| 00:00.02 | mmlj4 | hrm, lemme try |
| 00:00.15 | vlad2 | most likely that |
| 00:00.16 | vlad2 | sounds like that |
| 00:01.15 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-yuqlwugaoraslmpi) |
| 00:01.15 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:01.58 | vlad2 | mmlj4 does it work now |
| 00:02.18 | vlad2 | can anybody help me with my calls disconnecting from my asterisk to a cisco pastebin.com/7BWtTTX9 every call disconnects at 19 or 20 seconds |
| 00:02.35 | vlad2 | that is a sip trace of 1 call |
| 00:02.49 | *** join/#asterisk mihamina (~mihamina@41.188.46.121) |
| 00:03.23 | *** join/#asterisk fkurkowski (~fkurkowsk@177.18.100.36) |
| 00:07.37 | mmlj4 | disallow=all ; First disallow all codecs |
| 00:07.37 | mmlj4 | allow=ulaw ; Allow codecs in order of preference |
| 00:07.41 | mmlj4 | still the same error |
| 00:08.13 | [TK]D-Fender | vlad2: What has your IOS side said? |
| 00:08.25 | [TK]D-Fender | mmlj4: Show us the actual SIP comms in full |
| 00:08.26 | [TK]D-Fender | ~pb |
| 00:08.26 | 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. |
| 00:08.28 | [TK]D-Fender | ^^^ |
| 00:08.32 | [TK]D-Fender | "sip set debug on" |
| 00:08.59 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 00:15.59 | mmlj4 | http://pastebin.com/5QaSWbgv |
| 00:17.04 | [TK]D-Fender | mmlj4: Reliably Transmitting (no NAT) to 64.61.93.190:5060: Contact: <sip:s@192.168.2.59:5060> |
| 00:17.16 | [TK]D-Fender | mmlj4: Your server is not properly set up to work behind NAT |
| 00:17.37 | [TK]D-Fender | mmlj4: Go fix all of that right now |
| 00:18.36 | [TK]D-Fender | mmlj4: Next : <--- SIP read from UDP:192.168.2.1:5061 ---> INVITE sip:15042396550@centosvb SIP/2.0 Via: SIP/2.0/UDP 68.11.55.215:5061;rport;branch=z9hG4bK217773894 |
| 00:18.38 | slav3_kitten | nope still broken |
| 00:19.09 | [TK]D-Fender | mmlj4: Your linphone seems to be getting the IVA different than assuming it was LOCAL. |
| 00:20.55 | [TK]D-Fender | mmlj4: And more : Sttarting at line 408 : ransmitting (no NAT) to 64.61.93.190:5060: ACK sip:5042396550@jfk-primary.voicepulse.com SIP/2.0 |
| 00:21.06 | [TK]D-Fender | mmlj4: Where we see a shit-ton of codecs being offered. |
| 00:21.20 | [TK]D-Fender | mmlj4: -- Executing [15042396550@internal:1] Dial("SIP/linphone-00000002", "SIP/voicepulse-primary/5042396550") in new stack |
| 00:21.38 | [TK]D-Fender | mmlj4: Clearly [voicepulse-primary] was not fixed |
| 00:21.56 | mmlj4 | so my * behind NAT, acting as a SIP client... there's something I have to change on my end? I thought it was the other way around |
| 00:22.29 | [TK]D-Fender | a pile of settings. |
| 00:22.39 | [TK]D-Fender | read the sample configs |
| 00:22.42 | [TK]D-Fender | ~nat |
| 00:22.42 | infobot | i heard nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 00:22.48 | [TK]D-Fender | ^ |
| 00:23.07 | mmlj4 | I grepped sip.conf, but it didn't mention NAT |
| 00:23.25 | jmetro | seems like none of those settings mentioned above have the word NAT in them |
| 00:23.30 | [TK]D-Fender | The samples certainly do |
| 00:23.52 | vlad2 | can anybody help me with my calls disconnecting from my asterisk to a cisco pastebin.com/7BWtTTX9 every call disconnects at 19 or 20 seconds <<sorru for the repaste |
| 00:24.19 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/trunk/configs/sip.conf.sample?view=log <- I see *23* |
| 00:24.43 | [TK]D-Fender | [20:08][TK]D-Fender vlad2: What has your IOS side said? |
| 00:25.12 | mmlj4 | heh, I grepped the wrong sip.conf |
| 00:25.27 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/trunk/configs/sip.conf.sample?view=markup <- 52 in the actual file |
| 00:25.56 | [TK]D-Fender | infoboti heard nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 00:26.07 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 00:26.14 | [TK]D-Fender | and "directmedia=no" <- |
| 00:28.06 | navaismo | so i received the response on the ticket: "Do you have layer2_persistence enabled?" but as i wrote in the ticket Idont know about that option, anyone here know what is that? |
| 00:29.00 | *** join/#asterisk joesmoe (~joesmoe@c-76-112-110-95.hsd1.mi.comcast.net) |
| 00:30.07 | elguero | navaismo: look in your chan_dahdi.conf file |
| 00:33.01 | navaismo | thanks, never used that option |
| 00:33.30 | navaismo | never ever in 7years, but as always, there is a first time |
| 00:37.02 | navaismo | well i dont have that option so I guess Im using the libpri default which is leave_down |
| 00:43.13 | vlad2 | [TK]D-Fender the ios guy said it works for everyone else |
| 00:53.05 | Spengler1 | so just bought an intel nuc ; gonna throw centos on there and see how well asterisk runs |
| 00:53.25 | Spengler1 | got it for around $260 on newegg with 4gb memory and a 32gb msata drive |
| 00:53.45 | Spengler1 | was reading that it uses ~5-6 watts at idle |
| 01:04.53 | vlad2 | can anybody help me with my calls disconnecting from my asterisk 11.3.0 to a cisco ios siptraceurl: pastebin.com/7BWtTTX9 every call disconnects at 19 or 20 seconds |
| 01:06.15 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 01:08.16 | *** part/#asterisk pgrace (pgrace@vsix.me) |
| 01:10.44 | slav3_kitten | jesus god, anyone here good with cisco 2621 routers, asterisk, and sip |
| 01:11.12 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 01:12.26 | *** join/#asterisk sktyau (~sktenau@monitor.ic.ipera.net.au) |
| 01:13.31 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.smb.curriegrad2004.ca) |
| 01:14.00 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.smb.curriegrad2004.ca) |
| 01:14.49 | sktyau | Morning all. Quick Q that someone may be able to help with. We have several upstream peers that send us calls on 5061, 5062. We send calls to them on 5060. We haven't previously worried about matching a peer on inbound, but I now need to. Obviously I can't specify a peer to have multiple ports - I'm assuming the only way around this is going to be to crate 3x peer definitions for each of the |
| 01:14.49 | sktyau | upstream servers, unless someone has a better suggestion |
| 01:15.32 | sktyau | Noting that I'm using the term peer losely - not in the Asterisk sense of the word. |
| 01:16.35 | sktyau | Eh. Brain fart. Insecure will do what I wan't :-) |
| 01:17.58 | *** join/#asterisk mjordan (~mjordan@adsl-98-81-59-18.hsv.bellsouth.net) |
| 01:17.58 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:20.52 | navaismo | ok... change to keep_up the layer2 setting avoid the unresponsive cli, but if you want to call again the cli doesnt response, enable the pri debug show some info like PRI Span: 1 TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment) |
| 01:21.44 | navaismo | the pri show span 1 never get all the info, so I'm start to thinking an issue with dahdi or libpri or the system conf |
| 01:24.53 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 01:24.53 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 01:29.40 | *** join/#asterisk deo_ (~deo@203.177.214.75) |
| 01:37.28 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 01:37.40 | *** join/#asterisk Bradada (~Bradada@114-32-7-232.HINET-IP.hinet.net) |
| 01:41.50 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-rtsypifvfgqptbsz) |
| 01:44.43 | *** join/#asterisk Bradada (~Bradada@114-32-7-232.HINET-IP.hinet.net) |
| 01:54.45 | navaismo | I got this http://pastebin.com/rXdcmXpg on pri set debug |
| 01:55.08 | navaismo | right now used all configuration availbale in system.conf and chan_dahdi.conf |
| 01:55.19 | navaismo | maybe a bad card?? |
| 01:57.46 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 02:06.28 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 02:07.43 | navaismo | enough for today |
| 02:35.19 | *** join/#asterisk Spengler1 (~Spengler1@pool-98-117-215-165.bltmmd.fios.verizon.net) |
| 02:49.11 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
| 02:49.58 | hebber | Grandstream ATA disconnect detection is really hard to figure out. |
| 03:05.33 | *** join/#asterisk andy09usa (~Andrey@unaffiliated/andy09usa) |
| 03:10.42 | *** join/#asterisk deo__ (~deo@222.127.13.226) |
| 03:14.02 | slav3_kitten | so is there a way i can limit the number of rtp ports asterisk needs? |
| 03:14.09 | slav3_kitten | on sip |
| 03:14.12 | slav3_kitten | outbound |
| 03:14.18 | slav3_kitten | well from my itsp |
| 03:14.22 | slav3_kitten | god i'm failing |
| 03:24.31 | ectospasm | RTP!=SIP |
| 03:24.38 | ectospasm | slav3_kitten: see rtp.conf |
| 03:26.17 | slav3_kitten | ectospasm, sorry. it's been a shit day and i'm half asleep trying to fix the server |
| 03:26.56 | slav3_kitten | i'm 95% sure it's nothing wrong with my side, since my static IP changed and then everything went to hell. even after changing the static IP on everything in my rack |
| 03:27.19 | ectospasm | behind a nat? Did you change the externip? |
| 03:27.24 | ectospasm | in sip.conf? |
| 03:27.29 | slav3_kitten | yep changed all that |
| 03:27.51 | slav3_kitten | updated my dns entries pointing to my now different ip |
| 03:29.25 | ectospasm | do you have a static IP address from your ISP? Shouldn't you be taking them to task for changing it on you? |
| 03:31.08 | slav3_kitten | yea i would. except the tech support guy has been on vacation for the last week |
| 03:31.08 | slav3_kitten | he's going to be coming back tomorrow |
| 03:31.26 | ectospasm | they don't have an alternate? Get a better ISP |
| 03:31.38 | slav3_kitten | there is one isp for rural "high speed" |
| 03:32.14 | slav3_kitten | we bought the house after comcast assured us they could get us service. even sent out a site survey guy who gave thumbs up. |
| 03:33.02 | slav3_kitten | once we closed on the house and got comcast out they were all "ohhh you're more then 300 feet from the main road, can't get you cable" |
| 03:33.02 | slav3_kitten | so it's this shit wisp or dialup |
| 03:33.35 | ectospasm | jikes |
| 03:33.52 | ectospasm | you should have gotten Comcast to sign something they could provide service |
| 03:34.12 | ectospasm | but Comcast are bitches, anyhow. |
| 03:34.22 | slav3_kitten | yep, but hindsight is 20/20 |
| 03:34.33 | jmetro | why am i still working |
| 03:34.40 | jmetro | its 10:30 wtf |
| 03:34.55 | slav3_kitten | jmetro, for the bitches? |
| 03:35.13 | jmetro | true... |
| 03:42.13 | slav3_kitten | crap i can't remember |
| 03:42.58 | slav3_kitten | after editing rtp, what do i need to issue for it to take effect |
| 03:56.31 | slav3_kitten | core reload... how the crap did i forget that |
| 04:01.10 | mmlj4 | better than core dump |
| 04:01.12 | ectospasm | yeah, core reload should do it, or an Asterisk restart |
| 04:02.00 | mmlj4 | get a spool of cable and run your own |
| 04:02.16 | mmlj4 | then get the installer to just terminate it |
| 04:05.01 | slav3_kitten | ok well i've spent 9 hours trying to get it to work again. i'm 100% positive it's not my end that's fucked |
| 04:06.32 | mmlj4 | which provider, slav3_kitten? |
| 04:07.08 | slav3_kitten | mmlj4, my itsp i'm sure is fine as well |
| 04:07.24 | slav3_kitten | it's my ISP, a rural WISP that's on it's second name which fucked me |
| 04:07.36 | slav3_kitten | had everything working, and working well |
| 04:07.51 | slav3_kitten | then my static IP which i pay 10 a month to get changed |
| 04:08.02 | slav3_kitten | and it went to hell |
| 04:08.04 | mmlj4 | I'm tempted to try setting up a rural isp, about 2 hours from here |
| 04:08.21 | mmlj4 | I could live an hour from it, I guess that's sufficient |
| 04:09.05 | slav3_kitten | if you do, prepare to have people like me hate you |
| 04:09.08 | mmlj4 | them folks got dialup or satellite, that's it |
| 04:09.10 | slav3_kitten | hate you so much |
| 04:09.55 | mmlj4 | oh, comcast? |
| 04:10.05 | mmlj4 | or no, you can't get them, right? |
| 04:10.43 | slav3_kitten | 60 a month for a questionable quality 3 down 1.5 up link that's got questionable up time, costs 10 for a static IP, have no tech support, everyone goes home at 5, the main tech takes week long vacations... |
| 04:10.44 | slav3_kitten | nope, it's this wisp, sat, or 56k |
| 04:13.37 | mmlj4 | well, tech support should be until like 8 or so |
| 04:13.49 | mmlj4 | because real people have jobs, etc. |
| 04:14.16 | mmlj4 | how does the signal get to your house? wires? wireless? squirrel? |
| 04:14.44 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.161.3) |
| 04:15.37 | *** join/#asterisk deo_ (~deo@222.127.13.226) |
| 04:15.41 | slav3_kitten | i had no connectivity for 23 hours because crap went down literally at 5:01 and they didn't get my message until 3pm next day. despite me leaving 20 messages from 8am to 3pm |
| 04:16.18 | slav3_kitten | it's wireless. 2.1 miles to the water tower they have their gear with no backup power what so ever |
| 04:17.24 | jmetro | what the hell 60 a month for 3down 1.5 up? |
| 04:17.39 | jmetro | that gets you like 10down/1up here |
| 04:17.54 | slav3_kitten | yep |
| 04:17.58 | jmetro | Find out how many 9's youre allotted and ask for credit |
| 04:19.26 | slav3_kitten | and the link went to shit so i can' even get iax2 (only working protocol at the moment) to have an ineligible call |
| 04:24.23 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 04:24.53 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 04:26.20 | mmlj4 | what kind of wireless? I mean 802.11... can you see the device when you scan for networks with your laptop or whatever? or is it wimax or some other? |
| 04:26.42 | mmlj4 | jmetro: he's out in the woods, and takes what he can get |
| 04:26.50 | mmlj4 | jmetro: 1 9 |
| 04:27.33 | slav3_kitten | 802.11 2.4ghz |
| 04:27.51 | mmlj4 | hrm |
| 04:28.03 | slav3_kitten | via ubiquiti gear |
| 04:28.10 | mmlj4 | well, those are good |
| 04:28.33 | slav3_kitten | they are |
| 04:29.50 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 04:33.45 | *** join/#asterisk deo__ (~deo@222.127.13.226) |
| 04:35.50 | joako | I have a user that´s complaining that at random times they receive e.g. 6 calls from ¨101¨ but I´ve checked the server logs and there do not appear to be any such calls going through |
| 04:39.39 | slav3_kitten | odd |
| 04:45.26 | joako | I know, the only thing I can really conclude is that the user has a virus in their network |
| 04:47.31 | slav3_kitten | possibly |
| 05:18.20 | *** join/#asterisk deo_ (~deo@222.127.13.226) |
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| 06:35.38 | *** join/#asterisk bulkorok (~chatzilla@85.183.36.36) |
| 06:40.35 | bulkorok | hi |
| 06:50.22 | *** join/#asterisk jkroon (~jkroon@dsl-244-22-34.telkomadsl.co.za) |
| 06:50.58 | jkroon | hi guys, when issueing Hangup() - is 0 a legal parameter value or not? it seems to complain: [Apr 10 08:45:35] WARNING[19872][C-0000aa0d] pbx.c: Invalid cause given to Hangup(): "0" |
| 06:51.45 | wdoekes | 0 is, "0" isn't |
| 06:52.06 | wdoekes | I think |
| 06:53.03 | wdoekes | mm.. I'm wrong |
| 06:53.27 | wdoekes | the quotes are in the message |
| 06:53.34 | wdoekes | so 0 isn't legal |
| 06:53.41 | wdoekes | see causes.h for valid causes |
| 06:55.23 | wdoekes | note that you can supply a named cause.. Hangup(UNALLOCATED) == Hangup(1) |
| 06:55.57 | jkroon | wdoekes, actually, no, you can ONLY supply a name |
| 06:56.47 | jkroon | the code doesn't allow for anything else. |
| 06:57.04 | jkroon | and the quotes are added by asterisk when outputting. |
| 06:58.20 | wdoekes | cause = strtol((const char *) data, &endptr, 10); |
| 06:58.35 | wdoekes | if (cause != 0 ... chan->hangupcause = cause; |
| 06:58.59 | bulkorok | wishes he could really understand what that means... |
| 06:59.00 | jkroon | wdoekes, what version of ast? because 11.3.0 does this: if (!ast_strlen_zero(data)) { cause = ast_str2cause(data); |
| 06:59.15 | wdoekes | 1.8 |
| 06:59.36 | jkroon | and ast_str2cause doesn't allow for numeric cause numbers, only the names... |
| 06:59.40 | wdoekes | 11 looks different, but does the same |
| 06:59.41 | wdoekes | if (sscanf(data, "%30d", &cause) |
| 06:59.50 | *** join/#asterisk din3sh (2988fd82@gateway/web/freenode/ip.41.136.253.130) |
| 07:00.11 | jkroon | oh yea, you're right. |
| 07:00.28 | jkroon | however, check the && clause directly after, 0 is invalid. |
| 07:00.35 | jkroon | must be 1 or greater, so going to use 16 :) |
| 07:00.42 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 07:01.06 | wdoekes | that's what I said.. use one of the codes from causes.h, or better yet a name |
| 07:01.44 | wdoekes | you can use no arg at all to get NORMAL_CLEARING (16) |
| 07:01.47 | bulkorok | I remember using ISDN-cause codes from ringofsaturn.com was working every time... |
| 07:02.00 | jkroon | Hangup(${IF($["${HANGUPCAUSE}"!=""]?${HANGUPCAUSE}:20)}) <-- ok, so that's not helping. |
| 07:02.21 | jkroon | it means that HANGUPCAUSE is 0, which probably means no dial was attempted. |
| 07:02.26 | jkroon | looks further, thanks. |
| 07:02.42 | wdoekes | indeed, if qualify says that the peer is down, you get 0, iirc |
| 07:03.07 | wdoekes | you'd need to tweak that IF |
| 07:03.52 | *** join/#asterisk MariusKarthaus (~quassel@541F97BB.cm-5-8c.dynamic.ziggo.nl) |
| 07:04.51 | jkroon | probably, but more importantly i first need to figure out *why* it drops through. |
| 07:05.13 | jkroon | will just add that if it's 0 to treat it the same as "" |
| 07:05.20 | MariusKarthaus | Hi, we have some strange behavior in a queue. Sometimes the calling party will keep hearing the ringing tone after they have been picked up by the agent. Does this (sorry for the pun) ring any bells? |
| 07:06.43 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 07:09.51 | din3sh | hello all |
| 07:11.52 | din3sh | I have a couple of cisco phones on SIP firmware running on TCP port, each time I issue a "sip reload", the phones lose their registration leading to SIP retransmit critical request. Only way to make the phones register is to issue a "core restart now" |
| 07:12.34 | *** join/#asterisk James87 (~wiv@unaffiliated/james87) |
| 07:14.26 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:5125:1fd9:3732:271e) |
| 07:14.38 | wdoekes | din3sh: you're saying that the TCP connections are not cleared, but the peers registrations are removed? |
| 07:15.17 | wdoekes | can you confirm that the connections are still up using netstat ? |
| 07:15.39 | wdoekes | if the TCP connection are tore down, the phones would know, and begin re-registering |
| 07:16.06 | ChannelZ | they are Crisco! |
| 07:18.01 | din3sh | when i do a "sip reload", and try calling phone B from Phone A |
| 07:18.19 | din3sh | CLI shows "called Phone A", phone does not ring |
| 07:18.37 | din3sh | sip debug on shows invite retries 10 times |
| 07:18.45 | din3sh | then i get a sip retransmit error |
| 07:20.26 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:201b:6b5f:cad1:fe67) |
| 07:22.46 | din3sh | all the phones are on the same network, with no NAT settings, no firewall |
| 07:29.17 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 07:30.56 | jkroon | wdoekes, issue found and fixed, someone set a ring time of 0 (which is legal for my requirements, even though unusual), this short-circuited the Dial() statement and due to no hunting/voicemail/forwarding resulted in Hangup() being executed with 0. Changed it to Hangup(17) (busy) now if HANGUPCAUSE=0. Thanks for the help. |
| 07:33.45 | *** join/#asterisk bombev (~User@PPPoE-Static-40-132.UnicsBG.Net) |
| 07:34.06 | bombev | hi |
| 07:35.12 | bombev | is it possible to configure asterisk somehow when I got SIP/2.0 403 Forbidden I want in my asterisk log to see Hangup cause 19 |
| 07:35.14 | bombev | not 34 |
| 07:36.31 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:36.52 | bombev | which file should I modify chan_sip.c or causes.h |
| 07:38.28 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-nzgwxpuakaoeicce) |
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| 07:50.37 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 07:52.03 | jkroon | why 19? |
| 07:52.13 | *** join/#asterisk ghost75 (~trechber@dslb-188-105-016-160.pools.arcor-ip.net) |
| 07:52.21 | jkroon | 19 isn't appropriate, but you can remap 34 onto 19 i guess ... |
| 07:52.24 | jkroon | in dialplan |
| 07:54.09 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 07:54.10 | *** join/#asterisk k611 (~K610@cred.epid.ucl.ac.be) |
| 07:57.07 | jkroon | and your change if you really want to change the code itself would be chan_sip.c |
| 07:58.29 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.119) |
| 08:09.55 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 08:13.46 | ectospasm | changing the code should be a last resort, it might have unintended consequences... |
| 08:17.18 | fling | Hello! Is `Cisco ATA 186´ good? |
| 08:35.30 | andross | fling: yes better than mediatrix imho |
| 08:37.25 | fling | andross: great! thanks |
| 08:37.44 | fling | andross: can't find any pata in this small town |
| 08:38.03 | fling | andross: some guy is selling this cisco for $40 |
| 08:38.10 | coppice | the ATA186 is sooooo retro :-) |
| 08:38.11 | fling | andross: I'm about to buy |
| 08:38.21 | fling | coppice: retro but fine? |
| 08:38.52 | andross | for a 2 port ata i found them to be solid devices |
| 08:39.06 | andross | i used them with sip mostly |
| 08:39.20 | coppice | they are better than most |
| 08:39.26 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 08:41.59 | fling | great! |
| 08:43.48 | fling | should be better than D-Link DVG-2102S, right? |
| 08:46.29 | coppice | what's wrong with the D-Link? |
| 08:46.58 | TriJetScud | hey coppice |
| 08:47.24 | *** join/#asterisk Diffen (~Diffen@c-7476e555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 08:49.24 | Diffen | Hello, does anyone of you know of a good billing system for voip? It should be webb based and should not only handle cdr information but also subscriber information and so on. |
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| 09:35.10 | *** join/#asterisk MalMen (~MalMen@bl6-156-183.dsl.telepac.pt) |
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| 10:31.05 | kareena | hello |
| 10:31.06 | kareena | i have a sip account how to configure it for incoming calls? |
| 11:46.03 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 11:46.03 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.3.0 (2013/03/28), 10.12.2 (2013/03/27), 1.8.21.0 (2013/03/28), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 11:46.31 | bombev | WIMPy ok i will take a look, but i don't wanna mess something... :) |
| 11:47.03 | WIMPy | It's messy anyway. |
| 11:47.26 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:a42:224:1dff:fecd:234c) |
| 11:47.29 | bombev | yeah, I see |
| 11:48.17 | bombev | WIMPy can you see that row 23123 : http://doxygen.asterisk.org/trunk/d5/dfe/chan__sip_8c.html |
| 11:48.29 | WIMPy | no |
| 11:48.39 | Schreda | Point to multipoint |
| 11:48.49 | Schreda | this i knew already |
| 11:48.55 | Schreda | msni |
| 11:49.10 | WIMPy | Schreda: Ok, that makes DDI impossible anyway. |
| 11:49.19 | bombev | this is not what I need |
| 11:49.32 | kaldemar | MalMen: what is that config exactly? |
| 11:50.26 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.39) |
| 11:50.46 | kaldemar | MalMen: and what version are you using? |
| 11:50.54 | WIMPy | Schreda: I'm not really sure how you could configure dahdi/asterisk to ignore the MSN. Are you sure your telco sends it to you? |
| 11:51.13 | WIMPy | OTOH someone else had the same issue. So maybe it is possible. |
| 11:51.16 | Schreda | I see the phone number on the display |
| 11:51.27 | Schreda | but I just talk right now to the customer service... |
| 11:51.44 | Schreda | and seemse like that they didnt make all adjustments correct... |
| 11:51.49 | Schreda | Checking right no |
| 11:51.49 | Schreda | w |
| 11:52.52 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 11:56.37 | kaldemar | MalMen: following parameters in your config are invalid, feel free to remove these: username, registername, from, localnet. "allow" only takes one value per line. so you need to divide alaw&ulaw into to two separate allow lines. |
| 11:58.27 | ectospasm | kaldemar: allow can be a comma separated list: allow=g729,ulaw,alaw,gsm,etc... |
| 12:00.25 | *** join/#asterisk biomorph (~biomorph@91.85.204.16) |
| 12:00.32 | kaldemar | ectospasm: so it seems. and even disallow can be stuffed in an allow line with !. |
| 12:00.58 | ectospasm | that I did not know. Kinda cuts down on readability, doesn't it? |
| 12:01.10 | kaldemar | allow=!all,alaw,ulaw |
| 12:01.12 | kaldemar | oh well. |
| 12:01.26 | eirirs | ! negates |
| 12:01.41 | ectospasm | I understand that, but it's not as readable as "disallow=all" |
| 12:01.52 | bombev | WIMPy I got lost... in chan_sip |
| 12:01.53 | eirirs | it's for coders hehe |
| 12:02.07 | eirirs | when I read !yes I automatically understands it like no |
| 12:02.19 | gusto | ¬yes |
| 12:02.21 | eirirs | it's the same everywhere |
| 12:02.31 | kaldemar | it's like using "unless not" in perl or something. |
| 12:02.43 | gusto | i have ¬ on my keyboard :-P |
| 12:02.50 | eirirs | gusto: haxboard |
| 12:02.56 | gusto | no |
| 12:03.00 | gusto | UK layout |
| 12:03.05 | eirirs | hehe |
| 12:03.09 | eirirs | the lands of coders |
| 12:03.13 | WIMPy | bombev: That's normal. Maybe it will be better with chan_gulp. |
| 12:03.18 | gusto | no, i am in slovakia now |
| 12:03.27 | gusto | but you can get US and UK keyboards here |
| 12:03.35 | gusto | sometimes |
| 12:03.41 | gusto | ... ehm ... when they have them |
| 12:06.02 | bombev | what is the path location of chan_gulp do you know? WIMPy |
| 12:08.21 | eirirs | slovakia... Hostel movies <o> |
| 12:09.27 | WIMPy | bombev: It will be in Asterisk 12. |
| 12:09.48 | bombev | WIMPy well I use asterisk 1.8 |
| 12:13.45 | tzanger | hostel was a terrible movie. cliche I guess |
| 12:14.05 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 12:14.13 | tzanger | I waited so long to see it and finally saw it and felt even worse because I waited for it heh |
| 12:23.29 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
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| 12:38.39 | *** join/#asterisk DND (5ec8071a@gateway/web/freenode/ip.94.200.7.26) |
| 12:38.44 | DND | hi guys im getting this from asterisk cli: Span 1: D-channel is down! / Primary D-Channel on span 1 up |
| 12:38.49 | DND | over and over |
| 12:39.20 | WIMPy | Bad timing or hardware issue. |
| 12:39.55 | DND | i can see that i get 1 irq miss |
| 12:39.58 | bombev | [TK]D-Fender do you any idea.. about my question |
| 12:40.23 | [TK]D-Fender | bombev, I wasn't here for it. |
| 12:40.29 | WIMPy | Check all cables first. |
| 12:40.49 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 12:42.42 | bombev | [TK]D-Fender I wanna somehow to change the hangup cause to 20 instead 34, because when I get SIP/2.0 403 Forbidden in my asterisk log I see hangupcause 34 |
| 12:43.07 | bombev | this has to be done in chan_sip but I got lost there.. :( |
| 12:43.10 | [TK]D-Fender | bombev, vi chan_sip.c <- |
| 12:43.33 | [TK]D-Fender | bombev, Go make a feature request on the tracker if you can't find it for yourself. |
| 12:43.45 | kaldemar | bombev: can't you trap it in dialplan and do a Hangup(20)? |
| 12:45.02 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 12:45.07 | eXcAliBuR | Good Morning Everyone |
| 12:45.11 | eXcAliBuR | I'm back, did you miss me? |
| 12:45.21 | [TK]D-Fender | eXcAliBuR, With every bullet so far! |
| 12:45.23 | eXcAliBuR | waits for at least one person to say yes |
| 12:45.29 | eXcAliBuR | :[ |
| 12:46.04 | *** join/#asterisk sjs205 (~sjs205@host81-151-44-189.range81-151.btcentralplus.com) |
| 12:46.46 | eXcAliBuR | I got home, and tested my sip phone which connected through a router to the interweb, It can call out find, but not receive calls. I remember having this prob before when there was a firewall and natting happening, something with having to use TCP and not UDP or some other settings |
| 12:47.00 | eXcAliBuR | I only have port 5060 forwarding to the phone |
| 12:47.02 | bombev | kaldemar well I have in my dialplan HangupCause20 |
| 12:47.07 | bombev | but I still got 34.. |
| 12:47.08 | eXcAliBuR | any ideas? |
| 12:47.09 | bombev | :( |
| 12:47.24 | [TK]D-Fender | bombev, Show us |
| 12:47.25 | kaldemar | bombev: what ever you do, don't show it to anyone! |
| 12:48.21 | [TK]D-Fender | eXcAliBuR, qualify=yes <- NAT keep-alive. And don't forward on the phone-side |
| 12:48.54 | eXcAliBuR | do I put that in my sip.conf under the phone extension? |
| 12:49.11 | DND | [TK]D-Fender: i did check the cable i can see dahdi is saying "ok" |
| 12:49.30 | DND | but it still says up/down |
| 12:49.34 | [TK]D-Fender | eXcAliBuR, yes |
| 12:50.17 | DND | i did dahdi_maint -s 1 and the result has all value of -1 |
| 12:51.47 | bombev | [TK]D-Fender http://pastebin.ca/2354686 |
| 12:52.36 | bombev | and actually my cell phone is switched off and I got 403 Forbidden.. |
| 12:52.51 | bombev | my goal is to change 34 hangup cause to 20 |
| 12:53.52 | [TK]D-Fender | bombev, [2013-04-10 14:47:26] VERBOSE[19990] chan_sip.c: Reliably Transmitting (NAT) to 213.168.165.14:5060: <-- ITSP's are almost NEVER NAT'd. Fix your peer |
| 12:54.01 | sjs205 | Hello all, I'm trying to configure asterisk with opensips, but I'm aving some roblem with the database side of things. Am I correct in saying that one can choose either the odbc OR the mysql driver for the asterisk database? |
| 12:54.32 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 12:55.15 | *** join/#asterisk serafie (~erin@nat/digium/x-nehbjyjhcnrsbrmd) |
| 12:55.49 | [TK]D-Fender | bombev, and you have NOT done as kaldemar suggested you try to do |
| 12:56.10 | [TK]D-Fender | sjs205, yes |
| 12:56.14 | DND | [TK]D-Fender: any idea about my prob? is it config issue or hardware issue? |
| 12:56.14 | Schreda | Well now I have point to point connection |
| 12:57.23 | sjs205 | [TK]D-Fender, can you tell me whether there are any major differences between them? |
| 12:57.31 | [TK]D-Fender | DND, Could be anything.... because you've shown us nothing. |
| 12:57.36 | bombev | [TK]D-Fender about this what should I add to my peer.. |
| 12:57.51 | [TK]D-Fender | sjs205, One is MySQL direct. The other is.... whatever you set it up with. |
| 12:58.14 | kaldemar | bombev: it won't change to 20 by itself. you must do Hangup(20) in your dialplan when you get congestion. you've done it hard on yourself by using freepbx but it should be doable. |
| 12:58.16 | [TK]D-Fender | bombev, if I tell you thta you should NOT be treating it as though it were behind NAT ... what do YOU think? |
| 12:58.22 | sjs205 | Cheers [TK]D-Fender... |
| 12:58.25 | sjs205 | :) |
| 12:59.24 | DND | [TK]D-Fender: its a newly installed asterisknow. i configured it: ccs/hdb3, pri_cpe, national ISDN2, LBO=0, pridialplan=national, prilocaldialplan=national, context= from-digital |
| 12:59.30 | eXcAliBuR | what protocol should I be using? |
| 12:59.33 | eXcAliBuR | tcp or udp |
| 12:59.34 | [TK]D-Fender | DND, PASTEBIN. |
| 12:59.45 | [TK]D-Fender | DND, ALL of it. Don't make this a guessing game. |
| 13:00.16 | [TK]D-Fender | eXcAliBuR, that is a transport, not a protocol. And I just told you some settings to make for it and heard nothing back regarding your attemps. |
| 13:00.17 | DND | ok hold on |
| 13:00.33 | eXcAliBuR | well i don't have the phone at the office |
| 13:00.34 | WIMPy | DND: Setting explicit dialplans is usually a bad idea, but not your concern at the moment. |
| 13:00.35 | eXcAliBuR | :/ |
| 13:00.42 | eXcAliBuR | it's at home |
| 13:01.01 | eXcAliBuR | but i can get one to try |
| 13:01.24 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 13:01.38 | bombev | kaldemar but if I change congestion to hangupcause 20, when I get 404 user not found I get congestion asl well |
| 13:02.18 | [TK]D-Fender | bombev, So far you have not done what he said. |
| 13:02.57 | *** join/#asterisk polysics (~Adium@host91-247-dynamic.20-79-r.retail.telecomitalia.it) |
| 13:03.08 | bombev | [TK]D-Fender in my peer details I dont have NAT feature.. ? |
| 13:03.14 | DND | http://pastebin.com/raw.php?i=QPeKvF4H |
| 13:03.27 | DND | that is chan_dahdi.conf dahdi-channels.conf and system.conf |
| 13:03.29 | polysics | hello! we have a SIP DID configured but for some unknown reason it goes "dormant" when not used for a while, ie. stops accepting calls |
| 13:03.41 | eXcAliBuR | mauahahaha it worked |
| 13:03.44 | polysics | the first time someone dials out, it starts working again |
| 13:03.45 | [TK]D-Fender | bombev, maybe you should explixitly SET IT |
| 13:03.56 | polysics | is there a setting we are missing? |
| 13:04.09 | eXcAliBuR | the qualify=yes did it |
| 13:05.05 | DND | i havent modified anything on the context yet using freepbx 2.11 beta 2.8, asterisknow 3.0.0 |
| 13:05.19 | bombev | [TK]D-Fender by using nat=yes ? |
| 13:05.32 | WIMPy | DND: Your timing configuration is bad. YOu need to use the line as timing source. |
| 13:05.34 | [TK]D-Fender | bombev, You don't seem to have read what I told you.... |
| 13:05.55 | bombev | well I did.. |
| 13:06.06 | DND | WIMPy: which one? signalling? |
| 13:06.29 | [TK]D-Fender | bombev, what did I tell you? |
| 13:06.30 | WIMPy | span= |
| 13:06.36 | [TK]D-Fender | DND, What are you connecting that to? |
| 13:06.42 | WIMPy | You have 0 as timing which means provide timing. |
| 13:06.45 | [TK]D-Fender | indeed |
| 13:06.46 | polysics | do I need qualify to fix the above? |
| 13:06.47 | [TK]D-Fender | ^ |
| 13:07.05 | DND | pri card? te121 connecting directly to telco |
| 13:07.13 | [TK]D-Fender | polysics, no. you just need 5060 clear to your server |
| 13:07.26 | [TK]D-Fender | DND, your SPAN line is bad as WIMPy has told you |
| 13:07.44 | [TK]D-Fender | DND, You are acting as the timing source instead of TAKING timing from the telco |
| 13:07.52 | polysics | [TK]D-Fender: sorry, but I do not understand. You say it is an issue with networking? |
| 13:07.54 | [TK]D-Fender | DND, And you are not going to win that fight |
| 13:08.13 | [TK]D-Fender | polysics, Most likely. IS 5060 clear to your server? Forwarded, etc |
| 13:08.37 | DND | ok i will disable freepbx interfereing with dahdi configs |
| 13:08.42 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 13:08.43 | DND | and manually set it |
| 13:10.18 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:11.43 | Schreda | working now |
| 13:12.04 | Schreda | I reconfigure the whole Sangoma card |
| 13:12.12 | Schreda | setup all config files again |
| 13:12.19 | Schreda | KEY was Point to Point |
| 13:12.36 | Schreda | otherwise I need to pay for the extensions on my orivuder |
| 13:12.38 | Schreda | provider |
| 13:12.42 | Schreda | THANKS A LOT |
| 13:12.42 | eXcAliBuR | I loveyou long time |
| 13:12.42 | bombev | [TK]D-Fender I put in peer details nat=yes |
| 13:12.44 | WIMPy | You said ptmp earlier. |
| 13:12.53 | Schreda | GREAT COMMUNITY |
| 13:13.01 | Schreda | But now I just called them |
| 13:13.13 | Schreda | they switched it back to Point to point |
| 13:13.22 | Schreda | than i setuped everything again |
| 13:13.25 | Schreda | work done :D |
| 13:13.26 | Schreda | HAPPY |
| 13:13.27 | Schreda | :D |
| 13:13.30 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:13.30 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:13.30 | Schreda | THANKS A LOTE |
| 13:13.31 | polysics | [TK]D-Fender: 5060/TCP is forwarded |
| 13:13.31 | Schreda | LOT |
| 13:13.35 | *** join/#asterisk willryder (~tedryder@nc-184-3-98-65.dhcp.embarqhsd.net) |
| 13:13.37 | WIMPy | That should make calls impossible. |
| 13:13.49 | WIMPy | Not just a missing number. |
| 13:13.49 | Schreda | no its working fine |
| 13:13.59 | Schreda | all works as expected |
| 13:14.06 | Schreda | can dial directly the extension number |
| 13:14.15 | Schreda | for incoming calls |
| 13:14.17 | Schreda | works well :D |
| 13:14.28 | DND | WIMPy: i did span=1,1,0,ccs,hdb3 but its still the same |
| 13:14.32 | WIMPy | So you also got DDI now? |
| 13:14.49 | [TK]D-Fender | polysics, unless you have configured otherwise.... *'s SIP is UDP |
| 13:14.51 | WIMPy | DND: Did you reconfigure dahdi? |
| 13:15.20 | [TK]D-Fender | polysics, You should probably show us your registration to your provider. |
| 13:15.43 | sjs205 | Can anyone tell me how I can debug the mysql driver since it seems connect but I constantly get "invalid database specified" but I know it *should* be right |
| 13:15.53 | [TK]D-Fender | DND, And you stopped * completely, re-initialized DAHDI and restarted everything? |
| 13:16.05 | DND | i did amportal stop |
| 13:16.13 | DND | that also stops dahdi |
| 13:16.25 | polysics | [TK]D-Fender: reg here: https://gist.github.com/polysics/88b2646828190157f881 |
| 13:16.29 | DND | then amportal start also does dahdi_cfg automatically |
| 13:16.42 | [TK]D-Fender | DND, that is a PEER and has nothing to do with registration |
| 13:16.54 | [TK]D-Fender | polysics, ^ |
| 13:18.31 | *** join/#asterisk mihamina (~mihamina@41.188.46.121) |
| 13:18.48 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-qcvhwoktxoajmuzp) |
| 13:18.48 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:19.44 | DND | i did amportal start/stop, service dahdi restart, cli> dahdi restart, module unload/load chan_dahdi.so |
| 13:20.52 | DND | this is dahdi_scan btw: http://pastebin.com/raw.php?i=15L9rWNm |
| 13:21.42 | WIMPy | irq=0? |
| 13:21.47 | DND | by the way the telco only told me this config: double-framing and clocksource = 16 |
| 13:22.22 | WIMPy | You won't get that thing working without IRQs. |
| 13:22.34 | WIMPy | That's rather useless. |
| 13:23.06 | DND | WIMPy: which one? |
| 13:23.20 | [TK]D-Fender | polysics, SIP DEBUG from a registration.... |
| 13:23.25 | bombev | [TK]D-Fender I put nat=yes but in peer details still get Reliably Transmitting (NAT) to 213.168.165.14:5060:.. |
| 13:23.39 | WIMPy | DND: The information you got. |
| 13:23.54 | DND | the telco are not much help |
| 13:24.19 | eXcAliBuR | I gots a new problem |
| 13:24.20 | WIMPy | Anyway: Yu need to fix that interrupt thing. |
| 13:24.40 | eXcAliBuR | My asterisk box is in a different time zone, so my voicemail timestamp is wrong |
| 13:25.42 | WIMPy | You can even set the tz per mailbox. |
| 13:25.44 | DND | the irq right? |
| 13:25.54 | WIMPy | yes |
| 13:25.58 | [TK]D-Fender | <bombev> [TK]D-Fender I put nat=yes but in peer details still get Reliably Transmitting (NAT) to 213.168.165.14:5060:.. <- what part of "they are NOT behind NAT" is unclear? |
| 13:26.30 | bombev | well I dont understand where I can fix this :( |
| 13:26.36 | [TK]D-Fender | eXcAliBuR, So go set your timezone for that box |
| 13:26.44 | [TK]D-Fender | bombev, NAT=NO. |
| 13:26.53 | DND | ok i need to loog on how to fix that. i have setup 3 boxes already and this is the first time i had problems with dahdi |
| 13:26.54 | [TK]D-Fender | bombev, I tell you they are NOT behind NAT ... you se it to NO <- |
| 13:27.00 | DND | *look |
| 13:27.15 | bombev | aha |
| 13:27.35 | WIMPy | DND: Read your dmesg or just try another slot. |
| 13:29.02 | DND | seems the echo canceller is the start: http://pastebin.com/raw.php?i=45VguUK3 |
| 13:30.25 | WIMPy | I wouldn;t expect that to be related. But that looks like your dahdi installation has some issue. |
| 13:31.15 | Katty | i am not having a good morning :< |
| 13:31.25 | DND | i will change slot. i remember pci-x 1x will work on a 16x slot |
| 13:31.36 | Katty | infobot: crittercam |
| 13:31.36 | infobot | crittercam is probably Katty's Critter Cam http://tinyurl.com/b5k3lt4 |
| 13:31.47 | Katty | maybe these guys will make me feel better. |
| 13:32.42 | WIMPy | DND: Oh, if that's an express card there isn't much that can go wrong other than a kernel that doesn;t support it. |
| 13:33.17 | DND | its a newly installed 32bit asterisknow |
| 13:33.28 | bombev | [TK]D-Fender sorry for giving you hard time.. |
| 13:33.29 | DND | you think i will go with 64 bit? |
| 13:33.47 | WIMPy | No, 32-bit is fine. |
| 13:36.22 | *** part/#asterisk Schreda (~Schreda@88.116.251.174) |
| 13:40.33 | DND | WIMPy: still there |
| 13:40.41 | DND | up/down |
| 13:40.56 | WIMPy | Do you have an IRQ now? |
| 13:41.34 | DND | dahdi_scan says irq=0 compared to 1 earlier |
| 13:41.54 | DND | i still got 1 irq miss from dahdi_tool |
| 13:42.11 | WIMPy | The one you pasted said 0. |
| 13:42.25 | bombev | i have another question can you take a look at : http://pastebin.ca/2354726 and see at row 47 I get SIP/2.0 407 Proxy Authentication Required |
| 13:43.03 | DND | is there another way to get irq? |
| 13:43.18 | WIMPy | cat /proc/interrupts |
| 13:43.21 | bombev | which feature should I use in peer details insecure=very or authuser=user... |
| 13:43.22 | eXcAliBuR | is there any sample configs that I should remove for security reasons, i did my own sip.conf and extensions.conf |
| 13:43.33 | [TK]D-Fender | bombev, [2013-04-10 15:14:36] VERBOSE[3564] chan_sip.c: Transmitting (NAT) to 213.168.165.14:5060: <-- Still. Not. Fixed. |
| 13:44.18 | DND | http://pastebin.com/raw.php?i=n5RfJpLi |
| 13:45.13 | [TK]D-Fender | <bombev> which feature should I use in peer details insecure=very or authuser=user... <-- insecure is for inbound calls, not outbound. |
| 13:45.16 | WIMPy | That looks better. |
| 13:45.42 | WIMPy | DND: Do a dahdi_test |
| 13:46.02 | [TK]D-Fender | bombev, And you are not showing the full call. |
| 13:46.03 | DND | 99.998 |
| 13:46.18 | WIMPy | Ok |
| 13:46.38 | Katty | [TK]D-Fender: i'll show you the full call in a minute. |
| 13:46.41 | Katty | ...what? |
| 13:46.52 | Katty | that was so much cooler in my head. |
| 13:47.44 | jmetro | it still sounds cool if you read it in a suggestive samuel l jackson voice |
| 13:47.46 | bombev | [TK]D-Fender I just made new call: http://pastebin.ca/2354738 |
| 13:47.46 | DND | hmm |
| 13:48.01 | kaldemar | bombev: are you using users.conf? |
| 13:48.11 | bombev | btw I already have NAT=NO in peer details |
| 13:49.12 | bombev | [TK]D-Fender row 120 SIP/2.0 407 Proxy Authentication Required.. |
| 13:50.30 | DND | WIMPy: seems i ran out of options. im doing a research on google now |
| 13:51.56 | WIMPy | DND: Maybe the card is too new for your dahdi? I have no idea how current the stuff is. |
| 13:53.05 | DND | the card is bought a year ago. and this is a new software |
| 13:54.52 | WIMPy | Missing symbols don't look like a sane installation. |
| 13:55.41 | DND | WIMPy: found the solution |
| 13:55.51 | DND | http://forums.asterisk.org/viewtopic.php?p=74273 |
| 13:56.12 | DND | in the lat post it says to do this: echo 2 > /proc/irq/233/smp_affinity |
| 13:56.33 | DND | i change 233 to 0 because the card has irq-0 |
| 13:57.19 | DND | now i get irq=177 |
| 13:57.42 | WIMPy | 0 is the system timer |
| 13:57.58 | [TK]D-Fender | bombev, fromuser=YOURUSERNAME , sendrpid=yes , trustrpid=yes |
| 13:58.07 | [TK]D-Fender | bombev, Change. Retry |
| 13:58.24 | bombev | ok |
| 14:00.08 | bombev | [TK]D-Fender I did what you wrote, but again I get IP/2.0 407 Proxy Authentication Required |
| 14:02.26 | *** join/#asterisk Rico29 (~rico@oceanet-telecom-fttb-129-2.olm.fr) |
| 14:02.31 | Rico29 | hello all |
| 14:02.43 | Rico29 | is there a way to indicate q-value on a registration ? |
| 14:05.38 | *** join/#asterisk morfin (~morfin@morfin.telenet.ru) |
| 14:05.40 | morfin | hello |
| 14:07.37 | morfin | can anyone tell me how AMI handles multiple Action: keys? |
| 14:09.55 | Kobaz | a merging we will go... |
| 14:11.49 | mjordan | you shouldn't specify multiple Action: keys in a single AMI action. The first action header found will be the action that gets executed, which will result in behavior that is unpredictable. |
| 14:11.58 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 14:12.13 | Rico29 | is there a way to indicate q-value on a SIP registration ? |
| 14:12.16 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 14:12.29 | igcewieling | Somewhere (here or the mailing list) I've seen a couple of reports of random crashes with Asterisk 11. Has anyone here experienced that? |
| 14:13.36 | Kobaz | igcewieling: all asterisk versions will crash at some point, given enough time and use cases |
| 14:13.52 | eXcAliBuR | i wanna call someone |
| 14:13.57 | eXcAliBuR | just to say Hi |
| 14:14.05 | Kobaz | the important part is posting bug reports with lots of debug data so we can fix the problem |
| 14:14.08 | igcewieling | Good to know, Kobaz! We'll migrate to a commercial soltion then. |
| 14:14.23 | Kobaz | igcewieling: those crash too |
| 14:14.36 | igcewieling | Kobaz: There is no chance in hell of getting us to do that when our production servers crash. |
| 14:15.08 | Kobaz | igcewieling: i'm thinking by commercial you mean like 3cx or something like that? |
| 14:15.29 | igcewieling | We have already decided to revert to 1.8. I'm just wondering if the Asterisk 11 crashes people are experiencing lock the entire system or JUST Asterisk. |
| 14:15.42 | Kobaz | no idea |
| 14:15.50 | Kobaz | 1.8 has been my baby for a while |
| 14:16.05 | Kobaz | i get one crash maybe every 45-90 days with 1.8 |
| 14:16.07 | mjordan | igcewieling: have you reported the problems you're seeing on the issue tracker? |
| 14:16.20 | Kobaz | running 20+ systems |
| 14:16.26 | igcewieling | Kobaz: our old Genband/Nextone/VoipLogic box has not crashed since I've been with the company (at least 3 years). It has had other issues like the DBs filling up, etc. But no actual crashes. |
| 14:17.02 | Kobaz | if you get an entire system lock, it's probably not asterisk's fault |
| 14:17.12 | igcewieling | Kobaz: I manage around 60 endpoint Asterisk PBXs and 4 core asterusj pbxs. The endpoints can crash occasionally |
| 14:17.32 | Kobaz | yeap |
| 14:17.39 | igcewieling | the core should not. |
| 14:18.15 | igcewieling | We've been quite annoyed with Asterisk Realtime 8-| |
| 14:18.27 | Kobaz | yeah dont use realtime |
| 14:18.37 | file | grumbles realtime |
| 14:19.28 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 14:21.39 | igcewieling | mjordan: not with the single crash we had and we don't want another. |
| 14:21.57 | igcewieling | But my original question is still unanswered. |
| 14:23.01 | Kobaz | which was what? |
| 14:23.16 | igcewieling | (10:12:29 AM) igcewieling: Somewhere (here or the mailing list) I've seen a couple of reports of random crashes with Asterisk 11. Has anyone here experienced that? |
| 14:23.42 | Kobaz | oh that one |
| 14:23.56 | Kobaz | i havent used asterisk 11 yet i'm still porting my patchsets |
| 14:25.16 | Kobaz | do you have to use 11? |
| 14:25.22 | elguero | igcewieling: haven't had any issues with crashes using 11 |
| 14:25.28 | mjordan | igcewieling: Understandable, but if you do happen to get evidence with a crash (core file, etc.) - please consider creating an issue. We can't fix the problems if we don't know about them. As far as your question goes, that's hard to answer without knowing what components you're using. With respect to SIP, at least, I'm not aware of any Asterisk 11 specific crashes or problems. |
| 14:25.28 | Qwell | igcewieling: There are random crashes on every version. Whether they affect you or not, is what matters. |
| 14:25.55 | Kobaz | Qwell: yeah that's what i said |
| 14:25.55 | igcewieling | we normally wait until around the *.20 release, which in my experience is around the time Asterisk becomes stable enough for our use. Asterisk 11 uses different versioning, but I still should have waited. |
| 14:26.13 | igcewieling | Kobaz: no, which is why we are reverting to 1.8 tonight |
| 14:26.21 | eXcAliBuR | looks at Qwell with big sad eyes |
| 14:26.23 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:26.23 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:26.28 | Kobaz | igcewieling: has your 11 crashed ? |
| 14:26.36 | Qwell | eXcAliBuR: O.o |
| 14:26.39 | igcewieling | Kobaz: yesterday, yes it did. |
| 14:26.40 | eXcAliBuR | O.O |
| 14:26.49 | Kobaz | do you have a core dump? |
| 14:26.59 | igcewieling | Kobaz: no. Had to remote power cycle the system. |
| 14:27.10 | eXcAliBuR | Kobaz: I had a Mega dump last night |
| 14:27.25 | igcewieling | To me that says a system problem, not asterisk, but I would LOVE TO KNOW if anyone else has had asterisk 11 crashes and what the crash was like. |
| 14:27.29 | Kobaz | sounds like it could have been a kernel problem |
| 14:28.42 | igcewieling | Though we now have approval for an IP KVM for the NOC 8-) |
| 14:30.18 | jmetro | you didnt already have one? =p |
| 14:33.48 | igcewieling | jmetro: I've been asking for months |
| 14:34.04 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:34.12 | igcewieling | We never had crashes until we moved off 1.4/1.6 |
| 14:34.18 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 14:34.58 | igcewieling | at least 1.4, I don't recall if we had any actual crashes when we were on 1.6 or not. |
| 14:35.17 | igcewieling | 1.8 has been quite stable on the endpoint PBXs and pretty stable in our core |
| 14:36.51 | igcewieling | Qwell or file: Will Digium ever reset the "registration" count on g729 licenses? We replaced our software licenses with a hardware transcoding card and would love to have the reg count reset so we can register them on other systems. We had some ethernet changes which required us to reregister a couple of the licenses. |
| 14:37.52 | Qwell | igcewieling: You'd need to ask support. |
| 14:37.56 | igcewieling | Qwell: thanks. |
| 14:37.56 | Qwell | We have nothing to do with any of that. |
| 14:38.00 | drmessano | Thats what he told me too!!! |
| 14:38.09 | drmessano | !!! |
| 14:38.13 | igcewieling | I'm assuming "no", but I'll e-mail support |
| 14:38.57 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 14:43.03 | mmlj4 | hey, eric |
| 14:43.28 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-qcvhwoktxoajmuzp) |
| 14:44.09 | *** join/#asterisk navaismo (~navaismo@189.241.36.80) |
| 14:44.23 | igcewieling | hello mmlj4 |
| 14:44.24 | *** join/#asterisk k611 (~K610@cred.epid.ucl.ac.be) |
| 14:44.48 | mmlj4 | talked to John lately? |
| 14:44.54 | igcewieling | mmlj4: not in years |
| 14:44.58 | mmlj4 | heh |
| 14:46.34 | mmlj4 | hope everything's going well for you |
| 14:47.36 | igcewieling | reasonably well. Working for an ITSP/CLEC/ISP in NYC area, living in Pensacola |
| 14:48.47 | navaismo | Hi, back with the issue yesterday about the dahdi restart freezes the asterisk before I did the dahdi restart i got a pri debug with the following messages--->http://pastebin.com/rXdcmXpg |
| 14:49.01 | mmlj4 | oh, NYC? cool |
| 14:49.36 | mmlj4 | I'd live in p'cola, actually interviewed for a spot on one of the bases there, never made the first cut though |
| 14:50.02 | navaismo | I have tested all posible configs in system.conf and still seeing that errors, on pri show span 1 the cli freezes again.Downgraded dahdi to 2.6.1 persist the issue |
| 14:50.08 | igcewieling | I was in Huntsville for the past couple of years, lived at a campground for a few years before that |
| 14:50.34 | navaismo | If I disconnect the E1's cable the card doesnt show the alarm until I run dahdi_cfg |
| 14:50.43 | navaismo | and the same when i put it back the cable |
| 14:51.04 | mmlj4 | I knew about the nightmare on campground street |
| 14:51.37 | mmlj4 | I'm still stuck in the same apartment in metairie |
| 14:51.42 | igcewieling | navaismo: is this a Digium card or Sangoma card? |
| 14:53.04 | igcewieling | navaismo: Can you confirm the PRI works when connected to something else? |
| 14:53.37 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-jorucygnwcuvytqz) |
| 14:53.37 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:55.31 | navaismo | Digium card |
| 14:55.47 | navaismo | and no i cant confirm that |
| 14:56.27 | *** join/#asterisk deo_ (~deo@112.198.90.230) |
| 14:57.15 | slav3_kitten | god i have a log full of "flowroute is now unreachable" "voipms is now unreachable" and moments later them being reachable |
| 14:57.20 | slav3_kitten | my net blows |
| 15:03.45 | igcewieling | slav3_kitten: that happens even on non-bad network connections. Qualify is not as useful as it might seem. |
| 15:04.16 | igcewieling | try qualify=10000 and (maybe) qualifyfreq=120 |
| 15:05.39 | ghost75 | know any callmonitor software for asterisk under win? |
| 15:05.40 | slav3_kitten | igcewieling, also 40.6ms of jitter, and 36.2 kbps upload -1.5mbps upload randomly between those |
| 15:06.17 | igcewieling | 40 ms of jitter (not latency) is a bad network connection or excessive usage. |
| 15:07.15 | igcewieling | 40 ms of latency is not too bad |
| 15:08.09 | slav3_kitten | it's bad network connection. rural wisp isp of epic fucking fail |
| 15:08.47 | igcewieling | it is wireless, that is what you should expect. |
| 15:08.52 | slav3_kitten | also for some reason, not rtp transverses my network to wan after i pick up. i've checked all my router configs |
| 15:09.12 | slav3_kitten | and my static IP changed. shit was working perfectly before that happened |
| 15:09.34 | slav3_kitten | i did change all my configs to match the new static IP |
| 15:10.09 | jmetro | you paid for a non-static static ip? |
| 15:10.59 | slav3_kitten | nope i paid for a static ip, 10 bucks additional a month |
| 15:11.03 | slav3_kitten | on top of the 60 a month |
| 15:11.17 | jmetro | and then your static ip changed, making it not static? |
| 15:11.28 | slav3_kitten | yep |
| 15:11.42 | slav3_kitten | tech is out of the office until 2pm |
| 15:12.04 | jmetro | sounds like you should be getting a free static IP for the term of the contract if they cant gaurantee its static |
| 15:13.38 | Kobaz | heh |
| 15:13.40 | slav3_kitten | i also think they turned on the sip alg for my ubiquiti "modem" which is why sip is broken but iax works fine |
| 15:13.43 | [TK]D-Fender | No, it means there's a lot of static noise on the line which is why it drops :) |
| 15:14.18 | jmetro | oh you paid for a line with static on it... static ip! |
| 15:14.22 | slav3_kitten | lol |
| 15:14.24 | jmetro | you can edit those ubiquiti modems though |
| 15:14.41 | Kobaz | sip alg? |
| 15:14.50 | jmetro | application layer gateway |
| 15:14.52 | slav3_kitten | jmetro, i can't. they can but i can't |
| 15:14.59 | jmetro | you dont know the login info for it? |
| 15:15.08 | Kobaz | mm |
| 15:15.49 | slav3_kitten | actually that's a good idea... i'll see what happens if i set my IP to the ip the ubiquit device has |
| 15:16.54 | jmetro | you mean in the same subnet, so you can manage it? |
| 15:17.06 | *** join/#asterisk v0lZy (~Thunderbi@84-255-194-41.static.t-2.net) |
| 15:17.32 | Kobaz | only 24 more files to merge! |
| 15:19.00 | slav3_kitten | yep, nope, still broken |
| 15:19.14 | slav3_kitten | i do know 5060 is forwarding correctly |
| 15:19.22 | slav3_kitten | just none of the udp packets |
| 15:19.31 | slav3_kitten | i'll fuck with it after i vacuum an make some food |
| 15:22.43 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-144-164.user.veloxzone.com.br) |
| 15:24.06 | *** join/#asterisk michaely (~michaely@38.101.34.1) |
| 15:26.53 | *** join/#asterisk chris_n (~Chris@koha/developer/chris-n) |
| 15:30.54 | michaely | Hello, I make an application that provides visual voicemail functionality for Asterisk. The application manages the voicemails by making direct file system modifications to the voicemail directory. I noticed that when I move a voicemail from the INBOX directory to Old or if I delete the voicemail from the directory the MWI light on my phone does not update. Is there any specific phone or asterisk configuration that needs to be in place for this to work? |
| 15:30.55 | michaely | <PROTECTED> |
| 15:31.07 | jmetro | update to asterisk 11 |
| 15:32.02 | michaely | jmetro: was that message for me? |
| 15:32.05 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 15:32.06 | eXcAliBuR | i'm back |
| 15:32.25 | eXcAliBuR | when I call with my 1-888 number, it shows toll free for the name and not the name I set |
| 15:33.21 | navaismo | eXcAliBuR, your provider allows the CID rewrite? |
| 15:33.30 | eXcAliBuR | yes |
| 15:34.06 | eXcAliBuR | this is my line: exten => _1NXXXXXXXXX,1,SET(CALLERID(all)="NAME" <8880000000>) |
| 15:34.21 | eXcAliBuR | i just put in a fake name n number |
| 15:35.38 | [TK]D-Fender | eXcAliBuR, Your carrier probably doesn't permit you to change it on demand |
| 15:35.54 | [TK]D-Fender | eXcAliBuR, Very few do. Usually you have to ask them what you want it set as. |
| 15:36.06 | eXcAliBuR | I use flowroute |
| 15:36.36 | [TK]D-Fender | Go look at your debug to see if it's actually even passed |
| 15:36.59 | [TK]D-Fender | eXcAliBuR, And make sure first that your peer has "sendrpid=yes", and "trustrpid=yes" |
| 15:43.59 | eXcAliBuR | [TK]D-Fender: can I pm you? |
| 15:45.25 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 15:47.42 | [TK]D-Fender | eXcAliBuR, Do not PM. And if you aren't looking with SIP DEBUG enabled ... you aren't really looking. |
| 16:02.30 | elguero | eXcAliBuR: What are you calling to? Cellphone? |
| 16:12.21 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 16:25.14 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 16:29.07 | *** join/#asterisk vlad_starkov (~vlad_star@87.117.139.75) |
| 16:30.14 | *** join/#asterisk MrMeek (~meekhime@172-4-223-5.lightspeed.toldoh.sbcglobal.net) |
| 16:35.58 | *** join/#asterisk vlad_starkov (~vlad_star@87.117.139.75) |
| 16:39.11 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 16:46.16 | eXcAliBuR | i'm back, i went to lunch |
| 16:46.26 | eXcAliBuR | elguero: i'm caling anyphone with caller id |
| 16:46.37 | eXcAliBuR | it shows the number correctly, but toll free for the name |
| 16:46.45 | eXcAliBuR | i did a sip debug and will pastebin it |
| 16:48.38 | elguero | eXcAliBuR: ya, well, you will not be able to set the name through the dial plan |
| 16:48.50 | eXcAliBuR | i have to email the provider? |
| 16:49.04 | elguero | flowroute offers CNAM storage |
| 16:49.22 | eXcAliBuR | i saw that |
| 16:49.29 | eXcAliBuR | but didn't kno wwhat it was for so didn't click on it |
| 16:49.31 | elguero | you have to set the name you want displayed for that DID number you are setting |
| 16:50.36 | elguero | eXcAliBuR: since this is asterisk support, don't want to go into too many details, but flowroute has the instructions on their site |
| 16:51.21 | eXcAliBuR | i asked support and they told me to do CID with my PBX |
| 16:51.25 | eXcAliBuR | :/ |
| 16:52.17 | [TK]D-Fender | eXcAliBuR, the NUMBER yes... the NAME, no. |
| 16:52.28 | eXcAliBuR | I love smart people |
| 16:52.30 | elguero | yes, you can set the CID (number) through the PBX, but the other end that receives the call has to lookup the number to get the name... |
| 16:53.08 | igcewieling | Outbound CallerID name is not transported between carriers. Any attempt to do so is futile and a waste of time. |
| 16:53.44 | igcewieling | Additionally, in my experience toll free numbers don't have CNAM anyway. |
| 16:56.57 | eXcAliBuR | my system is pending a reboot |
| 16:56.58 | eXcAliBuR | bbl |
| 16:57.04 | *** part/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 16:58.18 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 17:02.02 | slav3_kitten | yay unreliable internet |
| 17:02.13 | slav3_kitten | packet loss suddenly 40% |
| 17:02.43 | jmetro | get dialup? |
| 17:03.13 | igcewieling | dialup or cellular internet. That is what I had to do at one time. |
| 17:08.23 | kareena | hi |
| 17:08.29 | kareena | i have this exten http://pastebin.ca/2354909 |
| 17:08.38 | kareena | how can allow any extesion to be called |
| 17:08.39 | kareena | ? |
| 17:08.54 | kareena | now i can call only 0033XXXXXXXXX |
| 17:09.25 | kareena | when i try other exten i got the error : |
| 17:09.35 | kareena | [Apr 10 13:10:47] NOTICE[3294]: chan_sip.c:15566 handle_request_invite: Call from '8003' to extension '0XXXXXXXXX' rejected because extension not found. |
| 17:09.50 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 17:10.42 | [TK]D-Fender | kareena, That dialplan snippit is incomplete and we have no idea what the actual call is specifically looking for. |
| 17:11.01 | [TK]D-Fender | kareena, pastebin the ENTIRE context and a failed call with SIP DEBUG enabled. |
| 17:11.04 | [TK]D-Fender | ~pb |
| 17:11.04 | 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. |
| 17:11.05 | [TK]D-Fender | ^^^ |
| 17:15.41 | kareena | http://pastebin.com/F9KyKX9X |
| 17:16.27 | WIMPy | Make an exten calld 0467274739. |
| 17:17.06 | [TK]D-Fender | kareena, that is not a COMPLETE context ... and that is not a call with SIP DEBUG enabled |
| 17:17.23 | kareena | i have set sip debug on |
| 17:17.35 | sjs205 | I am using the opensips integration found here: http://www.opensips.org/Resources/DocsTutAsterisk18, can anyone tell me if I am right in thinking that using this database configuration, one can not create a dynamic peer? |
| 17:17.56 | [TK]D-Fender | kareena, Clearly not with that pastebin |
| 17:18.08 | WIMPy | kareena: And that dialplan snippet looks like the context for your phones. |
| 17:18.11 | [TK]D-Fender | kareena, that is not a complete call with SIP debug. |
| 17:18.26 | [TK]D-Fender | kareena, COMPLETE context. COMPLETE call debug |
| 17:18.34 | sjs205 | It would seem that the host and the fromdomain and the defaultip and... are al linked to a single field in the opensips database, therefor cannot be changed independently... |
| 17:19.23 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 17:20.02 | kareena | http://pastebin.com/1kg1pJ6H |
| 17:20.29 | [TK]D-Fender | kareena, that is not a call. |
| 17:20.37 | [TK]D-Fender | kareena, And we still don't have your COMPLETE context |
| 17:25.55 | kareena | where i will find it? |
| 17:27.46 | [TK]D-Fender | If you placed a call you'd see it.\ |
| 17:28.05 | [TK]D-Fender | You don't even have the line your previous faliures showed so you are not testing properly |
| 17:33.14 | cusco | hi folks |
| 17:33.54 | cusco | when using originate, 2 channel legs are created with 2 different uniqueid's, is it possible to have them with only one uniqueid ? |
| 17:41.18 | *** join/#asterisk Jagster` (~chatzilla@unaffiliated/jagster/x-9084543) |
| 17:43.38 | *** join/#asterisk ska (~ska@unaffiliated/ska) |
| 17:46.23 | [TK]D-Fender | cusco, Where do you get 2 channel legs from? |
| 17:46.32 | [TK]D-Fender | cusco, Originate dumps them in the dialplan |
| 17:47.44 | ska | Is is practical to use VPS servers for Asterisk ? |
| 17:48.02 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 17:48.33 | [TK]D-Fender | ska, Do you see a reason it'd be practical for you? |
| 17:49.33 | *** part/#asterisk polysics (~Adium@host91-247-dynamic.20-79-r.retail.telecomitalia.it) |
| 17:52.50 | igcewieling | BTW, if anyone has considered the Sangoma transcoding cards, we installed some last night, they appear to be working fine, though the install docs could have been a little clearer. |
| 17:53.37 | igcewieling | 720 channels of g729 goodness |
| 17:56.07 | jmetro | I wonder when we will stop "Sip Trunking" and start "PSTN Trunking" and finally get off the old system |
| 17:56.30 | ska | [TK]D-Fender: The cost of entry is low, but latencies can be hell. |
| 17:56.35 | WIMPy | ? |
| 17:56.53 | ska | [TK]D-Fender: just looking to see if experienced guys have done it successfully or no. |
| 17:57.15 | [TK]D-Fender | ska, Yes, plenty have on multiple techs |
| 17:57.35 | igcewieling | jmetro: neither term means much. I prefer "handoff" as in "SIP Handoff", "POTS handoff", "PRI handoff", etc. |
| 17:57.55 | jmetro | ^ Those terms then. |
| 17:58.37 | igcewieling | 90%+ of our customers using sip take analog or PRI handoff from us. The SIP part is from the on-site media gateway to us |
| 17:59.15 | WIMPy | That's even worse than pure SIP. |
| 17:59.36 | igcewieling | WIMPy: when they get a SIP PBX the handoff is changed to SIP. |
| 17:59.55 | igcewieling | but only a few unusually clever or unusually stupid customer have sip capable PBXs |
| 18:00.01 | WIMPy | A little less bad. |
| 18:01.17 | igcewieling | WIMPy: maybe. half of them think that because we provide dialtone to them we are also supposed to fix inside wiring issues, email issues, and copier issues, i.e. we are now their IT department! *eye roll* |
| 18:01.42 | WIMPy | Haha |
| 18:01.45 | *** join/#asterisk k610 (~K610@87.66.232.163) |
| 18:02.19 | jmetro | copier issues! |
| 18:03.39 | mmlj4 | igcewieling: all for a large fee :-) |
| 18:04.02 | jmetro | 500$ an hour for us to learn your copier and fix it |
| 18:04.24 | mmlj4 | they could get away with $150/hr at least in NYC |
| 18:04.28 | igcewieling | mmlj4: we refer then to a couple of local IT companies |
| 18:04.48 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:04.48 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:04.55 | igcewieling | some of them are actually willing to pay, they just don't know who to call. Those are not too bad. |
| 18:43.08 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 18:43.08 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.3.0 (2013/03/28), 10.12.2 (2013/03/27), 1.8.21.0 (2013/03/28), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 18:44.11 | zoid_ | Hi, I'm implementing an IVR over SIP and this dobut came out: It is possible somehow to establish a one way audio channel? I don't really need the audio from the caller for anything and not having it would halve the required bw |
| 18:45.16 | file | that functionality does not exist within Asterisk at this time |
| 18:45.46 | jmetro | wouldnt one-way audio stop their DTMF... |
| 18:46.21 | file | it's possible |
| 18:46.29 | zoid_ | jmetro: not if it's SIP-signaled |
| 18:46.30 | file | (that it would stop it) |
| 18:46.46 | file | yeah, if it is SIP signaled then the world is happy |
| 18:47.37 | zoid_ | file: maybe then putting openser+rtpproxy in the middle? |
| 18:47.51 | file | I don't have any knowledge of that for such a purpose. I can not comment. |
| 18:48.03 | zoid_ | ok, thanks :) |
| 18:48.26 | SuperNull | hey guys.. anyway to tunnel vmnotify through nat or some other way to get voicemail notification (linksys ATA) |
| 18:48.35 | file | I can comment on the peanut butter cookie I had. It was good. |
| 18:48.49 | zoid_ | lol |
| 18:49.32 | igcewieling | The best way to play audio to the caller without receiving audio from the caller is to not answer the call when you playback. |
| 18:49.33 | file | SuperNull, if you have configured chan_sip for NAT it should work, provided the remote side's device doing NAT allows return UDP traffic to reach the device on the UDP mapping |
| 18:50.37 | jmetro | I think if you need to halve your call by forcing one way audio you might have bigger problems |
| 18:51.18 | igcewieling | jmetro: I'm sure there are lots of issues doing it that way, but his question was not about problems, just how to do it. |
| 18:52.15 | file | yeah my mind is in SIP land, where it's not possible to explicitly do that to the stream - but if you do early media it has a chance of working depending on the equipment/provider |
| 18:52.15 | zoid_ | I do have issues, that wasn't the question (so does my boss, who pays for bw) |
| 18:52.53 | zoid_ | right now I'm in the process of calculating specifications for the system |
| 18:54.15 | [TK]D-Fender | <file> that functionality does not exist within Asterisk at this time <- ummm there's an easy way (functionally speaking) :) |
| 18:54.35 | [TK]D-Fender | file, Except for the BW part :) |
| 18:54.42 | file | ha |
| 18:55.26 | [TK]D-Fender | * expects RTp and will kill any call without it... |
| 18:55.38 | [TK]D-Fender | This idea sounds like a very dead-duck. |
| 18:56.04 | zoid_ | Is it really that wrong? why? |
| 18:56.46 | [TK]D-Fender | zoid_, There is no "setup half" concept anywhere. Answered = bi-directional or else. |
| 18:57.56 | WIMPy | Counfigoure your externip to 127.0.0.2? |
| 18:59.08 | zoid_ | [TK]D-Fender: I see, It's kind of non-standard |
| 19:00.35 | [TK]D-Fender | WIMPy, * still expects the RTP. Failure to get = DOA |
| 19:00.43 | WIMPy | Fake it |
| 19:01.18 | [TK]D-Fender | WIMPy, Sounds like serious rtp.c / chan_sip.c hackery required... |
| 19:01.56 | WIMPy | You don't have to do it within Asterisk. |
| 19:05.11 | *** join/#asterisk vlad_starkov (~vlad_star@83.220.238.105) |
| 19:10.18 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
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| 19:21.32 | Katty | weeeeeeeeeeeeeeeeeeeeeeeeeeeee |
| 19:22.03 | WIMPy | But not here in public |
| 19:22.49 | *** join/#asterisk jsjc (~Adium@249.Red-83-46-236.dynamicIP.rima-tde.net) |
| 19:27.38 | igcewieling | Yay! VZ just repaired another cable failure from Sandy. 8-| |
| 19:27.48 | igcewieling | Way to go Verizon! |
| 19:27.56 | jmetro | http://cache.ohinternet.com/images/b/b2/Slowpoke.gif |
| 19:28.35 | igcewieling | There are a couple of sites with "Won't Repair" status. |
| 19:31.42 | [TK]D-Fender | WIMPy, Perhaps if you have a proxy that will satisfy * on the inside end.... |
| 19:33.54 | igcewieling | seems like it would be easier to write the IVR in the proxy. Kamailio can at least play back pre-transcoded audio files |
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| 19:43.40 | zoid_ | igcewieling: will look into that |
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| 19:55.16 | zpotoloom | good evening |
| 19:55.19 | zpotoloom | can anyone please point me to the right direction when dealing with "aborted tx frames" and "rx overrun errors" in wanpipe status on E1 |
| 19:55.35 | WIMPy | interrupts |
| 19:56.15 | zpotoloom | both are growing and there are some audio quality issues, most probably related to this |
| 19:56.48 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
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| 19:58.04 | WIMPy | Yes and if you can hear it on occasions other than listening to a dialtone, it must be very bad. |
| 19:58.32 | zpotoloom | it's like once a minute, half a second burst of noise |
| 19:59.08 | zpotoloom | when listening to MOH for example |
| 19:59.21 | WIMPy | Check for other I/O activity at that time like disk. |
| 20:00.51 | zpotoloom | thanks i'll check that, currently machine is idle, i'll do some monitoring |
| 20:01.20 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.104) |
| 20:01.23 | WIMPy | Is the cards IRQ shared? |
| 20:02.15 | leifmadsen | is that actually still a thing? :) |
| 20:02.34 | jmetro | apparently yes. |
| 20:02.50 | WIMPy | Not if you have a PCIe card. |
| 20:04.07 | jmetro | i had to remember all my IRQ knowledge trying to setup a wireless USB on linux with ndiswrapper |
| 20:08.54 | zpotoloom | hmm, I can see that all the interrupts are handled by only one CPU core |
| 20:10.36 | Kobaz | do de do |
| 20:10.45 | Kobaz | sitting onsite at a customer, watching their queue |
| 20:10.46 | Kobaz | fun fun |
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| 20:30.32 | *** part/#asterisk Matthias (~Matthias@195.16.243.99) |
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| 20:36.20 | eureka^ | b/ 34 |
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| 20:52.42 | igcewieling | zpotoloom: Sangoma provides free support for their products |
| 20:53.28 | zpotoloom | igcewieling, thanks for info |
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| 21:03.18 | pabelanger | Kobaz: what version of asterisk you running? |
| 21:03.35 | pabelanger | for your queues |
| 21:05.20 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:07.32 | Kobaz | 1.8 |
| 21:08.31 | Kobaz | not using app_queue though |
| 21:08.47 | Kobaz | i love this customer |
| 21:08.53 | Kobaz | they have unsweatened iced tea on tap |
| 21:09.26 | Kobaz | and a keg |
| 21:09.37 | Kobaz | not that i'm a beer person but that seems like a nifty perk |
| 21:11.02 | Kobaz | haha |
| 21:11.03 | Kobaz | Caution: Avoid Upgrading and Downgrading At The Same Time |
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| 22:27.10 | cusco | hi |
| 22:27.29 | cusco | on MixMonitor |
| 22:27.30 | cusco | [2013-04-10 23:24:35] WARNING[31521][C-0000001e]: file.c:1238 ast_writefile: No such format 'gsm' |
| 22:27.48 | cusco | modules.conf has a few lines instead of autoload |
| 22:27.53 | cusco | is there something I should add? |
| 22:32.01 | cusco | nvm |
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| 23:22.28 | Cartes | hi all |
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