| 00:00.29 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
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| 00:27.18 | volga629 | Hello Everyone, trying troubleshoot issue with RTP https://fpaste.networklab.ca/Pd8A/ What is mean type rtp in debug ? |
| 00:32.36 | *** join/#asterisk morfin (~morfin@morfin.telenet.ru) |
| 00:32.39 | morfin | hello |
| 00:33.23 | morfin | how can i add h extension in realtime? |
| 00:37.56 | ChannelZ | type really really quickly |
| 00:39.01 | morfin | lol |
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| 02:00.33 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 04:00.58 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 04:16.20 | volga629 | Hello Everyone, trying troubleshoot issue with RTP https://fpaste.networklab.ca/Pd8A/ What is mean type rtp in debug ? |
| 04:17.04 | [TK]D-Fender | link is no good |
| 04:17.15 | volga629 | ups just sec |
| 04:17.48 | volga629 | https://fpaste.networklab.ca/Pd8A/ I just opened it is not working for you ? |
| 04:18.11 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ccdtgpcxsmlcphsc) |
| 04:18.53 | volga629 | http://fpaste.org/rxtp/ |
| 04:21.18 | [TK]D-Fender | That's the message. What is the actual problem you are having that this seems to be related to? |
| 04:23.15 | volga629 | yes look like. I am trying track down |
| 04:23.31 | volga629 | yes you was write about first link |
| 04:24.44 | volga629 | it should come back on-line right now |
| 04:28.10 | volga629 | I checked MTU on tunnels I checked firewall setting to make sure no any nat module will be loaded. I checked both sip trunk setting |
| 04:29.08 | volga629 | And can't find any useful information online about this issue, except physical media |
| 04:33.03 | *** join/#asterisk radic (~radic@dslb-178-010-253-206.pools.arcor-ip.net) |
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| 05:15.05 | volga629 | I will continue work tomorrow on this issue |
| 05:18.43 | *** part/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
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| 07:09.46 | *** join/#asterisk schmidts (~schmidts@77.116.33.187.wireless.dyn.drei.com) |
| 07:09.48 | schmidts | good morning |
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| 07:34.48 | *** join/#asterisk bruce_ (~bruce_@41.177.76.17) |
| 07:34.57 | bruce_ | hey peeps |
| 07:35.04 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 07:35.06 | greenwolf | sup bruce? |
| 07:35.11 | greenwolf | how is everyone tonight? |
| 07:35.27 | bruce_ | greenwolf, yo man. I was told that you the guy to speak to with this issue I have |
| 07:35.28 | bruce_ | haha |
| 07:35.41 | greenwolf | really? |
| 07:35.57 | greenwolf | ok shoot |
| 07:36.22 | bruce_ | greenwolf, I have a SIP trunk from Asterisk to some VOIP provider out there... what would I look for in the full Asterisks logs if it were to go down? |
| 07:37.09 | bruce_ | have any idea? |
| 07:37.35 | greenwolf | depends on how the trunk went down |
| 07:37.53 | bruce_ | umm... give some examples man |
| 07:38.08 | greenwolf | check the log files for the sip packets for that trunk or provider |
| 07:38.35 | greenwolf | most likely will contain SDP packets for info passed between you and the carrier |
| 07:38.36 | bruce_ | these logs are awful |
| 07:38.42 | bruce_ | I think the verbose level is like 100000 |
| 07:38.44 | bruce_ | or something |
| 07:38.45 | greenwolf | hit cntrl+F |
| 07:38.51 | greenwolf | then search trunk name |
| 07:39.06 | greenwolf | scroll until you find the error msgs not the warnings |
| 07:39.28 | bruce_ | umm.... I'm trying to do something in BASH that plugs into Nagios |
| 07:39.33 | bruce_ | but I'll check this out |
| 07:39.33 | greenwolf | i personally use nano to open the logs and then search with control W |
| 07:40.07 | *** join/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 07:40.10 | greenwolf | i see...i haven;t used Nagios so couldn't help u there...sorry |
| 07:40.16 | bombev | hi all |
| 07:40.22 | greenwolf | yea its sometimes a pain to go thru the logs |
| 07:40.26 | greenwolf | i hate it myself |
| 07:40.34 | schmidts | bruce_ maybe you should start with a grep ni "error" logfile then you allready have the right line number |
| 07:40.34 | greenwolf | sup bombev? |
| 07:40.49 | schmidts | uhh missed the "-" before ni |
| 07:42.14 | kaldemar | bruce_: what logs are you looking at? |
| 07:42.24 | bruce_ | full |
| 07:42.33 | kaldemar | the file? |
| 07:42.37 | bruce_ | yip |
| 07:43.28 | bombev | greenwolf what ? :) |
| 07:44.05 | greenwolf | bombev:i said whats up? |
| 07:44.25 | bruce_ | kaldemar, wanna help? |
| 07:44.40 | kaldemar | bruce_: if you have qualify enabled for the peer and they answer those messages, you'll see "Peer '<peername>' is now UNREACHABLE". |
| 07:45.21 | kaldemar | bruce_: other than that, i don't know what you're trying to do. are you trying to find a cause for some downtime or find a reason why something is not working right now? |
| 07:45.55 | bruce_ | it's just for monitoring... |
| 07:46.12 | bruce_ | I need a way to monitor if the sip trunk is up or not |
| 07:46.23 | bruce_ | there is some sip proxy thing in the middle though |
| 07:46.40 | bruce_ | so using nagios' sip plugins are useless |
| 07:46.40 | kaldemar | AMI can be used for that too. you'll get events for such occurrences. |
| 07:46.46 | *** join/#asterisk ghghz (~ton@kluonis.kvb.lt) |
| 07:47.52 | ghghz | Hey. Is it possible to track the path, how call is traveled? I mean via what ISPs. |
| 07:47.54 | bruce_ | grr |
| 07:50.20 | bombev | what does mean that error: SIP/2.0 501 Unsupported Method |
| 07:51.06 | bombev | greenwolf not much what about you ;) |
| 07:51.17 | bruce_ | kaldemar, AMI? |
| 07:51.48 | greenwolf | bored doing some work then going to bed soon...very tired and its late here :) |
| 07:52.31 | kaldemar | bruce_: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Manager+Interface+(AMI) |
| 07:57.55 | bruce_ | 404 |
| 08:01.00 | bombev | any idea |
| 08:01.00 | bombev | ? |
| 08:01.11 | bombev | about SIP/2.0 501 Unsupported Method |
| 08:06.54 | schmidts | bombev incoming or outgoing call? |
| 08:09.06 | ghghz | So, is it possible to trace/track a call? |
| 08:09.14 | ghghz | which provaiders it passed |
| 08:11.18 | *** part/#asterisk Russ (~russ@pool-74-100-57-85.lsanca.fios.verizon.net) |
| 08:17.36 | bombev | schmidts incoming |
| 08:26.10 | schmidts | bombev then you have to take a look at the invite message sent from this client to your asterisk what kind of codecs they offer. maybe the try to set up t38 in the initial invite and you have set t38 support to no, then you will see a 501 reply or the client offers only codecs you havent allowed |
| 08:33.54 | mariusno | tormsl: mshaugla trenger en |
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| 08:53.17 | bruce_ | hah I'm making something with nsca work |
| 08:53.22 | bruce_ | awesome stuff |
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| 09:16.45 | ghost75 | for cisco phone i need to have injector with a 48v power supply? |
| 09:18.02 | ChannelZ | eh? |
| 09:18.29 | ghost75 | ? |
| 09:20.25 | ChannelZ | ‽ |
| 09:21.03 | ghost75 | what |
| 09:21.30 | bombev | schmidts well i am using g711 codec.. i dont thik this is the problem |
| 09:22.25 | kaldemar | bombev: your sip.conf and sip debug will probably show the reason. |
| 09:23.12 | bombev | kaldemar well I already did sip debug |
| 09:24.16 | kaldemar | consider showing it to others if you can't figure it out. |
| 09:26.16 | schmidts | bombev it could be a problem if the client doesnt support g711 or maybe has it deactivated but as kaldemar said, please pastebin us the sip debug output |
| 09:26.58 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 09:28.02 | bombev | oks I'll paste it in a minute |
| 09:36.12 | *** join/#asterisk tokozedg (5cf1517e@gateway/web/freenode/ip.92.241.81.126) |
| 09:37.16 | tokozedg | greeting, does anyone have any idea how can I drop call as soon as asterisk receives 180 Ringing message or 183? |
| 09:38.11 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 09:38.49 | schmidts | tokozedg for 183 you could use the rtplimit with 1 second, cause rtp will start after a 183 is received, but for 180 you should take a look at absolut limit, but i dont know why you would need something ;) |
| 09:39.11 | ChannelZ | crank calling |
| 09:42.00 | tokozedg | schmidts: thanks for answere, I'll take a deeper look. It's kinda call back service, one client informs other I don't have balance so call me :) with single ring tone |
| 09:42.20 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 09:43.23 | schmidts | tokozedg ah ok, even an easier way would be to set the timeout really short in the Dial app but this will also close the call before receiving a ringing |
| 09:44.06 | tokozedg | schmidts: yes, that's what I'm afraid of, some calls might take more than standard time to setup, and they might got dropped. |
| 09:44.42 | danfromuk | I'm struggling to diagnose an issue. Ive got two asterisk servers set up using the same mysql database for realtime SIP Peers. The SIP clients are pointing to sip.mycompany.com which only points to the ip address of server1. |
| 09:45.17 | danfromuk | Some how, occasionally, sip peers end up registered with server2. I have verified this by using sip set debug ip. |
| 09:45.32 | danfromuk | However, there is nothing in the client confirm thats pointing to server2. |
| 09:45.43 | danfromuk | The only connection is the realtime sip peers db |
| 09:46.36 | danfromuk | This occurs with multiple hardware vendors and other asterisk boxes connecting in to my servers. |
| 09:46.51 | bombev | kaldemar schmidts: http://pastebin.ca/2305227 |
| 09:47.20 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 09:47.51 | kaldemar | bombev: they don't like your OPTIONS messages which are caused by qualify. just ignore those. |
| 09:50.02 | schmidts | danfromuk do you really see register messages to server2 or only the keep alives? |
| 09:50.21 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:b4c3:d291:18b9:fd80) |
| 09:51.41 | danfromuk | schmidts: Just waiting for the next register message to trigger. But in the last 2 minutes, it looks like only keepalives. |
| 09:51.56 | danfromuk | Strangly, both servers are receiving the keepalives |
| 09:51.57 | danfromuk | http://pastebin.com/pPcMdrTD |
| 09:53.05 | danfromuk | Both servers have the extension as registered |
| 09:54.06 | danfromuk | This is one example. Currently, around 4% of peers are registered with both servers without being given the IP of server2 |
| 10:13.33 | danfromuk | schmidts: its only keepalives |
| 10:20.15 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
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| 10:30.06 | v0lZy | hi guys |
| 10:30.16 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.65) |
| 10:30.23 | v0lZy | security question... I have a remote location which i need to register the phone from to my location |
| 10:30.56 | v0lZy | whats my best option here.... VPN? would GRE suffice? |
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| 10:39.14 | ruben231 | hi guys any idea how to make my voicemail be emailed toa specified email add, using asterisk 1.4 |
| 10:42.49 | v0lZy | dont know about 1.4, but other than that its pretty trivial |
| 10:43.08 | schmidts | danfromuk even strange but i think its quite normal cause of the realtime data, maybe you can take a look at the astdb on server2 maybe you still have some data of these peers in the astdb and with the realtime db asterisk knows where to reach these peers |
| 10:43.13 | v0lZy | at least in my asterisk version, its not biggy. |
| 10:48.37 | ruben231 | v0lZy: any guide somehow.. |
| 10:54.33 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 10:56.12 | schmidts | ruben231 i dont understand what you want, just sending your voicemail to an email address or something different? just take a look at the configs/voicemail.conf:samples file in the directory where your asterisk sources are, its described in there how to setup an external mail prog to send voicemails |
| 10:58.37 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 11:00.37 | gavimobile | im having trouble over here with clid. http://pastebin.com/kMfM7fkz the polycom is listed in cdr as "Polycom" <111> and the other peer comes out as "110" <0000FFFF0000>. it should read as "Office Portable" <110> instead of "110" <0000FFFF0000> in the cdr. I don't believe this is an asterisk problem, but maybe asterisk can overide the ata/pap device |
| 11:01.18 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:03.04 | ruben231 | schmidts: juts basically sending my voicemail file into an email, thats all |
| 11:04.58 | *** join/#asterisk vfabi (~fabi@champagnely-update.volia.net) |
| 11:05.22 | schmidts | ruben231 then its easy. open your voicemail.conf, set attach=yes, sendvoicemail=yes and add a mail address to your voicemailbox itself, that should be everything if you have a mailsender allready configured on your system |
| 11:09.22 | danfromuk | schmidts: in that case, if i remove the ipaddr, fullcontact and port columns from the realtime db, the servers should start behaving? |
| 11:09.44 | danfromuk | if thats the case** |
| 11:10.44 | schmidts | danfromuk first do a database show key SIP/Registry on server2 if you have any entries in there, if yes you could remove it from there and it should be fine |
| 11:12.07 | danfromuk | Some are listed in astdb but not all of them. |
| 11:14.55 | danfromuk | There is a column called regserver which doesn't appear to be being used, but i would have assumed is asterisk's way of tracking which server a peer is currently registered to. |
| 11:16.37 | danfromuk | Actually, its an old bug thats marked as resolved. http://lists.digium.com/pipermail/asterisk-bugs/2010-September/085967.html |
| 11:16.47 | *** join/#asterisk Elleni (3ec00582@gateway/web/freenode/ip.62.192.5.130) |
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| 11:27.01 | danfromuk | Does regserver actually prevent the other servers from trying to use the data? |
| 11:31.49 | ruben231 | schmidts: the problem i dont know how to configure mail sende on the system |
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| 11:34.48 | schmidts | ruben231 what os do you use? |
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| 12:05.41 | ghost75 | wow poe needs even 56v psu |
| 12:18.50 | wdoekes | danfromuk: "Ignore this, please close it i'm an idiot. asterisk.conf [options] was commented out so the system name setting was not getting applied. As soon as i uncommented it it works." |
| 12:20.00 | Chainsaw | ghost75: 48V at the destination really, but yes, there is some loss in the cabling. |
| 12:20.29 | ghost75 | do you think these phones will run also with much lower voltage? |
| 12:22.18 | Chainsaw | ghost75: Qualify much. |
| 12:22.53 | Chainsaw | ghost75: Anything below 48 is unlikely to work. If you happen to have some oddball Cisco 48V PSU lying around though... yes, that might well work. |
| 12:24.49 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
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| 12:27.52 | danfromuk | wdoekes: i saw that and checked it but i have it uncommented and regserver still not being updated. Also, when it works, it doesnt appear to prevent asterisk from sending out keepalive packets. it appears to be used in dialplans to send calls to the correct server. |
| 12:28.08 | danfromuk | However, I'm looking into dundi as an alternative. |
| 12:28.16 | ghost75 | 24V, plugged directly into phone? |
| 12:29.46 | Chainsaw | ghost75: That's half of what you need. I told you that you need 48. |
| 12:30.15 | ghost75 | but nobody tried lower right? |
| 12:30.52 | Chainsaw | ghost75: It is unlikely but not impossible. |
| 12:31.33 | Chainsaw | ghost75: I would like to qualify though, that the unlikely is written in capital letters. |
| 12:32.51 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:35.24 | ghost75 | too low voltage shouldnt damage it, so maybe i try it |
| 12:47.31 | ruben231 | schmidts: ubuntu server 12.04 LTS, sorry late reply |
| 12:49.26 | *** join/#asterisk felipealmeida (~user@querubim.tecgraf.puc-rio.br) |
| 12:49.29 | schmidts | ruben231 first install sendmail with "aptitute install sendmail" and normally you should get a config menu set it up,normally its very easy to do |
| 12:49.35 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 12:50.02 | ruben231 | schmidts:ok i wil try |
| 12:50.45 | schmidts | k |
| 12:52.06 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 12:53.21 | weinerk | Hi please help with DUOBLING DTMF sygnals. |
| 12:53.21 | weinerk | I added this: dtmfmode=auto |
| 12:53.21 | weinerk | Now all user input started to double up like this: 1122334455 |
| 12:54.11 | gavimobile | im having trouble over here with clid. my polycom is listed in cdr as "Polycom" <111>which is good and my other peer comes out as "110" <0000FFFF0000>. it should read as "Office Portable" <110> instead of "110" <0000FFFF0000> in the cdr. I don't believe this is an asterisk problem, but maybe asterisk can overide the ata/pap device |
| 12:56.02 | schmidts | weinerk set dtmfmode to inband or rfc2633 this should help |
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| 13:12.51 | gavimobile | why is the output of ${CALLERID(name)} not coming out as specified in sip.conf |
| 13:13.37 | kaldemar | gavimobile: show what you're doing. |
| 13:15.34 | gavimobile | kaldemar: when calling from one of my peers, the wrong data is being saved into my database. here is peer info http://pastebin.com/HQPQBRNP. |
| 13:15.49 | gavimobile | this is what I see in my database "110" <0000FFFF0000> |
| 13:16.08 | gavimobile | kaldemar: what else can I show you please? |
| 13:17.15 | kaldemar | CLI output that shows an incoming call match that peer and show caller id output that differs from the value in sip.conf. |
| 13:17.47 | kaldemar | your database might have something else. |
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| 13:28.11 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:29.17 | gavimobile | kaldemar: I called from another peer locally this is what I get http://pastebin.com/F9c8G7wA |
| 13:29.43 | gavimobile | I don't think that's enough detail for you |
| 13:30.24 | gavimobile | my other peers that are NOT registered to my pbx with a pap/ata device work fine. seems my ata/pap device is taking control of the clid for this peer |
| 13:31.04 | kaldemar | gavimobile: your paste is completely useless. |
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| 13:31.15 | gavimobile | the pap/ata device has 2 fxs ports and both log wrong clid info. I've checked all my other peers from softphones to hardphone and they are all fine |
| 13:32.24 | kaldemar | 1. make an extension that outputs caller id. 2. enable sip debug. 3. make a call to the extension that outputs caller id. 4. pastebin result. |
| 13:33.04 | gavimobile | kaldemar: that sounds clearer than the first request |
| 13:33.08 | gavimobile | just a minute |
| 13:36.53 | gavimobile | kaldemar: hope nothing important got cut |
| 13:36.54 | gavimobile | http://pastebin.com/F0UHKhKQ |
| 13:38.21 | gavimobile | this is my dialplan for this btw http://pastebin.com/PKq4ArLj |
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| 13:40.15 | kaldemar | gavimobile: asterisk -rx "sip show peer 0000FFFF0000" | grep Callerid |
| 13:40.36 | gavimobile | Callerid : "Office Portable" <110> |
| 13:40.41 | gavimobile | I already gave you that info |
| 13:40.59 | kaldemar | no you did not. you showed a snippet of sip.conf. |
| 13:41.44 | fukuda76140 | hi, i'm a problem to install T38modem with hylafax and asterisk. I'm installed T38modem but on faxstat -s ==> Modem ttyT38-1 (): Waiting for modem to come ready |
| 13:42.10 | gavimobile | kaldemar: ok, well "Office Portable" <110> was set as the value for callerid in that sip snippet |
| 13:42.38 | gavimobile | kaldemar: could my pap device be taking over? it all started I believe once I upgraded the firmware for my pap device |
| 13:42.46 | gavimobile | if I am NOT mistaken |
| 13:42.57 | kaldemar | the rpid is probably to blame. and the rest of your sip.conf. |
| 13:43.08 | gavimobile | rpid? |
| 13:43.26 | gavimobile | what's that? and what's the fix? |
| 13:44.40 | fukuda76140 | In /var/spool/syslog => Jan 21 14:42:58 hylafax-projet1 FaxGetty[1680]: /dev/ttyT38-1: Can not open modem (No such file or directory) |
| 13:44.47 | kaldemar | trustrpid=no under [0000FFFF0000] |
| 13:46.01 | gavimobile | kaldemar: wow!!! bingo |
| 13:46.14 | gavimobile | thanks man! |
| 13:46.59 | gavimobile | This defines whether or not Remote-Party-ID is trusted. |
| 13:47.40 | gavimobile | kaldemar: what's recommended as default? |
| 13:47.48 | gavimobile | on or off for all other peers |
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| 13:55.38 | fukuda76140 | help please |
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| 14:00.17 | kaldemar | gavimobile: what's recommended depends on where it is used. for phones there is no need to use rpid because the values are defined in asterisk configs. |
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| 14:06.09 | pabelanger | ~itsp-us |
| 14:08.36 | fukuda76140 | it not find /dev/ttyT38-1 but ttyT38-1 redirect to /dev/pts/2 |
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| 15:00.38 | fukuda76140 | hi |
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| 15:43.15 | jeffspeff | I'm having some issues with faxing. It seems that some come through just fine and others don't. Here are the fax logs and dialplan context and macro i'm using. http://pastebin.com/FrSzi6BB http://pastebin.com/DZhyfW74 |
| 15:49.32 | leifmadsen | has anyone used sipp to perform some registration testing, and been able to get it to respond to OPTIONs? |
| 15:49.46 | *** join/#asterisk tarcert (c1a9b80e@gateway/web/freenode/ip.193.169.184.14) |
| 15:50.13 | leifmadsen | I can't get SIPp to respond to OPTIONs using either the -aa option (which isn't for OPTIONs anyways according to docs) or the -oocsf file to respond to out of call messages (because it appears to not match them as a OOC scenario) |
| 15:52.01 | tarcert | Hi All, i have 7911G with sipfirmware, if I rest that phone I'll lose the SIPFirmware? |
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| 15:56.33 | tarcert | anyone can help plz |
| 15:58.58 | igcewieling | jeffspeff: Best of luck. In my experience FoVoIP and FoIP both such. |
| 15:59.26 | jeffspeff | igcewieling, thanks |
| 15:59.34 | igcewieling | s/such/suck |
| 15:59.54 | igcewieling | jeffspeff: we get anywhere from a 50% to 80% completion rate |
| 16:00.10 | jeffspeff | wow |
| 16:00.19 | jeffspeff | even when using t.38? |
| 16:00.33 | igcewieling | only a moron would not use T.38 for fax. |
| 16:00.42 | igcewieling | jeffspeff: T.38 makes it closer to the 80% |
| 16:01.07 | jeffspeff | so far, our fax machine that's setup for using SIP has a much better success rate than the t.38 |
| 16:01.13 | jeffspeff | and i mean much much better |
| 16:02.03 | jeffspeff | fax machines should just be destroyed. there's no need for them anymore; they've been replaced with email |
| 16:02.03 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 16:03.04 | igcewieling | jeffspeff: Fax is needed because paying customers want it. |
| 16:03.31 | jeffspeff | unfortunately, they only want it because they don't know any better. |
| 16:03.33 | igcewieling | We often install POTS lines for fax. |
| 16:03.50 | igcewieling | I don't care why they want it. If we don't offer it they will go with a different company. |
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| 16:36.51 | pabelanger | So, if I wanted to change a queue members settings via the AMI, I guess I need to log them out then back in with the new settings? |
| 16:37.05 | pabelanger | for example, changing their penalty |
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| 16:49.02 | mjordan | pabelanger: does QueuePenalty not immediately affect the member? |
| 16:54.58 | pabelanger | mjordan, not sure, guess that is what I am asking. About to test it. But, also wanted to know if we exposed anything over the AMI to modify a member, aside from logout / login |
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| 17:00.44 | damg | does anyone know what's the right behaviour for rfc 5589 (sip call transfer) if the refer-to header contains uri parameters or headers? should the transfered UA invite with or without the params or headers? e.g. what should the user agent do about a Refer-To: sip:foobar@192.168.0.1;fancykey=1337 ? |
| 17:01.57 | jacekowski | i've been recently getting quite a lot of call failures with cause code 111, i'm on BT BRI (ISDN2e) lines |
| 17:02.06 | jacekowski | does anybody have any idea what is it? |
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| 17:29.39 | Katty | hello my asterisk does not work at all how to fix??? answer plz is urgent thx. |
| 17:31.37 | chuckf | Katty: press the shift key then the 8 key |
| 17:31.55 | pabelanger | Katty, I think you want to try freeswitch |
| 17:32.37 | igcewieling | *** Katty is now known as KattyTrolling |
| 17:32.43 | Katty | igcewieling: sshhhh |
| 17:32.46 | Katty | stuffs igcewieling in the closet |
| 17:33.14 | Katty | pabelanger: yes. i do. |
| 17:33.41 | Katty | pabelanger: but that is complicated since we don't sell that sort of thing here. |
| 17:34.06 | igcewieling | jacekowski: 111 is "unspecificd protocol error" |
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| 17:46.37 | Katty | man i am having a rough morning of converting oxygen into carbon dioxide |
| 17:49.07 | slav3_kitten | assists with this by starting a nice fire in the fire place |
| 17:49.30 | Katty | that'll work so much faster than my lungs |
| 17:49.30 | ChannelZ | That will make the Green people happy |
| 17:49.49 | slav3_kitten | did you get all sick Katty ? |
| 17:50.05 | Katty | well...a few weeks ago i caught something. |
| 17:50.13 | slav3_kitten | aww :( |
| 17:50.13 | Katty | not sure /what/ exactly, but benadryl kicked it to the curb. |
| 17:50.21 | Katty | i've been fine since. |
| 17:50.55 | Katty | still sinus drainage tho. and the occasional cough. |
| 17:50.59 | slav3_kitten | might be pneumonia, it's been going around recently |
| 17:51.03 | Katty | but nothin like it was. |
| 17:51.10 | slav3_kitten | oh i see |
| 17:51.12 | Katty | oh no, it's not pneumonia |
| 17:51.20 | Katty | my guess is some sort of allergic reaction to something |
| 17:51.24 | slav3_kitten | when you said lungs i was thinking hard time breathing an shortness of breath |
| 17:51.37 | slav3_kitten | i got it pretty damn terrible around christmas |
| 17:51.40 | Katty | oh no. i was just saying i wasn't doing anything this morning except breathing :P |
| 17:52.00 | slav3_kitten | well i got some ideas.... |
| 17:52.01 | Katty | but pneumonia would also be an acceptable answer. |
| 17:52.10 | Katty | it fits :P |
| 17:52.26 | slav3_kitten | you could drive over here an do some electronics repair for me :) |
| 17:53.15 | Katty | that sounds like work. |
| 17:53.25 | Katty | i think i'd rather convert oxygen to carbon dioxide |
| 17:54.03 | slav3_kitten | you could do some programming for me? |
| 17:54.14 | Katty | could? yes. |
| 17:54.16 | Katty | will? probably not. |
| 17:54.37 | Katty | :P |
| 17:55.12 | slav3_kitten | what if i cooked you a steak for your troubles? |
| 17:55.41 | Katty | you must really be desperate hehe |
| 17:56.00 | Katty | tho...honestly.. |
| 17:56.04 | Katty | i don't have a good steak recipe yet. |
| 17:57.00 | Qwell | recipe? |
| 17:57.14 | Qwell | remove steak from fridge. put steak on fire. remove steak from fire. eat. |
| 17:57.15 | Katty | welll....guidelines |
| 17:57.22 | Katty | >.< |
| 17:57.31 | Katty | i don't know how to cook steak. at all |
| 17:57.44 | Katty | and i'm not really entirely sure on how to operate the bbq grill either |
| 17:58.02 | *** join/#asterisk neohidra (538e1404@gateway/web/freenode/ip.83.142.20.4) |
| 17:58.11 | Qwell | Katty: feel the meaty part of your thumb, on your palm. The softest part == rare. Move up slightly, med. Move up more, well done. |
| 17:58.34 | Katty | interesting |
| 17:58.38 | Katty | like up towards finger |
| 17:58.41 | Katty | or up towards palm |
| 17:58.49 | Qwell | up the thumb |
| 17:59.04 | Qwell | by the joint |
| 17:59.18 | Katty | it's just skwish then bone :< |
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| 17:59.43 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 17:59.47 | Katty | so between the two joins would be medium, or well done? |
| 18:00.01 | Qwell | go in more |
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| 18:00.24 | Katty | goes in search of photo |
| 18:00.26 | Qwell | Katty: You'll need to see a picture |
| 18:00.28 | Qwell | yar, that |
| 18:00.46 | Qwell | http://www.ourbestbites.com/2008/05/great-tip-testing-steak-doneness/ |
| 18:00.51 | Qwell | or that |
| 18:01.34 | Katty | woah |
| 18:01.36 | dr0ck | you gotta know your grill, is all |
| 18:02.09 | Katty | i feel weird continuiously poking myself |
| 18:02.59 | slav3_kitten | kaldemar, to cook steak you take lump charcoal |
| 18:03.03 | slav3_kitten | you err Katty |
| 18:03.18 | Katty | i probably won't cook it on a grill |
| 18:03.58 | slav3_kitten | you then light it in a chimney, wait till coals are white hot and spread them out and flatten them down with a cast iron pan |
| 18:04.37 | Katty | i'll probably cook in a cast iron pan |
| 18:05.00 | slav3_kitten | but but... |
| 18:05.01 | Katty | that's a neat idea tho...to flatten it with cast iron |
| 18:05.45 | slav3_kitten | cooking a nice dirty steak is tasty |
| 18:05.56 | slav3_kitten | http://www.epicurious.com/recipes/food/views/Dirty-Steak-352992 |
| 18:06.10 | neohidra | i use asterisk with freepbx and managed to set it up at least for the LAN. As a client i use csipsimple and i am able to connect to the server via the mobile network. I have set a port forwarding rule (to the asterisk server) on the router and can dial/receive calls but i cannot hear anything - only the other side can hear me. Is there something i am missing regarding the port forwarding or it is most probably a problem with some o |
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| 18:20.25 | Katty | http://tinyurl.com/b5k3lt4 <- squirrlie nomnomnom |
| 18:21.05 | ChannelZ | neohidra: What ports did you forward? |
| 18:21.20 | chuckf | How i cook steaks... Alton Brown's method. http://www.youtube.com/watch?v=2wg0UDuU2-o |
| 18:21.34 | chuckf | That's for indoor cooking, not grilling |
| 18:21.53 | Katty | chuckf: ty! |
| 18:22.26 | chuckf | Katty: you're welcome. |
| 18:22.32 | neohidra | channelz: UDP 4590 on the router to 5060 on the asterisk machine |
| 18:23.08 | ChannelZ | That will get you SIP (I guess, if you point your phone at port 4590...) but you need a range of ports forwarded also for RTP |
| 18:23.33 | ChannelZ | For FreePBX I have no idea how you change it, but in vanilla asterisk it's whatever range you use in rtp.conf |
| 18:24.16 | ChannelZ | which by Asterisk default is some huge range like 10000-20000 |
| 18:24.59 | ChannelZ | oof actually it's worse than that.. "Defaults are rtpstart=5000 and rtpend=31000" |
| 18:25.07 | neohidra | channelz: yes i am pointing csipsimple to my asteriskserver:4590 but missed the rtp port. So there is not on esigle port but a range to forward? |
| 18:25.12 | igcewieling | neohidra: try directmedia=no or canreinvite=no (depending on your asterisk version) in the general section of sip.conf |
| 18:25.23 | igcewieling | ChannelZ: no, the default is 10,000 - 20,000 |
| 18:25.58 | *** join/#asterisk din3sh (~din3sh@196.20.246.153) |
| 18:26.07 | din3sh | Hello all |
| 18:26.13 | ChannelZ | igcewieling: well it is set for 10k-20k but I guess if you specify nothing it's the other? For FPBX no idea what the hell it's doing |
| 18:26.46 | igcewieling | default == no config file |
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| 18:27.59 | din3sh | Have deployed a couple of cisco 6945 SIP phones with Asterisk 1.18. I am getting "SIP 486 busy here" messages without any DND function actually active on the phones. |
| 18:28.11 | din3sh | any idea why ? |
| 18:28.51 | ChannelZ | neohidra: anyway yes, you need a range, as many as you would have simultaneous calls occuring on the system plus some padding |
| 18:29.33 | neohidra | I get ; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ; ; this file must be done via the web gui. There are alternative files to make ; ; custom modifications, details at: http://freepbx.org/configuration_files - i will dig into freepbx docs for now |
| 18:29.57 | neohidra | and may come cak if still have problems :) thank you all |
| 18:31.27 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 18:32.28 | din3sh | Have deployed a couple of cisco 6945 SIP phones with Asterisk 1.18. I am getting "SIP 486 busy here" messages without any DND function actually active on the phones. any idea why? |
| 18:33.45 | jmetro | did you watch the console and see why its happening? |
| 18:37.19 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 18:46.02 | din3sh | jmetro: only fires out a SIP 486 busy here |
| 18:46.56 | din3sh | not much info :s |
| 18:46.58 | jmetro | so with core set debug 400 and core set verbose 400 you dont see anything else. |
| 18:47.11 | *** part/#asterisk deo_ (~deo@112.198.90.185) |
| 18:47.24 | din3sh | or i can't interprete it maybe :s |
| 18:47.46 | jmetro | you should pastebin the log for us. |
| 18:50.10 | din3sh | http://pastebin.com/K0m1cARP |
| 18:50.14 | din3sh | here it is |
| 18:50.42 | din3sh | 1st call rings alright |
| 18:50.56 | din3sh | 2nd call which arrives gives a busy 486 message |
| 18:54.46 | Chainsaw | Hanging up zombie call. Be scared. |
| 18:55.11 | din3sh | i am scared |
| 18:56.45 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 18:59.32 | *** join/#asterisk tarcert (c1a9b80e@gateway/web/freenode/ip.193.169.184.14) |
| 19:00.36 | Katty | skurred |
| 19:01.51 | din3sh | :s |
| 19:02.31 | Katty | does a little dance, gets down tonight |
| 19:02.48 | Katty | Qwell: i had a russian cuisine recommendedation for the stl area :>>>>>>>> |
| 19:02.57 | Katty | Qwell: and the menu looks LEGIT! |
| 19:03.08 | ChannelZ | Turn off core debug, it's almost never helpful for what is probably configuration or device configuration issues |
| 19:05.04 | ChannelZ | there's so much going on in this paste |
| 19:07.23 | Katty | ChannelZ: amazing that the human brain can sort through so much input and make sense of it |
| 19:08.44 | Katty | ChannelZ: did you know that we process about 1.5MBs of data per second? |
| 19:11.11 | Katty | assuming you're awake for 15 hours a day...you brain sifts through nearly 80GB of data per day |
| 19:11.31 | ChannelZ | 90% of which it probably throws out as irrelevant |
| 19:11.34 | Katty | yes. |
| 19:11.38 | Katty | but it's still processed. |
| 19:11.48 | Katty | and it knows what to keep, and what to toss. |
| 19:11.50 | Katty | most of the time >.< |
| 19:11.58 | Katty | shame it doesn't catalog and tag it too |
| 19:12.55 | ChannelZ | well this log is only a few k and I can't get through it, it's a wreck :) |
| 19:14.02 | Katty | my whole day is a wreck. |
| 19:14.10 | Katty | but someone my brain still manages to sift through it....amazing! |
| 19:15.27 | jacekowski | igcewieling: i know that |
| 19:15.33 | jacekowski | igcewieling: i was wondering what that means on BT |
| 19:16.14 | ChannelZ | erhhmm... |
| 19:26.11 | navaismo | In asterisk 11.2 the profile-level-id in video mode is setted by default in the sdp or I need to patch the asterisk to include a profile-level-id?? |
| 19:36.26 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 19:51.28 | *** join/#asterisk feeshon (~gaston@ool-45787011.dyn.optonline.net) |
| 19:51.37 | feeshon | Hello all, I have a quick one I believe |
| 19:52.14 | feeshon | I need to made one change in my polycom config files (which is correct from what I read in the polycom manual) and my phones don't seem to pick up the change |
| 19:52.41 | feeshon | Is there something I should be doing for it to prase the new phone prov files? |
| 19:53.21 | feeshon | parse |
| 20:03.36 | *** join/#asterisk volga629 (~volga629@host7.pythian.com) |
| 20:07.25 | tarcert | Hi Gents, why cisco phone keep requesting ctlsep<mac>.tlv ? anyone can help me please ? |
| 20:07.29 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 20:12.32 | *** join/#asterisk gg608f (~Adium@client-7-157.visitor-network.oxuni.org.uk) |
| 20:14.49 | tarcert | Hi why cisco phone keep requesting ctlsep<mac>.tlv ? anyone can help me please ? |
| 20:16.34 | *** part/#asterisk volga629 (~volga629@host7.pythian.com) |
| 20:31.06 | *** join/#asterisk coreyf1513 (~coreyf151@108.250.153.45) |
| 20:39.46 | jacekowski | sounds like some kind of provisioning |
| 20:44.18 | [TK]D-Fender | tarcert, Because that is one of the files used in provisioning it with the version of firmware you have on it. www.cisco.com <- go download the admin guides and set them up. |
| 21:02.00 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.65) |
| 21:03.03 | jeffspeff | [TK]D-Fender, can you assist me with faxing troubles? here's the conf files and dialplan http://pastebin.com/Vs6N009S here's the fax log of completed faxes and errored faxes http://pastebin.com/5igT2sbK the problem is that all faxes aren't being received. as you can see on the last few attempts, i have the baud rate set for 2400 but it still somehow negotiates a 4800 speed. the others that transf |
| 21:03.03 | jeffspeff | er at 2400 fail with timeout errors. |
| 21:04.11 | *** join/#asterisk dwayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 21:05.21 | [TK]D-Fender | jeffspeff, What is the call coming in over? What ver of *? |
| 21:06.00 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-jcqnfluwvdkvahfo) |
| 21:07.34 | jeffspeff | the faxes are coming over t.38 using asterisk version 11.0 with latest FFA module from digium. |
| 21:07.44 | *** join/#asterisk Quest (~sync@pool2-80-210.brain.net.pk) |
| 21:07.52 | Quest | how do i know i have t38 support? i have asterisk pbx runing . i want fax over ip |
| 21:08.14 | jeffspeff | Quest, do a fax show stats |
| 21:08.22 | jeffspeff | if you have license for t.38 it will show there |
| 21:08.27 | Quest | what command? |
| 21:09.00 | jeffspeff | fax show stats |
| 21:10.01 | Quest | we never purchased a licence |
| 21:10.20 | jeffspeff | you can get a free license for 1 concurrent fax from digium |
| 21:10.26 | Quest | and if we do. theres no need for any additional software? |
| 21:10.46 | Quest | digium.com? |
| 21:10.51 | jeffspeff | i'm doing this to recieve a fax and then i wrote a script to email it |
| 21:10.57 | jeffspeff | i believe so |
| 21:11.04 | jeffspeff | just do a google search for digium FFA |
| 21:11.15 | Quest | i need to send faxes ... |
| 21:11.26 | jeffspeff | [TK]D-Fender, did you see my response to you above? |
| 21:11.54 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
| 21:12.05 | [TK]D-Fender | jeffspeff, Show the complete call |
| 21:12.15 | Quest | jeffspeff, thanks |
| 21:12.39 | jeffspeff | that is complete. i have the DID go straight to the fax extension |
| 21:12.57 | jeffspeff | that's line for line what shows on console, just without all the extra crap from people making phone calls |
| 21:15.38 | [TK]D-Fender | jeffspeff, there is no SIP DEBUG in there |
| 21:22.32 | jeffspeff | [TK]D-Fender, do i need to enable sip debug and try again |
| 21:23.40 | *** join/#asterisk myyrdin (~myyrdin@gateway/tor-sasl/myyrdin) |
| 21:25.00 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 21:25.59 | *** join/#asterisk tarcert (c1a9b80e@gateway/web/freenode/ip.193.169.184.14) |
| 21:28.39 | *** join/#asterisk jsjc (~Adium@91.Red-83-60-132.dynamicIP.rima-tde.net) |
| 21:37.22 | *** join/#asterisk bchia (~Adium@nat/digium/x-msmklfhomehsvqgo) |
| 21:41.47 | *** join/#asterisk DoSJustin (~justin@vpn.bctconsulting.com) |
| 21:45.19 | navaismo | anyone here using webrtc with asterisk 11.2?? |
| 21:46.31 | *** join/#asterisk greenwolf (42570089@gateway/web/freenode/ip.66.87.0.137) |
| 21:49.54 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 22:08.55 | *** join/#asterisk SteveWilliams (~chatzilla@59.162.182.218) |
| 22:09.13 | SteveWilliams | Good Morning everyone! |
| 22:10.35 | *** join/#asterisk tamiel (~tamiel@c-67-169-76-114.hsd1.ca.comcast.net) |
| 22:11.12 | SteveWilliams | I am facing voice quality issues with my asterisk based dialer. How do I seek paid support from Digium? Their link for commercial support is broken http://www.digium.com/en/supportcenter/asterisk.php |
| 22:12.16 | SteveWilliams | Guys help! Please.... |
| 22:13.29 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 22:14.21 | *** join/#asterisk ruyo (~ruyo@a213-22-221-38.cpe.netcabo.pt) |
| 22:14.22 | WIMPy | I think, commercial support is only available for the commercial products. For commercial Asterisk support you can ask on the asterisk-biz mailing list. For support here you have to |
| 22:14.24 | WIMPy | ~ask |
| 22:14.24 | 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. |
| 22:14.25 | navaismo | evening! |
| 22:15.44 | SteveWilliams | Alrighty Sir.... Let me start off by explaining my setup first... |
| 22:16.24 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-pwutaxjgeryagpnu) |
| 22:16.53 | SteveWilliams | My Dialer uses 3.0 Ghz Xeon Proc, 2 GB RAM, 250 GB Hard Disk and has asterisk 1.6.2.16-68 installed in it |
| 22:17.13 | SteveWilliams | it is connected to a 2 MBPS leased line |
| 22:17.14 | WIMPy | That's pretty old. |
| 22:17.24 | din3sh | update your asterisk man |
| 22:17.36 | SteveWilliams | okay.... |
| 22:17.45 | navaismo | LOL i know people using asterisk 1.2 & 1.4 |
| 22:17.58 | din3sh | how many concurrent calls? |
| 22:18.01 | navaismo | for commercial & professional CC |
| 22:18.02 | WIMPy | The S&M scene never dies. |
| 22:18.19 | SteveWilliams | near about 24 - 30 concurrent calls |
| 22:18.53 | WIMPy | What CODEC? |
| 22:19.01 | SteveWilliams | g729.a |
| 22:19.10 | WIMPy | ok |
| 22:19.11 | SteveWilliams | did i spell it right |
| 22:19.13 | SteveWilliams | ?? |
| 22:19.35 | SteveWilliams | next priority codecs are ulaw & alaw |
| 22:19.38 | mjordan | SteveWilliams: clicking on your link went to the Digium support page. |
| 22:19.53 | *** join/#asterisk SuperNull (~super@24-148-106-195.ip.mhcable.com) |
| 22:20.06 | SteveWilliams | i dunno, it didnt in my case... lemme check again |
| 22:20.07 | WIMPy | 30 calls of alaw/ulaw won't fit in to 2mbit. |
| 22:20.26 | SteveWilliams | but should work with g729 as the first codec |
| 22:20.30 | SuperNull | am i retarded ? how does one list loaded modules in ast 11 |
| 22:20.32 | SteveWilliams | or am i wrong |
| 22:20.53 | WIMPy | SteveWilliams: correct |
| 22:21.19 | WIMPy | SuperNull: module show |
| 22:22.36 | SteveWilliams | okay... my setup..... my hardware for agents to take calls.... 2 port LinkSys PAP2 phone adapter(supports g729) + Panasonic Telephony Basesets |
| 22:23.45 | SteveWilliams | i have a sangoma ut 50 usb dongle installed as well with my asterisk server... |
| 22:24.09 | ChannelZ | I think 'module show' is sort of a lie though |
| 22:26.06 | SteveWilliams | also, i have a netgear firewall acting as gateway for the asterisk server |
| 22:27.36 | SteveWilliams | okay guys... contacting support..... bye |
| 22:28.11 | ruyo | Anyone knows why an ISDN doesn't make a busy tone in an incoming call when the 2 lines are busy? (asterisk 1.4.X mISDN 1.1.9.2, I know it's very old) |
| 22:28.52 | sawgood | usually busy are SIP 300 reinvite messages (at least for me) maybe you have a reinvite concern? |
| 22:29.19 | WIMPy | ruyo: Tone? Where? Incomming from there? |
| 22:31.42 | ruyo | WIMPy, For instance, imagine 2 phones are making or receiving calls, occupying both ISDN lines. An outside person then makes a call to the ISDN phone number and they don't get a busy tone. They instead get either those fast busy tones or a message from the operator. |
| 22:31.56 | ruyo | Sometimes people think the number is out of order. |
| 22:32.26 | WIMPy | ruyo: That's correct. |
| 22:32.36 | WIMPy | It's a congestion. |
| 22:34.48 | ruyo | I remember having this question and you answering that like 2 years ago. :P But that's not how those ISDN phones or those Siemens Gigaset for instance, behave |
| 22:35.37 | ruyo | And when someone calls the number they think it doesn't exist anymore |
| 22:35.48 | WIMPy | The difference is that Asterisk doesn't support Call Waiting. |
| 22:37.16 | ruyo | Yeah, but I also tried disabling CW on the operator and they say it's disabled. Is there any way to send the "right" tone? Maybe that must be handled on the operator side? |
| 22:38.12 | WIMPy | If CW is disabled from the operator side, the 3rd call won't reach you. So it's up to them then. |
| 22:38.58 | ruyo | Hmm... In that case they're not doing their job. I do get some sort of message from the D-Channel when I try to make the 3rd call. |
| 22:39.38 | WIMPy | I haven't head anyone disabling CW anyway. |
| 22:39.44 | ruyo | This is the message: http://pastebin.com/j2FJHKJs |
| 22:40.58 | WIMPy | That's charging information for an ongoing call. |
| 22:41.19 | ruyo | Hmm |
| 22:41.29 | ruyo | Let me retry... |
| 22:42.59 | ruyo | Indeed. I tried twice before but it must have been coincidence. |
| 22:43.18 | ruyo | I don't get any debug information when placing the 3rd. |
| 22:44.22 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 22:52.10 | ruyo | Are the call waiting enable/disable codes a worldwide standard? |
| 22:53.12 | WIMPy | Definitely not worldwide. And not usually applicable to ISDN anyway. |
| 22:53.55 | ruyo | Grr |
| 22:55.03 | ruyo | So usually CW is enabled on ISDN and folks using asterisk don't disable it? |
| 22:55.33 | WIMPy | No, you usually disable CW on your terminal, not at the operator. |
| 22:56.45 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-vtnlktfxtbkqbysv) |
| 22:56.59 | ruyo | Yeah, but the call never gets to the terminal because it hits asterisk first and the two channels are already busy |
| 22:58.12 | WIMPy | Yes |
| 22:58.50 | WIMPy | You could probably patch the cause code to "fake" a user busy. |
| 22:59.00 | WIMPy | If you have CW enabled, off course. |
| 22:59.36 | ruyo | I tried that once and it worked for a while, but then it stopped working. |
| 23:00.19 | WIMPy | When? |
| 23:00.20 | ruyo | Maybe the call waiting was disabled in the meanwhile... |
| 23:00.36 | ruyo | About a year ago. |
| 23:00.53 | WIMPy | That would be an explanation. |
| 23:01.15 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:01.48 | ruyo | I think it only worked on PTP too... |
| 23:02.19 | WIMPy | That doesn't matter. |
| 23:02.29 | ruyo | Here it is: http://lists.digium.com/pipermail/asterisk-users/2010-November/256013.html |
| 23:03.38 | ruyo | Apparently I got it to work when the CW was enabled. At least I got messages when receiving the 3rd call. |
| 23:04.04 | WIMPy | Without it there's definitely nothing you can do. |
| 23:04.15 | *** join/#asterisk iztech (~iztech@76-246-226-131.uvs.irvnca.sbcglobal.net) |
| 23:04.29 | ruyo | Yeah, that's a start. :-) |
| 23:04.47 | *** join/#asterisk dxrt (~dxrt@unaffiliated/dxrt) |
| 23:09.36 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:14.34 | *** join/#asterisk Penguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
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| 23:18.13 | *** join/#asterisk elico (~Thunderbi@bzq-79-180-187-53.red.bezeqint.net) |
| 23:28.09 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
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| 23:30.23 | *** join/#asterisk ccrnp (~ccr@208-90-186-2.warp2biz.com) |
| 23:34.43 | *** join/#asterisk iztech (~iztech@76-246-226-131.uvs.irvnca.sbcglobal.net) |
| 23:36.52 | iztech | hi guys i have installed asterisk 11 on ubuntu and i got it to connect to callcentric but i would like to use SipStation instead but I don't have freePBX. has anyone configured sipstation/phonebooth if so i would like to see the sip.conf |
| 23:38.08 | iztech | fpbx-1-xxxxxxxxxx 184.72.227.214 a 5060 UNREACHABLE |
| 23:38.08 | iztech | fpbx-2-xxxxxxxxxx 6 50.56.59.168 a 5060 UNREACHABLE |
| 23:38.24 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 23:38.24 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 23:38.32 | iztech | this is what i get with sip show peers |
| 23:38.40 | *** join/#asterisk Mon|A|rch (~SBean@72.29.180.35) |
| 23:38.51 | Mon|A|rch | hey, [TK]D-Fender |
| 23:39.13 | Mon|A|rch | any idea how i might get a call to wait for a voicemail to be hit? |
| 23:39.28 | Mon|A|rch | or, at least to detect whether someone's picked up, or their voicemail was reached |
| 23:39.41 | [TK]D-Fender | Mon|A|rch: "core show application amd" |
| 23:39.51 | Mon|A|rch | cool |
| 23:39.52 | iztech | this is all the guys at sipstation provide - http://www.freepbx.org/freepbx-trunks |
| 23:40.07 | Mon|A|rch | thanks [TK]D-Fender |
| 23:40.51 | [TK]D-Fender | iztech: I'd start by actually looking at your registration attempts and inbound call debug |
| 23:42.06 | iztech | [Jan 21 15:21:58] NOTICE[4448]: chan_sip.c:14983 sip_reg_timeout: -- Registration for 'account@trunk2.phonebooth.net' timed out, trying again (Attempt #11) |
| 23:43.02 | iztech | both the ip addresses are resolved so i know its not a dns issue |
| 23:43.16 | iztech | is that what you mean [TK]D-Fender |
| 23:43.39 | [TK]D-Fender | I mean look at the SIP DEBUG |
| 23:43.46 | [TK]D-Fender | and see what's actually in there |
| 23:45.03 | saint_ | Can anyone tell me with this configuration : http://pastebin.com/nTb2xc9N why most of the time my Digium D70 does not have audio when it calls Asterisk ? |
| 23:45.20 | saint_ | D70 <-- private network --> Internet <-- Asterisk on public IP --> |
| 23:46.38 | iztech | so sip debug on at cli? |
| 23:46.51 | [TK]D-Fender | iztech: "sip set debug on" |
| 23:46.53 | [TK]D-Fender | ~pb |
| 23:46.53 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 23:46.54 | [TK]D-Fender | ^^^ |
| 23:47.17 | *** part/#asterisk coreyf1513 (~coreyf151@108.250.153.45) |
| 23:47.25 | [TK]D-Fender | saint_: hostile router by phone, bad firewall on server. Incorrect server config, etc. |
| 23:51.23 | *** join/#asterisk leowt (~leowt@233.163.108.93.rev.vodafone.pt) |
| 23:51.28 | *** join/#asterisk Molo (~Molo@c-76-31-26-101.hsd1.tx.comcast.net) |
| 23:51.43 | iztech | http://pastebin.com/4kujP0JX |
| 23:51.58 | iztech | [TK]D-Fender: i think this is what you are looking for |
| 23:52.48 | [TK]D-Fender | iztech: Contact: <sip:s@10.0.1.33:5060> <- your * is NOT on a public IP, and it transmitting its PRIVATE IP to them |
| 23:53.05 | leowt | hi there, ive got asterisk with sip clients. Wen connecting to a client outside the NAT i am able to talk to the 1st client, and the other ones get sip connection but no sound. |
| 23:53.12 | iztech | they told me it would work without static IP |
| 23:53.28 | [TK]D-Fender | iztech: It isn't configured properly for it |
| 23:54.07 | leowt | do anyone has a clue? |
| 23:54.36 | *** join/#asterisk MissionCritical (~MissionCr@unaffiliated/missioncritical) |
| 23:54.39 | saint_ | [TK]D-Fender: the firewall on the asterisk has ports 5060 udp/tcp and 10000:20000 udp all open |
| 23:54.44 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 23:54.44 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 23:56.20 | iztech | [TK]D-Fender: r u saying that it will work without a static IP but my config is wrong? if so is this a simple fix? |
| 23:57.12 | [TK]D-Fender | iztech: * needs to know it's WAN IP and this is everyday common stuff. |
| 23:58.05 | [TK]D-Fender | iztech: in [general] nat=yes , directmedia=no, externaddr=get.a.dyndns.type.provider , localnet=NETWORK/MASK |
| 23:59.11 | iztech | thx |