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| 01:05.06 | mrcina8989 | Hi everyone! |
| 01:05.33 | mrcina8989 | Anyone who wont to discuss about AMI ? |
| 01:08.59 | [TK]D-Fender | ~ask |
| 01:09.00 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 01:16.15 | MaliutaLap | free will is an option with *? |
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| 02:12.46 | onculayin | hey guys, I am trying to figure out is this case is possible. Lets say that calls are automatically directed to 5010, however sometimes someone might want to access 5001. Is it possible to dial 5001 directly while the calls are normally directed to 5010? |
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| 02:32.12 | eschmidbauer | if you send SIP 183 with SDP, does RTP get transmitted with that ? |
| 02:32.20 | eschmidbauer | RTP for the ringback |
| 02:32.22 | eschmidbauer | ? |
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| 03:08.32 | snadge | hmm.. asterisk 11.6 cert11 or whatever it is.. sip registration database appears corrupted |
| 03:08.45 | snadge | anyone seen this before? like multiple entries for the same thing.. and peers with username/default username mixed up |
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| 09:35.36 | jww | Hello. |
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| 09:39.11 | jww | is the sip kernel module required to run asterisk ? |
| 09:40.13 | AviiNL | you mean chansip? if so, then no, you could use pjsip as an alternative |
| 09:40.22 | WIMPy | The question is wrong. There is not _a_ sip module. |
| 09:40.51 | WIMPy | And no, you don't need it. One can help with FW rules, the other is likely to cause harm. |
| 09:45.50 | jww | I'm not very good with asterisk, but I'll try to ask better questions. |
| 09:46.16 | AviiNL | it might become more easy to answer if you explain what you want to achieve |
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| 09:47.49 | jww | I have to fix an old asterisk server, there were an upgrade of the virtualisation server from the isp, the vps was rebooted and since when I call the pbx I get a voice telling me all lines are busy. |
| 09:48.19 | jww | so I'm looking on what could be the problem. |
| 10:03.07 | carrar | haha upgrade only when broken |
| 10:03.15 | carrar | thats awesome |
| 10:03.37 | carrar | asterisk 1.0 is no longer supported ;) |
| 10:05.43 | carrar | jww: What is the error your getting? |
| 10:08.51 | carrar | wanders off to dinner |
| 10:13.32 | jww | I think it's [Feb 24 12:17:19] NOTICE[1561] chan_sip.c: Call from '0488369978' to extension '33488369978' rejected because extension not found. |
| 10:13.40 | jww | the context is default. |
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| 10:19.12 | jww | I also see an error 'SIP/2.0 404 Not Found' when enabling sip debug |
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| 11:01.55 | jww | it's strange because if I do dialplan show 33488369978@default , it show the extension. but still I get a 404 when calling it. |
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| 13:38.42 | jww | if anybody got an idea on why I have calls rejected because of extension not found( I guess ).I've put my logs and the errors message on http://pastebin.ca/3381642 . |
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| 14:02.19 | Kunsi | jww: no idea, but maybe try to use another context for testing? |
| 14:08.11 | whizzi | jww: Whatâs the context defined under the peer [budget-out] ? |
| 14:10.12 | jww | Kunsi: I tried , but I get the same issue. |
| 14:10.24 | jww | whizzi: I go check it out. |
| 14:10.28 | Kunsi | extension not found, or extension not found in context? |
| 14:10.56 | Kunsi | also, check for any non-printable characters in your extension |
| 14:12.26 | jww | whizzi: it's 'default' |
| 14:12.35 | whizzi | My guess is that under his [budget-out] peer, thereâs a context defined which doesnât include his 'default' |
| 14:12.50 | jww | Kunsi: I just have extension not found in the logs. |
| 14:14.09 | jww | whizzi: I'm not sure to understand, in sip.conf I have context=default in [budget-out] section . |
| 14:15.45 | whizzi | jww: Can we see the register part and peer definition from your sip.conf (without password obviously) |
| 14:16.03 | jww | whizzi: yes, I update the pastebin. |
| 14:16.18 | [TK]D-Fender | Thatt is an old version of a dea branch |
| 14:16.27 | [TK]D-Fender | The automatic response is "upgrade" |
| 14:16.57 | jww | Hello D-Fender. |
| 14:17.39 | [TK]D-Fender | dead* |
| 14:18.44 | jww | I gotta fix this problem before upgrading, I'm not very good with asterisk it could take time. |
| 14:19.29 | jww | whizzi: I updated the pastebin with peer info, but I can't find the 'register part' do you know in which file it is ? |
| 14:19.45 | whizzi | should be in sip.conf |
| 14:19.54 | whizzi | perhaps thatâs the issue ;) |
| 14:20.27 | jww | note i don't see any password either in sip.conf |
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| 14:21.11 | jww | I checked twice, there is nothing that looks like registration in sip.conf |
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| 14:24.11 | whizzi | no line that starts with register => |
| 14:24.12 | whizzi | ? |
| 14:24.32 | jww | no, this word isn't in the file. |
| 14:24.50 | jww | maybe it's because it's an old version. |
| 14:25.00 | whizzi | any line matching allowguest in there ? |
| 14:25.46 | whizzi | (should be in [general] of sip.conf ) |
| 14:26.09 | jww | no it's not there. |
| 14:27.13 | whizzi | Iâm not sure if 1.8-11 has allowguest default to off or on |
| 14:27.34 | jww | I got : allow=alaw if it can help. |
| 14:27.34 | whizzi | but either way, you want to be registered |
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| 14:30.32 | whizzi | you basically have a number coming in from a source and you havenât told Asterisk what to do with it |
| 14:30.42 | whizzi | mind the 'a source' |
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| 14:31.36 | whizzi | Iâm agreeing with [TK]D-Fender here that itâs best to upgrade |
| 14:32.02 | whizzi | you probably can get this to work if you place a context=default under the [general] section of sip.conf |
| 14:32.34 | jww | I try it |
| 14:32.46 | whizzi | and set allowguest = yes (which is not recommended) |
| 14:33.04 | whizzi | you better register the trunk by placing a register => line in your sip.conf |
| 14:35.40 | whizzi | and if you upgrade, make sure to fix your extensions.conf and replace the pipe symbols by commaâs ;) |
| 14:35.45 | whizzi | else it will fail |
| 14:36.20 | jww | okay so allowgest = yest ( context=default was already there ) didn't fixed the issue :\ |
| 14:36.29 | jww | erm s/yest/yes |
| 14:36.56 | whizzi | allowgest or allowguest ? |
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| 14:37.24 | jww | allowguest = yes , sorry. |
| 14:37.35 | whizzi | and you did a sip reload ? |
| 14:37.48 | jww | I restarted asterisk using init script |
| 14:38.37 | whizzi | you are also aware that Debian Squeeze support is ending in about 3 days |
| 14:40.46 | jww | it's not my server yet, I've just been called to fix that emergency. |
| 14:40.59 | jww | but if I do, maybe I could take care of the baby ! |
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| 14:41.21 | whizzi | wrong window closed :/ |
| 14:41.46 | jww | got my answers still ? |
| 14:41.56 | whizzi | what does surprise me is this line in the pastebin : Context 'default' created by 'pbx_ael' |
| 14:42.15 | whizzi | if your last answer was to fix this emergency, then yes |
| 14:42.37 | jww | yep. |
| 14:42.54 | jww | that pbx_ael is from extensions.ael file. |
| 14:43.34 | jww | what about that 404 error in the logs ? |
| 14:44.11 | whizzi | thatâs the reply of Asterisk back to the incoming call, number canât be found |
| 14:46.06 | jww | but when it's write Looking for 33488369978 in default (domain 80.248.221.165) , it doesn't write if it's context default from pbx_config or pbx_ael |
| 14:46.31 | jww | but in dialplan show there is an empty '[ Context 'default' created by 'pbx_config' ] |
| 14:46.44 | whizzi | Iâve never done anything with AEL |
| 14:47.13 | whizzi | my guess itâs going by default to the default of extensions.conf (the normale dialplan) |
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| 14:48.02 | jww | damnit !! |
| 14:48.21 | jww | I fixed it by renaming [default] context in sip.conf |
| 14:48.57 | jww | do the deer's dance |
| 14:49.12 | whizzi | in sip.conf ? |
| 14:49.17 | whizzi | not in extensions.conf ? |
| 14:49.33 | jww | yes it was in extensions.conf , sorry. |
| 14:49.51 | whizzi | Iâd be surprised if you renamed it in sip.conf and it still worked :P |
| 14:50.34 | whizzi | still, remove the allowguest=yes please, else youâll be getting a lot of incoming phone calls who will try to redirect to expensive numbers |
| 14:50.42 | whizzi | set it to no |
| 14:50.42 | jww | [TK]D-Fender: I guess it's time for upgrade ; |
| 14:50.53 | jww | whizzi: I commented it already |
| 14:51.03 | whizzi | and make a register => line |
| 14:51.10 | whizzi | no, you have to set it to 'no' |
| 14:51.18 | whizzi | by default itâs yes |
| 14:51.35 | jww | oh I'll do so. |
| 14:51.37 | whizzi | so Asterisk will accept INVITEâs from anybody |
| 14:51.53 | whizzi | if you leave it to 'yes' |
| 14:52.07 | jww | it won't break stuff if I set it to no ? |
| 14:52.23 | whizzi | it will if you donât have a register => line |
| 14:53.03 | whizzi | allowguest = no makes sure Asterisk only accepts INVITEâs from Registered trunks |
| 14:53.21 | jww | I have to read a bit the doc. |
| 14:53.37 | jww | thanks for your help guys ! |
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| 15:38.51 | zaf | hi.. i'm working with a fresh install of asterisk compiled with pjsip, and it looks like both chan_sip and chan_pjsip are enabled. is this right? Do i need to disable chan_sip manually? |
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| 15:43.09 | whizzi | zaf: probably yes. As far as I know Asterisk autoloads all modules |
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| 15:43.32 | zaf | seems weird that it would load both at once |
| 15:43.34 | whizzi | just put noload => chan_sip.so in modules.conf |
| 15:43.55 | zaf | k, thanks |
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| 15:49.40 | [TK]D-Fender | NEED to? No. WANT to? up to you |
| 15:50.02 | [TK]D-Fender | There are several cases where running both is beneficial. |
| 15:50.34 | zaf | but wouldn't they both be listening on the same ip/port bindings unless i change the configs for at least one of them? |
| 15:50.49 | [TK]D-Fender | They'll listen on what you configured them to listen to |
| 15:51.02 | [TK]D-Fender | And if you set them the same ... they will show their lack of appreciation :) |
| 15:51.30 | whizzi | :D |
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| 16:14.13 | zaf | ah, looks like in the sample configs, pjsip is not listening at all |
| 16:14.22 | zaf | thx for help |
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| 17:08.31 | _0x5eb_ | hi, is it possible to let Asterisk establish a direct RTP connection from a phone A to a peer B without serving as an RTP proxy? |
| 17:09.46 | _0x5eb_ | I tried directmedia=yes and directrtpsetup=yes both in the peer and in the phone sections of my sip.conf without any luck, RTP is still passing through Asterisk |
| 17:11.27 | [TK]D-Fender | it will go through * if it has a reason to |
| 17:12.56 | _0x5eb_ | yes, e.g. on hold, conference, voice mail etc., but here for a simple call, I don't see the need |
| 17:14.15 | [TK]D-Fender | RECORDING |
| 17:14.19 | [TK]D-Fender | DTMF OPTION |
| 17:14.21 | [TK]D-Fender | etc |
| 17:14.36 | [TK]D-Fender | transcoding |
| 17:18.25 | _0x5eb_ | is there a way to find out which option might make Asterisk take this decision? |
| 17:19.29 | [TK]D-Fender | look at the call |
| 17:19.51 | _0x5eb_ | you mean with sip debug? |
| 17:20.12 | [TK]D-Fender | Looking at less is not a smart idea |
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| 18:19.39 | MaliutaLap | [TK]D-Fender: it is if you don't actually want to solve anything |
| 18:20.09 | [TK]D-Fender | I consider "not wanting to solve" as "not smart" :) |
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| 18:40.56 | zaf | trying to set up upstream trunk provider using the examples at https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples, the aor section has a single contact=sip:$ip line. is it possible to have two contact lines there, or do i need to set up an AOR for each? |
| 18:42.05 | jhester | zaf: Yep, just enter a new line for the next contact entry below the first one. |
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| 18:42.36 | zaf | nice, thx |
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| 20:30.21 | lambda79 | hi everyone i am using Asterisk certified/13.1-cert3 x86_64 and i've got severe issues regarding cpu monopolyzation |
| 20:30.57 | lambda79 | top tells me 112,2 %CPUÂ is used |
| 20:31.38 | lambda79 | running an asterisk -r i am flooded with ERROR[30334]: stasis_cache.c:845 caching_topic_exec |
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| 20:33.35 | lambda79 | it that a known bug ? the closest i found was http://lists.digium.com/pipermail/asterisk-bugs/2015-July/147863.html |
| 20:37.41 | lambda79 | http://pastebin.com/886k0b7c |
| 20:39.42 | lambda79 | I notice cert4Â has been released should i uppgrade ? |
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| 20:43.43 | lambda79 | IÂ guess it is the 'realtime destroy' that is doing something nasty |
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| 20:51.27 | [TK]D-Fender | yes you should clearly upgrade |
| 20:51.38 | [TK]D-Fender | and Cert is worthless unless you have a service contract with Digium |
| 20:51.54 | [TK]D-Fender | so that puts it many releases behind |
| 20:52.12 | [TK]D-Fender | And they do REAL changes between even decimal versions in the 13 branch |
| 20:55.28 | lambda79 | ok thx i am downloading asterisk 13.7.2 hope it fixes realtime issues (i am using a proxy sip in front of *) |
| 21:00.05 | lambda79 | do you know how to get default values of asterisk sip driver ? i was thinking like something like in postfix postconf -d |
| 21:01.34 | lambda79 | i often spend many times looking for the meaning of specific options |
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| 21:06.01 | [TK]D-Fender | read the sample config |
| 21:06.10 | [TK]D-Fender | that's the place to go |
| 21:09.23 | lambda79 | you mean all the lines commented out is sip.conf ? that is the way to go |
| 21:14.00 | [TK]D-Fender | There are comments describing the various parameters including what is considered default |
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| 21:15.29 | lambda79 | ok they are pretty verbose that's true, specially since the 13.X releases i guess |
| 21:17.29 | [TK]D-Fender | the samples have been that way for a very long time |
| 21:17.46 | [TK]D-Fender | Of course there are simply MORE parameters with each branch |
| 21:19.01 | lambda79 | truth be told, it is pretty well explained, and it is a good linux philosophy |
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| 21:22.25 | [TK]D-Fender | "not bad". |
| 21:22.38 | [TK]D-Fender | There's room, but its a pretty decent start |
| 21:24.58 | lambda79 | yep regarding sip.conf only and having in mind rfc3261 explanations can come plentiful |
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| 21:35.00 | [TK]D-Fender | heads home |
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| 21:39.06 | dan_j | Hi. I'm trying to work out an issue with unattended transfers. With attended transfers, I'm able to see what the transferring peer is because it's doing the dialling before the call is transferred. But with unattended transfers, the call is simply passed on so I dont know who is doing the dialling. It's causing an issue because I'm unable to set the CallerID |
| 21:39.07 | dan_j | and instead the receiving peer sees 'asterisk' as the callerID. |
| 21:39.51 | dan_j | Two questions, is there any way to get the transferring peer during an unattended transfer? |
| 21:40.07 | dan_j | and if not, why is it displaying 'asterisk' instead of the original callerID? |
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| 22:29.04 | [TK]D-Fender | dan_j, Show us... |
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| 22:54.44 | dan_j | [TK]D-Fender: Sorry. All sorted. Thanks anyway. |
| 22:56.25 | dan_j | Btw, how do you stop asterisk changing the callerid to "via xxxxx". Its not in my dialplan. As soon as the channel is bridged, a new sip packet is sent and changes the callerid to 'via something' when a call is transferred. |
| 22:58.20 | [TK]D-Fender | Show us |
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| 23:15.51 | dan_j | Show you how? Theres nothing in the dialplan apart from the dial, but that command has finished executing when the call is finally transferred. And there is nothing in the CLI apart from the normal call transferred messages. Im not by my PC at the moment so will pastebin what i do have once i get back. |
| 23:16.41 | [TK]D-Fender | Like always, actual calls with actual dumps showing actual full proof of everything. |
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| 23:38.17 | F2Knight | Q: Having an issue with Queues not ringing agents. All agents are static members, and show up in queue. But show up as Ben (Local/720@from-queue/n from hint:720@ext-local) (ringinuse enabled) (Invalid) has taken no calls yet |
| 23:38.53 | F2Knight | Asterisk 13.7.2 |
| 23:39.13 | F2Knight | Any help would be greatly appreciated |
| 23:39.45 | F2Knight | http://pastebin.com/GfkZTiH2 |
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