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| 00:40.24 | qakhan | anyone worked on atcom gsm card with sms? |
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| 01:38.45 | ruben23 | hi guys i have a polycom Ip Phones conencted to my freepbx..wanted to dispplay a name on my ploycome screen..how to do it..? |
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| 02:14.18 | Penguin | ruben23: #freepbx |
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| 04:12.57 | jdzielny | i know this is a bit off topic, but does anyone have any idea what would cause an elastix 2.5 distro to not display any voicemails in the voicemails portion of the GUI? |
| 04:17.17 | Penguin | #freepbx |
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| 04:20.39 | jdzielny | Penguin, I asked there too. No luck :( |
| 04:41.22 | ChannelZ | bites his tongue |
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| 07:54.21 | Onyx47 | Hi all. We're working on a local version of sound files for Asterisk, but have a bit of a problem. In my language there are two plural forms, depending on the number that precedes the word. Specifically, numbers (ending with) 2, 3 and 4 should be followed by one form, while anything larger uses the second one. Is there some mechanism to facilitate this in Asterisk? |
| 07:55.19 | WIMPy | Not sure, but have a look at say.conf. |
| 07:56.24 | Onyx47 | ah, yes, I think I see a section that should do it |
| 07:56.31 | Onyx47 | completely missed that file. cheers :) |
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| 08:20.28 | elfranne | I have some issue with NAT: the call seems to ignore the nat=force_rport,comedia setting, call log (202 calling 203) : ) http://pastebin.com/sPB6FJEu |
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| 08:49.33 | babak | Hi,anybody worked with DAHDIRAS() ? |
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| 09:40.39 | Onyx47 | great... I have an asterisk install that keeps dying with code 127. /usr/sbin/asterisk is nowhere to be found, only /usr/sbin/safe_asterisk, /var/log/asterisk/messages has nothing useful... anyone have any pointers as to what to look for? |
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| 09:55.36 | dkakoti | HI all. I am stuck in a situation I have two PRI first one is g1 and second one is g2,I need a solution where suppose g1 PRI is dead or channel unavailable call will be automatically redirected through g2 |
| 09:56.05 | dkakoti | It is for outbound dial |
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| 10:47.43 | Skintkingle | hello! |
| 10:48.34 | Skintkingle | I wonder if anyone is able to clarify a quick query I have about reading some CLI info correctly. |
| 10:49.52 | wdoekes | ~ask |
| 10:49.52 | 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. |
| 10:50.13 | Skintkingle | good to know. |
| 10:50.17 | Skintkingle | Well, I shall elaborate then! :P |
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| 10:52.21 | Skintkingle | Stop me if i'm incorrect at any point. But my understanding is that, Upon a call connection, either inbound or outbound, I will see an "AGI" command fly through the CLI, with the first parameter being the unique identification of that channel. I.E AGI("SIP/blabla".... Would make "SIP/blabla" the identification. |
| 10:53.14 | Skintkingle | When coming to hanging up, Hangup("",""), The first Parameter in Hangup, that should be the identification of the channel you want to hangup, So the created call from the AGI, "SIP/blabla" should be viewed in the first parameter of the Hangup command? |
| 10:55.12 | Skintkingle | I would state, I am talking about the visual feedback from a CLI from Asterisk. I'm not directly calling either of these commands, Just trying to correctly link the events together. |
| 10:56.41 | Skintkingle | If what i've stated is the case, and correct information, Is there anything else in the "Executing Hangup in new stack" line, that would denote where the hangup came from, or anything else other than the unique identification of the requested channel. |
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| 11:01.05 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.0.0 (2014/10/25), 11.13.1 (2014/10/20), 1.8.31.1 (2014/10/20); Standard: 12.6.1 (2014/10/20); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 11:14.25 | simpleTon` | hi looking for call centre stats application like asternic..... opensource |
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| 11:25.45 | elfranne | I have some issue with NAT: the call seems to ignore the nat=force_rport,comedia setting, call log (202 calling 203) : ) http://pastebin.com/sPB6FJEu |
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| 12:20.29 | Skintkingle | So many questions, so little time. |
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| 12:36.54 | wdoekes | Skintkingle: not "the channel you want to hangup" |
| 12:37.04 | marceloamorim | guys, if you have access on the server where we get asterisk.org/downloads |
| 12:37.09 | marceloamorim | its down |
| 12:37.10 | wdoekes | but "the channel that the hangup application is run on" |
| 12:38.49 | wdoekes | normally, the Executing... would start with what part of the dialplan is being called |
| 12:39.17 | wdoekes | e.g. Executing [ID81571@osvpi_route_account:20] Goto("SIP/... |
| 12:39.56 | wdoekes | dialplan context osvpi_route_account with exten ID81571 on prio 20 calls Goto() |
| 12:40.34 | wdoekes | marceloamorim: http://www.asterisk.org/downloads/ is not down |
| 12:41.24 | wdoekes | marceloamorim: but you're right, both asterisk.org and www.asterisk.org work |
| 12:41.29 | wdoekes | but only the latter has proper links |
| 12:42.23 | Skintkingle | wdoekes: So the first parameter of the "Hangup" would show the channel that hangup occured, Either the local side, or the remote side? And not the call stack as a whole? |
| 12:42.28 | marceloamorim | was fixed =) |
| 12:42.49 | Skintkingle | I have Executing [<extensionnumber>@extensions:1] before the AGI call. |
| 12:42.59 | Skintkingle | and Executing [h@extensions:1] before the Hangup call. |
| 12:43.13 | wdoekes | no, the Hangup application, *not* any other hangup |
| 12:43.20 | wdoekes | either a side of the call hangs up |
| 12:43.38 | wdoekes | or someone (e.g. the dialplan) calls an application named Hangup |
| 12:44.35 | wdoekes | a call to Hangup() in the h extension seems superfluous |
| 12:44.43 | wdoekes | you're in h because the call got hung up |
| 12:44.49 | Skintkingle | right. I see. So the information in the square braces has caller information, I.E the source of the particular call. |
| 12:45.01 | Skintkingle | when I say call ,I mean call to the function. Not phone call. |
| 12:45.40 | wdoekes | yes |
| 12:46.00 | wdoekes | open up extensions.conf and find the [extensions] context |
| 12:47.44 | Skintkingle | yessir. i'm there. |
| 12:48.03 | wdoekes | there you should see the AGI call and the Hangup call |
| 12:50.35 | Skintkingle | I indeed see the AGI for my extension number, and just a global Hangup at the bottom of the stack |
| 12:50.37 | Skintkingle | as thus: |
| 12:50.38 | Skintkingle | exten => h,1,Hangup |
| 12:56.18 | wdoekes | you rarely jump to the h extension yourself. usually the dialplan engine (pbx core) jumps there when the call is hung up (because one side terminated the call) |
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| 12:59.46 | Skintkingle | i see, but you cant tell from the pbx core calling h's hangup where the hangup occured from. Just that the core had a hangup issue to it? |
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| 13:02.04 | Stefan27 | more OS than asterisk but how do I make asterisk wait 1 second between stop and start through 'systemctl restart asterisk' i created a systemd/asterisk.service file which has ExecStart=/usr/sbin/safe_asterisk but it always seems to stop and restart instantly which can cause problems sometimes |
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| 13:05.49 | Skintkingle | For example, this line, for the hangup command, straight out of the CLI: |
| 13:05.52 | Skintkingle | Executing [h@extensions:1] Hangup("SIP/300-00002cc2", "") in new stack |
| 13:06.05 | Skintkingle | There's nothing discriptive in that line that would state if it's local, or remote that caused the hangup to occur. |
| 13:06.16 | Stefan27 | i guess ill have to do something like systemctl stop asterisk && sleep 2 && systemctl start asterisk each time but seems ugly when there is systemctl is supposed to handle restart-sec-timeouts |
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| 13:09.45 | wdoekes | Skintkingle: you'll see a "== Spawn extension ... exited non-zero" before it |
| 13:14.00 | *** join/#asterisk dym (~patrick@unaffiliated/dym) |
| 13:14.14 | dym | okay - im having trouble. i rebooted one of my asterisk vm's and all of the sudden everything stopped working |
| 13:14.17 | dym | asterisk*CLI> core stop now |
| 13:14.20 | dym | No such command 'core stop now' (type 'core show help core' for other possible commands) |
| 13:14.24 | dym | sip commands are unknown too |
| 13:14.34 | dym | this is 11.12.0 |
| 13:14.38 | dym | and idea? |
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| 13:18.30 | dym | funny thing: console shows chan_sip warnings |
| 13:18.31 | dym | [Nov 3 15:15:33] WARNING[1166]: chan_sip.c:31287 display_nat_warning: !!! PLEASE NOTE: Setting 'nat' for a peer/user that differs from the global setting can make |
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| 13:21.33 | WIMPy | Stefan27: So why not just change the script? |
| 13:22.06 | Stefan27 | safe_asterisk? i guess i can, but thats overwritten with every new version of asterisk |
| 13:24.04 | WIMPy | Copy it as my_asterisk and use that? |
| 13:24.38 | WIMPy | And just out of interest, what kind of trouble do you experience by it restarting immediately? |
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| 13:25.30 | Stefan27 | Some modules dont load |
| 13:25.52 | Stefan27 | it doesnt happen all the time |
| 13:27.26 | Stefan27 | it often happens when i install a new asterisk while the older version is still running and then do restart |
| 13:27.57 | WIMPy | Always the same modules? |
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| 13:28.22 | Stefan27 | I think so, same pattern in the "module show" output but i havent verified 100% its the same |
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| 13:28.38 | agaran | Hello, does anyone know if ChanIsAvail() is supposed to work on SIP channels too? |
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| 13:29.21 | WIMPy | Might be a good idea to do so. I'd consider that a bug, if it doesn'r restart propperly. |
| 13:31.03 | Stefan27 | yeah ill investigate more whether its my fault or not |
| 13:32.03 | Skintkingle | hi wdoekes, sorry went afk. yes i see that line, what discriptive info should i take from it? |
| 13:35.38 | aruntomar | anybody using sipjs +webrtc+asterisk ? how is your experience? |
| 13:42.24 | Skintkingle | for EG: the Spawn Extension line for that example call earlier, shows like this: |
| 13:42.27 | Skintkingle | Spawn extension (extensions, <external number>, 1) exited non-zero on 'SIP/300-00002cc2' |
| 13:42.29 | Skintkingle | Executing [h@extensions:1] Hangup("SIP/300-00002cc2", "") in new stack |
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| 15:06.27 | workingcats | sorry for the stupid question, but where do i find the reference manual of all the asterisk options? wanting to use call-limit but i want to check if it's still present in *-11 as it's deprecated in *-1.8 |
| 15:06.51 | mjordan | workingcats: check the sip.conf.sample for that version of Asterisk. |
| 15:36.57 | *** join/#asterisk infobot (ibot@rikers.org) |
| 15:36.57 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.0.0 (2014/10/25), 11.13.1 (2014/10/20), 1.8.31.1 (2014/10/20); Standard: 12.6.1 (2014/10/20); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 15:37.35 | [TK]D-Fender | net_: rrmemory goes by oldest first as sorted by priority |
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| 16:29.07 | *** join/#asterisk atom0 (615bcbf9@gateway/web/cgi-irc/kiwiirc.com/ip.97.91.203.249) |
| 16:29.40 | atom0 | hello. anyone here atm? |
| 16:29.53 | file | maybe |
| 16:30.54 | atom0 | i'm looking for someone to hire to help me sort out an asterisk problem that's most likely extremely simple but i'm far too new and sucky at asterisk to fix lol |
| 16:35.56 | *** join/#asterisk areski (~areski@cust114-dsl56.idnet.net) |
| 16:37.33 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-nnkivmuozfmapsbb) |
| 16:39.30 | teknoprep | why hire someone |
| 16:39.33 | teknoprep | just tell us what the problem is |
| 16:39.42 | teknoprep | whats the problem... don't ask to ask |
| 16:46.14 | atom0 | i have a website that places calls via preset extensions and ami http "originate" command, one sip works flawlessly the other one has recently developed an issue and they say it's something on my end. |
| 16:46.14 | atom0 | the issue is either the call will send out immediately but will take forever to dial the phone in question, and then only rings for 1 second and stops |
| 16:46.14 | atom0 | OR |
| 16:46.14 | atom0 | it will say the call was answered within a second and start the call recording process, but the phone never rings, and the recording is simply a ringing sound. |
| 16:48.03 | *** join/#asterisk areski (~areski@cust114-dsl56.idnet.net) |
| 16:49.39 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 16:54.35 | *** join/#asterisk dym (~patrick@unaffiliated/dym) |
| 16:55.31 | dym | Hey again! For some reason my asterisk 11.12.0 has forgot about all sip commands, although i can see sip stuff going on (registration tries, etc) |
| 16:59.38 | Penguin | module show like sip |
| 17:01.39 | file | *sings* what's SIP got to do... got to do with it |
| 17:01.42 | *** join/#asterisk FreezeS (~FreezeS@ip4-89-238-223-70.euroweb.ro) |
| 17:08.28 | FreezeS | Hi guys! I'm running 11.7.0~dfsg-1ubuntu1 and cannot change astagidir. I've changed it in asterisk.conf but no effect. Checked the start script in init.d and it's nothing there. What can I do ? |
| 17:09.04 | [TK]D-Fender | Show us |
| 17:10.05 | *** part/#asterisk Sjors (~sgielen@2a01:4f8:202:63f2:f00::1) |
| 17:13.17 | *** join/#asterisk jsarrel (~jsarrells@24-158-61-198.static.hckr.nc.charter.com) |
| 17:19.49 | marceloamorim | there is any way to fix the loader.c: Module 'func_realtime.so' and 'res_realtime.so' when the log keeping show me that 2 modules already exists. |
| 17:27.59 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 17:28.54 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
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| 17:34.23 | FreezeS | [TK]D-Fender: http://pastebin.com/mA0FLxnw |
| 17:42.43 | rmudgett | FreezeS: [directories](!) is declared as a template in the samples file. Since nothing uses the template it is effectively a comment. Just remove the (!) on the end of the line. |
| 17:44.26 | FreezeS | aha, I was thinking about that... |
| 17:44.28 | [TK]D-Fender | yup |
| 17:44.51 | FreezeS | but I thought it might mean something else |
| 17:44.52 | FreezeS | thanks :) |
| 17:45.15 | Penguin | I think you can also define some of those values via /etc/sysconfig/asterisk. |
| 17:45.34 | Penguin | You'd have to look at the init script to see what values are usable, though. |
| 18:00.48 | *** join/#asterisk matt_ (~matt@ccpc-mwr.bath.ac.uk) |
| 18:01.45 | eppigy | hello |
| 18:01.47 | eppigy | i am dave |
| 18:01.50 | [TK]D-Fender | you are dave |
| 18:02.46 | eppigy | yes |
| 18:02.51 | eppigy | hello [TK]D-Fender |
| 18:02.56 | [TK]D-Fender | y0 |
| 18:03.12 | eppigy | I am finally learning python |
| 18:03.21 | eppigy | I should have done so like 6 years ago |
| 18:06.07 | [TK]D-Fender | It's a language... |
| 18:06.22 | eppigy | ur a lnaguage |
| 18:19.32 | *** join/#asterisk MadHatter42 (~tuwid@92.60.24.147) |
| 18:25.42 | marceloamorim | I found the problem, the problem is the sorcery.conf when you remove the [res_pjsip] about that loader.c |
| 18:40.51 | *** join/#asterisk miruoy (~quassel@d51a4ceb0.access.telenet.be) |
| 18:50.55 | dym | loader.c:1046 load_resource_list: *** Failed to load module chan_sip.so <--- i dont get it. can anyone suggest some debugging? |
| 18:51.00 | dym | the module is there |
| 18:51.44 | dym | wtf |
| 18:51.44 | dym | okay |
| 18:51.57 | dym | now that i removed the require, it started and sip commands work |
| 18:53.32 | newtonr | dym, if you debug it further, turn on the DEBUG channel and turn it up to 5 or above, then look at what Asterisk says around the module load failure |
| 18:54.13 | dym | newtonr: this was spat out on asterisk -vvvvv |
| 18:54.27 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 18:54.48 | dym | thanks! Ill look into it |
| 18:55.28 | newtonr | Yeah you would want to verify the debug logger channel is going to a file , be sure Asterisk starts with debug at 5 or above and then start Asterisk, stop asterisk and go search for file for where it fails. |
| 18:55.47 | newtonr | That wiki page linked and this one https://wiki.asterisk.org/wiki/display/AST/Logging should help. |
| 18:56.11 | newtonr | Just having a require shouldn't cause chan_sip.so to fail to load. |
| 18:59.49 | *** join/#asterisk theron (~theron@199.201.65.135) |
| 19:35.15 | *** join/#asterisk jhlavacek (~jirka@jix.nextradsl.cz) |
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| 20:05.42 | *** join/#asterisk windback (~jkleinerm@190.123.122.162) |
| 20:09.20 | windback | I'm receiving the following ISDN SETUP message on my asterisk server: http://pastebin.com/JW7W1bbX |
| 20:09.38 | windback | As you can see, I'm not receiving the calling party part |
| 20:11.08 | windback | I'm trying to write an specific callerid using SET(CALLERID(num)=XXXXX) and SET(CALLERID(name)=myName) |
| 20:11.53 | windback | As the trunk doesnt have the calling party part, I have not success |
| 20:12.23 | windback | I try the same in a DAHDI trunk which send the calling party in setup message without problems. |
| 20:12.42 | windback | Do you know a way to have write the caller id? |
| 20:13.26 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 20:16.17 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-fpgmfopghkljzmbm) |
| 20:16.37 | [TK]D-Fender | windback: Show the complete call |
| 20:19.50 | windback | [TK]D-Fender, http://pastebin.com/smQPmyni |
| 20:21.18 | [TK]D-Fender | windback: I don't see any of those SET commands in there |
| 20:24.05 | *** join/#asterisk Cuzner (~ccuzner@198.41.29.45) |
| 20:26.20 | windback | [TK]D-Fender, http://pastebin.com/ui6WiE3u |
| 20:27.26 | windback | [TK]D-Fender, I was trying with CONNECTEDLINE functino |
| 20:29.08 | Cuzner | Am I on a snipe hunt looking for European VSCs? I know they're generally carrier specific, and NANP we have ours because of AT&T, but did no one standardize something similar on telco equipment in europe, or did everyone just do their own thing so now there is no standard there? |
| 20:30.22 | Cuzner | i can find mention of UKs 1571, but not much else... |
| 20:31.32 | Cuzner | oh man... |
| 20:31.33 | Cuzner | http://en.wikipedia.org/wiki/Last-call_return |
| 20:31.43 | Cuzner | nevermind, this is going to take me a while i guess... |
| 20:31.58 | Cuzner | looking up each feature code one by one :P |
| 20:32.42 | *** join/#asterisk e4voip (uid13742@gateway/web/irccloud.com/x-lzlpxgvpwrwtrbin) |
| 20:54.16 | *** join/#asterisk giesen_ (~ggiesen@2001:4900:2:650::180) |
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| 21:42.29 | WIMPy | Can anyone tell me what 'ooh323_onReceivedSetup: Unacceptable ip <ip>' actually means? |
| 21:48.57 | mjordan | WIMPy: may (Alexander Anikin) would be your guy. You might want to ping the asterisk-dev list, as I don't see him online in either room (he is in Russia however) |
| 21:51.02 | ChannelZ-Wk | Browsing the source, it looks call-limity |
| 21:52.00 | WIMPy | It's obviousely someone doing some scanning, but the message itself doesn;t make much sense to me. |
| 22:00.22 | WIMPy | Debug isn't very interesting. |
| 22:01.09 | *** join/#asterisk pjensen00 (~per@ip-64-21-247-189.far.ideaone.net) |
| 22:02.52 | pjensen00 | I've been using asterisk 12 but have just tried to upgrade to 13. I can't seem to get asterisk to start, and I'm wondering which of these errors are causing A13 to not start |
| 22:02.53 | pjensen00 | http://pastebin.com/wRimU86s |
| 22:03.25 | pjensen00 | I see it's angry with my pjsip.conf entry for 'voiptrunk' and various modules |
| 22:04.46 | WIMPy | Possibly the one after the last you see. Add some verbose and debug. |
| 22:05.34 | pjensen00 | oh, I removed all entries of my pjsip.conf and it gets rid of those error messages, but it still dies after all the warnings for my dialplans. |
| 22:05.50 | pjensen00 | I'll add some more verbosity to the logging |
| 22:11.00 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 22:16.30 | pjensen00 | http://pastebin.com/LNVujBNm holy cow that's a lot more verbose. |
| 22:17.25 | pjensen00 | The fact that it seems to be choking on some pretty basic modules like res_agi worries me greatly as to what I possibly could have done on this install |
| 22:19.07 | WIMPy | As it again ends with corosync, maybe that was already it and not whatver would follow. |
| 22:23.26 | pjensen00 | ............ I'm an idiot. I thought I added it to my list of no_loads. You're right, it works now. |
| 22:24.54 | ChannelZ-Wk | Seems like you have every possible config file in your dir |
| 22:39.45 | *** join/#asterisk Micc (~Micc@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 22:40.05 | Micc | Is there an irc channel for help with digium phones? |
| 22:41.31 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 22:51.18 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.50) |
| 22:53.21 | *** join/#asterisk jkister (~jkister@pool-96-227-239-254.phlapa.fios.verizon.net) |
| 22:53.43 | jkister | anyone from digium around? |
| 22:54.05 | mjordan | yes? |
| 22:54.10 | jkister | if you click 'Downloads' from www.asterisk.org, it hrefs to http://www.asterisk.org/downloads which i get 404 |
| 22:54.13 | jkister | fyi |
| 22:54.16 | mjordan | thanks! |
| 22:54.35 | mjordan | We did know, and the hamsters are furiously running in their wheels trying to get it fixed |
| 22:54.48 | jkister | oh, ok. sorry for the noise. i checked the mailing list and didnt see anything. |
| 22:54.53 | mjordan | but you're the first person to tell us publicly :-) |
| 22:54.59 | mjordan | Nope! Definitely appreciate you telling us |
| 22:55.10 | mjordan | Noise of "this is broken" is always good |
| 22:55.36 | mjordan | particularly when it's the website ;-) |
| 22:56.16 | mjordan | urp, I was wrong, bklang hit that same page earlier |
| 22:56.32 | mjordan | (There's a few others that are missing too - it's like an easter egg hunt, only you get a 404 instead of a tasty treat) |
| 22:56.40 | jkister | haha |
| 23:04.42 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-lspaymzedqtubyyw) |
| 23:08.22 | file | a few others hit it EARLY this morning as well |
| 23:13.12 | WIMPy | Couldn't be much earlier than now :-) |
| 23:52.03 | *** join/#asterisk areski (~areski@cust114-dsl56.idnet.net) |