the ITSP. They upconvert to G.711 to hit the PSTN and then reach the cell where your already degraded audio get pushed through the cheese-grater AGAIN in going to GSM. DOUBLE CONVERSION |
| 09:42.45 | dandate2 | does this also apply for cell-to-cell calls? |
| 09:45.26 | ChannelZ | probably depends on the networks involved and how they might interconnect |
| 09:47.50 | xheliox | I suspect the vast majority of phone calls going across any network are degraded on some level at this point. |
| 09:49.14 | xheliox | I certainly wouldn't make any absolute conclusions. |
| 09:49.28 | xheliox | 1 + 1 is never 2 anymore. |
| 09:49.52 | WIMPy | No. 1 + 1 is 10 these days. |
| 09:50.41 | WIMPy | dandate2: I guess you might be lucky if both parties are on the same network. Otherwise I'm pretty sure it will go via G.711. |
| 09:51.30 | *** join/#asterisk mgbowman (~mgbowman@78.96.144.26) |
| 09:53.46 | xheliox | Big if. |
| 09:54.31 | ChannelZ | that would be IF then |
| 09:55.10 | mgbowman | I'm having an issue with TLS / SRTP :/ |
| 09:55.19 | mgbowman | everything works to a point |
| 09:55.43 | mgbowman | I can register (TLS) and call into VoiceMailMain / MeetMe / etc... |
| 09:55.49 | WIMPy | TLS day was yesterday :-) |
| 09:55.57 | mgbowman | I know WIMPy :) |
| 09:56.03 | mgbowman | I thought my problem was solved |
| 09:56.17 | mgbowman | I cannot establish a SRTP call between two extensions |
| 09:56.25 | mgbowman | between an extension and asterisk it works fine |
| 09:57.02 | mgbowman | but if I try and dial exten 1 to 2 ... with the Set(_SRTP_CRYPTO=enable) before the Dial |
| 09:57.04 | ChannelZ | sounds like a device problem if they're talking to each other directly |
| 09:57.23 | mgbowman | well I have canreinvite=no |
| 09:57.29 | mgbowman | it should go throu * no? |
| 09:57.34 | WIMPy | Indeed. Is it supposed to work with reinvites at all? |
| 09:57.49 | ChannelZ | what version of asterisk? canreinvite was deprecated |
| 09:57.54 | ChannelZ | it's called 'directmedia' or something now |
| 09:58.07 | mgbowman | and it's _SIPSRTP_CRYPTO ... sorry |
| 09:58.27 | *** join/#asterisk hband (~hband@ip68-2-140-46.ph.ph.cox.net) |
| 10:01.42 | mgbowman | [Oct 17 06:01:01] WARNING[12539]: app_dial.c:2030 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 10:01.43 | mgbowman | <PROTECTED> |
| 10:01.43 | mgbowman | <PROTECTED> |
| 10:01.59 | mgbowman | I understand this as SRTP is unavail |
| 10:03.30 | ChannelZ | it can also mean you have the wrong peer name or typed something wrong |
| 10:03.30 | mgbowman | if the audio goes through * |
| 10:04.06 | mgbowman | does asterisk try to initiate an SRTP channel |
| 10:21.32 | dandate2 | so its not really an issue if the call is transcoded to ulaw from g729 at the ITSP then hacked up to gsm? |
| 10:22.03 | dandate2 | because i tried converting gsm to ulaw via software to be converted to GSM at the cellular level and it sounded like crap |
| 10:22.36 | dandate2 | i.e. pbx decoded gsm to ulaw then sent to itsp |
| 10:24.18 | dandate2 | is there still open source g729 so i can make some tests |
| 10:39.58 | *** join/#asterisk nicola_pav (~chatzilla@mail2.tikalnetworks.com) |
| 10:40.16 | nicola_pav | anyone familiar with sipp? |
| 10:46.38 | tzafrir_laptop | dandre, the name "open source" is misleading there. While that code itself is under a free license, using it requires a non-free Intel library |
| 10:47.07 | tzafrir_laptop | But yes, it's still out there |
| 10:53.16 | mgbowman | so it seems like * is not offering SRTP to the peer |
| 10:53.48 | mgbowman | the extens are both set to always use srtp and i'm getting Status: 406 Not Acceptable from the exten |
| 10:55.02 | *** part/#asterisk saxa (~sasa@host242-95-static.223-217-b.business.telecomitalia.it) |
| 11:17.46 | mgbowman | if I want to co svn ... should I use trunk or branches/1.8 |
| 11:20.37 | tzafrir_laptop | mgbowman, what evrsion do you use now? |
| 11:20.50 | mgbowman | 1.8-rc3 tarball |
| 11:22.01 | mgbowman | I'm testing the TLS + SRTP but still having issues |
| 11:24.02 | mgbowman | I can call into * from 2 extens both using SRTP |
| 11:24.26 | mgbowman | but if I try to call in between the 2 extens (with and without canreinvite) it fails |
| 11:27.12 | *** join/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda) |
| 11:27.24 | mbrevda | anyone here that understands a2billing? (no, I dont need help setting it up) |
| 11:39.39 | mgbowman | ok so it looks like * is not forwarding the crypto in the callee's SDP to the called exten |
| 11:47.52 | mgbowman | sorry the caller's SDP has crypto but the calle's SDP does not - hence the SIP 406 Not Acceptable response |
| 11:53.35 | *** part/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda) |
| 11:58.53 | mgbowman | well what do you know |
| 11:59.10 | mgbowman | sip.conf : encryption=yes |
| 12:01.26 | mgbowman | now I have a TLS issue ... grrr |
| 12:01.31 | *** join/#asterisk af_ (~getsmart@78.134.22.91) |
| 12:01.49 | mgbowman | 1 exten with TLS and 1 exten with TCP |
| 12:02.01 | mgbowman | calls from TLS -> TCP work |
| 12:02.09 | mgbowman | calls from TCP -> TLS ... |
| 12:02.17 | mgbowman | SSL certificate ok |
| 12:02.18 | mgbowman | <PROTECTED> |
| 12:02.18 | mgbowman | [Oct 17 08:01:13] WARNING[32464]: tcptls.c:218 handle_tcptls_connection: FILE * open failed! |
| 12:03.00 | mgbowman | but the peer is already connected via TCP / TLS ... shouldn't it use the already connected socket |
| 12:19.48 | nicola_pav | hello. anyone familiar with sipp? |
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| 12:55.23 | *** join/#asterisk puzzled (~patrick@535335AA.cm-6-4a.dynamic.ziggo.nl) |
| 12:57.46 | mgbowman | anyone using tls here? |
| 13:00.42 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 13:04.46 | nicola_pav | i run moh show classes and i get nth |
| 13:04.48 | nicola_pav | any hint? |
| 13:10.05 | nicola_pav | fixed |
| 13:10.28 | nicola_pav | musiconhold.conf contained malformed entries |
| 13:16.37 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
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| 14:15.43 | mgbowman | ok I'm giving up |
| 14:15.52 | mgbowman | switching to FreeSWITCH |
| 14:17.13 | tzafrir_laptop | mgbowman, AFAIK that in that version TLS should work. Never got to try it myself. If it doesn't, it's a bug |
| 14:17.34 | mgbowman | i've tried rc3 |
| 14:17.40 | mgbowman | i've tried branch/1.8 |
| 14:17.51 | mgbowman | i've tried trunk |
| 14:18.05 | mgbowman | now I get tcptls.c:376 ast_tcptls_client_start: Unable to connect SIP socket to 192.168.1.167:49392: Connection refused |
| 14:18.22 | mgbowman | i can register over tls |
| 14:18.33 | mgbowman | but trying to make 2 calls in between 2 tls extens ... it borks |
| 14:19.16 | mgbowman | MeetMe works fine |
| 14:19.27 | tzafrir_laptop | that's odd |
| 14:19.56 | tzafrir_laptop | meetme, echo test and such work, but bridging to something else doesn't? |
| 14:20.05 | tzafrir_laptop | That something else is a SIP phone? |
| 14:21.10 | mgbowman | yeah ... |
| 14:21.24 | mgbowman | I have 2 iPhones running Bria configured w/ TLS + SRTP |
| 14:21.36 | mgbowman | I can call into MeetMe on both (tls + srtp) |
| 14:21.48 | [TK]D-Fender | mgbowman: disable reinvites <- |
| 14:21.53 | mgbowman | tried both |
| 14:22.07 | mgbowman | I assume it's still canreinvite? |
| 14:22.17 | mgbowman | the configs have changed over the years |
| 14:23.19 | [TK]D-Fender | mgbowman: Changed in 1.6.2+ to "directmedia" I believe |
| 14:24.52 | mgbowman | still the same |
| 14:24.52 | mgbowman | <PROTECTED> |
| 14:25.14 | mgbowman | why would it try to connect as a client |
| 14:25.33 | mgbowman | shouldn't if use the active tcp/tls conx established by the phone |
| 14:26.27 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 14:28.18 | mgbowman | i don't know why it says '192.168.1.167:49392' ... |
| 14:28.22 | mgbowman | netstat -atn shows |
| 14:28.39 | mgbowman | 192.168.1.167:49454 ESTABLISHED |
| 14:29.52 | [TK]D-Fender | different ports... |
| 14:30.23 | [TK]D-Fender | Also check your firewalls |
| 14:30.55 | mgbowman | no firewalls |
| 14:31.17 | mgbowman | so I must not understand sip over tcp correctly |
| 14:31.26 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 14:31.33 | mgbowman | it doesn't use the tcp socket from the register? |
| 14:37.45 | mgbowman | great ... no something completely different |
| 14:37.45 | mgbowman | [Oct 17 10:37:16] WARNING[18454]: tcptls.c:219 handle_tcptls_connection: FILE * open failed! |
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| 14:59.12 | v1s | if I am doing client1 <SIP> serv1<IAX> serv2 <SIP> serv3 should it could it be causing lots of drops? |
| 14:59.17 | v1s | should I just do all sip ? |
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| 15:05.55 | DanFromUK | Hi, can anyone help with an rfc2833 dtmfmode issue? |
| 15:06.26 | DanFromUK | Asterisk is able to process the signals, but I cant get any providers to play the tones to the other party. |
| 15:06.49 | DanFromUK | And if I use inband, my incall features stop working. |
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| 17:20.36 | oryxtec | hi all.. i m getting this error msg on asterisk cli |
| 17:20.37 | oryxtec | ERROR[6402]: cdr_addon_mysql.c:159 mysql_log: cdr_mysql: cannot connect to database server localhost. |
| 17:20.48 | oryxtec | please guide me how resolve this error msg |
| 17:27.14 | *** join/#asterisk a1fa (~awef@unaffiliated/a1fa) |
| 17:27.16 | a1fa | hm |
| 17:27.22 | a1fa | anyone know what does Voxel do? |
| 17:28.43 | *** join/#asterisk b14ck (~rdegges@cpe-24-24-128-47.socal.res.rr.com) |
| 17:28.52 | oryxtec | a1fa: i don't wht it does but i m havinf some issues on asterisk may be you can guied me |
| 17:28.53 | oryxtec | :) |
| 17:28.58 | WIMPy | Walt through space. |
| 17:29.00 | oryxtec | i m getting this error msg on asterisk cli |
| 17:29.03 | oryxtec | ERROR[6402]: cdr_addon_mysql.c:159 mysql_log: cdr_mysql: cannot connect to database server localhost. |
| 17:31.11 | a1fa | oryxtec: nvm.. looks like fring app is effing with me |
| 17:31.16 | a1fa | its getting uninstalled |
| 17:31.34 | oryxtec | how can i fix this? |
| 17:31.38 | oryxtec | plz help! |
| 17:37.26 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 17:39.02 | pabelanger | oryxtec: Seems pretty straight forward 'cdr_mysql: cannot connect to database server localhost.' |
| 17:39.24 | pabelanger | check your credentials |
| 17:39.40 | pabelanger | test your connect outside of asterisk |
| 17:40.02 | drmessano | Error 6402 is "You're not getting very far with Asterisk, maybe you should install Vista and play Farmville instead" |
| 17:40.17 | drmessano | or is that 6403? |
| 17:40.44 | bougyman | the meanness isn't necessary |
| 17:40.58 | bougyman | oryxtec: do you know how to test mysql outside of asterisk? |
| 17:41.04 | oryxtec | no |
| 17:41.15 | oryxtec | how can i test? |
| 17:41.18 | oryxtec | plz guide me |
| 17:41.33 | bougyman | http://support.addy.com/content/databases/command-line/ |
| 17:41.39 | bougyman | mysql -u youruser -h yourdomain.com -p yourdomain_com |
| 17:41.42 | bougyman | that's first. |
| 17:41.44 | drmessano | The lack of being able to take a joke isn't necessary either |
| 17:42.11 | bougyman | replace youruser with the user asterisk is using, yourdomain.com with 'localhost', and yourdomain_com with the password asterisk is using. |
| 17:42.33 | oryxtec | let me try |
| 17:43.22 | bougyman | drmessano: it still makes you irrelevant, to the unbiased observer. |
| 17:43.38 | bougyman | also: TOUCHDOWN STEELERS |
| 17:44.03 | oryxtec | i did this mysql -u root -h localhost -p and it took me to mysql command line |
| 17:44.30 | oryxtec | now wht? |
| 17:44.33 | bougyman | ok, you let asterisk connect as user root? |
| 17:44.57 | oryxtec | yes |
| 17:44.57 | drmessano | bougyman, irrelevant.. Good word.. Are you going to troubleshoot all of his FreePBX install for him too? Because his problem is that he's been breaking and unbreaking this same install for a couple weeks now, and yelling "HELP ME" in 3 different channels. |
| 17:45.03 | bougyman | change the credentials in /etc/asterisk/mysql_cdr.conf to match what you just used, if you want to do that. |
| 17:45.19 | bougyman | drmessano: seems like he doesn't know much about the fundamentals. |
| 17:45.34 | bougyman | you can give him fish, throw fish at him, or teach him how to fish. |
| 17:45.50 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 17:46.25 | bougyman | you can do #2 while I try #3, parallel tactics. |
| 17:46.44 | oryxtec | in mysql_cdr.conf... there is dbname=phonedb and table=admin_cdr |
| 17:46.54 | oryxtec | i don't see this DB in mysql |
| 17:46.55 | oryxtec | :S |
| 17:46.57 | bougyman | oryxtec: doesn't seem to match what you just used on the command line. |
| 17:48.04 | bougyman | http://dev.mysql.com/doc/refman/5.0/en/adding-users.html |
| 17:48.18 | bougyman | spend some time reading documentation for the system you are using. |
| 17:48.32 | bougyman | at the very least, as mysql related questions in a mysql forum. |
| 17:48.40 | bougyman | er ask |
| 17:49.30 | oryxtec | humm |
| 17:49.45 | oryxtec | can i disable mysql_cdr in asterisk? |
| 17:50.33 | pabelanger | don't load it |
| 17:50.37 | drmessano | bougyman, I am glad you think you know so much. I won't hold it against you that you don't completely understand the issue, nor time that I and others have personally put into the same. However, there is a reason for packaged installations like AsteriskNOW, and part of them is to negate the need to spend hours troubleshooting simple setup issues which are handled by the installer, and then broken needlessly. The simple solution is " |
| 17:50.37 | drmessano | reload and stop tweaking" |
| 17:50.54 | oryxtec | let me try |
| 17:50.55 | oryxtec | thanks |
| 17:51.14 | bougyman | drmessano: they don't teach much if you don't break them. |
| 17:51.25 | bougyman | i never learned a thing from winning a chess match. |
| 17:52.31 | drmessano | bougyman, they are not designed to teach.. they are a simple means to a deployment.. Why install from a packaged ISO, then reconfigure the entire install. Is that not the same as "starting from scratch", and does that not negate the purpose of the simplified install? |
| 17:52.57 | *** join/#asterisk mahogany (~root@h49n6-n-a31.ias.bredband.telia.com) |
| 17:53.02 | mahogany | Hello folks |
| 17:54.06 | drmessano | bougyman, you're buying a bike with training wheels already installed, then taking them off, and asking for help reinstalling them. That seems counterproductive to me |
| 17:54.34 | bougyman | hey mahogany |
| 17:55.12 | mahogany | According to various internet sources (google, PDFs, etc) I should be able to jump to n+101 if dial fails because of a busy channel. I have this in my dialplan: exten => 808,1,dial(SIP/EXT-808,20,jt) and further down I have exten => 808,102,dial(SIP/EXT-816,20,jt) but this does not work, I never reach priority 102 (Could I be missing some other configuration option?) |
| 17:57.39 | oryxtec | pbx_ael.c:4531 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael' |
| 17:57.50 | oryxtec | why do i get these warrning on reload |
| 17:58.11 | oryxtec | will these warning effect my system? |
| 17:58.44 | mahogany | oryxtec: That is neither a warning nor error. It simply tells you it has parsed the configuration file. |
| 17:59.04 | oryxtec | humm thanks |
| 17:59.04 | mahogany | http://codepad.org/GDkN5pQU complete section for extension 808 |
| 18:00.11 | mahogany | Bascially, what I want to achieve is that if EXT-808 is busy dial EXT-816. In the future I would like to add to that chain, if EXT-816 is busy dial EXT-817, etc |
| 18:01.54 | mahogany | forgot to check verbose asterisk output and does so now |
| 18:03.22 | pabelanger | mahogany: *CLI> core show application Diall |
| 18:03.26 | pabelanger | look at DIALSTATUS |
| 18:03.44 | pabelanger | *CLI> core show application Dial |
| 18:04.26 | pabelanger | n+101 is not used anymore. |
| 18:04.32 | mahogany | ah! |
| 18:06.25 | *** join/#asterisk EiNSTeiN_ (~einstein@unaffiliated/einstein/x-615171) |
| 18:13.28 | mahogany | pabelanger: The chain will be rather long, i.e. if phone1 is unavailable or buys or whatever, call phone2, then try phone3, then phone4, can I do this using dial() only instead of having dial call all phones at the same time? |
| 18:14.06 | WIMPy | You can use local channels with wait()/ |
| 18:14.30 | pabelanger | mahogany: or setup a queue |
| 18:14.56 | pabelanger | sounds like your trying to build an ACD |
| 18:21.17 | mahogany | pabelanger: I'm fixing a friends Asterisk. He has 6 phones which he wants asterisk to use for incoming calls. But phone2 should only be called if phone1 is busy or unavailable, etc. |
| 18:21.25 | tzanger | mahogany: yep, you can use Dial(sip/phone1&sip/phone2&sip/phone3&zap/3&whatever,,) |
| 18:21.46 | Docteh | that'll ring phone2 every time |
| 18:21.48 | tzanger | personally though if you're going to be doing this for anything other than ringing a few lines you want to set up a queue or something more realistic |
| 18:21.56 | tzanger | Docteh: exactly yes |
| 18:21.59 | pabelanger | mahogany: are all the phone local? |
| 18:22.01 | tzanger | oh |
| 18:22.02 | tzanger | I misread |
| 18:22.07 | pabelanger | otherwise you might want to setup followme |
| 18:22.10 | mahogany | pabelanger: Yes. |
| 18:22.11 | tzanger | I thought he said "or dial them all at the same time" |
| 18:22.17 | mahogany | tzanger: :) |
| 18:23.13 | pabelanger | mahogany: I would setup a queue then, same logic call centres use for agents |
| 18:23.19 | tzanger | I built my own version of followme so that when my wife's friends call the house and nobody answers it rings her phone, my frinds will route to my phone, anyone else can route to a phone if desired, and unknown can be routed similarly |
| 18:23.27 | tzanger | it's all done in astdb |
| 18:24.16 | tzanger | pretty straightforwrd and can be altered on the fly with an IVR |
| 18:31.05 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
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| 19:14.09 | cj | so's |
| 19:14.13 | cj | I've got this old winmodem |
| 19:14.21 | cj | can I use it with asterisk? |
| 19:14.43 | cj | can I make it play with DTFM |
| 19:16.11 | cj | http://www.modem-help.co.uk/Askey/V1456VQH89B-Lucent-Win-Modem.html |
| 19:16.34 | cj | it's got a sound card i/o knob on it |
| 19:16.52 | cj | s/knob/4-pin adaptor/ |
| 19:17.34 | cj | infobot: I didn't ask you to decode it... |
| 19:17.34 | infobot | You didn't ask you to decode it...? |
| 19:17.40 | cj | infobot: no botsnack |
| 19:17.40 | infobot | cj: thanks |
| 19:17.49 | cj | you deserved every one of them |
| 19:25.09 | tzafrir_laptop | cj, completely unrelated I suppose, but: https://reviewboard.asterisk.org/r/977/ |
| 19:29.57 | pabelanger | cj: no not worth it, buy a TDM card from Digium |
| 19:38.00 | *** join/#asterisk neurosys (~neurosys@c-65-34-190-58.hsd1.fl.comcast.net) |
| 19:40.56 | *** join/#asterisk momelod (~smelo@dsl-69-171-155-124.acanac.net) |
| 19:41.13 | momelod | greetings channel |
| 19:42.00 | momelod | im having some dificaulty loading my successfully compiled dahdi module |
| 19:42.19 | momelod | modprod dahdi produces the error: dahdi: disagrees about version of symbol module_layout |
| 19:42.29 | momelod | any thoughts? |
| 19:43.46 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
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| 19:47.15 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:55.34 | tzafrir_laptop | momelod, you should probably unload dahdi itself, and load the dahdi you built as well |
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| 20:36.34 | sabrina22 | hey |
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| 21:23.26 | a1fa | mgcp = teh suck |
| 21:26.45 | *** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 21:46.54 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 21:47.27 | boodu | hello |
| 21:47.57 | boodu | someone use smcdsp-200 phone ? I have problem with moh :O |
| 21:50.09 | [TK]D-Fender | boodu: What does your phone model have to do with MoH? |
| 21:51.15 | boodu | moh works fine with other phone model |
| 21:51.22 | boodu | but no with this |
| 21:51.49 | boodu | there nothing in console when I put the call in hold with this model |
| 21:53.21 | *** join/#asterisk d43mOn (~oscar@host69.190-229-185.telecom.net.ar) |
| 21:53.25 | *** part/#asterisk d43mOn (~oscar@host69.190-229-185.telecom.net.ar) |
| 21:54.37 | boodu | there are an option : Hold by RFC (now to "on") but nothing better |
| 22:06.52 | *** join/#asterisk ritztech (~ritztech@ip65-47-226-86.z226-47-65.customer.algx.net) |
| 22:14.45 | Besticles | I am having problems with MixMonitoring. I setup a extension on my dialplan to execute MixMonitor(test.wav), which works flawlessly. So then I setup a FastAGI command to send which is EXEC MIXMONITOR test.wav, and I am getting errors back: 'Oct 17 14:59:58] WARNING[21662]: file.c:1160 ast_writefile: No such format 'wav |
| 22:14.45 | Besticles | [Oct 17 14:59:58] ERROR[21662]: app_mixmonitor.c:323 mixmonitor_thread: Cannot open /var/spool/asterisk/monitor/test.wav |
| 22:14.55 | Besticles | I dont see what I am doing wrong. |
| 22:21.05 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 22:41.12 | heffer | tzafrir_laptop: i don't know if you remember me but i asked about an TEI assignment error some time ago: chan_dahdi.c: 1 Unable to receive TEI from network in state 2(Assign awaiting TEI)! |
| 22:41.18 | heffer | i have fixed it now by the way |
| 22:41.43 | heffer | patch against your git is here: http://fpaste.org/fBrG/raw/ |
| 22:42.07 | heffer | please feel free to correct the commit message as i don't really have a clue what i was doing there :D |
| 22:42.27 | heffer | nevertheless my dahdi channel works again |
| 22:42.43 | *** join/#asterisk darkdrgn2k3 (~darkdrgnk@bas2-toronto44-1242514582.dsl.bell.ca) |
| 22:43.40 | darkdrgn2k3 | whats the dialstring to append a number to a dialed number? |
| 22:43.57 | darkdrgn2k3 | ie if i dialed 555-555-5555 i want to make it 9 555 555 5555 |
| 22:44.29 | pabelanger | Dial(9${EXTEN}) |
| 22:45.05 | darkdrgn2k3 | hmm thast easy enough LOL |
| 22:47.03 | cj | if I wanted to have asterisk pick up an incoming call on channel 01 (fxo) and play screaming monkeys to the caller, what all do I need to do? |
| 22:47.20 | cj | dahdi_cfg -vv shows the fxo thus: |
| 22:47.24 | cj | Channel 01: FXO Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 01) |
| 22:47.40 | *** join/#asterisk oryxtec (~test@116.71.186.178) |
| 22:47.45 | cj | I guess I would be just as happy with logs showing an incoming call at this point :) |
| 22:48.19 | darkdrgn2k3 | ok im sorry my dialplan is slighly rusty |
| 22:48.20 | darkdrgn2k3 | exten => s,1,Dial(7{$EXTEN}) |
| 22:48.36 | *** join/#asterisk ElvisR (~elvis@horace.dionysian-mind.net) |
| 22:48.54 | darkdrgn2k3 | how do you match everyons in s |
| 22:49.12 | [TK]D-Fender | darkdrgn2k3: huh? |
| 22:49.29 | [TK]D-Fender | darkdrgn2k3: Match what? |
| 22:49.38 | darkdrgn2k3 | the first paramater of exten... |
| 22:50.00 | *** join/#asterisk dandate2 (~gtejkgjke@112.206.130.149) |
| 22:50.04 | [TK]D-Fender | darkdrgn2k3: In "s"? What is that supposed to mean? |
| 22:50.12 | darkdrgn2k3 | exten => s,1,Dial(7{$EXTEN}) <- S |
| 22:50.22 | dandate2 | fender: man you were right double conversion does sound horrible!! i pity da foo that use g729 to call cellular |
| 22:50.24 | pabelanger | darkdrgn2k3: That would dial 7s |
| 22:50.42 | [TK]D-Fender | darkdrgn2k3: that clearly holds a LETTER. So what good does referencing ${EXTEN } in there do you? |
| 22:50.46 | darkdrgn2k3 | where s is the phone number diead? |
| 22:50.48 | [TK]D-Fender | darkit doesn't hold a number. |
| 22:51.03 | [TK]D-Fender | darkdrgn2k3: "s" ... is a letter of the ALPHABET. |
| 22:51.15 | darkdrgn2k3 | [TK]D-Fender: i want to append 7 to a dailed call |
| 22:51.18 | [TK]D-Fender | darkdrgn2k3: Clearly isn't anumber. It doesn't hold ANYTHING. It is a LETTER |
| 22:51.24 | tzafrir_laptop | heffer, I think it is |
| 22:51.28 | pabelanger | darkdrgn2k3: no, s is s. You will want exten => 6135551234,1,Dial(9${EXTEN}) |
| 22:51.36 | [TK]D-Fender | darkdrgn2k3: What dialed call? Where do we see a "dialed call? Append to WHAT? |
| 22:51.42 | darkdrgn2k3 | pabelanger |
| 22:51.49 | darkdrgn2k3 | pabelanger: so what is the 6135551234 argument ? |
| 22:51.52 | ChannelZ | GET OFF MY LAWN! |
| 22:52.09 | heffer | tzafrir_laptop: it is what? |
| 22:52.15 | Besticles | Is cepestral easy to integrate with Asterisk? |
| 22:52.24 | [TK]D-Fender | darkdrgn2k3: Where do YOU see a NUMBER coming from in what you showed us? |
| 22:52.27 | pabelanger | darkdrgn2k3: the number in the context |
| 22:52.58 | pabelanger | You likely want exten => _NXXNXXXXX,1,Dial(9${EXTEN}) |
| 22:53.55 | tzafrir_laptop | heffer, does it actually fix this? |
| 22:54.17 | heffer | tzafrir_laptop: yes it does. if i switch it back i don't receive a TEI on the card |
| 22:54.52 | heffer | with DAHDI_D i do. and i can even phone in and out again |
| 22:55.03 | darkdrgn2k3 | pabelanger: ok i tried that but it seems that the 9 doesnt get diealed :-S |
| 22:55.35 | p3nguin | darkdrgn2k3: Show us what number you called on the phone. Show us the extension. |
| 22:55.46 | pabelanger | darkdrgn2k3: show use the call log |
| 22:55.50 | pabelanger | ~pb |
| 22:55.51 | 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.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. |
| 22:57.04 | heffer | tzafrir_laptop: i see you did exactly the opposite commit on july 31st. |
| 22:57.51 | heffer | and i just recently started using dahdi 2.4.0. before that i didn't compile a new version since 2.3.0.1 |
| 22:58.17 | heffer | and that commit was made after you updated your git to the 2.3 branch |
| 22:59.49 | tzafrir_laptop | yeah, I'm still trying to figure it out |
| 23:05.39 | Tech_Travis | Does * have a common place to store files/scripts run at bootup time? |
| 23:07.30 | darkdrgn2k3 | pabelanger: thanx for all the help... i think i got it now :) |
| 23:07.31 | *** part/#asterisk darkdrgn2k3 (~darkdrgnk@bas2-toronto44-1242514582.dsl.bell.ca) |
| 23:11.10 | [TK]D-Fender | pabelanger: Did he PM it to you? |
| 23:26.05 | *** join/#asterisk csnook (~chris@138.210.3.1) |
| 23:51.25 | *** join/#asterisk EiNSTeiN_ (~einstein@unaffiliated/einstein/x-615171) |