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06:36.58 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.4.0 (2015/06/04), 11.18.0 (2015/06/04), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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07:57.47 | nunne | Anyone had problems with icesupport=yes in sip.conf and rtp.conf and all of a sudden Asterisk stops accepting incomming SIP channels all together after a while. I can't see anything in any logs. It just stops right up. Really weird. |
07:59.53 | Chainsaw | nunne: Normally suggests your STUN server has stopped accepting connections. |
08:00.00 | Chainsaw | nunne: Do you host that yourself or are you using someone elses? |
08:00.16 | nunne | Chainsaw: using googles. but will have a look at it. thanks. |
08:01.12 | nunne | Chainsaw: but will remove it from my sip.conf general. I only need it for webrtc. and will install my own instead. maybe I flooded googles :P |
08:01.49 | Chainsaw | nunne: It may be an automated countermeasure on the Google server, yes. It's all speculation unless you capture the packets and compare a good run to a bad run. |
08:03.08 | nunne | Chainsaw: thanks for pointing me in the direction. I will have a look at it. But sounds like a plausible thing they might be doing. Specially since it might be over 100 call setups per second. So might as well setup a stun server myself to save on some network traffic. |
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08:04.24 | Chainsaw | nunne: "stuntman" is a decently maintained STUN server codebase. |
08:04.31 | Chainsaw | nunne: Most of the others appear to be long abandoned. |
08:05.45 | nunne | Chainsaw: thanks |
08:12.41 | yun1989 | hello every one, I need when I receive one call to "1099", generate one call from 1099 between 343 |
08:13.14 | yun1989 | I try to use the command Originate but i don't have sucess |
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08:46.52 | Ice_Strike | Is there public log of this channel? |
08:47.03 | pjensen00 | Don't think so |
08:48.19 | Chainsaw | Some people log it and put it online actually. |
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08:50.51 | Ice_Strike | Found it lol https://botbot.me/freenode/asterisk/ |
08:51.06 | Ice_Strike | real time log too, nice. |
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09:33.33 | yun1989 | hello every one, I need when I receive one call to "1099", generate one call from 1099 between 343 |
09:33.39 | yun1989 | <PROTECTED> |
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09:42.03 | Jason_weileilei | oh ,i am come in. |
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09:44.12 | Rico | is there a way to tell asterisk 13 (pjsip) tu add a route header four outgoing requests ? |
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09:47.41 | Rico | s/four/to |
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09:58.31 | Jason_weileilei | hi |
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10:23.24 | pjensen00 | hello |
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11:20.38 | cyford33 | hi |
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12:34.41 | johnny_|_ | Hi, if server responds with WWW-Authenticate: Digest nonce="1400543250:ab9fbe93a438d3bd72fd1b7553cc2847",algorithm=MD5,realm="sip.someprovider.net". Does that mean use 1400543250 nonce to get ab9fbe93a438d3bd72fd1b7553cc2847 response? |
12:37.46 | file | no, the nonce is used as-is in the response generation |
12:38.40 | johnny_|_ | thanks, I omitted the fact that it might not be asterisk, sorry |
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12:40.13 | file | the way it works is a defined way, the content of the nonce is opaque |
12:40.19 | file | if it has meaning then that's specific to the implementation |
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12:57.20 | tompaw | Morning |
12:58.03 | tompaw | Guys, I need a favor. Can someone please do a lsof | wc -l on their asterisk user (non-root) and report the results together with number of users using the pbx atm? |
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13:53.19 | tompaw | res_pjsip_pubsub.c: No registered publish handler for event presence << what's this? |
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13:55.50 | Katty | [TK]D-Fender: you are getting old. |
13:55.53 | Katty | [TK]D-Fender: but i still love you. |
13:56.47 | [TK]D-Fender | <3 |
13:56.48 | tompaw | Can I disable presence with pjsip? |
14:00.32 | MaliutaLap | hey Katty |
14:00.52 | Katty | waves to MaliutaLap |
14:02.09 | MaliutaLap | just went to the most amazing jazz gig in ages |
14:02.25 | MaliutaLap | better than Herbie Hancock and Chick Corea the other month |
14:04.05 | MaliutaLap | My mate was recording stuff on his phone - should have a bootleg mixed in a couple of days |
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14:07.43 | Katty | MaliutaLap: very nice. what sort of jazz? |
14:08.57 | Katty | we talking like big band swing? |
14:09.18 | MaliutaLap | Katty: wild, mad jazz - the kind that should be played somewhere like the Blue Note club where audience members aren't shushed for tapping beat on their legs, or can yell appreciation, and not it a concert hall where the old people suck the fun out of it |
14:09.18 | Katty | or miles davis |
14:09.32 | MaliutaLap | James Morrison with Megan Washington |
14:10.06 | Katty | that sounds lovely ^_^ |
14:10.14 | MaliutaLap | and don't mention davis around me - the man was the biggest thief in the jazz world, always took credit for other peoples music :) |
14:10.37 | MaliutaLap | the pianist was so good I expected them to pull out some Monk |
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14:16.30 | MaliutaLap | hehehe I have coined the term "Megasm" it's like an orgasm, but comes from listening to Megan Washington perform |
14:16.35 | MaliutaLap | and it's catching on |
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15:29.54 | Rico | bon week end ! |
15:34.40 | MaliutaLap | Rico it's 01:35 on a saturday - I have had a head start on you :) |
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16:13.54 | x86_64 | test |
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16:24.07 | florian333444_ | hello |
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16:27.55 | florian333444_ | question: (Asterisk 11) i need to IGNORE 1 msn. i tried to put a list into /etc/asterisk/chan_dahdi.conf in "msn=123,234,345,..." .... unfortunately this list may not be long enough, i needed to list 17 msn because i have 2 isdn-lines. ideas ? |
16:28.24 | florian333444_ | the 18th msn is the one id like to ignore, thu |
16:30.46 | florian333444_ | up to 6 or 7 msn are fine, but say msn 8-17 will be ignored (not in the msn-list ...) too |
16:31.56 | florian333444_ | wanna put it more simple: what is the easiest way to tell asterisk to ignore a specific msn ? |
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16:41.32 | rmudgett | florian333444_: The msn list in chan_dahdi.conf is per ISDN span and is only for when you have that span configured as PTMP CPE mode. The msn list string can be up to 79 characters. When used, Asterisk will ignore incoming calls with a callerid not matching a msn in the list. |
16:42.19 | florian333444_ | rmudgett thx. |
16:42.41 | florian333444_ | rmudget: but my msn-list is 17 msns long and far longer than 79 chars |
16:43.15 | florian333444_ | i tried wildcards in there like "*" or "X" to shorten my numchars. |
16:43.20 | florian333444_ | but that didnt work |
16:43.40 | florian333444_ | rmudgett: ideas ? |
16:44.52 | rmudgett | Why do you need so many MSNs? |
16:45.09 | florian333444_ | rmudgett its a small company |
16:46.20 | florian333444_ | irmudget: i just need asterisk to completly ignore ONE specific msn .... |
16:46.28 | florian333444_ | *rmudget |
16:46.31 | rmudgett | That list is to allow other devices on a multidrop configuration to share the line so Asterisk will not attempt to steal calls from another device. |
16:46.56 | rmudgett | If you don't have any other device on the line then you don't need to use the list. |
16:47.15 | florian333444_ | ok i understand |
16:47.28 | florian333444_ | mmm i have a fax-device on that line |
16:47.34 | florian333444_ | but its not fax, its edi |
16:48.14 | florian333444_ | so i cannot use forward-fax within elastiks because it doesnt recognize edi as fax |
16:48.37 | florian333444_ | so i want asterisk to ignore ONE msn |
16:49.02 | florian333444_ | *the protocol used is not fax, but edi |
16:49.39 | florian333444_ | rmudget: do you have an idea howto tell asterisk to ignore ONE msn = |
16:49.42 | florian333444_ | *? |
16:51.00 | rmudgett | When that list is not empty it tells asterisk which calls to accept. If it's not listed it won't respond. |
16:51.02 | florian333444_ | if the msnlist was up to say 250 characters i was fine, but its 79 |
16:51.17 | rmudgett | There isn't an option to do the opposite. |
16:52.01 | florian333444_ | can i use wildcards inside chan_dahdi.conf ? if so, could you tell me how than looks like ? |
16:52.53 | florian333444_ | ...or msn-abreviations or so ? |
16:54.18 | florian333444_ | ...or tell 2 lists, each one for a span ? |
16:57.06 | rmudgett | florian333444_: Just looked at the code and the list can be made up of dialplan extensions including dialplan patterns "_NXXNXXXXXX" |
16:57.32 | rmudgett | That should work for you |
16:58.12 | florian333444_ | ahhh... |
16:59.32 | florian333444_ | so msn="_2X3X,_4XX44" and so forth ? |
16:59.41 | rmudgett | yes |
16:59.55 | florian333444_ | cool ... so cool i check it right now off course ! |
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17:01.26 | rmudgett | Hah. It was even stated in the chan_dahdi.conf.sample file. |
17:01.29 | rmudgett | ; Here you can give a comma separated list of numbers or dialplan extension |
17:01.29 | rmudgett | ; patterns. An empty list disables MSN matching to allow any incoming call. |
17:01.47 | florian333444_ | really ? oh so bad i missed that im sorry |
17:02.02 | florian333444_ | would you be so kind an paste that line here ? |
17:03.55 | florian333444_ | rmudget: i think i did it............. HOORAYYY ! thx a lot !!!!!! |
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17:09.21 | florian333444_ | thx so much ! a bunch of flowers to your room !! |
17:09.24 | florian333444_ | cu !!! |
17:11.30 | ryao | I have never used Asterisk. At the moment, my home has an Obi100 ATA connecting the POTS phones to Google Voice as well as a fax machine. Would it be possible to install a FXS into a supermicro system running Open Indiana, setup Asterisk to connect to Google Voice and then be able to receive and send regular calls as well as faxes using something like mgetty or hylafax{,plus}? |
17:25.22 | cyford33 | is there a tool to check dialplans for errors |
17:29.26 | file | not really |
17:29.39 | file | syntax stuff will be output on load |
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18:22.10 | pjensen00 | Switch to ARI. Come to the dark side! |
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20:20.42 | JimDickenson | I am trying to get asterisk 13 going and VM_INFO seems to not be present |
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20:21.41 | johnny_|_ | Hi. regarding https://issues.asterisk.org/jira/browse/ASTERISK-25265 issue. I've raised this issue here one week ago. To recap: calls _to_ firefox jssip client fail. Now I've applied this * 13.1 patch to * 11.18.0. It's still not working (I'll post logs) but let me just ask this to make it clear. Can I make calls to firefox using SRTP media encryption (codec is opus and vp8 passthrough) or do I _have_ to |
20:21.43 | johnny_|_ | use DTLS? |
20:24.44 | JimDickenson | How does one find out if a mailbox exists in Asterisk 13? |
20:27.31 | [TK]D-Fender | voicemail <tab> from cli |
20:27.52 | [TK]D-Fender | and there are dialplan functions for that at that level as well |
20:28.31 | rmudgett | VM_INFO still exists. Is app_voicemail loaded? |
20:41.07 | JimDickenson | As far as I know. VMCOUNT is present but not VM_INFO. It does not show up in core show functions nor when I try to use it. |
20:42.47 | rmudgett | VMCOUNT is supplied by a different module. |
20:43.02 | rmudgett | VM_INFO is supplied by app_voicemail |
20:50.43 | JimDickenson | Thanks. I did not have app_voicemail.so |
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21:15.28 | rexwin_ | is it okay to stop fail2ban-server as it taking up a lot of resources? |
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22:21.12 | Ice_Strike | If a call center have 500+ agents for outbound calls and put all in 1 astrisk box is asking for trouble. What is your solution? |
22:22.00 | WIMPy | Didn't you find the answer to the 2nd part yesterday? |
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22:23.55 | Ice_Strike | Yesterday? |
22:24.04 | Ice_Strike | I don't remmeber about that answer |
22:24.09 | Ice_Strike | I will scroll up |
22:24.41 | cyford33 | probablly was dont put all in 1 box |
22:26.46 | Ice_Strike | WIMPy yea yesterday was about ARI - creating own Queues application to seperate server and then scale to mulitple servers. Yea that would work. |
22:30.44 | [TK]D-Fender | you said "outbound calls" |
22:30.49 | [TK]D-Fender | how does taht relate to Queues? |
22:31.24 | WIMPy | Optimizing bad impressions :-) |
22:31.46 | WIMPy | Or sometimes getting your line cut by some watchdog. |
22:33.30 | Ice_Strike | [TK]D-Fender Yea I got mixed up, my bad. |
22:34.18 | [TK]D-Fender | and you haven't clarified it either |
22:34.24 | [TK]D-Fender | so which is it |
22:34.37 | [TK]D-Fender | because 500 calls through 1 decent server isn't ahuge deal yet |
22:37.08 | Ice_Strike | Someone said to me yesterday that 500 agents on one host require very complex system(like kamailio with custom code). not doable on asterisk. |
22:39.39 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
22:39.56 | [TK]D-Fender | "someone said" sounds like vague crap. Especially since your needs are never expressed properly |
22:45.12 | Ice_Strike | [TK]D-Fender Remember diagram I show you few days ago? Basically Originate a call, Caller Answered, AMD Check Passed and then passed to the Queue |
22:45.21 | Ice_Strike | Using AMI to do all that |
22:45.42 | [TK]D-Fender | How many calls SITTING in the queue at a time? |
22:47.15 | cyford33 | [TK]D-Fender i found the reason my server goes bizzark after using busy().................... if i dont answer the call on a 1800 number an send a busy the networks lcr and re route the calls ... so with 1 call i end up with 4 calls to hangup |
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22:48.10 | WIMPy | That still doesn't explain the difference to using Hangup. |
22:48.11 | Ice_Strike | [TK]D-Fender It is assumption and scenario. Let say 300? |
22:48.20 | cyford33 | i noticed when it kepted originating on same priority it was using differnt channels |
22:49.14 | [TK]D-Fender | Ice_Strike, so 300 actually WAITING for agents? |
22:49.17 | cyford33 | someone said busy() and conjustion() did something to the channel |
22:49.21 | [TK]D-Fender | This is horrible |
22:49.33 | Ice_Strike | [TK]D-Fender Oh lol. No |
22:49.36 | cyford33 | <PROTECTED> |
22:49.48 | Ice_Strike | [TK]D-Fender Let say max 50 |
22:50.02 | [TK]D-Fender | Ice_Strike, so use that link as a reference |
22:50.10 | Ice_Strike | [TK]D-Fender 500 agents is already taken and talking to customers. |
22:50.18 | Ice_Strike | ok |
22:50.20 | [TK]D-Fender | "ready " is not a LOAD |
22:50.23 | [TK]D-Fender | LOAD = WORK |
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22:51.53 | Ice_Strike | Ah ok |
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22:52.52 | cyford33 | <rmudgett> Busy() and Congestion() are not meant to be used in the h exten. Both of those applications indicate to the channel driver a busy/congestion signal and wait forever for the channel to hangup. |
22:53.34 | WIMPy | Yes, but it's not being used in h. |
22:53.49 | Ice_Strike | Can Asterisk queues() use external database for storage? |
22:54.17 | cyford33 | yes, but as soon as my 1800 number provider gets the signal they reroute |
22:54.48 | cyford33 | if i use hangup. py provider keeps getting a reinvite and keeps the line active |
22:54.59 | cyford33 | for 20 secs or 3-4 retries |
22:55.17 | WIMPy | A reinvite after hangup? That doesn't make sense. |
22:55.24 | cyford33 | or invite |
22:55.37 | Ice_Strike | Ah just reading http://manuals.loway.ch/QM_UserManual-chunked/ar01s19.html |
22:55.41 | cyford33 | <PROTECTED> |
22:55.44 | Ice_Strike | Storing queue data on MySQL |
22:55.46 | WIMPy | Same |
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22:57.04 | Ice_Strike | Might be useful for asterisk queue failover destination? |
22:57.27 | cyford33 | so all senerios end up multiple invites if not answering the freaking 1800 number... but if i use hangup() the line never ends till the last one finnishes... and if i use busy() the line ends but the same call comes in like 3-4 times |
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22:58.43 | cyford33 | if i answer and hangup() its good, or answer and busy() its good.. but then i get hit with 85 cents payphone fees lol |
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