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| 02:06.50 | ruben23 | hi guys how do i delete recoridngs on asterisk which are wav file only coz they are mixed on one directory call recordings...i have mp3 and wav on it.. |
| 02:06.51 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 02:06.53 | ruben23 | any idea guys |
| 02:07.15 | WIMPy | rm |
| 02:09.01 | ruben23 | but there lots of wav files and mp3 on that rdirecorty mixed up |
| 02:09.29 | WIMPy | And what's the problem? |
| 02:09.31 | ruben23 | is it possible to have a commadn to dlete wav file only..? |
| 02:09.40 | ruben23 | in one time |
| 02:09.48 | WIMPy | rm *wav |
| 02:11.11 | ruben23 | WIMPy: somehow like 10K records of .wav file on that directory. |
| 02:11.57 | WIMPy | Ouch |
| 02:12.14 | WIMPy | Would surely make sense to do some housekeeping there. |
| 02:13.04 | ruben23 | yes can this be done in one time command acction to delete all .wav files only..? |
| 02:13.31 | WIMPy | rm *wav |
| 02:13.53 | WIMPy | You might have to use xargs or find or split them up if there are too many. |
| 02:25.17 | *** join/#asterisk raspberrypifan (~textual@71-22-220-224.gar.clearwire-wmx.net) |
| 02:25.50 | TazzNZ | with 10k of files and 5k being .mp3 (on avg) I'd say you will exceed the 4k arg limit |
| 02:31.30 | ruben23 | TazzNZ: any command i could do somehow..? |
| 02:34.45 | *** join/#asterisk hfp (~hfp@MTRLPQ0736W-LP130-01-2925269581.dsl.bell.ca) |
| 02:38.11 | TazzNZ | find /path/to/folder -name "*.wav" -type f -exec rm -f{} \; |
| 02:38.15 | TazzNZ | something like that |
| 02:38.22 | TazzNZ | that is from memory so make a test |
| 02:38.28 | TazzNZ | and run on there first |
| 03:08.18 | JTL | Hi all. Anyone used VICIDIAL before? Does it work with Asterisk 11.9? |
| 03:09.01 | JTL | Trying to look for an open source predictive dialer |
| 03:09.57 | ruben23 | no vicidial does not work with 11.. the lastes pacth version is 1.8 only |
| 03:10.12 | ruben23 | latest patch* |
| 03:11.52 | JTL | ruben23: Thanks. Please point me to some documentation if you know where I can look |
| 03:23.28 | *** part/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
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| 04:06.44 | Greek-Boy | anyone here have BLF working on SPA942? |
| 04:09.55 | ChannelZ | I have a 504g |
| 04:14.13 | ChannelZ | guess his internet doesn't work either |
| 04:38.14 | [TK]D-Fender | JTL: they have a website.... that's it. |
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| 07:03.59 | Zogot | ahoyhoy |
| 07:05.01 | r00f | sup |
| 07:20.13 | TazzNZ | land ahead ! :D |
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| 07:36.10 | iulhk | switching from asterisk to kamailio: in asterisk for dialplan we use extensions.conf , i have configured kamailio, want peer to peer calls ? where to handle dialplan in kamailio? |
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| 07:52.17 | *** join/#asterisk al_nz1 (al_nz1@121-99-93-219.bng1.tvc.orcon.net.nz) |
| 07:52.24 | al_nz1 | Hi All |
| 07:52.42 | al_nz1 | I have PIAF running on a VMware PC at home - its great |
| 07:53.01 | al_nz1 | i have a few Cisco phones on the LAN at home that connect to it |
| 07:53.14 | al_nz1 | but I am putting the VM machine in the cloud with azure |
| 07:53.51 | al_nz1 | Is the best way to get the phones onto the server once the server is in the cloud to setup a site to site VPN, or open up some ports? |
| 07:54.57 | r00f | if router is powerful enough to handle all the voip traffic in vpn, i'd prefer vpn |
| 07:55.41 | al_nz1 | r00f: Yeah, I will probably use a router running DD-WRT for the site to site VPN |
| 07:55.47 | al_nz1 | should be ok |
| 07:56.13 | al_nz1 | I was more worried about the extra hops to the Azure server |
| 07:57.00 | al_nz1 | I kinda dont know why I am doing it because I know Im going to inflict a world of frustration on myself |
| 07:57.27 | r00f | we mostly use site-to-site vpns for branches, but in my experience i've seen some poor home-class routers that would loose packets in very loaded encrypted vpn. modern routers should be ok, i guess. still can depend on amount of traffic. |
| 07:57.45 | al_nz1 | maybe im a mascocist |
| 07:57.48 | al_nz1 | sic |
| 07:58.09 | al_nz1 | r00f: ok |
| 07:58.21 | al_nz1 | I use openvpn nailed up |
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| 09:15.33 | *** join/#asterisk andycol (~andycol@41.79.82.246) |
| 09:16.05 | andycol | Hey Guys |
| 09:16.13 | *** join/#asterisk hindi (~hindi@030-138-088-212.ip-addr.vsenet.de) |
| 09:16.42 | andycol | in my macro i have a GotoIf section that points to the dial command how do I tell it what number to call |
| 09:16.50 | andycol | currently it just calls the context |
| 09:16.51 | andycol | eg |
| 09:16.58 | andycol | exten => s-Medivision,1,Set(CDR(accountcode)=${accountcode}) |
| 09:16.58 | andycol | <PROTECTED> |
| 09:16.58 | andycol | <PROTECTED> |
| 09:17.08 | andycol | i need it to show the no dialed |
| 09:26.42 | *** join/#asterisk StaRetji (~vencuga@178.79.6.17) |
| 09:27.09 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 09:28.01 | StaRetji | TazzNZ: ping |
| 09:28.02 | StaRetji | :) |
| 09:29.43 | TazzNZ | hi StaRetji |
| 09:29.51 | StaRetji | hi |
| 09:30.04 | StaRetji | remember that noop you said I should put in a2billing to read dnid |
| 09:30.10 | TazzNZ | yeah |
| 09:30.14 | StaRetji | it was reading 993366123123 |
| 09:30.25 | StaRetji | but a2billing then strips and mess things |
| 09:30.33 | StaRetji | so, I added noop to asterisk context |
| 09:30.36 | StaRetji | and here is what I get |
| 09:30.42 | StaRetji | <PROTECTED> |
| 09:30.45 | StaRetji | so, that is problem |
| 09:30.53 | TazzNZ | correct |
| 09:31.06 | StaRetji | that is why it is not matched with astdb |
| 09:31.11 | andycol | can you catch a number dialed and declare it as variable |
| 09:31.12 | StaRetji | so, now I have 2 choices |
| 09:31.31 | StaRetji | actually only 1 choise |
| 09:31.42 | StaRetji | because that c363 is always different |
| 09:31.54 | StaRetji | so, I need to strip Local/ and @asterisk.... |
| 09:32.28 | StaRetji | actually, I need to strip Local/0 and @asterisk-..... |
| 09:32.37 | StaRetji | and add 9933 in front |
| 09:33.03 | andycol | any ideas if it is possible? |
| 09:33.08 | TazzNZ | yip |
| 09:33.08 | StaRetji | problem is, I know how to strip from start, but no idea how to strip the rest :( |
| 09:33.10 | TazzNZ | it is |
| 09:33.21 | andycol | how lol |
| 09:33.40 | TazzNZ | I would do a gotoif for that |
| 09:33.41 | StaRetji | let me paste for the start |
| 09:33.44 | TazzNZ | something like |
| 09:34.18 | TazzNZ | gotoif($[${EXTEN:1:1} = "0"]?dial the right number) |
| 09:34.19 | StaRetji | exten => _X.,n(dnid),GotoIf(${DB_EXISTS(whitelistdnid/${CALLERID(dnid)})}?callout,9933${EXTEN:7},1:gotofilter) |
| 09:34.37 | StaRetji | this should strip Local/0 |
| 09:34.37 | TazzNZ | that will take from char position 7 |
| 09:34.42 | StaRetji | and add 9933 |
| 09:34.45 | StaRetji | I think so |
| 09:34.52 | TazzNZ | you need 9933${EXTEN:2} |
| 09:34.53 | StaRetji | but problem is how to strip @asterisk-cccc |
| 09:34.59 | TazzNZ | the Local/0 isn't counted |
| 09:35.05 | StaRetji | really? |
| 09:35.07 | StaRetji | hm |
| 09:35.17 | TazzNZ | and the @asterisk is the context |
| 09:35.30 | StaRetji | yes, I know, but that is how it see if I do this: |
| 09:35.31 | TazzNZ | Local is a channel |
| 09:35.40 | TazzNZ | jus tlike "SIP" or "DAHDI" |
| 09:35.47 | StaRetji | exten => _X.,1,NoOp(${CALLERID(dnid)}) |
| 09:35.54 | StaRetji | if I put this before a2b |
| 09:35.55 | andycol | Tazz will that work? |
| 09:36.01 | StaRetji | it shows 993366123123 |
| 09:36.03 | andycol | can i show u my macro and u can see what im trying to do? |
| 09:36.13 | StaRetji | if I put after a2billing, it shows Local/0 |
| 09:36.14 | TazzNZ | andycol: kinda busy with StaRetji - give me a min |
| 09:36.20 | andycol | ok cool no prob :) |
| 09:36.38 | TazzNZ | StaRetji: pastebin your exten.conf again please |
| 09:37.06 | TazzNZ | I still believe the problem is inside a2billing |
| 09:37.14 | TazzNZ | it shouldn't re-write the number |
| 09:37.24 | TazzNZ | but even if it does, you should be able to catch it |
| 09:37.37 | TazzNZ | btw - did you fix that GotoIf with $[] ? |
| 09:37.38 | StaRetji | yes, moment |
| 09:37.51 | StaRetji | no, i haven't touch anything |
| 09:38.23 | TazzNZ | you need to do that first |
| 09:38.27 | StaRetji | http://pastebin.com/3igANkuM |
| 09:51.14 | *** join/#asterisk aness (~aness@2a02:fe0:c310:57f0:3421:b911:2937:e0b5) |
| 09:55.10 | andi | Hi |
| 09:55.53 | andi | Can somebody tell me what OPTIONS packets are doing within the SIP protocol? Are they transmitted through 5060 udp as all other sip packets? |
| 09:57.05 | andi | My provider is telling me that my asterisk is not accepting OPTIONS packets. How can I check that? |
| 09:58.10 | Tim_Toady | andi: ofc they go through 5060 as the rest of the SIP signaling, its part of the sip protocol, its a method that used to discover information about what each endpoint supports |
| 09:58.44 | Tim_Toady | probably your asterisk doesnt reply back to their OPTIONS message |
| 09:59.46 | Tim_Toady | seems like they are not reaching your server (firewalling, nat?) |
| 10:07.10 | *** join/#asterisk jiku (~jiku@182.73.253.194) |
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| 10:25.36 | *** join/#asterisk glambert_ (~glambert@37.157.50.80) |
| 10:26.19 | glambert_ | any way to play an announcement in Dial() using the A(/path/to/message) and have it actually wait for the call to be answered before playing? If the call isn't answered straight away the majority of the message is missed |
| 10:27.03 | *** join/#asterisk michael_work (~michael@bzq-82-168-31-134.red.bezeqint.net) |
| 10:31.44 | michael_work | i have sip trunk configurated. and there is one extension that is connected to asterisk thru kamailio and there is no sound. until i use hold/unhold. both set to canreinvite=yes and directmedia=no. The weird part is when i comment out type on sip trunk it automagically starting to work. |
| 10:33.52 | file | glambert_, what are you dialing out on... |
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| 10:51.31 | iulhk | hi |
| 11:07.26 | andi | Tim_Toady: That's exactly the point. I have another trunk configured which works. Therefore sip communication is working in any way. The firewall is open and at the moment the ip and the port is reachable from everywhere. |
| 11:07.50 | andi | I feel like I have a cheese firewall lots and lots of wholes... |
| 11:08.01 | andi | *holes |
| 11:09.43 | *** join/#asterisk jasnejac (~kvirc@81.91.106.59) |
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| 11:14.58 | Tim_Toady | andi: thes get a network trace and inspect if you actually get the OPTIONS messages and sending back a reply (200 OK) |
| 11:15.19 | Tim_Toady | s/thes/then/ |
| 11:16.52 | andi | How often do such option packages occur? |
| 11:16.55 | andi | Every minutre? |
| 11:17.02 | andi | s/minutre/minute/ |
| 11:17.22 | Tim_Toady | depends how the other end is configured |
| 11:18.07 | andi | Is it enough to do a tcpdump -i eth0 host <their host> or do I need special udp settings? |
| 11:19.11 | Tim_Toady | host <IP> and udp and port 5060 will create a more clear trace only with sip traffic, but well, it isnt that importand |
| 11:19.32 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:19.59 | andi | I will skip the port 5060 part because I'd like to know if there is any other traffic from their host to my asterisk, thanks. |
| 11:24.16 | andi | I there an easy way to send an OPTIONS packet to my own asterisk? |
| 11:24.27 | andi | Some testing tool or something similar? |
| 11:24.56 | Tim_Toady | sipsak |
| 11:25.02 | Tim_Toady | another atserisk |
| 11:25.06 | Tim_Toady | a phone |
| 11:25.15 | Tim_Toady | lots of options |
| 11:28.49 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
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| 11:36.37 | andi | A simple sipsak -vv -s sip:<my-voip-ip> gives me a 200 OK. So OPTIONS packets should work. |
| 11:40.03 | *** join/#asterisk yottanami (~yottanami@130.255.255.35) |
| 11:40.28 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:41.23 | *** part/#asterisk StaRetji (~vencuga@178.79.6.17) |
| 11:41.57 | yottanami | I wan Elastix user now I want to switch pure Asterisk :-) Now I want to create this Inbound/Outbound route,How can I do it ? |
| 11:41.57 | yottanami | http://tarvandco.com/blog/wp-content/uploads/2013/12/12-10-2013-10-14-45-AM.png |
| 11:41.57 | yottanami | http://tarvandco.com/blog/wp-content/uploads/2013/12/12-10-2013-10-15-56-AM.png |
| 11:42.34 | andi | The tcpdump does only show the OPTIONS packets which my asterisk is sending for qualify. Over 10 minutes I cannot see one single OPTIONS packet from their site. |
| 11:44.08 | WIMPy | yottanami: You write your extensions.conf. |
| 11:46.41 | yottanami | WIMPy: It is default I just add one line "[my-phones] exten => 2000,1,Dial(SIP/2000)" |
| 11:46.54 | WIMPy | What is default? |
| 11:49.55 | yottanami | WIMPy: http://dpaste.com/3GM7S9Q/ |
| 11:51.08 | yottanami | WIMPy: I also add this lines at the end of /etc/asterisk/sip.conf http://dpaste.com/22PWFPB/ |
| 11:51.38 | WIMPy | Sample configs are samples and not defaults to be used. |
| 11:52.15 | *** join/#asterisk NirS (~NirS@5.28.148.71) |
| 11:52.22 | NirS | Greetings all |
| 11:52.25 | WIMPy | Especially extensions.conf needs to be written to YOUR needs. |
| 11:52.37 | yottanami | WIMPy: OK I will remove contents |
| 11:52.46 | NirS | guys, I'm having a very weird problem |
| 11:53.10 | WIMPy | yottanami: Start with the book |
| 11:53.15 | WIMPy | ~book |
| 11:53.15 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 11:53.18 | NirS | If I change the default SIP port from 5060 to something else, although netstat will report the port as binded - Asterisk will not accept the inbound packets |
| 11:53.21 | NirS | any ideas ? |
| 11:53.38 | Tim_Toady | NirS firewall |
| 11:53.39 | WIMPy | Firewall? |
| 11:53.42 | NirS | no firewall |
| 11:53.43 | NirS | no nothing |
| 11:53.50 | Tim_Toady | no packets |
| 11:54.03 | NirS | I can see the packet coming in on port 6060 |
| 11:54.09 | NirS | but Asterisk doesn't seem to get it at all |
| 11:54.11 | NirS | really odd |
| 11:54.12 | yottanami | WIMPy: I am reading http://the-asterisk-book.com but I dont know what is Inbound/outbound Route is exactly |
| 11:54.23 | Tim_Toady | NirS: listening on different interface? |
| 11:54.33 | NirS | same physical interface |
| 11:54.58 | Tim_Toady | what netstat shows, what IP asterisk listens to |
| 11:55.01 | WIMPy | yottanami: That also good, but a little out of date. Probably not an issue for you now. |
| 11:55.27 | WIMPy | yottanami: Asterisk has no concept of incomming/outgoing. Every call comes in and goes out. |
| 11:55.53 | NirS | netstat shows Asterisk binded to the IP number and port I've assigned |
| 11:55.54 | yottanami | WIMPy: which book do you prefer ? |
| 11:56.01 | NirS | but yet, it will not accept my SIP requests |
| 11:56.12 | NirS | only if I bind to port 5060 it will work |
| 11:56.27 | NirS | for example, tried binding TLS to 5063, not working, only 5061 |
| 11:56.34 | WIMPy | yottanami: Actually I did prefer the one you're reading. Look above at the information Infobot gave. |
| 11:56.52 | Tim_Toady | NirS: are your phones/peers/endpoint configures to connect to the new port? |
| 11:57.02 | Tim_Toady | NirS: are your phones/peers/endpoints configured to connect to the new port? |
| 11:57.53 | NirS | of course |
| 11:58.10 | yottanami | WIMPy: I have a Newrock Gateway that I need to make contact between Astrisk and Newrock Gateway. |
| 11:58.11 | yottanami | I find just this Elastix tutorial http://tarvandco.com/blog/1392/09/19/%D8%A7%D8%B1%D8%AA%D8%A8%D8%A7%D8%B7-%DA%AF%DB%8C%D8%AA-%D9%88%DB%8C-%D9%87%D8%A7%DB%8C-newrock-%D8%A8%D8%A7-elastix/ ( non English ) |
| 12:04.01 | NirS | ok, this is really odd |
| 12:04.13 | NirS | when I added the port entry to /etc/services file, it started working |
| 12:04.28 | NirS | could this be a security restriction for CentOS 6.5 ? |
| 12:04.32 | NirS | doesn't make any sense |
| 12:06.26 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:06.50 | *** join/#asterisk bmurt (~brendan@static-96-245-76-214.phlapa.fios.verizon.net) |
| 12:14.08 | Tim_Toady | NirS: selinux is enabled? |
| 12:16.53 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 12:17.35 | *** join/#asterisk 77CAAL933 (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
| 12:20.25 | WIMPy | Does anyone her have an idea what's going on at sipgate? Did they break something or is it a new policy? All calls to simquadrat give "Forbidden". |
| 12:29.39 | *** join/#asterisk `mc (~mc@c-71-235-50-162.hsd1.ct.comcast.net) |
| 12:30.18 | `mc | Looking for an experienced asterisk developer for contractor work. Please /msg me for details.. |
| 12:31.08 | WIMPy | `mc: You should not expect serious answers unless you give details about what you need to be done. |
| 12:33.26 | `mc | as I said, /msg me for details |
| 12:33.51 | WIMPy | As I said: Don't expect anyone serious. |
| 12:34.11 | `mc | oh well |
| 12:34.16 | marceloamorim | guys, yesterday I asked about how can I handle it with astdb, I save at the beginning of call the callerid with the uniqID, than I need to transfer this call to a number, but when I transfer I just create a new channel, because of that they told me that I should use astdb to save as I did, but now I need to use that key that I saved it on this second channel, how can I do that? the pastebin is > http://pastebin.com/gBaWWtdq |
| 12:35.06 | WIMPy | How do you create the 2nd channel? |
| 12:35.15 | WIMPy | Is it a transfer? |
| 12:35.48 | [TK]D-Fender | it's really a new call... |
| 12:36.15 | WIMPy | How/why is the 2nd call started? |
| 12:37.28 | marceloamorim | because I wish to do a research with the clients, but not to everyone |
| 12:39.04 | marceloamorim | in my thoughts these way was the best way, so I did it this way, but when I transfer to number 200, |
| 12:40.03 | marceloamorim | I wish to get those db that I create at the beginning, but I didn't know how |
| 12:41.30 | marceloamorim | I did the same thing with global var, but when I did that, when I have multiples calls the variable wasn't the right var |
| 12:43.08 | WIMPy | You are the only one who can make the link between those calls. Now you need to find out if you can describe that link. |
| 12:43.59 | marceloamorim | I just use uniqID |
| 12:44.13 | marceloamorim | but I don't know if is the right way |
| 12:45.01 | WIMPy | You already found out that it's the way to find a call, but you also already know that you need to know what call you're looking for. |
| 12:45.11 | [TK]D-Fender | [08:39]marceloamorimin my thoughts these way was the best way, so I did it this way, but when I transfer to number 200, <- do a BLIND transfer and that original channel will have all it's variables intact as-is |
| 12:47.51 | *** join/#asterisk acassio (~acassio@177.20.3.249) |
| 12:52.44 | marceloamorim | ohh |
| 12:54.17 | marceloamorim | I'll do this blind transfer than [TK]D-Fender, ty, I didn't know that |
| 12:54.21 | marceloamorim | awesome |
| 12:55.41 | *** join/#asterisk dfighter (~someone@arcemu/staff/dfighter) |
| 12:55.47 | marceloamorim | but WIMPy, how do you suggest that I set a db for one call and this db is uniq and I could get the right db |
| 12:55.51 | *** join/#asterisk PLMg (~PLMg@78.96.151.225) |
| 12:56.32 | [TK]D-Fender | You don't need this at all.... |
| 12:56.54 | WIMPy | And you can't unless you hve a way to link th e calls. |
| 12:56.59 | marceloamorim | I know now, but this can be helpful in another situation |
| 12:57.01 | [TK]D-Fender | If the variables were set prior to the call to the person doing the transfer then they are already in the originating channel. |
| 12:57.04 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 13:00.26 | PLMg | hello, I am trying to see inbound calls on a specific number that routes the call to a queue used by another number aswell. Any ideea how I can do this in mysql using cdr? Thus far I have SELECT * FROM cdr WHERE dst = '2'(queue number) ORDER BY calldate DESC; But since it is used by another number this query is a bit pointless. |
| 13:02.06 | [TK]D-Fender | your goal is a little vague... |
| 13:02.08 | WIMPy | PLMg: Maybe you should have set the called number as account code, for example. |
| 13:04.37 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 13:06.20 | PLMg | this is not my setup, I just inherited it. My goal is to see the inbound calls since january til april on a number that shares a queue with another number. |
| 13:06.49 | WIMPy | If the information wasn't saved, it's too late. |
| 13:07.10 | PLMg | the calls are logged in cdr |
| 13:07.34 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:f970:75fb:3d4f:2365) |
| 13:07.37 | WIMPy | With the information you want? |
| 13:07.50 | PLMg | yes |
| 13:08.23 | WIMPy | I guess I didn't get the question then. |
| 13:09.00 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 13:09.40 | PLMg | how to query in mysql or the freepbx report for inbound calls made to a number between january and april 2014 |
| 13:11.09 | [TK]D-Fender | Have your "where" check the date-range then. |
| 13:14.53 | PLMg | Let us name the number I wish to get this information for "X". When someone dials the number X, the call gets put in a queue. That queue is used by another number aswell, lets call that number "Y". How can I see the inbound calls for the number X between january and april? |
| 13:15.26 | [TK]D-Fender | First, stop just saying "X". Is that actually in the DST field? |
| 13:15.28 | WIMPy | Where have X and Y been stored? |
| 13:15.39 | [TK]D-Fender | And the CALLDATE field IS there. |
| 13:15.47 | WIMPy | Have they been stored at all? |
| 13:15.47 | [TK]D-Fender | and you can obviously query on that |
| 13:16.12 | [TK]D-Fender | Look at your CDR. Can you, with your eyes, identify the kind of calls you want to select? |
| 13:17.18 | PLMg | [TK]D-Fender, yes I can. WIMPy the calls are stored in asteriskcdrdb that I can access via mysql |
| 13:17.42 | WIMPy | That's not what I asked. |
| 13:18.07 | [TK]D-Fender | plgwhats the problem then? set your WHERE and be done with it... you're talking about 2 simple tests.... |
| 13:19.53 | PLMg | The calls are stored on the same machine as asterisk. [TK]D-Fender the destinatio "2" is the queue used for 2 numbers. The number that I want the onfo for and another number. |
| 13:19.53 | snmp | select * from cdr where dst='X' and lastapp='Queue' |
| 13:21.39 | snmp | PLMg: whats wrong with this query? |
| 13:22.23 | PLMg | snmp, it does not diferentiate between the 2 numbers that share the queue |
| 13:22.55 | snmp | someone dials the number X, the call gets put in a queue. |
| 13:23.04 | PLMg | yes |
| 13:23.27 | snmp | and cdr record for this is dst x and lastapp queue |
| 13:23.36 | PLMg | yes |
| 13:23.58 | snmp | and this is not what yout need? |
| 13:25.03 | PLMg | it is sprt of. the same rule applies for the number "y" aswell. Someone dials the number Y. the call gets put in a queue, the same queue as the number X uses. |
| 13:25.36 | snmp | you want some stats on a queue member? |
| 13:26.20 | [TK]D-Fender | [09:22]PLMgsnmp, it does not diferentiate between the 2 numbers that share the queue <- where do YOU see the difference in the CDR record? |
| 13:28.18 | PLMg | snmp, I think it is used for statistics and also to get info on members. |
| 13:30.48 | PLMg | [TK]D-Fender, I do not know how to differentiate the 2 number via a query. But I know for a fact that the agents answering the phones answer bouth numbers since one is a number for the tech support for a product and the other number is also tech support for another product. The agents get call for support on both prudcs thus the 2 numbers are both in use. |
| 13:32.45 | WIMPy | I guess my first answers were the right ones, but it's impossible to help if no questions are answered. |
| 13:34.06 | PLMg | WIMPy, can you make me understand you questions? :) |
| 13:35.30 | WIMPy | Does the information you're looking for exist? If so where? |
| 13:36.00 | *** join/#asterisk Takapa (vegard@junior.svanberg.no) |
| 13:36.48 | PLMg | All the calls are stored in cdr, so the information I need is tored inside. But I do not know how to filter for it. |
| 13:37.16 | [TK]D-Fender | PLMg: where do YOU see the difference in the CDR record? |
| 13:37.21 | WIMPy | You still didn't answer the question. |
| 13:37.36 | snmp | looks like its a dstchannel field |
| 13:37.49 | WIMPy | I didn't ask if the calls are there, but if the information you want (i.e. the called number) is there. |
| 13:38.02 | [TK]D-Fender | Nothing "looks" like anything.... we haven't been shown 2 separate samples. |
| 13:38.26 | snmp | cmon hes talking about queues |
| 13:38.40 | snmp | dst contains only extension |
| 13:38.51 | [TK]D-Fender | He's asking about querying CDR. |
| 13:39.21 | PLMg | this is a sample from dstchannel: Local/111@from-queue-7e7f,1 |
| 13:39.33 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:39.41 | snmp | and 111 is your X? |
| 13:39.47 | [TK]D-Fender | PLMg: answer the question and show us 2 clear samples. |
| 13:39.48 | PLMg | WIMPy, no, the called number is not there |
| 13:40.01 | WIMPy | Ok, great. |
| 13:40.06 | [TK]D-Fender | PLMg: show us the identifying piece of information from your CDR records. |
| 13:40.09 | WIMPy | If it's not there, it's not there. |
| 13:40.18 | [TK]D-Fender | ^^^^^^ |
| 13:41.06 | WIMPy | You can't find things that don't exist. |
| 13:41.33 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 13:43.45 | WIMPy | Change your dialplan so that you can find the information in the future. |
| 13:43.52 | snmp | weve lost him, doctor |
| 13:44.09 | WIMPy | Which is what I said 40 minutes ago. |
| 13:44.42 | PLMg | testing something out so I can provide the right information |
| 13:45.09 | snmp | drumroll |
| 13:45.30 | [TK]D-Fender | What is there to test. Look at your CDR records with your own eyes. If you can't tell by looking at them, then there is nothing to query for. |
| 13:46.04 | [TK]D-Fender | As WIMPy said, either you have enough identifying information, or you don't. |
| 13:52.59 | *** part/#asterisk AlHafoudh (~AlHafoudh@echo.freevision.sk) |
| 13:59.47 | matt_ | does anybody know if a company that allows you to verify and spoof caller id's to the PSTN and also allows the caller id's to be set from asterisk SetCallerID? and is UK based? |
| 14:00.41 | PLMg | my issue is almost done. |
| 14:00.54 | PLMg | they have 2 numbers for the same service, just like a cab company has multiple numbers for the same service. Will contact them now to set things straight |
| 14:01.18 | PLMg | thank you for the help! |
| 14:04.39 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 14:10.57 | jasnejac | matt_: It's illegal to do that in the UK I believe |
| 14:11.34 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 14:13.55 | *** join/#asterisk cyborg-one (~cyborg-on@62.16.23.192) |
| 14:16.15 | matt_ | jasnejac: not if you prove you own the number, i am already doing it but the campany I use drops all caller id and sets when I have entered into the web interface |
| 14:16.50 | matt_ | jasnejac: all I want is the ability for asterisk to set that value, I can set multiple at the moment and switch between then, but again I have to do it from a web interface |
| 14:17.43 | jasnejac | matt_: If that's the case (you own the numbers) I may be able to help |
| 14:17.51 | *** part/#asterisk yottanami (~yottanami@130.255.255.35) |
| 14:22.12 | *** join/#asterisk mattcen (~mattcen@c110-22-201-130.sunsh4.vic.optusnet.com.au) |
| 14:22.20 | *** join/#asterisk BeachBall (~eXcAliBuR@206.162.174.6) |
| 14:22.50 | BeachBall | I'm trying to get my asterisk box to be able to route sip calls from my video conference unit, i don't know what to put in the extensions.conf |
| 14:23.11 | [TK]D-Fender | Same as anything else. |
| 14:23.23 | [TK]D-Fender | look what it is dialing. Make something to amtch it. |
| 14:23.26 | [TK]D-Fender | match* |
| 14:23.28 | BeachBall | video conf unit is reg to asterisk... and dials out a sip address... ex. beachball@sipppy.sip.com |
| 14:24.02 | BeachBall | like i know how to make it work with phone numbers, but how do i match letters? |
| 14:25.55 | *** join/#asterisk Qwell (north@pdpc/sponsor/digium/Qwell) |
| 14:25.55 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 14:26.01 | [TK]D-Fender | exten => johndoe,1,NoOp(duh) |
| 14:27.08 | *** join/#asterisk timahvo1 (~rogue@197.237.131.169) |
| 14:27.10 | BeachBall | i'm not explaning it correctly |
| 14:27.35 | [TK]D-Fender | Show an actual call and we'll see what it's looking for. |
| 14:28.29 | BeachBall | we have to call a sip uri d-fender@sip1.com from our video conference unit that regs to our internal asterisk box. sip1.com in external |
| 14:29.08 | BeachBall | our internal asterisk box needs to be able to call the sip uri that we enter in our video conf unit... so it will change |
| 14:29.44 | [TK]D-Fender | So make a match for "anything" |
| 14:31.12 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-aiupdpwfjdijdyhm) |
| 14:31.12 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:44.36 | *** join/#asterisk jaflong (5bec7504@gateway/web/freenode/ip.91.236.117.4) |
| 14:44.55 | jaflong | asterisk 12 is listening on port 5060 |
| 14:45.03 | jaflong | netstat dp 0 0 0.0.0.0:5060 0.0.0.0:* 6372/asterisk |
| 14:45.15 | jaflong | tcpdump 10.1.15.65.5090 > 10.1.15.61.sip: SIP, length: 368 REGISTER sip:gstream1@10.1.15.61 SIP/2.0 |
| 14:45.33 | jaflong | any idea why asterisk does not pick up the sip regster |
| 14:46.49 | [TK]D-Fender | Show us |
| 14:49.27 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-wldyvrrofkvfgivi) |
| 14:49.27 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:49.35 | jaflong | I found this info http://yohann.martineau.free.fr/blog.php/blog.php?url=asterisk-11-listen-on-port-5060&lang=en and subsequently added openssl-devel and rebuilt however still not resolved |
| 14:52.17 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-aawaoxzrauxyfour) |
| 14:53.18 | [TK]D-Fender | Finding a random blog entry is not "looking at your system". |
| 14:53.31 | [TK]D-Fender | Go to * CLI enable SIP debug an show wus what's arring. |
| 14:53.34 | [TK]D-Fender | arriving* |
| 14:53.39 | *** join/#asterisk mattcen (~mattcen@c110-22-201-130.sunsh4.vic.optusnet.com.au) |
| 14:53.58 | [TK]D-Fender | And verify that you don't have a firewall in the way |
| 14:54.03 | jaflong | that the problem nothing is arriving |
| 14:55.14 | [TK]D-Fender | check your firewall. |
| 14:56.19 | jaflong | tcpdump -v dst port 5060 shows the sip messaging coming in. Firewall is not activated |
| 14:57.31 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 14:58.58 | [TK]D-Fender | Show us the dump and * CLI |
| 15:02.05 | BeachBall | [May 28 10:59:52] WARNING[2179]: chan_sip.c:4176 retrans_pkt: Retransmission timeout reached on transmission 7e069ac703d0209242d169d21eb5dd63@10.75.2.190:5060 for seqno 102 (Critical Request) |
| 15:02.09 | BeachBall | thats what i get |
| 15:02.31 | BeachBall | i think the other end is trying to contact on our private ip |
| 15:02.34 | BeachBall | :{ |
| 15:03.08 | [TK]D-Fender | Look > Think |
| 15:03.28 | BeachBall | i don't know how to confirm |
| 15:04.07 | [TK]D-Fender | SIP DEBUG obviously |
| 15:04.19 | [TK]D-Fender | We've been through this several times... |
| 15:04.29 | jaflong | i will try to get core debug log |
| 15:04.36 | BeachBall | yes and i keep forgetting the debug command |
| 15:04.43 | BeachBall | sip debug on doesn't work |
| 15:05.05 | [TK]D-Fender | "core show help" |
| 15:05.20 | BeachBall | ok debug is on |
| 15:06.10 | BeachBall | oh wow |
| 15:06.13 | BeachBall | that is great |
| 15:09.43 | BeachBall | your so smart |
| 15:09.44 | BeachBall | :} |
| 15:09.45 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 15:12.22 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:12.22 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 15:17.03 | shidan | hello, does jingle support webrtc or is it just chan_sip ? |
| 15:18.10 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 15:18.17 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 15:18.22 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 15:19.45 | mjordan | shidan: "Support WebRTC" doesn't really have much meaning, as WebRTC isn't a standard, and even if it was, it doesn't mandate signaling. chan_sip and chan_pjsip (12+) support websockets as a transport. Client libraries exist for the browsers that support sending SIP over WS. |
| 15:20.20 | mjordan | shidan: chan_motif/res_xmpp do not currently support a websocket transport |
| 15:22.18 | *** join/#asterisk navaismo (~navaismo@200-52-45-221.dynamic.axtel.net) |
| 15:26.34 | shidan | mjordan: Thanks, that answers it ... I was referring to webrtc support as specified on the asterisk.org site with res_http_websocket |
| 15:39.41 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:45.33 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 15:55.33 | adeeln | on a 1.6 box, is there any way for the verbosity to be increased without a user manually issuing either a 'core set verbose X' or connecting to asterisk via 'asterisk -vvvr' ? |
| 15:56.18 | WIMPy | asterisk.conf |
| 15:57.55 | adeeln | but that wouldn't take affect without restarting asterisk i believe |
| 15:58.14 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 15:59.26 | WIMPy | I think you're right. |
| 16:06.17 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
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| 16:15.05 | litn | anyone here have experience with attended transfer? |
| 16:15.23 | wdoekes | ~ask |
| 16:15.23 | 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. |
| 16:15.40 | navaismo | maybe is a poll |
| 16:15.42 | wdoekes | ~polls |
| 16:15.42 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 16:15.50 | navaismo | ouw |
| 16:16.22 | litn | if I do an attended transfer to someone who is on DND, I have a delemma... If I hang up then the person is transfered to this mid voicemail prompt, and the voicemail can't hang up to give the caller back to me |
| 16:16.34 | adeeln | i want to validate that a value returned by a db lookup matches a toll-free number and do an ExecIf...i'm trying to do this via a regex, but is it better to use the REGEX() function, or use the expression handler ($[ foo =~ bar ] ) for this? |
| 16:17.24 | *** join/#asterisk dumby_PC (~dumby@204.246.140.162) |
| 16:17.46 | wdoekes | litn: odd phone. my spa lets me hang up the 2nd call and return to the original. only by pressing xfer does the transfer happen |
| 16:18.48 | litn | I don't know if this is a phone thing, we do supervised transfer and normal transfer via AMI application |
| 16:19.18 | litn | the way the supervised transfer has been working, is if we hang up the phone then the other person is transfered the call, if they hang up the phone the caller is put back on with us |
| 16:19.23 | litn | is that unusual? |
| 16:20.27 | wdoekes | I have no idea what is normal in an office environment. but you do indeed have a dilemma if peers-on-DND redirect calls to voicemail then |
| 16:21.01 | [TK]D-Fender | [12:16]litnif I do an attended transfer to someone who is on DND, I have a delemma... If I hang up then the person is transfered to this mid voicemail prompt, and the voicemail can't hang up to give the caller back to me <- since when should this call that you transferred suddenly decide to call back to you? |
| 16:21.08 | Qwell | One of the benefits of an attended transfer over blind, is the ability to cancel the transfer. |
| 16:21.22 | Qwell | You're removing the ability to cancel. That's where your problem lies. |
| 16:21.35 | litn | if you cancel then you should get the call back to you, right? |
| 16:21.48 | *** join/#asterisk dumby_PC (~dumby@204.246.140.162) |
| 16:21.50 | litn | so I guess that's the question, how to cancel an attended transfer... |
| 16:22.12 | *** join/#asterisk Zogot (~Adium@90-145-116-55.bbserv.nl) |
| 16:22.18 | [TK]D-Fender | [12:18]litnI don't know if this is a phone thing, we do supervised transfer and normal transfer via AMI application <-- I'd read the AMI docs for this.... |
| 16:22.21 | Qwell | The outbound leg of an attended transfer is a second call. You just hang up that second call. |
| 16:22.53 | [TK]D-Fender | Qwell: doesn't that complete the transfer? |
| 16:24.35 | Qwell | If your phone is stupid, maybe. |
| 16:25.31 | [TK]D-Fender | Qwell: He's referring to using AMI to do the transfer, not the phone |
| 16:26.23 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
| 16:26.28 | mjordan | AMI attended transfers are a bit odd. IIRC, they 'fake out' the DTMF that would normally trigger a features/core attended transfer. |
| 16:26.37 | mjordan | double checks |
| 16:27.06 | mjordan | yup. |
| 16:27.07 | *** join/#asterisk AlHafoudh (~AlHafoudh@echo.freevision.sk) |
| 16:27.32 | mjordan | from the perspective of the 'phone', sending the DTMF to cancel the transfer should work |
| 16:28.51 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
| 16:32.40 | *** join/#asterisk tuxd00d (~tuxd00d@ip24-251-34-116.ph.ph.cox.net) |
| 16:35.25 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 16:35.48 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 16:37.36 | *** join/#asterisk tuxd00d (~tuxd00d@ip24-251-34-116.ph.ph.cox.net) |
| 16:38.31 | *** join/#asterisk dumby_PC (~dumby@204.246.140.162) |
| 16:38.46 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 16:39.29 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 16:53.02 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
| 16:59.35 | *** join/#asterisk dumby_PC (~dumby@204.246.140.162) |
| 17:07.43 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
| 17:10.20 | *** join/#asterisk Sprocks (~Sprocks@70.49.35.13) |
| 17:15.02 | BeachBall | how to use video codecs with asterisk? |
| 17:15.09 | BeachBall | says no compatible codecs |
| 17:15.11 | BeachBall | :{ |
| 17:17.52 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 17:18.15 | [TK]D-Fender | then specify them in your peers |
| 17:19.48 | BeachBall | i did |
| 17:19.58 | BeachBall | i did allow = h263 ..;. and others |
| 17:20.49 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 17:20.49 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 17:20.50 | BeachBall | show video codecs gives a list |
| 17:20.51 | *** join/#asterisk Sprocks (~Sprocks@70.49.35.13) |
| 17:23.36 | [TK]D-Fender | And did you enable videsupport under general? |
| 17:24.22 | BeachBall | yes |
| 17:24.24 | BeachBall | [May 28 13:24:08] WARNING[23333][C-00000a54]: chan_sip.c:10215 process_sdp: Ignoring video stream offer because port number is zero |
| 17:24.32 | BeachBall | i made it = yes |
| 17:26.28 | [TK]D-Fender | And what version are you running? |
| 17:26.58 | BeachBall | 11 |
| 17:28.51 | [TK]D-Fender | now try the whole number.... |
| 17:29.32 | *** join/#asterisk TechSmurf (~jdaniel@unaffiliated/techsmurf) |
| 17:30.00 | BeachBall | 11.8.1 |
| 17:31.30 | [TK]D-Fender | upgrade first |
| 17:31.49 | BeachBall | why? |
| 17:31.51 | [TK]D-Fender | then come back with SIP debug for the failed call. |
| 17:31.56 | BeachBall | :{ |
| 17:32.03 | [TK]D-Fender | Because I've seen posts of bugs having been fixed concerning this |
| 17:32.18 | *** join/#asterisk bmurt (~brendan@static-96-245-76-214.phlapa.fios.verizon.net) |
| 17:39.05 | *** join/#asterisk af_ (~af@93-43-45-195.ip90.fastwebnet.it) |
| 17:43.36 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
| 17:44.21 | Micc | Does asterisk have problems playing mp3 files sometimes? I have customers reporting speed up and slow downs of the audio in mp3 moh. |
| 17:44.35 | BeachBall | i just do my make && make install |
| 17:44.35 | BeachBall | to upgrade? |
| 17:44.48 | [TK]D-Fender | if compiling over, yes |
| 17:45.02 | [TK]D-Fender | Micc: no VBR, no ID3 |
| 17:50.32 | *** join/#asterisk aness (~aness@2a02:fe0:c310:57f0:ac5c:f276:fbc2:7e18) |
| 17:51.42 | *** join/#asterisk dumby_PC (~dumby@204.246.140.162) |
| 17:58.12 | *** join/#asterisk dumby (~dumby@204.246.140.162) |
| 18:00.52 | *** join/#asterisk mokmeister (~mokmeiste@78.17.37.109) |
| 18:02.20 | BeachBall | D-fender |
| 18:02.21 | BeachBall | http://pastebin.com/Le8NF3ah |
| 18:02.24 | BeachBall | there is my debug |
| 18:02.29 | BeachBall | i upgraded |
| 18:03.06 | *** join/#asterisk Zogot (~Adium@90-145-116-55.bbserv.nl) |
| 18:03.21 | Zogot | hey all, any reason what can cause a registered sip peer to have the wrong registered contact? |
| 18:03.31 | *** join/#asterisk dumby_PC (~dumby@204.246.140.162) |
| 18:03.33 | Zogot | someone is calling clienta-100 and reaching clientz-400 |
| 18:04.03 | [TK]D-Fender | Show us |
| 18:04.08 | Zogot | when I do sip show peer clienta-100 it shows Reg. Contact sip:clenz-400@clientzip |
| 18:04.27 | Zogot | sorry, @asterisk ip, but the Add->IP is the clientzip |
| 18:04.32 | [TK]D-Fender | because a client reg'd to that peer and that's the return contact they provided |
| 18:05.18 | Zogot | its safe for me to paste the output of sip show peers? |
| 18:05.38 | Zogot | this is also slightly odd as this changed middle of the day |
| 18:06.03 | Zogot | i restarted asterisk and that didn't solve it. I get a custom to restart their phone and they get the current registered IP but still the wrong registered contact |
| 18:06.10 | Zogot | customer* |
| 18:06.18 | Zogot | correct ip* |
| 18:07.03 | Zogot | [TK]D-Fender: a line from sip show peers https://gist.github.com/zogot/cf327bcdb2121d6ccfd0 |
| 18:07.41 | [TK]D-Fender | Doesn't help. You need to look at the registration attempt itself |
| 18:07.52 | [TK]D-Fender | Restarting * doesn't make the phone reregister either.... |
| 18:07.57 | Zogot | o |
| 18:07.59 | Zogot | what does? |
| 18:08.21 | [TK]D-Fender | Restarting the phone |
| 18:08.23 | Zogot | sip unregister for each broken one? |
| 18:09.59 | Katty | you know what. |
| 18:10.09 | Katty | life can go shove it this month. |
| 18:10.13 | [TK]D-Fender | I know what. Don't ask me why. |
| 18:10.26 | Zogot | [TK]D-Fender: no other solution other than restarting the phone? |
| 18:10.39 | Katty | Zogot: i have a great solution |
| 18:10.42 | Katty | Zogot: quit. |
| 18:10.47 | Zogot | ? |
| 18:10.47 | Katty | Zogot: go find a job in a less stressful occupation. |
| 18:10.51 | Zogot | lol |
| 18:10.55 | [TK]D-Fender | Zogot: You could sit around and wait for the registration to expire and that it re-reg's.... |
| 18:11.05 | Zogot | [TK]D-Fender: fair enough |
| 18:11.13 | Zogot | [TK]D-Fender: thanks for your help though, much appreciated |
| 18:11.24 | Zogot | its a holiday tomorrow so that should give it enough time |
| 18:11.33 | [TK]D-Fender | goes to watch the latest episode of "Paint Drying vs Astro-Turf Growing" |
| 18:11.48 | Zogot | sip reload couldn't do this could it btw? |
| 18:11.50 | Zogot | if you would know |
| 18:12.02 | Zogot | mess up the registered peers that is |
| 18:12.16 | *** join/#asterisk Geriatrix (~Geriatrix@mail.vancouveronthenet.com) |
| 18:12.22 | [TK]D-Fender | what you see is what * received |
| 18:12.34 | _Corey_ | Katty: try rebooting... |
| 18:12.41 | Geriatrix | hey guys - anyone know about sending SMS to mobile via SMSC ? |
| 18:12.56 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 18:13.05 | Katty | _Corey_: my dad is having an angioplasty tmw morning |
| 18:13.11 | Katty | _Corey_: i don't think rebooting is going to fix that one |
| 18:13.43 | _Corey_ | Katty: Ouch, I hope all goes smooth... |
| 18:13.50 | Katty | _Corey_: you and me both. |
| 18:14.44 | *** join/#asterisk Geriatrix (~kvirc@mail.vancouveronthenet.com) |
| 18:15.07 | Geriatrix | or any other method of sending sms to mobile |
| 18:16.41 | [TK]D-Fender | your provider probably has an e-mail to SMS gateway |
| 18:16.53 | [TK]D-Fender | Or you could use chan_mobile + a cell phone. |
| 18:17.02 | [TK]D-Fender | or chan_dongle + applicable device |
| 18:17.10 | [TK]D-Fender | Get Googling.... |
| 18:17.17 | Katty | or quit. |
| 18:23.39 | Zogot | hmm i do an unregister on a test extension, and it comes back with the right ip this time |
| 18:23.46 | Zogot | but still the wrong registered contact |
| 18:24.12 | Zogot | new one: https://gist.github.com/zogot/3d3465e0122f17dc5381 |
| 18:24.28 | Qwell | Get rid of the broken proxy/ALG on the clients end. |
| 18:25.50 | Zogot | Qwell: this has effected many clients though |
| 18:25.59 | Zogot | on the same system, but many different 'tennants' |
| 18:26.52 | [TK]D-Fender | Zogot: the client is doing what it is doing. have them fix their end |
| 18:27.20 | Zogot | another one: https://gist.github.com/zogot/f48927f4366e7409cd50 what to fix though? |
| 18:27.25 | *** join/#asterisk Sprocks (~Sprocks@70.49.35.13) |
| 18:27.40 | [TK]D-Fender | Have them fix their end |
| 18:35.34 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 18:36.00 | rrittgarn | whats the ~infobot tag for info on analog cards? |
| 18:36.31 | [TK]D-Fender | Depends on what info |
| 18:36.56 | rrittgarn | specifically looking for info on brands / options for an 8 port fxo card |
| 18:37.13 | rrittgarn | i know there are many, but rather go with tried and true, vs. offbrand |
| 18:37.38 | [TK]D-Fender | basically the top 2 are Digium & Sangoma |
| 18:38.04 | [TK]D-Fender | Out of those left, there is Openvox & Rhino.... then everyone else |
| 18:38.21 | rrittgarn | and your preference of the top 4? |
| 18:38.39 | [TK]D-Fender | Nothing is really changing in that field lately, nor would I suspec much movement in the nex few odd years |
| 18:38.47 | [TK]D-Fender | Personal is Sangoma. |
| 18:39.14 | rrittgarn | Cool thanks |
| 18:43.02 | drmessano | ~analog |
| 18:43.02 | infobot | extra, extra, read all about it, analog is Analog refers to a representation of a quantity that varies over any continuous range of values. Analog signals can be thought of as pure in nature and not processed. Thus, the debate over whether record albums (analog representation of sound) sound better than CDs (digital representation of sound). Think of nature as analog. Values are exact, but it is impossible to correct errors in reproduction. |
| 18:43.23 | drmessano | ~fxo |
| 18:43.23 | infobot | [fxo] foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx. This is the type of port found on phone or fax machine devices. See also: http://www.digium.com/index.php?menu=fxsvfxo |
| 18:43.29 | drmessano | Hmmm |
| 18:55.46 | Geriatrix | [TK]D-Fender: i know about dongling with bluetooth - however my blackberry notr my nokia support sms for some reason via asterisk |
| 18:56.09 | Geriatrix | [TK]D-Fender: however - there is a way to go directly to SMSC ? |
| 18:56.53 | [TK]D-Fender | via what? |
| 18:57.04 | [TK]D-Fender | Asterisk is a piece of software. It has to talk using SOMETHING |
| 18:57.28 | Geriatrix | i am assuming that SMSC has a protocol of some sort asterisk can send message to via internet ? |
| 18:57.47 | Geriatrix | just liek you can email to SMS on some providers |
| 18:57.57 | *** join/#asterisk Sprocks (~Sprocks@BMTNON3746W-LP140-03-845507938.dsl.bell.ca) |
| 18:57.57 | [TK]D-Fender | no, everything else as an intermediary is a gateway. |
| 18:58.05 | Geriatrix | i dont' want to depend on phone bing in bluetooth range |
| 18:58.30 | Geriatrix | or have to have another phone with monthly plan just to send sms - as there is no just sms plan - you need voice etc... |
| 18:58.34 | [TK]D-Fender | Ans as I said, many cell carriers have their own e-mail > SMS gateways |
| 18:58.51 | rrittgarn | You could also look into SMS Gateway services |
| 18:59.00 | [TK]D-Fender | And there are plenty of other services that offer SMS sending. |
| 18:59.19 | [TK]D-Fender | But there is no magical global Asterisk option for this. |
| 18:59.27 | [TK]D-Fender | Service comes from service providers. |
| 18:59.39 | jameswf | In the US I don't know of any national carrier that doesn't support email sms |
| 19:00.16 | Geriatrix | but thats one way only |
| 19:00.19 | *** join/#asterisk Dovid (~Dovid@185.23.174.1) |
| 19:00.30 | Geriatrix | i could not receive as i dont' have an accoutn with that provider |
| 19:00.51 | jameswf | I guess you are canadian so maybe YMMV |
| 19:01.20 | Geriatrix | no - in canada i can send out too |
| 19:01.32 | jameswf | http://www.emailtextmessages.com/ |
| 19:01.35 | Geriatrix | but how woudl i receive sms - email to sms solves only outgoing |
| 19:01.46 | jameswf | google voice |
| 19:02.15 | jameswf | nothing out of the box but surely you could hack something |
| 19:02.18 | [TK]D-Fender | depends what that service offers you for inbound. |
| 19:02.37 | [TK]D-Fender | taht's going to be for a number. You're going to be paying SOMEONE for that number... |
| 19:02.44 | Geriatrix | no google voice in canada yet that i am aware of |
| 19:02.59 | rrittgarn | http://bandwidth.com/business/add-voice-sms-your-apps |
| 19:03.09 | [TK]D-Fender | About to be no Google Voice ANYWHERE..... |
| 19:03.12 | Geriatrix | i know i have to pay for the service - i can't find what service offers what i need :) |
| 19:03.18 | rrittgarn | They have an API, as im sure others have too |
| 19:12.41 | TazzNZ | Geriatrix: I have yet to hear of a provider that will allow public access to the SMSC |
| 19:13.11 | Geriatrix | i dont' mean pulic - i figured i needed an account |
| 19:21.05 | *** join/#asterisk Cubber (~ronny@mail.adirondackitsolutions.com) |
| 19:21.55 | Cubber | I have been trying to update an AsteriskNow installation for months now but getting the following error about dadhi on a yum update: http://pastebin.com |
| 19:23.09 | *** join/#asterisk darkdrgn2k (~darkdrgn2@209.90.253.66) |
| 19:23.32 | darkdrgn2k | a=fmtp:102 0-15 <- is this the video codec |
| 19:24.27 | Cubber | I appologize the full pastebin url did not copy over: http://pastebin.com/mQycj8as |
| 19:24.57 | Qwell | pokes mjordan |
| 19:26.06 | drmessano | OOPS |
| 19:27.13 | *** join/#asterisk zerick (~eocrospom@190.118.43.113) |
| 19:30.52 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 19:33.37 | Geriatrix | Cubber: what kernel is installed ? |
| 19:38.06 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
| 19:41.36 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:45.16 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-26-100.pn.at.cox.net) |
| 19:58.22 | *** join/#asterisk mattcen (~mattcen@c110-22-201-130.sunsh4.vic.optusnet.com.au) |
| 20:08.18 | *** part/#asterisk ManxPower (~igcewieli@ip98-183-26-100.pn.at.cox.net) |
| 20:21.42 | *** join/#asterisk t4nk523 (40cbec93@gateway/web/freenode/ip.64.203.236.147) |
| 20:23.35 | t4nk523 | anyone care to take a crack at one hell of a puzzle? I can't figure out for the life of me, why my incoming calls are fine, but outgoing ring busy on a broadvoice SIP trunk. |
| 20:23.49 | t4nk523 | full problem was posted here, with logs and such http://forum.elastix.org/viewtopic.php?f=3&t=128837 |
| 20:24.28 | t4nk523 | and i can post any other config you may want to see, firewall, router, asterisk, or otherwise |
| 20:25.21 | t4nk523 | ill mail a beer to whoever can crack it, because I'm at the end of my rope after stumping two levels of broadvoice techs, and every soul kind enough to hear me through to the end. |
| 20:26.50 | sekil | what does sip traces show |
| 20:26.55 | elguero | t4nk523: May 28 14:02:46WARNING[5856] chan_sip.c: |
| 20:26.55 | elguero | Received response: "Forbidden" from '"8436363398" <sip:8436363398@64.253.17.237>;tag=as02e0f0de' |
| 20:27.11 | sekil | looks like 404 |
| 20:27.12 | sekil | 403 |
| 20:27.32 | *** join/#asterisk CeBe (~CeBe@brsg-4dbd770f.pool.mediaWays.net) |
| 20:28.07 | sekil | probably something in the invite bothers them |
| 20:28.25 | [TK]D-Fender | "sip set debug on" <---- which you should be looking at |
| 20:28.38 | sekil | yeah |
| 20:29.19 | [TK]D-Fender | fromuser=8436363398 |
| 20:29.28 | [TK]D-Fender | fromuser=8436363398 <- change to "defaultuser=..." |
| 20:29.32 | [TK]D-Fender | oops.. |
| 20:29.35 | [TK]D-Fender | add "fromuser |
| 20:29.41 | t4nk523 | yeah, I'll dump a debug in a moment. |
| 20:29.49 | [TK]D-Fender | username=8436363398 <- change THIS one to "defaultuser" |
| 20:33.07 | [TK]D-Fender | checkout time, BBIAB |
| 20:34.16 | *** join/#asterisk aness (~aness@cm-84.215.80.229.getinternet.no) |
| 20:37.55 | *** join/#asterisk Qwell (north@pdpc/sponsor/digium/Qwell) |
| 20:37.57 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 20:40.50 | t4nk523 | http://pastebin.com/iu14bzBp |
| 20:40.57 | t4nk523 | last few pages of debug |
| 20:41.17 | t4nk523 | still offering a beer to whoever can figure it out. I'm gonna be reading over it now |
| 20:43.52 | sekil | SIP/2.0 403 From: URI not recognized |
| 20:46.25 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 20:46.52 | Cubber | Geriatrix: 2.6.18-348.6.1.el5 |
| 20:47.20 | Cubber | Geriatrix: CentOS release 5.9 (Final) it is an AsteriskNow installation |
| 20:50.41 | mjordan | Qwell: huh? |
| 20:56.12 | *** join/#asterisk Nugget (nugget@rennsport.macnugget.org) |
| 20:57.58 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:59.49 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 21:05.07 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 21:11.44 | *** join/#asterisk hfp (~hfp@MTRLPQ0736W-LP130-01-2925269581.dsl.bell.ca) |
| 21:11.51 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:12.34 | *** join/#asterisk DelphiWorld (~TheVoicer@openvpn/user/DelphiWorld) |
| 21:13.01 | DelphiWorld | hey guys |
| 21:13.17 | DelphiWorld | guys, anyone have a click to dial js code for webrtc? |
| 21:14.34 | *** join/#asterisk KNERD (~KNERD@24.175.253.226) |
| 21:16.06 | navaismo | for free stuff google is your friend |
| 21:16.24 | DelphiWorld | navaismo: yeah, mr doubango |
| 21:16.53 | navaismo | uh? |
| 21:17.37 | DelphiWorld | navaismo: isn't you use the Doubango framework? |
| 21:17.37 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 21:18.04 | navaismo | more than jssip yes |
| 21:18.18 | DelphiWorld | navaismo: yeah i knew you |
| 21:18.24 | navaismo | so based on that im also a mr Asterisk too |
| 21:18.27 | KNERD | What is the methos to completely remove dahdi? Cannot do a "make uninstall " |
| 21:19.02 | DelphiWorld | navaismo: so help me with sipml5? |
| 21:19.22 | DelphiWorld | navaismo: no registration, no user / pass. i just want to dial a sip uri |
| 21:20.40 | navaismo | use this and dont register-->http://asterisktools.blogspot.mx/2013/07/sistema-de-atencion-al-cliente-con.html |
| 21:23.09 | DelphiWorld | navaismo: lol, my tts is dumb:) |
| 21:24.07 | KNERD | Seems there is no script to remove dahdi (how stupid is that?) |
| 21:24.56 | KNERD | WHat about removing the ability of DAHDI to not load certain modules? |
| 21:25.09 | Chainsaw | That's a double negative. |
| 21:25.25 | Chainsaw | What are we doing? Blacklisting modules? Overriding blacklists because we can't change the original config files? |
| 21:25.35 | navaismo | DelphiWorld, just use the code discard all |
| 21:27.38 | KNERD | I have a module which was compiled with incorrect hardware |
| 21:27.44 | KNERD | I deleted that module |
| 21:28.01 | KNERD | now I am gettin an error saying it cannot find it |
| 21:28.18 | KNERD | wanpipe: FATAL: Module wanpipe not found. |
| 21:28.58 | *** join/#asterisk Qwell (north@pdpc/sponsor/digium/Qwell) |
| 21:28.59 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 21:31.32 | navaismo | recompile wanpipe |
| 21:31.52 | KNERD | ghee...why didn't I think of that? </rolls eye> |
| 21:32.09 | KNERD | I get errors when I di |
| 21:32.11 | KNERD | do |
| 21:32.22 | jzaw | is anyone using blink sip client with asterisk ? (either with or without tls srtp etc) ? |
| 21:32.53 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 21:33.10 | navaismo | well if you pastebin all and provide all the info i can stop guessing |
| 21:34.35 | KNERD | there is nothing to paste bin |
| 21:34.57 | KNERD | I need to get DAHDI to stop trying to load that wanpipe configuration |
| 21:36.00 | KNERD | Loading DAHDI hardware modules: wanpipe: FATAL: Error inserting wanpipe (/lib/modules/2.6.18-371.8.1.el5/kernel/drivers/net/wan/wanpipe.ko): Unknown symbol in module, or unknown parameter (see dmes |
| 21:36.04 | KNERD | there you fo |
| 21:36.05 | KNERD | go |
| 21:36.26 | navaismo | see dmesg |
| 21:36.33 | navaismo | or the wanpipe log |
| 21:36.47 | KNERD | then when I try to recompile wanpipe - Configuring T1/E1 cards [A101/A102/A104/A108/A116/T116] ------------------------------------------------------- No Sangoma ISDN T1/E1 cards detected |
| 21:36.51 | navaismo | or see if is in the /etc/dahdi/modules |
| 21:37.18 | KNERD | AGAIN: I need to get DAHDI to stop using that module |
| 21:37.33 | navaismo | AGAIN: or see if is in the /etc/dahdi/modules |
| 21:37.45 | KNERD | it was compiles with an incorrect option |
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| 21:38.46 | navaismo | ok, usually a "recompile all " policy works but ... |
| 21:38.51 | navaismo | shrugs |
| 21:40.01 | KNERD | make: *** No rule to make target `recompile'. |
| 21:40.04 | KNERD | I shrug too |
| 21:40.43 | navaismo | i mean: make clean && make install etc etc |
| 21:41.11 | navaismo | but seriously check if /etc/dahdi/modules has the module in it, if so delete or comment it |
| 21:42.01 | KNERD | yes yes I tried all that business |
| 21:42.21 | KNERD | I solved it..I just compiled an older version of dahdi |
| 21:42.34 | KNERD | they really need to fix that dahdi garbage |
| 21:43.03 | KNERD | but thanks for the input anyway |
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| 22:23.06 | Cubber | jameswf: so just run yum update on the kernel only before the full system? This is a production system I don't want to just try things to try them because there are almost 60 phones on this system. |
| 22:26.05 | spicyramen_ | ciao anyone has deployed asterisk 11+ version with HD video phones (720p video quality), if so which phone is recommended? |
| 22:36.44 | jmmills | is there a way to add a codec allow from cli for a sip user? |
| 22:37.06 | TazzNZ | jmmills: not that I know off or have seen |
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| 22:59.26 | Cubber | I just updated my AsteriskNow system and now I get a segfault when asterisk starts and freepbx does not come up instead I get an error unknown type: [fselect] |
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| 23:05.01 | willwh | bah no moy :( |
| 23:05.26 | willwh | so looks like chrome 35 broke any chance of straight sipjs type clients and native asterisk |
| 23:05.29 | willwh | blah |
| 23:22.29 | navaismo | add another 1 to the chrome 35 list |
| 23:22.51 | willwh | :P |
| 23:22.53 | willwh | navaismo: haha |
| 23:22.56 | willwh | you did warn me >.> |
| 23:23.01 | willwh | at least I am pretty sure it was you |
| 23:25.01 | navaismo | maybe |
| 23:25.31 | navaismo | but sipml5 forum and asterisk forum are filled with that issue already |
| 23:25.41 | willwh | https://groups.google.com/forum/#!topic/doubango/iQ_8rgpbm0k |
| 23:25.51 | navaismo | o the joys of non standard "protocol"(?) |
| 23:26.00 | willwh | \m/ |
| 23:27.44 | willwh | DTLS in asterisk is a no go? |
| 23:28.51 | navaismo | dont know a while ago i gave up with webrtc |
| 23:28.58 | willwh | wuss :) |
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| 23:41.23 | navaismo | in the jira page some people was submitting patches |
| 23:42.37 | mjordan | I'm going to be brutally honest here. If you can't develop the systems you are prototyping WebRTC with, you probably shouldn't be playing around with it. |
| 23:43.31 | mjordan | If you can, then by all means, rock it out. But you've got three moving targets: the browsers (which are all over the map), the client libraries, and - slightly more rarely - Asterisk (because we aren't chasing Google, did that in the past, it sucked a lot, thanks please come again) |
| 23:44.17 | mjordan | We've had DTLS in Asterisk since 11.0.0. At the time, no one had an implementation that was free - no browsers, no soft phones, no one. Nada. We tested it between two Asterisk instances, checked the spec, yada yada, it worked. |
| 23:44.36 | mjordan | Since then, there's been a few minor tweaks, but nothing major. Now the browsers _finally_ mandate it, and everyone's all in a rush? Yeah, okay. |
| 23:45.08 | mjordan | Let me go drop the legitimate features, bugs, and everything else in the world to go chase after the world, now that the world has _finally_ caught up. |
| 23:45.09 | mjordan | Pass. |
| 23:45.14 | mjordan | It'll get worked out. |
| 23:45.20 | mjordan | But I'm not losing any sleep over WebRTC. |
| 23:45.23 | mjordan | waves |
| 23:46.50 | navaismo | good can we replicate this answer like in... everywhere |
| 23:46.55 | navaismo | oh too late |
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