| 00:00.04 | *** join/#asterisk Phil (~phil@2001:0:5ef5:79fd:2835:f17f:26f2:749b) |
| 00:01.39 | Phil | I have an Asterisk server mostly configured with two analogue phones and a SIP phone |
| 00:02.09 | Phil | I'm in the UK so I changed Caller ID Start to polarity on the trunk which allowed the SIP phone to see the CID |
| 00:02.23 | Phil | however the analogue phones do not get it still |
| 00:02.40 | Phil | I'm thinking this is because it's sending the CID incorrectly for UK phones |
| 00:02.44 | Phil | how can I configure this? |
| 00:03.29 | navaismo | http://www.voip-info.org/wiki/view/Asterisk+and+UK+Caller+ID |
| 00:04.08 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 00:04.59 | Phil | thanks, I'll try that |
| 00:05.15 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
| 00:05.44 | *** join/#asterisk SeRi (~wtf@c-76-31-169-54.hsd1.tx.comcast.net) |
| 00:11.44 | *** join/#asterisk k-man (~k-man@unaffiliated/k-man) |
| 00:16.34 | Phil | awesome, navaismo :) |
| 00:16.35 | Phil | thanks |
| 00:17.43 | navaismo | np |
| 00:21.09 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:21.09 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 01:03.26 | FiReSTaRT | anybody with a bit of shorewall knowledge? if i wanna allow all established connections from net to $fw on 5060, i would just put 'ACCEPT net $FW udp 5060' in the ESTABLISHED section? |
| 01:11.20 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 01:13.23 | jpsharp | There's really no concept of "established" connections with UDP, though. |
| 01:13.40 | jpsharp | dunno anything about shorewall, though. |
| 01:16.47 | *** join/#asterisk arii (ari@lol.ebnj.net) |
| 01:18.10 | *** join/#asterisk jsjc (~Adium@77.209.224.3) |
| 01:18.56 | arii | hey.. so i have a digium card with some FXO ports connected to some phones, and everything about asterisk is working fine, but when making a call to/from an FXO phone, after the other side hangs up, i just get a busy signal on the FXO phone instead of a hangup |
| 01:19.08 | arii | the asterisk console seems to think it's hung up the FXO side |
| 01:22.36 | arii | anything i should try? |
| 01:24.59 | *** join/#asterisk volga629 (~slava@76-10-130-18.dsl.teksavvy.com) |
| 01:25.03 | volga629 | ERROR[11559] chan_sip.c: No SRTP module loaded, can't setup SRTP session. |
| 01:25.25 | volga629 | asterisk 1.8.7.1 |
| 01:26.10 | p3nguin | arii: Phones don't connect to FXO ports on the card. Perhaps that's part of the problem. |
| 01:26.28 | volga629 | Asterisk 1.8.7.1 |
| 01:26.31 | arii | then i reversed the description and they're connected to FXS |
| 01:26.40 | arii | but everything about them works fine - they can get a dial tone, make calls, etc. |
| 01:26.43 | p3nguin | Be sure. |
| 01:26.55 | arii | i'm sure |
| 01:39.21 | p3nguin | seri: Are you around? |
| 01:46.10 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 01:50.53 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 02:11.05 | ruben23 | hi guys |
| 02:12.11 | *** part/#asterisk arii (ari@lol.ebnj.net) |
| 02:12.17 | SeRi | p3nguin: waz up |
| 02:12.30 | ruben23 | hi guys can anyone help me how to remove this warning error on my asterisk its populating the whole CLI screen ----> http://pastebin.com/pLCbfMpq |
| 02:13.28 | SeRi | p3nguin: I just got in. Today was my first day at the new job. |
| 02:14.28 | g_r_eek | hello channel, i have an asterisk connnectic behind a att gsm modem and it seems that att using a proxy the registration port changing constantly so the registar server does not find the peer in between this registrations is a short re expire time on the peer a solution to that problem? |
| 02:14.50 | g_r_eek | *reg expire |
| 02:16.00 | sysdef | i have capi running (by Fritz!Box Remote CAPI over TCP) and "capi info" is listing the devices. i expect i will see any reaction (logs) when i call the number but nothing happens |
| 02:18.03 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 02:18.19 | ruben23 | any one have idea guys..? |
| 02:19.50 | *** join/#asterisk alexoc (~alexoc@187.64.254.47) |
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| 02:24.31 | p3nguin | seri: I wanted to go over something with you... |
| 02:24.40 | SeRi | msg me |
| 02:24.46 | p3nguin | Okay. |
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| 02:29.08 | *** join/#asterisk liban (~liban@CPE00222d90c50a-CM00222d90c506.cpe.net.cable.rogers.com) |
| 02:29.20 | liban | hello everyone, |
| 02:29.49 | liban | i am at the end of the road with a problem i have can someone give me a direction?? |
| 02:30.09 | liban | other than jump off a cliff which i am about to do |
| 02:30.28 | p3nguin | ~ask |
| 02:30.29 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 02:30.54 | *** join/#asterisk magicrhesus (~magicrhes@aether.hipocoon.be) |
| 02:31.00 | liban | i have timing slips crc4 erros and e-bit erros |
| 02:31.24 | liban | no irq misses |
| 02:31.57 | liban | everything is configured correctly on the span |
| 02:32.07 | liban | double and triple checked by digium techs |
| 02:32.36 | liban | pap loop test didnt pass |
| 02:32.49 | liban | cyclic test was good |
| 02:33.07 | liban | any ideas what to do |
| 02:33.40 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 02:36.40 | liban | anyone? |
| 02:40.30 | *** join/#asterisk Gugge (gugge@kriminel.dk) |
| 02:49.31 | liban | if anyone can help that'd be great |
| 02:51.56 | WIMPy | sysdef: I've done it once, just to see if it's possible. have you tried all interfaces? |
| 02:59.30 | *** join/#asterisk tomaw (tom@freenode/staff/tomaw) |
| 03:03.59 | liban | CRC4 error count: 178002 E-bit error count: 188676 Timing slips: 20 |
| 03:04.02 | liban | thats what i get |
| 03:05.20 | WIMPy | That looks really bad. Is the timing source configured correctly? |
| 03:05.44 | WIMPy | Is it only one port or multiple? |
| 03:05.53 | liban | yes and its multiple ports |
| 03:05.55 | liban | dual span |
| 03:06.39 | WIMPy | Both with errors? |
| 03:06.47 | liban | jst one is enable rright now |
| 03:06.54 | liban | span 1 to be more specific |
| 03:07.32 | WIMPy | Is the cabling correctly terminated? |
| 03:08.02 | liban | hmm i would imagine so |
| 03:08.30 | liban | are u referring to the T1 circuit |
| 03:09.02 | WIMPy | It's a T1? |
| 03:09.18 | liban | PRI ... T1 yes |
| 03:09.42 | WIMPy | In that case I wonder what E bits are meant by that message. |
| 03:10.35 | WIMPy | was thinking of bits in the E-channel. |
| 03:10.50 | liban | Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) B8ZS/ESF ClockSource CRC4 error count: 193423E-bit error count: 205013Timing slips: 21 |
| 03:11.00 | liban | thats what i get now |
| 03:11.03 | liban | its going crazy |
| 03:11.32 | WIMPy | yes |
| 03:12.33 | liban | any ideas? |
| 03:12.42 | liban | or relse ima get my dick chopped off tomorrow |
| 03:12.51 | liban | ima constantly have to restart this box a million times tomorrow |
| 03:13.06 | liban | during business hours :( |
| 03:13.18 | WIMPy | You are able to make calls? |
| 03:13.33 | liban | yes but static and clicks |
| 03:13.48 | liban | and its cut off after sometime |
| 03:14.35 | WIMPy | Not too bad. |
| 03:14.42 | liban | ive done everything with the digium tech support, i thoght id get a second opinion |
| 03:15.05 | liban | timing slips will go to over 200 |
| 03:15.20 | WIMPy | Do you have any other equipment to check if the line is actually in a usable state? |
| 03:15.42 | liban | u know |
| 03:15.50 | liban | thats a good idea |
| 03:15.51 | liban | lol |
| 03:15.54 | liban | i do but |
| 03:16.28 | liban | i doubt id be able to make calls out, i havent configured the system i dont have access to it |
| 03:16.37 | liban | i'll try that thought |
| 03:16.40 | liban | anything else? |
| 03:17.56 | WIMPy | If the timing is correct, I'd expect it to be somethign on the T1 side of the card. |
| 03:18.23 | liban | have u heard of digium cards going bad cant keep timing? |
| 03:18.30 | *** join/#asterisk [ALT][F4] (~altf4@183.89.92.232) |
| 03:18.31 | liban | cause i doubt its the telecom side |
| 03:18.44 | WIMPy | No |
| 03:18.46 | liban | if their timing is off then a million other customers be complaining |
| 03:19.23 | WIMPy | Timing is just one possible cause of the problem. |
| 03:19.27 | liban | well is the timing the root cause? |
| 03:19.49 | liban | and the crc4 e-bit erros are a result i assume right? |
| 03:20.01 | WIMPy | It can be any kind of transmission error. |
| 03:20.45 | liban | i see |
| 03:21.36 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 03:22.15 | liban | ill try to change |
| 03:22.20 | liban | the phone system to check |
| 03:22.28 | liban | any other ideas? |
| 03:22.47 | WIMPy | nope |
| 03:23.25 | liban | VGA and Digium card on same IRQ .... no irq misses, u think it can still be a problem? |
| 03:24.23 | WIMPy | If it is, it should tell you. |
| 03:24.32 | WIMPy | Did you try dahdi_test? |
| 03:26.06 | liban | yea looks good |
| 03:27.20 | [ALT][F4] | does anyone have a clue how to "apend" numbers to CID on incomming SIP Trunks ? |
| 03:27.58 | liban | are u using this for incoming fax? |
| 03:28.18 | WIMPy | Set(CALLERID(num)... |
| 03:28.57 | [ALT][F4] | WIMPy: i try that already .. but somehow it shows either nothing or only the number (probably caused by my stupidity in setting it up) :P |
| 03:29.23 | [ALT][F4] | i sinply want to hardcode the country code on incomming calls and strip the leading 0 |
| 03:30.23 | WIMPy | ${cc}${CALLERID(num):1} |
| 03:32.02 | liban | thank you |
| 03:32.21 | liban | wimpy for ur help |
| 03:33.18 | WIMPy | That e bit still puzzels me |
| 03:33.33 | liban | what is e-bit? |
| 03:33.39 | liban | im so clueless with this stuff |
| 03:34.20 | WIMPy | NFI. I thought of bits from the E-channel. But I don't think a PRI has one. |
| 03:34.30 | WIMPy | Still trying to find out. |
| 03:38.28 | [ALT][F4] | thanks . that looks very simple ;) |
| 03:38.54 | [ALT][F4] | now just have to get rid of the timeout issues .. |
| 03:38.57 | [ALT][F4] | to test it |
| 03:39.48 | [ALT][F4] | does anyone know a good advice on how to set router (prefered routeros) to have asterisk connect nicely to remote SIP Trunks ? |
| 03:41.53 | *** join/#asterisk dimbulb (~totallyre@tacomatelematics.com) |
| 03:43.05 | [ALT][F4] | since few days having "NOTICE[204]: chan_sip.c:7810 in sip_reg_timeout: -- Registration for '9413091@sipgate.de' timed out, trying again (Attempt #3553)" in the logs … sure something about the NAT and routes |
| 03:43.30 | [TK]D-Fender | ~sipnat |
| 03:43.30 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 03:43.32 | [TK]D-Fender | ^^ |
| 03:43.57 | [ALT][F4] | routeros have some helper system for SIP behind nat… but i think it screws things up instead of make it smooth |
| 03:44.15 | WIMPy | They all do |
| 03:45.53 | [ALT][F4] | it worked perfectly with my SIP Phone for years … but asterisk seems to have some trouble |
| 03:45.56 | dimbulb | This may seem like a dumb question, but my google is weak. How do I direct calls to chan_alsa in a dialplan? |
| 03:46.32 | WIMPy | console/alsa |
| 03:47.34 | dimbulb | thanks! |
| 03:48.53 | *** join/#asterisk Guest8383 (~Geek@unaffiliated/cain) |
| 03:56.20 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
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| 03:59.33 | *** join/#asterisk Synoptic (~synoptic@modemcable171.137-83-70.mc.videotron.ca) |
| 03:59.38 | Synoptic | Hi there |
| 04:00.04 | Synoptic | Hi there, i'm trying to display the value of ${CALLERID(name)} in the log |
| 04:00.25 | Synoptic | what is the fucntion to do that ? I'm in from-pstn-custom context for that |
| 04:00.39 | Synoptic | which loads first (almost) when a call comes in. |
| 04:01.07 | Synoptic | I'd like to know what is that value when someone is blocking it's CID name, because I need to do something with that variable after |
| 04:02.46 | Synoptic | NoOP |
| 04:02.59 | *** join/#asterisk liban (~liban@CPE00222d90c50a-CM00222d90c506.cpe.net.cable.rogers.com) |
| 04:04.27 | liban | CRC4 error count: 277782 E-bit error count: 299398 1TE2/0/1/1 Clear (In use) YELLOW (EC: VPMOCT064 - INACTIVE) |
| 04:04.45 | liban | this is what i get now... crazyyy |
| 04:04.52 | liban | no one using the system |
| 04:05.27 | WIMPy | If the link is broken, it's broken. No matter what the users do. |
| 04:06.39 | liban | can i ask the telecom to do something from their end any test to suggest? |
| 04:07.18 | WIMPy | Sure, but only to the NT. |
| 04:09.26 | liban | NT? |
| 04:09.50 | *** join/#asterisk NetNut404 (~Net@stephenson.cc) |
| 04:11.30 | WIMPy | The network terminator. |
| 04:13.28 | liban | well.. thanks wimpy i appriciate it |
| 04:15.23 | NetNut404 | I need some info on troubleshooting the email notifications of asterisk .. logs say in debug the message was sent, but it never reaches the mail server logs.. but from from what I read in the source code file it says it would complain if it had trouble writing the file in /tmp that it mails correct ? |
| 04:16.04 | NetNut404 | I guess might make a different on the version.. it's 1.8.7 (centos rpms) |
| 04:25.19 | *** join/#asterisk meest (~alpaca@155-23-88-118.dcwest.net.au) |
| 04:29.31 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 04:49.26 | NetNut404 | I am quite lost in telling where to look for why emails are not being sent even though the logs claim they are |
| 04:54.23 | ChannelZ | are you talking about voicemails? |
| 04:57.03 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 04:58.34 | NetNut404 | yup |
| 04:59.01 | NetNut404 | the emails for notifications of voicemails.. or the voicemails themselves |
| 05:02.09 | ChannelZ | asterisk doesn't contain a mailer, it just calls some other commandline mta - default sendmail. See voicemail.conf mailcmd I think |
| 05:04.23 | ChannelZ | if your server isn't setup to do mail, you can get something like msmtp and have it send to some other SMTPD that can deliver it from there. That's what I do as my Asterisk box does not have a mail server running on it |
| 05:04.35 | ChannelZ | (nor do I want one on it) |
| 05:08.25 | meest | Does dahdi allocate memory for the dahdi_chan.writechunk? |
| 05:12.38 | NetNut404 | right it says it send with sendmail -t |
| 05:13.41 | NetNut404 | s/send/sent |
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| 05:46.00 | meest | Where can I found dahdi documentation? |
| 05:49.56 | *** join/#asterisk Lipsum (~lipsum@77.40.154.242) |
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| 05:58.38 | kaldemar | meest: http://docs.tzafrir.org.il/ |
| 05:59.05 | kaldemar | meest: other than that, DAHDI is kind of poorly documented. |
| 05:59.41 | meest | yeah, I have noticed that. I have read that document many times lol |
| 06:00.06 | meest | I have pretty much everything set up, got the channels registered and all that |
| 06:00.31 | meest | but i cant seem to access the write and readchunk stuff in the channels |
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| 06:58.48 | Prabalan | Hi all |
| 06:59.02 | Prabalan | I am getting an error in Asterisk |
| 06:59.06 | Prabalan | ERROR[3806]: chan_sip.c:13072 register_verify: Peer 'prasanth' is trying to register, but not configured as host=dynamic |
| 06:59.39 | Prabalan | In sip i have configured 'prasanth' as its IP address |
| 07:00.21 | kaldemar | why? |
| 07:02.14 | Prabalan | http://pastebin.com/913qpME9 |
| 07:02.25 | Prabalan | This is my sip conf. |
| 07:03.14 | Prabalan | I order for the asterisk to know the peer, i have done like that |
| 07:03.24 | Prabalan | Any problem there |
| 07:05.41 | kaldemar | Prabalan: ip addres is just one way to know a peer. |
| 07:06.14 | Prabalan | Ok. |
| 07:06.30 | kaldemar | Prabalan: you should either define it as dynamic or configure the peer to not register to you. |
| 07:06.31 | Prabalan | But why am getting the error i mentioned above |
| 07:06.40 | Prabalan | ERROR[3806]: chan_sip.c:13072 register_verify: Peer 'prasanth' is trying to register, but not configured as host=dynamic |
| 07:07.08 | Prabalan | so do i have to change the host to dynamic |
| 07:07.09 | kaldemar | registrations are only allowed for devices that are defined as dynamic. |
| 07:08.28 | Prabalan | My device 'prasanth' is an IP phone. I have set it not as dynamic. |
| 07:08.43 | Prabalan | So my question is still why it is trying to register |
| 07:09.10 | *** join/#asterisk s[X] (~s_x_@eth589.qld.adsl.internode.on.net) |
| 07:09.28 | *** join/#asterisk felipe_ (~felipe@unaffiliated/felipe) |
| 07:09.50 | kaldemar | Prabalan: i don't even know what your phone is. |
| 07:14.57 | Prabalan | My phone is an IP Polycom Phone. Do i have to set in the phone setting that it is not dynamic |
| 07:16.23 | Prabalan | kaldemar: Am not able to find any settings in the IP phone related to registration |
| 07:18.24 | kaldemar | Prabalan: the option is called "register" in the GUI and reg.X.server.Y.register in the XML configs. |
| 07:18.37 | kaldemar | Prabalan: however, there's no reason to disable registration for a phone. |
| 07:19.04 | kaldemar | just configure the peer as dynamic. |
| 07:19.35 | Prabalan | ok i will configure the peer as dynamic |
| 07:23.49 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 07:26.19 | NetNut404 | I dont get the voicemail here.. the mailcmd does nothing for notification I even put a small script there instead of mail to see if it ran.. and it didnt.. but logs on debug say they do |
| 07:33.17 | *** join/#asterisk s[X] (~s_x_@eth589.qld.adsl.internode.on.net) |
| 07:40.25 | ChannelZ | Your script might have failed for any number of reasons (like permissions, or environment) |
| 07:40.38 | ChannelZ | Do your mail logs show the mail or an error? |
| 07:42.37 | *** join/#asterisk roham (~ali@31.184.187.2) |
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| 07:52.46 | NetNut404 | no |
| 07:52.52 | NetNut404 | the mail logs show nothing |
| 07:52.59 | NetNut404 | no connection at all |
| 07:53.19 | NetNut404 | and a echo to a file in /tmp should never fail |
| 07:53.25 | NetNut404 | it would not even do that |
| 07:54.27 | NetNut404 | but the debug asterik log claims it runs.. there is never any evidence of it running |
| 07:57.07 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
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| 08:13.04 | ChannelZ | not sure what to say |
| 08:13.09 | ChannelZ | mailcmd=/bin/echo farting >/tmp/farting |
| 08:13.14 | ChannelZ | works here |
| 08:14.06 | *** join/#asterisk Jimmy890 (~chatzilla@92.60.238.0) |
| 08:16.20 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 08:16.27 | elliot98 | politely enteres |
| 08:17.22 | elliot98 | ok, so the Asterisk server needs to authenticate with every call it sends to a certain provider, so using the register directive in sip.conf is not good enough...where do username/password authentication go? |
| 08:18.29 | elliot98 | would be adding username/password in the peer details enough? |
| 08:22.18 | kaldemar | elliot98: registration has little to do with call authentication. put credentials in the peer definition. |
| 08:23.20 | *** join/#asterisk stix (~stix@193.89.191.209) |
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| 08:30.21 | krotos | hi al : |
| 08:30.25 | krotos | hi all : |
| 08:32.20 | elliot98 | kaldemar: ok, good, thanks...just to understand authentication a bit better, the username/password inside either a peer, user, or friend, works in either direction for call authentication? |
| 08:33.02 | elliot98 | if the asterisk server places a call and needs to be authenticated, it will use those credentials, and if an incoming call needs credentials (ie. without insecure=invite set), it will use the same credentials |
| 08:33.18 | elliot98 | greetings krotos |
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| 08:41.25 | kaldemar | elliot98: simplified. depending what you dial, fromuser may also be needed. |
| 08:44.38 | elliot98 | kaldemar: ok, so if the remote server and local Asterisk server needs different credentials for authenticating a call, one should create a separate peer and user account? |
| 08:44.52 | krotos | i'm upgrading a old server that use a openvox B400P/B200P (with 2 bri port). Is correct if i load the wcb4xxp for this card and blaclist hfcmulti and msISDN? |
| 08:48.48 | kaldemar | elliot98: or define remotesecret for the peer. |
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| 09:11.24 | elliot98 | kaldemar: aha..ok, making sense now, basically, the remote will be associated with the particular peer according to IP address, but if authetication is needed, it will look the remotesecret. |
| 09:19.09 | kaldemar | elliot98: see "Naming devices" here: http://svn.digium.com/svn/asterisk/tags/10.0.1/configs/sip.conf.sample |
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| 10:08.35 | jacc0 | it's a sad day for the internet in the netherlands |
| 10:08.43 | jacc0 | the strated blocking websites |
| 10:08.46 | jacc0 | *they |
| 10:08.54 | jacc0 | :S |
| 10:09.17 | jacc0 | so much for my free country |
| 10:10.34 | cusco | jacc0: :( |
| 10:10.53 | cusco | produce a public letter |
| 10:11.06 | cusco | and make people send a copy of it to their local MP's |
| 10:11.34 | jacc0 | there is a lot of ongoing action |
| 10:11.57 | jacc0 | mirrors of blocked websites keep poping up @ a raid of about 40 an hour |
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| 10:20.20 | krotos | hi, someone have used a openvox b400p/b200p? |
| 10:20.47 | krotos | what module should i use? if i load the wcb4xxp and blacklist hfcmulti and msISDN is correct? |
| 10:22.35 | *** join/#asterisk [ALT][F4] (~altf4@ppp-58-11-96-124.revip2.asianet.co.th) |
| 10:23.11 | [ALT][F4] | anyone an idea how to obtain the user-status with asterisk-gui ? (what ajax call is doing that) |
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| 10:27.32 | [ALT][F4] | ah found it .. |
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| 11:19.54 | Prabalan | HI ALL |
| 11:20.28 | Prabalan | i am usinf an fxo/fxs card TDM410P |
| 11:20.55 | Prabalan | It is having 2 ports active; port 1 and 4 |
| 11:22.15 | Prabalan | Port one i have connected to an analog phone at my desk and to port 4 i have connected to the intercom line of my office |
| 11:23.59 | Prabalan | Call to my analog from intercom i have configured. But my analog phone is not giving a ring |
| 11:26.48 | Prabalan | This is my chan_dahdi.conf: http://pastebin.com/HWMQ8YZd |
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| 11:27.09 | Prabalan | Plz have a look and give suggestion |
| 11:30.32 | Prabalan | ... |
| 11:30.35 | Prabalan | Hi |
| 11:31.30 | Prabalan | How to configure my analog phone connected to FXO/FXS card for getting calls from intercom line..???? |
| 11:31.34 | Prabalan | Plz help |
| 11:38.27 | Prabalan | Any more details needed?? |
| 11:42.27 | kaldemar | do you get a call when DAHDI/1 is dialed and you pick up the handset? |
| 11:42.49 | kaldemar | can you dial from the phone? do you get dialtone when you pick up the handset? |
| 11:44.21 | Prabalan | When DAHDI/1 is been called am getting ring in asterisk but not from by analog phone which is connected to port 1 |
| 11:45.09 | Prabalan | No my actuall problem is that am not getting the dial tone in my analog phone connected to PORT 1 of FXO/FXS card |
| 11:51.16 | Prabalan | This is my extension |
| 11:51.18 | Prabalan | http://pastebin.com/eWQ1qBfK |
| 12:01.58 | kaldemar | Prabalan: is the port really an FXS? |
| 12:02.39 | kaldemar | do you see the channels in asterisk with "dahdi show channels"? |
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| 12:13.26 | Prabalan | dahdi show channel o/p: http://pastebin.com/ff3suKk0 |
| 12:15.02 | Prabalan | The port is FXS itself |
| 12:15.51 | Prabalan | This is the auto generated system.conf file: http://pastebin.com/UpqiUAAr |
| 12:16.25 | Prabalan | Auto generated by the command dahdi_genconf -v |
| 12:16.44 | kaldemar | and channel 4 works? |
| 12:16.57 | Prabalan | Ya channel 4 is working fine |
| 12:17.07 | kaldemar | maybe the phone is not properly connected, the cable is bad, the module is broken or something else. |
| 12:17.12 | Prabalan | am able to call out to my intercome using a soft phone |
| 12:17.30 | Prabalan | No that all part i have properly checked |
| 12:17.38 | Prabalan | that is not the reason |
| 12:17.41 | kaldemar | with other equipment? |
| 12:18.21 | Prabalan | ya i have cross checked |
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| 12:18.44 | Prabalan | do we get the ring tone normally once phone is connected to the port?? |
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| 12:26.07 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 12:27.09 | aberrios | Hey folks. Is there a way of going to the previous step in the dial-plan? I have an i extension that will catch invalid input from a number of steps, and I want it to go back to the step that was last...if that makes sense |
| 12:29.30 | jacc0 | use labels ? |
| 12:29.46 | jacc0 | goto(label) ? |
| 12:30.11 | aberrios | but i wont know which label to go to necessarily..... |
| 12:30.13 | aberrios | unless |
| 12:30.17 | aberrios | I set a variable |
| 12:30.22 | aberrios | that could work |
| 12:30.34 | aberrios | goto(${labelvar}) |
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| 12:32.09 | kaldemar | aberrios: how would you not know? |
| 12:34.09 | aberrios | kaldemar, I would if I set a variable. But for example I have 0,1,SayText("Press 0 for a taxi from XYZ") and s,n,SayText(If you would like to speak to an operator, press 1) any invalid input from these would to to i,1, but extension i would know which step the invalid input came from. |
| 12:34.34 | aberrios | wouldnt know* |
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| 12:35.49 | kaldemar | maybe you should handle the input validation in the extension itself, and not let it jump to i. |
| 12:36.35 | aberrios | kaldemar, i did start off that way, but its easier and less messy using the interrupt provided by SayText("Text",i) |
| 12:36.56 | aberrios | i just needed to get around the i extension knowing where to send the caller back to |
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| 12:55.10 | aberrios | hm this could be useful ${INVALID_EXTEN} |
| 12:56.11 | aberrios | or maybe not |
| 13:00.56 | leifmadsen | aberrios: what are you trying to do? |
| 13:01.57 | aberrios | leifmadsen, its okay i've got it. I've set a variable in each extension, like exten => 0,1,Set(position=${EXTEN},${PRIORITY}) and then i looks like |
| 13:02.04 | aberrios | exten => i,1,SayText("Sorry that is an invalid option, please try again") |
| 13:02.04 | aberrios | exten => i,n,GoTo(${position}) |
| 13:02.34 | leifmadsen | aberrios: that's a GoSub() |
| 13:02.40 | leifmadsen | pretty much anyways |
| 13:02.50 | aberrios | GoSub eh,,, I'll look it up |
| 13:03.10 | leifmadsen | I'm not sure I like the idea of placing a comma in the channel variable name. |
| 13:03.15 | leifmadsen | s/name/value/ |
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| 13:06.00 | aberrios | leifmadsen, hmm I'm not sure how i'd use GoSub in my IVR.. I know I'd need i to look like exten => i,1,Return but I'm not sure where the GoSub would be in each of my extensions... maybe 0,1,GoSub()?? |
| 13:06.36 | leifmadsen | There are other tools you should probably be using, like GoSub(), GoSubIf(), DIALPLAN_EXISTS(), etc. |
| 13:06.53 | leifmadsen | I think you're relying on the 'i' too much |
| 13:06.53 | aberrios | leifmadsen, I'm using SayText("Wibble",i) and they give an invalid extension here,,where does the GoSub come into it? |
| 13:07.17 | leifmadsen | For 'i' in an auto-attendant, I usually just return to the top, or near the top and give them the menu again |
| 13:07.30 | leifmadsen | exten => aa,n(start),NoOp() |
| 13:07.31 | leifmadsen | <PROTECTED> |
| 13:07.39 | leifmadsen | exten => i,n,Goto(aa,start) |
| 13:07.56 | aberrios | leifmadsen, It did cross my mind that'd be easy. But I'd rather not people go straight to the top of it and have to listen to a load of stuff again just for accidently pressing the wrong button |
| 13:08.02 | leifmadsen | aberrios: I came in well after when you described your problem |
| 13:08.13 | aberrios | leifmadsen, :) |
| 13:08.38 | leifmadsen | ya, I don't like you using the comma though. I'd prefer to see it changed to a hyphen or something, then use CUT() to return |
| 13:09.10 | leifmadsen | exten => i,n,GotoIf($[${EXISTS(${position})}]?${CUT(position,-,1),${CUT(position,-,2)}) |
| 13:09.16 | leifmadsen | given that is significantly crazier |
| 13:09.47 | leifmadsen | just that the method you're using is an old parlance for multi-set |
| 13:10.02 | leifmadsen | (which is now MSet() |
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| 13:49.01 | sekil | hi |
| 13:49.17 | sekil | I'm still on * 1.2...have to move for one installationt to 1.6+ version |
| 13:49.28 | sekil | so the answer is ...do I use 1.6 or 1.8 version? |
| 13:49.43 | sekil | I need to configure digium b410p with dahdi/libpri |
| 13:49.53 | sekil | err answer=question |
| 13:50.55 | leifmadsen | sekil: 1.6.0 and 1.6.1 are end-of-life, and 1.6.2 is security fixes only (about another 5-6 months). 1.8 is LTS and currently supported. |
| 13:50.58 | leifmadsen | ~asterisk-versioning |
| 13:50.58 | infobot | it has been said that asterisk-versioning is http://blogs.asterisk.org/2009/06/24/about-the-new-asterisk-versioning-method/ |
| 13:51.06 | leifmadsen | ~asteriskversions |
| 13:51.30 | leifmadsen | ~asterisk-versions |
| 13:51.30 | infobot | Information about Asterisk maintenance support and when branches will move into security fix only mode, and eventually end-of-life is available at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 13:51.44 | leifmadsen | sekil: ^^^ you can make a decision based on that |
| 13:52.04 | sekil | hm..I guess 1.8.x is the answer |
| 13:52.39 | sekil | any known issues with digium b410p bri isdn libpri/dahdi? |
| 13:52.48 | sekil | currently there is 1.2 with misdn |
| 13:52.52 | sekil | on the site |
| 13:53.05 | leifmadsen | can't answer as he doesn't use that hardware |
| 13:53.45 | leifmadsen | sekil: either way, you're going to build on a separate devleopment box first and test, because upgrading a 1.2 production box to 1.8, without prior testing, isn't something you would ever consider thinking |
| 13:54.05 | sekil | that's going to be done |
| 13:54.13 | sekil | I'm just asking what would new customers do |
| 13:54.26 | sekil | go with 1.8 or 1.6 |
| 13:54.41 | sekil | but you answered that indirectly |
| 13:54.58 | leifmadsen | right, it would be silly to deploy a non-supported version of software |
| 13:55.07 | sekil | right |
| 13:55.27 | sekil | thanks |
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| 13:59.12 | leifmadsen | sekil: also fyi, there is no 1.6; it's 1.6.0, 1.6.1, and 1.6.2 (all major versions) |
| 13:59.53 | sekil | I see...I haven't been following the development recently |
| 14:00.25 | sekil | anyhow..I downloaded the newest 1.8.X and will start from there |
| 14:00.40 | leifmadsen | that's why you'll see them outlined on the asterisk versions table separately |
| 14:01.06 | sekil | oh yes |
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| 14:02.00 | leifmadsen | it was an experiment which didn't work as planned (there was supposed to be more frequent major releases, but that didn't appear to be possible |
| 14:02.09 | leifmadsen | so 1.8 was born, much like 1.2, 14, etc. |
| 14:02.15 | leifmadsen | s/14/1.4/ |
| 14:09.11 | sekil | so 1.6.x should not be used |
| 14:09.29 | leifmadsen | unless you already have it deployed, it doesn't make sense, so no. |
| 14:09.38 | sekil | yes |
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| 14:09.53 | leifmadsen | I like 1.8 a lot, and it has some nice features |
| 14:10.09 | leifmadsen | and the latest O'Reilly book is written against 1.8 (there is no 1.6.x based book) |
| 14:10.16 | sekil | I have couple of 1.2 boxes with pris where ISDN aoc is used...there are some patches involved to make AoC work |
| 14:10.32 | sekil | but I see that ISDN AoC support in libpri is more or less complete now.. |
| 14:10.35 | leifmadsen | gotcha, hopefully using DAHDI and 1.8 makes it so you don't need to |
| 14:10.43 | sekil | correct |
| 14:10.51 | leifmadsen | ya, I think it is, but I'm uncertain as I haven't deployed AoC stuff |
| 14:10.56 | leifmadsen | it sounds familiar though |
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| 14:51.21 | rv | hi |
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| 14:53.51 | rv | anyone know how to do remote restart polycom 321? |
| 14:54.35 | rv | via web conf page... when i change something and then submit.. phone save configuration but without restart |
| 14:55.45 | WIMPy | sekil: It took me some time to find the parameter AOCenable, but after I found that, it actually worked. |
| 14:56.07 | kaldemar | rv: *CLI> sip notify polycom-check-cfg |
| 14:56.47 | ruben23 | hi guys how do i disable warning on asterisk CLI..? |
| 14:57.03 | [TK]D-Fender | ruben23, "exit" |
| 14:57.06 | kaldemar | ruben23: all warnings? |
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| 14:58.10 | rv | kaldemar: ? from rasterisk? |
| 14:58.32 | rv | but this phone is not yet registered |
| 14:58.49 | rv | i have access via polycom web page |
| 15:01.16 | kaldemar | rv: yes, from asterisk. assign an ip for it and it doesn't matter if it's registered or not. |
| 15:01.41 | kaldemar | rv: or just send it a SIP notify with "Event: check-sync" as you wish. |
| 15:01.46 | ruben23 | kaldemar:yes, is it possible to limit to a particular warning only..? |
| 15:02.13 | sekil | WIMPy: the one in zapata.conf...or whatever is the name now? |
| 15:02.20 | kaldemar | ruben23: short answer, no. |
| 15:02.35 | kaldemar | ruben23: what exactly do you mean by "warning"? |
| 15:02.40 | rv | kaldemar: ill try. thanks. |
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| 15:03.14 | WIMPy | sekil: chan_dahdi.conf, yes |
| 15:03.39 | sekil | WIMPy: does it work when sip side hangs up first? |
| 15:04.59 | ruben23 | kaldemar: this one ---->http://pastebin.com/pLCbfMpq |
| 15:05.31 | WIMPy | sekil: There is an option to delay hangup, but I haven't tried it. I only checked AOC-D. |
| 15:05.54 | sekil | WIMPy: that's to wait E |
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| 15:06.04 | WIMPy | yes |
| 15:06.46 | sekil | there was an issue before when SIP side hangs up ...there was no aoc in dp/cdr....because it wouldn't wait for RELEASE COMPLETE on the isdn side...but killed the call |
| 15:07.13 | sekil | structure needed to be expanded |
| 15:08.00 | WIMPy | Yes, well Asterisks architecture doesn't really allow for anything to happen after a hangup. |
| 15:08.32 | sekil | there's no reporting state or something similar |
| 15:08.43 | WIMPy | Nope |
| 15:08.52 | WIMPy | And as I said, I haven't tried that delay thing. |
| 15:09.02 | sekil | I don't think delay thing is the issue... |
| 15:09.17 | sekil | that's just to pickup E data |
| 15:09.40 | sekil | test would be to make a call..start receiving data from isdn aoc-s...and hangup from sip side.. |
| 15:09.48 | sekil | and see if it's there or not ..in the cdr |
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| 15:12.15 | sekil | WIMPy: looking at chan_sip.c in 1.8.x ..there have been patches |
| 15:12.51 | [TK]D-Fender | ruben23, fix your file |
| 15:16.16 | kaldemar | ruben23: you could always remove the warnings from format_wav.c:140, file.c:386 and res_musiconhold.c:261 but that would be idiotic. instead fix the problem you've made in your configuration or file usage. |
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| 15:26.53 | Katty | i can haz caffeines now plz? |
| 15:27.21 | pigpen | not only can you have it, it is highly recommended. |
| 15:27.32 | Katty | :>>>> |
| 15:27.40 | WIMPy | Drugs are bad! |
| 15:28.07 | pigpen | no doubt I need more (drugs and caffeine) |
| 15:28.17 | pigpen | (good drugs, not bad drugs) |
| 15:28.27 | pigpen | I guess that is relative. |
| 15:28.59 | leifmadsen | drugs are awesome, mmmk |
| 15:29.03 | [TK]D-Fender | Stay in milk! |
| 15:29.07 | [TK]D-Fender | Don't do school! |
| 15:29.12 | [TK]D-Fender | Drink your drugs! |
| 15:29.27 | Katty | how can you have any pudding if you don't, drink your...caffeines? |
| 15:30.01 | pigpen | I had a friend who said he could write code better when he was high. Now, he is a proud resident in Huntsville Prison. ;-) |
| 15:30.27 | Katty | those kinds of drugs are bad. |
| 15:30.47 | pigpen | heh, for your career. |
| 15:31.01 | pigpen | life, health, etc... |
| 15:31.08 | *** join/#asterisk ccesario (~ccesario@189.29.54.144) |
| 15:31.15 | [TK]D-Fender | Oh please... prescription drugs kill infinitely more people than all illegal ones combines... |
| 15:31.47 | [TK]D-Fender | Lets not even start on cigarettes & alcohol... |
| 15:31.54 | pigpen | true. true. But you will do more time for the illegal ones. (if you get caught that is) |
| 15:32.25 | pigpen | Coffee for that matter. Legalized addictive drink. (speaking of which, I need more...bad.) |
| 15:32.30 | Katty | [TK]D-Fender: that also falls into 'those kinds of drugs' |
| 15:33.10 | pigpen | [TK]D-Fender, I hope you are proud. I am talking with you and not even asking you a single question. Is that improvement or what? |
| 15:33.21 | pigpen | Dam. there's that question. |
| 15:33.32 | Katty | i'd just ignore fender. |
| 15:33.33 | [TK]D-Fender | pigpen, yes, the point was the reverse scale of demonization and criminalization... |
| 15:33.42 | Katty | i meani do it all the time ;) |
| 15:34.05 | Katty | i imagine if we ever /actually hung out/ i'd have to duct tape his mouth shut often. |
| 15:34.22 | [TK]D-Fender | pigpen, http://smily-domination.webs.com/photos/Funny-Junk/shipment_of_fail.jpg |
| 15:34.33 | pigpen | My wife would probably add me to to duct tape list. |
| 15:35.07 | [TK]D-Fender | Katty, Oh... you could try :) |
| 15:35.27 | Katty | you'd let me. |
| 15:35.28 | pigpen | [TK]D-Fender, cool. Hmm?I wonder who the captain was. I bet he went to work for a cruise line. |
| 15:35.53 | [TK]D-Fender | pigpen, You followed the news about the Italian cruise ship from a few weeks ago? |
| 15:36.21 | [TK]D-Fender | pigpen, Do you know how close they were to shore? |
| 15:36.24 | pigpen | A little. Mostly thoughts how "I" would chop that ship up. |
| 15:36.33 | pigpen | From what I could tell about 300 yards |
| 15:36.36 | [TK]D-Fender | pigpen, neither did their captain ;) |
| 15:36.38 | *** join/#asterisk Kalidarn (~unknown@unaffiliated/kalidarn) |
| 15:37.02 | pigpen | haha. yeah. he was being a dumbass. |
| 15:37.25 | pigpen | Did you like his comment?"I ran it aground so it wouldn't sink!!!" |
| 15:39.12 | pigpen | Wikipedia definition of dumbass: "See Captain Francesco Schettino." |
| 15:39.27 | pigpen | Well dam. I need to work. |
| 15:39.30 | Katty | was he that one guy that one meme was based off of |
| 15:39.37 | Katty | one does not simply.... |
| 15:40.47 | pigpen | You watch, you will that ship on ebay. |
| 15:40.49 | [TK]D-Fender | pigpen, the best was all of the news comparisons to being just like the Titanic... except for being a quick swim to shore off a mediterranean paradise :p |
| 15:41.37 | pigpen | heh, yeah. No mention of succumbing to the freezing waters, ice, Leonardio DeCaprio, or anything. |
| 15:41.41 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 15:41.57 | [TK]D-Fender | pigpen, And worst of all.... CELINE DION |
| 15:42.28 | pigpen | yeah. true. |
| 15:43.31 | pigpen | Too bad Oprah wasn't on it. The Titanic that is. |
| 15:44.19 | [TK]D-Fender | Yeah.... Oprah was a huge pusher of junk science... |
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| 16:10.47 | pigpen | [TK]D-Fender, any luck hanging a fax machine off of a Audiocodes FXS using T38 to the asterisk box where it has a DAHDI PRI/Analog card? |
| 16:11.27 | pigpen | I know that I can do a fxs->fxo passthrough on the same unit and have faxes pass fine. |
| 16:11.56 | [TK]D-Fender | pigpen, Not sure on * 10, but previous could not terminate T.38 with provided lib. Spandsp was supposed to be capable fo this. Coppice could fill you in on the particulars when he's about I'm sure... |
| 16:12.27 | [TK]D-Fender | pigpen, I don't have any direct experience with T.38 to offer you though. |
| 16:12.47 | *** join/#asterisk dxd828 (~dxd828@host81-133-31-249.in-addr.btopenworld.com) |
| 16:12.59 | pigpen | k, yeah I couldn't find much on T.38 on 1.8 and saw that 1.10 was to have some abilities built in. |
| 16:13.24 | pigpen | but I am in the same boat. I know it exists in the wild, but nothing of real implementation. |
| 16:13.32 | pigpen | I hate faxes. |
| 16:15.48 | zkn | OK, so it's possible to use dial codes to pause the agent in the queue, suppose one needs more statuses other than only pause, how could that be implemented without using dial codes ? soft phones usually have also the option to specify account status but these seem to have no effect to Asterisk, what's the deal with that? |
| 16:16.54 | [TK]D-Fender | zkn, Because * is not a SIP proxy and the concept of a device advertising its state like that isn't something that it supports on any protocol |
| 16:17.07 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:17.21 | [TK]D-Fender | zkn, And "queue pause" is not the same thing as general SIP presence |
| 16:17.49 | *** join/#asterisk wesphillips (~wphill04@137.237.233.124) |
| 16:18.31 | zkn | so how is this achieved? |
| 16:19.08 | [TK]D-Fender | zkn, It isn't |
| 16:19.56 | *** join/#asterisk DefV (~Jan@node-hahmcop4no4a1ruk1w.ipv6.as30961.net) |
| 16:20.33 | zkn | OK, so this is not possible with Asterisk |
| 16:21.29 | DefV | This is probably the question most asked inhere, but can I setup a voip server with this card (http://www.modem-help.co.uk/Puretek/PT-3517-56k-Data-Fax-Speakerphone-PCI.html) to receive incoming calls over PSTN? |
| 16:21.36 | Dovid | is there any way of settign directrtp from the dial plan or only in a peer? |
| 16:22.32 | Qwell | DefV: No. |
| 16:22.46 | Qwell | wait, what? |
| 16:22.52 | Qwell | pure voip? with analog hardware? |
| 16:23.15 | kaldemar | Dovid: for a peer, yes. |
| 16:23.21 | Qwell | needs more coffee |
| 16:23.26 | DefV | well no, incoming calls would be from a regular phone line |
| 16:23.37 | Qwell | DefV: still no. You need real telephony hardware. |
| 16:23.50 | DefV | OK |
| 16:23.59 | DefV | I figured as much, since I couldn't get it to work |
| 16:24.04 | Qwell | something like this, with an FXO, for example: http://www.digium.com/en/products/analog/4-port/ |
| 16:24.19 | DefV | tnx for clarifying :-) |
| 16:26.01 | DefV | any affordable cards? :-) |
| 16:27.21 | [TK]D-Fender | DefV, Describe your precise needs and expectations |
| 16:28.34 | DefV | OK. At the moment we're doing regular phone'ing with a normal phone (PSTN, not ISDN). We have some fancy Cisco phones that do VOIP. I want all calls coming in over the phone line to be sent to the Cisco VOIP phones |
| 16:28.39 | DefV | and I have a server with hw |
| 16:28.44 | DefV | just not a good voice card :-) |
| 16:28.49 | kaldemar | DefV: an ATA costs 10 times less. you could also use an ITSP for PSTN connectivity. |
| 16:29.53 | [TK]D-Fender | DefV, Linksys SPA-3102 ~$70 USD |
| 16:30.00 | _Corey_ | DefV: For whatever it's worth, ISDN is not an alternative to the PSTN... |
| 16:30.33 | WIMPy | Huh? ISDN is PSTN. |
| 16:30.52 | [TK]D-Fender | Or at least a common tech for reaching it :) |
| 16:31.24 | [TK]D-Fender | <PROTECTED> |
| 16:31.49 | DefV | jup, sorry |
| 16:32.01 | WIMPy | That would make more sense. |
| 16:32.01 | DefV | my telecom provider calls it PSTN / ISDN |
| 16:32.09 | DefV | and I just copied that :-) |
| 16:32.12 | *** join/#asterisk navaismo (~navaismo@189.249.54.116) |
| 16:32.46 | _Corey_ | DefV: Well, make sure you actually have analog lines before you buy anything. ISDN handoff requires different equipment... |
| 16:32.51 | DefV | [TK]D-Fender: well, I have some custom need, server accepting the voip also needs to call into an OpenVPN network and route to the phones |
| 16:33.15 | DefV | _Corey_: yeah, I'm sure it's analog and not ISDN |
| 16:33.25 | WIMPy | If you have ISDN available, you should check that out. The interfaces are dead cheap. |
| 16:33.32 | DefV | I don't :-( |
| 16:34.20 | [TK]D-Fender | DefV, I was only asking about the PSTN interface requirements |
| 16:35.37 | DefV | so, to summarize, a card/hw that I can plug into a computer that Asterisk supports, I can plug my phone cable (POTS) into, and is affordable? :-) |
| 16:39.28 | DefV | *tumbleweed* |
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| 16:40.29 | [TK]D-Fender | <[TK]D-Fender> DefV, Linksys SPA-3102 ~$70 USD <--------- please pay attention |
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| 16:42.06 | DefV | ah, sorry |
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| 17:27.11 | krotos | hi all :) |
| 17:30.01 | WIMPy | lo you |
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| 18:20.51 | krotos | hei guys, i need a confirmation about a rtp . If user A and B are both registered on Asterisk Box K. all with canreinvite=no, directmedia=no. If a call b the rtp stream follow this path? A-->rtp-->K-->rtp-->B |
| 18:20.56 | krotos | right? |
| 18:22.19 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 18:22.35 | [TK]D-Fender | krotos, Correct |
| 18:23.46 | krotos | [TK]D-Fender: thankyou :) So i can prioritize VoIP simple marking packet that have ip to my Asterisk Box |
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| 18:54.57 | talntid | Best SIP provider for a call center environment? |
| 18:57.04 | [TK]D-Fender | The same as for a non-call-center environment |
| 19:08.44 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 19:11.47 | akrohn | this might be a dumb question, but if it's a call center, why not just get a PRI? What's cost diff between PRI and SIP provider? Maybe I only say that because I dislike 'cloud' services. |
| 19:15.33 | [TK]D-Fender | SIP provider != cloud. And cost is an obvious factor. |
| 19:18.28 | cekz | akrohn, there could be long distance fees, and many more reasons |
| 19:20.01 | akrohn | good call on the LD fees cekz, hadn't considered that. [TK]D-Fender I know it's not cloud per sae, but because of customers, I find myself referring to everything that isn't sitting in my datacenter to be the 'cloud' =/ |
| 19:23.54 | [TK]D-Fender | You should smack that guy :) |
| 19:26.52 | talntid | I currently have a PRI |
| 19:26.56 | talntid | it sucks. |
| 19:27.00 | talntid | the provider sucks |
| 19:27.12 | talntid | so, I want to switch to SIP. |
| 19:32.49 | [TK]D-Fender | "it sucks" <- care to qualify that a bit? |
| 19:33.05 | p3nguin | Or at least quantify it. |
| 19:34.35 | [TK]D-Fender | p3nguin, I would never expect any math skills given the source ;) |
| 19:34.49 | p3nguin | Poor guy. |
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| 19:40.11 | *** join/#asterisk fireman_biff (~biff@65.48.133.102) |
| 19:40.44 | fireman_biff | Hi, is there a way to check from the Asterisk CLI if software echo cancellation is being used on a DAHDI channel? |
| 19:40.47 | *** part/#asterisk fireman_biff (~biff@65.48.133.102) |
| 19:40.56 | wdoekes2 | lol |
| 19:41.01 | *** join/#asterisk fireman_biff (~biff@65.48.133.102) |
| 19:42.41 | [TK]D-Fender | fireman_biff, Do you have HWEC? |
| 19:43.36 | fireman_biff | [TK]D-Fender: Yes, but they users say they're getting echo, so I'm wondering if both software and hardware are in use |
| 19:43.52 | [TK]D-Fender | No, HWEC only. You don't get to override it if it's there |
| 19:44.35 | fireman_biff | so what should I check if they have echo with hwec? |
| 19:44.39 | fireman_biff | the gain? |
| 19:45.27 | [TK]D-Fender | for starters. |
| 19:46.22 | fireman_biff | I would want to decrease both the rx and tx right? |
| 19:46.35 | [TK]D-Fender | I'd start at 0 across the board first. |
| 19:46.46 | [TK]D-Fender | And make sure the HWEC is indeed properly ack'd |
| 19:47.17 | fireman_biff | how do i check the hwec? |
| 19:48.15 | *** join/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 19:49.26 | [TK]D-Fender | Depends on what you've got. |
| 20:01.04 | fireman_biff | [TK]D-Fender: any idea how i would check for an analog rhino card? |
| 20:05.38 | *** join/#asterisk kessius (~cassio@186.206.8.118) |
| 20:06.35 | [TK]D-Fender | EWWWWW |
| 20:06.39 | [TK]D-Fender | "no" |
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| 20:32.22 | talntid | [TK]D-Fender, I could talk about the symptoms as to why it sucks, and then you can tell me that it is a configuration issue, or that I am not doing it right. so why? Hired 3 different people to check it out. Still fucks up. |
| 20:33.05 | [TK]D-Fender | talntid, And I could ask 10 times and probably not get a clear coherent answer with all the pertinent details... |
| 20:34.30 | fireman_biff | is reloading asterisk enough to put into effect changes to rx/tx gain? |
| 20:34.59 | talntid | sometimes, people don't know all the pertinent details to provide you. Not everyone is as experienced as you. For some people, this is the only PRI they have ever tried, and maybe they don't know everything about how to do it right. |
| 20:35.09 | [TK]D-Fender | reload chan_dahdi |
| 20:35.18 | fireman_biff | [TK]D-Fender: thanks |
| 20:35.19 | [TK]D-Fender | base reload = no. And this WILL drop calls |
| 20:35.40 | [TK]D-Fender | talntid, Yes, but you continue to give none.... |
| 20:36.10 | talntid | [TK]D-Fender, every few days, my PRI shows "channel unavailable". an asterisk restart fixes it. |
| 20:37.46 | talntid | I'll go back and grab a log, if you think you can help |
| 20:38.55 | talntid | [Jan 30 09:44:28] WARNING[27227] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) |
| 20:38.59 | talntid | that is what shows up, over and over. |
| 20:39.00 | [TK]D-Fender | talntid, * version? Card? |
| 20:39.36 | [TK]D-Fender | Cause 34 is often used as an actual "busy" by some telcos and shows that the other side is at least getting the request. |
| 20:39.42 | [TK]D-Fender | and "Zap" tells me a lot already... |
| 20:39.44 | talntid | Asterisk 1.4.17, on Ubuntu. Reason Asterisk is so old is because I have not been able to successfully upgrade it AND get the zaptel stuff working again |
| 20:39.48 | [TK]D-Fender | Carbon-dated..... |
| 20:40.25 | talntid | and it's a sangoma... A200 |
| 20:40.42 | [TK]D-Fender | .... |
| 20:40.57 | [TK]D-Fender | A200 = analog... how are you getting a ISDN cause-code back from that>? |
| 20:41.17 | talntid | 02:01.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card |
| 20:41.52 | talntid | I don't know a ton about how PRI's work.. what I know is this: |
| 20:42.04 | talntid | I have a "voip gateway" box, in my rack... |
| 20:42.14 | talntid | it has an internet connection going into it.. |
| 20:42.24 | talntid | and it has an output, which goes to the card in my asterisk box |
| 20:42.34 | [TK]D-Fender | A200 = analog.. so how the hell is this a PRI issue? |
| 20:43.17 | talntid | see, this is where I'm not sure what you are asking. I don't know shit about PRI, and why the card being analog is a "duh" moment for you. |
| 20:43.42 | [TK]D-Fender | You'd done the equivalent of asking where the gas-tank is on your bicycle... |
| 20:43.53 | [TK]D-Fender | A200 = analog. Not digital circuit. Not a PRI. |
| 20:44.09 | [TK]D-Fender | And you've not hinted at something else that may be relaetd... |
| 20:44.57 | *** join/#asterisk chigambamukoko (~chatzilla@fl-76-3-22-71.dhcp.embarqhsd.net) |
| 20:45.05 | talntid | so I'm guessing what is happening is.... |
| 20:45.38 | talntid | PBX (digital) -> Sangoma (Analog) -> Voip Gateway Input (Analog) -> Voip Gateway Output (Digital) -> Provider |
| 20:46.16 | [TK]D-Fender | That makes no sense..... |
| 20:46.32 | [TK]D-Fender | for PRI. |
| 20:46.42 | talntid | http://www.accessline.com/smartvoiceservice/interior.asp?nav=nHIW |
| 20:46.57 | [TK]D-Fender | And You're using an A200 to reach a gatway that turns it right back to some digital format? That is retarded... |
| 20:47.34 | talntid | Yeah, I built it 5 years ago. I didn't/don't know the difference between PRI's and what I should have, etc... |
| 20:47.43 | talntid | hence, wanting to go SIP. I know what that is. |
| 20:48.03 | talntid | I think there are too many wrong steps out/in |
| 20:48.05 | [TK]D-Fender | There is no PRI. You introduced that term with nothing backing it up. |
| 20:48.09 | talntid | ok. |
| 20:48.14 | talntid | that's how it was sold to me |
| 20:48.24 | [TK]D-Fender | It seems to have been lodged in there for years. |
| 20:48.36 | talntid | buzzwords. |
| 20:48.58 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
| 20:49.09 | [TK]D-Fender | No, not a "buzz" word. A technical word that will get you killed like mistaking hydrochloric acid for water and taking a big gulp |
| 20:49.30 | talntid | I don't know what else to call something that was introduced to me as a PRI. |
| 20:49.37 | [TK]D-Fender | talntid, Ask your provider what protocol their gatway is talking and get * to talk direct to the provider and do away with all that conversion nonsense |
| 20:49.56 | talntid | I know that they don't support SIP. I have asked them that |
| 20:50.03 | talntid | but yeah, I can ask them |
| 20:50.19 | [TK]D-Fender | Odds are H.323 |
| 20:50.37 | [TK]D-Fender | Which you could probably get away with for this need. |
| 20:50.53 | talntid | http://www.shopmania.co.za/phones-faxes/p-quintum-dx-4120-rack-mount-4-e1-voip-gateway-120-channels-1787678 |
| 20:51.01 | talntid | that's what I have in the rack.. they provided that. |
| 20:51.25 | [TK]D-Fender | That is a GIANT f'n gateway... and a digital one. |
| 20:51.28 | [TK]D-Fender | A200 = analog. |
| 20:51.36 | [TK]D-Fender | again the pieces do not add up |
| 20:52.29 | talntid | might not be that exact model. But it is the same brand, and looks the same. I'm not on-site, so I can't get exact model |
| 20:54.10 | *** join/#asterisk [ALT][F4] (~altf4@ppp-58-11-96-124.revip2.asianet.co.th) |
| 20:54.40 | [TK]D-Fender | Was that a peanut allergy I had... or a penicillin? Guess I'll just take an aspirin |
| 20:54.42 | [TK]D-Fender | dies |
| 20:56.53 | [TK]D-Fender | wanpipe configs should tell you this already along with the rest of their CLI suite. |
| 20:57.20 | [TK]D-Fender | And you're no doubt on a seriously decrepit version of their driver as well.. |
| 20:59.01 | talntid | yeah. Seems they no longer maintain packages for it |
| 20:59.22 | p3nguin | If they have a gateway on site, tell them to replace it with a SIP gateway so you can use SIP over Ethernet between it and Asterisk. |
| 20:59.36 | kessius | * Does not register for incoming calls |
| 20:59.46 | p3nguin | kessius: Do you want it to? |
| 20:59.58 | pigpen | I was just informed that I need to be able to connect a Musac streaming music via a headphone/rca cable into Asterisk's music on hold. The catch is, I would need to have it be a usb audio input. Asterisk 1.8.7+ Doable? |
| 21:00.04 | pigpen | I have never done it. |
| 21:01.00 | [TK]D-Fender | pigpen, Sure. |
| 21:01.20 | [TK]D-Fender | pigpen, there are docs for taking an ALSO line-in as MoH. |
| 21:01.22 | pigpen | just any usb linux supported sound card and a special config? |
| 21:01.23 | [TK]D-Fender | ALSA* |
| 21:01.24 | p3nguin | I missed where you got from a headphone or other coaxial audio cable to a USB cable. |
| 21:01.49 | pigpen | ALSA, yeah, that's right. |
| 21:01.55 | Qwell | USB doesn't matter, it's just a soundcard |
| 21:02.02 | Qwell | at least as far as Asterisk is concerned. |
| 21:02.09 | pigpen | k, didn't know if it mattered to asterisk. |
| 21:02.33 | pigpen | then just pass it into asterisk via the moh config. |
| 21:02.42 | pigpen | Easy enough. |
| 21:02.43 | pigpen | tks. |
| 21:03.05 | talntid | Had a person onsite check the exact model, [TK]... DX2030 |
| 21:03.59 | [TK]D-Fender | talntid, Looking like yuo have no clue on what card you have. |
| 21:04.11 | [TK]D-Fender | look at your wanpipe config sand the hwprobe |
| 21:04.49 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 21:05.59 | talntid | http://pastebin.com/rpZ3Xt1p |
| 21:09.49 | talntid | and my lspci says: 02:01.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card |
| 21:12.37 | talntid | I get the impression that these pieces of hardware aren't supposed to work together. |
| 21:14.28 | [TK]D-Fender | That config is clearly not analog |
| 21:14.39 | [TK]D-Fender | wanroute hwprobe |
| 21:14.42 | [TK]D-Fender | wanrouter hwprobe |
| 21:14.43 | [TK]D-Fender | ^^ |
| 21:15.00 | talntid | http://pastebin.com/kk5dicE6 |
| 21:15.18 | [TK]D-Fender | AFT-A101-SH : SLOT=1 : BUS=2 : IRQ=5 : CPU=A : PORT=1 : HWEC=32 : V=34 |
| 21:15.24 | [TK]D-Fender | A101 != A200 |
| 21:15.37 | [TK]D-Fender | That is their oldest 1-port T1/E1 interface |
| 21:16.00 | talntid | was $800ish back in the day. heh |
| 21:16.12 | [TK]D-Fender | TDMV_HWEC = YES |
| 21:16.12 | talntid | I also had a Rhino r1t1 card too... |
| 21:16.16 | talntid | but couldn't get it to work |
| 21:16.20 | [TK]D-Fender | Odd... A101 didn't have HWEC |
| 21:16.24 | [TK]D-Fender | this shouldn't be there |
| 21:16.32 | talntid | yeah |
| 21:16.37 | talntid | echo cancellation right? |
| 21:16.55 | talntid | "Choose the Sangoma A101D and A101DE, equiped with world class DSP hardware to achieve carrier-grade echo cancellation and voice quality enhancement functions for your telephone systems." |
| 21:17.01 | [TK]D-Fender | A101d did, but that came after the A104d was first introduced (fall 2005). |
| 21:17.24 | [TK]D-Fender | It should list as a "d"... |
| 21:17.36 | [TK]D-Fender | anyway, I'll leave that be for a bit. |
| 21:17.48 | talntid | I do recall buying it specifically because it did have echo cancellation |
| 21:17.54 | [TK]D-Fender | enable PRI debug when you start getting the congestion warnings and PB a sample. |
| 21:18.24 | talntid | k |
| 21:18.47 | talntid | pri debug span 1? |
| 21:19.34 | [TK]D-Fender | looks right |
| 21:20.38 | talntid | heh, turned it on. now trying to figure out how to turn it off |
| 21:21.46 | talntid | oh well. it's fine running |
| 21:22.54 | talntid | so, does this setup make more sense now? |
| 21:23.04 | talntid | or is it still a clusterfck? :) |
| 21:23.17 | [TK]D-Fender | Well... it's still wasted hardware... |
| 21:23.21 | [TK]D-Fender | but at least it adds up. |
| 21:23.39 | [TK]D-Fender | Not an A200. Is an A101. That makes sense... in a "you could..." way |
| 21:24.52 | talntid | gotcha |
| 21:25.00 | talntid | wonder why lspci shows a A200 |
| 21:28.09 | [TK]D-Fender | No idea... and on that note... checkout time. Long walk home. |
| 21:28.11 | [TK]D-Fender | BBL |
| 21:28.27 | talntid | have a good one |
| 21:29.51 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-benawpehpbbajqnr) |
| 21:29.52 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 21:33.50 | pabelanger | kessius: ask here! |
| 21:37.04 | leifmadsen | Qwell: ping |
| 21:37.10 | Qwell | pong? |
| 21:37.21 | leifmadsen | you're going to hate this question... |
| 21:37.30 | leifmadsen | but what are you calling the asterisk 10 RPMs (what is your file naming convention?) |
| 21:37.39 | Qwell | They are all asterisk now |
| 21:37.46 | Qwell | see the new asterisk-1.8, asterisk-10 repos |
| 21:37.48 | leifmadsen | separate repos? |
| 21:37.50 | Qwell | yep |
| 21:38.06 | leifmadsen | ok, and how do you handle having both 1.4 and 1.8 repos enabled at the same time? (or do you just, not?) |
| 21:38.09 | Qwell | I did that to fix exactly that problem |
| 21:38.16 | Qwell | I used to call them asterisk14, asterisk18 |
| 21:38.18 | leifmadsen | I'm trying to figure out Requires: stuff |
| 21:38.24 | Qwell | but those were screwy |
| 21:38.27 | leifmadsen | ya |
| 21:38.37 | Qwell | oh, how do I handle them being enabled. it doesn't matter |
| 21:38.57 | leifmadsen | I have a Requires: asterisk-devel >= 1.4.19 in an addons RPM and it's finding the 1.8.8.2 I created |
| 21:39.04 | pabelanger | leifmadsen: separate repos is the way to go, unless you some how plan to run both version of asterisk at the same time |
| 21:39.13 | Qwell | I think you can do like |
| 21:39.28 | Qwell | Requires: asterisk-devel >= 1.4.19 && asterisk-devel <= 1.6 |
| 21:39.30 | Qwell | or something |
| 21:39.32 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 21:39.40 | leifmadsen | Qwell: ok I was thinking about that too, trying |
| 21:39.47 | Qwell | might be & |
| 21:42.44 | leifmadsen | Qwell: I think that is illegal, but rpmlint doesn't complain about something liek asterisk >= 1.4.19, asterisk < 1.6 |
| 21:43.00 | p3nguin | talntid: lspci -Qs 02:01.0 |
| 21:45.00 | p3nguin | Or, actually, lspci -Qs 02:01 |
| 21:45.45 | saxa | hi , anybody willing to help me out of a no audio problem ? The phone is behind a nat and the server is behind another nat. I opened the ports 5060 and 10000-10200 UDP I get the phone registered, it rings, calls but audio doesnt goes thru. Interesting is that it was working perfectly, until I added another port in the port forwarding on the sserver side modem/router. |
| 21:46.19 | saxa | I already tried many things, and cant understand where the problem occurs ? |
| 21:50.02 | p3nguin | saxa: Perhaps that is the wrong port range. The typical range is 10000-20000 until changed to something else. |
| 21:53.58 | saxa | p3nguin: i added this port range in rtp.conf |
| 21:54.18 | saxa | there is one problem, my modem permits me to open max 256 ports |
| 21:54.26 | saxa | Huawei HG521 |
| 21:54.52 | Qwell | so then limit your port range to 256 |
| 21:56.02 | saxa | so I opened from 10000 to 10200 |
| 21:56.30 | leifmadsen | did you reload asterisk after your changes? did you look at the sip trace to see what ports the audio is being setup on? |
| 21:59.13 | krotos | hi all! :) I'm having a small problem with a OpenVox B400p/B200p . Should i use wcb4xxp driver for this card? |
| 21:59.47 | *** join/#asterisk k-man_ (~jason@unaffiliated/k-man) |
| 21:59.48 | Qwell | krotos: Ask the vendor. |
| 21:59.55 | Qwell | that driver does not explicitly support that card |
| 21:59.59 | saxa | leifmadsen: yes i restarted it many times, after each change |
| 22:00.32 | saxa | the audio is sent on 5074, 5004 , random ports |
| 22:00.32 | k-man_ | when i call my linksys phone from my iphone voip client, on the linksys it says incomming from "jasonsiphone" rather than the extension number - is there any way to show the extensionnumber? |
| 22:00.49 | saxa | those are just some of them i could see |
| 22:02.00 | p3nguin | k-man_: Fix the callerid value. |
| 22:02.28 | krotos | Qwell: ok.. |
| 22:02.58 | p3nguin | k-man_: Typically, something like this: callerid=Jason iPhone <1234> |
| 22:06.35 | *** join/#asterisk Gaiax (be19ebae@gateway/web/freenode/ip.190.25.235.174) |
| 22:10.04 | k-man_ | p3nguin: thanks heaps! that did the trick |
| 22:13.45 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 22:17.03 | *** join/#asterisk wwalker (~wwalker@208.92.232.27) |
| 22:18.34 | wwalker | does anyone have a SQL tool that reads CDRs and outputs some set of "port in use, start time, end time" ? |
| 22:19.17 | saxa | ok I see now the audio is sent on 10010 |
| 22:19.23 | saxa | see here |
| 22:19.26 | saxa | http://pastebin.ca/2108306 |
| 22:19.36 | saxa | the client is casasip |
| 22:19.37 | *** part/#asterisk wesphillips (~wphill04@137.237.233.124) |
| 22:20.07 | saxa | hmm, let me check my router on the client side |
| 22:22.28 | saxa | http://pastebin.ca/2108314 |
| 22:22.38 | saxa | this is a second call I did right now |
| 22:23.47 | saxa | I have also opened on my client side nat the ports 5060 TCP/UDP and 10000-10200 UDP |
| 22:38.06 | p3nguin | You should never need to forward ports for a phone. |
| 22:38.34 | p3nguin | If asterisk is configured to support NAT, a phone behind a NAT should work correctly. |
| 22:43.31 | saxa | http://pastebin.ca/2108347 <--- thats a sip set debug on |
| 22:44.08 | saxa | ok I can get off those open ports on the client side nat |
| 22:44.37 | saxa | client side nat is the router I have at home, where casasip sits behind it. |
| 22:46.14 | *** join/#asterisk fireman_biff (~biff@65.48.133.102) |
| 22:47.44 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:59.12 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-benawpehpbbajqnr) |
| 23:03.14 | *** join/#asterisk [ALT][F4]1 (~altf4@180.183.125.177) |
| 23:15.54 | *** join/#asterisk chasing`Sol (~cS@197.132.90.181) |
| 23:17.11 | saxa | p3nguin: so the opened ports should be only on the router in front of asterisk server ? |
| 23:18.23 | p3nguin | In my experience, where you have asterisk behind one NAT and a phone behind a different NAT, it is sufficient to only forward the relevant ports inbound to asterisk. |
| 23:19.19 | p3nguin | This is how I have my own system deployed. I have asterisk behind a NAT and phones behind another NAT. Port forwarding is done only on the asterisk side. |
| 23:19.37 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 23:19.49 | saxa | ok, i see |
| 23:20.25 | saxa | i have removed the opened ports |
| 23:20.42 | saxa | and in fact it works the same way it was, still no audio :) |
| 23:20.47 | saxa | but it registers |
| 23:21.05 | p3nguin | Make sure you to not have any ALG enabled anywhere. Also, make sure your phone is not trying to do NAT traversal on its own -- allow asterisk to handle it. |
| 23:21.31 | Maliuta | no audio sounds like no RTP is getting through |
| 23:22.39 | p3nguin | That is a very astute observation. |
| 23:23.48 | Maliuta | although all audio sounds like something ;P |
| 23:24.23 | p3nguin | Even the silence is deafening at times. |
| 23:25.57 | Maliuta | plays p3nguin the "Sounds of Silence" |
| 23:30.17 | saxa | p3nguin: so I should disable STUN on my phone ? |
| 23:31.01 | p3nguin | If you need STUN, use it. But you may not need it. |
| 23:31.42 | saxa | NAT Traversal (STUN): No No, but send keep-alive Yes |
| 23:32.19 | saxa | i had this enable before it stopped to work. |
| 23:33.23 | saxa | what is rfc3581 used for ? |
| 23:33.36 | saxa | symetricrouting |
| 23:34.32 | kessius | good night greetings |
| 23:34.45 | saxa | night kessius |
| 23:35.26 | kessius | friend back question, asterik no registred - rule register=user:password@domain:port/extension |
| 23:39.29 | kessius | asterisk has face from internet ? I need help, brothers - i am novice on asterisk |
| 23:39.42 | [TK]D-Fender | Talks does funny Yoda hhhhmmmmmMMMMMM??!?!?!? |
| 23:40.36 | [TK]D-Fender | kessius: "sip set debug on" <- look at your registration attempt |
| 23:42.22 | kessius | ok I will see - i post Fender |
| 23:42.41 | [TK]D-Fender | ~pb |
| 23:42.41 | 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. |
| 23:42.42 | [TK]D-Fender | ^^^ |
| 23:44.46 | Maliuta | [TK]D-Fender. He's the Obi Wan of #astersik |
| 23:45.07 | Maliuta | "Search your heart ... you know it to be true." |
| 23:49.08 | *** join/#asterisk nfsnobody (~whitey@ppp191-96.static.internode.on.net) |
| 23:49.16 | nfsnobody | hey all |
| 23:50.25 | nfsnobody | I've recently upgraded from asterisk 1.6 to 1.8 and I use MySQL Realtime for my queue_log for the purposes of custom applications. It seems the data structure has changed to a pipe delimited data field (e.g. 1|834|23|) instead of seperate fields (data1 is 1, data2 is 834, data3 is 23, etc) - can anyone point me in the right direction of fixing this? |
| 23:50.59 | nfsnobody | I could update all my MySQL triggers to suit this, but I strongly feel that the seperate fields is the better way, as well as worrying about the consistency of my old and new data |
| 23:51.18 | *** part/#asterisk fireman_biff (~biff@65.48.133.102) |
| 23:52.28 | nfsnobody | I'm wondering if there's a setting somewhere that I'd set on the old version that I haven't on the new version, as this doesn't seem to be a bug fix or feature |
| 23:54.13 | nfsnobody | any ideas anyone? I'm willing to try whatever needs trying :) |
| 23:54.57 | kessius | no registered * - sip set debug on OK - where look -> registration attempt ? |
| 23:57.18 | [TK]D-Fender | kessius: Yes |
| 23:58.22 | [TK]D-Fender | "sip show registry" <- this will show is * even saw your config files having a register directive in the right place. Then SIP debug should show the actual attempts |
