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| 19:08.54 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 19:08.54 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.7.0 (2013/12/17), 1.8.25.0 (2013/12/17); Standard: Asterisk 12.0.0 (2013/12/20); DAHDI: DAHDI-linux 2.8.0.1 (2013/12/19), DAHDI-tools 2.8.0 (2013/12/06); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 19:15.27 | Blashyrkh | i have a problem with dahdi and asterisk, i have wanpipe 7 with dahdi 2.8.0.1 with HWEC and asterisk 10.3 running. |
| 19:15.42 | *** join/#asterisk navaismo (~navaismo@189.241.98.34) |
| 19:16.08 | Blashyrkh | when i call some numbers with a registered client, i dont get any sound, but when i record it with dahdi_monitor i can hear it |
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| 19:33.47 | boom^time | Does anyone know if there is any theoretical limit to the amount of dial plan asterisk can handle? |
| 19:36.15 | *** join/#asterisk wizbit (~wizbit@unaffiliated/wizbit) |
| 19:36.30 | wizbit | when i start asterisk i get this message: Privilege escalation protection disabled! |
| 19:37.00 | [TK]D-Fender | wizbit: This is a warning for AMI |
| 19:37.02 | Penguin | Is it configured to run as root? |
| 19:37.13 | [TK]D-Fender | new option for function calls |
| 19:37.14 | wizbit | yes im running it as root user |
| 19:37.21 | wizbit | is root user dangerous for asterisk? |
| 19:37.40 | [TK]D-Fender | potentially lethal |
| 19:38.05 | wizbit | is there a conf inside /etc/asterisk to pick another user? |
| 19:38.20 | [TK]D-Fender | any weakness in AMI = hijacked system |
| 19:38.30 | wizbit | :-S |
| 19:38.37 | [TK]D-Fender | wizbit: it's runs as whatever user your startup script told it to |
| 19:38.44 | wizbit | ill check |
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| 19:40.01 | wizbit | my rc script doesnt do much |
| 19:40.02 | wizbit | <PROTECTED> |
| 19:40.03 | wizbit | <PROTECTED> |
| 19:40.45 | wizbit | -U <user> Run as a user other than the caller |
| 19:40.47 | Penguin | If the startup script does not define the user, will asterisk.conf's setting of user/group be obeyed? I can't remember if I tried it. |
| 19:40.47 | wizbit | i see! |
| 19:41.06 | wizbit | ill create a user called asterisk |
| 19:41.11 | wizbit | what doesnt have a login shell |
| 19:41.33 | Penguin | asterisk:x:5060:5060:Asterisk daemon:/var/lib/asterisk:/bin/false |
| 19:41.39 | wizbit | aye up! |
| 19:41.52 | [TK]D-Fender | ~book |
| 19:41.52 | 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 |
| 19:41.57 | [TK]D-Fender | ^^^ it's in there |
| 19:42.23 | wizbit | thanks |
| 19:42.29 | wizbit | another problem is to do with my polycom |
| 19:42.35 | wizbit | when it boots this shows in the log: |
| 19:42.36 | wizbit | 0203113857|copy |3|00|Download of '0004f2251870.cfg' FAILED on attempt 1 (addr 1 of 1) |
| 19:42.47 | wizbit | 0004f2251870.cfg exists and i can access it via ftp |
| 19:42.56 | wizbit | the phone logs in via ftp ok for the log file |
| 19:43.50 | wizbit | 0203113857|copy |3|00|transport res: 78 respCode 550 |
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| 19:47.40 | *** join/#asterisk bsdice (~bsdice@embinet.eu) |
| 19:48.41 | bsdice | good news |
| 19:49.01 | bsdice | my phone still works :) |
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| 19:59.23 | BarthezZ | hmm, is there a way to preload realtime sip peers on 1.8? I have one peer identified by IP which currently is in realtime config, but before someone called that peer it isn't loaded; so it seems calls for it are rejected |
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| 20:10.35 | leifmadsen | BarthezZ: sip show peer <realtime_peer> load |
| 20:11.09 | leifmadsen | BarthezZ: beyond that, they are not loaded until register or call from peer after a system restart / chan_sip reload |
| 20:11.30 | BarthezZ | yeah i know that one... but i should have it triggered after a sip reload |
| 20:11.34 | BarthezZ | which was exactly what i feared :) |
| 20:11.34 | BarthezZ | :(* |
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| 20:30.31 | BarthezZ | hmm but leifmadsen, I'm wondering if I need to create a bug report... Basically the only real identifier on the peer is the IP... So shouldn't the realtime engine do a lookup for the host column? |
| 20:30.49 | leifmadsen | BarthezZ: it's not a bug |
| 20:31.18 | BarthezZ | I can totally understand how this is by design though... Just this use case isn't supported (and might be for the good) |
| 20:32.38 | leifmadsen | basically if you have rtcachefriends=yes and rtupdate=yes (default) then calls to the peer will happen even after a restart since Asterisk will attempt to place a call to the last known location |
| 20:33.06 | leifmadsen | but if the NAT connection was closed due to the router closing the connection (e.g. no OPTIONs for a period of time) there isn't anything else to do other than wait for registration |
| 20:33.11 | leifmadsen | which I've usually set to no more than 120s |
| 20:36.53 | BarthezZ | yeah, but this upstream provider is annoying and doesn't do *any* authentication... just "Hey, this is your ip... accept my call or else!" |
| 20:39.35 | *** join/#asterisk MarkS- (~mark@unaffiliated/mark21) |
| 20:41.51 | MarkS- | Hello, what function/variable should I look at when I want to get information from Asterisk related to who (what sip account/phone number) answered a call? I'm looking for an option to mail the one who was called with some information related to the call he did just make (time, caller ID, who answered, etc). |
| 20:42.26 | MarkS- | This is for a situation where multiple persons can pickup the call (Dial(SIP/1&SIP/2&SIP/3,,Tt)) |
| 20:43.15 | MarkS- | and sometimes it is forwarded using an external trunk (Dial(SIP/1&SIP/2&SIP/012345678@trunk,,Tt) for an idea) |
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| 21:03.47 | funky1 | could anyone help me analyse/compare wireshark output? my problem is described with links to wireshark output http://forums.digium.com/viewtopic.php?f=1&t=84744&p=196341&sid=e9062afa18c8455d3fa752f2f79bda2d#p196341 |
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| 21:06.52 | *** join/#asterisk powerunits (272a14eb@gateway/web/freenode/ip.39.42.20.235) |
| 21:07.57 | powerunits | i have a question. which dongles are easy compatible with asterisk.. |
| 21:09.12 | *** join/#asterisk gtjoseph (~gtjoseph@unaffiliated/gtj) |
| 21:09.19 | pabelanger | dongles? |
| 21:09.30 | powerunits | team any help on dongle version... |
| 21:10.01 | powerunits | so i can make sip call using GSM network using dongle |
| 21:10.25 | pabelanger | well, you could use bluetooth and a cell phone |
| 21:11.25 | MarkS- | or you could use a special sip -> gsm gateway |
| 21:11.49 | MarkS- | portech has a few options available that work (we use it) |
| 21:12.36 | drmessano | If you google for chan_dongle you will likely find a blog, wiki, or forum post with compatible hardware |
| 21:12.46 | powerunits | @pabelanger: what is more reiable ?? useing dongle or bluetooth? |
| 21:13.16 | drmessano | Actually, I just did and found a list |
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| 21:13.59 | powerunits | drmessano: what list you found |
| 21:14.00 | powerunits | ? |
| 21:14.03 | MarkS- | powerunits: probably dongle as bluetooth has a few additional things that can go wrong |
| 21:14.20 | drmessano | powerunits, google for "chan_dongle hardware" |
| 21:14.29 | powerunits | ok |
| 21:14.38 | drmessano | Just as I said |
| 21:14.54 | powerunits | one more qustion can i connect more then 1 dongle with asterisk and make multiple calls ? |
| 21:14.58 | drmessano | Theres a list, forum posts, the whole works |
| 21:15.11 | pabelanger | powerunits, I would use neither. Just get an external device and connect to it via sip |
| 21:16.01 | powerunits | drmessano: external dervices are very expensive |
| 21:16.06 | powerunits | i wana use dongle.. |
| 21:16.34 | powerunits | can i connect more then 1 dongle with asterisk and make multiple calls ? |
| 21:16.59 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 21:17.37 | powerunits | ? |
| 21:19.27 | pabelanger | asterisk doesn't have a native chan_dongle, so you need to ask who ever supports it |
| 21:19.51 | pabelanger | like I said, get an external device is your best bet |
| 21:20.01 | pabelanger | otherwise, you'll be hacking stuff together |
| 21:21.14 | powerunits | ok thanks |
| 21:26.45 | skrusty | evening |
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| 21:29.49 | powerunits | @pabelanger: can i connect more then 1 bluetooth with asterisk so i can make more then 1 call? |
| 21:30.27 | pabelanger | should be able too |
| 21:30.59 | powerunits | kool |
| 21:31.16 | powerunits | i never used it.. please can you tell me how will be the call quality? |
| 21:31.52 | pabelanger | unknown, never used it |
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| 21:33.57 | powerunits | ? |
| 21:34.04 | drmessano | chan_mobile works fine, but you can only make 1 call |
| 21:34.21 | drmessano | Per BT dongle |
| 21:34.37 | drmessano | So you need one phone and one bluetooth dongle per call |
| 21:34.48 | drmessano | Talk about expensive and complex |
| 21:35.04 | powerunits | if will will attach 4 bluethooth and 4 phone |
| 21:35.14 | powerunits | then in that case can i make 4 calls |
| 21:35.18 | drmessano | 4 bluetooth devices are going to work horribly |
| 21:35.32 | powerunits | humm |
| 21:35.34 | drmessano | Too much interferences from those 4 dongles together |
| 21:35.39 | drmessano | interference |
| 21:36.13 | drmessano | You'll be lucky to get two bluetooth dongles working, nevermind 4 |
| 21:36.35 | drmessano | If this was for ONE call, that would be different.. for 4 concurrent, you need a real gateway |
| 21:36.57 | powerunits | yes .. i can understand now.. |
| 21:37.54 | drmessano | http://www.ebay.com/itm/GoIP-4I-VOIP-Gateway-GSM-Converter-SIP-IP-Phone-Adapter-Asterisk-SIP-IMEI-change-/321292251657 |
| 21:38.17 | drmessano | Thats $350.. If thats too expensive, then you need to come up with another solution or use Skype |
| 21:38.27 | powerunits | the issue is in our country these devices are baaned |
| 21:38.33 | drmessano | Ohhhhhhh |
| 21:38.39 | powerunits | i can't import them |
| 21:38.42 | drmessano | I guess you're stuck then |
| 21:38.54 | drmessano | and this conversation makes me an accomplice to a crime |
| 21:39.02 | powerunits | :( |
| 21:39.03 | powerunits | yes |
| 21:39.18 | powerunits | lol |
| 21:39.26 | powerunits | rite |
| 21:39.51 | drmessano | DISCLAIMER: I NO WAY INTENT TO ASSIST IN THE DEFRAUDING OF THE GOVERNMENT OF NEW GUZBEKISTAN |
| 21:40.00 | drmessano | There |
| 21:40.05 | drmessano | Damnit |
| 21:40.07 | drmessano | INTEND |
| 21:40.18 | powerunits | ok dear |
| 21:40.44 | drmessano | If the gateway is banned, the dongles should and would be banned too |
| 21:41.04 | drmessano | Even if they slip past import, they accomplish the same |
| 21:41.13 | drmessano | So you are pretty much out of options |
| 21:41.34 | drmessano | Have you tried tying two coconuts together? |
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| 21:43.04 | powerunits | ? |
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| 21:43.05 | *** mode/#asterisk [+o file] by ChanServ |
| 21:43.05 | powerunits | i didn't get what do you mean by coconuts |
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| 22:34.29 | flippyjigger | Anyone else having Grandstream 1405s going absolutely insane? |
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