| 00:14.14 | cj | does asterisk require that each peer have its own IP address? |
| 00:16.13 | *** join/#asterisk lucasb (~lucasb@S0106000c42710923.ok.shawcable.net) |
| 00:16.46 | rogersja | cj: i think you can use host=dynamic if you can't specify the exact address |
| 00:17.58 | cj | they're all going to be the same IP with differing ports |
| 00:18.00 | cj | will this be okay? |
| 00:21.13 | rogersja | i think it should work |
| 00:21.22 | rogersja | give it a try, i never have done that |
| 00:21.28 | cj | alright. I wonder what the deal is. only one of the phones is able to successfully log in. |
| 00:24.11 | rogersja | actually you can specify port along with host |
| 00:24.27 | cj | and dynamic won't build that for you? |
| 00:25.49 | rogersja | i think you can add port when you have host=dynamic |
| 00:26.25 | cj | I don't know that I can get the devices to use a consistent source port... and if I could, it wouldn't scale... |
| 00:26.30 | rogersja | when you say 'log in' you mean the device is registered? |
| 00:26.41 | cj | right, register, not 'log in'. |
| 00:27.04 | rogersja | hrm, in not sure then |
| 00:27.15 | rogersja | seems to me that should work |
| 00:27.23 | cj | alright. I'll see if I can figure out what's going on. |
| 00:28.28 | cj | http://paste2.org/p/1389808 |
| 00:28.49 | cj | so asterisk thinks it's registered, but the phone doesn't. |
| 00:31.05 | cj | http://paste2.org/p/1389814 |
| 00:32.51 | cj | so it looks like our load balancer isn't able to correctly reverse the NAT |
| 00:34.23 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 00:42.38 | drfreeze | Here are the logs from the remote phone trying to register |
| 00:42.39 | drfreeze | http://pastie.textmate.org/private/tskbp3pg0utkj3rtnpvng |
| 00:47.38 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v013-117.mobile.uci.edu) |
| 00:52.57 | WiretapSeven | drfreeze, have you tried a different softphone? |
| 00:56.56 | drfreeze | http://pastie.textmate.org/private/betmztsex1afbkd868pag |
| 00:57.17 | drfreeze | WiretapSeven: just tried zoiper and the actual polycom |
| 00:57.44 | drfreeze | the zoiper will register immediately if connected thru a vpn via the computer |
| 00:58.01 | WiretapSeven | drfreeze, why are you clipping the logs, we want the whole lot |
| 00:58.31 | drfreeze | I'll get more |
| 00:58.38 | WiretapSeven | and do I see NAT there? |
| 00:58.42 | WiretapSeven | also is there a firewall? |
| 00:59.59 | *** join/#asterisk coppice (~chatzilla@62.166.232.220.dyn.pacific.net.hk) |
| 01:01.08 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 01:03.14 | drfreeze | WiretapSeven: https://gist.github.com/0fff70227db82c03595c |
| 01:04.18 | drfreeze | connection is: asterisk <- cloud -> WRT54GL -> Polycom |
| 01:04.35 | drfreeze | asterisk and WRT have VPN connection |
| 01:07.41 | WiretapSeven | drfreeze, is the WRT54GL NATing, or not |
| 01:07.51 | WiretapSeven | they have a habit of doing it even if its off if you're running DD-WRT |
| 01:11.42 | WiretapSeven | drfreeze, with the non-clipped log I see a 401, make sure your peer is set to dynamic and that the shared secret is correct |
| 01:12.47 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 01:12.56 | rogersja | leifmadsen: ping |
| 01:21.33 | *** join/#asterisk [netman] (~netman@20.Red-80-39-52.staticIP.rima-tde.net) |
| 01:35.01 | Freeaqingme | With AGI I'm using EXEC DIAL <something> |
| 01:35.19 | Freeaqingme | is it possible to somehow receive dtmf stuff in my app? |
| 01:36.29 | leifmadsen | rogersja: pong? |
| 01:38.25 | rogersja | just looking through your book, trying to setup ISN dialing |
| 01:38.35 | *** join/#asterisk galaxywatcher (~galaxywat@pdpc/supporter/active/galaxywatcher) |
| 01:38.45 | rogersja | got the outgoing bit no problem, but having difficulties with the incomming |
| 01:39.24 | rogersja | asterisk seems to be sending a 401 in reply to an incoming call |
| 01:40.22 | rogersja | any ideas? |
| 01:41.08 | leifmadsen | rogersja: 401 Unavailable usually means authentication errors I guess |
| 01:41.14 | rogersja | yep |
| 01:41.24 | rogersja | even though I have allowguest=yes |
| 01:42.11 | leifmadsen | would have to see some additional information like console output etc |
| 01:42.16 | leifmadsen | otherwise it's just random stabs in the dark :) |
| 01:42.21 | leifmadsen | wouldn't want to hit any innocents |
| 01:42.34 | rogersja | standby |
| 01:48.33 | *** join/#asterisk Micc (~Micc@c-98-232-41-66.hsd1.wa.comcast.net) |
| 01:50.26 | rogersja | leifmadsen: http://pastebin.com/rPP4zjgF |
| 01:50.37 | rogersja | console output |
| 01:50.45 | rogersja | with sip debug on |
| 01:52.29 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 01:59.19 | rogersja | actually this may be more relevant: http://pastebin.com/RG6bcDuZ |
| 01:59.44 | rogersja | this is just the logs from the incomming request, not in and out as the previous one was. |
| 01:59.53 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 01:59.56 | rogersja | outgoing is fine, i can dial other ISN's |
| 02:02.58 | \DSAFEW\ | rogersja, what's the upstream server? |
| 02:04.07 | rogersja | both are asterisk servers |
| 02:04.40 | rogersja | i used an asterisk server i have at chunk host to dial via ISN to my asterisk server here |
| 02:05.02 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 02:05.03 | leifmadsen | rogersja: well it looks normal to me -- a peer is coming in, it is being matched by IP, and Asterisk is required to ask for authentication |
| 02:05.03 | rogersja | the NAPTR record points to the server here |
| 02:05.17 | leifmadsen | rogersja: you need to connect from an IP that isn't setup to require authentication in your sip.conf file |
| 02:05.23 | leifmadsen | I bet it works if I call it :) |
| 02:05.33 | leifmadsen | what is your ISN? |
| 02:05.33 | rogersja | give it a shot |
| 02:05.48 | rogersja | try 1*1407 |
| 02:06.11 | leifmadsen | MoH |
| 02:06.15 | rogersja | damn |
| 02:06.27 | \DSAFEW\ | rogersja, triple check the auth info |
| 02:06.36 | leifmadsen | rogersja: that's not right? |
| 02:06.41 | rogersja | no thats right |
| 02:06.43 | leifmadsen | seemed to work as far as I could tell |
| 02:06.53 | rogersja | yes i saw the call come through as well |
| 02:07.09 | leifmadsen | rogersja: try 7659*460 |
| 02:07.23 | leifmadsen | rogersja: I hung up pretty quick if you were wondering why the call didn't stay up long |
| 02:07.44 | leifmadsen | ITAD_RESULT=1@asterisk.rogerswest.net |
| 02:07.47 | leifmadsen | all looks good |
| 02:07.54 | rogersja | i dont love polycoms though |
| 02:08.15 | leifmadsen | everyone does! |
| 02:08.21 | leifmadsen | there is a song about it! so it must be true |
| 02:08.27 | leifmadsen | goes off to pack for the weekend |
| 02:08.33 | rogersja | well you've got me there |
| 02:08.54 | WiretapSeven | so glad I didn't buy polycoms by the sounds of it :P |
| 02:09.25 | rogersja | WiretapSeven: dial 7659*460, you'll see why |
| 02:09.39 | WiretapSeven | I don't have ISN set up |
| 02:09.40 | leifmadsen | I don't use anything but polycoms. They just work. |
| 02:09.45 | WiretapSeven | so I just get NU |
| 02:09.46 | leifmadsen | WiretapSeven: weak sauce! |
| 02:09.55 | leifmadsen | sip:polycom@shifteight.org |
| 02:10.12 | WiretapSeven | leifmadsen, if you can point me to setting it up for freepbx I'll gladly do it, I did a breif googlng and there was no answer |
| 02:10.30 | leifmadsen | WiretapSeven: check nerdvittles |
| 02:11.02 | rogersja | so i wonder why i can't dial myself via ISN |
| 02:12.32 | rogersja | leifmadsen: thanks, btw! |
| 02:13.19 | WiretapSeven | leifmadsen, looks interesting |
| 02:13.24 | WiretapSeven | might look at that soon enough |
| 02:13.35 | WiretapSeven | but I'll save it for once I have this phone up, I hate provisioning on the SPA922 :P |
| 02:16.36 | rogersja | \DSAFEW\: sorry, which auth info was that |
| 02:17.06 | rogersja | i still cant get my remote asterisk box to dial my ISN |
| 02:19.08 | WiretapSeven | rogersja, if the call is coming from a peer which under other circumstances local asterisk expects should authenticate, it won't work |
| 02:19.09 | \DSAFEW\ | rogersja, oh can you edit the remote box's config? |
| 02:19.43 | rogersja | yes |
| 02:20.04 | \DSAFEW\ | rogersja, I was saying be sure sip user auth is correct, but you want it to work with anyone? |
| 02:20.34 | rogersja | well it seems to work with anyone else dialing in, as leif just did |
| 02:20.46 | rogersja | and i can dial ISN's out fine from both boxes |
| 02:21.11 | rogersja | i just cant dial out from the remote box via ISN in to my local box |
| 02:21.15 | \DSAFEW\ | rogersja, not sure if this matters, what's the extension's type? friend? |
| 02:21.16 | WiretapSeven | \DSAFEW\, ISN does kinda rely on it working with 'just anyone' |
| 02:22.00 | rogersja | WiretapSeven: exactly, and thats what I want. |
| 02:22.17 | rogersja | Im just trying to figure why remote box cant ISN dial my local box |
| 02:22.31 | rogersja | the two are not connected in any fashion |
| 02:22.46 | rogersja | neither one registers to the other |
| 02:23.38 | WiretapSeven | nor is peered in a non-registering peer to the other? |
| 02:25.18 | rogersja | they used to be IAX peers with each other, but i have shut that off on both, while i troubleshoot this |
| 02:25.52 | rogersja | sorry 'IAX friends' with eachother |
| 02:28.42 | WiretapSeven | shut off as in disabled or removed from the config? |
| 02:28.51 | \DSAFEW\ | rogersja, do you have this? transport=tcp,udp |
| 02:29.13 | rogersja | the iax register entires are commented out in both iax.conf severs |
| 02:29.36 | rogersja | \DSAFEW\: no i don't have that |
| 02:29.37 | *** join/#asterisk marlowe (~marlowe@ip68-100-150-54.dc.dc.cox.net) |
| 02:30.30 | WiretapSeven | rogersja, comment out or remove the entire peer/friend definition |
| 02:30.42 | WiretapSeven | just commenting out the register doesn't stop the lookup |
| 02:33.13 | rogersja | just did that, no change |
| 02:34.20 | rogersja | the only thing i can think of, is that the softphone im using to connect and dial out of the remote box, is on the same lan as the local box |
| 02:34.47 | rogersja | perhaps some nat conflict |
| 02:36.02 | leifmadsen | rogersja: like I said, it's because it's matching the IP and the authentication is failing |
| 02:36.59 | rogersja | i just can't figure where its matching the IP, unless you are agreeing with what i said above :s |
| 02:46.58 | rogersja | well its not the issue with the softphone being on the same lan as the local box |
| 02:47.21 | rogersja | i just had someone else register with remote box and try from there, same problem |
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| 04:47.01 | Preytell | is there an issue with IAX between 1.8.0 and 1.8.3? I update one system to 1.8.3, have a trunk between the two that worked before the update. Now when I make a call I get the bearercapibility_notavail hangup code 58 error. |
| 04:59.04 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 05:02.55 | rogersja | Preytell: im running rc3 and do not have issues with IAX between boxes |
| 05:06.54 | toresbe | Damn. No matter what I do, my Asterisk won't accept any incoming DTMFs in a DISA. |
| 05:07.57 | toresbe | Well, no matter what I do with the exception of DTMF signalling, that is; my router has previously worked fine with translating pulse dialling into OOB DTMF signals. |
| 05:09.42 | rogersja | toresbe: nat issue? |
| 05:10.05 | toresbe | Why would that affect _only_ DTMF signalling, though? |
| 05:10.08 | toresbe | Sound is fine. |
| 05:10.21 | rogersja | hrm, oh that is odd |
| 05:13.31 | toresbe | I'm using RTP NTE RFC2833 signalling, from a Cisco 2621XM. |
| 05:16.17 | *** join/#asterisk logicwrath (~no@c-68-62-24-205.hsd1.mi.comcast.net) |
| 05:18.23 | Preytell | rogersja: Are both boxes running rc3? in my case one is 1.8.0 and the other is 1.8.3, was working when both were 1.8.0. |
| 05:18.32 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 05:18.42 | Preytell | I can upgrade the other box, see if that fixes it but will have to wait to do that. |
| 05:18.58 | rogersja | yes both on rc3 |
| 05:31.32 | toresbe | Got RTP packet from 88.87.32.14:16570 (type 101, seq 005133, ts 019840, len 000004) |
| 05:31.36 | toresbe | Got RTP RFC2833 from 88.87.32.14:16570 (type 101, seq 005133, ts 019840, len 000004, mark 0, event 00000010, end 0, duration 00000) |
| 05:31.39 | toresbe | Got RTP packet from 88.87.32.14:16570 (type 101, seq 005134, ts 019840, len 000004) |
| 05:31.42 | toresbe | Got RTP RFC2833 from 88.87.32.14:16570 (type 101, seq 005134, ts 019840, len 000004, mark 0, event 00000010, end 0, duration 00000) |
| 05:31.57 | toresbe | This _does_ mean it _is_ receiving the RTP RFC2833 packets correctly, right? |
| 05:34.19 | Preytell | yes and no, I am having the same issue now, and it seems to be related to the duration of the 2833 packet. |
| 05:35.04 | Preytell | yes it's sending the digit, but things like duration, volume, and other things are related to the failure. |
| 05:35.15 | toresbe | aha. Damn. |
| 05:40.53 | *** join/#asterisk brianc (~brian@99-61-68-214.lightspeed.austtx.sbcglobal.net) |
| 05:46.56 | kaldemar | toresbe: set dtmf on the console line of logger.conf, it will show you when the channel detects DTMF. |
| 05:47.30 | toresbe | Thanks! |
| 05:48.29 | kaldemar | also, some clients seem to want an Aswer before the DISA to even send any DTMF. |
| 05:48.45 | Preytell | by the way, it sucks but I had to switch to dtmfmode=inband to get around the issue. I could not get 2833 to work reliably. |
| 05:49.03 | toresbe | Preytell: Well, I have pulse dialling phones, I'm screwed ;) |
| 05:49.18 | Preytell | wow. |
| 05:49.44 | Preytell | well I'm off to bed. good night all. |
| 05:50.30 | toresbe | This is so annoying! This WORKED! Very recently! |
| 06:04.15 | rogersja | im off as well, can't even spell weasels right its so late |
| 06:04.21 | rogersja | takes dog for walk |
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| 06:52.23 | jacc0 | good morning all :0 |
| 06:54.28 | kleszcz | morning |
| 06:54.35 | *** join/#asterisk unhack|wrk (~unhack@unhack.olma.ru) |
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| 07:07.33 | wdoekes2 | morning |
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| 07:45.30 | unhack|wrk | greetings |
| 07:46.20 | unhack|wrk | can you help me with one problem. I need to replace the backslash symbol in asterisk var |
| 07:46.28 | unhack|wrk | http://pastebin.com/3x1haW3k |
| 07:47.06 | unhack|wrk | it is the description of my attempts |
| 07:48.28 | kaldemar | unhack|wrk: which version of asterisk are you using? |
| 07:49.21 | unhack|wrk | asterisk -V |
| 07:49.22 | unhack|wrk | Asterisk 1.8.3.2 |
| 07:53.00 | kaldemar | \, is a literal comma character for the REPLACE function. |
| 07:55.53 | kaldemar | Set(__reception4=${FILTER(a-zA-Z1-9\,,${reception3})}) would be one approach |
| 07:57.42 | unhack|wrk | thx |
| 07:57.48 | unhack|wrk | i understand an idea |
| 08:09.04 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
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| 08:22.38 | jacc0 | Woei!!! I've got a coredump of a crash!!!!!!! finaly!!! |
| 08:23.31 | kaldemar | unhack|wrk: Set(__reception4=${REPLACE(reception3,\\\\\)}) will work too. the find argument is passed on to ast_get_encoded_str which returns 3 or less \'s as an empty string. four will be returned as a single \. fubar? |
| 08:23.31 | sxpert | \o/ |
| 08:30.26 | *** join/#asterisk tatramaco (~IceChat77@chello089173132123.chello.sk) |
| 08:30.46 | tatramaco | good morning |
| 08:31.06 | tatramaco | <---newbiw looking for some help |
| 08:31.28 | tatramaco | connecting my brand spanking new asterisk to my sip provider t-com.sk |
| 08:31.51 | tatramaco | I get incoming calls fine but a "CONGESTED" message when dialing out |
| 08:31.55 | tatramaco | any ideas ? |
| 08:38.02 | jacc0 | @kaldemar: could you point me to the document about backtracing one more time? |
| 08:39.39 | jacc0 | ~backtrace |
| 08:39.39 | 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 |
| 08:39.48 | jacc0 | did it myself ;) |
| 08:41.36 | unhack|wrk | kaldemar, thank you very much |
| 08:42.01 | unhack|wrk | it's very impressive |
| 08:42.09 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:8c4b:5b82:a6ac:6e48) |
| 08:43.42 | tatramaco | anybody else using t-com.sk as ITSP ? |
| 08:52.14 | jacc0 | I get this error when starting gdb : warning: Can't read pathname for load map: Input/output error. |
| 08:52.19 | jacc0 | is that a problem? |
| 09:01.28 | jacc0 | do I need to attach bbacktrace.txt and gdb.txt to the issue? or just gdb.txt? |
| 09:03.52 | kaldemar | http://www.asterisk.org/developers/bug-guidelines |
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| 10:21.57 | *** join/#asterisk psilvao (~psilvao@190.20.47.180) |
| 10:25.58 | wdoekes2 | that's odd jacc0: it shouldn't crash with num=1, len=1248.. calloc should return NULL or succeed (re: #19201).. do you have memory issues? something in dmesg? |
| 10:28.31 | jacc0 | @wdoekes2 |
| 10:28.36 | jacc0 | will have a look for you |
| 10:31.35 | jacc0 | nothing strange in dmesg |
| 10:32.22 | jacc0 | it doesn't seem I have memory issues i guess |
| 10:32.24 | *** part/#asterisk cfh (~luca@net-2-36-38-229.cust.dsl.vodafone.it) |
| 10:33.25 | jacc0 | random crashes happen on multieple machines from differend vendors |
| 10:33.48 | jacc0 | not sure it is exaclty the same issue all the time |
| 10:34.13 | jacc0 | but I gussing it is; that would exclude the option of it being a driver issue |
| 10:34.58 | jacc0 | could you explain what calloc is? and what you mean with : "calloc should return NULL or succeed" |
| 10:36.17 | jacc0 | this is what I get from /var/log/messages : asterisk[20357]: segfault at e ip b74be19f sp b5393c0c error 4 in libc-2.7.so[b744d000+155000] |
| 10:36.39 | jacc0 | is this usefull to add to the bug report? |
| 10:36.48 | *** join/#asterisk wonderworld (~ww@port-92-201-17-48.dynamic.qsc.de) |
| 10:38.42 | wdoekes2 | nah.. it just states that you had a segfault |
| 10:39.09 | wdoekes2 | install manpages-dev and look at 'man 3 calloc' |
| 10:40.19 | wdoekes2 | if it really crashes there.. and the report looks like it, it shouldn't be a driver issue.. unless that driver is randomly destroying your memory |
| 10:43.32 | wdoekes2 | you could install libc6-dbg and rerun gdb, it might reveal something |
| 10:44.44 | jacc0 | will do |
| 10:50.14 | wdoekes2 | (btw.. you filed it in the wrong category of asterisk-gui instead of asterisk) |
| 10:50.39 | jacc0 | okay, sorry for that |
| 10:50.50 | jacc0 | can it be realocated? or should I create a new one? |
| 10:54.02 | jacc0 | updated issue : https://issues.asterisk.org/view.php?id=19201 |
| 10:59.46 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 11:00.09 | jacc0 | added new gdb trace |
| 11:00.17 | *** join/#asterisk kaushal (~kaushal@182.72.14.170) |
| 11:00.19 | kaushal | hi |
| 11:00.24 | jacc0 | hello |
| 11:01.07 | wdoekes2 | jacc0: I guess you should look at memory debugging of asterisk (valgrind, MEMORY_DEBUG..) |
| 11:02.15 | kaushal | Can someone please recommend me the Hardware Server Configuration/8 or 4 port PRI Card to make Outbound Call at the rate of around 320 outbound Calls/min ? |
| 11:07.33 | *** join/#asterisk k3asd` (~k3asd`@static-217-133-87-244.clienti.tiscali.it) |
| 11:11.56 | jacc0 | pabelanger@asterisk-bugs tells me: jacc0 looks like an issue with CDRs |
| 11:26.40 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 11:27.50 | k3asd` | hi |
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| 11:30.53 | kaushal | checking in again for the query ? |
| 11:32.23 | *** join/#asterisk kwk (~kleine@carbon.gonicus.de) |
| 11:34.13 | kwk | Has anybody an idea for this bug? https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=19154 This link contains some visual images of flow graphs for SIP: https://issues.asterisk.org/file_download.php?file_id=29218&type=bug |
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| 12:22.45 | jacc0 | what deos this meen: |
| 12:22.49 | jacc0 | <PROTECTED> |
| 12:22.49 | jacc0 | [Apr 29 14:17:35] ERROR[24500]: lock.c:407 __ast_pthread_mutex_unlock: features.c line 5872 (bridge_exec): mutex 'current_dest_chan' freed more times than we've locked! |
| 12:22.49 | jacc0 | [Apr 29 14:17:35] ERROR[24500]: lock.c:438 __ast_pthread_mutex_unlock: features.c line 5872 (bridge_exec): Error releasing mutex: Operation not permitted |
| 12:26.26 | \DSAFEW\ | jacc0, which version of asterisk do you use? |
| 12:26.40 | jacc0 | 1.8.4-rc3 |
| 12:28.43 | \DSAFEW\ | jacc0, any aggressive cflags? |
| 12:29.12 | \DSAFEW\ | jacc0, did this work fine with earlier versions using the same kernel, and toolchain? |
| 12:31.04 | \DSAFEW\ | you might want to do a stack trace and fill a full bug report, but if you're in any sort of hurry to get this working, the best option is to use a previous stable build |
| 12:32.12 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:32.29 | jacc0 | okay |
| 12:32.51 | jacc0 | I have non_optimized flag en debug flags |
| 12:33.35 | \DSAFEW\ | jacc0, you can grep 24500 /var/log/asterisk/messages for a bit more info, but I think you're just running into a bug here, not a config error |
| 12:34.12 | *** join/#asterisk ]loy[ (~nobody@95.72.54.173) |
| 12:34.41 | jacc0 | same info in messages |
| 12:35.24 | \DSAFEW\ | jacc0, go to the asterisk bug tracker and submit a full bug report if you have time, developers love looking at those |
| 12:35.26 | jacc0 | I've seen it happen in asterisk 1.8.3 also |
| 12:36.14 | jacc0 | will do |
| 12:36.35 | \DSAFEW\ | has it ever worked without that error? |
| 12:37.06 | leifmadsen | jacc0: when you do that, you'll probably need to provide a backtrace and a 'core show locks' |
| 12:37.26 | leifmadsen | in Compiler Flags in menuselect, make sure DONT_OPTIMIZE and DEBUG_THREADS is enabled |
| 12:38.05 | leifmadsen | just attach to the running asterisk process with gdb, reproduce the issue, do "bt full" and "bt thread apply all", then from the Asterisk CLI do "core show locks" |
| 12:38.21 | leifmadsen | provide that information as a text file attachment to the issue, which describes how to reproduce the error |
| 12:40.28 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 12:45.34 | jacc0 | @\DSAFEW\: it works normaly; sometimes it gevis this result |
| 12:46.21 | jacc0 | @leif : will make a backtrace of this issue |
| 12:46.34 | jacc0 | I've made a backtrace of some other issue :) |
| 12:46.59 | jacc0 | only posted it in the wrong category : https://issues.asterisk.org/view.php?id=19201 |
| 12:47.29 | jacc0 | radom crash issue |
| 12:47.54 | jacc0 | in 1.8.4-rc3 |
| 12:50.53 | jacc0 | will try to write a dialplan to reproduce the bridging error |
| 12:53.21 | leifmadsen | jacc0: i can change the category for you |
| 12:53.54 | leifmadsen | moved to Asterisk project now |
| 12:56.32 | jacc0 | ;) ty |
| 12:56.41 | leifmadsen | anyone thing of an effective way in the dialplan to check CALLERID(num) to verify it is a number? |
| 12:56.58 | leifmadsen | I could probably use REGEX() somehow right? |
| 12:57.51 | jacc0 | GotoIf($[${REGEX("^[0-9]+" ${SipNumber})}]?isNumeric) |
| 12:57.51 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 12:58.36 | leifmadsen | perfect thanks! |
| 12:59.05 | kaldemar | or "${FILTER(0-9,${CALLERID(num)})}" != "" |
| 12:59.24 | leifmadsen | oh that could work too, I hadn't thought o that |
| 12:59.32 | leifmadsen | I might use the FILTER() trick actually |
| 12:59.33 | leifmadsen | thanks! |
| 13:00.16 | leifmadsen | ${LEN(FILTER(0-9,${CALLERID(num)}))} > 0 |
| 13:01.19 | jacc0 | it will fail if there is no CALLERID(num) set |
| 13:01.23 | kaldemar | re-thinking it, it may contain characters too. |
| 13:01.24 | jacc0 | I guess |
| 13:02.12 | kaldemar | if CALLERID(num) is 123abc123, FILTER 0-9 will just return 123123 but the id is still not a number. |
| 13:03.04 | kaldemar | maybe the REGEX is the way to go. |
| 13:03.06 | jacc0 | wouldn't it return abc? |
| 13:03.50 | leifmadsen | jacc0: no, it returns everything in the string that matches the chars specified |
| 13:04.01 | jacc0 | okay |
| 13:04.08 | leifmadsen | kaldemar: I'm also checking length to make sure it matches 10 and is >0 |
| 13:04.10 | kaldemar | func REPLACE would be for character removal. |
| 13:04.20 | leifmadsen | it's not perfect, but it'll work in the majority of the cases |
| 13:05.23 | jacc0 | ${LEN(FILTER(0-9,${CALLERID(num)}))} =${LEN(${CALLERID(num)})} |
| 13:05.33 | kaldemar | $[${LEN(FILTER(0-9,${CALLERID(num)}))} = ${LEN(${CALLERID(num)})}] would work as a single expression. |
| 13:05.46 | jacc0 | ;) |
| 13:05.50 | kaldemar | echoes in here... |
| 13:05.55 | jacc0 | lol |
| 13:08.20 | jacc0 | and ; do you consider 0xff a number? |
| 13:08.21 | leifmadsen | something like this... |
| 13:08.22 | leifmadsen | http://pastebin.com/ht7sq6L5 |
| 13:08.39 | leifmadsen | untested :) |
| 13:08.47 | leifmadsen | should help my buddy get close enough |
| 13:09.30 | jacc0 | missing a ) |
| 13:09.39 | jacc0 | GotoIf($[${LEN(${CID_NUM})} > 0 & ${LEN(${CID_NUM} = 10]?insert:skip) |
| 13:09.52 | jacc0 | and missing a } |
| 13:10.07 | jacc0 | GotoIf($[${LEN(${CID_NUM})} > 0 & ${LEN(${CID_NUM})} = 10]?insert:skip) |
| 13:10.07 | leifmadsen | oops yes |
| 13:10.12 | leifmadsen | you're right |
| 13:10.16 | leifmadsen | good catch, I didn't check my code :) |
| 13:10.29 | leifmadsen | someone else who is as good at catching missing braces as me :) |
| 13:10.34 | jacc0 | ${LEN(${CID_NUM})} > 0 is always true if ${LEN(${CID_NUM})} = 10 |
| 13:10.46 | jacc0 | that would make it obsolete |
| 13:11.07 | kaldemar | leifmadsen: that will still allow caller id's that have non-numeric characters. if it was the point to actually test it. |
| 13:11.18 | kaldemar | we're shooting your code to peaces. ;) |
| 13:11.43 | jacc0 | ;) |
| 13:11.44 | leifmadsen | kaldemar: it'll work for this guy well enough -- it's not going to get anything too crazy :) |
| 13:12.28 | leifmadsen | I guess I'm doing two checks that are unnecessary though |
| 13:12.39 | jacc0 | ;0 |
| 13:12.47 | *** join/#asterisk ack_syn (~Jedi@unaffiliated/ackz0r) |
| 13:12.51 | ack_syn | hi guys |
| 13:13.02 | jacc0 | hello ack_syn |
| 13:13.03 | ack_syn | hi guys, do you know what is PDD? It is the time between the INVITE and the first MEDIA diferent of a empty noise, right? |
| 13:13.23 | ack_syn | is there something you know to I measure or analyze it? |
| 13:13.36 | ack_syn | something I mean a plugin or a daemon |
| 13:13.37 | jacc0 | tshark ? |
| 13:13.56 | ack_syn | jacc0: the Idea is to measure it in real time and record it in a database |
| 13:14.51 | ack_syn | btw I need to analyze RTP traffic that is where the media itself is |
| 13:15.00 | ack_syn | I dont think tshark could help |
| 13:15.03 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 13:15.36 | jacc0 | tshark -R "sip && rtp" |
| 13:16.16 | ack_syn | jacc0: the point isnt to capture the traffic, for that I use tcpdump or whatever |
| 13:16.27 | ack_syn | the goal is to analyze the media traffic (rtp) |
| 13:17.11 | ack_syn | actually I use media proxy (ag project) wich tell me a PDD based on signaling (from INVITE to SESSION PROGRESS) but sometimes I receibe a SESSION PROGRESS method, but no media yet |
| 13:17.16 | ack_syn | tells * |
| 13:17.23 | anonymouz666 | I hate mediaproxy. |
| 13:17.28 | anonymouz666 | it sux. |
| 13:17.54 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 13:18.23 | anonymouz666 | sometimes playing with kernel-space could be like playing with fire, hehe. |
| 13:18.40 | ack_syn | anonymouz666: ok it doesnt care, I just would like to know if there's a deamon or a simple binary wich could help me doing a media analysis |
| 13:18.55 | ack_syn | anonymouz666: ok, I'd like to know why |
| 13:19.07 | ack_syn | media proxy work fine to me |
| 13:19.22 | anonymouz666 | good to you. some conntrack limitations doesn't work for me. |
| 13:19.30 | ack_syn | I use in my plataform asterisks (signaling only), opensips and media proxy (ag project) |
| 13:19.55 | anonymouz666 | do you want to analysis the RTP that pass-through your mediaproxy? |
| 13:20.03 | ack_syn | anonymouz666: why not? how much simultaneous calls do you have ? |
| 13:20.19 | ack_syn | anonymouz666: yes, I could make a script or something |
| 13:20.42 | anonymouz666 | ack_syn: better, what is the question? maybe I can help you |
| 13:20.46 | ack_syn | I just can't make the media pass through my asterisks because it cause jitters in the calls |
| 13:21.05 | anonymouz666 | mediaproxy and rtpproxy adds jitter too. |
| 13:21.11 | ack_syn | anonymouz666, ok, let me explain it to you |
| 13:21.17 | anonymouz666 | rtpproxy about 5ms |
| 13:21.24 | anonymouz666 | if you lower that, you add CPU load |
| 13:21.31 | ack_syn | anonymouz666: yes, but asterisk increase much than media proxy by my tests |
| 13:21.54 | anonymouz666 | what RTP mode? |
| 13:22.02 | anonymouz666 | Asterisk P2P bridge is really FAST |
| 13:23.14 | ack_syn | anonymouz666: I have many clients that sometimes claim that the PDD is very high (greather than 10s) I'd like to detect it before them (the clients). The media proxy can tell me the PDD based on: from INVITE to the first SESSION PROGRESS. but it doesnt measure when in fact the media starts without empty noise. do you understand anonymouz666 ? |
| 13:23.36 | anonymouz666 | I should use the proxy as signalling |
| 13:23.36 | ack_syn | anonymouz666: ok, I can test it if there's something about my problem wich asterisk helps me |
| 13:24.28 | ack_syn | I have some proxies, some media servers and some asterisk acting like b2bua (only singnaling) where I make some programations about the calls time (Credit) and others |
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| 13:25.24 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:25.49 | ack_syn | anonymouz666: the question maybe is a bit complex |
| 13:25.51 | anonymouz666 | ack_syn: I think what you are trying to accomplish is not so trivial. |
| 13:26.17 | anonymouz666 | you want to make sure that RTP in early media in fact contains "valid stuff" |
| 13:26.42 | anonymouz666 | am I right? |
| 13:27.09 | anonymouz666 | leifmadsen: hello |
| 13:27.23 | ack_syn | anonymouz666: yes, you are right |
| 13:27.48 | ack_syn | or a RINGING or a BUSY TONE, or even a voice, but something != of empty noise |
| 13:28.27 | anonymouz666 | ack_syn: you are in trouble. not so easy, and in my opinion not useful at all. |
| 13:28.38 | ack_syn | It wont increase jitter in the calls since the call doesnt need to wait it be calculated. I just need this information in a database to I plot a graphic |
| 13:28.53 | ack_syn | well, I am sure useful it will be |
| 13:28.54 | anonymouz666 | sometimes the progress tone could take a lit bit and before that you could listen nothing but at this moment the rtp already started |
| 13:29.12 | ack_syn | yes, that's why I need to analyze the rtp traffic |
| 13:29.51 | ack_syn | anonymouz666: I have many graphics to all my carriers, with many informations about them |
| 13:29.55 | leifmadsen | anonymouz666: hello |
| 13:30.05 | ack_syn | now I want to increaase the PDD information, but based in what I told you |
| 13:31.11 | ack_syn | I dont think it is hard like it seems |
| 13:32.46 | ack_syn | I could do something wich could join the packets contaning the rtp traffic untill it is != to a audible noise (different from a empty noise, I can find a pattern range of frequency) |
| 13:33.05 | ack_syn | but before I start doing that I want to know if already exists something like it |
| 13:33.44 | anonymouz666 | leifmadsen: I read in the book about FAXING (pass-through) and the internal timing issues that could make fax a little bit unstable (multiple pages). This issue is Asterisk limitation at the moment or this is a technology limitation sip/rtp + g711 sending the call (LAN) through T1/E1 card |
| 13:33.57 | anonymouz666 | ? |
| 13:34.19 | anonymouz666 | ack_syn: I don't know how to help you futher |
| 13:34.28 | ack_syn | ok |
| 13:35.42 | leifmadsen | anonymouz666: not sure -- I know very little (if anything) about faxing |
| 13:35.56 | leifmadsen | you'll have to bug someone like file for that answer :) |
| 13:36.20 | anonymouz666 | file used to chat more in here |
| 13:36.55 | jacc0 | I've made a very stupid dialplan to reproduce the bridging error |
| 13:36.57 | jacc0 | :P |
| 13:37.12 | jacc0 | will post a link 2 the bug report here later |
| 13:37.21 | leifmadsen | file chats more in #asterisk-scf-dev now because that's the team he is on |
| 13:38.02 | anonymouz666 | ahhh ok |
| 13:42.13 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
| 13:45.47 | *** join/#asterisk ks3 (~ksandy@74.203.195.1) |
| 13:46.05 | jacc0 | @leifmadsen: how can I report an issue in the right category? when I click on report issue I only see categorys: Card detection,General,NewFeature and Service providers/trunks |
| 13:46.15 | leifmadsen | jacc0: you're in the wrong project then |
| 13:46.21 | leifmadsen | upper right corner, select Asterisk |
| 13:46.25 | leifmadsen | not Asterisk-GUI |
| 13:46.44 | jacc0 | okay, ty |
| 13:46.45 | *** join/#asterisk saisoma (~saisoma@client105.jdcc.edu) |
| 13:47.06 | leifmadsen | np |
| 13:47.44 | jacc0 | applicatio/app_bridge is not in the category list : should I select Applicatio/general? |
| 13:48.02 | leifmadsen | there is an apps/app_bridge.c ? |
| 13:48.16 | ack_syn | is there a plugin to make voice analysis in asterisk? |
| 13:48.36 | leifmadsen | ack_syn: no, but you can check out AQuA by http://sevana.fi |
| 13:48.43 | leifmadsen | it's quite good |
| 13:48.54 | *** join/#asterisk Sipster_ (~Sipster@modemcable143.199-202-24.mc.videotron.ca) |
| 13:48.54 | leifmadsen | (we're using it in the testing framework for load testing) |
| 13:49.14 | ack_syn | ok, I will take a look |
| 13:49.51 | ack_syn | I'd like to make analysis in a rtp traffic in real time |
| 13:49.57 | ack_syn | is it possible with that plugin leifmadsen ? |
| 13:50.56 | pigpen | Hi all. Quick question. I deployed an asterisk 1.8.3.3 system without DAHDI. Asterisk had a panic about ever 24 hours. We then re-compiled it with DAHDI. Now it runs fine. |
| 13:51.02 | pigpen | We are also seeing: Apr 27 22:30:53 fw kernel: dahdi: Detected time shift. |
| 13:51.26 | pigpen | every hour or so. |
| 13:51.37 | pigpen | So, is DAHDI still required?? |
| 13:52.20 | leifmadsen | ack_syn: anaylsis of RTCP data is only as good as the RTCP data you get -- and asterisk has some issues with it. Check out the http://svn.asterisk.org/svn/asterisk/team/oej/pinefrog-trunk/ for RTCP stuff. If you want to analyze the RTP directly, check out the AQuA site. It's not a "plugin" but rather a toolkit you can use to build data analysis toosl. |
| 13:52.21 | leifmadsen | tools* |
| 13:52.35 | leifmadsen | pigpen: well you still need a timing module of some sort |
| 13:52.55 | leifmadsen | pigpen: did you have some sort of res_timing_* module loaded? |
| 13:53.18 | ack_syn | leifmadsen: nice, I will take a look. |
| 13:53.18 | leifmadsen | Honestly, I still use DAHDI for timing because I find it the most stable, and it only takes a few minutes to compile. |
| 13:53.32 | ack_syn | to be honest I dont use asterisk to my media, I use media proxy (ag project) hehe |
| 13:53.37 | pigpen | leifmadsen,: res_timing_timerfd.so Timerfd Timing Interface 1 |
| 13:53.38 | *** join/#asterisk golam (4531720a@gateway/web/freenode/ip.69.49.114.10) |
| 13:53.47 | pigpen | leifmadsen, yes sir. |
| 13:53.55 | pigpen | before and after dahdi is loaded. |
| 13:53.57 | leifmadsen | pigpen: use timerfd if it works -- if it is unstable, use DAHDI |
| 13:54.17 | pigpen | sorry, dahdi is not "loaded" it is compiled. |
| 13:54.39 | pigpen | meaning, dahdi is compiled only, module not loaded |
| 13:55.00 | pigpen | the timerfd is the only timer previous loaded and is still currently the only timer loaded. |
| 13:55.39 | pigpen | Just curious .....not a big deal, I just thought because of these new timers it removed the necessity to even compile dahdi. |
| 13:55.51 | leifmadsen | it does |
| 13:56.04 | leifmadsen | if they are not stable on your platform though, then it seems silly to use them :) |
| 13:56.14 | leifmadsen | timerfd is a kernel module |
| 13:56.18 | pigpen | heh...yeah. It may be a 32bit thing. |
| 13:56.19 | leifmadsen | or part of the kerne somehow |
| 13:56.22 | leifmadsen | who knows |
| 13:56.30 | pigpen | right, fd is kernel, pt is glibc right? |
| 13:56.35 | leifmadsen | I don't find compiling dahdi to be a burden so I just do it regardless |
| 13:56.40 | leifmadsen | something like that |
| 13:56.49 | leifmadsen | pthread timing is not very good |
| 13:56.59 | leifmadsen | it works for some people, but I see a lot of issues around it |
| 13:57.06 | leifmadsen | if you have the option of using dahdi, then use it |
| 13:57.12 | leifmadsen | it'll give you the most reliable module |
| 13:57.20 | pigpen | sure...but since I work with a kernel dev, he feels if it isn't needed or used, why compile it. And yeah, fd is preferred. |
| 13:57.27 | leifmadsen | only people using things like OSX should really be using an alternate timing module in my opinion |
| 13:57.35 | pigpen | but like I said, dahdi is only compiled, not loaded. |
| 13:57.41 | leifmadsen | pigpen: but it *is* needed if you want it rock solid :) |
| 13:57.50 | pigpen | heh. hopefully. |
| 13:57.51 | leifmadsen | that is my opinion on the matter |
| 13:57.53 | pigpen | time will till. |
| 13:58.00 | pigpen | s/till/tell |
| 13:58.06 | leifmadsen | people get a little too uppity and make things much more complicated than they need to be |
| 13:58.29 | pigpen | yeah, it is nice to keep a clean kitchen, but if it means you can't cook, then get it messy. |
| 13:59.06 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 13:59.06 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:59.06 | leifmadsen | it's like asking someone to use a skillet to make a roast :) |
| 13:59.17 | leifmadsen | use the tools that are right and appropriate for the job |
| 13:59.27 | leifmadsen | even if it means you can't put the pan in the dishwasher at the end |
| 14:05.19 | *** join/#asterisk rogersja (~RogersJa@S0106000f6695039f.gv.shawcable.net) |
| 14:07.46 | jacc0 | filed a bug report about the bridging error : https://issues.asterisk.org/view.php?id=19203 |
| 14:08.52 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 14:09.38 | ack_syn | hey is there a plugin to asterisk make voice quality monitoring? or maybe another daemon Idk |
| 14:15.38 | jacc0 | http://www.voipmonitor.org/ maybe |
| 14:16.08 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 14:16.39 | ack_syn | ok |
| 14:23.23 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 14:30.33 | jacc0 | almost weekend!! 30minutes :) |
| 14:30.33 | rogersja | leifmadsen: solved the ISN issue with insecure=invite :s thanks for the help!:) |
| 14:30.43 | leifmadsen | no problem! |
| 14:30.50 | leifmadsen | jacc0: heh, only 10:30am here ;) |
| 14:30.53 | leifmadsen | so ya, 30 mins sounds right |
| 14:31.13 | jacc0 | tomorrow it's queensday here |
| 14:31.20 | jacc0 | so one-big-party |
| 14:31.28 | jacc0 | I'm looking forward to it |
| 14:31.32 | jacc0 | :0 |
| 14:32.37 | ack_syn | jacc0: do you know it? the voip monitor? have u tested it? |
| 14:34.51 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:42.06 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:42.14 | *** join/#asterisk sigius (~sigius@93-125-185-45.dsl.alice.nl) |
| 14:42.18 | sigius | h |
| 14:42.59 | jacc0 | @ack_syn: never used it, someone came into this channel some days ago talking about it |
| 14:43.06 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 14:43.31 | jacc0 | @ack_syn: I'm planning on looking into this for some projects where we have audio quality problems |
| 14:44.57 | ack_syn | jacc0: right, I'm about to test it. the guy, did he talk something good or bad about it? |
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| 14:46.49 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-4.saucontech.com) |
| 14:47.12 | jacc0 | @ack_syn: he was just asking if anyone even used it :p |
| 14:47.18 | jacc0 | *ever |
| 14:47.30 | fullstop | Hi. I'm trying to connect a metaswitch provider with Asterisk. They require md5 digest auth on outbound calls, but do not require any authentication for inbound calls. |
| 14:47.47 | ack_syn | ok :T |
| 14:47.49 | fullstop | For some reason, I'm having a hard time with this. |
| 14:48.00 | fullstop | I have outbound calls (the digest auth) working |
| 14:48.01 | fullstop | but |
| 14:48.22 | fullstop | inbound calls are failing, and it looks like asterisk is sending a 401 unauthorized because it is expecting digest auth. |
| 14:48.59 | sigius | How should I destablish origin of calls to my US DID. When I get a call from Baltimore the CallerID on my asterisk start with 30.., when I have a call from Greece it also start with 30. So how can I tell where it came from ? |
| 14:54.22 | jacc0 | have a nice weekend all!!!!!!! |
| 14:54.24 | jacc0 | bye |
| 14:55.27 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 14:55.35 | *** join/#asterisk ixyd_ (~denzs@carbon.gonicus.de) |
| 14:55.42 | ixyd_ | hi guys |
| 14:56.14 | ixyd_ | does anyone have a working example for executing a sub beforee the caller gets conencted by a queue? |
| 14:56.32 | ixyd_ | like parameter U() for dial the recent version of Queue() supports calling a sub too |
| 14:56.42 | ixyd_ | but iam not sure about the correct syntax?! |
| 14:56.54 | ixyd_ | i tried the same like i use for Dial U() |
| 14:56.59 | ixyd_ | but it doesnt work :( |
| 15:10.42 | *** join/#asterisk snigavig (~snegovik@72-58-112-92.pool.ukrtel.net) |
| 15:11.16 | snigavig | Hi Guys, I have a question about asterisk and t38, can anybody help me? |
| 15:13.27 | ChannelZ | ~ask |
| 15:13.27 | 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. |
| 15:14.11 | Freeaqingme | I want to determine on a per call basis whether or not CLID should be enabled. Does that mean that I need to add the same context twice per context? |
| 15:16.12 | snigavig | ok, sorry, I need to decrease the T38MaxBitRate field in INVITE's SDP part. It is 14400 now but the gateway needs it to be 9600 or lower. I tried to change it in the udptl.conf, but it seems not to work...The question is, how to define the T38MaxBitRate? |
| 15:19.20 | ixyd_ | or maybe my question should be, does the gosub parameter of Queue support parameters? |
| 15:21.08 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
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| 15:28.01 | fullstop | any reason why "sip reload" wouldn't update the context of a sip user? |
| 15:28.13 | fullstop | I changed it, but it is still routing calls to the old context. |
| 15:28.32 | sxpert | I've had issues with that in the past. dunno if stuff was fixed |
| 15:28.39 | sxpert | and probably depends on your version |
| 15:28.53 | fullstop | bummer |
| 15:28.59 | fullstop | 1.6.2.11 right now |
| 15:29.31 | snigavig | I need to decrease the T38MaxBitRate field in INVITE's SDP part. It is 14400 now but the gateway needs it to be 9600 or lower. I tried to change it in the udptl.conf, but it seems not to work...The question is, how to define the T38MaxBitRate? |
| 15:30.08 | *** join/#asterisk karmicdude (~chatzilla@223.141.102.97.cfl.res.rr.com) |
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| 15:38.13 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:39.58 | fullstop | Also, I am connecting to a metaswitch, which expects digest auth for outbound calls but no authentication for inbound calls. |
| 15:40.02 | fullstop | I eventually got this working.. |
| 15:40.34 | fullstop | but I did it by creating the peer first and then the user in sip.conf. I think that if I reverse the entries that it will not work. |
| 15:42.04 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
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| 15:51.21 | gruvfunk | greetings all! easiest way to upload/push recordings to a remote server? |
| 15:51.42 | gruvfunk | (during an IVR session, say when the caller hangs up or just prior) |
| 15:52.10 | gruvfunk | I've tried a System call to ncftpput but that didnt' work |
| 15:52.34 | sxpert | probably because it blocks stuff |
| 15:52.47 | Tozz_ | try ncftpbatch |
| 15:53.04 | Tozz_ | and then cron ncftpbatch once every 5 mins or so |
| 15:53.05 | gruvfunk | a manual ncftpput works fine |
| 15:53.12 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 15:55.05 | *** join/#asterisk jovandeb (~jovan@host219-228-static.22-87-b.business.telecomitalia.it) |
| 15:55.12 | jovandeb | Hi to all |
| 15:55.41 | jovandeb | Are there any negotiation codec patch for Asterisk 1.6.2? |
| 15:57.11 | jovandeb | I have a Voip phone with Ulaw and G729 codec support. I have two outgoing trunk: 1 DAHDI channel and 1 SIP Provider that only support g729 |
| 15:57.48 | jovandeb | I'd like to avoid codec transaction when I call from Voip Phone using SIP Provider trunk |
| 15:58.24 | jovandeb | and force pass through when I use SIP trunk :) |
| 15:59.01 | jovandeb | SIP_CODEC and SIP_CODEC_OUTBOUND did not solve |
| 15:59.22 | Tozz_ | why not just force the phone to use g729? |
| 16:00.05 | jovandeb | Because If I foce g729 on the phone I have codec translaction when outgoing line is DAHDI |
| 16:01.11 | jovandeb | I need a sort of codec negotiation before Dial ... |
| 16:01.35 | jovandeb | and force SIP device to use its own g729 codec ... |
| 16:02.39 | jovandeb | now I have disallow=all and allow=g729 on SIP trunk |
| 16:03.04 | Tozz_ | sounds right |
| 16:03.08 | jovandeb | and disallow=all allow=ulaw allow=alaw allow=g729 on SIP phone |
| 16:03.55 | jovandeb | but I need something able to choose between these codecs ... |
| 16:05.37 | *** join/#asterisk logicwrath_ (~no@c-68-62-24-205.hsd1.mi.comcast.net) |
| 16:09.01 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 16:09.30 | gruvfunk | say does anyone know if ncftpbatch processes the spool in order? I need a specific file sent last... |
| 16:13.35 | jovandeb | gruvfunk: ncftpbatch is for batch ftp? |
| 16:13.59 | gruvfunk | believe so |
| 16:14.05 | gruvfunk | as mentioned above |
| 16:14.42 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
| 16:15.53 | jovandeb | gruvfunk: The jobs are spool files written to a user's $HOME/.ncftp/spool directory and have a special format and file-naming convention (which contains when the job is to be run) |
| 16:16.15 | jovandeb | look at file-naming convention :) |
| 16:16.57 | gruvfunk | great, problem is - I see Asterisk calling up System ncftpput -bb and the ncftpbatch, but I don't see the files actually uploading to the destination (outside asterisk, works fine) |
| 16:18.16 | *** join/#asterisk MrTelephone (~MrTelepho@h697179-171.picriverisp.net) |
| 16:19.47 | MrTelephone | Is there something I can troubleshoot for not getting a ring on long distance calls? The remote party eventually picks up but the ringing sometimes doesn't work. The calls for going through the telephone company PRI. |
| 16:20.10 | gruvfunk | Am I unable to use an Asterisk variable I set myself within the System call? |
| 16:20.22 | jovandeb | MrTelephone: use r option in Dial |
| 16:21.03 | MrTelephone | It used to ring without the r option. I wonder why it doesn't now :( |
| 16:21.38 | jovandeb | gruvfunk: I don't know I have never use ncftp ... I read that ncftpput needs ncftpbatch istance running |
| 16:22.53 | gruvfunk | jovandeb: ncftpput works beautifully from command line or shell script, without batching -- to use ncftpbatch, you have to tell ncftpput -bb (which I've tried also) - again works beautiful from command line or script) |
| 16:24.19 | gruvfunk | I'm doing this from IVR: seems the variable doesn't pass... System(/usr/bin/ncftpput -b -f /tmp/my.ftp.creds / /usr/share/asterisk/sounds/recordings/${file}) |
| 16:24.51 | jovandeb | maybe permission problem? |
| 16:25.53 | gruvfunk | can somebody verify that I can pass my own set variable to System ? |
| 16:27.45 | jovandeb | post Set string |
| 16:28.48 | jovandeb | You can try to set command string before ... |
| 16:28.51 | jovandeb | for example |
| 16:28.52 | golam | how can I write the command out from CLI to a file. example #CLI> core show version ---> log.txt |
| 16:28.53 | jovandeb | ;;exten => _0Z.,1,Set(filename=${EXTEN}_${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}_${CALLERID(num)}.wav) |
| 16:28.56 | jovandeb | ;;exten => _0Z.,n,MixMonitor(${filename}|v(0)V(0)) |
| 16:29.09 | jovandeb | with NoOp |
| 16:29.12 | MrTelephone | I'm getting EVENT_RINGING On the PRI but the phones are not ringing. I guess the sip message is not making it to the phone |
| 16:29.31 | jovandeb | you can debug dialplan and print string with NoOp |
| 16:29.39 | *** join/#asterisk grayhame (~chatzilla@74-94-250-169-Nashville.hfc.comcastbusiness.net) |
| 16:30.09 | jovandeb | for example NoOp(${CALLERID(num)}) |
| 16:31.29 | jovandeb | Someone able to help me with codec transaction :) ? |
| 16:33.44 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 16:34.11 | *** join/#asterisk sourcode (~code@ppp-61-90-15-188.revip.asianet.co.th) |
| 16:34.37 | grayhame | hello. i have a carrier telling me that my pbx (asterisk, sangoma AFT-A101 PRI) has started sending out tons of line code and path code violations but I can't find anything to indicate a problem. |
| 16:34.47 | grayhame | does anyone have a recommendation for determining what's causing the violations to start up? |
| 16:39.46 | *** join/#asterisk el3slave (~email@ip68-4-133-145.oc.oc.cox.net) |
| 16:41.01 | MrTelephone | I changed long distance providers for our PRI and ever since then I've been having issues with long distance services. |
| 16:41.47 | MrTelephone | I figured it didn't matter who the provider was but I guess their equipment is shotty |
| 16:43.17 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 16:44.32 | jovandeb | Maybe I need something like that http://www.rtpproxy.org/wiki/AsteriskCodecNegotiationPatch |
| 16:51.08 | snigavig | Hi All, I need to decrease the T38MaxBitRate field in INVITE's SDP part. It is 14400 now but the gateway needs it to be 9600 or lower. I tried to change it in the udptl.conf, but it seems not to work...The question is, how to define the T38MaxBitRate? |
| 16:53.51 | *** join/#asterisk snigavig (~snegovik@72-58-112-92.pool.ukrtel.net) |
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| 16:59.52 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:00.13 | keith4 | MrTelephone: as in "full of shot"? |
| 17:07.02 | MrTelephone | yeah, maybe it could be an asterisk problem but the sound quality seems kind of poor |
| 17:07.57 | MrTelephone | I set the max bitrate to 9600 as well by setting FAXOPT but it still wants to connect at 14400. I'm not sure if that is because the remote device forces it to 14400. |
| 17:08.08 | *** join/#asterisk snigavig (~snegovik@72-58-112-92.pool.ukrtel.net) |
| 17:16.08 | gruvfunk | got my ftp issue resoved, thanks jovandeb for making me pay closer attention to my Set(s) |
| 17:16.41 | gruvfunk | can indeed use our own variables in System |
| 17:17.17 | gruvfunk | another beautiful example of why I love working with Asterisk so much - any challenge has at least one viable solution (more than one way to skin a badger) |
| 17:19.25 | *** join/#asterisk snigavig (~snegovik@72-58-112-92.pool.ukrtel.net) |
| 17:19.26 | gruvfunk | of course, two important factors: 1) time and 2) this channel |
| 17:19.57 | MrTelephone | Gruvfunk, what are you ftping? |
| 17:20.29 | gruvfunk | MrTelephone: call recordings as part of a larger IVR system |
| 17:21.28 | MrTelephone | nice |
| 17:21.50 | gruvfunk | just one of many projects |
| 17:22.05 | MrTelephone | I hear you. I'm swamped |
| 17:22.13 | gruvfunk | it's a good thing |
| 17:23.57 | *** join/#asterisk b0ot (~Jinxed---@147.177.57.74) |
| 17:24.04 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 17:24.04 | b0ot | is there a more generic voip help room? |
| 17:24.54 | gruvfunk | MrTelephone and snigavig - I have set my FAXOPT(minrate)=2400 |
| 17:25.23 | snigavig | gruvfunk, how did you do it? |
| 17:25.32 | gruvfunk | maxrate=14400 |
| 17:25.49 | gruvfunk | Set(FAXOPT(maxrate)=14400) ; Set(FAXOPT(minrate)=2400) |
| 17:25.56 | leifmadsen | b0ot: not that I'm aware of |
| 17:26.04 | MrTelephone | Did you try 9600? |
| 17:26.11 | snigavig | Where should I put it? Could you please be more specific? |
| 17:26.11 | b0ot | alright thanks leifmadsen |
| 17:26.32 | gruvfunk | I had issues with any other minrate.. but maybe due to the fact that my fax machine is on a GV line... dunno |
| 17:26.48 | gruvfunk | or could be the ATA sitting between Fax machine and * |
| 17:27.11 | gruvfunk | snigavig: describe your platform, are you working in the .conf files? |
| 17:27.33 | snigavig | gruvfunk, yes, only conf files or CLI, no web interface... |
| 17:28.10 | gruvfunk | right, so create a context like [outboundfax] |
| 17:28.44 | snigavig | I tried to put "T38MaxBitRate = 9600" line to the udptl.conf, but it did not work.. |
| 17:29.33 | gruvfunk | I'm talking extensions.conf, create a context for what you want to do (process faxes) |
| 17:29.34 | *** join/#asterisk fireman_biff (~biff@65.48.133.102) |
| 17:29.53 | gruvfunk | in it, I set my FAXOPT(min and maxrate |
| 17:31.34 | snigavig | hm, I think it will not work for me, I do not really use extensions.conf, everything is done in a self-made AGI daemon.. |
| 17:31.58 | snigavig | is there another way to do it? Like a static option in the config like udptl.conf? |
| 17:32.01 | gruvfunk | and how do you call that AGI? not from within a context? |
| 17:33.05 | snigavig | it's done within context, but this agi daemon is one for the whole system, and it determines if it is a fax or voicemail or call queue etc. |
| 17:33.27 | snigavig | so I can not add anything to the extensions.conf |
| 17:34.14 | snigavig | but anyway thank you very much for your help, now I know how to set it, so I'm closer to my goal! Thank you ! |
| 17:35.25 | fireman_biff | hi, can anybody recommend an online training program that covers freepbx and the basics of asterisk itself (troubleshooting etc) |
| 17:36.12 | *** join/#asterisk hairyraven (~nobody@95.72.54.173) |
| 17:38.22 | leifmadsen | fireman_biff: the only online training I'm aware of is the new program by Digium |
| 17:39.01 | *** join/#asterisk coppice (~chatzilla@62.166.232.220.dyn.pacific.net.hk) |
| 17:39.05 | fireman_biff | leifmadsen: I'm actually checking that out now but I'm not seeing any mention of freepbx, do you know if they include this or not? |
| 17:39.26 | leifmadsen | I doubt it |
| 17:39.35 | leifmadsen | for FreePBX you'll probably have to buy a book |
| 17:39.39 | *** join/#asterisk phl4kx (~phl4kx@190.223.54.18) |
| 17:39.41 | phl4kx | hi all |
| 17:39.48 | leifmadsen | maybe the freepbx people offer training on it... not sure |
| 17:39.54 | fireman_biff | leifmadsen: oh ok, thanks |
| 17:40.03 | leifmadsen | I'm not sure what it would entail... it's pretty point-and-click :) |
| 17:40.04 | phl4kx | I need some solution in asterisk for listening calls anytime like using Flash Operator, any idea? |
| 17:40.09 | gruvfunk | fireman_biff maybe ask in #freepbx |
| 17:40.11 | *** join/#asterisk pgrace (~pgrace@hermes.vsix.me) |
| 17:40.15 | leifmadsen | phl4kx: do you actually mean "listening" ? |
| 17:40.27 | leifmadsen | like, hearing the audio? |
| 17:40.30 | phl4kx | yes |
| 17:40.33 | phl4kx | hearing the audio |
| 17:40.37 | phl4kx | enter in a conversation |
| 17:40.39 | leifmadsen | core show application ChanSpy |
| 17:40.51 | leifmadsen | core show application ExtenSpy |
| 17:40.52 | gruvfunk | leif is super quick |
| 17:40.59 | leifmadsen | types at 80wpm :) |
| 17:41.06 | therawr | slow |
| 17:41.09 | pgrace | Hey, anyone from digium here who can answer a question? I'm wondering how the licensing for the skype channel driver works; if I buy two licenses now and choose to add 20 more later, do I need to get a new license code for 22 total licenses or can you add license keys at will? |
| 17:41.15 | leifmadsen | therawr: I'm only slow in bed |
| 17:41.19 | *** join/#asterisk w4llnu55 (~w4llnu55@89.204.153.74) |
| 17:41.24 | therawr | leifmadsen: 80 wpm? psh |
| 17:41.28 | phl4kx | some console for easy access for my SALES MANAGER? |
| 17:41.32 | gruvfunk | leif's cpu processing speed has more to do with it than just wpm typed |
| 17:41.32 | therawr | if you're not 100+ then you're slow, grandpa |
| 17:41.35 | leifmadsen | pgrace: I doubt it -- Digium HQ is offline except for critical infrastructure |
| 17:41.42 | MrTelephone | SKYPE. will customers go for that? |
| 17:41.48 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 17:41.50 | phl4kx | my sales manager like to hear the audio of the sales anytime, some solution? |
| 17:41.52 | leifmadsen | therawr: I'm saying 80 wpm with high accuracy :) |
| 17:41.57 | therawr | hahahaha |
| 17:42.03 | leifmadsen | phl4kx: the solution is what I just described |
| 17:42.07 | phl4kx | thanks |
| 17:42.08 | pgrace | leif: oh, right, crap, the tornados. Everyone OK? Anyone heard? |
| 17:42.17 | phl4kx | have a GUI access like Flash Operator? |
| 17:42.23 | leifmadsen | create something for your SALES MANAGER (why are we yelling?) that is easy |
| 17:42.29 | pgrace | mrtelephone: in our case, we'll be using it as a backdoor into our Lync conferences :) |
| 17:42.31 | leifmadsen | phl4kx: no, GUI access is not the job of Asterisk |
| 17:42.45 | pgrace | some people in our organization DEMAND skype and we can't get them on Lync yet due to licensing so... skype! |
| 17:42.54 | leifmadsen | pgrace: everyone seems to be fine, but no power for about 5 more days at the current estimate |
| 17:43.00 | MrTelephone | I'd like to offer skype numbers to our clients for a fee per month. That would be awesome. |
| 17:43.12 | pgrace | leif: gotcha, so any orders through the digium website would also likely be delayed |
| 17:43.16 | leifmadsen | pgrace: I think you just add multiple license keys |
| 17:43.22 | phl4kx | some solution with GUI access?, its going to be difficult for access to my SALES manager |
| 17:43.28 | MrTelephone | I'm afraid skype is losing popularity because there are some other crappy clients out there now. |
| 17:43.28 | leifmadsen | pgrace: that's how G729 works and is what I would expect for all the other modules |
| 17:43.51 | leifmadsen | phl4kx: go ahead and build it -- otherwise you need to create some sort of java or activeX based interface to communicate with asterisk |
| 17:43.57 | pgrace | leif: cool, thanks for the info. |
| 17:44.17 | phl4kx | thanks friend |
| 17:44.24 | *** part/#asterisk fireman_biff (~biff@65.48.133.102) |
| 17:44.34 | phl4kx | Some Solution for a CALL Center for SALES ??? |
| 17:44.49 | leifmadsen | stop using caps in your sentences |
| 17:45.10 | leifmadsen | phl4kx: you're not asking the right questions because I have no idea what it is you're expecting |
| 17:45.22 | leifmadsen | asterisk does provide an interface to listen to calls -- it's called ChanSpy() |
| 17:45.28 | leifmadsen | if you need a web interface, you will need to build it |
| 17:45.33 | leifmadsen | that is the bottom line |
| 17:45.43 | phl4kx | sorry |
| 17:45.52 | karmicdude | what if I lol really loud, can I use LOL in caps? |
| 17:45.54 | phl4kx | thanks friend |
| 17:45.55 | phl4kx | I understand |
| 17:45.59 | leifmadsen | LOL |
| 17:46.00 | phl4kx | sorry |
| 17:46.01 | karmicdude | :) |
| 17:46.02 | leifmadsen | that is acceptable |
| 17:46.05 | phl4kx | hahaha |
| 17:46.18 | leifmadsen | but you DON'T need to capitalize CERTAIN work for effect. |
| 17:46.23 | leifmadsen | s/work/words/ |
| 17:46.29 | MrTelephone | you guys chew up a lot of bandwidth for some questionable material, that's for sure. |
| 17:46.41 | leifmadsen | dances around the questionable material |
| 17:46.42 | malcolmd | for affect, use * characters, like you were bolding something. e.g. that is *so* lame |
| 17:46.42 | phl4kx | lets, my solution is Flash Operator + ChanSpy, another program for add to my soution???? I like to implement a call center |
| 17:46.50 | MrTelephone | Where's the next astricon |
| 17:47.02 | leifmadsen | MrTelephone: see www.astricon.net |
| 17:47.02 | malcolmd | denver |
| 17:47.04 | leifmadsen | (Denver) |
| 17:47.10 | MrTelephone | are you guys going? |
| 17:47.25 | leifmadsen | I'll probably be on a cruise ship for my honeymoon |
| 17:47.29 | leifmadsen | depends on when we leave |
| 17:47.40 | MrTelephone | I went to key west last month, I recommend it |
| 17:47.44 | leifmadsen | it'll be the first astricon I'll have missed since 2004 |
| 17:47.46 | malcolmd | yop |
| 17:47.46 | coppice | sounds like a honeymoon for old people |
| 17:47.58 | leifmadsen | coppice: I'm 57... |
| 17:48.05 | MrTelephone | Get a room at the Casa Marina Resort for your honeymoon |
| 17:48.28 | malcolmd | leifmadsen: you're a very spry 57 |
| 17:48.36 | gruvfunk | <PROTECTED> |
| 17:48.42 | coppice | leifmadsen: i'm 56, but I'd need to be a lot older to go on a cruise for my honeymoon |
| 17:48.54 | egeste | brb, #oldguyhoneymoon |
| 17:48.57 | MrTelephone | I want to get married there too. Rent a scooter and fly off the pier james bond style. |
| 17:49.06 | gruvfunk | funny, we stayed at the Reach Resort (sister to Casa Marina) |
| 17:49.21 | leifmadsen | malcolmd: yes, I'm quite sprightly |
| 17:49.35 | MrTelephone | We stayed at the days inn because we blew all our money in orlando. :( $200/night for a crappy room |
| 17:49.59 | coppice | so you couldn't afford to stay nights? |
| 17:50.15 | MrTelephone | You ever come back from holidays and request a raise from your company because you realize how prices are over inflated everywhere? |
| 17:50.20 | carrar | w00t |
| 17:50.47 | MrTelephone | Couldn't afford to stay more than 1 night yeah |
| 17:51.04 | MrTelephone | Astricon 2012 Location : Key Largo |
| 17:51.08 | karmicdude | what about Vegas? you can hire Elvis and all |
| 17:51.57 | phl4kx | friends |
| 17:51.59 | MrTelephone | How about getting married by people dressed in star trek uniforms instead? |
| 17:52.08 | phl4kx | some solution for Reports Call in asterisk? |
| 17:52.58 | gruvfunk | I know, way off topic, but when we wed on beach, there was a Pirate convention going on |
| 17:53.09 | gruvfunk | arrrrghh |
| 17:53.54 | gruvfunk | phl4kx you may wish to visit #freepbx instead - they have a GUI for Call Reports, etc |
| 17:54.26 | phl4kx | thanks |
| 17:54.48 | gruvfunk | or if callcenter, check http://www.asternic.biz/ |
| 17:55.01 | gruvfunk | I use that for a customer with Queues |
| 17:56.16 | *** part/#asterisk snigavig (~snegovik@72-58-112-92.pool.ukrtel.net) |
| 17:56.25 | MrTelephone | gruvfunk, that is cool. A friend of mine considered getting married on a pirate ship (rented) in toronto. Who would want to do that? |
| 17:56.34 | citywok | happy birthday to me! |
| 17:56.55 | gruvfunk | hb |
| 17:57.00 | karmicdude | hb |
| 17:57.27 | coppice | is it still pirate if its rented? don't you have to steal it? |
| 17:57.46 | MrTelephone | Someone restored an old pirate ship and they rent out for special occasions |
| 17:57.48 | gruvfunk | lol |
| 17:57.56 | coppice | did it have a somali crew? |
| 17:58.06 | gruvfunk | pretty neat, I wonder if they use asterisk inside as a communications system |
| 17:58.26 | MrTelephone | Don't get me started on somalians. Forgive me for saying but that country has got to go |
| 17:58.47 | MrTelephone | Can we not photoshop in a boat into the asterisk logo? |
| 18:00.43 | coppice | a replica of the HMS Bounty, made for the movie, stands just down the road from here (though not actually in the road :-\ ), but I've never seen them unfurl the sails. kinda sad. it would look great in full sail |
| 18:01.07 | MrTelephone | Where do you live? |
| 18:01.19 | coppice | right VVV here |
| 18:01.48 | coppice | http://www.coppice.org/DiscoveryBay.jpg |
| 18:02.04 | MrTelephone | pri related question. I did a trace on my pri and asterisk is sending out two identify frames everytime there is something to send. Any ideas? |
| 18:02.05 | *** join/#asterisk galaxywatcher (~galaxywat@pdpc/supporter/active/galaxywatcher) |
| 18:03.17 | gruvfunk | fireman_biff fwiw, I just found this perusing the nets http://www.freepbx.org/open-telephony-training-seminar |
| 18:03.36 | *** join/#asterisk hairyraven (~nobody@95.72.54.173) |
| 18:16.33 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 18:16.51 | fullstop | Can asterisk handle different sip settings for inbound vs outbound calls, or should I be looking into something like Kamailio? |
| 18:19.19 | leifmadsen | yes |
| 18:19.22 | leifmadsen | peer vs user |
| 18:22.52 | *** join/#asterisk knorkeknie (~hans@p5496E30B.dip.t-dialin.net) |
| 18:23.30 | *** join/#asterisk Z_God (~julius@80.187.212.63) |
| 18:25.11 | gruvfunk | is Dahdi just for HW support now in *1.8? or does it still support conferences and things? |
| 18:25.37 | *** join/#asterisk logicwrath (~no@mail.vistitude.com) |
| 18:26.27 | karmicdude | openser, opensis, kamilio. are they all the same or is one better than the other? |
| 18:26.42 | karmicdude | s/opensis/opensips/ |
| 18:29.50 | leifmadsen | gruvfunk: you can use DAHDI for timing still (res_timing_dahdi) or you can use res_timing_timerfd, or res_timing_pthread (reliability goes in about that order) |
| 18:29.59 | leifmadsen | I still use res_timing_dahdi because I find it the most stable |
| 18:30.16 | gruvfunk | righton thx |
| 18:30.19 | leifmadsen | some people have luck with timerfd, and SOME have luck with pthread, but I'd say they are the minority |
| 18:30.43 | leifmadsen | unless you're using a machine that can't load a module into the kernel (OSX for example) I recommend using dahdi |
| 18:30.57 | *** part/#asterisk pgrace (~pgrace@hermes.vsix.me) |
| 18:30.59 | coppice | timerfd can do some nice things for you, if you're nice to it |
| 18:31.41 | fullstop | leifmadsen: Can you explain a little bit more w/peer vs user? I've tried that, unsuccessfully. |
| 18:39.41 | fullstop | Am I correct in thinking that peer is for outbound and user is for inbound ? |
| 18:41.47 | *** join/#asterisk l1nuxman (~l1nuxman@CPE0021296828b2-CM00111ae6f860.cpe.net.cable.rogers.com) |
| 18:46.22 | leifmadsen | fullstop: yes |
| 18:48.09 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 18:48.56 | *** join/#asterisk nikip (~k@dslb-094-217-080-228.pools.arcor-ip.net) |
| 18:50.44 | fullstop | both user and peer have the same ip address and port, and it looks like asterisk is picking the peer for inbound calls. |
| 18:51.25 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:51.56 | leifmadsen | yes, because something is causing the 'user' to not match |
| 18:52.12 | leifmadsen | disable the peer for now and get the user to match first for incoming calls -- the peer (matching on IP) is the failover |
| 18:52.20 | leifmadsen | so if it is getting that far, it's not matching on the user |
| 18:52.28 | fullstop | let me make another call and see what user it is sending.. if any. |
| 18:52.48 | leifmadsen | I'm finishing building a system then I have to leave in about 8 mins, so I probably won't be able to respond anymore |
| 18:55.09 | *** join/#asterisk jdoe (jdoe@falseprophet.ca) |
| 18:56.26 | fullstop | I don't get it.. why does it attempt to find a peer for an _inbound_ call? |
| 18:57.11 | fullstop | I get "No matching peer for '3842412500' from '72.88.66.92:5600' |
| 18:58.12 | leifmadsen | fullstop: like I said, it's trying to match the IP |
| 18:58.33 | fullstop | leifmadsen: is '3842412500' the username it is trying to match? |
| 18:58.49 | fullstop | and, because it does not find it, it goes to ip? |
| 19:01.43 | *** join/#asterisk volker- (~volker@84.201.4.133) |
| 19:01.45 | volker- | hi |
| 19:02.23 | l1nuxman | I'm trying to hook up my analog landline to analog phones in my house and also to my PC with asterisk with a FXO ATA adapter. Should I connect like this?: landline->FXO LAN PORT->ROUTER->MODEM ? OR: landline->(FXO WAN PORT->ROUTER & FXO LAN PORT->ASterisk->ROUTER)...I dunno how can I do it? Right now I have a set up Landline->Analog Phones & Cable Modem->Router->PC Asterisk |
| 19:02.27 | volker- | i googled for a while but didnt find an actual (not 2005) answer: is it possible that one useraccount connects two times and calls gets forwarded to both ip registered from? |
| 19:07.26 | *** join/#asterisk l1nuxman (~l1nuxman@CPE0021296828b2-CM00111ae6f860.cpe.net.cable.rogers.com) |
| 19:08.51 | ariel_ | ping tzafrir question about setting up a astrixbank as a T1 CAS, with Dahdi 2.2.1 it always wants to see it as a PRI |
| 19:10.34 | *** join/#asterisk ananian (~ananian@mail.libertynet.com.br) |
| 19:29.53 | *** part/#asterisk nikip (~k@dslb-094-217-080-228.pools.arcor-ip.net) |
| 19:32.02 | jaytee | I don't need DAHDI installed for * 1.8 if I'm not going to use conferencing and all calls are SIP only, right? |
| 19:33.14 | jaytee | and conf_bridge uses the system timer, not DAHDI_DUMMY so I could still use that app if I needed to. |
| 19:38.51 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 19:43.45 | volker- | bye |
| 20:01.48 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 20:07.52 | *** join/#asterisk cerberus_za (~coert@196-215-103-120.dynamic.isadsl.co.za) |
| 20:09.16 | *** join/#asterisk eerie (~mime@gateway/shell/bshellz.net/x-axuyhzuahyejodml) |
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| 20:22.26 | sunfone | anyone have any pointers for Cacti templates for call traffic? |
| 20:24.29 | gruvfunk | Asterisk can support "multi-tenant", right? I'm thinking contexts to contain each "tenant" ? |
| 20:29.33 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 20:30.50 | sunfone | gruvfunk: sure thing |
| 20:31.10 | gruvfunk | thx |
| 20:32.01 | *** join/#asterisk powerunits (~exe@119.152.41.240) |
| 20:35.26 | powerunits | http://www.facebook.com/pages/See-Wh0-Viewed-Y0ur-Profile/181639041887279 |
| 20:36.12 | *** part/#asterisk powerunits (~exe@119.152.41.240) |
| 20:39.25 | ariel_ | anyone know how to configure a Astribank T1 CAS setup? |
| 20:40.43 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 20:41.41 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
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| 20:54.38 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 20:56.11 | paulc | In which version of Asterisk did "same =>" become valid in extensions.conf? |
| 20:57.11 | *** join/#asterisk m_tadeu (~quassel@89.180.212.224) |
| 20:57.11 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 21:00.20 | wdoekes2 | paulc: $ grep '"same"' asterisk-*/pbx/pbx_config.c -l | sort | head -n1 |
| 21:00.20 | wdoekes2 | asterisk-1.4.x/pbx/pbx_config.c |
| 21:00.45 | paulc | doffs hat to wdoekes2 - thank you sir! |
| 21:00.56 | wdoekes2 | could be 1.2.. I don't have that.. but I doubt it |
| 21:02.59 | paulc | Ah.. no.. poking around in the code, that's the "same" priority piece.. I meant "same" as a replacement for "exten => xxxx" |
| 21:03.14 | paulc | so you can have a single "exten" followed by a bunch of "same" lines, all using "n" as priority |
| 21:04.15 | wdoekes2 | oh.. in that case.. 1.6 :) |
| 21:05.34 | paulc | 1.6.1.0 doesn't seem to have it though.. might be time for an upgrade at this site.. |
| 21:07.14 | wdoekes2 | mm.. 1.6.1.0 is exactly 2 years old |
| 21:07.27 | wdoekes2 | but, considering that 1.6 is EOL, you could consider moving to 1.8 |
| 21:10.01 | paulc | Yeah, I think you're probably right.. good weekend for an upgrade? |
| 21:10.27 | paulc | actually, now that I saw a bug about AMI and meetme redirects/hangups solved, 1.8 may be where I should be now |
| 21:13.40 | cusco_ | i tested 1.8 once and mysql queue log uses time as datetime instead of unixtimestamp |
| 21:14.02 | cusco_ | there should be a document stating differences to watch out for when moving to 1.8 |
| 21:14.24 | *** join/#asterisk mpe (~mpe@2.109.40.231) |
| 21:15.13 | *** join/#asterisk l1nuxman_ (~l1nuxman@CPE0021296828b2-CM00111ae6f860.cpe.net.cable.rogers.com) |
| 21:15.25 | l1nuxman_ | do you need somethign called Dahdi to use Asterisk? |
| 21:15.32 | cusco_ | no |
| 21:16.18 | l1nuxman_ | I'm trying to install Asterisk on Ubuntu |
| 21:16.33 | cusco_ | so... |
| 21:16.44 | gruvfunk | oh lords of *, will you please shed some guidance on a pestering issue? Have a customer who is tied to a provider (Gafachi), and no matter how we configure * or where we host it (cloud, private vm's, physical boxes, behind nat, in front of the firewall: all routes explored), we keep experiencing the same problem: call drops and we get this Warning: chan_sip.c:3540 retrans_pkt:... |
| 21:16.46 | gruvfunk | ...Hanging up call ed891be743a81d8c0b5624ef46492576@IP.ADD.RE.SS - no reply to our critical packet (see doc/sip-retransmit.txt) |
| 21:16.53 | l1nuxman_ | looking for good documentation for it? Guess the wiki? |
| 21:17.58 | *** join/#asterisk psilvao (~psilvao@190.20.47.180) |
| 21:18.16 | gruvfunk | l1nuxman see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages |
| 21:18.28 | gruvfunk | flawless forme |
| 21:19.27 | cusco_ | l1nuxman_: I used lots of voip-info.org help... its not official tho. wiki.asterisk.org has v1.8 onwards |
| 21:19.39 | l1nuxman_ | what the heck is this dahdi |
| 21:19.41 | gruvfunk | So, if I've exhausted configuration options, hosting options, network options (and by association, ISP provider options), what's left to blame for my call drops and chan_sip.c:3540 retrans_pkt: Hanging up call ed891be743a81d8c0b5624ef46492576@IP.ADD.RE.SS - no reply to our critical packet (see doc/sip-retransmit.txt) |
| 21:20.30 | gruvfunk | (the ITSP provider?) |
| 21:22.00 | cusco_ | I don't believe you've exhausted configuratons |
| 21:22.05 | cusco_ | and I do believe its configuration |
| 21:22.11 | cusco_ | what are you connecting asterisk to ? |
| 21:22.52 | cusco_ | l1nuxman_: dahdi is a set of proprietary modules for linux and modules for asterisk |
| 21:23.06 | cusco_ | basically linux modules are drivers for PRI cards |
| 21:23.16 | gruvfunk | cusco_ * is connecting to Gafachi using their provided configuration |
| 21:23.22 | l1nuxman_ | ahh k |
| 21:23.27 | cusco_ | I dunno what gafachi is |
| 21:23.40 | gruvfunk | voip provider, did's, inbound, outbound |
| 21:24.03 | gruvfunk | Packet timed out after 32000ms with no response |
| 21:24.13 | gruvfunk | so I basically get 32 seconds of connectivity, then boom |
| 21:25.48 | cusco_ | im sure you can twist asterisk to suit almost any sip configuration |
| 21:26.08 | cusco_ | gruvfunk: you should ge a sip debug and get some expert's help in here |
| 21:26.37 | gruvfunk | have never had an issue like this for sure |
| 21:35.23 | cj | is there a way to set up the dialplan so that the local sip peers list is referenced first, and if the peer is not currently connected, the call is forwarded to a gateway? |
| 21:38.44 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 21:39.07 | cj | or maybe there is a way to announce to a pool of asterisk servers which sip peers are currently registered |
| 21:39.15 | cj | maybe that xmpp pubsub thing |
| 21:52.05 | *** join/#asterisk russellb (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 21:52.05 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:58.26 | cusco_ | cj: yes |
| 21:58.52 | citywok | w00t friday |
| 21:59.35 | cusco_ | dial sip/peer or queue that rings in them or something.. then evaluate DIALSTATUS to know if it was answered or noanswer or whatever |
| 22:04.06 | *** join/#asterisk psilvao (~psilvao@190.20.47.180) |
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| 22:32.31 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 22:34.53 | l1nuxman_ | so I have asterisk installed now, but how can I make asterisk receive voicemail when my analog landline phone say rings like 6 times? |
| 22:35.54 | gruvfunk | l1nuxman_ see Dial command |
| 22:36.23 | paulc | l1nuxman_: Is it ringing a SIP phone? Or do you want your house phones to ring and Asterisk is just acting like an answering machine? |
| 22:36.26 | gruvfunk | each ring is separated by about 3 seconds, so if you want 6 rings, you are likely desiring a 20 second ring |
| 22:36.38 | l1nuxman_ | paulc, yes the second one |
| 22:36.51 | l1nuxman_ | I have no SIP Phones |
| 22:37.27 | paulc | l1nuxman_: Have your trunk point to a context, where the s extension does Dial(Local/1234@somecontext), then in that extension do a Wait(20), then Answer, then Voicemail(somemailbox) |
| 22:37.47 | paulc | do you have telephony hardware in your Asterisk box for FXO and FXS? Or only FXO? |
| 22:38.23 | l1nuxman_ | I hope I purchased the right hardware. paulc i bought an ATA with FXO and FXS, LAN and WAN |
| 22:39.43 | paulc | l1nuxman_: Sounds like a SPA-3102 or similar. So... sounds like you want to connect your phone line to FXO, your house phones to FXS, have them all ring a few times, then go to voicemail on Asterisk? |
| 22:40.25 | l1nuxman_ | paulc, I think that is correct |
| 22:41.13 | l1nuxman_ | I have asterisk PBX started now, but what does that mean? It is receiving calls or something? |
| 22:42.09 | paulc | l1nuxman_: You need to configure your ATA to register with Asterisk (create 2 peers in sip.conf, one for FXO port, one for FXS port) |
| 22:42.42 | paulc | then create some dial plan logic to deal with what to do with calls to/from both peers (ie what to do with incoming PSTN calls, what to do when FXS phones dial different digits etc) |
| 22:45.18 | l1nuxman_ | paulc, is the FXS port the "LINE" port? Where my landline comes in? Isn't that the only thing I need to add? And then it uses LAN port to go to my router |
| 22:45.33 | l1nuxman_ | sorry MY Pc not router |
| 22:46.22 | paulc | l1nuxman_: FXO = connection to the telco/line. FXS = where you plug telephones into. |
| 22:46.53 | paulc | l1nuxman_: If you just want Asterisk to be your answering machine, it's fairly simple. If you want all your calls to flow through it (from your house phones) it's a bit more work (but worth it) |
| 22:48.07 | l1nuxman_ | why worth it? They're not VOIP phones so they can't communicate paulc |
| 22:48.29 | Aut0ExeC | l1nuxman_: there is a tut on the interweb for spa3102 and asterisk |
| 22:48.40 | Aut0ExeC | l1nuxman_: google it and it will show u step by step |
| 22:48.45 | *** join/#asterisk neurosys (~neurosys@c-65-34-188-197.hsd1.fl.comcast.net) |
| 22:48.47 | Aut0ExeC | l1nuxman_: thats what I started with also |
| 22:49.01 | paulc | l1nuxman_: If you wire your house phones into the FXS port, you can use your phone line like you do now to make/receive calls |
| 22:49.28 | paulc | l1nuxman_: BUT.. you can also use those phones to dial other places via VoIP, use Voicemail, other clever stuff you create in the dial plan |
| 22:51.00 | l1nuxman_ | hmm k thx |
| 22:51.31 | Aut0ExeC | l1nuxman_: http://forum.voxilla.com/cisco-linksys-sipura-voip-support-forum/starter-spa3102-asterisk-setup-18612.html |
| 22:51.43 | Aut0ExeC | l1nuxman_: that will get u all set up if u have an spa3102 |
| 22:51.56 | paulc | Ahh.. Voxilla forums :-) I used to work for Voxilla, years ago |
| 22:52.03 | Aut0ExeC | :) |
| 22:52.21 | l1nuxman_ | mines a HandyTone HT503 |
| 22:52.27 | Aut0ExeC | oh :( |
| 22:52.49 | Aut0ExeC | well you can either google that and/or still look at the link for ideas |
| 22:53.01 | l1nuxman_ | ty |
| 22:53.03 | Aut0ExeC | k |
| 22:53.11 | paulc | l1nuxman_: In which case... replace the SPA-3102 configuration pieces with the relevant/equivalent for the HT503.. the Asterisk configuration should be fairly similar though |
| 22:53.20 | Aut0ExeC | yup |
| 23:00.16 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 23:07.24 | *** part/#asterisk GameGamer43 (~GameGamer@li119-60.members.linode.com) |
| 23:17.05 | carrar | cj |
| 23:23.15 | *** join/#asterisk ketas- (ketas@ketas6-sixxs.si.pri.ee) |
| 23:27.07 | *** join/#asterisk kaushal (~kaushal@115.246.218.125) |
| 23:27.09 | kaushal | hi |
| 23:27.21 | kaushal | Can someone please recommend me the Hardware Server Configuration/8 or 4 port PRI Card to make Outbound Call at the rate of around 320 outbound Calls/min ? |
| 23:28.47 | pabelanger | kaushal: you can use an erlang calculator |
| 23:29.34 | kaushal | pabelanger: ok |
| 23:30.29 | kaushal | pabelanger: what would be the maximum ports available in a PRI Card ? |
| 23:30.46 | pabelanger | 4 port |
| 23:31.00 | kaushal | pabelanger: I have 8 port card too |
| 23:31.11 | pabelanger | well, from Digium |
| 23:31.15 | kaushal | ok |
| 23:31.33 | kaushal | pabelanger: what are the recommended cards for Asterisk ? |
| 23:31.46 | kaushal | Sangoma or Digium or Dialogic ? |
| 23:31.51 | pabelanger | Digium |
| 23:32.32 | kaushal | pabelanger: what are the availabe PRI cards |
| 23:33.07 | *** join/#asterisk afink (~chatzilla@wsip-70-184-217-180.om.om.cox.net) |
| 23:33.22 | cusco_ | just browse the website |
| 23:33.32 | kaushal | cusco_: ok |
| 23:33.54 | kaushal | cusco_: can you please point me to the URL ? |
| 23:34.12 | cusco_ | http://www.digium.com/en/products/ |
| 23:34.36 | cusco_ | Telephony Interface Cards |
| 23:34.55 | cusco_ | http://store.digium.com/telephony_card_selector.php seems cool |
| 23:35.25 | pabelanger | kaldemar: ^ |
| 23:36.58 | kaushal | pabelanger: Thanks |
| 23:37.00 | kaushal | cusco_: Thanks |
| 23:37.11 | cusco_ | np |
| 23:37.56 | kaushal | Also What specs are recommended to run Asterisk to handle traffic of around 320 Outbound Calls per min ? |
| 23:38.06 | kaushal | I mean Server |
| 23:38.11 | gruvfunk | woa |
| 23:38.45 | kaushal | Is Asterisk a CPU Intensive or Memory Intensive Application ? |
| 23:39.20 | kaushal | gruvfunk: woa ? |
| 23:39.22 | gruvfunk | 320 concurrent calls? |
| 23:39.27 | kaushal | gruvfunk: yes |
| 23:39.40 | gruvfunk | i've just never seen that many, first time for everything |
| 23:39.59 | *** part/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 23:40.12 | kaushal | gruvfunk: Do i need to go with Blade Servers ? |
| 23:40.19 | gruvfunk | i'm handling under 100 call agents on an 8cpu with 8GB, with no issues |
| 23:40.21 | kaushal | gruvfunk: please suggest |
| 23:40.43 | gruvfunk | sorry 4CPU 8GB RAM |
| 23:41.10 | kaushal | gruvfunk: is High Availability available in Asterisk ? |
| 23:41.30 | kaushal | I mean redundanct |
| 23:41.32 | gruvfunk | kaushal: different question, target the audience |
| 23:41.41 | kaushal | I mean redundancy |
| 23:41.45 | kaushal | gruvfunk: ok |
| 23:49.39 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 23:56.05 | kaushal | checking in again for the query ? |
| 23:56.06 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
