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| 01:12.55 | MacintoshCat | i need help getting the latest asterisk (with mysql module) on ubuntu can anyone help point me on the right way? |
| 01:27.13 | [TK]D-Fender | Depends onew what you mean by "mysql module" and where you're at now |
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| 01:36.22 | file | falls over |
| 01:40.17 | MacintoshCat | well i tried a few things and im at the point where i decieded to start witha fresh ubuntu install |
| 01:40.40 | MacintoshCat | i need the mysql driver |
| 01:40.46 | MacintoshCat | sqlite3 isnt good enough for me |
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| 01:52.57 | [TK]D-Fender | To do what? |
| 01:53.58 | MacintoshCat | i guess no one knows how to do it :P lol |
| 01:55.19 | [TK]D-Fender | WHICH one are you talking about |
| 01:55.29 | [TK]D-Fender | For what purpose? |
| 01:55.59 | [TK]D-Fender | Careful not to talk about a thing as though the name "mysql" only applies ONE WAY. |
| 01:56.22 | [TK]D-Fender | And that there aren't multiple ways to acheive the same end |
| 01:59.19 | MacintoshCat | asterisk realtime mysql module |
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| 01:59.56 | MacintoshCat | i need to make sure i have it, im assuming the latest version of asterisk (13.2) has it back (dont know why it was removed in earlier versions) |
| 02:00.21 | [TK]D-Fender | What do you see in menuselect? |
| 02:01.04 | MacintoshCat | i need help getting the latest asterisk (with mysql module) on ubuntu |
| 02:01.09 | file | it has never been "removed" |
| 02:01.26 | MacintoshCat | it has because i installed on ubuntu and it wasnt there |
| 02:01.38 | MacintoshCat | i had sqlite3 tho lol |
| 02:01.53 | MacintoshCat | and im pretty sure it was an old version (v11) |
| 02:02.05 | file | that doesn't mean anything - if building by source then it won't build if the dependencies aren't meant |
| 02:02.12 | file | if using a package then it's up to the packager |
| 02:02.24 | MacintoshCat | so anyways back to square one |
| 02:02.27 | MacintoshCat | i need help getting the latest asterisk (with mysql module) on ubuntu |
| 02:02.30 | MacintoshCat | thats the bottom line |
| 02:02.51 | MacintoshCat | if you dont know what im talking about asking me what i need it for doesnt help |
| 02:02.55 | MacintoshCat | :\ |
| 02:03.34 | [TK]D-Fender | <MacintoshCat> it has because i installed on ubuntu and it wasnt there <- installed HOW? From their repos? |
| 02:05.53 | MacintoshCat | [TK]D-Fender: your not helping |
| 02:06.00 | [TK]D-Fender | You're not answering |
| 02:06.06 | MacintoshCat | im asking how to do it right, not how i did it wrong |
| 02:06.13 | [TK]D-Fender | Because if you are using their packages... then they have to package |
| 02:06.14 | [TK]D-Fender | <PROTECTED> |
| 02:06.18 | [TK]D-Fender | We didn't make those |
| 02:06.18 | MacintoshCat | i was askig if anyone know how to get latest asterisk on ubuntu |
| 02:06.26 | MacintoshCat | i understand that man |
| 02:06.26 | [TK]D-Fender | Compile it yourself |
| 02:06.38 | MacintoshCat | aite |
| 02:06.56 | [TK]D-Fender | Download *, meet the DEPENDENDECIES for it to include MySQL support... and it will have it |
| 02:08.08 | [TK]D-Fender | menuselect will tell you if and what you're missing. |
| 02:14.06 | MacintoshCat | ok thanks |
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| 07:17.54 | UncleKiwi | Hey just wondering about how getting a call back after a blind-transfer when there is now answer |
| 07:20.48 | UncleKiwi | I thought it would be automatic to return the call back to the transfer'er if the target person did not answer |
| 07:21.02 | UncleKiwi | but it just seems to keep ringing and ringing |
| 07:21.20 | UncleKiwi | until the preson being transfered gives up |
| 07:25.50 | UncleKiwi | anyone ? |
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| 07:41.18 | ChannelZ | I don't think it's easily done |
| 07:41.32 | UncleKiwi | thanks ChannelZ for the reply |
| 07:41.45 | UncleKiwi | Im sure I have configured it in the past |
| 07:41.50 | UncleKiwi | hmmm |
| 07:45.15 | UncleKiwi | whats the norm for this kind of situation |
| 07:45.19 | ChannelZ | It depends. Is it a feature code transfer or a SIP phone function transfer? |
| 07:46.15 | ChannelZ | Feature code transfers have a callback of some sort I think - see features.conf |
| 07:47.00 | UncleKiwi | yes i do recall this being part of the issue. im trying to get the best of both |
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| 07:47.42 | UncleKiwi | yeah by default features.conf is configured to work in this way |
| 07:49.01 | UncleKiwi | its probably the sip phones transfer feature |
| 07:49.17 | ChannelZ | That's more difficult |
| 07:49.23 | UncleKiwi | i guess i can disable it |
| 07:50.35 | UncleKiwi | im sure i have had it working in the past... when i figure it out ill have to note it down |
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| 08:16.05 | UncleKiwi | Hey ChannelZ you say its more difficult to get that functionality working what does it involve ? |
| 08:17.43 | UncleKiwi | Attendant Console Call Park Code |
| 08:17.51 | UncleKiwi | seem to have this this might be a hint |
| 08:18.25 | UncleKiwi | maybe it can park the call as part of the transfering ? |
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| 08:41.54 | WIMPy | The thing is that it's not that easy to find out who transferred the call. |
| 08:42.19 | UncleKiwi | ah |
| 08:42.30 | WIMPy | With chan_sip you can find the device name as part of the channel name. |
| 08:43.11 | UncleKiwi | its interesting when i use a a phone attached to an ata and i do a blind transfer using the flash hook it returns the call |
| 08:43.15 | UncleKiwi | after 15 seconds |
| 08:43.35 | UncleKiwi | but when i use the soft keys on a cisco spa514g |
| 08:43.41 | UncleKiwi | it dont work |
| 08:47.00 | UncleKiwi | How would i find that out WIMPy ? |
| 08:47.59 | WIMPy | I'm pretty sure, google can come up with some examples. |
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| 09:21.00 | UncleKiwi | yay I go it going haha |
| 09:21.13 | UncleKiwi | using the soft keys also |
| 09:21.52 | UncleKiwi | http://www.voip-info.org/wiki/view/Asterisk+config+features.conf |
| 09:22.16 | UncleKiwi | the very last bits of this page for asterisk 11 - if your interested |
| 09:23.49 | WIMPy | Trouble is it only works for one channeltype and realies on undocumented behaviour. |
| 09:24.16 | UncleKiwi | yeah it feels a bit messy :-/ |
| 09:24.27 | UncleKiwi | your right |
| 09:24.47 | UncleKiwi | its not really the answer im after |
| 09:25.08 | WIMPy | But it's the best AFAIK. |
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| 10:34.12 | skrusty | anyone seen a situation before where dial calls a sip uri the first time, but subsiquent times it says called, but does nothing. |
| 10:35.01 | WIMPy | Which sip? |
| 10:35.09 | skrusty | chan sip |
| 10:35.25 | skrusty | dial(sip/some.uri,60,tr) |
| 10:35.27 | WIMPy | And nothing as in sitting, waiting or as in exit immediately? |
| 10:35.40 | skrusty | just says "called some.uri" and then nothing |
| 10:35.45 | skrusty | works fine the first time |
| 10:36.20 | skrusty | it'sn also an old version of asterisk! :/ 1.6.2.24 |
| 10:36.20 | WIMPy | First time after what? |
| 10:36.44 | WIMPy | Ugh. Replace that. |
| 10:36.50 | skrusty | as in, i can call the sip uri the first time no problems, try a second call after the first and nothing |
| 10:37.12 | skrusty | yeah, i know. not my setup though, so may be a lot of work to upgrade them |
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| 10:52.22 | cervajs2 | hi, is it possible use simultaneously asterisk STATIC realtime and extensions*.conf files ? i saw that in asterisk 13 this can be configured in stasis.conf |
| 10:52.33 | cervajs2 | my question is for asterisk 11 |
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| 15:11.29 | cervajs2 | <PROTECTED> |
| 15:11.29 | cervajs2 | <PROTECTED> |
| 15:12.41 | *** join/#asterisk zamba (marius@flage.org) |
| 15:13.04 | zamba | i'm getting the following error: [Mar 2 16:12:31] WARNING[1688]: chan_sip.c:16405 check_auth: username mismatch, have <video-test>, digest has <> |
| 15:13.40 | zamba | [Mar 2 16:13:08] NOTICE[1688]: chan_sip.c:28003 handle_request_register: Registration from '<sip:video-test@asterisk.lan>' failed for '192.168.238.57:5060' - Username/auth name mismatch |
| 15:13.52 | zamba | this is from a cisco tandberg c40 system |
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| 15:26.22 | cervajs2 | zamba: send your sip.conf section where you defined video-test user. do you have correct credentials for video-test user in tandberg? |
| 15:27.37 | zamba | cervajs2: i really think so |
| 15:28.12 | cervajs2 | zamba: wireshark is your friend |
| 15:28.54 | [TK]D-Fender | No need for 3rd party... * SIP debug dumps the packets just the same... |
| 15:29.05 | catphish | i have an asterisk pbx connected to an upstream sip trunk, i notice that calls arrive from the upstream provider and are then transferred internally, asterisk generates additional INVITE packets to the upstream SIP provider, is there any way to prevent this? the upstream provider does not need to know as i'm proxying the media anyway |
| 15:29.31 | zamba | http://pastie.org/9993779 |
| 15:31.29 | [TK]D-Fender | catphish: Only reason should be is if you're allowing REINVITES |
| 15:32.04 | catphish | [TK]D-Fender: that was my thought too, i was sure i'd disabled them, i'll have to check again |
| 15:33.49 | zamba | i see username is empty in the authorization.. is that the problem? |
| 15:34.35 | [TK]D-Fender | username mismatch, have <video-test>, digest has <> <---- looks pretty clear.... |
| 15:34.55 | [TK]D-Fender | Registration looked like ONE thing... call looked like another |
| 15:35.02 | zamba | so the client isn't sending the username? |
| 15:35.33 | catphish | [TK]D-Fender: sip show peer shows "DirectMedia : No" on the peer that the INVITES are being sent to |
| 15:36.02 | [TK]D-Fender | catphish: Show us an actual call and actual configs to match masking only the secret |
| 15:37.55 | catphish | [TK]D-Fender: i'd like to, though pulling the relevant data may be quite difficult |
| 15:43.13 | *** join/#asterisk mjordan (mjordan@nat/digium/x-sfmbrlgfjdezjmhl) |
| 15:43.13 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 15:51.12 | catphish | cloudshark isn't working :( |
| 15:52.51 | pabelanger | obligatory shark: http://i3.kym-cdn.com/photos/images/newsfeed/000/380/681/ef1.jpg |
| 15:53.04 | catphish | :) |
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| 15:55.41 | [TK]D-Fender | pabelanger: http://i.imgur.com/OXfzugf.png |
| 15:57.57 | pabelanger | <PROTECTED> |
| 15:58.40 | catphish | here's my problematic call, my host is 185.22.208.8, the call comes in from the upstream peer, but my asterisk then appears to reinvite it with what seems like a nonsense remote party iid |
| 15:58.43 | catphish | https://www.cloudshark.org/captures/b38fe1579d5b |
| 16:00.32 | catphish | some config: http://paste.codebasehq.com/pastes/2y9446v2n74rlnzr4n |
| 16:02.35 | [TK]D-Fender | peer4-3198a7 87.238.72.130 <- peer dump |
| 16:02.51 | [TK]D-Fender | 3.95697387.238.73.130 <--- debug |
| 16:03.03 | [TK]D-Fender | 72 != 73 |
| 16:03.10 | catphish | oh yeah :) |
| 16:05.02 | catphish | fixed: http://paste.codebasehq.com/pastes/ygr4hq4ky8ol5p3qrg |
| 16:08.34 | [TK]D-Fender | <PROTECTED> |
| 16:09.16 | catphish | RPID certainly seems to be what's killing the call, do you think enabling rpid might enable reinvites? |
| 16:09.17 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:10.47 | [TK]D-Fender | No, I think it's trying to update the identity and causing them to freak out |
| 16:11.55 | rrittgarn | I've actually had RPID update break L3 media gateways losing audio in one direction. I had to disable it on my ITSP Peers. |
| 16:12.15 | catphish | the peer actually told me previously that my rpid headers were breaking some calls, i thought disabling reinvite would fix it |
| 16:12.23 | rrittgarn | presented differently though. Call would process until they hit an IVR, when we'd send an update the audio stream dropped out |
| 16:13.10 | catphish | rrittgarn: this is what happens to me, incoming calls work, they hit an IVR, then when put through to a real handset a reinvite is sent and audio drops |
| 16:13.45 | catphish | in this case, the peer actually drops the whole call because the address in the RPID header is total nonsense |
| 16:13.57 | catphish | at least nonsense from my lay interpretation |
| 16:17.13 | rrittgarn | Yeah... turn off RPID Update |
| 16:17.16 | rrittgarn | for your upstream peer |
| 16:17.39 | rrittgarn | I don't use re-invite for my ITSPs, i keep the * box in the middle at all times |
| 16:17.52 | catphish | me too |
| 16:17.58 | catphish | i thought i'd disabled reinvite :) |
| 16:18.15 | catphish | how might i disable rpid update |
| 16:20.34 | [TK]D-Fender | sendrpid=NO |
| 16:21.47 | catphish | i guess i might have to do that, was hoping i could still send it in initial invites |
| 16:22.04 | catphish | but i don't think its particularly important for me to do so |
| 16:25.51 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
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| 16:58.57 | catphish | thanks both, i'll try disabling sendrpid, need to test the impact first |
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| 17:14.10 | DivideBy0 | When I originate a channel into my stasis app, but it doesn't answer or is a bad #, I don't get any info until it times out. I think this is by design, but is there a way to tell before the timeout (maybe with the AMI?) |
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| 17:37.15 | *** join/#asterisk infobot (ibot@rikers.org) |
| 17:37.15 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.2.0 (2015/02/06), 11.16.0 (2015/02/06), 1.8.32.2 (2015/01/28); Standard: 12.8.1 (2015/01/28); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 17:40.06 | *** join/#asterisk infobot (ibot@rikers.org) |
| 17:40.06 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.2.0 (2015/02/06), 11.16.0 (2015/02/06), 1.8.32.2 (2015/01/28); Standard: 12.8.1 (2015/01/28); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 17:54.37 | DigiDaz | Hi all, can anybody tell me please if outboundproxy is a global setting or can it be applied per trunk? |
| 17:55.19 | [TK]D-Fender | both |
| 17:55.25 | AL13N | what's a good windows opensource voip client that supports VP8 and opus? i myself don't use windows, but if i want to use voip/sip instead of skype, some people will have to switch too... |
| 17:55.42 | AL13N | i will give them accounts on my asterisk, which has VP8 and opus... |
| 17:55.51 | AL13N | anyone know what's a good windows sip client? |
| 17:56.29 | [TK]D-Fender | DigiDaz: You can see this clearly in the sample config as well. |
| 17:56.59 | DigiDaz | [TK]D-Fender: Thanks for that, I have a guy having trouble connecting to one of my boxes and I was hoping I can fire just his one trunk at kamailio as an outbound proxy, that should be possible then :) |
| 17:57.31 | DigiDaz | [TK]D-Fender: I don't use asterisk for years, it was just something I vaguely remembered |
| 17:58.00 | [TK]D-Fender | DigiDaz: I'd hate to thing you'd need a proxy for this issue... |
| 18:00.37 | DigiDaz | [TK]D-Fender: I'm 99% sure I don't but the client his having difficulty configuring his asterisk, what I'm trying to get him to do is configure a trunk direct to the carrier. What struck me though is that being as I do not have access to the client side either, it leaves me kinda blind, so though it is probably not needed, it does allow me to put something in the path so that I can at least do a bit of diagnostics when I come across clients like this |
| 18:01.20 | [TK]D-Fender | DigiDaz: So "marginally better than nothing". Gotcha |
| 18:03.18 | DigiDaz | <[TK]D-Fender I can also probably use it to sanitise caller ID etc, I'm only gonna use Kamailio in its simplest form so it should be very lightweight. - Thanks anyway |
| 18:04.10 | [TK]D-Fender | DigiDaz: Unfortunate that you'd have to sit in between for this. If that's the access that you've got and it does the job ... well that's what you need done. |
| 18:04.57 | DigiDaz | I'm still gonna send media direct so it will just be the SIP |
| 18:19.43 | *** join/#asterisk jiku (~jiku@1.23.119.146) |
| 18:29.54 | pabelanger | Not sure if anybody know |
| 18:30.02 | pabelanger | but when you setup call forward on polycom |
| 18:30.19 | pabelanger | does is use 181 Call is Being Forwarded or 302 Moved Temporarily SIP message? |
| 18:30.32 | pabelanger | Setting up test now |
| 18:30.36 | pabelanger | figured I'd ask |
| 18:30.41 | *** join/#asterisk Alex_Bkash (7c06ecf5@gateway/web/freenode/ip.124.6.236.245) |
| 18:31.58 | [TK]D-Fender | 0 |
| 18:32.01 | [TK]D-Fender | 302 |
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| 18:55.52 | overyander | are there any commercial asterisk conference systems comparable with webex or at&t connect? |
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| 19:43.34 | *** join/#asterisk Danielzt (bb264c84@gateway/web/freenode/ip.187.38.76.132) |
| 19:44.13 | Danielzt | Is there any way to forward a Response Header in asterisk 11? |
| 19:44.29 | Danielzt | The header can have any name... |
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| 20:04.06 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 20:14.35 | cervajs2 | <PROTECTED> |
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| 20:27.26 | *** join/#asterisk salz212 (~chatzilla@182.178.216.82) |
| 20:27.55 | salz212 | i am having severe problem. Asterisk server getting stuck on MYSQL dialplan application on connect |
| 20:32.06 | *** part/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 20:33.04 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 20:42.29 | Danielzt | Is there any way to forward a Response Header in asterisk 11? I need to tell originator some information of the answered call... |
| 20:43.43 | WIMPy | What exactely are you looking for? |
| 20:44.30 | Danielzt | My topology is: Customer -> Asterisk -> Proxy -> Provider |
| 20:44.51 | Danielzt | Sorry... Proxy A -> Asterisk -> Proxy B -> Provider |
| 20:45.04 | Danielzt | I need to tell Proxy A which provider was chosen by Proxy B |
| 20:46.08 | Danielzt | Like on Freeswitch: sip_rh_XProvider=123 |
| 20:46.39 | WIMPy | core show function CONNECTEDLINE |
| 20:47.42 | *** join/#asterisk CustosLimen (~CustosLim@unaffiliated/cust0slim3n) |
| 20:50.21 | Danielzt | Interesting... I'll check deep thanks |
| 20:52.29 | salz212 | i am unable to get a core dump, asterisk sever just get stuck and it does not crach. any hints there? |
| 20:53.21 | *** part/#asterisk mjordan (mjordan@nat/digium/x-goaupfisrylwhfwq) |
| 20:54.05 | WIMPy | That is unfortunatly the norm. But IIRC you can provoke a core dump with the right signal. |
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| 20:57.50 | salz212 | looking for such command then |
| 20:58.58 | WIMPy | The command is kill. |
| 20:59.14 | salz212 | i could get to find much help fom old mailing list related to mysql application stuck asterisk |
| 21:01.24 | *** part/#asterisk kharwell (kharwell@nat/digium/x-mfrsqmozvyovtiwi) |
| 21:01.41 | pabelanger | salz212, sounds like a deadlock: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationForADeadlock |
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| 21:04.16 | salz212 | pabelanger: the issues is has started happening to all 3 asterisk servers I have and all at the same time. plus its production so i cant really re-compile |
| 21:05.40 | pabelanger | sounds like a database lock or something |
| 21:05.50 | WIMPy | Astersk seems to be a very social software. |
| 21:07.57 | salz212 | rebooted the databases |
| 21:08.01 | salz212 | and they keep on stucking |
| 21:08.40 | pabelanger | well, without looking at asterisk and seeing what it is doing, you are basically just grasping at straws |
| 21:08.48 | pabelanger | the link above is the solution |
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| 21:30.20 | Danielzt | WINPy: connectedline doesn't seem to work... What I actually need is to insert a custom header on 200 OK |
| 21:30.23 | Danielzt | Is this possible? |
| 21:30.50 | WIMPy | Not with chan_sip. |
| 21:31.54 | Danielzt | =( Or maybe, is there any "standard" header that could traverse on the 200 OK? |
| 21:32.18 | WIMPy | That would be the caller ID stuff. |
| 21:33.04 | Danielzt | This is too public... ok, i try to figure out some other solution. Thanks |
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| 21:59.40 | WIMPy | Hmm. I just enabled debug_threads, yet when I checked, Asterisk seems to be unusually fast. Has there be some extreme optimization lately? |
| 22:00.24 | *** part/#asterisk cervajs2 (~cervenka@178.148.broadband4.iol.cz) |
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| 22:50.40 | rmudgett | Not for DEBUG_THREADS |
| 22:51.50 | WIMPy | No. I t must be somewhere else. It is in fact much faster that what I'm used without debugging enabled. |
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| 23:09.05 | eschmidbauer | hi |
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| 23:33.33 | *** join/#asterisk prelude2004c (prelude200@dsl-67-55-28-3.acanac.net) |
| 23:34.09 | prelude2004c | hi everyone, anyone can point me in the right direction why the confbridge has such bad quality. Regular sip calls are perfect but just the confbrige gives us distortion on the line when people are in it. Not sure what is causin git and i have looked at everything |
| 23:36.22 | Penguin | Check the codecs that are in use. |
| 23:36.37 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 23:39.24 | prelude2004c | ulaw |
| 23:39.34 | prelude2004c | it is very odd, it only happens when i am calling from the PSTN lines |
| 23:39.42 | Penguin | All of the participants are using ulaw? |
| 23:39.43 | prelude2004c | when i do a call internal from system to system its fine |
| 23:39.57 | prelude2004c | yup, all using ulaw which is coming from PSTN |
| 23:40.49 | prelude2004c | the odd part is.. when i make a regular call from pstn to SIP its perfectly clear |
| 23:40.55 | prelude2004c | when i joing the bridge it messes up |
| 23:45.16 | prelude2004c | could it have something with the allow = ulaw:20 ? |
| 23:45.20 | prelude2004c | the ptime setting |
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| 23:58.18 | prelude2004c | can anyone help? i am not sure where to look |
