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| 01:20.53 | Penguin | wimpy: I've been messing with that conntrack stuff, and it's a terrible failure. Using the vyatta config to set the conntrack sip ports on my router causes one-way audio when a remote phone calls me, but two-way when I call it. Setting the conntrack sip ports on the remote router causes no-way audio. |
| 01:21.12 | Penguin | I deleted that configuration and it's back to two-way audio. |
| 01:21.33 | Penguin | I didn't see any difference when the nf_nat_sip module was loaded or unloaded. |
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| 02:49.41 | bluefirecorp | Sorry, I'll probably setup a irc bouncer tomorrow. |
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| 03:55.23 | gartral | could someone explain this error to me? [Nov 3 22:52:38] WARNING[11313]: chan_sip.c:29322 sip_send_keepalive: sip_send_keepalive to 192.168.1.8:5060 returned -2: Bad file descriptor |
| 03:57.10 | gartral | bluefirecorp: znc took me 10 minutes |
| 04:01.20 | gartral | it would be nice if asterisk would tell me WHAT file decriptor was bad so I have a snowball's chance in hell of fixing it >.< |
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| 04:32.48 | gartral | anyone? |
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| 08:09.42 | davlefou | hi, i use asterisk 1.8, i need to listen some line. What i used? |
| 08:11.47 | kaldemar | davlefou: ChanSpy |
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| 08:24.51 | nir | Hello, i'm trying to bind TLS for port 443 and i'm getting the following error: tcptls.c: Unable to bind SIP TLS server to 0.0.0.0:443: Permission denied |
| 08:25.44 | nir | ofcourse i'v disabled other daemons using that port. and selinux is disabled. |
| 08:26.14 | nir | do you know if there is a limitaion in the appliction for the port range ? |
| 08:27.34 | kaldemar | what user is running asterisk? |
| 08:29.06 | gerritfromsa | TLS over SIP uses port 5061 if you've got transport=tls in sip.conf |
| 08:30.04 | gerritfromsa | Port 443 will most likely hit Apache |
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| 08:31.34 | nir | we wan to use it over port 443 |
| 08:32.12 | nir | we might suspect that ITLS are blocking SIP |
| 08:32.55 | nir | so if it's encrypted and over 443 it will probarly pass their Firewall |
| 08:33.37 | nir | it's already working with port 5061, and i'v disabled port 443 (apache ).. |
| 08:34.09 | nir | kaldemar: user apache |
| 08:34.12 | gerritfromsa | wireshark it and see if the remote sides asterisk responds |
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| 08:35.01 | gerritfromsa | if you execute asterisk -rx "sip show tcp" - what do you get? |
| 08:35.01 | nir | telnet localhost 443 is good indication and also telnet -tupln |
| 08:35.01 | Nugget | telnet is eeeeeeevil! |
| 08:35.27 | nir | telnet -tupln = netstat |
| 08:36.11 | nir | gerritfromsa: empty result |
| 08:39.34 | gerritfromsa | You should be getting something ... |
| 08:40.00 | gerritfromsa | Typical output :Host Port Transport Type |
| 08:40.17 | *** join/#asterisk afournier (~admin@mx1.wisp-e.com) |
| 08:40.23 | gerritfromsa | 222.222.222.222 53483 TLS Server |
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| 08:41.56 | gerritfromsa | natting done? |
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| 08:47.47 | nir | noting, it might be connected to the error im getting on the logs (full) : [Nov 4 02:54:29] ERROR[2603] tcptls.c: Unable to bind SIP TLS server to 0.0.0.0:443: Permission denied |
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| 08:55.10 | gerritfromsa | Try to binfd it with the server ip |
| 08:56.03 | gerritfromsa | run a netstat -a | grep 443 |
| 08:56.43 | nir | same issue |
| 08:57.06 | nir | i'm getting the same error on the logs (full) |
| 08:57.52 | gerritfromsa | what is the result of a #netstat -a | gep 443 |
| 08:58.33 | nir | gep: command not found :) |
| 08:58.43 | nir | but, really |
| 08:58.45 | nir | it's empty |
| 08:59.05 | nir | i'v tried to chown asterisk to root |
| 08:59.40 | gerritfromsa | sorry must be grep |
| 09:00.09 | gerritfromsa | wait a minute , you installed asterisk as non-root? |
| 09:00.33 | kaldemar | i asked for the user over 30 minutes ago... |
| 09:00.47 | gerritfromsa | sorry was not here |
| 09:00.47 | kaldemar | 443 is a privileged port. |
| 09:01.07 | kaldemar | you were. :) |
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| 09:01.27 | gerritfromsa | just not behind the screen |
| 09:02.58 | nir | yea, compiled asterisk as root |
| 09:03.21 | nir | and asterisk user/group is apache:apache |
| 09:03.30 | nir | chowned it to root:root |
| 09:04.01 | nir | also i'v ran a simple perl scrip to listen to port 443 and it's working fine .. |
| 09:04.35 | gerritfromsa | nir , you cant just chown asterisk , there is A LOT of files that is used by asterisk |
| 09:05.08 | gerritfromsa | why add apache as uder/group anyway ? |
| 09:05.53 | gerritfromsa | nir , I would seriously consider re-compiling asterisk as root to see if this is nor resolved that way |
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| 09:07.32 | kaldemar | nir: chown does not define what user is running asterisk, it just sets file ownership. ps aux | grep asterisk |
| 09:09.05 | nir | kaldemar: seems you right |
| 09:09.10 | nir | it's still with apache user |
| 09:11.07 | nir | it's fixed |
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| 09:11.49 | linocisco | hi all |
| 09:11.55 | nir | changed : runuser and rungroup on asterisk.conf |
| 09:12.26 | nir | from apache to root . |
| 09:12.36 | nir | thanks kaldemar and gerritfromsa :D |
| 09:12.55 | linocisco | http://my.digium.com/en/products/ivr/audio-converter/ does not work for my conversion of mp3 file |
| 09:13.18 | gerritfromsa | nir , glad you're sorted |
| 09:14.57 | nir | linocisco: in which format the file you are trying to upload ? |
| 09:15.26 | linocisco | nir, mp3 |
| 09:17.03 | *** join/#asterisk crazed1 (4c5a1503@gateway/web/freenode/ip.76.90.21.3) |
| 09:17.24 | nir | i never tried to use that tool, but most of the chances that there is a problem with your MP3 file. are you getting any kind of message telling you what is the reason for the failure ? |
| 09:17.40 | crazed1 | hey i've got an install that i copied over and now i can't connect. i enabled sip set debug, and i see that its return "401 Unauthorized" but the user is definitely in sip_additional.conf |
| 09:18.02 | nir | you can use a cmd tool like: mpg123 -w output.wav input.mp3 |
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| 09:19.07 | nir | crazed1: asterisk -rx "sip show users" | grep <YOUR USERNAME> |
| 09:19.18 | nir | check that the username and passwords are matched |
| 09:19.33 | gerritfromsa | crazed1, is the IP's the same , maybe it's your permit= statement ? |
| 09:20.12 | crazed1 | the ip is different, i was looking for something like that but thought i found them all |
| 09:20.24 | crazed1 | nir: i see my password along with YES YES |
| 09:21.18 | crazed1 | all of the extensions are deny:0.0.0.0 permit=0.0.0.0 |
| 09:23.43 | crazed1 | what's the website i post data to? |
| 09:24.16 | gerritfromsa | pastebin.com |
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| 09:26.03 | crazed1 | pastebin.com/3YvXeu6b |
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| 09:31.14 | gerritfromsa | crazed1, this is NORMAL for asterisk to challenge the phone on the 1st register |
| 09:31.51 | gerritfromsa | crazed1, it ask fo the md5checksum , the sip client must then respond with a 2nd reg but this time with the md5sum |
| 09:32.12 | gerritfromsa | onlt then will you get a successful reg |
| 09:32.20 | gerritfromsa | where is the rest of the trace ? |
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| 09:33.48 | eloycoto | HI, Can I use dynamic features with patterns? Any idea about it? |
| 09:33.53 | gerritfromsa | O'm not too familiar with Jitsi , is there different authentication options ? md5/rsa ... |
| 09:34.14 | eloycoto | My idea is use pattern like *XXXX* into the features map |
| 09:34.26 | eloycoto | do you know if it possible make this? |
| 09:34.28 | gerritfromsa | eloycoto, elaborate on dynamic features |
| 09:35.10 | eloycoto | yes, but dynamic features doesn't support patterns |
| 09:35.56 | gerritfromsa | eloycoto, what is it that you want to match ? |
| 09:36.29 | eloycoto | Now I'm matching like ** with my custom Macro and works ok |
| 09:36.46 | eloycoto | but I need to match *\d{4}* |
| 09:37.16 | gerritfromsa | Alpha numeric ? |
| 09:37.50 | eloycoto | only numeric |
| 09:38.32 | gerritfromsa | and the wildcards ? |
| 09:38.57 | eloycoto | yes |
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| 09:41.30 | crazed1 | pastebin.com/u0f2XGKv gerritfromsa |
| 09:43.49 | gerritfromsa | If you use Set(ARG1) before this feature , you can call it with the AppArguments in features , about the only way I know |
| 09:44.16 | crazed1 | gerritfromsa, i don't have any trouble connecting using jitsi to another pbx that is an exact copy |
| 09:44.35 | gerritfromsa | crazed1, wireshark it |
| 09:44.50 | crazed1 | this is nuts i need sleep lol |
| 09:44.57 | crazed1 | ugh i also need this fixefd -.- |
| 09:45.15 | crazed1 | isn't there any stupid thing i can check first |
| 09:45.44 | gerritfromsa | crazed1, add insecure=peer,invite and try |
| 09:46.29 | eloycoto | app => **,caller,SET,ARG1 |
| 09:46.34 | eloycoto | like this? |
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| 09:49.09 | gerritfromsa | eloycoto, No you'll have to set it before Dial() then pass the argument to the AppArgument parameter in features.conf |
| 09:49.48 | gerritfromsa | <FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,<AppArguments>[,MOH_Class]] |
| 09:50.57 | linocisco | nir, i m not on linux. i m on windows 7 |
| 09:50.59 | eloycoto | yes, but my agent was on call. and she press *111* to send alert to extension *111*, but maybe is 111 or 222 |
| 09:51.01 | gerritfromsa | What dies the 4 digits represent ? |
| 09:51.06 | eloycoto | I can't setup before the dial |
| 09:51.22 | gerritfromsa | create 2 different features then |
| 09:51.32 | eloycoto | random numbers |
| 09:51.40 | gerritfromsa | problem |
| 09:51.55 | eloycoto | I know :-) |
| 09:52.23 | eloycoto | and If I use option M of dial? And read the dtmf from the MACRO? |
| 09:52.34 | gerritfromsa | wait , you can have the agent transfer a call and that way send the alert and have the call return to the agent with confirmation |
| 09:52.40 | gerritfromsa | even better |
| 09:53.16 | gerritfromsa | so instead of *111* you dial #111 |
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| 09:53.56 | gerritfromsa | or #*111 and then in the dialplan you can use the Pattern _*XXX,1,App() |
| 09:54.50 | crazed1 | might there be another place i have to change the ip? |
| 09:55.24 | gerritfromsa | under [general] maybe |
| 09:55.51 | eloycoto | umm |
| 09:55.57 | eloycoto | yes, is a valid way |
| 09:56.25 | gerritfromsa | You do you send the alert ? |
| 09:56.37 | eloycoto | can you show me any piece of code for this transfer? |
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| 10:03.17 | crazed1 | hmm looks like i'm not getting the return message |
| 10:05.37 | crazed1 | i just tried it from local network and its still getting rejected |
| 10:07.23 | gerritfromsa | exten => _*XXX,1,Set(CALLERID(name)=ALERT ${EXTEN:1)) ; _*XXX,2,Dial(SIP/${EXTEN:1},,r) ; _*XXX,3,Transfer(SIP/${CALLERID(num)}) |
| 10:07.59 | eloycoto | thanks, I'm testing |
| 10:07.59 | eloycoto | :-) |
| 10:08.20 | gerritfromsa | So the CALLED party will have ALERT 111 displayed in the phone when dialling 111 from ext 100 , but when it done it will Transfer BACK too 100 |
| 10:08.58 | gerritfromsa | maybe change _*XXX,2,Dial(SIP/${EXTEN:1},,r to _*XXX,2,Dial(SIP/${EXTEN:1},,1 - will only dial for 1 sec then return |
| 10:09.14 | gerritfromsa | nope that must be _*XXX,2,Dial(SIP/${EXTEN:1},1,r |
| 10:09.54 | gerritfromsa | quickly going to test it myself ... |
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| 10:16.30 | gerritfromsa | back |
| 10:17.13 | gerritfromsa | that dialplan needs a bit of work |
| 10:17.55 | gerritfromsa | it transferring to the wrong agent |
| 10:18.04 | eloycoto | I know |
| 10:19.10 | gerritfromsa | must be CHANNEL but cut out ... |
| 10:23.17 | gerritfromsa | got it |
| 10:23.26 | gerritfromsa | will publish soon |
| 10:33.21 | gerritfromsa | eloycoto, pastebin.com/EL1yUP126 |
| 10:33.59 | eloycoto | sorry mate, this pastebin doesn't work |
| 10:34.21 | gerritfromsa | forgot the captha try again |
| 10:34.43 | gerritfromsa | pastebin.com/EL1yUP26 |
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| 10:51.44 | gerritfromsa | eloycoto, does this work for you? |
| 10:51.59 | eloycoto | I'm testing |
| 10:52.05 | eloycoto | but make sense |
| 10:52.06 | eloycoto | :-) |
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| 10:55.52 | AikiLinux | Morning |
| 10:56.48 | AikiLinux | how can i trigger SIPp to make a single sip call ? |
| 11:00.18 | eloycoto | https://github.com/mojolingo/sippy_cup |
| 11:00.33 | eloycoto | sippy_cup is a wrapper for sipp |
| 11:00.35 | eloycoto | is more easy |
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| 11:05.12 | AikiLinux | i'll try it,thanks |
| 11:05.51 | gerritfromsa | anyone knows what the limitations is when writing Monitor() files to NAS/SAN ? |
| 11:06.05 | AikiLinux | but what i realy need is a sip client i can call from a CLI ( or more to the point trigger it as a nagios notification function) |
| 11:06.59 | gerritfromsa | AikiLinux, you can call from the CLI> using console dial |
| 11:07.25 | gerritfromsa | or use call files ... |
| 11:16.12 | gerritfromsa | eloycoto, thanks for the sippy_cup link ! |
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| 12:17.49 | m1tO | hi there! |
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| 12:26.30 | phix | hai |
| 12:27.35 | gerritfromsa | eloycoto, still testing my brother ? |
| 12:27.52 | eloycoto | yes and no |
| 12:28.01 | eloycoto | it works as expected |
| 12:28.22 | eloycoto | but the problem is when the user send #1234 it's ok |
| 12:28.31 | eloycoto | but the caller has moh |
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| 13:04.34 | fprior | Hi all. I'm searching a way to check if a call from a NPA-NXX number to another NPA-NXX is a local call or not, as does this page http://goo.gl/vAw4OY (It show Y or N if call is local); is there any way to execute this check in Asterisk (dialplan, database) ? |
| 13:08.06 | *** join/#asterisk george_sand (~georgesan@194.6.230.193) |
| 13:08.13 | george_sand | hyl |
| 13:08.16 | george_sand | hello |
| 13:09.31 | gerritfromsa | fprior, use the ${CHANNEL} variable to check |
| 13:10.22 | *** join/#asterisk geeksteve (~geeksteve@emh-nat.poundbury.com) |
| 13:11.15 | kaldemar | fprior: asterisk itself does not have such information. would you want to use that service on every call or do you have some other means to access the locality information? |
| 13:11.15 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 13:11.17 | george_sand | can some help me with low success rate for receiving fax ? when i send fax i don't have any issue. |
| 13:11.56 | gerritfromsa | george_sand, i can try help if it's Hylafax you're using |
| 13:12.41 | *** join/#asterisk acidfu (~nib@unaffiliated/acidmen) |
| 13:13.03 | george_sand | not quite , i build a platform using java as fax controler and asterisk as a media gateway for fax |
| 13:13.40 | fprior | gerritfromsa, can you give me more details ? |
| 13:13.54 | gerritfromsa | most common problem on receiving is usually buad rate and ECR settings |
| 13:14.17 | gerritfromsa | fprior, let's say before you use the Dial() |
| 13:14.35 | gerritfromsa | you 1st interrogate the CHANNEL variable to see if its local or not |
| 13:14.53 | gerritfromsa | and then you follow with a condition statement like GotoIf() |
| 13:15.14 | george_sand | gerritfromsa you mean ECM ? |
| 13:15.36 | fprior | kaldemar, do you know if exists another resource like that page ? |
| 13:15.39 | gerritfromsa | sorry yes Error Correction Mode |
| 13:16.24 | george_sand | thanks for the info i will rechekc there to see what to do |
| 13:16.57 | fprior | gerritfromsa, I need to check if call will be local or not before execute DIAL command, only on NPA-NXX basis. If local I'll Dial on PRI, if not, I'll dial on SIP Trunk |
| 13:17.39 | gerritfromsa | fprior, you'll do this simply with a Pattern_match |
| 13:17.50 | kaldemar | fprior: no. |
| 13:18.19 | gerritfromsa | fprior, what is your local prefix and what is your international/national prefix |
| 13:19.00 | kaldemar | gerritfromsa: in that case "local area" has multiple prefixes. |
| 13:19.24 | gerritfromsa | no problem - multiple pattern matching then |
| 13:19.34 | fprior | yes, my customer has multiple prefixes |
| 13:20.21 | *** join/#asterisk porkchop85 (~porkchop@50.200.28.42) |
| 13:20.31 | fprior | first idea was create a databases with multiple combination of local prefixes; I would ask here if someone has solved the issue in a different way |
| 13:21.10 | fprior | I'm from Argentina and I need to learn more about this topic, to implement better solution |
| 13:21.10 | gerritfromsa | If you want to store it in a Database , no problem either |
| 13:21.53 | gerritfromsa | so in sql you'll have a table with Pattern and Trunk .... |
| 13:22.01 | fprior | gerritfromsa, localcallingguide.com doesn't offer to download database; do you know another resource ? |
| 13:22.12 | gerritfromsa | but it would be a lot easier in the dialplan |
| 13:22.30 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:22.47 | gerritfromsa | Try your local Telecoms company's website |
| 13:23.09 | gerritfromsa | does Argentina Telecom exist ? or something like that ? |
| 13:23.45 | gerritfromsa | The Telecom Argentina S.A website ... |
| 13:24.19 | gerritfromsa | Thats what worked for me here in my country |
| 13:24.30 | fprior | yes, but I need to implement that check with U.S. Telecom |
| 13:25.13 | gerritfromsa | For each state/city in the US? |
| 13:25.21 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:25.21 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:25.45 | fprior | no, for few cities only |
| 13:26.23 | gerritfromsa | I stumbled on to this website : www.telcodata.us that's got it , but there is a fee |
| 13:27.40 | gerritfromsa | $15 to become a "member" |
| 13:28.23 | gerritfromsa | Give it conveniently is csv format - ready to be uploaded in SQL |
| 13:28.52 | fprior | someone can explain me what mean LATA ? |
| 13:29.40 | gerritfromsa | wait maybe I can help |
| 13:29.49 | gerritfromsa | with the US cities I mean ... |
| 13:30.14 | fprior | could be possible that all calls between NPA-NXX in same LATA are local ? |
| 13:30.57 | gerritfromsa | what? |
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| 13:32.46 | fprior | no, ignore it; in http://goo.gl/Od3rdI I tested two NPA-NXX from same LATA but they are NOT local |
| 13:32.46 | *** join/#asterisk dronacid (~dronacid@vpn.openroot.de) |
| 13:35.28 | gerritfromsa | fprior, there you go http://pastebin.com/fqHtw3Kb |
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| 13:44.03 | fprior | gerritfromsa, thanks |
| 13:44.10 | [TK]D-Fender | Except that has no city-level |
| 13:53.48 | fprior | Interesting document to understand more about rates http://goo.gl/oZoaIs |
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| 14:15.28 | *** join/#asterisk serafie (~erin@nat/digium/x-awbrvwtgcxrhdafk) |
| 14:17.51 | fprior | I found a solution to determine local calls by NPA-NXX at thie page: http://goo.gl/b87sxB Thanks to all for help |
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| 14:59.12 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:59.14 | sprink | Any suggestions how I can troubleshoot a polycom phone, asterisk pbx, the phone is in a ring-all call queue, but sporadically the phones only rings once? |
| 14:59.30 | sprink | While the rest of the phones continue to ring |
| 14:59.31 | *** join/#asterisk TSM (~the_softw@fw-lon1.wenn.com) |
| 15:00.42 | [TK]D-Fender | Show us the call |
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| 15:02.02 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 15:02.19 | jmetro | do you have ringbusy yes/no |
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| 15:03.07 | sprink | [TK]D-Fender: Should I capture all the traffic to the phone and post it here? |
| 15:03.45 | [TK]D-Fender | sprink: not just the traffic, full CLI so we can see what * is deciding |
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| 15:42.00 | gerritfromsa | is back (gone 00:23:37) |
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| 15:54.39 | gerritfromsa | is away: Taking a walk outside ... |
| 15:54.57 | Greenlight | What is this, Twitter? :) |
| 15:55.19 | jmetro | gerritfromsa: While youre out there, let me know what the fox says. |
| 15:57.16 | ipengineer | Hello everyone! Is anyone aware of where the new cdr schema is for Asterisk 12? The schema in the docs say it was deprecated in 1.8. https://wiki.asterisk.org/wiki/display/AST/MySQL+CDR+Backend |
| 16:00.21 | kaldemar | ipengineer: where does it say so? |
| 16:01.39 | gerritfromsa | is back (gone 00:06:59) |
| 16:02.02 | ipengineer | kaldemar In red at the top.. Well it says "cdr_mysql module below has been deprecated" but I see no reference to that below outside of the MYSQL CDR module. The other thing that led me to believe that was that I had read that there are some new fields being used now. |
| 16:02.21 | Greenlight | You want to use the odbc one now, |
| 16:04.52 | gerritfromsa | *12 cdr spec : https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification |
| 16:05.41 | ipengineer | gerritfromsa: Thanks! That is what I needed. |
| 16:05.52 | kaldemar | ipengineer: it says the native MySQL CDR module was deprecated, but nothing about the schema. |
| 16:06.44 | Greenlight | The scehme is very flexible with odbc; a lot of columns are optional and you can add custom columns |
| 16:07.13 | gerritfromsa | Maybe it will be an add-on like before ... |
| 16:08.19 | jmetro | gerritfromsa: did you figure out what the fox says? |
| 16:09.03 | gerritfromsa | jmetro, no it remains a mystery ... |
| 16:10.47 | jmetro | amazing. Very mystery. such fox. |
| 16:11.02 | Greenlight | The quick brown fox ? |
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| 16:22.22 | gerritfromsa | is away: Taking a walk outside ... |
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| 16:26.52 | ipengineer | kaldemar: Yea, I see that now that you mention it but not when I read it originally.. |
| 16:27.01 | Alex_h | how, im really stumped trying to find some documentation that explains the output of the cli command "pri show spans", iv checked the wiki but cant seem to see anything that explains the different statuses, iv checked tzafir's dahdi documentation, and iv had a look through the sample config files in the build directory, but cant seem to find an explanation of them. Can anyone point me to a resource that explains this command? |
| 16:27.26 | jmetro | points at d-fender |
| 16:30.31 | m1tO | hi there!!! |
| 16:31.38 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
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| 16:49.56 | anonymouz666 | how easy is to configure T.38 gateway... added two lines in the dialplan, one in sip.conf and first FAX over WAN just worked. |
| 16:49.57 | anonymouz666 | nice. |
| 16:51.36 | coppice | anonymouz666: shame isn't it? what happened to all the skill that was needed in days of yore? |
| 16:54.13 | anonymouz666 | coppice: this is the only advantage of being late on a particular topic |
| 16:59.21 | runfromnowhere | Mixmonitor, using different files for recording send and recieve audio, those files are of unequal length. Is there any logic to how/why this happens so I can correct it in the postprocessing? |
| 17:00.42 | Greenlight | runfromnowhere: One side wasn't producing audio at some point |
| 17:01.25 | runfromnowhere | Right - the problem is that I need to mix the two files together and if that mix is off by a second or two the call quickly becomes nonsense to listen to |
| 17:02.05 | runfromnowhere | So if those omissions only occur at the beginning of a call, I can make sure to line them up so they both end at the same point |
| 17:02.17 | runfromnowhere | I would imagine that it couldn't occur at the end? |
| 17:02.28 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 17:02.53 | runfromnowhere | Or do I need to get back into the mixmonitor source again and ensure that I'm recording both files even if one side isn't producing audio? :( |
| 17:02.57 | Greenlight | This used to happen with Monitor a lot, I've not seen it too much with mixMonitor |
| 17:03.11 | runfromnowhere | I'm seeing mixmonitor output files with unequal lengths |
| 17:03.15 | Greenlight | Best bet is to listen to the calls and find *where* they go out of sync |
| 17:03.20 | runfromnowhere | Not huge differences, maybe 0.5-1s |
| 17:03.48 | Greenlight | Remember as well; MixMonitor uses audiohooks, so you can safely attatch multiple MixMOnitors to a single call. |
| 17:04.03 | Greenlight | Eg, you could have on, capturing to a single file. and another capturing the in/out separately. |
| 17:04.23 | Greenlight | But prior to going down that road, I'd recommend invesitgating where they get out of sync and why |
| 17:05.56 | runfromnowhere | Yeah - I can actually capture all three at once |
| 17:06.08 | *** part/#asterisk sprink (~sprink@you.cant.haxit.org) |
| 17:06.08 | runfromnowhere | I'm just not sure how to determine where they go out of sync |
| 17:08.07 | runfromnowhere | OK I'm learning a little bit - it's always the "out" side that's shorter |
| 17:09.54 | Greenlight | Perhaps you get early media on the "in" side ? |
| 17:12.37 | runfromnowhere | I think that's the case |
| 17:12.47 | runfromnowhere | It's just a matter of alignment.... |
| 17:13.05 | Greenlight | Does MixMonitor take a "b" option, like ChanSpy does ? |
| 17:13.25 | Greenlight | Yes, it does. |
| 17:13.27 | Greenlight | Try that |
| 17:13.47 | Greenlight | https://wiki.asterisk.org/wiki/display/AST/Application_MixMonitor |
| 17:13.53 | Greenlight | b - Only save audio to the file while the channel is bridged. |
| 17:14.01 | runfromnowhere | What does "while the channel is bridged" imply? |
| 17:15.45 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 17:15.59 | Greenlight | Honestly, I'm not entirely sure if that's after the remote end has Asnwered or not, internally early media does bridge |
| 17:16.03 | Greenlight | SO, try and see :) |
| 17:16.05 | runfromnowhere | Heh |
| 17:16.21 | runfromnowhere | Well I can't quite do that now... |
| 17:16.24 | runfromnowhere | Production data :( |
| 17:16.37 | gartral | could someone explain this error to me? [Nov 3 22:52:38] WARNING[11313]: chan_sip.c:29322 sip_send_keepalive: sip_send_keepalive to 192.168.1.8:5060 returned -2: Bad file descriptor |
| 17:17.31 | Greenlight | gartral: Sounds like something serious has gone wrong... What version are you running ? |
| 17:17.47 | gartral | Greenlight: 11.5 |
| 17:18.00 | gartral | Greenlight: hang on, i'll get the exact version |
| 17:18.56 | gartral | Greenlight: 11.5.1 |
| 17:20.21 | Greenlight | What causes the issue ? |
| 17:20.36 | Greenlight | And do you have plenty spare descriptors ? |
| 17:21.08 | gartral | Greenlight: cause is unknown, and I should, hold on |
| 17:21.23 | Greenlight | As in, what's happening when the issue occurs? |
| 17:23.33 | gartral | Greenlight: again, unknown, it's just throwing that error.. over.. and over again, no other output |
| 17:24.12 | gartral | Greenlight: it seems to happen only when a client is trying to connect |
| 17:24.28 | *** join/#asterisk heffer (felix@fedora/heffer) |
| 17:24.31 | Greenlight | Sounds like a bug of some sort. |
| 17:25.32 | Greenlight | Hopefully someone more familiar with the codebase will cast their eyes over it, and it may mean more to the, |
| 17:26.43 | gartral | Greenlight: ok, I know this system has 393916 FDs.. how do I tell how many are used? >.> |
| 17:28.21 | *** join/#asterisk jploh (~textual@49.144.120.20) |
| 17:28.26 | gartral | nevermind, I'm only using about 3000 of them |
| 17:30.54 | gartral | Greenlight: this all started i think when I activated the keepalive ocnfig in sip.conf |
| 17:32.56 | runfromnowhere | Greenlight: It looks like that might well have fixed it |
| 17:35.17 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 17:39.41 | gartral | AAHAH! it was a mangled comment in my sip.conf! grrr! |
| 17:40.59 | *** join/#asterisk andyoutside (cec00d96@gateway/web/freenode/ip.206.192.13.150) |
| 17:41.09 | *** join/#asterisk ipso (~ipso@184.68.39.238) |
| 17:42.38 | ipso | I recently upgraded from Asterisk to v1.8 and now my queues don't work... All the static queue members (ie: SIP/remoteserver/700) are marked as INVALID and can't take calls. Any ideas how to correct this? |
| 17:43.24 | Greenlight | You need to be more verbose with your defenitons and specify a device |
| 17:43.34 | Greenlight | Umm two secs I'll grab an example |
| 17:43.40 | [TK]D-Fender | ipso: "sip show peer remoteserver", "queue show", and your queues.conf please... |
| 17:43.41 | [TK]D-Fender | ~pb |
| 17:43.42 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:43.43 | [TK]D-Fender | ^^^^ |
| 17:45.07 | Greenlight | member => SIP/sip_callmanager/30502,0,30502,SIP/sip_callmanager,yes |
| 17:45.12 | Greenlight | Like that |
| 17:45.19 | andyoutside | ok changing over from cdr_mysql to cdr_adaptive_odbc I am using the same table. I am able to get cel_odbc working on a different table. So I know the connection is good. What should I be looking at to see why records are not being put in?\ |
| 17:45.34 | Greenlight | andrewyager: The debug log... |
| 17:45.54 | ipso | [TK]D-Fender, http://www.pastebin.ca/2473709 |
| 17:46.16 | Greenlight | ipso: DO what I say will fix it. |
| 17:47.09 | ipso | Greenlight, okay, I will try to lookup what all the arguments are for. |
| 17:47.26 | Greenlight | syntax: member => interface,[,penalty][,membername][,state_interface][,ringinuse] |
| 17:47.35 | *** join/#asterisk d45h (~d45h@unaffiliated/dash-/x-7576607) |
| 17:47.38 | Greenlight | It's the state_interface that's important when using this syntax. |
| 17:51.11 | ipso | Greenlight, hrmm, doesn't seem to help: member => SIP/office/790,0,790,SIP/office,yes |
| 17:51.44 | Greenlight | Isn't it "mycompany" from your PB ? |
| 17:52.07 | ipso | Greenlight, sorry, I wasn't consistent in my replacements, but yes, its the same. |
| 17:52.47 | Greenlight | Ok. Lets see a "queue show" again. And which memeber did you change ? |
| 17:52.57 | *** join/#asterisk salzh (18b0c186@gateway/web/freenode/ip.24.176.193.134) |
| 17:53.01 | ipso | Greenlight, when I just do this: exten => _7901,1,Dial(SIP/mycompany/790,20,r) it rings and works fine. |
| 17:53.09 | Greenlight | Yes |
| 17:53.16 | Greenlight | Your alternative is to use a Local channel |
| 17:53.25 | *** join/#asterisk jpoz (~jpoz@107-1-105-37-ip-static.hfc.comcastbusiness.net) |
| 17:53.56 | Greenlight | But that shouldn't be neccissary, as I've had it working with the syntax I gave |
| 17:54.06 | ipso | Greenlight, For the global_queue it shows: 790 (SIP/mycompany/790) (Invalid) has taken no calls yet |
| 17:54.44 | Greenlight | Doesn't seem to have picked up the config change |
| 17:54.49 | Greenlight | Did you "queue reload all" ? |
| 17:55.34 | ipso | Greenlight, I tried a full restart, before it was showing: SIP/mycompany/790 (dynamic) (Invalid) has taken no calls yet |
| 17:55.49 | ipso | Now its showing: 790 (SIP/mycompany/790) (Invalid) has taken no calls yet |
| 17:55.59 | ipso | So it has picked up some change anyways... |
| 17:56.25 | *** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 17:56.31 | Greenlight | Yes, sorry, didn't see |
| 17:57.01 | Greenlight | "sip show peers" |
| 17:57.14 | *** join/#asterisk hugobulnes (~Thunderbi@189.215.47.77.cable.dyn.cableonline.com.mx) |
| 17:57.27 | [TK]D-Fender | relod the module |
| 17:57.30 | [TK]D-Fender | reload* |
| 17:57.30 | ipso | mycompany/mycompany 66.183.160.211 D N 5060 OK (87 ms) |
| 17:57.38 | [TK]D-Fender | perhaps channel-loading didn't occur first |
| 17:58.22 | ipso | [TK]D-Fender, thats what I tried first from Googling this problem... Didn't seem to help though, even after adjusting my preload orders |
| 17:58.40 | ipso | [TK]D-Fender, just did: reload app_queue.so and it didn't make any difference. |
| 17:58.47 | Greenlight | I had this *exact* same issue before adding the state_interface params |
| 17:59.04 | Greenlight | You might be better just using Local channels then... |
| 17:59.21 | ipso | Greenlight, seems odd that it was working prior to the upgrade... |
| 17:59.38 | Greenlight | ipso: I never attempted it in 1.8 so can't compare on that front |
| 18:00.33 | ipso | Greenlight, I did attempt the local workaround too, never seemed to work... I guess I will try it again |
| 18:00.47 | Greenlight | Lets see your queues.conf quickly first |
| 18:01.37 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 18:01.54 | andyoutside | Greenlight: when you say debug log. I assume looking ate the console and setting the debug on cdr_adaptive_odbc.c to 7 hsould work. I do not see any thing related to cdr |
| 18:02.29 | ipso | Greenlight, http://www.pastebin.ca/2473712 |
| 18:02.55 | ipso | Greenlight, I just changed one line to start switching to the Local method... |
| 18:03.26 | runfromnowhere | Oh dear god |
| 18:03.38 | runfromnowhere | The Queue application has its own Mixmonitor invocation? |
| 18:04.47 | andyoutside | the queues would not be setup on the other server would they? |
| 18:05.28 | gartral | the more I read into confs for asterisk, the more I feel like I'm dealing with an operating system held together with duct-tape and pixie-dust >.< |
| 18:07.39 | Greenlight | ipso: Can't see anything wrong with the entry in that queues.conf. It worked for me ;/ |
| 18:08.05 | ipso | Greenlight, it may be that I'm not defining the local extensions... |
| 18:08.51 | Greenlight | ipso: Yes, the local way will certainly allow you to do what you need. |
| 18:09.01 | Greenlight | Do you have the extensions defined in the context "mainmenu" ? |
| 18:09.06 | ipso | Greenlight, yes |
| 18:10.01 | ipso | Greenlight, mainmenu context has: '_790' => 1. Background(pls-wait-connect-call) [pbx_config] |
| 18:12.51 | ipso | Greenlight, grr... The member entry in the queue using Local works fine from the dialplan itself, yet it still shows invalid... |
| 18:14.00 | Greenlight | Try disabling callcounter |
| 18:14.29 | [TK]D-Fender | "invalid" typically is a syntax issue with either the call device, or the state device, possibly due to a module load order issue |
| 18:14.39 | ipso | Greenlight, hrmm, I just cut it back to the basic and it seemed to work: member => Local/790@mainmenu |
| 18:14.43 | runfromnowhere | If I'm using "Set" to set a variable, and that variable has parenthesis in it, does a quoted string work? |
| 18:14.50 | [TK]D-Fender | it should not actually poll the remote server directly, but rather rely on the local stat as usual |
| 18:14.58 | ipso | Greenlight, when I try to use: member => Local/790@mainmenu,0,790,Local/790@mainmenu,yes it doesn't |
| 18:15.06 | runfromnowhere | i.e. Set(OPTION="r(file.wav)") |
| 18:15.29 | Greenlight | Just use Local/790@mainmenu |
| 18:15.31 | Greenlight | Leave the rest off |
| 18:16.16 | Greenlight | That state_interface bit was only if you were using the SIP/server/exten syntax |
| 18:18.11 | [TK]D-Fender | That shouldn't even matter |
| 18:18.22 | [TK]D-Fender | <PROTECTED> |
| 18:18.52 | [TK]D-Fender | state is a DEVICE state which is tied to * counting channels against it |
| 18:19.00 | [TK]D-Fender | You Do have those values set ... right? |
| 18:20.16 | Greenlight | He's setting callcounter from queues.conf |
| 18:20.29 | Greenlight | Although I wonder if that's somehow not being set right on the devices |
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| 18:26.24 | ipso | [TK]D-Fender, my modules.conf has this near the top: preload = pbx_config.so preload = chan_sip.so preload = chan_local.so preload = app_queue.so |
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| 18:29.10 | andyoutside | Greenlight: I do not see anything from cdr_adaptive_odbc.c in the debug log |
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| 18:44.09 | hugobulnes | . |
| 18:53.24 | btcquant | Hello - is there any easy way to setup RetryDial in Freepbx? |
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| 18:57.41 | WIMPy | Try to ask in #freepbx |
| 19:00.30 | runfromnowhere | Well, in case anyone's interested, it seems there are some barely documented variables that can be set to make the queue record in a different or more exciting way |
| 19:02.39 | paulc | describe a more exciting way of recording.. does it have overlaid sound effects and audience applause etc? cos THAT would be cool.. |
| 19:04.24 | [TK]D-Fender | The voices add that for me already.... |
| 19:04.39 | runfromnowhere | Hahah I mean you can actually set MixMonitor options and ending exec |
| 19:04.50 | runfromnowhere | As opposed to just getting standard mixmonitor output |
| 19:05.06 | runfromnowhere | I find that exciting, but that might just be because I've spent WAY to long working on getting stereo recording working :) |
| 19:07.13 | paulc | nods - fair play. We did a thing with a pseudo ACD type thing (but not using Queue), where we wanted agent audio in one ear and caller audio in the other.. so we could tell who was talking over who.. or moaning.. as the case was.. |
| 19:07.30 | runfromnowhere | Hahah interesting line of work |
| 19:08.30 | paulc | hehe.. yes.. "We use Asterisk to connect callers paying a few bucks a minute to 'agents' who get paid cents per minute to talk.. uh.. 'racily' or however the paying caller wants them to talk" |
| 19:10.55 | andyoutside | I am try to use cdr_adaptive_odbc I can see the config laoding without errors. With debug on I do not see any lines from cdr_adaptive_odbc.c. What should I be looking into to have cdr putinto the db? |
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| 19:19.21 | gerritfromsa | is back (gone 02:56:58) |
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| 20:14.58 | andyoutside | ok after some digging the records are being put in just not calldate has the name changed? |
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| 20:48.37 | anonymouz666 | it is possible to also distribute PRESENCE STATE? |
| 20:48.44 | anonymouz666 | device state is OK |
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| 21:38.54 | anonymouz666 | sruffell: do you already tested modem v.90 with wct24xxp driver? |
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| 21:40.03 | sruffell | Nothing specifically. Normally that is covered by fax testing. |
| 21:41.42 | anonymouz666 | we have a lots of improvements in chan_dahdi about FAX, but I think the modulation is different from v.90 and FAX stuff. |
| 21:42.29 | sruffell | The driver typically doesn't care at that level. The only catch is to make sure there aren't overruns / underruns. |
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| 21:49.21 | raub | I am getting a ton of messages like this: [Nov 4 02:25:30] NOTICE[1872] chan_sip.c: Failed to authenticate device 202<sip:202@128.227.101.87>;tag=cc1bfa76 |
| 21:49.33 | raub | thing is we don't use that extension |
| 21:49.46 | jblack | one of your devices thinks it's 202. |
| 21:50.13 | jblack | well, something at 128.227.101.87 is trying to contact youur * server |
| 21:52.07 | raub | That is the server's ip |
| 21:52.40 | raub | (or at least represents it for this discusssion) |
| 21:52.52 | drmessano | or its a SIP attack bot thingo |
| 21:52.55 | drmessano | and can be ignored |
| 21:53.36 | drmessano | Spoofed IP.. Buddy of mine gets those all the time |
| 21:55.08 | raub | Any way to find out the real ip? |
| 21:55.44 | gavit | anyone know how I get to the asterisk command line interface? |
| 21:57.19 | wdoekes | gavit: asterisk -r |
| 21:57.54 | gavit | wdoekes: thanks, however sip reload doesn't do a thing :S |
| 21:58.59 | wdoekes | gavit: given too little verbosity, you won't see that it does a thing |
| 21:59.11 | wdoekes | core set verbose 5 |
| 21:59.12 | drmessano | core set verbose 10 |
| 21:59.15 | drmessano | or something |
| 21:59.24 | drmessano | 5, 7, 999 |
| 21:59.45 | gavit | wdoekes: thanks, worth mentioning that in the asterisk admin guide :) |
| 21:59.46 | wdoekes | (haha, I use 20, but I didn't want to get into that discussion) |
| 22:01.33 | drmessano | verbose and -v are both explained in any Asterisk admin documentation |
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| 22:05.43 | gavit | drmessano: I guess it will be, but I'm @ page 45, and no mention of it yet, but they give me output which doesn't match |
| 22:07.28 | drmessano | Page 104 |
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| 22:18.10 | gavit | drmessano: I believe its there, I am just stating they should mention it sooner or at leats notify that it should be entered (no questions asked) |
| 22:18.32 | gavit | when running sip show peers I see as port 0 rather than 5060 |
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| 22:46.51 | wdoekes | gavit: peers that haven't registered yet |
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