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| 00:05.57 | fury_ | WIMPy: thanks. will do. meanwhile, a non-superuser (or non-'asterisk' user) cannot interact with the asterisk CLI. this could be fixed if I add the user to the asterisk group, but the permissions are 755 on the socket file. is there a way to get asterisk to set chmod 775 on its socket file? |
| 00:08.43 | fury_ | nevermind found it! |
| 00:18.33 | Penguin | fury_: I use netcat to connect to the manager (AMI). |
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| 01:31.00 | turbzy | is there a new method of viewing SIP debug information on Asterisk 13.0.0 |
| 01:31.09 | turbzy | using sip set debug on in the CLI doesn't seem to do anything |
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| 07:42.39 | Pernat | how to make outgoing call from SIP to mobile via GSM modem? |
| 07:44.42 | Pernat | i have working gsm modem and asterisk use it to get calls |
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| 08:11.59 | saxa | Pernat: I have no experience with GSM modem, but if you can make calls from * with that modem probably a simple Dial() should work. |
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| 08:13.19 | Pernat | yeah but i want to call by softphone entering a number which i want to call |
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| 08:33.06 | Pernat | i need to call all gsm numbers by gsm dongle which can anwser calls. anybody? |
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| 08:47.44 | kaldemar | Pernat: find out how you dial using it, and then make an extension that uses it for your gsm numbers. |
| 08:52.28 | Pernat | i got this: chan_sip.c:25624 handle_request_invite: Call from '2500' (192.168.1.2:5060) to extension '+48XXXXXXXXX' rejected because extension not found in context 'from-internal'. and don't know what now |
| 08:52.31 | tparcina | Pernat: How is your GSM modem connected to *? |
| 08:53.04 | Pernat | tparcina it is connected via USB using chan_dongle |
| 08:53.17 | tparcina | Pernat: Why don't you add +48xxx extension to your from-internal context? |
| 08:54.57 | Pernat | i got _48531XXXXXX gsm number in USB dongle and i want to call to +48530XXXXXX |
| 08:55.50 | Pernat | im first time installing asterisk and im very new in it |
| 08:55.53 | kaldemar | exten => +48530XXXXXX,1,Dial(Dongle/DONGLE_NAME_FROM_dongle.conf/${EXTEN}) |
| 08:57.00 | kaldemar | replace DONGLE_NAME_FROM_dongle.conf with the name inside square brackets in your dongle.conf. the sample config has [dongle0] for example. |
| 08:57.44 | Pernat | if i write exten => +48XXXXXXXXX i should can call all gsm numbers? |
| 08:57.58 | kaldemar | X means 0-9 |
| 08:58.54 | kaldemar | so that will match all numbers that start with +48 and have length of 12. |
| 08:59.23 | kaldemar | https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
| 09:00.13 | kaldemar | you need an underscore there aswell to denote a pattern, so exten => _+48530XXXXXX |
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| 09:09.04 | Pernat | okay it works. thanks :) |
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| 09:23.27 | justdave | so my IVR menus suddenly in the last week or two have started dropping random digits when people try to dial extension numbers, and I haven't changed any config lately that I'm aware of. What might I poke at to debug that? |
| 09:24.05 | justdave | using rfc2833 for dtmfmode across the board |
| 09:24.30 | justdave | and I can replicate it direct-dialing into an IVR from a local extension, not just from the incoming SIP trunks |
| 09:25.24 | justdave | it's acting like it's doing inband dtmf and has the backoff timeout set too long |
| 09:25.46 | justdave | but everything I can find says to use rfc2833 |
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| 09:56.16 | justdave | if I dial slowly it works. |
| 09:56.45 | justdave | I thought it wasn't supposed to matter with rfc2833 how fast you dialed because it was just a signal and not interpreting tones |
| 10:04.35 | justdave | (and this used to work no matter how fast you dialed) |
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| 10:35.38 | DND | hi guys, i have a 32 channel pri line. and i wanted to know how many is being used currently. is there any command in asterisk that does that? |
| 10:36.14 | DND | thing is, we are getting "all lines are busy now" and im suspecting that all 31 channels are being used |
| 10:39.14 | evilman_work | 30 calls. one timeslot is busy by signaling, another one is busy by synchronization. |
| 10:42.31 | DND | evilman_work: how can i get a realtime status/ |
| 10:43.41 | stevenm | could anyone with a 504G tell me if 'auto answer' is an option on it's web GUI? I can't seem to find it on mine - but it might be locked down. |
| 10:44.13 | DND | dahdi show channels is not useful |
| 10:45.32 | evilman_work | DND: depends on your PRI-card. for zaptel conftool there is command '/sbin/ztcfg -vv'. |
| 10:46.41 | evilman_work | DND: for digium PRI card you can run 'cat /proc/zaptel/<card-number>' to check status of timeslots. |
| 10:46.52 | DND | im using digium card. i tried lsdahdi and here's the output: http://pastebin.com/raw.php?i=SK5N28eG |
| 10:47.06 | DND | evilman_work: ok will try that |
| 10:47.46 | DND | i dont have zaptel. sorry i forgot. i used asterisknow |
| 10:47.58 | DND | its an old version. not the new version |
| 10:48.57 | DND | evilman_work: ok i used cat /proc/dahdi/1 |
| 10:49.54 | DND | here's the output. what does the "EC:" means in line 1? http://pastebin.com/raw.php?i=d9QhDGWJ |
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| 10:56.21 | theahindle | Hi |
| 10:56.28 | theahindle | Does Asterisk support remote SIM servers/hosting? |
| 10:56.43 | theahindle | Getting SIM data from another server, to use on the local Asterisk box? |
| 10:59.46 | file | SIM data? |
| 11:03.37 | theahindle | Whatever's needed for the GSM module to use that SIM - ICCID/IMSI |
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| 12:06.55 | workingcats | theahindle, pretty sure the SIM has to more than that |
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| 12:08.43 | theahindle | workingcats: I don't believe so, but remote sim servers _do_ work/exist, I just wondered if it's possible with Asterisk |
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| 12:13.36 | file | the GSM stuff isn't part of mainline Asterisk, so that would be a question for wherever it came from |
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| 12:50.40 | theahindle | Thanks file |
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| 13:08.52 | Katty | morning |
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| 13:09.47 | Rico | hui |
| 13:09.49 | Rico | hi |
| 13:09.55 | Rico | I'm having a strange problem with asterisk |
| 13:11.43 | Rico | how can I know which group asterisk runs in ? |
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| 13:15.15 | [TK]D-Fender | Rico: Look at how you start it |
| 13:15.41 | Rico | rungroup is commented |
| 13:16.23 | [TK]D-Fender | Good odds it mirrors the user |
| 13:16.23 | Rico | so I only tell it a user to run in |
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| 13:16.33 | Rico | ok |
| 13:22.52 | Rico | thanks |
| 13:22.59 | Rico | so now second question |
| 13:23.20 | Rico | if I run a shell in asterisk user, I can access to the file I need (rx) |
| 13:23.25 | Rico | but asterisk can not |
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| 13:24.16 | Rico | Unable to open /home/asterisk_sounds/closed/demo/fermeture (format 0x4 (ulaw)): Permission denied |
| 13:24.59 | [TK]D-Fender | that folder doesn't look normal |
| 13:25.02 | [TK]D-Fender | as a name |
| 13:25.27 | Rico | http://pastebin.com/LsJ3AziX |
| 13:25.40 | Rico | yes I know, but that's not the problem (I guess) |
| 13:26.07 | Rico | I do my "Playback()" with full path |
| 13:26.33 | [TK]D-Fender | what about the rest of the path? |
| 13:26.43 | Rico | mmmmh, can it be a selinux problem ? |
| 13:26.53 | Rico | the rest of the path ? |
| 13:26.54 | [TK]D-Fender | It could, since * does not support it |
| 13:27.03 | [TK]D-Fender | SELINUX needs to be disabled |
| 13:27.27 | Rico | thanks |
| 13:28.02 | Rico | I think that's my problem |
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| 15:19.02 | wasanzy | hello |
| 15:19.15 | wasanzy | am trying to do ssh forward to a sip server but it is not working. |
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| 15:22.38 | [TK]D-Fender | wasanzy: ok |
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| 15:33.33 | Dimitripietro | Hi, I'm having hard time emulating a PRI between Asterisk and a Nortel CS-1000. Call from the CS-1000 to Asterisk work fine but call from Asterisk to CS-1000 I get Hangup Cause 31 from the CS-1000 |
| 15:33.58 | Dimitripietro | I know for sure I'M sending the right pattern (4digits) to the CS-1000 |
| 15:34.31 | Dimitripietro | I'm used to this kind of setup, first time it don't work |
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| 15:40.09 | vlad_starkov | Question: how can I set the channel's language? `__LANGUAGE=en` or `__LANGUAGE()=en` or `__LANGUAGE(en)`? |
| 15:40.19 | Qwell | Set(CHANNEL(language)=en) |
| 15:42.39 | vlad_starkov | Qwell: Should I prefix it with double underscore ? |
| 15:43.04 | Qwell | It's a function. I'm not sure if that works there. It may. Try it. |
| 15:44.59 | vlad_starkov | Qwell: If I use AMI to send 'Originate' action, how should I apply this function? |
| 15:45.38 | Qwell | Variable: CHANNEL(language)=en |
| 15:45.40 | Qwell | or something |
| 15:46.45 | Dimitripietro | You don't need the __ |
| 15:47.36 | vlad_starkov | Dimitripietro: because this is not the "variable" in classic sense so the variable inheritance rules is not applied here? |
| 15:49.02 | rmudgett | Functions are not channel variables. The __ prefix only applies to channel variables. |
| 15:49.09 | vlad_starkov | Qwell: Variable: CHANNEL(language)=en worked fine! Thanks! |
| 15:49.46 | Qwell | rmudgett: So how does that work in the case of a channel language? Is there no way to inherit? |
| 15:49.47 | Dimitripietro | vlad_starkov : Yep, that's what I think |
| 15:51.00 | rmudgett | Qwell: It is not inheritable. You could use a pre-dial handler to override the channel language. |
| 15:54.22 | vlad_starkov | So should I do something like CHANNEL(__language)=en ? |
| 15:55.06 | Qwell | rmudgett: makes sense |
| 15:55.48 | rmudgett | CHANNEL() only works on the one channel |
| 15:56.28 | rmudgett | The only exception would be local channels where it is reflected on the paired channel. |
| 16:02.19 | *** join/#asterisk F2Knight (~F2Knight@70-89-188-5-or.portland.hfc.comcastbusiness.net) |
| 16:04.29 | F2Knight | Q: Trying to get sip trunking setup with a vender (RCN) who uses a Metaswitch... sadly they do not know how to interface with Asterisk, and keep telling me that we need to send the 'hostname' in the invite packets... specifically they want to see their hostname. I have attempted to set the 'fromdomain=xxxx' but this still has no effect and we only send the IP address... any suggestions? |
| 16:04.59 | [TK]D-Fender | F2Knight: Show us the config & the call |
| 16:05.33 | F2Knight | let me get it on pastbin |
| 16:09.13 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
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| 16:13.30 | F2Knight | [TK]D-Fender: http://pastebin.com/WvqYKvMA |
| 16:14.31 | [TK]D-Fender | F2Knight: You aren't showing the dial itself ther and core settings are not involved. |
| 16:14.33 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:14.56 | F2Knight | [TK]D-Fender: you want the dial from the CLI? |
| 16:15.03 | [TK]D-Fender | F2Knight: You have eitehr masked or failed to show anything usefule for this |
| 16:17.00 | F2Knight | [TK]D-Fender: I only masked the account code and public IP. As for the Dial it is a stock dial out of FreePBX |
| 16:17.19 | [TK]D-Fender | F2Knight: We don't see what PEER you're using or any of it's settings |
| 16:17.24 | [TK]D-Fender | You've shown us nothing really... |
| 16:17.31 | vlad_starkov | Not forgotten to mention: thanks for your help!) |
| 16:17.44 | F2Knight | [TK]D-Fender: ahh sorry did not put the peer up let me correct that (doh) |
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| 16:19.12 | F2Knight | [TK]D-Fender: http://pastebin.com/DfdQJWrf |
| 16:19.19 | F2Knight | that is the peer settings |
| 16:20.09 | F2Knight | I suppose the heart of my question is.. how do I tell asterisk to put a hostname and not the IP address on the INVITE line? |
| 16:20.11 | [TK]D-Fender | If you're going to mask everything, I'm going to assume youa ren't even using the peers whose nameless settings you're showing us |
| 16:21.53 | F2Knight | [TK]D-Fender: The system is connected directly to the internet, as such I am required to mask that information. |
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| 16:54.01 | catphish | what would cause asterisk to generate SIP invites from anonymous@anonymous.invalid when passing on a SIP call that previously had a valid callerid? |
| 16:59.16 | newtonr | catphish, which SIP channel driver? |
| 16:59.27 | Stefan27 | anyone had to change any sysctl settings to get asterisk to work nicely on fedora/linux system (under high-medium load) ? say net.core.rmem_max or other parameters? |
| 16:59.42 | newtonr | catphish, I'm not sure I know the answer, but that is a crucial part of the question. |
| 16:59.59 | catphish | newtonr: interestingly i didn't know there was more than one |
| 17:00.56 | newtonr | catphish, chan_pjsip was added in 12 . https://wiki.asterisk.org/wiki/display/AST/New+in+12 |
| 17:01.20 | catphish | oh, this system is 1.8, using chan_sip |
| 17:02.40 | catphish | calls arrive with From: "441225812345" <sip:441225812345@80.93.123.123>;tag=jD18KHHHj5y7D (apologies for the redactions) |
| 17:02.57 | catphish | but are passed on with: From: "Anonymous" <sip:Anonymous@anonymous.invalid>;tag=as346e5f17 |
| 17:03.58 | catphish | it's very odd, this is the only system with the same config that experiences this behaviour, so i assume it somehow relates to some other field in the incoming INVITE |
| 17:04.50 | catphish | i'll have a read of what other SIP headers asterisk uses for caller id |
| 17:05.48 | newtonr | catphish, perhaps the setting of the "callingpres" option in sip.conf? I've honestly not had to mess with it much so I can't remember. |
| 17:06.01 | [TK]D-Fender | F2Knight: ^ |
| 17:11.37 | [TK]D-Fender | catphish: rather |
| 17:13.46 | catphish | i suspect the problem relates to either 1) the "Contact" header, this contains nonsense from this particular upstream or 2) more likely, "trustrpid" is enabled, but the remote provider does not send rpid |
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| 19:08.08 | mic__ | what are you using for generating calls (testing purposes) - but with RTP?| |
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| 19:39.00 | pabelanger | mic__, sipp can |
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| 19:48.22 | bklang | mic__: Sippy Cup can generate easy sipp profiles with RTP for testing: https://mojolingo.github.io/sippy_cup |
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| 20:39.28 | thekkid | join #nagios |
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| 20:54.50 | mic__ | bklang: thanks |
| 20:55.01 | mic__ | pabelanger: I tried sipp with the latest option for RTP, but w/o success. |
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| 21:47.59 | madax | Hi guys i'm using asterisk 11 when i'm doing a call i'd like to know and propagate (i mean during an attended transfer) the callee number |
| 21:48.23 | madax | i see that there is a variable called DIALEDPEERNUMBER but it seems broken |
| 21:48.51 | madax | any ideas how to do that especially the progagation of the info |
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| 21:50.11 | madax | since i have the info as the EXTEN at the start of the call, i need to keep the info during and attended transfer |
| 21:55.52 | WIMPy | That usually happens automagically. |
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| 21:58.14 | madax | WIMPy: You mean it's stored in a asterisk variable name ? |
| 21:58.40 | WIMPy | No it's displayed to the user. |
| 22:03.22 | madax | ah then maybe it's because i'm using atxfer as a command |
| 22:03.56 | madax | the info is not displayed to the user after the attended transfer |
| 22:04.19 | *** join/#asterisk nny (~Scott@174.107.201.51) |
| 22:06.52 | nny | Hi. I have a question that has been asked so many times it hurts. I have one way audio (NON-NAT -> NAT). Here is the sip debug http://pastebin.com/wLWuhWa5 and here is the sip.conf relevant section http://pastebin.com/DsxtqkuB I had nat=yes for the test. This is Asterisk 11.6-cert6 , I understand nat=yes is deprecated (but should still work?) |
| 22:07.05 | nny | firewall is disabled for the test entirely |
| 22:07.21 | nny | and traceroute to the sip client appears correct |
| 22:07.32 | nny | actually I have no audio.. not one way. Sorry |
| 22:11.06 | nny | I see the bad events and some other oddities. Not sure what to make of it. On my 1.8 asterisk the same sip client works ok. Not sure if maybe this is something to do with the relationship between the two, but if I VPN in and test on the local interface it works fine |
| 22:16.48 | nny | testing some things now. May be something I can work out.. I have other similar setups that work so no big deal |
| 22:23.18 | nny | fixed it with directmedia=no. makes sense |
| 22:23.21 | *** part/#asterisk nny (~Scott@174.107.201.51) |
| 22:23.36 | Penguin | ~sipnat |
| 22:23.36 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 22:23.46 | Penguin | I never would have thought. |
| 22:28.06 | WIMPy | madax: Maybe it's not enabled or your phone doesn't support it. |
| 22:28.13 | WIMPy | (or both) |
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| 23:04.57 | *** join/#asterisk bibz2 (bibz2@193-83-135-143.adsl.highway.telekom.at) |
| 23:05.06 | bibz2 | hey there |
| 23:05.53 | bibz2 | my asterisk stopped writing the status of the phones (peer status) into the mysql table. its still reading values like username, password etc but it doesn't update the status anymore.. |
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| 23:34.55 | *** join/#asterisk cj (~cjac@2607:ff08:f5:3a::3) |
| 23:34.57 | cj | hey folks |
| 23:35.03 | cj | have any of you used these phones? |
| 23:35.04 | cj | http://www.digium.com/en/products/ip-phones |
| 23:35.51 | WIMPy | yes |
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| 23:40.14 | cj | did you order them at high quantity? |
| 23:40.35 | cj | and would you advise me to consider them if I knew my PBX was going to be asterisk? |
| 23:40.51 | WIMPy | Nope. I have two for testing. |
| 23:41.06 | WIMPy | Depends on what you intend to do. |
| 23:41.27 | justdave | I was just going to say, I have one of them sitting in front of me, but also, just testing so far, havn't deployed them at scale yet |
| 23:42.32 | WIMPy | The handling of multiple calls is way better than what I've seen on others *IF* you only use one account. |
| 23:42.44 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 23:42.49 | WIMPy | If you have multiple accounts it's at the bad end. |
| 23:44.04 | cj | so if each phone has its own account, it makes it harder? |
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| 23:44.30 | WIMPy | No, per phone. |
| 23:45.50 | cj | oh, that sounds reasonable. one phone per desk. one desk per person. |
| 23:46.44 | WIMPy | I often use multple accounts to differentiate what the caller tried to reach. |
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| 23:54.22 | madax | WIMPy: yes maybe i'm going to check |
| 23:54.42 | madax | anyone from Brasil or anyone no the aligera product ? |
| 23:54.53 | madax | *knows |
| 23:56.15 | justdave | Anyone know of any tips for troubleshooting DTMF issues? I seem to have exhausted what I can find on Google (most seem to talk about mismatched transmission types (sipinfo, rfc2833, inband)) and I've ensured we've got rfc2833 across the board. My IVR menus are dropping digits at random, even if I feature-code into it from a local phone. |
| 23:56.37 | justdave | just started happening in the last couple weeks but I haven't been able to track down anything that actually changed in that time period |
| 23:57.26 | justdave | except for new versions of openssl, guntls, and bash of course :) |
| 23:57.43 | justdave | I'd hardly suspect those of screwing with dtmf though |
| 23:58.33 | justdave | asterisk 1.8.26.1 |
| 23:59.35 | justdave | (which is the newest version available in epel - I know there's newer, but since none of the newer have any security fixes for features that are compiled in it they haven't felt the need to upgrade I guess) |