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| 02:34.54 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.3.2 (2015/04/08), 11.17.1 (2015/04/08), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); 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 |
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| 03:36.08 | leftleg | hell all |
| 03:36.13 | leftleg | hello all |
| 03:41.27 | ChannelZ | hi |
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| 04:14.10 | leftleg | anyone use freepbx here? |
| 04:15.49 | leftleg | i'm looking for a robodialer for it. |
| 04:16.18 | leftleg | for a school to notify parents when school is not in session. |
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| 04:47.39 | LiuYan | leftleg: for asterisk, i used to create a cron job, so it repeatly scan a database, and fetch information from it, and make a .call file, feed it to asterisk, asterisk will dial the number. |
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| 05:02.46 | danielbw | why would asterisk 13 be inserting silence and talking spurts on a dahdi to sip call |
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| 07:20.58 | NoobSaibot | if 'show sip peers' is displaying a phone with an extension that is active with Status: OK, but the phone itself is reporting itself as being registered, where's the first place to look? |
| 07:21.07 | NoobSaibot | *Not being registered |
| 07:21.33 | NoobSaibot | god it's late, i need to proofread better. sorry about that jumble of garbage. |
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| 07:45.02 | cookiez | Hello all, in comedian mail. Is there anyway to disable some menu-choices? I'm thinking mostly about disabling all other folder choices except new and old messages. Thank you in advance! |
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| 07:56.51 | NoobSaibot | Figured out the problem - erroneous internal DNS entry pointing to the wrong server. sip.ourdomain.com aimed at an esxi server's IP. |
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| 08:38.51 | Guest9903 | Hi, is it possible to store sip traffic from asterisk to a file instead of using sip set debug on which will splatter the CLI? |
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| 08:52.24 | Chainsaw | Guest9903: You could do a wireshark capture and handle the traffic that way? |
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| 09:04.06 | zizo_ | Hi all, sorry for my non-technical language. I have a fully working asterisk, which, among usual thigs, provide a callback service. If a certain number calls the server the call hangs and asterisk calls-back the number asking to digit a number to call. Is there a way for the first calling number to send dtmf tones to asterisk without answering? |
| 09:06.38 | zizo_ | Or better, there is a way to read tones without an answer? Thanks |
| 09:07.26 | skrusty | i dont think so |
| 09:07.29 | Kunsi | i don't think that's possible. maybe you want extension numbers instead? |
| 09:10.10 | Guest9903 | Chainsaw: not a very nice solution indeed. |
| 09:10.33 | Chainsaw | Guest9903: It stores the SIP traffic like you asked. It does not splatter the CLI. |
| 09:10.50 | Chainsaw | Guest9903: You will have to qualify "nice", as it ticks all the boxes. |
| 09:10.50 | Guest9903 | Can it be run as a service? |
| 09:11.37 | Chainsaw | Guest9903: Wireshark? Sure. It can write to the file continuously, and your filters can be set up such that only the traffic you are interested in makes it to disk. |
| 09:14.57 | Guest9903 | I think I'll go with tcpdump for this one though. All together I wished to avoid any external programs. |
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| 09:31.21 | wdoekes | Guest9903: you could drop the verbose messages from the cli altogether and only write to file (logger.conf), but that may be more silent than you wish for |
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| 10:55.10 | jkroon | hi all, with ReceiveFAX, it used to be that FAXMODE was set. I like to log what fax mode was used for transfer (it helps to track incorrectly configured SIP gateways that doesn't have t.38 enabled), however, it seems this variable hasn't been set for a while and I can't find the appropriate replacement in FAXOPT() ... has this been lost or is there some other mechanism? |
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| 13:01.31 | vader- | uhhg, i have to figure out why this office which has about (5) Polycom 650s and (4) Polycom 350s, are rebooting randomly now. They have been in place for 2-3 years. No updates. Running 4.0.2 code on them with FreePBX. Any ideas? |
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| 13:28.28 | DivideBy0 | vader-: bad power? |
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| 13:48.34 | vader- | they are PoE |
| 13:53.55 | DivideBy0 | still sticking with power, is it a old POE switch? |
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| 14:03.18 | file | needs more power! |
| 14:03.45 | Kunsi | unlimited power! |
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| 14:10.48 | vader- | Netgear GSM7224P |
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| 14:45.26 | vader- | it looks like it might be 1 phone causing the issue which is causing the switch to knock out a few PoE ports |
| 14:45.45 | vader- | I displayed the "culprit" phone's port and it seems to have leveled the rest of the ports off |
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| 15:49.53 | xochilpili | hi all, im having issues tryingg to provisioning an h323 avaya phone with asterisk. I have loaded the module, and the phone seems to seek to asterisk, but asterisk, is not displaying anything |
| 15:50.04 | xochilpili | does anyone made this? |
| 16:03.27 | [TK]D-Fender | What does "seem to seek" mean? |
| 16:08.04 | xochilpili | hey [TK]D-Fender :D |
| 16:08.58 | xochilpili | i meant, when i turn on the phone, search http to asterisk http server, and read the config files |
| 16:09.07 | *** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl) |
| 16:09.29 | xochilpili | but in phone's display, just said : discover 10.0.1.55 (asterisk server) |
| 16:10.22 | xochilpili | the real question is if i can connect an h323 phone to asterisk |
| 16:11.17 | xochilpili | if it's possible to make it, researching on internet, everyone make a trunk between asterisk an avaya, im not trying to use avaya pbx. |
| 16:31.01 | [TK]D-Fender | xochilpili: * can talk H323 |
| 16:31.10 | [TK]D-Fender | But that has NOTHING to do with it looking for HTTP for configs |
| 16:31.15 | [TK]D-Fender | Do not mix up topics like that |
| 16:32.53 | xochilpili | what you mean? the phone is calling to HTTP to read his settings. (im not mixing this up with *), the question is about, how does ooh323 is communicated with the phone |
| 16:38.05 | [TK]D-Fender | H.323 is H.323 |
| 16:38.18 | [TK]D-Fender | The fact the phone wants configs is SEWCONDARY |
| 16:38.28 | [TK]D-Fender | Put it's configs wherever it is loking to get them from |
| 16:38.36 | [TK]D-Fender | That has NOTHING to do with Asterisk |
| 16:44.58 | xochilpili | [TK]D-Fender, thanks again! |
| 16:44.59 | xochilpili | :D |
| 16:45.04 | xochilpili | brb |
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| 18:39.32 | ch4plin | hi everyone |
| 18:40.10 | ch4plin | I have a question, is it possible an agent logged in to a queue to dial manually? |
| 18:40.38 | [TK]D-Fender | "agents" don't dial |
| 18:40.41 | [TK]D-Fender | Devices dial |
| 18:41.06 | [TK]D-Fender | Also, what defines "manually" in this case? |
| 18:42.56 | ch4plin | I'm creating a web app with elastix, so I'm using the Call center module, my app works with ECCP protocol, but they dont have the dial command implemented yet, that's why I looking for alternative, to let the agents put a number and send it to dial |
| 18:43.12 | ch4plin | hope you can get what a means |
| 18:43.33 | [TK]D-Fender | What protocol? |
| 18:44.16 | ch4plin | it's a own elastix protocol, ECCP is named, works with XML languague |
| 18:45.06 | [TK]D-Fender | Well whatever that actually means and does is purely an Elastix convention, and is not supported here, along with their entire GUI |
| 18:45.56 | ch4plin | I know, just want to need if there was an option by using asterisk core to make a call with an agent |
| 18:46.28 | ch4plin | so I guess i goint to develop a service to dial using ami |
| 18:46.39 | ch4plin | thanks for your comments |
| 18:47.13 | *** join/#asterisk areski (~areski@80.174.128.121.dyn.user.ono.com) |
| 18:48.14 | [TK]D-Fender | "Agent" isn't really a thing with Asterisk |
| 18:48.22 | [TK]D-Fender | The terms are vague |
| 18:48.46 | [TK]D-Fender | AMI (Originate) dials TO a device instead of you dialing FROM a device |
| 18:49.11 | [TK]D-Fender | So in the end that's jsut a device ending up on a call . Difference being which end well ... originates it |
| 18:49.34 | [TK]D-Fender | And if the device is capable of recieving a call.. it sure is capable of placing one itself |
| 18:51.11 | ch4plin | I was thinking to originate to a channel like SIP/{number}@trunk, then "transfer" to exten where a Queue, something like |
| 18:51.36 | [TK]D-Fender | How does that relate to your concept of "agent"? |
| 18:51.58 | [TK]D-Fender | You're calling "something" and when they answer are dumpign them INTO a queue... not as a member, but as a caller. |
| 18:52.01 | WIMPy | You wan to call someone to put him in to a queue? |
| 18:52.17 | WIMPy | I don;t think I understand anthing. |
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| 18:55.10 | ch4plin | something like that, it just an idea, I'll try and if I succeed I'll let you know |
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| 19:00.27 | *** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 19:00.59 | hexanol | I'm doing some tests on asterisk 13 to eventually update to it from asterisk 11 |
| 19:01.13 | hexanol | and, using chan_sip and when direct media is activated |
| 19:01.35 | hexanol | I've found that features code (for example, my feature code to hangup a call is *0) needs an additionnal # key |
| 19:01.42 | hexanol | to be accepted |
| 19:01.44 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-odwhejsxpdjlgufu) |
| 19:02.08 | hexanol | i.e. if I'm in a call with direct media enabled, I need to press *0# to end the call, while if direct media is disabled, I only need to press *0 |
| 19:02.27 | hexanol | am I crazy or is this something that happened to other people ? |
| 19:02.35 | [TK]D-Fender | You're crazy.... |
| 19:02.53 | hexanol | alright, so could this be related to my configuration ? |
| 19:02.58 | [TK]D-Fender | * can't do direct media AT ALL if features are enabled |
| 19:03.14 | hexanol | hum that's not what I'm seeing here |
| 19:03.20 | hexanol | I have "rtp set debug on" activated |
| 19:04.29 | hexanol | Dial("SIP/je5qtq-0000000a", "SIP/4wiilx,10,hHtT") in new stack |
| 19:04.29 | hexanol | ... |
| 19:04.29 | hexanol | > Remotely bridged 'SIP/4wiilx-0000000b' and 'SIP/je5qtq-0000000a' - media will flow directly between them |
| 19:06.27 | [TK]D-Fender | Hrm... |
| 19:14.45 | mjordan | uhm |
| 19:14.58 | mjordan | I'm not sure how you would be getting DTMF at all, unless it was via SIP INFO |
| 19:15.11 | mjordan | much less why it is remotely bridging with hHtT parameters added. |
| 19:15.29 | hexanol | I'm using SIP INFO, yes |
| 19:16.12 | mjordan | well, then it is theoretically possible to do what you are doing |
| 19:16.48 | hexanol | it does work on asterisk 11 |
| 19:17.22 | mjordan | I didn't say it didn't |
| 19:17.35 | mjordan | SIP INFO is just not what I'd typically recommend for DTMF, but okay |
| 19:18.34 | hexanol | what are the reason ? |
| 19:18.56 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:19.31 | mjordan | it has no beginning or end, and duration has to be completely emulated |
| 19:19.36 | mjordan | this can cause some rather strange effects |
| 19:19.56 | mjordan | it also doesn't interrupt the media on the sending side, but can interrupt the media on the receiving side when passed through, causing more weird effects |
| 19:20.18 | *** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta) |
| 19:22.02 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 19:23.16 | hexanol | doing some more tests, and in fact I just need to press any DTMF after pressing *0 for the feature code to be considered, not # in particular |
| 19:24.05 | hexanol | do you think it's worth opening a bug on the asterisk's bug tracker (if it's not already opened)? |
| 19:25.56 | wanda_ | hiiii people |
| 19:26.21 | wanda_ | I organised 2 BarCamp about voip open source :) |
| 19:26.43 | wanda_ | participants want an other one with mummmmmble : http://barcamp.org/w/page/97007235/BarCampMumble-VoIP-OpenSource |
| 19:27.38 | wanda_ | so, please. let us know your news or favorite tips about VoIP and we could discuss together: wednesday 17th june |
| 19:28.26 | wanda_ | 16H France, 10H north america coast , for the meeting, with mumble :) - Thank you ! (I'm here #xivo) :] |
| 19:47.48 | Dovid | Does anyone know wht would cause this? "res_musiconhold.c: getcwd() failed: No such file or directory". I cheked and the files are on the system and readable. we had an issue earlier with nfs but it was resolved. what's interesting is asterisk gives the same errors for files that are stored locally and files that are on the NFS server |
| 19:49.03 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 19:50.59 | WIMPy | hexanol: It's probably worth looking closely in what's actually happening. I got a similar issue with native bridging. |
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| 20:00.50 | *** join/#asterisk polysics (~polysics@93-38-164-94.ip71.fastwebnet.it) |
| 20:00.53 | polysics | hello! |
| 20:00.59 | leftleg | hey |
| 20:01.09 | polysics | are there some tricky differences between Asterisk 11 and 13 media wise? |
| 20:01.41 | polysics | because I am in a situation where an 11 box can dial a peer on the PBX fine and play music (or have media), while 13 with the exact same config can't |
| 20:03.34 | Chainsaw | polysics: If you are using the pjsip stack with 13, then yes, quite significant differences. |
| 20:04.18 | polysics | I did not install this box myself, so I don't know. What is hte best way to check quickly? |
| 20:05.22 | *** join/#asterisk pchero (~pchero@0x555140b5.adsl.cybercity.dk) |
| 20:06.51 | polysics | for what it's worth I do not think that is the case, though, since this was an upgrade of an existing build script |
| 20:07.19 | *** join/#asterisk caseyd (~Casey.Dav@olivaw.mdteam.com) |
| 20:08.46 | newtonr | polysics, 'module show like chan_sip.so' and 'module show like chan_pjsip.so' |
| 20:09.03 | newtonr | also, you can check for a configured sip.conf or pjsip.conf |
| 20:09.12 | polysics | they seem to both be there |
| 20:09.28 | newtonr | both are loaded? |
| 20:09.35 | polysics | yes |
| 20:10.06 | newtonr | then you can check which config file has actual configuration vs a sample configuration/comments |
| 20:10.10 | polysics | if I change something in sip.conf it does affect calls |
| 20:10.15 | caseyd | I'm having some trouble, with I guess getting sip to work with my checkpoint firewall. It works if I plug the phone straight into the internet, or a more simple firewall. I'm not seeing anything dropped in the firewall logs. I've tried all sorts of different rules, I've even allowed all content and its still not working. I also tried setting the static nat to match the asterisk server. Does anyone have any ideas? |
| 20:10.31 | polysics | newtonr: pjsip.conf is definitely a sample |
| 20:11.24 | newtonr | polysics, then it sounds like you are 'using' chan_sip. However you also have pjsip loaded and you probably don't want that if you are not using it. |
| 20:11.26 | newtonr | polysics, https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip#Migratingfromchan_siptores_pjsip-Disablingres_pjsipandchan_pjsip |
| 20:11.33 | *** part/#asterisk Kunsi (kunsmannf@unaffiliated/kunsi) |
| 20:11.42 | newtonr | That talks about how to disable pjsip while you are still using chan_sip |
| 20:11.57 | polysics | might that be causing media issues? calling a peer on the same box does have media, whatever that means |
| 20:12.07 | newtonr | If you don't understand how to configure them both and use them at the same time then you shouldn't have them both loaded and running |
| 20:12.22 | newtonr | and for 99.99% of people you will only want one or the other |
| 20:12.40 | leftleg | caseyd it might be the app control on the firewall |
| 20:13.08 | newtonr | polysics, possibly. I'd make sure the pjsip modules don't load to avoid any issues. |
| 20:13.15 | leftleg | i don't have that brand, but i've run into that problem where is firewall identifies voip traffic and woudl block if so enabled. |
| 20:13.43 | polysics | ok, the modules are gone |
| 20:14.15 | polysics | media is still not there but w/e :) |
| 20:14.27 | polysics | the same exact sip.conf works with 11 so it must be some new option |
| 20:15.04 | polysics | iptables has no rules... no idea on where to look now |
| 20:15.26 | caseyd | leftleg, I wish.. we don't have that. We only have the basic firewall |
| 20:16.06 | newtonr | polysics, how is your media issue presented to the user? Does a call between two phones not have any audio? |
| 20:16.27 | polysics | originate to the same PBX peer from the CLI has music in one case, no music in the other |
| 20:16.32 | polysics | same for a Dial()ed call |
| 20:16.41 | polysics | in fact ,the call is not even acknowledged |
| 20:16.47 | newtonr | what is the difference between the two cases? |
| 20:16.49 | polysics | the call goes down after 5-6 seconds |
| 20:16.59 | [TK]D-Fender | We don't know |
| 20:17.05 | polysics | basically, the Asterisk version |
| 20:17.06 | [TK]D-Fender | You haven't shown us anything yet |
| 20:17.31 | polysics | they are in the same network, and using the same peer to call the PBX |
| 20:17.36 | polysics | what could I show you? |
| 20:17.49 | newtonr | polysics, https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 20:18.12 | newtonr | you can leave debug on 0 |
| 20:18.32 | newtonr | the important parts will be verbose and the "sip set debug on" |
| 20:21.26 | polysics | gist is ok? |
| 20:22.22 | newtonr | sure |
| 20:23.04 | polysics | here's what I got out of that |
| 20:23.05 | polysics | https://gist.github.com/polysics/5635b593b5909c2caecd |
| 20:25.03 | newtonr | polysics, what were the parameters to originate? |
| 20:25.35 | polysics | originate SIP/talkbox-peer/3997 application musiconhold |
| 20:28.47 | newtonr | the 200OK and ACK are repeating. Perhaps the ACK is not reaching the far end? |
| 20:30.07 | newtonr | yeah the peer even hangs up with "X-Asterisk-HangupCause: No user responding" |
| 20:31.35 | newtonr | polysics, well, you have something to dig into there. |
| 20:31.53 | polysics | at what level might the issue be? networking? |
| 20:32.46 | newtonr | could be that or could be NAT related config options in sip.conf (on either end) that are misconfigured |
| 20:33.10 | polysics | are there any differences between 11 and 13 on that side of config? |
| 20:33.57 | newtonr | Probably. You could search through the upgrade and changes files included with Asterisk |
| 20:34.11 | polysics | yeah, I am reading through those |
| 20:34.29 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Updating+or+Upgrading+Asterisk |
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| 20:45.31 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 21:20.28 | *** join/#asterisk wmchris (~Krim@ip-176-198-200-19.hsi05.unitymediagroup.de) |
| 21:21.27 | wmchris | hi, need some help with my asterisk installation. i can connect to my sip account, but when i add the register => command in the sip.conf, i can't connect anymore. dunno why. |
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| 21:22.58 | wmchris | using Asterisk 1.8.10.1 on a openwrt router. |
| 21:23.18 | wmchris | really need help :( trying around this since 2 days... |
| 21:23.29 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 21:23.53 | wmchris | i don't even know how i can debug it, because there are no error messages in console or log file... |
| 21:24.50 | wmchris | noone online right now? |
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| 21:38.26 | newtonr | wmchris, yo |
| 21:38.39 | newtonr | wmchris, you can turn on "sip set debug on" on the console to see the SIP traffic |
| 21:38.57 | newtonr | then you can look to see if a REGISTER is going out |
| 21:39.35 | wmchris | already did... nothing. |
| 21:39.56 | wmchris | there is really nothing in the console |
| 21:40.31 | newtonr | what do you see with "module show like chan_sip.so" ? |
| 21:40.34 | wmchris | it's like the whole sip subsystem crashes when i add a register command |
| 21:40.51 | wmchris | module loaded. |
| 21:40.51 | wmchris | chan_sip.so Session Initiation Protocol (SIP) 0 |
| 21:41.03 | *** part/#asterisk mjordan (mjordan@nat/digium/x-mclygcetabdjsqrn) |
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| 21:41.16 | newtonr | so chan_sip is loaded |
| 21:41.25 | newtonr | you could reload it and watch for console messages |
| 21:41.36 | newtonr | you need to make sure of some things first |
| 21:41.42 | newtonr | "logger show channels" |
| 21:41.57 | newtonr | see what log message types are enabled for the console |
| 21:42.12 | wmchris | after module reload chan_sip the console stops working |
| 21:42.13 | newtonr | and remember that both verbose and debug have indepdent log levels |
| 21:42.59 | wmchris | if i open a new one and redo the command i'll get OpenWrt*CLI> module reload chan_sip |
| 21:42.59 | wmchris | The previous reload command didn't finish yet |
| 21:43.11 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 21:43.36 | wmchris | console has notice, error and warnings active. seems enough for me |
| 21:43.57 | newtonr | in a case like this you want verbose and debug, turn them both up to 5 |
| 21:44.03 | newtonr | *in addition |
| 21:44.13 | wmchris | done |
| 21:44.30 | newtonr | then try the reload again |
| 21:44.34 | newtonr | from your original state |
| 21:44.38 | *** join/#asterisk bkruse (~Adium@64.89.97.100) |
| 21:44.46 | wmchris | wait a sec, core set debug 5 doesn't enable verbose and debug |
| 21:44.54 | wmchris | so what is the command to add verbose and debug on console? |
| 21:45.15 | newtonr | safest to do it from the config file logger.conf |
| 21:45.38 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Logging+Configuration |
| 21:45.41 | wmchris | done |
| 21:46.05 | newtonr | but you can do it from the console too |
| 21:46.06 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Basic+Logging+Commands |
| 21:46.08 | wmchris | and again.... after module reload chan_sip => asterisk crashes |
| 21:46.18 | wmchris | or at least the command line interface stops working. |
| 21:46.23 | wmchris | without an error message |
| 21:46.42 | wmchris | if i comment the register=> command out, everything works fine. |
| 21:47.06 | newtonr | can you pastebin the register line? |
| 21:47.44 | wmchris | it doesn't matter, every register line results in this behavior. |
| 21:47.57 | wmchris | even a register => user@nothing |
| 21:48.28 | newtonr | ah, very odd |
| 21:50.30 | newtonr | I'd say file a bug and post logs to it, but not only is that version really old within the 1.8 branch, but the 1.8 branch is only supported for security issues now. |
| 21:50.54 | newtonr | I'd recommend comparing at least the latest version of 1.8 |
| 21:52.27 | wmchris | i'm using an old router right now with openwrt. |
| 21:52.51 | wmchris | so it's really complicated to compile things for |
| 21:53.52 | newtonr | Good luck! Not sure what else to try if absolutely any register line causes Asterisk to hang when you load chan_sip.so |
| 21:54.04 | newtonr | It certainly shouldn't do that |
| 21:55.35 | wmchris | maybe there is another solution for my problem. |
| 21:56.24 | wmchris | i just need to proxy the sip from my home line to a VPN line. first i thought using siproxd - didn't work. :( asterisk doesn't work either... |
| 21:56.31 | wmchris | maybe there is another solution? |
| 21:58.29 | wmchris | ah dammit, i'm installing a debian machine. |
| 21:58.48 | wmchris | i tried using a router... i failed... ^^ |
| 21:59.03 | wmchris | thank you very much for your help :) |
| 21:59.08 | wmchris | (no irony!) |
| 21:59.27 | wmchris | at least now i'm sure it's not a fault of my config file. ^^ |
| 22:00.58 | *** join/#asterisk vader- (~Adium@2600:1002:b123:c26e:bdac:13ac:78f1:d41b) |
| 22:01.53 | newtonr | wmchris, can you pastebin your entire config file? |
| 22:02.19 | newtonr | wmchris, there is always a chance that the location of the register line, or a combination of configuration elements in the file have caused some strange issue |
| 22:03.20 | *** join/#asterisk theHub (~chatzilla@ool-182e21ee.dyn.optonline.net) |
| 22:04.03 | wmchris | i'm using this setup (http://www.ip-phone-forum.de/showthread.php?t=250556&s=9a0411e6e35b6e22f560a302dc87170b&p=1856801&viewfull=1#post1856801) |
| 22:04.16 | wmchris | ofc extensions.conf and sip.conf are inversed here ;)# |
| 22:04.45 | *** join/#asterisk bkruse (~Adium@172.56.20.205) |
| 22:04.56 | wmchris | but i also tried the automatic configuration of the module luci-btx |
| 22:06.02 | wmchris | and this config works fine on the debian machine i'm running right now :) |
| 22:06.41 | wmchris | so problem not solved, but it's definitly a bug in asterisk 1.8 on ar71xx chipset |
| 22:08.26 | newtonr | did you try disabling srvlookup? |
| 22:10.39 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 22:13.00 | wmchris | nope |
| 22:13.04 | wmchris | why should i disable this? |
| 22:14.50 | newtonr | see https://issues.asterisk.org/jira/browse/ASTERISK-21378 and duplicates |
| 22:16.07 | wmchris | hm i'll try |
| 22:16.35 | wmchris | but i can do a lookup manually and i don't get a timeout even after 8hours.... |
| 22:16.41 | wmchris | so i doubt this is the bug :/ |
| 22:16.57 | wmchris | (and tbh.. i just made my first call with the debian machine ;-)) |
| 22:17.13 | newtonr | i doubt it will affect it in this case either, but it is worth a shot |
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| 23:08.20 | _danielbw | hello |
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