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| 02:38.17 | shealyw2 | hi guys, I am just now looking into pbx and am looking for a system that I can connect my twilio sip to, that will run on a windows computer and use usb phones and finally allow me to transfer calls to another number. |
| 02:39.01 | shealyw2 | I don't want to ask "is asterisk capable of this", more of am i going in the right direction looking at asterisk? |
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| 08:07.28 | stevenm | Lo, anyone know of some simple utility I can use to query a broadsoft-based platform with Xsi ? |
| 08:08.09 | stevenm | got a SIP client here that talks "Xsi" to a server to get it's sip and xmpp credentials as well as probably doing other crap like setting call forwards and such |
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| 08:19.12 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.12.0 (2014/08/19), 1.8.30.0 (2014/08/19); Standard: Asterisk 12.5.0 (2014/08/19); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 09:10.05 | lorenzo | hi! I've just finished writing my first asterisk macro |
| 09:10.09 | lorenzo | http://pastebin.com/z0rURKmF |
| 09:10.13 | lorenzo | does it look acceptable? |
| 09:10.22 | lorenzo | any stuff I'm missing maybe? |
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| 09:23.51 | lorenzo | one thing I can't seem to be passing to the macro |
| 09:23.57 | lorenzo | is stuff like user1&user2 |
| 09:24.06 | lorenzo | for exten => s,1,Dial(SIP/${ARG1},20) |
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| 09:32.52 | BlackDex | Hello there |
| 09:32.59 | BlackDex | i get the following error |
| 09:33.03 | BlackDex | Function SIPPEER cannot be written to |
| 09:33.24 | BlackDex | i'm trying to get CFwdALL to work from our cisco phones |
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| 10:41.36 | jastrup | Hello. Back in the days i used a function in the dialplan so when someone called in on an extension they got a new tone where they could use the dialpad to push in a new number which they would want to call out to |
| 10:41.44 | jastrup | any idea which function i am talking about? |
| 10:42.22 | file | DISA? |
| 10:42.32 | jastrup | aha |
| 10:42.33 | jastrup | maybe |
| 10:42.44 | jastrup | is it still working in asterisk 11? |
| 10:43.01 | file | sure |
| 10:43.29 | jastrup | I'm gonna need to investigate this, he he :) thanks alot |
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| 11:42.53 | Zogot | ahoyhoy |
| 11:44.38 | file | hi |
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| 11:49.08 | Zogot | is there a way to change the caller id on transfer? |
| 11:49.57 | Zogot | John Smith calls UserA, UserA transfers to UserB but UserB sees a caller id of something like T: John Smith |
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| 12:43.07 | cusco | hi folks |
| 12:43.31 | cusco | I've submitted a issue a couple of weeks ago: https://issues.asterisk.org/jira/browse/ASTERISK-24163 and forgot the backtrace |
| 12:43.38 | cusco | then I added a comment with backtrace as a attachment |
| 12:43.44 | cusco | but it is still pending feedback |
| 12:43.51 | cusco | should I add the backtrace some other way? |
| 12:49.05 | Zogot | include in a dialplan is always processed first correct? |
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| 12:58.59 | Tuju | i tried to get 1.8 confbridge to work with exten => 8888,n,ConfBridge(${EXTEN}, cs) but it just doesn't do anything. |
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| 13:03.54 | [TK]D-Fender | Tuju: Show us |
| 13:04.15 | Tuju | how do i turn debugging on for that app? |
| 13:04.45 | Tuju | it answers but there is silense on line, after a minute or so, it starts alarming again. |
| 13:05.02 | [TK]D-Fender | "alarming"? |
| 13:05.10 | [TK]D-Fender | as in? |
| 13:05.59 | Tuju | 'tuuu, tuuuu' |
| 13:06.12 | Tuju | i can hear alarm sound on line |
| 13:06.30 | Tuju | is there an example that *should* work? |
| 13:06.37 | [TK]D-Fender | Tuju: Show us the call |
| 13:06.51 | Tuju | confbridge looks so dummy that it shouldn't be that hard. |
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| 13:20.48 | Tuju | now it works, thanks a lot [TK]D-Fender |
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| 15:01.33 | orn | How does * choose which IP address to use for media, when it's bound on multiple IPs? I have asterisk receiving signaling on one IP and sending out media from another |
| 15:04.32 | [TK]D-Fender | source IP is based on your kernel routing stack |
| 15:04.44 | [TK]D-Fender | Asterisk does not "choose". |
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| 15:14.08 | newtonr | [TK]D-Fender, Asterisk chooses all! |
| 15:14.42 | [TK]D-Fender | YOU HAVE CHOSEN ... POORLY! |
| 15:15.35 | newtonr | withers away into a skeleton |
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| 15:21.49 | orn | thanks d-fender |
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| 15:23.50 | orn | got me pointed in the right direction -- i needed a route for the specific host with the desired source address -- i didn't have the route for the SIP address, only the media address, but the SIP address is what made asterisk construct the SDP, obviously |
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| 15:52.02 | hexanol | wdoekes: I've finally had some time to check for the "microsecond issue" regarding cel_odbc |
| 15:52.08 | hexanol | I don't know if you remember, you sent me a patch |
| 15:52.15 | hexanol | like 1 week ago |
| 15:52.31 | hexanol | I've updated it: http://fpaste.org/127083/raw/ |
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| 15:53.16 | hexanol | and now I have microsecond precision in my cel table (in postgresql) using cel_odbc |
| 15:53.53 | hexanol | I guess it should work with most ODBC driver, since the timestamp it generated seems to be ODBC compliant, as far as I understand ODBC (i.e. not much) |
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| 15:59.48 | hexanol | someone things it would make sense to try to incorporate this upstream ? especially knowing that cel_odbc.conf.sample implies that you should be able to have eventtime with microsecond precision, which is not the case, at least not in asterisk 11 |
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| 16:41.08 | KNERD | Anyone ever heard of a mass NAT failure when people on many different networks are trying to connect their extensions on a network, but cannot do it because they keep getting a SIP 403 "Not Authorized" because their LAN address is being reported to Asterisk, and not their public IP address? |
| 16:43.10 | [TK]D-Fender | look at your peers. |
| 16:43.15 | [TK]D-Fender | And just show us your examples |
| 16:43.25 | KNERD | no peers..offline |
| 16:43.30 | KNERD | because they all getting 403 |
| 16:43.59 | KNERD | despite connecting to from different networks to the same one |
| 16:44.38 | [TK]D-Fender | peer CONFIG |
| 16:45.09 | KNERD | the configs are fine This just started |
| 16:45.19 | KNERD | but I not actually asking for troubleshooting help, just just an idea of what may be happenbing |
| 16:45.47 | KNERD | if I connect an extrension through OpenVON, works fine...so it has the be the ISP network |
| 16:45.55 | KNERD | *OpenVPN |
| 16:47.34 | [TK]D-Fender | moves on |
| 16:47.54 | KNERD | I knew you could not know what it was :-) |
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| 16:58.36 | anonymouz666 | Who uses respoke here? |
| 16:58.52 | anonymouz666 | I just have found it's a interesting API |
| 17:04.18 | lorenzo | hi! |
| 17:04.23 | lorenzo | I can't seem to pass arguments with & |
| 17:04.34 | lorenzo | to this: http://pastebin.com/UF1SREqL |
| 17:05.55 | [TK]D-Fender | lorenzo: I don't see you trying anywhere in there |
| 17:07.11 | lorenzo | [TK]D-Fender: I'm calling the macro from outside |
| 17:07.24 | [TK]D-Fender | lorenzo: I don't see anything calling anything in there |
| 17:07.31 | lorenzo | e.g. exten => 101,1,Macro(users-enlab-ext,lorenzo&lorenzo-mobi,3420202020) |
| 17:07.47 | lorenzo | the problem is in the first Dial |
| 17:07.52 | [TK]D-Fender | show us the call |
| 17:10.33 | lorenzo | http://pastebin.com/9f4iYFWA |
| 17:11.12 | [TK]D-Fender | Macro("SIP/l.test-test-00000040", "users-enlab-ext,l.test&l.test-mobi,3420xxxxx") |
| 17:11.13 | *** join/#asterisk JerJer (~Adium@asterisk/original-h323-guy/JerJer) |
| 17:11.17 | [TK]D-Fender | <PROTECTED> |
| 17:11.26 | [TK]D-Fender | It is VERY clearly passing the & |
| 17:11.49 | anonymouz666 | JerJer is back on |
| 17:12.13 | JerJer | woot |
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| 17:14.52 | lorenzo | [TK]D-Fender: hmm, I wonder why it tells me that the format is wrong |
| 17:14.57 | lorenzo | it uesd to work outside the macro |
| 17:15.51 | [TK]D-Fender | lorenzo: No, that formatting is also clearly wrong. |
| 17:16.17 | [TK]D-Fender | lorenzo: "core show application dial" |
| 17:19.47 | anonymouz666 | JerJer: using much h323 signalling? |
| 17:20.06 | JerJer | not with Asterisk |
| 17:20.26 | JerJer | sadly Cisco Telepresence uses H.323 still |
| 17:20.54 | anonymouz666 | Cisco su... |
| 17:21.02 | JerJer | agrees |
| 17:21.37 | JerJer | Cisco turned a decent Tandburg product into a big joke |
| 17:21.47 | JerJer | tanburg |
| 17:22.18 | JerJer | oh +D :) |
| 17:22.32 | anonymouz666 | JerJer: what about the virtualization field? I remember you were pretty ahead on this topic |
| 17:23.11 | JerJer | running asterisk virtualized? sure |
| 17:23.12 | lorenzo | right I was missing a SIP/ :) |
| 17:23.57 | anonymouz666 | JerJer: it is a big deal |
| 17:24.29 | JerJer | heh - i've been doing it since almost day one |
| 17:24.52 | anonymouz666 | JerJer: external gateways and virtualization |
| 17:25.04 | JerJer | I started with linux-vserver, then xen, then vmware |
| 17:25.39 | JerJer | lately i've been playing with vagrant |
| 17:31.03 | [TK]D-Fender | JerJer: Haven't given KVM a whirl yet? |
| 17:32.39 | JerJer | hmm⦠not that i can remember |
| 17:33.20 | JerJer | but for some reason it sounds familiar |
| 17:33.23 | *** join/#asterisk timahvo1 (~rogue@197.237.134.227) |
| 17:38.09 | *** join/#asterisk [Outcast] (~jjones@64.206.121.41) |
| 17:42.45 | anonymouz666 | KVM works just fine |
| 17:51.11 | [TK]D-Fender | anonymouz666: Tried PCI() passthrough for DAHDI by any chance? |
| 17:51.42 | *** join/#asterisk JerJer (~Adium@asterisk/original-h323-guy/JerJer) |
| 17:53.56 | *** join/#asterisk navaismo (0c0adbe1@gateway/web/freenode/ip.12.10.219.225) |
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| 17:56.39 | anonymouz666 | nops, did you? |
| 17:59.31 | [TK]D-Fender | anonymouz666: Nope, I've only just started looking at KVM (via ProxMox) as an alternative to the ESXi my head office was suggesting. |
| 17:59.53 | [TK]D-Fender | For general use, not specifically Asterisk |
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| 18:42.12 | linocisco | hi all |
| 18:42.33 | linocisco | what is cheap and reliable international calls to landline and mobile? |
| 18:50.44 | *** join/#asterisk txapelgorri (5f3c173e@gateway/web/freenode/ip.95.60.23.62) |
| 18:52.21 | *** join/#asterisk mirela666 (~mirko.bra@95.180.116.173) |
| 18:52.46 | linocisco | what is cheap and reliable international calls to landline and mobile? |
| 18:56.34 | txapelgorri | Hi, I'm trying to make Asterisk go with call transfers (blind and attended). I have configured features.conf but somehow the DTMF is being 'ignored'. Log shows me that Asterisk is getting the DTMF tones but I suspect that it is not taking care of them because it says that they are being queued. For example, when I press just the wildcard this is what I get: |
| 18:57.05 | txapelgorri | DTMF[19157]: DTMF end '*' received on SIP/test_voip_1-0827e248, duration 0 ms DTMF[19157]: DTMF begin emulation of '*' with duration 100 queued on SIP/test_voip_1-0827e248 DTMF[19157]: DTMF end emulation of '*' queued on SIP/test_voip_1-0827e248 |
| 18:57.47 | txapelgorri | Any idea to deal with this problem? |
| 18:58.23 | *** join/#asterisk JerJer (~Adium@asterisk/original-h323-guy/JerJer) |
| 18:59.09 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 19:02.46 | [TK]D-Fender | txapelgorri: Show us the complete call |
| 19:03.09 | [TK]D-Fender | ~pb |
| 19:03.09 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:03.11 | [TK]D-Fender | ^^^ |
| 19:12.06 | txapelgorri | You mean the Asterisk CLI output of a complete call? |
| 19:12.29 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 19:12.37 | [TK]D-Fender | yes |
| 19:14.16 | *** part/#asterisk Sean-Der (~Sean-Der@siobud.com) |
| 19:16.08 | txapelgorri | http://pastebin.com/A6GXyBM4 |
| 19:17.53 | txapelgorri | Between the last 2 lines I have pressed some DTMF keys, but nothing appears in the console. I have to activate the DTMF logging to the syslog. There is where I get the previous commented log lines. |
| 19:23.10 | [TK]D-Fender | txapelgorri: "core show application dial" <- |
| 19:23.20 | [TK]D-Fender | txapelgorri: You have left off options. |
| 19:26.27 | txapelgorri | I paste if you want but all the sections in the output of that command are telling the same thing "Not available" :( |
| 19:27.09 | txapelgorri | I paste if you want but all the sections in the output of that command are telling the same thing: "Not available" :( |
| 19:27.59 | [TK]D-Fender | txapelgorri: You did not specify the option required to ALLOW transfers |
| 19:28.10 | txapelgorri | I see |
| 19:28.22 | [TK]D-Fender | txapelgorri: Go re-read Dial's instructions |
| 19:28.22 | txapelgorri | Where must I specify it? |
| 19:28.31 | [TK]D-Fender | [15:23][TK]D-Fendertxapelgorri: "core show application dial" <- |
| 19:30.59 | *** join/#asterisk [Outcast] (~jjones@64.206.121.41) |
| 19:32.55 | txapelgorri | I don't get it, sorry: when you tell me to re-read Dial's instructions, what does it mean?. Currently the output of "core show application dial" is http://pastebin.com/8nQ2WL1D |
| 19:33.21 | txapelgorri | Is there a previous command that I'm missing? |
| 19:34.10 | [TK]D-Fender | Something's migging on your install |
| 19:34.12 | [TK]D-Fender | missing* |
| 19:34.15 | txapelgorri | Aha |
| 19:34.34 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Application_Dial |
| 19:36.13 | txapelgorri | I see, so probably is my version of Asterisk, right?. I'm going to see an update or similar... |
| 19:40.12 | [TK]D-Fender | It's how it was installed that it didn't have the standard instructions bundled with it |
| 19:47.04 | *** join/#asterisk mokmeister (~mokmeiste@86-43-185-179-dynamic.b-ras1.pgs.portlaoise.eircom.net) |
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| 20:02.52 | txapelgorri | [TK]D-Fender: thanks a lot. It seems that my version is quite obsolete and the OS version is discontinued too, so I have go figure another way of solve this problem. As I said: thanks anyway :) |
| 20:03.32 | [TK]D-Fender | txapelgorri: Does stop you from fixing your dialplan or using other docs |
| 20:03.39 | [TK]D-Fender | doesn't* |
| 20:05.39 | txapelgorri | [TK]D-Fender: I see, I but I tought that without Dial's instructions I was in a dead end, wasn't I? |
| 20:05.54 | [TK]D-Fender | No, you just don't get to view them from CLI. |
| 20:05.59 | [TK]D-Fender | So follow the link I gave you |
| 20:09.03 | txapelgorri | [TK]D-Fender: I have already try the "t" and "T" options in the dialplan, but with no positive results :( You mean another option maybe? |
| 20:09.16 | [TK]D-Fender | show us |
| 20:09.26 | *** part/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 20:09.32 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 20:10.36 | txapelgorri | exten => 1005,1,Dial(SIP/1005@192.168.0.200,14,t) |
| 20:10.48 | txapelgorri | <PROTECTED> |
| 20:11.00 | txapelgorri | For both users |
| 20:11.47 | [TK]D-Fender | Show us the CALL. |
| 20:11.53 | txapelgorri | Oh |
| 20:12.28 | txapelgorri | That is the previous one, the one I pasted on pastebin previously... |
| 20:13.42 | txapelgorri | This one: http://pastebin.com/S0R5gtiM |
| 20:14.00 | [TK]D-Fender | <PROTECTED> |
| 20:14.13 | *** join/#asterisk aross42 (~aross@CPEbcc8100f769b-CMbcc8100f7698.cpe.net.cable.rogers.com) |
| 20:14.51 | txapelgorri | What is it wrong with that? |
| 20:15.03 | [TK]D-Fender | Where do you see the dial options required on that line? |
| 20:16.13 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 20:17.24 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 20:17.56 | txapelgorri | You are not referring to the dialplan, are you? |
| 20:18.40 | [TK]D-Fender | I have not seen a call where you are using to appropriate options yet |
| 20:29.07 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 20:29.22 | [TK]D-Fender | checkout time, heading home |
| 20:33.00 | *** join/#asterisk DynamicFail (~DynamicFa@147.177.152.171) |
| 20:33.22 | DynamicFail | Anyone familar with TETRA know if it is possible to integrate it into SIP to make group calls? |
| 20:34.41 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 20:35.26 | sekil | can one put something like Dial(SIP/101@a.com@ip:port) and hope for call to end on the server with the mentioned ip? |
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| 20:42.36 | txapelgorri | Thanks, I think I have solved :) |
| 20:44.24 | *** join/#asterisk newtonr (~newtonr@173-17-135-67.client.mchsi.com) |
| 20:44.24 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 20:52.19 | *** join/#asterisk k5673 (~jcapurro@190.52.189.6) |
| 20:53.33 | sekil | comes down to..how to change the request uri domain part on outbound invite in the dialplan? |
| 20:55.25 | k5673 | Hello! I have a setup where i got one Elastix connected to the PSTN, acting as a gateway, and another Elastix with a Virtual Fax configured. When i call to the gateway and dial the Virtual Fax ext., i got fax tone, but asterisk complains in /var/log/asterisk/full with "chan_sip.c FAX CNG detected but not fax extension". I've already installed Free Fax for Asterisk. Somebody knows why this isn't working? |
| 20:55.57 | *** join/#asterisk iceslice (~Rashed@unaffiliated/iceslice) |
| 21:01.03 | *** join/#asterisk areski (~areski@80.174.128.86.dyn.user.ono.com) |
| 21:02.09 | WIMPy | sekil: you want to have something different in the to: than the configured host? |
| 21:02.47 | sekil | no |
| 21:03.07 | sekil | diff domain in the SIP R-URI |
| 21:03.32 | WIMPy | 1. where? and 2. different from what? |
| 21:04.51 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:05.50 | sekil | invite request uri |
| 21:06.12 | sekil | not the ip of the host sending to..but the domain |
| 21:07.13 | WIMPy | That's still a little vague. What line of the SIP request? |
| 21:07.27 | sekil | vague? |
| 21:07.54 | WIMPy | Only the INVITE line itself? |
| 21:09.01 | sekil | ok..on invite first line is i.e INVITE sip:105@172.27.33.163:5060 SIP/2.0 |
| 21:09.14 | sekil | that's the invite req uri.. |
| 21:09.32 | sekil | now I want it to be sip:105@blah.com |
| 21:10.18 | sekil | INVITE sip:105@blah.com SIP/2.0 ...to be precise |
| 21:10.35 | WIMPy | Define a peer with host=blah.com and outboundproxy=172.27.33.163,force. |
| 21:12.21 | *** join/#asterisk aness (~aness@2a02:fe0:c310:3d0:54e:ccf2:94bd:a3f5) |
| 21:12.29 | sekil | yeah...should work .. |
| 21:13.03 | k5673 | Hello! Somebody knows why asterisk says "chan_sip.c FAX CNG detected but not fax extension" in /var/log/asterisk/full, even if the ext. is giving fax tone? |
| 21:14.01 | WIMPy | k5673: That's what it says. |
| 21:14.21 | WIMPy | "not" fax extension or "no"? |
| 21:16.02 | k5673 | WIMPy: I have configured the iax2 extension and when you call to it, the virtual fax answers, but that notice appears and the fax machine in the PSTN can't send... |
| 21:16.18 | sekil | hm.. |
| 21:16.24 | sekil | it goes to resolve blah.com |
| 21:17.09 | WIMPy | k5673: Read the message again. it says "no fax extension". And that's what it is. You haven't configured a fax extension. |
| 21:18.06 | WIMPy | sekil: I think that's a known issue. But unless you get delays if it doesn't work, it shouldn't matter. |
| 21:18.18 | k5673 | WIMPy: Do you know where is some documentation about the syntax in how to create it? |
| 21:18.50 | [TK]D-Fender | k5673: exten => fax,1,...... |
| 21:19.04 | WIMPy | k5673: It might be hard to find someone who knows how to configure elastix. |
| 21:19.07 | [TK]D-Fender | k5673: Go read the dialplan basics about Asterisk Standard Extensions |
| 21:19.07 | sekil | WIMPy: it disregards outboundproxy setting apparently |
| 21:19.15 | *** join/#asterisk DynamicFail (~DynamicFa@147.177.70.163) |
| 21:19.25 | WIMPy | sekil: Did you use ",force"? |
| 21:19.39 | k5673 | [TK]D-Fender: Thanks! |
| 21:19.50 | sekil | yup |
| 21:20.43 | WIMPy | In that case I have no idea. |
| 21:22.17 | sekil | it resolved blah.com to external ip which ofcourse don't have sip listening..and made the peer qualify as unreachable |
| 21:22.29 | sekil | it does not send signalling to op |
| 21:25.16 | *** join/#asterisk Qwell (north@asterisk/developer/Qwell) |
| 21:25.16 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 21:25.55 | [TK]D-Fender | sekil: Show your config and the debug to match |
| 21:26.52 | WIMPy | sekil: Did you restart Asterisk or unload/load chan_sip? |
| 21:27.18 | sekil | just sip reload |
| 21:27.35 | WIMPy | "sip reload" often doesn't work. |
| 21:28.49 | *** join/#asterisk DynamicFail (~DynamicFa@147.177.70.163) |
| 21:29.04 | WIMPy | It's ok if you just added new peers, but if you modified something it usually fails. |
| 21:29.24 | sekil | I just restarted * |
| 21:29.28 | sekil | still the same |
| 21:29.56 | sekil | outboundproxy=172.27.33.163,force |
| 21:30.14 | sekil | host=test.com ...this is actually the domain to test on my opensips |
| 21:30.29 | sekil | * resolved test.com to an ip... |
| 21:30.48 | sekil | os-test 50.23.225.49 No No 5060 UNREACHABLE |
| 21:36.19 | sekil | WIMPy: cracked it |
| 21:36.28 | sekil | WIMPy: one must turn qualify off |
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| 23:10.33 | bibz2 | ok, now I removed any form of NAT and connected asterisk directly to the internet |
| 23:10.56 | bibz2 | the amount of dropped calles significantly decreased |
| 23:11.23 | bibz2 | but there are still some calls which drop, but not after a static amount of time, but rather randomly.. some drop after 4 minutes, some after 6-7.. |
| 23:28.55 | *** join/#asterisk aross42 (~aross@192-0-133-151.cpe.teksavvy.com) |
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