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| 00:14.23 | Sudravirodhin | Anyone having issues with 12.4's modules.conf autoload? |
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| 01:23.27 | tinklebear | Anyone home? |
| 01:23.35 | Mango45 | waves |
| 01:24.47 | tinklebear | Can anyone tell me how to make a call using the AMI without having an actual sip phone |
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| 06:12.37 | rock1979 | hello |
| 06:13.06 | rock1979 | which version of spandsp is suggested for asterisk 11 |
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| 07:11.32 | p00h | xit |
| 07:11.34 | p00h | exit |
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| 07:59.13 | jkroon | hi all, is it possible to confirm that a remote peer is receiving UDPTL frames? |
| 08:01.52 | jkroon | i've got udptl debug enabled and it looks like asterisk sends a udptl frame to initially wrong port, but once a frame is received from the remote end it updates the destination, but the initially sent frame (seq=0) does not seem to get resent ?! |
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| 08:14.02 | coppice | does the first frame matter that much? |
| 08:17.49 | rock1979 | how can i configure asterisk to send sip 488 to the re-invite of T38 |
| 08:19.30 | rock1979 | currently it is sending SIP 200 OK with g711 in SDP |
| 08:24.23 | jkroon | coppice, not sure actually. but the first frame == part of negotiation phase. |
| 08:25.12 | jkroon | rock1979, t38_udptl=no |
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| 08:25.36 | coppice | jkroon: the first packet is usually just a no signal, CED or CNG indicator, so it doesn't matter a whole lot if it gets lost |
| 08:25.52 | jkroon | coppice, cool. |
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| 08:36.31 | rock1979 | jkroon i set t38_udptl=no in sip.conf for the peer and execute sip reload in asterisk cli but still asterisk is not sending SIP 200 with g711 in SDP |
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| 08:44.56 | jkroon | rock1979, run a sip show peer on the peer and check what it says regarding t38 |
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| 08:45.12 | jkroon | asterisk by default does not do t38 <=> passthrough gateway (there are patches out there) |
| 08:45.30 | jkroon | so by default if one legs switches to t38, the other leg has to as well. |
| 08:49.19 | coppice | jkroon: you only need to patch old versions of asterisk to act as a T.38 gateway. No patches are needed for Asterisk 10 or 11 |
| 08:50.22 | rock1979 | astertisk is working as gateway sip to dahdi call termination |
| 08:50.48 | rock1979 | what ever i do asterisk does not send sip 488 to t38 re-invites |
| 08:51.51 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 09:00.13 | jkroon | rock1979, can you disalbe t38 on the remote end? |
| 09:00.32 | jkroon | and then you need to have disallow=all; allow=alaw&ulaw in sip.conf for that peer. |
| 09:01.02 | jkroon | coppice, it's good to know that those patches has been integrated. |
| 09:01.39 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 09:02.03 | rock1979 | jkroon, the remote side is a very big telecom company and they are insisting that i should send them sip 488 if i dont support T38 |
| 09:02.30 | rock1979 | thus since i am not able to send SIP 488 they are not finalizing the interconnection test |
| 09:02.34 | jkroon | rock1979, what version of asterisk? if >=11.X, why not use T.38? |
| 09:03.02 | rock1979 | Asterisk 11.5.0 |
| 09:03.20 | rock1979 | it is working as media gateway sip to dahdi call termination |
| 09:03.24 | rock1979 | with libss7 |
| 09:03.25 | jkroon | ok, so do the gateway thing, run t38 between yourself and them, and ulaw/alaw to the DAHDI side. |
| 09:04.42 | rock1979 | yes that is exactly the thing that i am trying to do |
| 09:05.27 | jkroon | t38_udptl=yes,redundancy |
| 09:05.39 | jkroon | and if they're behind NAT, t38pt_usertpsource=yes |
| 09:05.54 | jkroon | that should result in a Trying + 200 OK for t38. |
| 09:06.00 | wdoekes | "t38pt_udptl" |
| 09:06.20 | jkroon | ag yes, of course, thanks for spotting wdoekes |
| 09:07.53 | rock1979 | actually i am behind nat |
| 09:08.38 | rock1979 | but ports for 5060, 10000-20000 and 4000-4999 are forwared to asterisk server on router |
| 09:09.05 | *** join/#asterisk areski (~areski@80.174.128.112.dyn.user.ono.com) |
| 09:09.06 | jkroon | and you have set the appropriate options to let it know what it's public IP is? |
| 09:09.10 | jkroon | or set up STUN? |
| 09:10.04 | rock1979 | yes everything expect fax is fine , i mean for normal calls |
| 09:16.34 | wdoekes | rock1979: what does asterisk return on the t38 reinvite then, if it isn't 488? |
| 09:17.05 | rock1979 | SIP 200 OK with G711u,G711a,G729 in SDP |
| 09:18.21 | wdoekes | that's odd. show us the entire INVITE and the 200 (sans sensitive info) |
| 09:19.40 | nunne | just moved from godaddy to geotrust. and created a .pem file (in the same way i did with godaddy). 1. geotrust.crt 2. geotrust-intermediate. 3. geotrust-end 4. private key .. But i get "SSL error loading private key file. </etc/ssl/private/geotrust_wild.pem>" . using openssl verify on the file it says that it's okay |
| 09:19.53 | nunne | this is asterisk 11.10.0 |
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| 09:21.32 | rock1979 | how can i send you the traces with removing sensitive info , i have trace as pcap file |
| 09:22.32 | jkroon | nunne, permissions most likely |
| 09:22.33 | wdoekes | tcpdump -nnvs0 -r the.pcap | sed -e 's/1.2.3.4/8.8.8.8/g;s/4.3.2.1/9.9.9.9/g' |
| 09:22.57 | wdoekes | | pastebinit |
| 09:23.38 | nunne | jkroon: it's stored in the same place as my godaddy cert with exactly the same permissions |
| 09:23.46 | jkroon | ownership? |
| 09:24.43 | rock1979 | ?? |
| 09:24.49 | nunne | root, but asterisk is running as root |
| 09:27.59 | rock1979 | wdoekes i couldnt understood what you mean |
| 09:28.17 | rock1979 | is it possible to do it in wireshark |
| 09:31.01 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 09:31.20 | wdoekes | sorry, I don't have time to explain how to extract text data out of a pcap. google? |
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| 09:47.04 | rock1979 | wdoekes: i found how to extract from pcap |
| 09:47.19 | rock1979 | will be sending in a few minutes |
| 09:52.04 | *** join/#asterisk tris- (tristan@2001:1868:a00a::4) |
| 09:55.51 | rock1979 | here is the trace as txt file https://www.sharefest.me/b8ff09a2b726ff2443d9e3848f475b25 |
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| 09:57.32 | rock1979 | wdoekes are you there ? |
| 10:07.59 | wdoekes | ok, that looks like a legit bug then. have you tried the latest 11.12 of asterisk? |
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| 11:12.15 | rock1979 | wdoekes , i was out of office just came back |
| 11:12.48 | rock1979 | i didnt tried with latest version * 11 vers'on |
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| 12:30.24 | masus | hi all, what options do we have for sending and receiving sms through asterisk without "bluetooth - chan_mobile". |
| 12:31.35 | masus | and chan_dongle has some bugs can't receive sms longer than 160 characters. |
| 12:37.22 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 12:37.50 | [TK]D-Fender | Nothing else for GSM like that |
| 12:37.58 | [TK]D-Fender | Asterisk is not an SMS platform. |
| 12:38.14 | [TK]D-Fender | You should stop trying to treat it as such and find some other platform and hardware |
| 12:39.17 | file | or service |
| 12:40.12 | [TK]D-Fender | Well.. maybe this: |
| 12:40.13 | [TK]D-Fender | http://www.openvox.cn/en/products/telephony-cards/gsmwcdma-cards/g400e.html?page=shop.product_details&flypage=flypage.tpl&product_id=79&category_id=6 |
| 12:40.22 | [TK]D-Fender | Eitehr way.... get googling. |
| 12:40.41 | [TK]D-Fender | This is not something we actively support |
| 12:41.04 | rock1979 | Fender |
| 12:41.23 | tparcina | [TK]D-Fender: If it isn't why it wouldn't become SMS platform? |
| 12:41.23 | masus | [TK]D-Fender: Thank You |
| 12:41.33 | rock1979 | i am still looking for a fix with T38 faxing |
| 12:41.39 | rock1979 | sip to dadi |
| 12:41.49 | pa | WIMPy, have you ever tried to use mISDN standalone? |
| 12:41.57 | rock1979 | do you mind if i send you the sip trace |
| 12:41.59 | [TK]D-Fender | rock1979: I still never saw the debug or got real details. |
| 12:42.12 | [TK]D-Fender | rock1979: I asked for * CLI with its SIP debug. |
| 12:42.18 | [TK]D-Fender | rock1979: AND the configs to match |
| 12:44.34 | rock1979 | Fender do you mind if i ask if you can provide me consultancy to solve this issue |
| 12:44.59 | *** join/#asterisk aross42 (~aross@CPEbcc8100f769b-CMbcc8100f7698.cpe.net.cable.rogers.com) |
| 12:45.05 | pa | as far as i understand, the standalone mISDN can be installed side by side with the kernel driver |
| 12:45.19 | pa | the only thing i don't understand is: how do i select what driver to use? |
| 12:45.28 | pa | if the standalone or the one that comes with the kernel? |
| 12:45.33 | [TK]D-Fender | rock1979: I am no expert in this particular aspect, but I can tell if steps have been skipped. And I have nothing to go on for this. |
| 12:47.46 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 12:48.13 | *** join/#asterisk bmurt (~brendan@8.39.115.8) |
| 12:49.31 | rock1979 | how can i check if spandsp is availble on my server or not ? |
| 12:52.09 | [TK]D-Fender | check your package management |
| 12:52.52 | rock1979 | ?? what did you mean by pachage management. It is an asterisk no frontend gui |
| 12:53.14 | [TK]D-Fender | DISTRO PACKAGE MANAGEMENT |
| 12:53.51 | [TK]D-Fender | works in caps too often |
| 12:54.36 | *** join/#asterisk theron (~theron@66.220.145.150) |
| 12:55.24 | rock1979 | :( |
| 12:55.38 | rock1979 | i have no idea what you are offering Fender |
| 12:55.42 | *** join/#asterisk ruied (~ruied@25.172.37.188.rev.vodafone.pt) |
| 12:56.58 | rock1979 | found it in google you talking about yum,apt-get etc... right ? |
| 12:57.29 | *** join/#asterisk lance (~ShapeShif@unaffiliated/shapeshifter499) |
| 12:57.29 | [TK]D-Fender | .... yes... |
| 12:57.32 | [TK]D-Fender | this is linux basics |
| 12:57.39 | [TK]D-Fender | Do you have no clue what you're running at all? |
| 12:58.38 | rock1979 | i didnt installed spandsp just checked in linux with command "whereis spandsp.so" |
| 12:58.50 | rock1979 | and where is didnt found anything |
| 12:59.11 | [TK]D-Fender | You haven't confirmed why you think you should ahve it. |
| 12:59.31 | [TK]D-Fender | And you're assuming that is a relevant file to its use |
| 13:00.36 | [TK]D-Fender | is thinking today is a good day to NOT go on a fishing trip. |
| 13:00.37 | rock1979 | i execute asterisk -rx "core show applications like fax" |
| 13:00.48 | rock1979 | and it shows |
| 13:00.58 | rock1979 | Matching Asterisk Applications =- |
| 13:00.58 | rock1979 | <PROTECTED> |
| 13:00.58 | rock1979 | <PROTECTED> |
| 13:00.58 | rock1979 | <PROTECTED> |
| 13:01.30 | file | those are the common dialplan applications, they still require a fax module to work - which is separate |
| 13:02.47 | rock1979 | file could you please help me how to check if spandsp is installed or not |
| 13:03.13 | rock1979 | <PROTECTED> |
| 13:03.41 | file | if you don't know if you have spandsp installed you most likely don't |
| 13:04.06 | rock1979 | yes i think so :) |
| 13:04.50 | [TK]D-Fender | hasn't seen qualified "thought" yet. |
| 13:07.10 | rock1979 | file : is this the reason that i am not able have faxing with T38 gateway sip to dahdi |
| 13:07.21 | file | if you don't have a fax module, then it won't work |
| 13:07.35 | rock1979 | it is asterisk 11.5.0 with libss |
| 13:07.44 | rock1979 | it is asterisk 11.5.0 with libss7 |
| 13:08.28 | rock1979 | calls are coming from sip and sent to telco with liss7 dahdi call |
| 13:09.12 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 13:09.26 | rock1979 | i set t38pt_udptl=no and setvar=FAXOPT(gateway)=no |
| 13:09.40 | rock1979 | but still when asterisk receives t38 reinvite |
| 13:09.58 | rock1979 | it is sending sip 200 ok with g711u,g711a,g729 in sdp |
| 13:10.02 | [TK]D-Fender | you DISABLED the gateway feature.... so why do you think it should WORK? |
| 13:10.15 | [TK]D-Fender | setvar=FAXOPT(gateway)=no <- FAIL |
| 13:10.52 | rock1979 | i am trying asterisk to send SIP 488 if t38 reinvite is sent |
| 13:11.28 | rock1979 | if i am able to send SIP 488 to t38 invite then the remote side will try t30 fax |
| 13:12.58 | rock1979 | i have also tried t38pt_udptl=yes and setvar=FAXOPT(gateway)=yes but when asterisk receives t38 reinvite it is sending sip 200 ok with g711u,g711a,g729 in sdp |
| 13:18.20 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 13:18.20 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:32.49 | *** join/#asterisk BKhan (ca7d90e3@gateway/web/freenode/ip.202.125.144.227) |
| 13:33.00 | BKhan | hi |
| 13:33.19 | BKhan | I need to install codec g729 for asterisk 1.4 |
| 13:33.36 | BKhan | any link |
| 13:33.40 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 13:33.58 | Qwell | ~upgrade asterisk |
| 13:33.59 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 13:34.02 | [TK]D-Fender | BKhan: www.digium.com |
| 13:36.44 | BKhan | D-Fender: thanks but in it also start from 1.8 |
| 13:37.07 | BKhan | http://my.digium.com/en/docs/G729/g729-download-overlay/?width=500&height=425&iframe=true |
| 13:37.48 | [TK]D-Fender | then you aren't looking very smart |
| 13:38.03 | [TK]D-Fender | one must search smarter to find the older bits |
| 13:38.23 | [TK]D-Fender | so look at one it DOES offer you ... and then look deeper |
| 13:38.28 | [TK]D-Fender | already found it |
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| 13:44.16 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 13:45.12 | linocisco | anybody from Bangkok , Thailand? |
| 13:46.07 | Mango45 | I know a guy who does VoIP there. Would that help? |
| 13:46.07 | Mango45 | http://www.dslreports.com/profile/1232816 |
| 13:47.55 | linocisco | would like to meet asterisk folks and asteriks product shops while I am in Bangkok |
| 13:48.51 | linocisco | because microsoft is based in Singapore and Thailand. Meaning MS would also put his foot on country where proprietary products are being used heavily |
| 13:49.24 | linocisco | I am wondering if there are asterisk folks in Bangkok |
| 14:03.15 | *** join/#asterisk jploh (~textual@49.146.179.249) |
| 14:03.23 | pa | WIMPy, just tried lcr-master with the mISDN standalone w. l1loop module |
| 14:03.43 | WIMPy | Hi pa |
| 14:03.44 | pa | well.. it seems it was better before, even. now asterisk->lcr does not work, and lcr->asterisk still does not work |
| 14:03.55 | pa | but i want to send you the patch |
| 14:04.00 | WIMPy | I haven't tried the standalone version for ages. |
| 14:04.07 | pa | maybe you can make it work, and anyway can put it online |
| 14:04.41 | WIMPy | The best place would be the isdn4linux mailing list. |
| 14:05.17 | WIMPy | Actually I'm not really sure why we have that standalone version at all. |
| 14:06.14 | pa | well, according to the cologne guys, the loop module is not in the kernel mISDN tree |
| 14:06.48 | pa | anyway, i sent you the url to the patch |
| 14:06.50 | WIMPy | A week or two ago I read that it didn't compile with recent kernels, which is pretty interesting, considering the kernels contain a version that works. |
| 14:07.01 | pa | feel free to post wherever you think it's most suitable |
| 14:07.09 | WIMPy | Ok, that would be a point then. |
| 14:07.38 | pa | WIMPy, now lcr 1.14 does not complain about the loopback module when executed |
| 14:07.47 | pa | but yeah, still doesnt work |
| 14:07.52 | pa | or maybe i just misconfigured it |
| 14:07.53 | WIMPy | Which makes it a little more questionable if using it in LCR is a good idea. |
| 14:08.14 | pa | i mean i still get all the events |
| 14:08.19 | WIMPy | But you did have the old version running, didn't you? |
| 14:08.21 | pa | but at some point lcr does not connect |
| 14:08.27 | pa | yes i have that one too |
| 14:08.34 | pa | yours |
| 14:12.06 | WIMPy | It really needs some tidting up. Looks like some change in the usb core made hfcsusb fail to work in NT mode. |
| 14:12.19 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 14:13.27 | pa | afaik ISDN is still widely used, if not home at least at telco level. it's a pity it got so abandoned on linux |
| 14:13.47 | pa | cologne seems still alive though, and responsive (they provided the patch) |
| 14:13.59 | pa | pity there's no direct support for asterisk |
| 14:14.17 | WIMPy | Well, it's abandoned everywhere. Even if there's nothing to replace it. :-( |
| 14:14.43 | sekil | isdn gets obsoleted every day |
| 14:15.33 | pa | sekil, here you can still buy ISDN lines |
| 14:15.43 | pa | i guess in germany too |
| 14:15.49 | pa | and in several other countries |
| 14:15.57 | pa | maybe it's not as popular as it used to be |
| 14:16.01 | sekil | yeah here too (PRI mostly) but nobody invests in new ISDN networks |
| 14:16.12 | pa | but for example it's the only way to get an additional cable for an ADSL line |
| 14:16.27 | sekil | everybody moves to SIP towards to customers |
| 14:17.42 | *** join/#asterisk Stefan27 (~Stefan27@212.247.4.149) |
| 14:18.16 | WIMPy | It is possible in germany if you are really persistant. Usually you will just get a parcel with a SIP-ISDN gateway. I don't have to tell you how well that works. |
| 14:20.37 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 14:23.06 | Stefan27 | I have 2 simple bridges in asterisk, [SIP/A,SIP/B] and [SIP/C,SIP/A]. A has put C on hold, which caused asterisk to start music on hold on channel SIP/C. I then call manager action Bridge with channel1: SIP/B and channel2: SIP/C . This destroys the two previous bridges and starts a new one with [SIP/B,SIP/C] however SIP/C still has music on hold... How can I stop the music on hold (with my |
| 14:23.06 | Stefan27 | bridge action?) |
| 14:23.36 | WIMPy | I don't know the current fugures, but 2011, Some 30% of all phone lines were (read) BRIs. |
| 14:23.36 | Stefan27 | "unhold"? |
| 14:25.41 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 14:28.57 | coppice | in most countries BRI is still around its historic peak market share |
| 14:29.38 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:33.46 | *** join/#asterisk keiths_ (~keiths@goatfarm.vianet.ca) |
| 14:34.44 | keiths_ | Hey guys, I am trying to park a call (blind transfer) and then have the parking number display on screen of the cisco phone, is this something anyone has done? |
| 14:36.13 | [TK]D-Fender | have you looked at the channel after transfer? |
| 14:36.44 | [TK]D-Fender | there is at least 1 obvious variable you could use... |
| 14:37.23 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-lwyscehvtucjcwan) |
| 14:37.23 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:37.41 | keiths_ | think I was over thinking this. Thanks [TK]D-Fender |
| 14:39.21 | WIMPy | tries to reboot his PCF and hopes he will have ethernet thereafter... |
| 14:44.47 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 14:47.27 | WIMPy | didn't get far enough to find out. |
| 14:55.59 | Stefan27 | there's no way to stop music on hold on a channel from AMI or CLI? |
| 14:56.28 | *** join/#asterisk mimage (~mimage@fedora/mimage) |
| 14:58.55 | [TK]D-Fender | Stefan27: What have you tried? |
| 15:05.26 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:06.32 | Stefan27 | A and B is talking with 2-way-audio. A also has a call with C but C is on hold because A sent a Re-Invite with (a=sendonly). From this state I want to unhold C and Bridge B and C (thus eliminating A) with some sequence of actions. I tried the AMI action bridge SIP/B SIP/C but this didn't stop the music on hold for C |
| 15:09.06 | [TK]D-Fender | So you tried a bridge... nothing else? |
| 15:09.39 | [TK]D-Fender | I think you should look at other options than this one approach you too that didn't turn out as you'd hoped |
| 15:10.34 | Stefan27 | I guess i could try redirect both into a parking lot and then bridge? |
| 15:11.45 | [TK]D-Fender | Sounds like progress.... |
| 15:12.09 | Stefan27 | Do you have a solution in mind? |
| 15:15.50 | [TK]D-Fender | steconsider your targets... and go follow up with your first idea |
| 15:16.22 | *** join/#asterisk lantizia (~lantizia@cpc15-stok15-2-0-cust34.1-4.cable.virginm.net) |
| 15:16.40 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-tssqwytuyzaxfrew) |
| 15:16.44 | lantizia | Hey - Does anyone know what other brands/manufacturers you might see a snom 320 being displayed as? |
| 15:16.56 | *** join/#asterisk ggayan (~ggayan@190.215.47.74) |
| 15:17.11 | lantizia | I seem to recall Elmeg for some reason - any others I don't know about (no not the Lync ones HP are doing) |
| 15:17.55 | lantizia | Only reason I ask is I've seen one rather cool looking re-branded/re-coloured snom 320 in that new film Lucy - used quite expensively even |
| 15:23.32 | Stefan27 | I don't really know what i have to play with. Redirect action may put both SIP/B and SIP/C back into the dialplan simultaneously, from there i can bridge them, but im not sure if that's a good solution |
| 15:24.23 | [TK]D-Fender | Stefan27: "good" is what works |
| 15:26.29 | Stefan27 | i guess but the bridge ami action seemed so much smoother, if only it had unholded and stopped the music |
| 15:29.24 | WIMPy | Maybe it's a good idea to discuss the possibility it might be condidered a bug? |
| 15:34.31 | [TK]D-Fender | I concur |
| 15:34.46 | [TK]D-Fender | or treated like one anyway. |
| 15:35.01 | lantizia | http://www.youtube.com/watch?v=nUfDN59PTvc |
| 15:35.23 | lantizia | There! that has a good quality video of it - it's a snom 320 - but what about those colours!? |
| 15:35.27 | Stefan27 | it's unwanted behaivour for me, but it's hard for me to judge if it's a bug since i haven't fully grasped how asterisk handles channels and bridges yet |
| 15:35.31 | lantizia | unless it's not a snom 320!? |
| 15:36.20 | Stefan27 | only just started reading the documentation https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+ManagerAction_Bridge it doesn't say much? |
| 15:37.12 | WIMPy | Stefan27: Well, I think either it has to stop MOH automatically or you need a way to do it manually. But I don't see how to do that. |
| 15:39.56 | *** join/#asterisk mokmeister (~mokmeiste@86-43-185-179-dynamic.b-ras1.pgs.portlaoise.eircom.net) |
| 15:42.18 | Stefan27 | so if it is a bug, can anyone report it at https://issues.asterisk.org/ ? |
| 15:42.26 | Stefan27 | or does one need to be member somehow |
| 15:43.12 | WIMPy | Create an account and do it. |
| 15:43.44 | Stefan27 | ok i will |
| 15:44.41 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
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| 16:06.30 | Mango45 | For future reference |
| 16:06.31 | Mango45 | exten => *,1,Goto(1) |
| 16:06.35 | Mango45 | This will cause issues |
| 16:07.21 | [TK]D-Fender | SMRT |
| 16:08.05 | WIMPy | "You asked for it, you got it" |
| 16:08.31 | Mango45 | It should just know that's not what I intended to type. |
| 16:08.53 | [TK]D-Fender | Which is precisely the opposite of how computers work |
| 16:09.00 | WIMPy | How would it know you don't intend to do some load testing? |
| 16:10.18 | Mango45 | I'd install Lync or 3CX for that. |
| 16:12.39 | *** join/#asterisk dms1 (~dms1@65.207.151.254) |
| 16:34.07 | mjordan | Stefan27: I think we could consider it a bug, but at the same time, you're trying to perform an attended transfer after a user already initiated one. That's kind of odd. |
| 16:37.57 | *** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
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| 16:41.05 | Stefan27 | User A calls C then holds C and calls B and then user A sends 'manager action bridge' to bridge B and C (through web (php)). Why is it odd? |
| 16:41.41 | Stefan27 | Yeah, it's an attended transfer scenario - i guess |
| 16:43.15 | *** join/#asterisk MarcoZink (~marcozink@187-177-157-192.dynamic.axtel.net) |
| 16:43.41 | Stefan27 | i described it in ASTERISK-24281 (not in much more detail though) but it should be reproducable (if it indeed is unwanted) |
| 16:45.27 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 16:45.54 | Stefan27 | user A has both a phone and a web interface to communicate with asterisk |
| 16:46.25 | *** join/#asterisk CeBe (~CeBe@brsg-4dbbef81.pool.mediaWays.net) |
| 16:50.09 | Stefan27 | im using a work-around now - redirecting both B and C with manager redirect instead, and then bridge. gtg back tomorrow :) |
| 16:51.31 | WIMPy | So a redirect stops MOH? |
| 16:51.53 | Stefan27 | I only did one test so far, but yes |
| 16:51.59 | Stefan27 | will test again tomorrow |
| 16:52.20 | WIMPy | Well, I guess bridge should do it as well then. |
| 16:52.55 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
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| 17:36.45 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 18:02.38 | *** join/#asterisk dms1 (~dms1@65.207.151.254) |
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| 18:07.32 | *** join/#asterisk pjensen00 (~per@ip-64-21-247-189.far.ideaone.net) |
| 18:08.14 | pjensen00 | Hey, I'm using ARI. I can use the channel API just fine, but when I try to use the bridge API I get "Bridge not in Stasis application" |
| 18:08.26 | pjensen00 | I checked and I have the res_ari_bridges.so module loaded |
| 18:08.58 | pjensen00 | There weren't any stasis modules I could see that I would need to load, is this something I've misconfigured? |
| 18:09.43 | file | what are you trying to do? |
| 18:09.58 | pjensen00 | just do a GET bridges/ |
| 18:10.18 | pjensen00 | I currently have a bridged call up |
| 18:11.14 | file | in ARI? |
| 18:12.21 | pjensen00 | Oh wait, my bad. I'm trying to do a bridges/{bridgeID}removeChannel |
| 18:12.41 | file | yeah that would not work, need another / in there |
| 18:13.06 | pjensen00 | ..... boy do I have egg on my face. |
| 18:23.34 | *** join/#asterisk its_jeremy_ (~omghax@gateway/tor-sasl/itsjeremy/x-75806909) |
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| 18:26.05 | *** join/#asterisk hfp (~hfp@MTRLPQ0736W-LP130-02-1279513144.dsl.bell.ca) |
| 18:28.05 | hfp | Hey all, it seems my cell phone provider made some changes to their network and now I have no sound. My SIP client on my mobile phone says "Local NAT type detected: port restricted". I'm guessing the issue is because my Asterisk server is behind a NAT on my network and my cell phone is behind another NAT on my provider's network. Should I look into solving this on Asterisk's side or my SIP client's side? |
| 18:28.08 | pjensen00 | Is it no possible to manipulate a bridge with ARI that was not created by Stasis? |
| 18:29.04 | file | pjensen00, nope |
| 18:36.54 | pjensen00 | Interesting, but apparently channels not created through Stasis can be manipulated. |
| 18:38.05 | file | to a very very limited extent |
| 18:38.12 | file | WELL, created is wrong |
| 18:38.24 | file | channels not within Stasis can be manipulated to a very very limited extent |
| 18:41.35 | pjensen00 | How limited? As in a few of the channel commands from the Channel API will work (I got hangup to work) but others wont'? |
| 18:41.47 | file | I wouldn't expect any of the others to work |
| 18:42.43 | file | snoop will because of the way I wrote it, but there's nothing to say that both of them won't change in the future to only work for channels within Stasis |
| 18:43.04 | pjensen00 | Gotcha. Thanks for the insight. |
| 18:43.34 | pjensen00 | this channel has been very helpful for integrating with asterisk |
| 18:43.35 | file | fundamentally for most things you can't do two things at once on a channel unless you, well, don't |
| 18:44.14 | file | and two things can't safely own/control a channel for everything |
| 18:48.57 | *** join/#asterisk dhardison (uid20219@gateway/web/irccloud.com/x-dffrikkrmtdhrmva) |
| 18:54.23 | file | bonus points if someone can tell me how you can sorta do two things at once to a channel |
| 18:54.57 | file | Note: Bonus points are not redeemable for prizes. |
| 18:55.18 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 18:57.48 | *** join/#asterisk areski (~areski@80.174.128.112.dyn.user.ono.com) |
| 19:19.06 | pjensen00 | Two things at once, like playing music on an AGI controlled channel and sending DTMF tones to that channel with the AMI? |
| 19:24.08 | *** join/#asterisk lancelott64 (~lancelott@208.69.10.6) |
| 19:27.55 | keiths_ | anyone know why this wont work? |
| 19:28.12 | keiths_ | exten => 50,1,Set(mypeer=${BLINDTRANSFER:4:11}@) |
| 19:28.12 | keiths_ | exten => 50,n,Set(foo=${SIPPEER(${mypeer}:ip)}) |
| 19:30.41 | keiths_ | it returns |
| 19:31.03 | keiths_ | 500-Postfix@'' |
| 19:33.18 | [TK]D-Fender | Perhaps you should show us |
| 19:34.01 | *** join/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 19:36.30 | ChannelZ | is :ip even valid? : ? |
| 19:36.53 | keiths_ | exten => 50,1,Set(mypeer=${BLINDTRANSFER:4:11}@) |
| 19:36.53 | keiths_ | exten => 50,n,Set(foo=${SIPPEER(${mypeer}:ip)}) |
| 19:36.53 | keiths_ | exten => 50,n,Set(msg=${mypeer}${foo}) |
| 19:36.59 | WIMPy | no |
| 19:37.20 | WIMPy | And that @ probably doesn't do anythin good there, either. |
| 19:38.33 | ChannelZ | I think at minimum you need SIPPEER(${mypeer},ip) not :ip |
| 19:39.15 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 19:40.01 | [TK]D-Fender | keiths_: that is not looking at the CALL |
| 19:40.14 | [TK]D-Fender | keiths_: Because I'm not seeing how that breaks down in actual run-time. |
| 19:40.27 | keiths_ | Set("SIP/metaswitch-0008ddad", "foo=") |
| 19:40.56 | keiths_ | sec, ill use pastebin |
| 19:41.06 | ChannelZ | yay! |
| 19:41.45 | ChannelZ | though I think you just answered your own question: foo= |
| 19:42.21 | ChannelZ | You're not getting what you want from SIPPEER, probably for the reason I pointed out |
| 19:43.29 | *** join/#asterisk navaiso (0c0adbe2@gateway/web/freenode/ip.12.10.219.226) |
| 19:43.39 | keiths_ | ChannelZ: If i replace Set(foo=${SIPPEER(${mypeer}:ip)}) with Set(foo=${SIPPEER(200:ip)}) It works fine, the issue is calling the var in the function |
| 19:43.51 | *** part/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 19:44.11 | WIMPy | Where did you find that syntax? |
| 19:44.47 | WIMPy | And, BTW, I pointed out another reason why you probably don't get what you want. |
| 19:45.08 | keiths_ | http://www.voip-info.org/wiki/view/Asterisk+func+sippeer |
| 19:45.18 | keiths_ | removing the @ made no differance |
| 19:45.49 | WIMPy | Better use wiki.asterisk.org or do a 'core show function SIPPEER'. |
| 19:46.32 | [TK]D-Fender | keiths_: Where's the pastebin? You seem to be doing everything except showing us the actual call |
| 19:49.30 | keiths_ | http://pastebin.com/CV6Zccrv |
| 19:50.51 | file | pjensen00, I will accept that! |
| 19:51.34 | keiths_ | brb. |
| 19:53.17 | [TK]D-Fender | keiths_: Where do we see you showing us the peer? |
| 19:54.04 | WIMPy | And why do the parts not fit together? |
| 19:58.37 | *** join/#asterisk dms1 (~dms1@65.207.151.254) |
| 20:02.04 | *** join/#asterisk dms1 (~dms1@65.207.151.254) |
| 20:04.47 | keiths_ | [TK]D-Fender: not sure what you want to see? WIMPy: what does not fit? |
| 20:05.05 | [TK]D-Fender | keiths_: Where do we see that that is a valid peer? |
| 20:05.44 | WIMPy | Prio 1 of your dialplan ends with an @. I don't see that in the log. Yet, two lines later it magically appears. |
| 20:07.05 | keiths_ | ill take the filtering out of it, show it RAW |
| 20:08.39 | WIMPy | And as [TK]D-Fender said, we don't knnow if "250-CUSTOMER@" is an existing peer. |
| 20:09.43 | keiths_ | i wasted all your time :( |
| 20:10.06 | keiths_ | I guess I did not remove the @ when I said I did |
| 20:11.53 | keiths_ | added the @ to exten => 50,n,Set(msg=${mypeer}@${foo}) and the whole config is working fine now. sorry guys |
| 20:22.05 | *** join/#asterisk twanny796 (~Twanny@c103-220.i07-26.onvol.net) |
| 20:22.27 | twanny796 | which linux distribution do the developers use? |
| 20:22.38 | [TK]D-Fender | CentOS primarily |
| 20:22.55 | twanny796 | ok |
| 20:23.05 | Qwell | [TK]D-Fender: disagree. Ubuntu, for Asterisk. |
| 20:23.12 | drmessano | They're canadian |
| 20:23.21 | drmessano | Canadians love CentOS |
| 20:23.28 | drmessano | It's like the Linux of Canada |
| 20:23.34 | drmessano | Official* |
| 20:24.11 | [TK]D-Fender | Unlike the US system, we negotiated a cheaper version of the brand-name the the US uses ;) |
| 20:24.42 | drmessano | Yes, the more expensive Brown Pants Linux |
| 20:25.22 | twanny796 | does it really matter which distribution is used ? |
| 20:25.30 | Qwell | twanny796: Not in the least. |
| 20:25.40 | Qwell | Except Gentoo, because they're dumb. |
| 20:25.46 | drmessano | Not unless you want to fight about it |
| 20:25.47 | *** join/#asterisk gugaua (~gugaua@unaffiliated/gugaua) |
| 20:25.53 | drmessano | Which is perfectly reasonable |
| 20:26.09 | twanny796 | Qwell, ok I'll use OpenSUSE ? |
| 20:26.20 | drmessano | <I don't know which distro you are using, but whatever it is sucks and mine is better even if it's the same/> |
| 20:26.30 | Qwell | oh, right, OpenSUSE too. |
| 20:27.25 | Qwell | If you want SIP to work, don't use Gentoo. If you want DAHDI to work, don't use OpenSUSE. |
| 20:27.32 | twanny796 | Qwell, I thinks CentOS is better, but I use OpenSUSE at work |
| 20:27.52 | twanny796 | Qwell, I want DAHDI |
| 20:27.54 | drmessano | If you want upgrading to work, don't use CentOS.. oh wait |
| 20:28.37 | keiths_ | use Debian :) |
| 20:28.38 | twanny796 | I want DAHDI so I cannot use OpenSUSE? |
| 20:28.58 | Qwell | twanny796: I wouldn't say "cannot". I'd use a much less threatening "should not". |
| 20:29.31 | drmessano | "It would be bad" |
| 20:29.36 | keiths_ | drmessano: My linux can beat up you linux, any day of the week |
| 20:29.40 | Qwell | They don't know how to patch a kernel properly, is all. |
| 20:29.44 | twanny796 | trixbox is on CentOS, yes? |
| 20:29.58 | twanny796 | and FreePBX |
| 20:30.15 | Qwell | All of the distros use CentOS or derivatives. |
| 20:30.21 | drmessano | Try to imagine all life as you know it stopping instantaneously and every molecule in your body exploding at the speed of light. |
| 20:30.30 | WIMPy | wouldn't use any distro with mangled kernels. They often make things worse rahter than better. Still. |
| 20:30.48 | keiths_ | trixbox is not something you should use unless the box is behind your LAN with heavy firewall rules. So many back doors |
| 20:30.49 | Qwell | WIMPy: So, LFS for you then? |
| 20:30.57 | drmessano | trixbox is extinct |
| 20:31.01 | Qwell | keiths_: You could have stopped that sentence much shorter... |
| 20:31.19 | WIMPy | Qwell: Should work. |
| 20:31.32 | drmessano | trixbox is spyware running Asterisk that masquerades as a Fisher Price PBX |
| 20:31.33 | Qwell | WIMPy: That's about the only thing that wouldn't have some sort of mangling. |
| 20:31.36 | twanny796 | ok, will install CentOS in the weekend!!! |
| 20:31.59 | keiths_ | Qwell:punctuation? aint nobody got time for that |
| 20:32.35 | WIMPy | Qwell: Slackwar doesn't either. And I'm sure there are lots of other that don't, eiter. Just stay away from the mainstream stuff. |
| 20:32.46 | drmessano | http://ecx.images-amazon.com/images/I/417YMSTP13L.jpg <-- trixbox endpoint |
| 20:33.07 | Qwell | WIMPy: They'll have to patch it once they finally add PAM support. |
| 20:33.41 | Qwell | also didn't Patrick like die or something? |
| 20:34.27 | WIMPy | NFI. |
| 20:34.43 | twanny796 | Qwell, Slackware was my first linux ;) |
| 20:34.57 | twanny796 | way back in 1997 |
| 20:56.19 | Nugget | I use an old slackware AP Disk Set for my computer desk. |
| 20:57.56 | file | Nugget, CHOOSE FILE |
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| 21:14.15 | drmessano | ~happyclownpbx |
| 21:14.15 | infobot | currently in closed beta, approaching 12GB in size, uses Asterisk for its core, it pwns, eats your children (seriously), and is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone |
| 21:15.50 | tm1000 | ~piaf |
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| 21:38.36 | pjensen00 | Trying to connect to websocket, getting "Origin header 'http://localhost:8088' does not match an allowed origin." I have the rest of the output here. http://pastebin.com/CqgBsugQ |
| 21:39.25 | navaismo | your oring has an * |
| 21:39.32 | navaismo | or only an IP |
| 21:41.32 | pjensen00 | Hrm, I'm supplying essentially the same connection string as was listed in the "getting started with ARI". I don't see any * or a standalone IP |
| 21:44.04 | mjordan | pjensen00: add allowed_origins=* to your ari.conf |
| 21:44.17 | mjordan | pjensen00: I might need to update the example... :-) |
| 21:45.44 | pjensen00 | oh thanks! |
| 21:46.14 | *** join/#asterisk mavrix (~kvirc@49.204.108.203) |
| 21:47.41 | mavrix | i'm having troubles comparing a var against the star key |
| 21:47.47 | mavrix | this |
| 21:47.55 | mavrix | GotoIf($[ ${secret2} = "*" ]?read1) |
| 21:48.15 | mavrix | returns an error |
| 21:49.08 | mavrix | i tried with or without quotes but always getting that ast_yyerror. how to do that? |
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| 22:03.02 | newtonr | mavrix, pastebin your full error with debug. That expression works fine for me. |
| 22:03.08 | newtonr | mavrix, what version of Asterisk are you using? |
| 22:03.49 | mavrix | asterisk-12.1.0-rc1 |
| 22:04.10 | newtonr | mavrix, in your debug, please print the value of your secret2 var before that GotoIf with a NoOp or Verbose call |
| 22:04.12 | mjordan | pjensen00: did that fix your issue? |
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| 22:04.41 | mavrix | ok |
| 22:05.29 | pjensen00 | mjordan: yes that got rid of that error, but I'm getting immediatley disconnected |
| 22:05.44 | pjensen00 | but I have to troubleshoot that to make sure it's not my library I'm using with perl |
| 22:06.03 | newtonr | mavrix, https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information <-- how to get the full log with DEBUG |
| 22:07.45 | mavrix | newtonr: the value of secret2 is user input from Read() app |
| 22:07.53 | mavrix | here is log http://pastebin.com/Rfnr3sPs |
| 22:13.12 | newtonr | mavrix, and pastebin your actual diaplan (that is copy paste the dialplan you are using, instead of typing it out here) |
| 22:13.24 | mavrix | ok |
| 22:17.43 | mavrix | ok here is the chunk http://pastebin.com/3A8EvjNs |
| 22:18.29 | mavrix | it waits for a passowrd (123) while playing rings on early media |
| 22:19.06 | mavrix | it checks digits one by one |
| 22:21.59 | newtonr | Hmm, try setting secret2 to "*" with Set right before the GotoIf as a test to see what happens |
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| 22:22.54 | mjordan | interesting, perl |
| 22:30.03 | mavrix | not working, getting syntax erro again.. |
| 22:31.09 | newtonr | oh! |
| 22:31.44 | newtonr | i think i might know what it is, one second |
| 22:32.25 | mavrix | maybe there is a different way to represent the star key. not sure.. |
| 22:33.18 | newtonr | its actually the line below that is failing |
| 22:33.23 | newtonr | i glazed over your error |
| 22:33.39 | mavrix | ? |
| 22:38.12 | newtonr | see lines 7 and 8 on your pastebin |
| 22:38.27 | newtonr | Though.. I still can't reproduce it |
| 22:38.56 | newtonr | lines 7 and 8 on your debug pastebin http://pastebin.com/Rfnr3sPs |
| 22:40.01 | mavrix | thanks for checking. is there a way around? |
| 22:40.06 | newtonr | Let me try actually using the Read app, as I'm not doing that at the moment |
| 22:40.16 | newtonr | mavrix, this feels buggy, unless i'm totally missing something |
| 22:40.41 | newtonr | mavrix, you said it still errors out even when set with the Set app right? |
| 22:40.51 | mavrix | yes |
| 22:41.07 | mavrix | does it work on your system? |
| 22:41.27 | newtonr | yup, and i installed 12.1.0-rc1 to test |
| 22:42.03 | mavrix | oh really thanks for efforts |
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| 22:43.22 | mavrix | I'm on a DEBIAN VPS using ssh |
| 22:44.07 | mavrix | but it |
| 22:44.25 | mavrix | but it's probably has nothing to do with the issue |
| 22:45.23 | newtonr | can you get the debug again, but make sure you have the "DEBUG" logger channel on and turned up to 5 |
| 22:45.49 | newtonr | "logger show channels" to see what logger channels are going where, see logger.conf to edit it, and "core set debug 5" to turn it up from the CLI |
| 22:46.20 | mavrix | ok |
| 22:46.23 | newtonr | if done correctly you should see lines like [Aug 28 17:44:23] DEBUG[27280][C-00000005]: pbx.c:3675 ast_str_retrieve_variable: Result of 'pass' is '"123"' |
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| 22:49.05 | mavrix | moment |
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| 22:53.58 | mavrix | ok it shows WARNING[4328][C-00000013]: |
| 22:54.20 | newtonr | nope it should show DEBUG messages too |
| 22:57.03 | mavrix | ok moment |
| 22:59.39 | newtonr | mavrix, https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 23:00.08 | mavrix | one second' |
| 23:01.58 | mavrix | ok i got the file saved. it's full of Debug lines |
| 23:03.34 | mavrix | here is that part http://pastebin.com/hmjvWwgF |
| 23:05.13 | newtonr | mavrix, I talked with putnopvut - try quoting your references to secret1, secret2, secret3 |
| 23:05.32 | newtonr | mavrix, it looks like Read is putting the * in the variable, but not as a string |
| 23:05.54 | mavrix | ok moment |
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| 23:09.35 | newtonr | mavrix, that should work for you. That means, instead of ${secret2}, do "${secret2}" |
| 23:09.43 | newtonr | mavrix, I've got to go now. Later! |
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| 23:18.43 | mavrix | just had a power cut.. |
| 23:18.51 | mavrix | are you there? |
| 23:22.52 | mavrix | just to l let you know your advise worked for me |
| 23:23.11 | mavrix | the quotes removed that error.. |
| 23:23.20 | mavrix | and thank you very much :) |
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