| 00:00.01 | WIMPy | I think so. |
| 00:00.11 | gusto | that's quite f**8 bad |
| 00:00.31 | WIMPy | You get used to it. |
| 00:00.42 | gusto | goes to sleep |
| 00:00.50 | WIMPy | It's only one of a whole lot of bad things. |
| 00:01.37 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 00:07.08 | sam555 | anyone use freepbx? |
| 00:11.20 | WIMPy | The guys in #freepbx |
| 00:11.53 | *** join/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
| 00:12.27 | Get_The_Fish | MichaelB8225 there is Bria on OS X, actually works on Win OS X and Ubuntu |
| 00:12.40 | Get_The_Fish | lol WIMPy |
| 00:13.31 | volga629 | Hello Everyone, I reset g729 the license , but still get this message WARNING[11146] translate.c: empty buf size, you need to supply on |
| 00:17.17 | volga629 | I will come back soon |
| 00:25.21 | *** join/#asterisk sam555 (~sysadmin@udp124488uds.hawaiiantel.net) |
| 00:34.36 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
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| 00:57.56 | Get_The_Fish | ...but I dont wanna learn sed and awk |
| 01:00.34 | volga629 | online |
| 01:05.26 | volga629 | where I can find g729a nocona to download |
| 01:07.51 | Get_The_Fish | http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC |
| 01:08.09 | jpsharp | http://downloads.digium.com/pub/telephony/codec_g729/ |
| 01:08.14 | Get_The_Fish | Only $10 a license! Hurry while supplies last! |
| 01:08.30 | volga629 | thanks |
| 01:08.57 | volga629 | I have already :-) |
| 01:09.58 | jpsharp | Just follow the link I posted for the right version of Asterisk & whether or not you're running 32 or 64 bit OS. |
| 01:10.54 | volga629 | yes, done finally I see translation and no errors :-) |
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| 01:12.45 | volga629 | thanks again |
| 01:13.19 | volga629 | yes, only one think is left how to stabilize srtp any tips ? |
| 01:15.50 | jpsharp | Not off the top of my head. |
| 01:16.50 | volga629 | I tried rebuild and change version of lib srtp, but still crashing |
| 01:17.49 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 01:19.26 | volga629 | very interesting, but tls for registration working no issue |
| 01:22.08 | volga629 | is something need in dial like encryption capable ? |
| 01:24.40 | jpsharp | No. If encryption is enabled in the Asterisk peer setting it should try to negotiate it. |
| 01:30.52 | volga629 | I wonder if there some patch for srtp which not included in tar 10.05 |
| 01:33.24 | volga629 | res_srtp.c:301 res_srtp_unprotect: SRTP unprotect: authentication failure this one I reported look snom 370 |
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| 01:35.31 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:38.11 | volga629 | another issue with Imap storage [2012-06-18 21:37:32] WARNING[4755]: app_voicemail_imapstorage.c:1800 vm_imap_delete: msgnum 0, mailbox message 0 is zero. |
| 01:38.47 | volga629 | and it delivery only to INBOX to folder which specified in config ? |
| 01:40.09 | *** part/#asterisk sam555 (~sysadmin@udp124488uds.hawaiiantel.net) |
| 01:46.01 | *** part/#asterisk GGD (~deberle@pool-173-72-200-109.clppva.fios.verizon.net) |
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| 02:10.24 | KNERD | That digium fax register app is returning an error: Could not generate Host-ID. Make sure that you have eth0 enabled. |
| 02:11.17 | *** join/#asterisk Node_403 (b4ea5a4c@gateway/web/freenode/ip.180.234.90.76) |
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| 02:44.10 | volga629 | I am getting this error IMAP Error: parse error: invalid message sequence number |
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| 03:10.14 | greenwolf | how come when i use alcazar networks free tollfree outbound service the asterisk places the call and hands it off to the remote network but then my asterisk console disconnects? |
| 03:10.26 | greenwolf | any ideas why? When i use my flowroute account its works just fine |
| 03:13.15 | KNERD | greenwolf: that is like asking: "I put Exxon gasoline in my car, but it just won't start now. I put Shell in in Earlier today"" |
| 03:14.40 | KNERD | try posting a call progress |
| 03:17.17 | greenwolf | ok is |
| 03:21.55 | greenwolf | ok now i cant even call into my box |
| 03:23.39 | greenwolf | i have posted my sip.conf file. any reason why non of my calls are making it to my box? |
| 03:23.40 | greenwolf | http://pastie.org/4112225 |
| 03:27.23 | greenwolf | nevermind i got it |
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| 03:58.43 | ChannelZ | Is there something like 'top' for files? |
| 03:59.10 | ChannelZ | I have some task which is sucking CPU and hammering the HD and I'd like to see what it thinks it's doing |
| 04:01.34 | volga629 | might lsof you can check what open and who accessing |
| 04:02.05 | ChannelZ | ah thanks |
| 04:08.28 | volga629 | http://danielmiessler.com/study/lsof/ |
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| 04:16.33 | greenwolf | why do i keep getting chan_sip.c:3695 retrans_pkt: Retransmission timeout reached on transmission 629bf6f63cbe2e3549a27c6f543c1354@66.54.140.46 for seqno 102 (Critical Response) |
| 04:16.58 | volga629 | might be nat firewall or rtp |
| 04:17.17 | greenwolf | how can i check ? |
| 04:18.34 | volga629 | where on pbx set nat ? |
| 04:19.08 | volga629 | if you need it. |
| 04:27.46 | greenwolf | its connected right to the cable modem |
| 04:27.49 | greenwolf | no router in the way |
| 04:27.58 | greenwolf | so i dont understand how NAT is doing this to my sip packets |
| 04:33.20 | KNERD | greenwolf: your router can be if it i not setup |
| 04:35.10 | ngharo | he said he's directly connected to his modem |
| 04:35.37 | ngharo | unless his ISP is doing NAT which is very possible and easy to check (is your WAN IP a private address or not) |
| 04:40.03 | jpsharp | Or your ISP is blocking SIP packets. |
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| 04:43.15 | greenwolf | yes for some odd reason its private |
| 04:43.29 | greenwolf | and there is network switch so it must have it build in? |
| 04:44.25 | ngharo | login to it and look to enable bridge mode |
| 04:47.09 | greenwolf | they wont give me the damn login creds for this modem |
| 04:47.22 | greenwolf | i thought that was the most craziest thing i ever heard |
| 04:47.47 | greenwolf | they said to bring my own router if i wanted to control that..they turned it into bridge mode or passthru mode already for me |
| 04:48.08 | *** join/#asterisk udhv (~uddhava@62.16.94.6) |
| 04:49.02 | ngharo | so where are you seeing the private IP? on your SIP device? |
| 04:49.06 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-aqaocilbpnrzqvck) |
| 04:50.32 | volga629 | Are using linux workstation use nc to test rtp port if connection hangs you good if not something on the way |
| 04:50.35 | greenwolf | no on the server |
| 04:50.47 | greenwolf | i have the server plugged directly into the modem bypassing the router |
| 04:51.00 | greenwolf | so i shouldnt receive a 192.X.X.X i dont understand y i am |
| 04:51.09 | greenwolf | mybe a DHCP enabled? |
| 04:52.09 | volga629 | on server it set static ip or dhcp ? |
| 04:52.20 | ngharo | then youre not bridged/passthru |
| 04:52.26 | volga629 | nc -zu ip address port number |
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| 04:55.21 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 04:57.43 | KNERD | Modems now days tend to have routers built in |
| 05:00.37 | jpsharp | The ISPs don't want you pulling down half a dozen IPs via DHCP. |
| 05:01.29 | ngharo | they limit per CPE MAC address server side |
| 05:02.02 | ngharo | at least every TW/Charter/Comcast all do or any ISP worth a shits |
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| 05:48.59 | KNERD | That digium fax register app is returning an error: Could not generate Host-ID. Make sure that you have eth0 enabled. |
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| 06:01.10 | din3sh | mrning all |
| 06:03.50 | KNERD | enenin' guvnor |
| 06:04.35 | *** join/#asterisk Vince-0 (c4d7a482@gateway/web/freenode/ip.196.215.164.130) |
| 06:04.51 | greenwolf | so is there a setting i must set in sip.conf for it to work around NAT? |
| 06:05.30 | volga629 | nat=yes |
| 06:05.42 | kaldemar | ~sipnat |
| 06:05.42 | 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, directmedia, externhost or externaddr, and localnet. |
| 06:05.50 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:05.52 | schmidts | good morning |
| 06:05.57 | volga629 | or shortcut :-) |
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| 06:17.11 | din3sh | anybody implemented phone directory via microbrowser in Polycom phones? |
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| 06:24.14 | bulkorok | hi |
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| 06:24.33 | bulkorok | we have sth like that for snom xml browser... |
| 06:27.34 | schmidts | din3sh maybe this could be interesting for you: http://sourceforge.net/projects/ipphone-xml/ AFAIK they only support snom and cisco by now but maybe polycom use the same xml format |
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| 06:44.23 | din3sh | thanks schmidts, taking a look at that |
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| 07:39.49 | CommaCrazy | has any one ever worked with Topex SoftSwitch and EoneS from Topex? |
| 07:40.57 | CommaCrazy | or better yet has anyone ever heard of Topex at all... |
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| 08:07.12 | schmidts | CommaCrazy nope for me ;) |
| 08:08.52 | _val_ | leifmadsen: you helped me yesterday with exten=_0X.,3,Set(CALLERID(name)=${dialername}) |
| 08:08.58 | _val_ | exten=_0X.,4,Set(CALLERID(number)=${dialernumber}) |
| 08:09.14 | _val_ | but this does not work. Any other solution? |
| 08:10.10 | schmidts | _val_ leifmadsen is from canada and its imho a little bit too early for him ;) |
| 08:11.19 | _val_ | schmidts: das stimmt :> aber kann du mich helfen? :-p |
| 08:13.38 | _val_ | ok schmidts now in English. They suggested me using exten=_0X.,3,Set(CALL.... exten=_0X.,4,Set(CALL... and exten=_0X.,5,Set(CALL... |
| 08:13.46 | _val_ | this however does not work. |
| 08:14.19 | _val_ | is it possible to use exten=_0[1-15].,3,Set(CALL... and so on? |
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| 08:18.50 | CommaCrazy | schmidts, lucky you... I wish I have never had the chance to work with it.... it is terrible... |
| 08:19.09 | *** join/#asterisk drhonk (~drhonk@static.57.112.4.46.clients.your-server.de) |
| 08:19.48 | schmidts | _val_ do you have exten = or exten => cause if you miss the ">" after the = then it will not work |
| 08:19.57 | CommaCrazy | but unfortunately I have to work with it... and I can not find the proper documentation for it, nor is their wiki good. |
| 08:20.34 | schmidts | commaCrazy :D haha i really know what you mean. right now i have to learn working with a 20 year old legacy switch from siemens. Pure POTS nothing dynamic |
| 08:21.38 | CommaCrazy | oh man those are a pain but once you learn them you will know all legacy systems and they are not that hard once you get the logic |
| 08:21.43 | CommaCrazy | you can do it :) |
| 08:22.00 | _val_ | schmidts: http://pastie.org/4113169 |
| 08:23.23 | *** join/#asterisk adolfomaltez (~taro@190.62.235.62) |
| 08:24.47 | schmidts | CommaCrazy at all the whole system is just stupid hard config, no chances, no transport stuff only hard wires like in the beginning of telephony, the old lady plugin cables from one port into another but in a large scale with around 7000 E1 :D |
| 08:25.24 | schmidts | _val_ you missed the > it should look like this: exten => _0XXXXXXXXX,3 or also the new one but you miss the > |
| 08:26.42 | _val_ | schmidts: otherwise extensions.conf won't load right? There is no configuration error. |
| 08:27.12 | schmidts | _val_ just do a "dialplan show" and see if these extensions are there |
| 08:28.02 | _val_ | schmidts: doing so. Thanks. I'll have to run to other machine and check it. |
| 08:28.39 | KNERD | Any idea on installing that Asterisk fax on a VPS without access to eth0? |
| 08:29.45 | schmidts | KNERD why do you need eth0? any kind of ip connection is good enough |
| 08:30.10 | KNERD | because the register app requires it |
| 08:31.01 | KNERD | Could not generate Host-ID. Make sure that you have eth0 enabled. |
| 08:31.13 | schmidts | ah you mean the license for it |
| 08:31.44 | schmidts | i guess you should ask digium directly about it, or maybe in #asterisk-dev |
| 08:32.26 | KNERD | I dont think they like asking those sort of questions in -dev |
| 08:32.39 | *** join/#asterisk benasse (~chatzilla@130.79.58.2) |
| 08:38.48 | schmidts | thats why i said ask digium directly first ;) |
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| 08:58.55 | wdoekes | KNERD: can't you use udev to rename the device to eth0 ? |
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| 09:02.19 | KNERD | wdoekes: I don't know...never done that, but I do know REGISTER will want a MAC address |
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| 09:06.58 | *** join/#asterisk Curs0r (~Curs0r@unaffiliated/curs0r) |
| 09:07.07 | Curs0r | o/ |
| 09:07.13 | Curs0r | Hi all :) |
| 09:07.22 | KNERD | \o/ |
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| 09:08.32 | CommaCrazy | schmidts, holy snap yeah that is large scale... nice, wish i could work on that for the exp :D |
| 09:09.14 | _val_ | schmidts: dialplan show , shows the extension being loaded.. but I'm still unable to make a call. |
| 09:09.32 | Curs0r | I don't suppose anyone could link me to an extremely novice-friendly how-to about setting up dahdi. Apparently I got the card talking to the system, that's good. But I see already compared to using SIP with asterisk I might be in over my head a little (for now) |
| 09:09.50 | _val_ | exen => _0X., ... seems not to work... or I'm doing something wrong |
| 09:11.00 | Curs0r | Wildcard TDM410P Board 1 OK 1 0 0 CAS Unk Y <-- I am at least hoping that's good news so far hehe |
| 09:13.15 | kaldemar | schmidts: actually, "=" and "=>" both work in extensions. |
| 09:13.22 | kaldemar | _val_: how does it not work? |
| 09:13.50 | kaldemar | _val_: with "core set verbose 10", what do you see in CLI when you make a call? |
| 09:14.12 | _val_ | kaldemar: UNKOWN |
| 09:14.26 | _val_ | I'm not at the machine running Asteriks at this moment. |
| 09:15.00 | _val_ | kaldemar: I now have _0X[1-14].,.... |
| 09:15.15 | kaldemar | "UNKOWN" does not say much. |
| 09:15.19 | _val_ | I don't know if that's right or not. |
| 09:15.22 | kaldemar | what says that? |
| 09:15.59 | kaldemar | [1-14] is obviously invalid. it will match 1 or 4. |
| 09:16.17 | kaldemar | anything inside square brackets matches a single digit or character only. |
| 09:16.49 | kaldemar | it will not match 1 through 14. the syntax says it as a range 1-1 and 4. |
| 09:17.02 | kaldemar | s/says/sees/ |
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| 09:21.40 | _val_ | I understand |
| 09:21.47 | _val_ | but how to match any number any lenght? |
| 09:21.48 | _val_ | :< |
| 09:22.05 | bulkorok | _X. |
| 09:22.43 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:24.19 | Curs0r | Uh oh, show status doesn't see it anymore hehe *facepalms* |
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| 09:25.48 | kaldemar | _X. matches two or more |
| 09:27.25 | bulkorok | he asked for any number, any length... |
| 09:28.49 | kaldemar | which X. does not match. X! matches one digit numbers too. |
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| 09:47.24 | _val_ | guys please how to put this in? |
| 09:47.46 | _val_ | exten=_0XXXXXXXXXXXX,3,Set(CALLERID(name)=${dialername}) |
| 09:48.05 | _val_ | exten=>_X.,3,Set(CALL.... |
| 09:48.08 | _val_ | shoult this suffice? |
| 09:48.43 | Get_The_Fish | _val_ you are asking for exten => _0X. |
| 09:48.48 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:49.16 | Get_The_Fish | I dont think that's probably really what you want to do, but there it is... |
| 09:50.00 | _val_ | Get_The_Fish: that did not work though. I'm going to check it once again Thasnk |
| 09:52.33 | teloniusz | hi guys, from SIP debug: |
| 09:52.37 | teloniusz | Using INVITE request as basis request - 0fc639b47b5437272c2b6d3842ab75d6@10.10.10.7:5060 |
| 09:53.25 | teloniusz | No matching peer for '(extension)' from '(NAT address):(port)' |
| 09:53.42 | teloniusz | and incoming SIP connection can't go into proper context |
| 09:54.20 | kaldemar | make a peer that matches |
| 09:54.28 | teloniusz | I did, I think |
| 09:54.39 | teloniusz | still... |
| 09:54.40 | kaldemar | you thought wrong. |
| 09:54.51 | teloniusz | did I forget insecure = setting? :> |
| 09:54.56 | kaldemar | no. |
| 09:55.42 | teloniusz | To which IP should this peer be assigned? 10.10.10.7 (behind NAT) or NAT address? |
| 09:55.46 | kaldemar | the issue is that no peer matches. what the insecure parameter does is handled after a matching peer is found. |
| 09:56.02 | kaldemar | ~sipnat |
| 09:56.02 | 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, directmedia, externhost or externaddr, and localnet. |
| 09:56.23 | teloniusz | thx |
| 10:04.52 | jacobw | is it possible to set callerid when func_callerid.so is unloaded? |
| 10:06.32 | wdoekes | jacobw: func_callerid providers the necessary dialplan functions (only). a callerid e.g. set in sip.conf will still work |
| 10:06.44 | wdoekes | s/provider/provide |
| 10:07.08 | *** join/#asterisk dandate2 (~dan@180.190.195.83) |
| 10:07.18 | dandate2 | are outbound providers charging us for ring time or only actual talk time? |
| 10:07.58 | jacobw | my problem is that the uriencode function of func_uri.c/so escapes the leading + in my callerid to %2b which confuses my provider's gateway |
| 10:08.30 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 10:09.23 | wdoekes | dandate2: normally only talk time.. unless someone up the stream decides to answer before forwarding the call |
| 10:09.33 | jacobw | so i'm trying to avoid this function being called, unloading func_uri.so did do that, unloading func_callerid.so avoids callerid dialplan mangling including setting of ${CALLERID(num)} |
| 10:10.10 | wdoekes | jacobw: that's a problem in asterisk. it shouldn't escape + indeed. which version? do you have pedantic=yes? |
| 10:10.36 | wdoekes | jacobw: it doesn't work like that |
| 10:10.39 | jacobw | *unloading func_uri.so did NOT do that* correction |
| 10:10.52 | wdoekes | func_uri is just dialplan functionality for uri escaping |
| 10:11.08 | *** join/#asterisk Vince-0 (c4d7a482@gateway/web/freenode/ip.196.215.164.130) |
| 10:11.22 | jacobw | wdoekes: according to RC3261 and various reviewboard discussions escaping special characters like '+' is the correct behaviour |
| 10:11.30 | Get_The_Fish | ewwww |
| 10:11.39 | wdoekes | no it isn't |
| 10:12.03 | *** join/#asterisk tommy87it (~tommy87it@testing.assicurazione.it) |
| 10:12.13 | wdoekes | rfc3261 is a pain in the butt, in the user-part, the + needs not be escaped. check the abnf |
| 10:14.46 | Curs0r | Ugh... I'm losing my mind trying to set this card up |
| 10:15.36 | _val_ | Hey guys. What does htis mean? [Jun 19 12:13:28] NOTICE[26654]: chan_sip.c:20163 handle_request_invite: Call from 'it' to extension '231233212321' rejected because extension not found. |
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| 10:16.45 | kaldemar | _val_: you don't have an extension that matches 231233212321 |
| 10:17.02 | _val_ | kaldemar: but what about _0X.,..? |
| 10:17.17 | kaldemar | _val_: do you see a zero in 231233212321? |
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| 10:24.25 | Curs0r | Now the dahdi commands have disappeard from asterisk cli... how is that even possible? lol |
| 10:25.23 | Curs0r | No help for Curs0r. I should just leave |
| 10:25.48 | *** part/#asterisk Curs0r (~Curs0r@unaffiliated/curs0r) |
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| 10:29.38 | jacobw | wdoekes: so the + in the contact: header doesn't have to be escaped? |
| 10:30.11 | wdoekes | hm? you're talking about the contact? |
| 10:30.33 | wdoekes | it needn't be.. but the provider shouldn't care what's in the contact either |
| 10:32.05 | jacobw | yeah, the + in the user part of my contact header is being escaped to %2b |
| 10:32.07 | wdoekes | if the From/RPID/PAI does not escape the +, but the contact does, you're seeing a minor bug (that doesn't affect many people, because the contact should be used verbatim and not scanned for anything) |
| 10:32.29 | jacobw | ah, that's exactly the case |
| 10:32.53 | wdoekes | file a bug report on jira |
| 10:33.47 | jacobw | my RPID (SipAddHeader(Remote-Party-ID:...)) is unaffected by URI encoding |
| 10:33.59 | wdoekes | no it's not ;) |
| 10:34.10 | jacobw | it's not? |
| 10:34.12 | wdoekes | try to put a @ in the user-part ;) |
| 10:34.16 | wdoekes | or a % |
| 10:34.56 | jacobw | SipAddHeader(Remote-Party-ID: "Me" <sip:${CALLERID(num)}@${SIPDOMAIN}> |
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| 10:35.29 | wdoekes | we aren't talking about manual header setting here |
| 10:35.44 | wdoekes | in that case you can set the Contact manually too (and watch how things break) |
| 10:36.17 | jacobw | that's what i've tried to do |
| 10:36.47 | G00r | Howdy all, I've had no luck getting this dialplan to work, if anyone's interested in loggin in remotely, getting SIP trunk to ring my extensions and vise versa (extensions dialing out via SIP trunk), I'd be happy to pay em for their troubles. Msg me if you can help |
| 10:38.31 | jacobw | Set{CALLERID(num)=+CCXXX) in [macro-dialout-trunk-predial-hook] produces the same behaviour |
| 10:42.30 | wdoekes | jacobw: you're trying stuff that won't work. is the contact "wrong" for incoming calls or outbound calls or both? |
| 10:43.57 | *** join/#asterisk General_Z0D (~Z0D@184.178.241.149) |
| 10:44.02 | jacobw | outgoing |
| 10:44.18 | General_Z0D | what is a good GUI to use for asterisk that is already running on a debian system |
| 10:44.19 | jacobw | i know what i'm trying to do doesn't make actual sense |
| 10:44.57 | jacobw | but my provider has their own requirements |
| 10:45.12 | Get_The_Fish | jacobw, it's that some freepbx code? I recognize that macro name |
| 10:45.35 | jacobw | i'm using freepbx for the web interface |
| 10:47.18 | wdoekes | and which asterisk version? |
| 10:47.45 | jacobw | 1.6.2.24 |
| 10:48.17 | wdoekes | in that care, you simply need to upgrade, I think |
| 10:48.33 | wdoekes | s/care/case |
| 10:49.01 | wdoekes | uri escaping doesn't mind where it is (user-part or other-part) in 1.6.2 |
| 10:49.10 | wdoekes | so it will unconditionally escape the + |
| 10:49.28 | wdoekes | (but because you have pedantic=no, it won't in the From) |
| 10:49.45 | General_Z0D | can anyone recommend a good web interface for Asterisk? |
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| 10:51.16 | jacobw | ah i see |
| 10:51.27 | jacobw | good, i'll test with a more recent version |
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| 10:55.19 | jacobw | thanks for your help wdoekes, i've been stalled with this issue for some time now. |
| 10:55.27 | *** part/#asterisk BuSyAnToS (~31749@93-44-80-247.ip96.fastwebnet.it) |
| 11:02.14 | _val_ | kaldemar: no zero there |
| 11:02.25 | _val_ | sorry I was away for a moment |
| 11:11.02 | _val_ | kaldemar: ;allowexternaldomains=no |
| 11:11.18 | _val_ | so default is yes |
| 11:11.38 | _val_ | what could be wrong here. Can you explain? |
| 11:17.31 | kaldemar | 13:16 < kaldemar> _val_: you don't have an extension that matches 231233212321 <-- show a CLI output of a call with sip debug enabled, your sip.conf and your extensions.conf and i will give you a better answer. |
| 11:21.02 | _val_ | kaldemar: I can't show the sip.conf because it's full with valid accounts :< |
| 11:21.15 | _val_ | I hope you understand ti. |
| 11:21.56 | _val_ | What will be relevant for the problem to troubleshoot? |
| 11:22.17 | _val_ | I'm willing to provide you anything that's relevant for this matte.r |
| 11:25.32 | kaldemar | are you allowing guest calls? |
| 11:26.04 | _val_ | ;allowguest=no ; Allow or reject guest calls (default is yes) |
| 11:26.46 | kaldemar | is 'it' a guest or should it match a peer or a user? |
| 11:27.15 | *** join/#asterisk tommy87it (~tommy87it@testing.assicurazione.it) |
| 11:27.44 | _val_ | kaldemar: http://pastie.org/4113899 |
| 11:27.52 | _val_ | It is a user |
| 11:28.44 | kaldemar | Dial(SIP/it, 30,tTr) <-- remove the white space |
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| 11:30.01 | kaldemar | 1. make it match a defined defice in sip.conf. 2. configure a context for it unless already configured. 3. make an extension or extensions that match incoming calls. |
| 11:30.31 | _val_ | kaldemar: I want to make outgoing calls. Outgoing calls aren't matched. |
| 11:30.40 | _val_ | Allrigh I'm removing the white space |
| 11:31.04 | *** join/#asterisk tommy87it (~tommy87it@testing.assicurazione.it) |
| 11:31.45 | kaldemar | _val_: everything is matched. "Call from 'it' to extension..." is what you get when a call comes IN to asterisk. |
| 11:32.16 | kaldemar | you must tackle those 3 steps before even thinking about anything going out of asterisk. |
| 11:39.49 | *** join/#asterisk wonderworld (~ww@dsdf-4db54bc2.pool.mediaWays.net) |
| 11:47.47 | jol | Is there someway to stop the current AGI command, making it end early, and let the AGI continue? Such as STREAM FILE on a long sound file, and you want to stop the playback and let the AGI continue. |
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| 12:06.42 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 12:06.46 | jacc0 | hi all! |
| 12:06.51 | jacc0 | :) |
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| 12:25.40 | teloniusz | strange thing: |
| 12:25.43 | teloniusz | Using INVITE request as basis request - 77007fdd2e0c318255e536a3627141b0@10.10.10.5:5060 |
| 12:25.58 | teloniusz | Found peer 'incoming-maur2' for '(number)' etc. |
| 12:26.10 | teloniusz | but i have no trunk incoming-maur2 |
| 12:26.28 | teloniusz | (sip show peers does not show anything like this) |
| 12:26.55 | teloniusz | I had one, but I've removed it from users.conf and reloaded Asterisk |
| 12:27.43 | teloniusz | Do i have to restart it or is there any special way to make Asterisk forget about it? |
| 12:27.53 | jacc0 | sip reload |
| 12:29.17 | teloniusz | nope, still the same... |
| 12:29.38 | jacc0 | did you check sip.conf also? |
| 12:30.29 | kaldemar | teloniusz: you either have it in sip.conf, users.conf or some included file or... |
| 12:30.32 | kaldemar | ~users.conf |
| 12:30.33 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 12:31.02 | kaldemar | or what you reloaded did not do the job, so do a restart. |
| 12:32.14 | teloniusz | # grep incoming-maur2 * |
| 12:32.14 | teloniusz | users.conf:;[incoming-maur2] |
| 12:32.34 | teloniusz | commented out |
| 12:32.50 | teloniusz | still, there is always restart |
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| 12:49.07 | *** join/#asterisk kervan (5561126b@gateway/web/freenode/ip.85.97.18.107) |
| 12:50.33 | kervan | Hello, I'm a newbie. I want to redirect inbound calls to local voip server, that's why I want to setup my own. Do you please suggest me a easy managable voip server? |
| 12:51.54 | [TK]D-Fender | kervan, Asterisk. |
| 12:52.13 | kervan | Is Asterisk also a Voip Server? |
| 12:52.23 | [TK]D-Fender | ~b2bua |
| 12:52.23 | infobot | from memory, b2bua is a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 12:52.24 | [TK]D-Fender | ^ |
| 12:52.44 | [TK]D-Fender | "VoIP server" is a vague term |
| 12:54.27 | jacc0 | I would sugest Asterisk (like most people in here) |
| 12:54.48 | jacc0 | it's like walking into a BMW dealer and asking what the best brand of car would be |
| 12:54.52 | jacc0 | :P |
| 12:55.14 | leifmadsen | jacc0: is it also a mode of transportation? |
| 12:55.30 | din3sh | hey jacc0! |
| 12:55.39 | din3sh | sorry for your team man :p |
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| 12:59.38 | leifmadsen | which team is that? |
| 13:03.03 | wdoekes | the orange team that sucked ~ instead of kicked |
| 13:03.09 | din3sh | hehehe |
| 13:03.19 | din3sh | yeah the orange team |
| 13:03.28 | leifmadsen | oh ya... Danmark! :) |
| 13:03.29 | din3sh | anti-climaxed |
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| 13:04.55 | din3sh | leifmadsen:nope denmark is red :D |
| 13:05.05 | din3sh | this is the orange team |
| 13:05.17 | WIMPy | red and white |
| 13:06.15 | din3sh | right WIMPy |
| 13:06.57 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 13:07.24 | teloniusz | the secret of being satisfied with oneself national team is not having too big expectations >:> |
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| 13:08.32 | din3sh | unless you are spanish !? |
| 13:08.35 | din3sh | ;) |
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| 13:29.38 | kervan | [TK]D-Fender: Were you talking about this : http://developer.berlios.de/projects/b2bua/ ? Or were you talking about a built in module? |
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| 13:43.52 | bulkorok | how can I check that ${FAXOPT(status)} is set!? |
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| 13:46.16 | kaldemar | bulkorok: func ISNULL will help you with that |
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| 13:47.15 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:47.20 | bulkorok | great... I'll check that |
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| 13:54.10 | *** join/#asterisk polysics (~polysics@ip-87-238-15-145.static.adsl.cheapnet.it) |
| 13:54.15 | polysics | hello! |
| 13:54.25 | polysics | I was researching attended transfer a little |
| 13:54.57 | polysics | A calls B, A presses something and B gets on hold, A dials C, C picks up, chats with A, A hangs up and B is now bridged to C |
| 13:55.28 | polysics | notice this is A initiating everything, not the standard "secretary" scenario |
| 13:55.41 | polysics | is that possibile using standard * functionality? |
| 14:00.21 | bulkorok | kaldemar: works for me :-) |
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| 14:14.45 | *** join/#asterisk ThinkGNU- (~ThinkGNU-@216.67.183.210) |
| 14:15.32 | ThinkGNU- | I'm looking to send text to the LCD screen of my phones and I'm not finding a way to do this. I tried SendText() but no luck. Anyone have any ideas? |
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| 14:16.17 | WIMPy | Get a phone that supports it? |
| 14:16.27 | ThinkGNU- | I thought that might be the issue |
| 14:16.51 | ThinkGNU- | Is that a common feature with SIP phones? |
| 14:16.55 | WIMPy | Maybe you're lucky and it's just disabled. |
| 14:17.27 | ThinkGNU- | Well, I doubt that. It's not mentioned anywhere in the documentation for the phone (that I could find) |
| 14:17.31 | ThinkGNU- | It's a Digium D50 |
| 14:17.32 | WIMPy | Featue and SIP phone usually don't go together very well. |
| 14:20.19 | ThinkGNU- | Oh well. I've got the majority of what I was trying to do with the feature. I just thought the text part would be a nice little touch. |
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| 14:24.47 | malcolmd | ThinkGNU-: it would be a nice feature to be able to send text to the phone, yes. it's not currently possible. it's a feature request that we've previously received |
| 14:25.58 | malcolmd | yup |
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| 14:26.24 | ThinkGNU- | Well I do like the D50 overall |
| 14:26.39 | ThinkGNU- | I've had a very easy time provisioning them and I appreciate that |
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| 14:27.02 | malcolmd | yay :) docs were able to get you through the process okay? |
| 14:27.14 | ThinkGNU- | Mostly |
| 14:31.25 | bulkorok | I can not unload res_fax_spandsp.so because: Soft unload failed, 'res_fax_spandsp.so' has use count 10 |
| 14:32.05 | bulkorok | ther is no session running right now... so what it wants from me?! I don't want to restart asterisk... |
| 14:39.33 | *** join/#asterisk cmnky (~cmnky@unaffiliated/cmnky) |
| 14:41.28 | cmnky | I asked about blocking telemarketing calls yesterday, and was advised I could use my SPA2102 and asterisk to do this ... but I'm confused as to whether it will work and I think there was some miscommunication ... I have analog phone service .. the SPA2102 has 2 FXS ports ... can I plug the line into 1 port, and the phone into the other ? that would be an FXO and FXS config wouldn't it ? |
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| 14:45.40 | drmessano | No |
| 14:45.50 | drmessano | FXS connects to a PHONE, FXO connects to a LINE |
| 14:45.57 | drmessano | You should have gotten an SPA 3102 |
| 14:47.23 | cmnky | drmessano, i had this box lying around .... so i've not actually purchased any equipment yet ... im still researching ... and after i thought about yesterdays conversation i realized this might be an issue, as the person i was talking to about it probably didn't realize i don't have VoIP service ... its verizons digital voice .. which as far as my gear is concerned, is POTS |
| 14:48.27 | cmnky | so since i have 2 phone lines ... i need 2 FXO ports, to connect verizons ONT -> asterisk box .... then I can use this SPA2102 to connect asterisk box <-> analog phones ... correct ? |
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| 14:49.50 | drmessano | Yep |
| 14:50.09 | cmnky | okay .. thanks for the clarification |
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| 14:52.19 | gtTuna | what are some disadvantages to using a FreePBX distro rather than just installing it ontop of CentOS ? |
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| 14:56.35 | leifmadsen | gtTuna: more maintenance I suppose |
| 14:56.47 | leifmadsen | longer lead time to configuration |
| 14:56.56 | Qwell | he said disadvantages |
| 14:57.43 | Qwell | There are really very few, if you choose the right one. |
| 14:59.05 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 15:00.23 | leifmadsen | ah yes, I inverted the question |
| 15:03.00 | gtTuna | we're having an argument about it here in the office...i think it's just as easy to do a fresh CentOS install |
| 15:03.11 | Qwell | Well you're wrong. :) |
| 15:03.17 | gtTuna | haha, fair enough |
| 15:03.30 | leifmadsen | what Qwell said |
| 15:03.51 | Qwell | But really, you'd be spending more time trying to either build packages for everything you need, or finding some repository to get them from |
| 15:03.55 | leifmadsen | manually installing something on top of an OS is always going to be some degree more difficult than letting the installation media set it up for you |
| 15:07.37 | gtTuna | understood |
| 15:08.40 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 15:33.15 | RZero | Hi guys is it possible to do voice activated recordings on asterisk, current using backgrounddetect which works but there is a slight delay between voice activity and the start of the recording so it misses the first part of the message of. |
| 15:34.22 | RZero | Does any one know of a better solution rather than backgrounddetect method |
| 15:36.15 | WIMPy | Do it after recording. |
| 15:37.03 | *** join/#asterisk tommy87it (~tommy87it@testing.assicurazione.it) |
| 15:37.30 | *** part/#asterisk cmnky (~cmnky@unaffiliated/cmnky) |
| 15:38.23 | Katty | hello my asterisk does not work at all how to fix plz |
| 15:38.29 | aberrios | :S |
| 15:38.49 | aberrios | Katty: please provide more information....8| |
| 15:39.04 | Katty | what ifnormation would u like plz?? |
| 15:39.14 | Katty | where is Qwell |
| 15:39.24 | aberrios | he's probably in #asterisk |
| 15:39.27 | aberrios | hangon |
| 15:39.32 | Katty | ^_- |
| 15:39.35 | Katty | you, sir, have insaned. |
| 15:39.36 | aberrios | DOH! |
| 15:39.52 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 15:39.52 | aberrios | indeed... 5 redbulls later.... |
| 15:39.57 | Katty | GASP |
| 15:40.04 | Katty | you will dies. |
| 15:40.38 | RZero | WIMPy do you mean do the backgrounddetect after the recording ? |
| 15:41.14 | Katty | where is everyone |
| 15:41.15 | aberrios | Katty: do you have any * cli output or is it just not running at all? |
| 15:41.19 | Katty | what? |
| 15:41.21 | RZero | Im trying to build some thing like Grunt detection |
| 15:41.22 | Katty | oh |
| 15:41.28 | Katty | aberrios: you were just cleverly trolled, dear. |
| 15:41.34 | Katty | aberrios: pay no attention to me. |
| 15:41.34 | aberrios | doh |
| 15:41.40 | aberrios | :P~ |
| 15:41.57 | aberrios | maybe I'll just /away |
| 15:42.03 | Katty | mkay |
| 15:43.11 | aberrios | grabs one last redbull for the day |
| 15:43.41 | Katty | NO! |
| 15:43.43 | Katty | put it down. |
| 15:43.50 | Katty | you will induce congestive heart failure. |
| 15:44.21 | aberrios | Hell I might even smoke too.... You drove me to it! You and your trolling. |
| 15:44.34 | Katty | admit it. you liked it. |
| 15:45.20 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 15:45.53 | aberrios | i thought I was in #freepbx... maybe i should just close down all these root terminals I have open. I'm probably going to regret some typos somewhere..... |
| 15:46.54 | Katty | i should scold you over having root terminals open, rather than suing. |
| 15:46.55 | generalhan | i dont know what my issue is with remembering FXO v FXS ... but everytime i put together a new system i have to ask to make sure i get it right. I have 4 phone lines coming in from my provider and 8 POTS, analog, phones. so i need 4 FXO ports, and 8 FXS ports, right ?! |
| 15:47.36 | *** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:5:0:558:12) |
| 15:47.54 | aberrios | You can typo with su too.... |
| 15:48.18 | Katty | i'll su your mom in a minute |
| 15:48.22 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 15:49.06 | *** join/#asterisk ccesario (~ccesario@189.29.37.189) |
| 15:49.11 | Katty | generalhan: put this on a post it note |
| 15:49.13 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 15:49.20 | Katty | generalhan: and tape it to your wall |
| 15:49.22 | Katty | generalhan: or monitor |
| 15:49.34 | Katty | generalhan: optionally your forehead. |
| 15:49.40 | generalhan | pfft, Tattoo FTW |
| 15:49.45 | Katty | generalhan: FX_O_ = P_O_Ts |
| 15:50.12 | Katty | FX_S_ = analog phones |
| 15:50.27 | Katty | post it? post it? good? great. |
| 15:50.55 | Katty | foreign exchange Office |
| 15:50.59 | Katty | Foreign exchange station |
| 15:51.08 | Katty | forehead! |
| 15:51.12 | volga629 | I have some problem with IMAP storage, it not delivery into specify folder |
| 15:51.50 | Katty | eat two cookies and drink half a bottle of whipped cream vodka |
| 15:51.54 | Katty | then call me in the morning |
| 15:52.06 | generalhan | Katty: the forehead tattoo is no good ... i would have to look in the mirror before i could install. im thinking maybe, back of the hand |
| 15:52.12 | aberrios | thinks Katty has also been on the RedBull |
| 15:52.24 | Katty | no i'm naturally like this. |
| 15:52.27 | Katty | i blame organic fruit. |
| 15:53.01 | Katty | generalhan: i think a post it note would do nicely. |
| 15:53.18 | Katty | generalhan: i've got a bazillion of them, everywhere |
| 15:54.13 | generalhan | what if you were offsite doing an installation ?! |
| 15:54.23 | generalhan | tattoos are more mobile ! |
| 15:54.24 | volga629 | imap - S: 0000060e BAD parse error: invalid message sequence number: 1 This error from zimbra imap |
| 15:54.43 | Katty | generalhan: do you carry a smart phone? |
| 15:54.53 | generalhan | doesnt everybody |
| 15:55.06 | Katty | well there you go. problem solved. |
| 15:56.07 | generalhan | why even write your stuff down on a post it at all then ? |
| 15:56.31 | Katty | helps drive it into the brain |
| 15:58.11 | aberrios | WIKI! |
| 15:58.24 | aberrios | Wiki > PostIT |
| 15:59.15 | WIMPy | RZero: Yes, that was my idea. |
| 15:59.23 | Katty | tiki > wiki |
| 15:59.27 | Katty | tiki torches are epic. |
| 15:59.31 | Katty | and flamey!! oooo. |
| 16:02.52 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 16:07.40 | KNERD | Any idea on installing that Asterisk fax on a VPS without access to eth0? |
| 16:13.36 | RZero | WIMPy sorry to bug you, but how would that work in detecting the voice ? |
| 16:14.31 | WIMPy | Some audio tool should be able to do it. |
| 16:15.15 | WIMPy | If it can be done to split recordings of albums in to single tracks by detecting silence, it should be possible to remove it as well. |
| 16:16.25 | RZero | ah I see |
| 16:23.20 | coppice | KNERD: why do you need access to eth0? |
| 16:23.51 | KNERD | coppice: because the REGISTER application wants it |
| 16:24.10 | pigpen | so anybody here use Audiocodes FXO's that has experienced audio quality issues? |
| 16:24.15 | coppice | KNERD: then use spandsp, and avoid the problem |
| 16:24.16 | pigpen | ie: sip gateway |
| 16:24.27 | KNERD | coppice: thanks |
| 16:24.50 | anonymouz666 | hehe.. who saw the video of linus torvalds talking about nvidia? |
| 16:25.29 | anonymouz666 | that was what I call love |
| 16:25.36 | coppice | anonymouz666: there must be more to that than has been made public |
| 16:26.25 | anonymouz666 | why nvidia make things harder for kernel devs? |
| 16:27.45 | coppice | nvidia makes sure their stuff works really well on linux. AMD has gone open source, but their support it hopeless. they don't even have opencl for linux. who uses opencl on anything but linux |
| 16:34.01 | drmessano | I have had a love/hate relationship with Nvidia hardware in Linux, but never to the point that I expected a comment like that to come from Linus. Really, there is hardware that is FAR FAR FAR worse under linux |
| 16:34.48 | WIMPy | It's probably about the amount of users affected. |
| 16:35.11 | coppice | drmessano: the only badly stuff these days seems to be scanners |
| 16:35.26 | coppice | s/badly/badly supported |
| 16:37.01 | jacobw | scanners that can mail are cool |
| 16:37.45 | drmessano | So is Fire.. Been around for almost as long too :) |
| 16:44.58 | jacobw | :) |
| 16:45.36 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 16:47.18 | *** join/#asterisk slav3_kitten (~kitten@unaffiliated/slav3-kitten/x-0866809) |
| 16:48.02 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:48.03 | slav3_kitten | well con was a roit, my friend isn't feel well though. she somehow happened upon poison ivy and bad food |
| 16:48.33 | slav3_kitten | oh well, still going to see if i can arrange to get her and her boyfriend back out next year |
| 16:48.47 | jpsharp | con death plague |
| 16:48.50 | *** join/#asterisk bn-7bc (~bjarne-im@2001:16d8:ee6c:1:d5b0:75c3:846b:7ce) |
| 16:48.54 | volga629 | Is any tips for srtp how stabilize ? |
| 16:49.16 | slav3_kitten | jpsharp, yea :( |
| 16:50.01 | slav3_kitten | plus side is she's being a real trooper and we are going to go see some attractions she always wanted to see in america, and i'm getting stories about her RAF stuff and snuggles |
| 16:51.53 | *** join/#asterisk Defraz (~Defraz@67-60-210-130.cpe.cableone.net) |
| 16:52.00 | *** join/#asterisk wonderworld (~ww@dsdf-4db54bc2.pool.mediaWays.net) |
| 16:52.08 | drmessano | stablize srtp? |
| 16:52.26 | coppice | give it training wheels |
| 16:52.34 | drmessano | ah |
| 16:52.39 | slav3_kitten | duct tape, that's always the answer |
| 16:53.19 | drmessano | I didnt know srtp had to be stabilized. Perhaps it requires a crystal oven or an inert gas? |
| 16:55.15 | drmessano | srtp_tcxo.so |
| 16:55.42 | drmessano | Provides temperature stability for SRTP endpoints. |
| 16:57.26 | *** join/#asterisk gusto (~gusto@ip-109-84-0-114.web.vodafone.de) |
| 16:57.29 | jpsharp | I prefer GPS disciplined XO. |
| 16:57.43 | jpsharp | *smack* STAY IN PHASE |
| 16:58.02 | drmessano | lol |
| 16:59.05 | drmessano | Is that anything like GPS time stabilization on a device that is read indirectly? |
| 16:59.35 | slav3_kitten | can't you use a ntp server? |
| 16:59.44 | drmessano | No |
| 17:00.12 | slav3_kitten | < knows little about srtp but is reading the rfc now |
| 17:00.27 | drmessano | Synchronization of audio is a pain in the ass |
| 17:00.43 | slav3_kitten | *nods* |
| 17:00.51 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 17:01.01 | jpsharp | NTP using a GPS receiever + 1 PPS output? |
| 17:01.14 | coppice | rubidium is the answer |
| 17:02.16 | jpsharp | That'll give you excellent relative time, but you still need a marker that says "Okay, it is 16:02:00 UTC....wait for it...wait for it.....NOW!" |
| 17:02.18 | drmessano | We use the 1 PPS output on the device, but it's polled in realtime. There's a 10MHZ clock between the devices and the GPS is used to used to reset the clock sync when it drifts |
| 17:02.36 | drmessano | NOT polled in realtime, rather |
| 17:03.02 | drmessano | Its pretty stupid. The whole thing was designed poorly from the start |
| 17:03.28 | slav3_kitten | aah |
| 17:03.40 | jpsharp | Sounds like it. Hell, my crappy linux box here keeps better time than that. |
| 17:03.54 | drmessano | They used ReiserFS, a very old implementation of it.. So the box wigs out and kills itself every so often. Happy accident on the pun. |
| 17:04.30 | jpsharp | My GPS receiver feeds NMEA + 1 PPS into NTPD. It is never more than a couple ms off of time.nist.gov. |
| 17:04.35 | coppice | does it have a "Reiser Inside" sticker on it? |
| 17:04.41 | drmessano | lol |
| 17:05.25 | drmessano | Wouldnt be much of a point in putting a sticker on the cover.. its spends most of its time with the cover off, while I am recovering it |
| 17:05.38 | drmessano | Actually, "restoring it" would be a better way of putting it |
| 17:06.05 | drmessano | "Oh look, the FS pulled a Hans again. Time for a reload" |
| 17:06.23 | *** join/#asterisk tzanger (tzanger@wallace.mixdown.ca) |
| 17:06.58 | jpsharp | Your server killed your wife? |
| 17:07.03 | drmessano | lol |
| 17:07.55 | drmessano | Its actually whats called an HD Exporter.. Its what generates HD Radio signals before they are fed to the transmitter |
| 17:08.07 | drmessano | HD Radio .... All runs on Linux |
| 17:08.55 | jpsharp | OTA HD Radio? |
| 17:09.18 | drmessano | Yep |
| 17:09.56 | jpsharp | Does it actually sound that much better? I've never listened to it. |
| 17:10.28 | drmessano | It does. Its amazing how much analog FM removes from a piece of audio |
| 17:10.58 | slav3_kitten | digital radio works great so long as you don't have a weak signal |
| 17:11.10 | drmessano | Yep |
| 17:11.14 | jpsharp | digital anything |
| 17:11.16 | volga629 | How can check if srtp working ? |
| 17:11.41 | slav3_kitten | with FM you can still make out what's being said in poor conditions, with digi you're screwed |
| 17:12.11 | slav3_kitten | which is why you don't see dstar emergency repeaters |
| 17:12.20 | drmessano | Your radio is supposed to fall back to Analog, seamlessly, when the digital drops out. You should never hear that |
| 17:12.31 | jpsharp | That and nobody wants to pay the damn AMBE tax :) |
| 17:12.42 | *** join/#asterisk Node_403 (b4ea7d07@gateway/web/freenode/ip.180.234.125.7) |
| 17:12.50 | coppice | slav3_kitten: the emergency services were some of the earliest adopters of digital radio |
| 17:13.11 | jpsharp | volga629: tcpdump your traffic and see if wireshark can see the RTP as something it interpret or no. |
| 17:14.11 | drmessano | We put a significant amount into syncronizing the HD and Analog audio.. Most of the time its near perfect blend when youre getting near the edge of the city grade contour |
| 17:14.31 | slav3_kitten | coppice, i know of not a single dstar emergency repeater in my local area. there are a number of echolink and dstar repeaters but none are RACES or ARES |
| 17:14.33 | drmessano | If thats not what youre getting from a particular station, you should complain to the engineer |
| 17:14.54 | volga629 | yes, tried tcpdump -i eth0 -vvvvvnnnnnn host ip |
| 17:15.17 | drmessano | slav3_kitten, I thats a matter of ARES not wanting to exclude anyone. ARES almost begs for operators most of the time. |
| 17:15.26 | volga629 | I see rtp ports, but nothing mention about encryption |
| 17:15.36 | drmessano | You move an ARES net to a DSTAR repeater and you hear crickets |
| 17:15.37 | slav3_kitten | i know police an ambulance use digital for crypto around the area |
| 17:16.10 | slav3_kitten | but i've had many cops bitch about it's performance on the fringes during terrible storms |
| 17:16.30 | slav3_kitten | speaking of DSTAR i really need to get a better radio for my car... |
| 17:16.40 | jpsharp | volga629: use wireshark to interpret the stream. if it is encrypted, it should show up as RTP but not be able to see what kind of codec it is. If it is not encrypted it can say "oh, this is G711 or G729 or whatever you're using". |
| 17:16.53 | drmessano | Hams are crybabies... You move an ARES or Skywarn net to a Digital repeater.... or hell, one with PRIVATE LINE, and they go NUTS |
| 17:17.01 | volga629 | in asterisk I should see line == Using SIP RTP TOS bits 184 |
| 17:17.10 | volga629 | or SRTp |
| 17:17.17 | volga629 | let me try |
| 17:17.27 | slav3_kitten | drmessano, actually our skywarn uses PL 102.7hz |
| 17:17.33 | coppice | slav3_kitten: REAL emergency services |
| 17:17.34 | slav3_kitten | no one has ever bitched that i heard |
| 17:17.42 | jpsharp | I don't have a problem with digital repeaters. Its when, in a real emergency, the guy who's running the link to the repeater, his house blows away or floods. |
| 17:17.59 | drmessano | slav3_kitten, how many of your operators are over the age of 60? |
| 17:18.09 | slav3_kitten | drmessano, none... |
| 17:18.13 | slav3_kitten | i'm not even 30 |
| 17:18.15 | drmessano | .... lol |
| 17:18.42 | volga629 | RTP: set_destination RP74 adr=ip:17774 this from snom look srtp not going |
| 17:18.46 | slav3_kitten | so you're saying old hams can't figure out DCS or PL tones? |
| 17:19.08 | drmessano | I would say 40% of the active hams in this area are over 60. I put PL on my 440 repeater and they acted like I have commited some sort of genocide |
| 17:19.21 | jpsharp | YOU'RE LOCKING IT OUT! |
| 17:19.37 | slav3_kitten | ... but PL tones prevent things like... storms an noise from transmitting... |
| 17:19.37 | jpsharp | Same thing happened at an old club of mine. |
| 17:20.04 | drmessano | Yeah, WHY DONT YOU WANT US TO USE THE REPEATER BOX |
| 17:20.20 | slav3_kitten | ... |
| 17:20.29 | drmessano | lol |
| 17:20.46 | slav3_kitten | well fuck em, they are idiots |
| 17:21.32 | jpsharp | Then again, these guys vapour lock on anything over 30Mhz. |
| 17:21.43 | drmessano | We had a push to put up a 146.94/34 repeater.. and I told everyone that someone needs to build it tube type and put it on a tower at the old folks home |
| 17:22.09 | jpsharp | Heh...9-4 10-4 |
| 17:23.03 | drmessano | But what can you do |
| 17:23.06 | slav3_kitten | i want to face palm so hard |
| 17:23.09 | drmessano | 1. Tell them to stop being 100 |
| 17:23.15 | drmessano | 2. Drive them to the cemetary |
| 17:23.37 | jpsharp | But to make it true amateur style, you have to put a 500W PA and a badly tuned duplexer on it so you can hear it 200 miles away but not be able to get into it from more than 5 miles away. |
| 17:23.42 | drmessano | 3. Take their dues and make sure theres lots of repeaters with no PL and CW IDers |
| 17:23.50 | drmessano | hahha |
| 17:24.32 | drmessano | mistune the input filter cavity on the transmit side so a spur keeps the repeater keyed up 12 to 14 hours a day |
| 17:24.50 | drmessano | WohWohWohWohWohWohWohWohWohWoh |
| 17:25.07 | slav3_kitten | i think you guys have either had really bad experiences |
| 17:25.14 | slav3_kitten | or i've had really good ones |
| 17:25.18 | jpsharp | Both. |
| 17:25.32 | slav3_kitten | then again most my life has been pretty good to me aside from the fight with gravity five years ago |
| 17:25.57 | *** join/#asterisk TheKernel[work] (~tcrowe@unaffiliated/the-kernel) |
| 17:26.08 | *** join/#asterisk bjweeks (~bjweeks@wikipedia/Brandon) |
| 17:26.18 | slav3_kitten | my british friend i flew out has been talking with me a lot and really become even closer friends then we had been. aside from her not feeling good and such it's been a wonderful week or so |
| 17:26.28 | jpsharp | But not just in the realm of Part 97. I've seen commercial systems that are some doozies. |
| 17:27.26 | jpsharp | the crack about the 500W PA and duplexer came from a sheriff's dept radio system in Texas. |
| 17:27.26 | Nugget | Don't mess with Texas. |
| 17:27.42 | slav3_kitten | i'm off to go grab some food guys. sorry to run but i'm starving |
| 17:28.13 | jpsharp | The transmitter was northwest of San Antonio, you could hear it in corpus christi, but not get into it from 20 miles away. |
| 17:30.11 | volga629 | if I see a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline this TLS no SRTP ? |
| 17:31.10 | jpsharp | If that's in your SIP invite, then that's SRTP. |
| 17:32.02 | volga629 | let me see if it is INVITE |
| 17:35.30 | *** join/#asterisk RubyRails (~justin@209.33.214.243) |
| 17:35.49 | RubyRails | in Asterisk 1.8, are files recorded with MixMonitor still processed by Sox? |
| 17:36.15 | volga629 | Look like babytel have different rtp ports then I open on firewall Peer audio RTP is at port 204.101.5.68:23666 |
| 17:36.16 | TheKernel[work] | Hi does any one know where I can find the change log between version 1.8.8 and 1.8.12? |
| 17:36.56 | Qwell | TheKernel[work]: The same place you'd get 1.8.12. |
| 17:38.54 | Katty | glomps Qwell |
| 17:39.18 | jpsharp | averts his eyes. |
| 17:42.00 | Katty | all my coworkers make me wanna AHHHHHHHHHHHHHHH |
| 17:42.59 | TriJetScud | Katty: when did you learn how to "glomp" people? |
| 17:43.09 | jpsharp | They wouldn't be coworkers if they didnt' |
| 17:45.11 | drmessano | lol |
| 17:45.33 | *** join/#asterisk Micc (~Micc@50.46.224.219) |
| 17:45.53 | *** join/#asterisk oej (~olle@95.209.36.241.bredband.tre.se) |
| 17:46.16 | drmessano | jpsharp: One of my managers has a repeater near a ridge. Sounds great on the other side, but the repeater is deaf as a hammer |
| 17:46.43 | drmessano | RF is a beautiful thing |
| 17:47.21 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:49.53 | TriJetScud | drmessano: did you ensure that the antennas were pointing in the right direction? |
| 17:50.17 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 17:51.12 | drmessano | Pointing? I am sure this was omnidirectional |
| 17:52.39 | TriJetScud | drmessano: most antennas are directional to some degree |
| 17:52.48 | TriJetScud | look up what a radiation pattern is |
| 17:53.02 | drmessano | LOL |
| 17:53.17 | drmessano | I am well aware of radiation patterns |
| 17:55.35 | drmessano | A typical repeater antenna, consisting of stacked folded dipoles is going to have a lobe to the horizon in a 360 degree pattern, minus the small null from the tower or mounting bracket |
| 17:59.52 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-120-227.socal.res.rr.com) |
| 18:02.59 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:04.48 | *** join/#asterisk logicwrath_work (~no@74-94-239-197-Michigan.hfc.comcastbusiness.net) |
| 18:06.20 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 18:07.56 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:15.25 | tzanger | god why does anyone use elastix |
| 18:16.08 | tzanger | guy called support to configure a te405 for two spans PRI one span CAS T1. support says the card doesnt' support it. funny but it was working for 4 years before the HDD crash |
| 18:19.40 | volga629 | <PROTECTED> |
| 18:21.31 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:26.02 | volga629 | <PROTECTED> |
| 18:32.41 | *** join/#asterisk din3sh (~din3sh@41.136.87.147) |
| 18:36.29 | drmessano | Elastix is horrid |
| 18:37.17 | KNERD | Qwell: How can I get around the Asterisk FAX register app needing eth0 and a MAC address as I am on a VPS. I was looking at SpanDSP, but it indicates it needs 3 files in which you use to recompile asterisk (even 1.8) with, but they seem to be no longer available. |
| 18:37.55 | Qwell | I don't think you need eth0. |
| 18:38.22 | KNERD | the REGISTER app won't continue because it cannot find eth0 |
| 18:39.00 | KNERD | -> Could not generate Host-ID. Make sure that you have eth0 enabled. |
| 18:39.04 | leifmadsen | Qwell: register app depends on eth0 I think |
| 18:39.19 | leifmadsen | KNERD: call digium support |
| 18:39.31 | leifmadsen | also, I'm not sure there is a work around |
| 18:39.38 | [TK]D-Fender | Make an alias |
| 18:39.39 | KNERD | Will the say "Credit Card please" |
| 18:39.48 | leifmadsen | I don't know, I'm not digium support |
| 18:40.13 | leifmadsen | if you purchased a product (which I assume you did if you're trying to use the register application) support should be included for an incident |
| 18:40.18 | KNERD | The VPS does not even have access to the host machine eth0 information |
| 18:40.43 | leifmadsen | this isn't the appropriate forum for support with digium commercial products |
| 18:40.59 | KNERD | well..it was the free fax license |
| 18:41.12 | p3nguin | of a commercial product. |
| 18:41.26 | KNERD | okay..i guess I wil give em a try |
| 18:41.27 | KNERD | thanks |
| 18:42.03 | Qwell | leifmadsen: it doesn't anymore |
| 18:42.41 | *** join/#asterisk chasing`Sol (~cS@41.232.125.163) |
| 18:43.16 | Qwell | KNERD: Do me a favor and pastebin the output of ifconfig? Just curious. |
| 18:43.29 | KNERD | sure... |
| 18:43.55 | Qwell | actually, ifconfig -a |
| 18:43.59 | leifmadsen | Qwell: maybe he is using an older version of the registration client |
| 18:44.09 | Qwell | Then he needs to use something built in the last 18 months. :p |
| 18:44.17 | leifmadsen | :) |
| 18:44.25 | leifmadsen | throws out random and wild guesses |
| 18:44.32 | KNERD | Qwell: http://pastebin.ca/2162981 |
| 18:44.35 | Qwell | I have a hunch. |
| 18:44.46 | Qwell | Yep. Nailed it. |
| 18:44.48 | Qwell | Fix your VPS. |
| 18:44.56 | KNERD | "fix" it? |
| 18:45.01 | Qwell | yes |
| 18:45.12 | KNERD | Please explain |
| 18:45.59 | Qwell | Your network interfaces are not valid interfaces. Some (many?) switches would actually discard all packets from them. |
| 18:46.39 | KNERD | Not sure how I can fix it. I do not have access to the host |
| 18:50.16 | Qwell | This is why you don't use VPSes. |
| 18:50.24 | Qwell | They are garbage. Always. Every single one. |
| 18:50.53 | KNERD | they are good for us poor folks |
| 18:51.00 | Qwell | VMs... |
| 18:51.25 | Qwell | Zero overhead, no custom kernel crap. |
| 18:54.21 | *** join/#asterisk FinboySlick (~shark@74.117.40.10) |
| 18:56.39 | KNERD | maybe someday I will be able to gradumuate to a real server |
| 18:56.58 | KNERD | but for $100 a year for VPS....gotta gor for that |
| 18:57.38 | Qwell | How much time have you spent dealing with stupidity? How much time do you think you will, over that year? |
| 18:57.48 | Qwell | Certainly more than the $100. |
| 18:58.37 | FinboySlick | Hello. I'm trying to relay calls from my asterisk box to a metaswitch system. It's a pretty straightforward Dial(SIP/metaswitchtrunk/metaswitch-connected-phone) when asterisk gets the call and it works. My problem is that the callerID of the call does not get relayed. doing a tcpdump of the sip communication shows the callerid name in the From: field but I don't see the calling number anywhere in the INVITE. Where should it be and how do I put it th |
| 18:59.25 | Qwell | FinboySlick: Search the sip.conf sample config for rpid. Enable that. |
| 18:59.53 | FinboySlick | Qwell: Sounds like a winner. Thanks for enlightening me :) |
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| 19:04.36 | volga629 | what is mean ? __sip_xmit: sip_xmit of 0x2ec9b80 (len 823) to :49664 returned -2: No such file or directory |
| 19:07.06 | FinboySlick | Qwell: The INVITE message looks good... But the metaswitch won't let it through. Still, many thanks. |
| 19:08.53 | FinboySlick | OK, that last bit was a typo, it lets it through, just seems to ignore the Remote-Party-ID header. |
| 19:09.16 | FinboySlick | I wish that big expensive dummy thing was as well made as asterisk. |
| 19:09.53 | Qwell | FinboySlick: get rid of it :p |
| 19:10.27 | FinboySlick | Qwell: For $400k, I'll let you get it off my hands ;) |
| 19:10.44 | FinboySlick | Just make sure nobody's around when you steal.. I mean, take it away. |
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| 19:20.09 | ibercom | Anybody know what means "Unable to forward frametype: 2" ? |
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| 19:21.39 | infinity_ | I have a phone behind NAT and I can't get it to register. it says UNREACHABLE. http://pastebin.com/s1vff4Qa |
| 19:21.53 | infinity_ | its a polycom phone behind NAT. what should i look for? |
| 19:22.33 | ibercom | I have a lot of errors of this type. What should I look for ? |
| 19:24.36 | Katty | throws things, shreds curtains |
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| 19:35.50 | chuckf | bad Katty |
| 19:46.43 | RubyRails | anyone know of an IRC place to talk about sox ? |
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| 19:50.51 | drmessano | #shuuz |
| 19:52.34 | ngharo | let's talk about sox, baby |
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| 20:01.37 | malcolmd | let's talk about resampling |
| 20:05.25 | anonymouz666 | i'd choice to use sox to output the recording calls to a dat file |
| 20:05.36 | anonymouz666 | and plot the result into a waveform |
| 20:06.01 | anonymouz666 | then we associate the waveform to the record and show to the callcenter leads |
| 20:10.28 | anonymouz666 | then, just by looking at the waveform (and not manually listening one by one) the leads could identify if a certain call is a possible candidate to problem or no |
| 20:10.39 | anonymouz666 | make sense? |
| 20:15.31 | Get_The_Fish | anonymouz666 thats a pretty damn cool idea, I really like that. |
| 20:16.22 | anonymouz666 | good |
| 20:16.28 | Get_The_Fish | you could get a lot of info from the change in vol vs. mean, especially if you looked at each side of the conversation individually. |
| 20:17.03 | Get_The_Fish | on the agent side you would have a pretty good dataset to sample from for mean vol after a while. You would be able to tell when they are yelling very quickly |
| 20:17.07 | *** join/#asterisk sam555 (~sysadmin@udp124488uds.hawaiiantel.net) |
| 20:17.14 | Get_The_Fish | or when someone is yelling into an IVR |
| 20:17.34 | sam555 | does anyone know if you need special drivers to run asteriskNOW on raid drives? |
| 20:17.59 | Get_The_Fish | (which I do all time hoping that the more sophisticated ones pick it up and route me to an agent immediately. It works for some places (amex, chase) |
| 20:19.46 | *** join/#asterisk omani (~hasan@33.37.69.80.in-addr.net-lab.net) |
| 20:20.30 | omani | with my asterisk box everything works fine, except one thing: when I intiate a call and wait til ringing, after hang up it takes too long to hangup the line |
| 20:20.34 | omani | what can I do? |
| 20:22.25 | omani | for testing purposes I call my self (from context "out") and hang up. asterisk tells "spawn extension extied non-zero...", but still the phones are ringing |
| 20:22.46 | omani | after few seconds (which takes too long imho) the ringing ends. |
| 20:24.25 | anonymouz666 | Get_The_Fish: I don't know if we can split each side of the conversation, because they are mono at 8000Hz, unless you save in another format |
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| 20:24.37 | anonymouz666 | If I understand correctly in mono recording we have only one channel |
| 20:25.05 | Get_The_Fish | I'll bet there is a way. Asterisk has to mix them. |
| 20:25.11 | TriJetScud | throws a cat at Katty |
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| 20:42.07 | RubyRails | @anonymouz666 - are you using Monitor or MixMonitor |
| 20:42.08 | RubyRails | ? |
| 20:47.13 | anonymouz666 | mixmonitor, sure! |
| 20:47.19 | anonymouz666 | monitor is very old stuff |
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| 20:57.40 | RubyRails | we just had to switch our main app from mix monitor to monitor |
| 20:57.41 | RubyRails | :( |
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| 21:02.46 | [TK]D-Fender | Fail to see how that's "bad" |
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| 21:14.38 | volga629 | for lack of RTP activity in 31 seconds how make sure that rtp traffic is going except firewall |
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| 21:16.52 | volga629 | ~nat |
| 21:16.52 | infobot | hmm... nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 21:24.16 | volga629 | tcptls.c:397 ast_tcptls_client_start: Unable to connect SIP socket to :32656: Connection timed out |
| 21:24.42 | volga629 | have some nat issue which can't track down |
| 21:25.17 | volga629 | and rtp is flow properly sometime go sometimes no |
| 21:25.56 | volga629 | and Incorrect password '' for user DTMF not detecting |
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| 21:48.23 | KNERD | Anyne messed with SpanDSP on Asterisk and where to get those 3 files used to compile with Asterisk? app_rxfax.c, app_txfax.c and apps_makefile.patch |
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| 22:02.42 | pigpen | anybody have experience with the 1.8 function DENOISE() ? |
| 22:02.47 | pigpen | on analog lines? |
| 22:02.59 | pigpen | or with analog lines ultimatly. |
| 22:04.06 | p3nguin | Is that really what you're trying to find out? |
| 22:04.42 | pigpen | p3nguin, well, true. how does it work for you? |
| 22:05.03 | pigpen | so ultimatly: a. used it b. thoughts |
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| 22:40.41 | Get_The_Fish | I know its OT, but any git users here? Any way to automagically accept the host verification on initial ssh connection (so it can be scripted)? |
| 22:41.00 | Get_The_Fish | I just figured out a way around it (known_hosts), but for posterity... |
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| 22:51.18 | talntid | [Jun 19 15:49:44] WARNING[8994]: app_voicemail.c:7767 play_message: No origtime?! |
| 22:51.22 | talntid | this is a new one on me... |
| 22:54.45 | talntid | cleared the vm dir, left new vm, and it works fine. |
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