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| 02:00.05 | ipengineer | Does anyone know what would cause a flood of: chan_sip.c:13640 in handle_response: Remote host can't match request NOTIFY to call '776843471-5060-81@BJC.BGI.B.BCJ'. Giving up. |
| 02:01.15 | ipengineer | For whatever reason this just started the last couple of days and has degraded performance greatly. Our receive queue from netstat is way high |
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| 06:00.11 | phix | Afternoon gang |
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| 06:01.43 | ChannelZ | Hey team! |
| 06:05.25 | phix | :) |
| 06:05.30 | phix | <3 |
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| 06:25.03 | *** join/#asterisk rock1979 (~rock@46.252.100.15) |
| 06:25.38 | rock1979 | Hello i am experiencing problems with ASterisk 11.5.0 with T38 faxing |
| 06:26.29 | rock1979 | i am working on the issue for the last 10 days |
| 06:26.41 | rock1979 | but cant find a solution |
| 06:27.04 | shido6 | t.38 enabled provider? |
| 06:27.18 | rock1979 | i am the provider |
| 06:27.23 | rock1979 | :) |
| 06:27.34 | rock1979 | the server is used with sangoma cards |
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| 06:27.41 | shido6 | what devices are in the call? |
| 06:27.54 | rock1979 | the calls are from sip to dahdi |
| 06:28.06 | shido6 | ata? softphone? |
| 06:28.18 | rock1979 | ata spa112 |
| 06:28.25 | rock1979 | it has t38 cababilities |
| 06:28.42 | rock1979 | so the calls are coming to my server with sip |
| 06:28.50 | rock1979 | and i am sending to telco with dahdi |
| 06:28.51 | coppice | is the SPA112 firmware up to date? |
| 06:29.31 | rock1979 | yes, |
| 06:29.47 | rock1979 | actually the wholesale voice company is doing the test |
| 06:30.01 | rock1979 | and we are tring to make interconnection |
| 06:30.10 | rock1979 | but since faxing is not successfull |
| 06:30.24 | rock1979 | we are not able to finalize the test |
| 06:30.33 | shido6 | ahhh |
| 06:30.36 | rock1979 | and continue with termination |
| 06:30.53 | rock1979 | so ata sends me the call |
| 06:31.04 | rock1979 | and i send the call to telco with dahdi |
| 06:31.12 | rock1979 | at the begging everyrhing is fine |
| 06:31.13 | shido6 | is there a Multifunction printer connected to the ata? |
| 06:31.49 | rock1979 | At the beginning of the call everything is fine ( i mean codec negotiation) . Then the remote sip side detects the fax tone that remote end sends and Then the sip remote side sends the t38 invite and asterisk sends SIP 200 OK message with G711 and G729 codec in SDP. |
| 06:32.13 | rock1979 | My customer is saying that if i am sending SIP 200 OK to his T38 invite .in SDP of SIP 200 OK there should be only T38. |
| 06:32.33 | rock1979 | Also disabling t38 with t38pt_udptl=no didnt change anything. Still asterisk is sending SIP 200 OK message with G711 and G729 codec in SDP as reply to T38 invite. |
| 06:32.37 | shido6 | is the wholesale carrier using SIP? |
| 06:32.38 | coppice | Do you have T.38 gateway mode enabled in the config files? |
| 06:33.04 | rock1979 | Setting faxopt(gateway)=yes and t38pt_udptl = yes,redundancy,maxdatagram=400 didnt work. |
| 06:33.23 | shido6 | <you>âSIPââ> WHOLESALEâSIPâ> LCR of OTHER PROVIDERs where 2 out of 7 dont offer t.38 ? |
| 06:33.31 | shido6 | does this look familiar? |
| 06:34.17 | rock1979 | shido i am the guy that providing the service |
| 06:34.18 | shido6 | pastebin.ca a sip trace or upload a pcap :) |
| 06:35.01 | rock1979 | so calls i coming to me from sip and i am sending to telco with dahi |
| 06:35.16 | rock1979 | i mean dahdi |
| 06:35.23 | coppice | is T.38 faxing to the asterisk box itself working OK? |
| 06:36.06 | rock1979 | coppice what did you mean ?? |
| 06:36.28 | coppice | faxing from the ATA to a file on the asterisk box |
| 06:37.29 | rock1979 | i didnt installed anything on asterisk box, it is standart Asterisk 11.5.0 (i mean noting like Hylafax,spandsp etc) |
| 06:37.53 | coppice | how is it supposed to be a T.38 gateway without spandsp? |
| 06:38.40 | rock1979 | did i understood incorrect, as i understood after asterisk 10 , you dont need to install spandsp |
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| 06:39.42 | rock1979 | Sip+T38--->Asterisk----dahdi----->Telco |
| 06:40.29 | rock1979 | i found this page in web http://www.voip-info.org/wiki/view/Asterisk+T.38+Gateway |
| 06:41.00 | rock1979 | it is exactly what i am tring to do |
| 06:41.42 | rock1979 | i am connected with TELCO with E1 over ss7 |
| 06:42.05 | coppice | that means asterisk 10 can use a T.38 gateway engine. You still need one installed |
| 06:43.29 | rock1979 | coppice from asterisk box to telco with dahdi , does this means that i am sending fax from * to telco with T30 right ? |
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| 06:44.03 | coppice | yes. |
| 06:44.49 | rock1979 | so how can i prevent asterisk to send sip200 ok to t38 invites |
| 06:44.55 | coppice | there is no other way to send a FAX over the PSTN (unless you are that rare group who can do group 4 FAX over ISDN) |
| 06:45.49 | rock1979 | i set t38pt_udptl=no in sip.conf for that peer |
| 06:45.55 | coppice | why do you not want it to say OK? You need to make it say OK with T.38 |
| 06:47.00 | rock1979 | yes, you are right , but currently t38 invite is hittting to * box and asterisk is sending OK with G711 |
| 06:47.32 | coppice | unless you have spandsp installed it can't do anything else |
| 06:48.07 | rock1979 | so the easiest solution that i found was when a t38 invite come to asterisk if asterisk send sip 488 then the sip remote side will try t30 fax |
| 06:48.28 | rock1979 | then it will be a t30 faxing |
| 06:48.36 | coppice | is that what you want it to do? |
| 06:49.00 | rock1979 | that is why i set t38pt_udptl=no |
| 06:49.13 | rock1979 | in sip.conf of peer |
| 06:49.33 | coppice | I thought you were trying to use T.38 between the ATA and the asterisk box |
| 06:49.52 | rock1979 | it is the best solution to support t38 |
| 06:50.17 | rock1979 | but just regretting t38 and fallthrout t30 would also solve my problem |
| 06:50.23 | coppice | the ATAs are pretty flaky trying to do FAX by audio |
| 06:51.13 | rock1979 | i thought that i need spandsp if i need to send fax inside asterisk from file |
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| 06:51.28 | rock1979 | or receive fax to asterisk to a file |
| 06:52.04 | coppice | you can do that with either spandsp or the digium FAX module. If you want a gateway only spandsp offers that |
| 06:53.39 | rock1979 | what i want is to be able to receive fax from sip and send to telco with dahdi |
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| 06:54.15 | rock1979 | on my case i have only one way interconnection( i mean from sip to dahdi , not from telco to sip) |
| 06:54.54 | coppice | either you just use G.711 end to end, which can be somewhat flaky at the ATA end, or you install spandsp and use T.38 gateway operation |
| 06:55.49 | rock1979 | is it possbile to send SIP 488 if i receive T38 invite |
| 06:57.16 | rock1979 | as far as i read on web if t38 is disabled in * , then sip 488 should be sent back |
| 06:58.23 | rock1979 | that is why i set t38pt_udptl=no in sip.conf |
| 06:58.58 | rock1979 | but still instead of sending SIP 488 , asterisk is sending SIP 200 OK with G711 in sdp |
| 06:59.40 | coppice | sending 488 will generally kill the call |
| 07:01.02 | rock1979 | no, i talked with the sip side , they are saying that if i send them back sip 488 their softswitch will fallback to T30 |
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| 07:02.38 | rock1979 | so if i send SIP 488 , they will send a new invite with G711 in SDP |
| 07:03.07 | coppice | could be. it varies between systems. isn't standardisation wonderful? |
| 07:03.25 | rock1979 | as far as i read if t38 is disabled in peer configuration ,then asterisk should send SIP488 |
| 07:03.43 | rock1979 | but it is not in my case |
| 07:04.07 | coppice | that may have been changed because a lot of things kill the call after a 488 |
| 07:06.28 | rock1979 | how can i be sure if that behaviour is changed is asterisk 11.5.0 or not |
| 07:06.50 | rock1979 | i searched in web for "t38 sip 488" |
| 07:07.27 | rock1979 | and what i found is if t38 is disabled than * will send SIP 488 |
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| 07:15.43 | rock1979 | can i solve this problem with digium FAX module |
| 07:17.29 | rock1979 | did you mean "Fax for Asterisk" when you said digium fax module |
| 07:18.25 | coppice | yes. FAX for Asterisk cannot work as a gateway |
| 07:21.37 | rock1979 | so only solution on my case is spandsp |
| 07:21.56 | coppice | you make that sound like a desperate last resort |
| 07:22.07 | rock1979 | since i am not able to reply sip 488 to t38 invites |
| 07:22.47 | rock1979 | yes the server is in production and i didnt installed spandsp before :) |
| 07:22.54 | *** join/#asterisk niekniek (~niek@77-172-219-210.ip.telfort.nl) |
| 07:23.56 | coppice | if you are worried about new things, why would FAX for Asterisk be a better choice? its a lot flakier than spandsp |
| 07:25.03 | rock1979 | no i didnt mean it is a better choice |
| 07:27.24 | rock1979 | so if asterisk still needs spandsp , what is the fax gateway support that come up in asterisk 10 |
| 07:28.12 | coppice | asterisk 10 added the code needed to make use of the T.38 gateway engine in spandsp |
| 07:29.01 | rock1979 | coppice can i add you on skype ? |
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| 07:32.54 | rock1979 | it seems that at the end i will need profferssional support to solve this problem , either from freelancers of from the forum of asterisk. |
| 07:33.12 | rock1979 | so this will be a project with bughet |
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| 07:36.28 | rock1979 | are you interested with this ? |
| 07:36.34 | rock1979 | coppice ? |
| 07:42.32 | rock1979 | coppice are you there ? |
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| 07:56.43 | rock1979 | does anyway have experience with t38 faxing |
| 07:56.54 | rock1979 | does anyone have experience with t38 faxing |
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| 08:03.43 | paolo_ | rock1979: Yes I do. And I am sure others have experience, too =) |
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| 09:15.47 | rock1979 | coppice are you there ? |
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| 09:40.20 | rock1979 | Hello we are looking for proffesional support to add t38 gateway capability to asterisk 11+libss7 server |
| 09:41.19 | Chainsaw | In what part of the world please? There are several consultants here but if you want on-site support it helps if they are local. |
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| 09:44.35 | rock1979 | Not necesssarily on-site support |
| 09:44.55 | rock1979 | remote support will be fine |
| 09:47.29 | Chainsaw | Asterisk 11 has native T38 gateway support, so that is good news. |
| 09:47.38 | Chainsaw | I will not DCC chat with you, please keep the discussion central. |
| 09:47.59 | rock1979 | yes sure :) |
| 09:48.42 | rock1979 | it is an asterisk server that we use for terminating calls to telco |
| 09:49.27 | rock1979 | calls are coming form owr customers from sip |
| 09:49.38 | rock1979 | and we are sending calls to telco with dadhi |
| 09:50.01 | rock1979 | it is a one way interconnection from asterisk to telco |
| 09:50.17 | rock1979 | so no calls are coming from telco to asterisk |
| 09:51.13 | rock1979 | we want to add t38 faxing capability to asterisk |
| 09:51.24 | *** join/#asterisk af_ (~af@93-43-38-121.ip90.fastwebnet.it) |
| 09:51.50 | rock1979 | anyone interested to provide professional supprt for this are welcome |
| 09:52.11 | rock1979 | anyone interested to provide professional support for this are welcome |
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| 10:07.32 | rock1979 | any consultant interesed with this ? |
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| 11:38.18 | matthew-moretalk | Hi All asterisk xmpp is driving me a little mad. in the output it just says - Waiting to request TLS?? |
| 11:38.29 | matthew-moretalk | any ideas what im doing wrong? |
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| 13:50.31 | rock1979 | Hello we are looking for proffesional support to add t38 gateway capability to asterisk 11+libss7 server |
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| 13:56.30 | WIMPy | feels bad being reminded of fax |
| 13:56.34 | [TK]D-Fender | rock1979: https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway |
| 13:56.56 | [TK]D-Fender | rock1979: Don't know why you'd need "professional support" for a basic option in the dialplan plus a few in your peer |
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| 14:01.41 | jameswf | was just reading about SS7 and privacy issues... neat stuff |
| 14:02.23 | WIMPy | privacy issues? |
| 14:03.07 | jameswf | WIMPy: it seems with SS7 you can locate any cell phone user in the world. Get all CSI.... |
| 14:03.45 | WIMPy | If you're allowed to do so. |
| 14:03.47 | jameswf | SS& security was sort of an after thought |
| 14:04.14 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 14:04.18 | WIMPy | It's not intended for customers. |
| 14:04.47 | jameswf | WIMPy: scary thing is customers aren't the ones (ab)using it |
| 14:05.25 | *** join/#asterisk shido6 (~shido6@98.234.178.27) |
| 14:05.36 | jameswf | Governments (likely all of them) are using SS7 for warrantless locating |
| 14:06.16 | WIMPy | They use everything they can anyway. Legal or not. |
| 14:06.24 | shido6 | like irc logs? |
| 14:06.34 | jameswf | was never here |
| 14:06.50 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 14:07.08 | jameswf | ^^ and you cant just put stuff on the internet that's not true.... |
| 14:10.26 | coppice | [TK]D-Fender: that page doesn't seem to mention that spandsp needs to be installed for the FAX gateway to work. This is what confused rock1979 |
| 14:10.58 | rock1979 | Fender , i visited all the web pages related with Asterisk T38, and did all the neccessary thinks |
| 14:11.15 | [TK]D-Fender | coppice: I'd wonder if that function would compile without it... |
| 14:11.31 | [TK]D-Fender | coppice: Which if it wouldn't should lead you back to menuconfig and a glaring dependency... |
| 14:12.00 | [TK]D-Fender | coppice: ... but that's probably just my silly idealism ;) |
| 14:12.22 | coppice | [TK]D-Fender: it would be better if the requirements were pointed out, though |
| 14:13.00 | [TK]D-Fender | coppice: If I'm right.. they are. |
| 14:13.27 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:13.56 | rock1979 | <PROTECTED> |
| 14:14.21 | WIMPy | Fax is rather scary. Too many options with very different reports on success. And some very strict dependencies requiring very specific versions. |
| 14:15.01 | [TK]D-Fender | rock1979: So show us the failure. |
| 14:15.09 | shido6 | :) |
| 14:15.58 | rock1979 | asterisk is sending sip 200 ok with G711 in sdp to t38 invite |
| 14:18.29 | [TK]D-Fender | by "show us the failure" I did not mean "give us a truncated 1-line attempt at a summary". |
| 14:18.38 | [TK]D-Fender | If you want an autopsy, give us a body. |
| 14:19.39 | rock1979 | ok i uploading pcap to pastebin.ca |
| 14:19.44 | rock1979 | ok i am uploading pcap to pastebin.ca |
| 14:20.16 | rock1979 | will it be fine ? |
| 14:20.27 | [TK]D-Fender | * CLI w/ verbose 10, SIP debug. |
| 14:21.06 | rock1979 | the server is in production with 120 calls on it |
| 14:21.11 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 14:21.38 | rock1979 | how can i get cli with verbosity level 10 |
| 14:21.39 | rock1979 | :) |
| 14:21.48 | shido6 | http://face2phase.com/sipdebugwireshark |
| 14:22.26 | shido6 | kewl pastebin pcap⦠sorry |
| 14:22.53 | [TK]D-Fender | "core set verbose 10" |
| 14:23.57 | rock1979 | shido6 , www.pastebin.com right ? |
| 14:24.13 | shido6 | do you have chrome? |
| 14:24.52 | rock1979 | yes |
| 14:24.55 | shido6 | capture the pcap with tcpdump and use https://www.sharefest.me |
| 14:25.38 | shido6 | paste the link when done |
| 14:25.55 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:25.55 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:26.28 | rock1979 | https://www.sharefest.me/b43b5244966522dc76acd1732b2cb9c9 |
| 14:27.50 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:28.06 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:30.15 | rock1979 | it doesnt matter whether t38pt_udptl=no or yes , asterisk sends sip 200 ok with sdp codes g711... to t38 invite |
| 14:31.23 | rock1979 | i am checking sip t38 flow diagrams and asterisk should send sip 200 ok with sdp t38 if faxing is t38 otherwise it should send sip 488 |
| 14:31.38 | coppice | t38pt stands for T.38 passthrough. If you are going from SIP to the PSTN passthrough is not possible, so that parameter does nothing |
| 14:32.10 | shido6 | passthrough? |
| 14:33.42 | shido6 | no t38 media... |
| 14:34.42 | shido6 | is this a netnet? |
| 14:34.51 | rock1979 | sorry ? |
| 14:35.07 | *** part/#asterisk c|oneman (cloneman@1337.montrealdark.com) |
| 14:35.53 | rock1979 | what is netnet ? |
| 14:37.03 | shido6 | I saw acmepack and thought acme packet net net |
| 14:37.47 | rock1979 | it is the softswitch of my customer that is testing the T38 fax |
| 14:39.27 | *** join/#asterisk wolrah_ (~wolrah@24.239.210.140) |
| 14:41.03 | shido6 | does their router have ALG? |
| 14:41.08 | shido6 | enabled. |
| 14:41.18 | shido6 | ? |
| 14:43.10 | rock1979 | as far as i know it is not alg enabled |
| 14:44.01 | rock1979 | what he told me is that their fax is working fine with other sip accounts |
| 14:45.34 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 14:45.41 | shido6 | test that through your system to theirs. and tcpdump it |
| 14:45.50 | shido6 | âgood callâ bad call comparison |
| 14:46.55 | shido6 | echo correction is enabled.. |
| 14:47.01 | shido6 | can u disable ec ? |
| 14:48.01 | rock1979 | is it enabled on my side or on the remote side ? |
| 14:48.14 | shido6 | looks like remote |
| 14:48.58 | rock1979 | where did you find that echo is enabled on the remote side ? |
| 14:49.19 | *** join/#asterisk random2 (~deepak@122.179.39.63) |
| 14:51.17 | random2 | does ${RECORDED_FILE} have complete path or is it just filename? |
| 14:52.19 | shido6 | no 6. |
| 14:52.26 | shido6 | media attribute |
| 14:58.48 | rock1979 | i am looking to the 6. record in sip trace |
| 14:59.06 | shido6 | nothing blocking 5069? :) |
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| 15:04.46 | rock1979 | i am only listeing on 5060 |
| 15:04.54 | rock1979 | *listening |
| 15:10.31 | rock1979 | shido6 is it fine that asterisk is sending sip200 ok with sdp g711 as reply to t38 |
| 15:11.36 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 15:15.43 | rock1979 | any one have idea how i will manage to send fax from sip to pstn |
| 15:16.02 | rock1979 | without T38 everything is fine |
| 15:16.18 | rock1979 | i mean with G711 codecs |
| 15:16.18 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 15:17.36 | rock1979 | coppice are you there ? |
| 15:18.16 | rock1979 | fender did you have change to look to the sip trace |
| 15:18.47 | [TK]D-Fender | <PROTECTED> |
| 15:18.48 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:1dd0:21b:63ff:fe31:8426) |
| 15:19.56 | rock1979 | you meant core set verbose 10 ? right ? |
| 15:21.31 | rock1979 | server is in production with many calls on it |
| 15:21.40 | *** join/#asterisk u0m3_ (~u0m3@92.80.109.44) |
| 15:21.55 | rock1979 | the log will be quite confusing |
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| 15:26.55 | *** join/#asterisk u0m3 (~u0m3@92.80.109.44) |
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| 15:59.36 | babak | Hi, Is there any document about how I can read a E1 timeslot data from DAHDI driver ? |
| 16:00.21 | WIMPy | There was a patch to allow you to write pcap files somewhere. |
| 16:01.23 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 16:01.54 | babak | is there any documents about DAHDI APIs ? |
| 16:04.01 | WIMPy | I don't know any. But the osmocom guys made their stuff to work with dahdi. So either they fond some docs or they reversed it. |
| 16:04.56 | newtonr | babak, No, there isn't. There is source code! |
| 16:05.54 | babak | Is there a easy sample source code ? ;) |
| 16:06.24 | drmessano | WakeOnLan is WakeOnStupidFace |
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| 17:42.23 | Katty | looks in |
| 17:44.11 | WIMPy | Come in and you can look out. |
| 17:44.22 | [TK]D-Fender | looks on |
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| 18:04.21 | Katty | [TK]D-Fender: rock fest sounds toasty |
| 18:04.29 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 18:04.49 | Katty | [TK]D-Fender: wear sunblock |
| 18:04.53 | [TK]D-Fender | Katty: Yeah I used a "select all" tool and didn't hand-pick. So .. you comin'? ;) |
| 18:05.13 | Katty | [TK]D-Fender: no i'm actually leaving friday for springfield |
| 18:05.15 | [TK]D-Fender | Katty: I'm actually just working sound for the show but figured I'd help with exposure for them. |
| 18:05.25 | Katty | [TK]D-Fender: i have a date with a huge arcade. |
| 18:05.52 | [TK]D-Fender | Katty: Pretty much every state including where I live has a Springfield. That's why it was chose for The Simpsons |
| 18:05.54 | Katty | [TK]D-Fender: that's nice of you. |
| 18:06.11 | Katty | [TK]D-Fender: the one here in missouri. nothing fancy. but they do have a lovely cupcake shop |
| 18:08.43 | [TK]D-Fender | mmmmm |
| 18:12.11 | file | moo |
| 18:12.17 | *** join/#asterisk shido6 (~shido6@98.234.178.27) |
| 18:45.51 | hexanol | someone knows how much time it usually takes to get a response from security@asterisk.org ? |
| 18:46.21 | hexanol | I've sent an email last week ago |
| 18:46.24 | hexanol | still no response |
| 18:46.27 | hexanol | so I'm wondering |
| 18:47.34 | Qwell | nudges newtonr |
| 18:48.54 | file | hexanol, what was the email address that it was sent from? |
| 18:53.50 | *** join/#asterisk Nugget (nugget@rennsport.macnugget.org) |
| 18:54.32 | newtonr | Qwell, sorry i was at the grocery store! File is on the case. |
| 18:54.42 | Qwell | newtonr: slacker! |
| 18:54.54 | file | I wouldn't say I'm on the case :P |
| 18:55.29 | file | hexanol, it wouldn't hurt to resend it too |
| 18:55.49 | *** join/#asterisk gugaua (~gugaua@unaffiliated/gugaua) |
| 18:56.01 | hexanol | well I guess I could |
| 18:56.06 | hexanol | do you want to be in copy ? ;) |
| 18:56.17 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:56.27 | newtonr | file, sure you are on the case. look at you go. |
| 18:56.51 | gugaua | Hello I want to update Asterisknow modules but for 1 module I get an error |
| 18:56.51 | gugaua | EndPoint Manager cannot be upgraded: |
| 18:56.52 | gugaua | <PROTECTED> |
| 18:56.52 | gugaua | Please try again after the dependencies have been installed. |
| 18:57.02 | Qwell | gugaua: Try #freepbx |
| 18:57.04 | file | hexanol, actually it looks like it was found! |
| 18:57.14 | gugaua | @Qwell: okay thanks |
| 18:57.18 | hexanol | hooray |
| 19:01.03 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 19:02.34 | file | hexanol, it shall be handled at a time in the future as we do not have time travel |
| 19:03.48 | hexanol | file: I'm not sure I understand (I'm not a native english speaker by the way) |
| 19:04.12 | file | hexanol, we'll respond soon |
| 19:04.13 | Qwell | hexanol: translated from file - "Thank you for your report. We will look into the issue as time permits." |
| 19:04.15 | hexanol | is it considered a security vulnerability or not ? or people will look at it later ? |
| 19:04.19 | hexanol | ok, thanks |
| 19:04.27 | file | Qwell, you speak file? |
| 19:04.36 | Qwell | file: as well as one can ever hope to |
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| 19:40.47 | TazzNZ | ~~morning all |
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| 19:48.06 | ipengineer | When using the System() app if trying to move a file and I dont have permissions the operation hangs and ties the channel up. Would trySystem() be a better fit here so the dialplan progresses in the event this happens |
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| 19:48.29 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 19:50.59 | [TK]D-Fender | ipengineer: background it. |
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| 20:11.23 | ipengineer | [TK]D-Fender: How do you do that. I am not seeing anything online for backgrounding a task |
| 20:12.29 | [TK]D-Fender | ipengineer: call a shell script... have that call a task in the background |
| 20:13.36 | ipengineer | [TK]D-Fender: Ok makes sense |
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| 20:24.48 | [TK]D-Fender | heads off to the gym... |
| 20:24.51 | *** join/#asterisk ra21vi (~xavbook@125.21.241.186) |
| 20:25.25 | ra21vi | Hi. I am using asterisk 11 (elastix), and my soft sip client gets hangup at exactly 6 sec |
| 20:25.42 | ra21vi | what can be the reason? :) |
| 20:26.16 | ra21vi | I searched google, didn't find why it drops call at exactly 6 secs. |
| 20:26.45 | ra21vi | If I call though Cisco hard phone, call goes well |
| 20:26.48 | *** join/#asterisk digiv (~digiv@drupal.org/user/102818/view) |
| 20:26.53 | ra21vi | any idea, please suggest |
| 20:29.36 | Qwell | ra21vi: Sounds like you've got rtptimeout set, and the softphone doesn't send RTP when there's silence. |
| 20:29.44 | Qwell | just a guess though |
| 20:33.45 | *** join/#asterisk digiv (~digiv@drupal.org/user/102818/view) |
| 20:40.56 | ipengineer | Isnt there a place where you can set a channel to hangup if it exceeds X minutes? I thought I saw that in a config at one point but am having problems finding it again |
| 20:42.40 | newtonr | I don't remember such an option |
| 20:43.36 | ipengineer | newtonr: So a call could essentially go on indefinitely then. There is no option to kill a channel if duration hits a threshold |
| 20:44.15 | newtonr | I'm not saying there isn't. I just don't remember one if there is one. |
| 20:44.35 | ipengineer | Ok.. I will keep digging thanks. I could just be dreaming |
| 20:44.37 | newtonr | seems like it would be an argument to Dial if it existed |
| 20:44.40 | ipengineer | wouldnt be the first time |
| 20:45.08 | newtonr | " S(x): Hang up the call <x> seconds *after* the called party has |
| 20:45.08 | newtonr | <PROTECTED> |
| 20:45.08 | newtonr | " |
| 20:45.37 | newtonr | ipengineer, is that what you are looking for? |
| 20:45.52 | ipengineer | I think it was global. I know that option is there. I think it is set and hanging up my calls |
| 20:46.09 | ipengineer | For whatever reason I thought it was in asterisk.conf but not seeing it |
| 20:47.31 | newtonr | oh there is the L argument for Dial as well |
| 20:49.25 | ipengineer | hmm.. Ok I will dig through my dialplan and make sure I am not setting it anywhere |
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| 21:01.44 | themrrobert | I've been having sporadic system crashes. Sometimes 2 or 3 per month, sometimes several months without an issue. I restart the asterisk server each night at midnight. max file limit is 5mil. It seems it's just unable to make new channels and everything stalls, even though the asterisk daemon is still running. I have a core file generated when i service asterisk restarted, but it has sensitive |
| 21:01.44 | themrrobert | information, where can i securely upload it? |
| 21:05.01 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 21:08.35 | themrrobert | here's the last 1000 lines of the log http://pastebin.ca/2835818 |
| 21:08.55 | themrrobert | the last 50 or so are where you can see theres a problem |
| 21:14.25 | shido6 | hrmm |
| 21:14.41 | ra21vi | ok let me check |
| 21:14.42 | shido6 | whats the rtp port range look like themrrobert ? |
| 21:15.24 | shido6 | rtp.conf - |
| 21:40.02 | *** join/#asterisk pjensen00 (~per@ip-64-21-247-189.far.ideaone.net) |
| 21:41.18 | pjensen00 | looking at the Bridge ARI REST api, I see that I can do something simple like "add channel to bridge". Is there a limit to the number of channels I can add? |
| 21:42.19 | ra21vi | \q |
| 21:46.40 | file | pjensen00, there is no hard-coded limit |
| 21:47.28 | file | but obviously you can't add an infinite number as the act of bridging channels takes X and if that exceeds the available time then you will get interesting results |
| 21:47.43 | pjensen00 | Man, I might have to switch over to ARI. I can't seem to figure out how to do that with AMI |
| 21:48.00 | pjensen00 | it seems so strait forward with ARI (Add channel X to bridge Y) |
| 21:48.14 | file | indeed, as that is what ARI is for |
| 21:48.35 | pjensen00 | Is there a reason why AMI and ARI such a different tool set? |
| 21:48.40 | pjensen00 | *have such |
| 21:49.43 | file | because they are for different purposes |
| 21:49.56 | file | https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573 |
| 21:52.30 | pjensen00 | thanks |
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| 23:08.36 | Kattyroo | fenderbender. |
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| 23:39.34 | hursjohn | Is this the channel to be in for help with libss7? #asterisk-ss7 is a channel I found, but it's a empty channel. |
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| 23:45.04 | hursjohn | How do I get SS7 to send carrier ID in IAM? pastebin.com/1YfftWW6 |
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