| 00:02.13 | *** part/#asterisk socomm (~socomm@96-40-131-63.dhcp.mtpk.ca.charter.com) |
| 00:47.14 | *** join/#asterisk epinky (~epinky@unaffiliated/trismegisto) |
| 00:51.54 | epinky | Hello, I'm very new to Asterisk, the person in charge has set up an schema I'm trying to understand, we have an 8 port E1 card installed in our CAT6500, as I understand we use Asterisk as a PBX for every building we have, but all the traffic(calls) must go first to Call Manager, then to the E1 to get to PSTN or between Asterisk PBXs. How this schema is called? is there any recipe to make... |
| 00:51.55 | epinky | ...this Asterisk integrate with CallManager? |
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| 01:32.03 | WIMPy | We don't know anything about your setup other than what you tell us. |
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| 02:22.18 | gradyo | hello. |
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| 02:22.32 | gradyo | looking into some yealink phones |
| 02:23.06 | WIMPy | What's inside? |
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| 02:26.23 | gradyo | playing with 3cx. and asterisk. |
| 02:26.35 | gradyo | trying to find a pbx that will work for me |
| 02:33.53 | mjordan | volga629: please don't PM me. It isn't a bug as you aren't specifying a resource for your buddy - which is what the error message is telling you. |
| 02:33.56 | mjordan | or rather, NOTICE message. |
| 02:34.10 | mjordan | If you need more assistance in how to set that up or looking at it further, you should e-mail the asterisk-users mailing list. |
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| 02:35.23 | volga629 | buddy=betta@networklab.ca/nl-status02 |
| 02:36.18 | volga629 | [Arguments] |
| 02:36.20 | volga629 | account |
| 02:36.22 | volga629 | <PROTECTED> |
| 02:36.24 | volga629 | jid |
| 02:36.26 | volga629 | <PROTECTED> |
| 02:36.28 | volga629 | <PROTECTED> |
| 02:36.51 | volga629 | this from core show function JABBER_STATUS |
| 02:40.40 | mjordan | xmpp is not a valid resource for that buddy |
| 02:40.47 | mjordan | exten => s,n,Set(JID_TO=${DB(AMPUSER/${DEXTEN}/xmpp)}) |
| 02:40.59 | mjordan | since it isn't found for that buddy, you don't get the status |
| 02:41.49 | volga629 | this is entry in ASTdb which pull jid |
| 02:42.12 | volga629 | <PROTECTED> |
| 02:42.33 | mjordan | that isn't what your dialplan is doing |
| 02:45.51 | volga629 | I am sure that dial plan is not in question |
| 02:45.53 | volga629 | http://fpaste.org/83871/19518139/ |
| 02:47.24 | volga629 | xmpp01*CLI> database showkey AMPUSER/102/xmpp |
| 02:47.26 | volga629 | /AMPUSER/102/xmpp : yuliia@networklab.ca |
| 02:47.28 | volga629 | 1 results found. |
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| 03:09.00 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 03:09.55 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 03:54.26 | volga629 | Matt, in function man it says that possible use without resource |
| 03:55.21 | volga629 | and in really must have use resource |
| 03:55.30 | volga629 | reality |
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| 04:42.37 | snadge | you should be able to match non numeric chars with a dial plan yeah? |
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| 04:52.40 | ChannelZ | yes |
| 04:55.43 | snadge | i think i may have run into a freepbx issue then ;) .. its documentation doesn't mention it.. matching non numeric chars in a dial plan, seems to be an unusual thing |
| 04:59.34 | snadge | collegue is writing a plugin which will be used for call screening on pbxes |
| 04:59.53 | snadge | and it returns "ok-"number (without the quotes) |
| 05:06.20 | snadge | blacklists based on dialled number or ip address |
| 05:45.58 | volga629 | in dial how to do or in GotoIf |
| 05:46.00 | volga629 | GotoIf($[$["${DIALSTATUS}" = "ANSWER"] | $["${DIALSTATUS}"="NOANSWER"]]?notifyinanswer:notifyinnoswer) |
| 05:50.39 | Penguin | Clarify what you are wanting to achieve, not how you intend to get there. |
| 05:59.17 | volga629 | I trying if statement if dial status answer or no anwser. |
| 06:00.51 | volga629 | based on if statement answer jump not notification |
| 06:00.55 | Penguin | You want to test if it is ANSWER first, and then test if it is NOANSWER if it wasn't ANSWER? |
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| 06:03.16 | volga629 | close if ANSWER send notification with one template and if NOASNWER then send another notification, but different template |
| 06:04.22 | Penguin | GotoIf($["${DIALSTATUS}" = "ANSWER"]?notifyanswer); |
| 06:04.22 | Penguin | GotoIf($["${DIALSTATUS}" = "NOANSER]?notifynoanswer); |
| 06:05.17 | Penguin | Oops. Missed a quote. |
| 06:05.21 | Penguin | GotoIf($["${DIALSTATUS}" = "NOANSER"]?notifynoanswer); |
| 06:05.39 | Penguin | GotoIf($["${DIALSTATUS}" = "NOANSWER"]?notifynoanswer); |
| 06:05.42 | Penguin | and misspelled it. |
| 06:06.15 | Penguin | So you first test to see if it is ANSWER. If it is, send it to the notifyanswer label. |
| 06:06.32 | Penguin | If it is not ANSWER, move on to the next line of dialplan. |
| 06:06.54 | Penguin | Second test to see if it is NOANSWER. If it is, send it to the notifynoanswer label. |
| 06:07.04 | Penguin | If it is not NOANSWER, move on to the next line of dialplan. |
| 06:07.09 | volga629 | yes, first label evaluate true |
| 06:07.31 | volga629 | I did |
| 06:07.33 | volga629 | exten => s,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?notifyinnoanswer:notifyinaswer) |
| 06:07.35 | volga629 | exten => s,n(unavailable),NoOp(XMPP Server reported user status: ${STATUS}. No notification will be send.) |
| 06:07.37 | volga629 | exten => s,n(notifyinanswer),NoOp(Trying send XMPP message on inbound call) |
| 06:07.50 | volga629 | if statement need separate |
| 06:08.03 | volga629 | ok that make sense thank you |
| 06:08.53 | Penguin | Your GotoIf() is only checking to see if it is NOANSWER. If it is NOANSWER, you are sending it to notifyinnoanswer. If it is not NOANSWER, you are assuming it was answered and send it to notifyinanswer. |
| 06:09.21 | Penguin | But there are many other dial status besides ANSWER and NOANSWER. |
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| 06:09.40 | Penguin | I would explicitly test for those two status if those are the ones you care about. |
| 06:10.30 | volga629 | yes only 2 VMSTATE will be different story |
| 06:17.56 | Penguin | Good luck with your project. I'm off to bed now. |
| 06:19.58 | volga629 | thank again |
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| 06:20.09 | volga629 | and I tested project working |
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| 06:39.10 | D30 | hi all, good day,... what does this actually mean? " Asked to transmit frame type ulaw, while native formats is (g729) read/write = ulaw/ulaw " |
| 06:39.50 | D30 | when trying to call for a number outside, i can see lots of WARNING message from the cli ^ |
| 06:40.24 | MaliutaLap | someone wants you to use g729? |
| 06:43.11 | *** part/#asterisk bpietro (~pietro@82.51.236.132) |
| 06:49.49 | D30 | hmmn i dont really know MaliutaLap |
| 06:50.02 | D30 | but some numbers i dialed, i dont see any warnings |
| 07:00.11 | MaliutaLap | are they going over different trunks? |
| 07:01.53 | D30 | theyre on the same trunk MaliutaLap |
| 07:16.59 | MaliutaLap | Odd then |
| 07:17.47 | volga629 | I am trying execute bash script from dial plan |
| 07:17.55 | volga629 | and this way not working |
| 07:17.57 | volga629 | exten => s,n,Set(RESOURCE=${SHELL(${ASTAGIDIR}AstPostBird ${JID_TO})}) |
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| 08:01.34 | D30 | hmmn ive tested several numbers to call but it only appears when calling 212 areacode in US :( |
| 08:01.49 | D30 | still there's [2014-03-10 03:58:26] WARNING[4786][C-00000003]: chan_sip.c:7296 sip_write: Asked to transmit frame type ulaw, while native formats is (g729) read/write = ulaw/ulaw |
| 08:02.00 | D30 | kinda weird.. |
| 08:03.24 | MaliutaLap | Something to do with your ITSP? |
| 08:05.28 | wdoekes | D30: sounds like g729 was negiotiated, yet it wasn't. do you have g729 codecs? what does the 'core show channel X' say about the format? |
| 08:06.04 | D30 | wdoekes: unfortunately dont have a g729 codecs installed :( |
| 08:06.31 | D30 | whats "X" in core show channel? |
| 08:06.32 | wdoekes | and the sdp negotiation selected ulaw? |
| 08:06.36 | wdoekes | the channel name :) |
| 08:06.42 | wdoekes | use tab completion |
| 08:07.01 | D30 | ahhh |
| 08:07.31 | D30 | yes wdoekes ulaw is used |
| 08:07.52 | D30 | ohh wait |
| 08:07.55 | D30 | allow=ulaw&g729 |
| 08:08.06 | D30 | do i need to remove g729? |
| 08:08.24 | D30 | i actually copied it from our provider |
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| 08:09.06 | D30 | MaliutaLap: you mean the sip provider? |
| 08:10.22 | wdoekes | D30: if you have no g729 codecs, you should not allow= it |
| 08:10.38 | wdoekes | disallow=all, allow=ulaw |
| 08:11.38 | D30 | hmmn okay will test it... thanks wdoekes |
| 08:13.30 | *** part/#asterisk jsjc (~Adium@117.Red-81-47-171.staticIP.rima-tde.net) |
| 08:14.37 | D30 | hmm wdoekes it works, :) no more warnings related to g729 |
| 08:15.06 | D30 | but when it stop ringing, i heard nothing :( does it mean i need g729 for it? |
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| 08:40.55 | emaberga81 | Hi guys, when I add a new register string and I then do a "sip reload" asterisk send a new REGISTER for all the register strings in my sip.conf file. If I am in a call and asterisk send a new REGISTER for my number then the proxy sends me a BYE and the call is dropped. Is there some way to send REGISTER only for new register strings after a sip reload? |
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| 09:38.56 | IMS77 | Hi, in my dialplan i have a macro with a line "exten => s,n,Wait(20)" to wait 20 sec (it's a test) => after around 15 sec the macro exit. Does someone knows which parameter (in a file .conf) to change to be able to wait 20 sec. Thanks by advance for any help |
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| 09:46.18 | IMS77 | When i call wait() is it using a queue => if so the parameters should be in queues.conf, non ? |
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| 10:53.19 | IMS77 | I'm still trying to wait 20 sec in my dialplan => wait(20) does no work but if i call progress() before to call wait(20) it's working. When I read the doc about progress, the function is used for early media => so what is the difference between the two states so in the 1st state i can't wait 20 seconds ! I tryed to change many timeout but without success. |
| 10:57.49 | IMS77 | No help ? |
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| 11:06.53 | gryffus | hi folks. Is it possible to use tls encryption without transfering client certificates to clients? Thanks for any help |
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| 12:12.08 | kalib | Hello guys, I´m receiving this message on my CLI: chan_sip.c:12723 handle_response_peerpoke Peer '8551' is now Reachable. (53ms / 2000ms) |
| 12:12.18 | kalib | Couldn´t find anything about this on google. Is this an error? |
| 12:13.25 | MaliutaLap | no, it just means that you had a SIP timeout on a link |
| 12:13.32 | MaliutaLap | normally a network issue |
| 12:15.43 | kalib | thanks |
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| 12:44.25 | gryffus | I'm trying to configure tls-only asterisk. I have created certificates like in this howto: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial , but when i try to connect from an Android SIP client (linphone), i get "tcptls.c: == Problem setting up ssl connection: error:14094418:lib(20):SSL3_READ_BYTES:tlsv1 alert unknown ca" and "tcptls.c: FILE * open failed!" errors in asterisk log file... What i did wrong? According to linphone tls |
| 12:44.26 | gryffus | howto, i have to import rootca.pem file to my android device, but there is no rootca.pem file in /etc/asterisk/keys ... Am i missing something? Is there no possibility to setup tls with certificates only on server, just like in HTTPS? |
| 12:45.34 | gryffus | as you can see, i don't have many experiences with tls, so any information is valuable to me... Thanks |
| 12:46.47 | emaberga81 | Hi guys, when I add a new register string and I then do a "sip reload" asterisk send a new REGISTER for all the register strings in my sip.conf file. If I am in a call and asterisk send a new REGISTER for my number then the proxy sends me a BYE and the call is dropped. Is there some way to send REGISTER only for new register strings after a sip reload? |
| 12:47.03 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 12:49.00 | gryffus | note that these are self-signed certificates... Is this even possible? |
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| 12:52.20 | WIMPy | emaberga81: No, and it will also send a REGISTER in the regulat intervalls anyway. Thare is no relation to ongoing calls. |
| 12:55.01 | emaberga81 | hi WIMPy so you think it is a problem on the sip proxy? Because as soon as it gets the REGISTER for the numbers that are in a call it sends a BYE |
| 12:56.03 | WIMPy | There is something going very wrong somewhere, yes. |
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| 13:05.42 | gryffus | anyone here with working tls sip setup with self-signed certs? |
| 13:06.55 | emaberga81 | ok, thanks WIMPy! The sip proxy is from another vendor so I will have to check with their support |
| 13:07.09 | emaberga81 | but a question: is it against some RFC to send a REGISTER while in a call? |
| 13:08.35 | WIMPy | Which of the many RFCs concerning SIP? |
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| 13:42.00 | emaberga81 | WMIPy I did some trace and I see that when Asterisk send the REGISTER because the timer is expired then the call does not drop |
| 13:42.25 | emaberga81 | the only difference between this REGISTER and the one that Asterisk send after a sip reload is that the first one already has a Authorization header |
| 13:42.39 | emaberga81 | after the sip reload it does not have the auth header |
| 13:42.45 | tuxx- | hiya guys. any asterisk developers here that can point me to the place where the SIP headers (from sipaddheader) are stored? in the ast_channel or sip_pvt struct somewhere? |
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| 13:43.20 | WIMPy | tuxx-: That looks like a question for #asterisk-dev |
| 13:43.28 | tuxx- | yeah, just thought about that. ill go bug the people there :-) |
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| 14:14.04 | banane_ | can anyone help with configuring or debugging dahdi/pri for a german landline? i´m using an openvox card and currently only get a yellow alarm on the active pri spans |
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| 14:52.42 | Chainsaw | Digium K-lined from Freenode. Now I've seen everything. |
| 14:52.54 | file | yeah ummm yeah |
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| 14:53.51 | Chainsaw | For what it's worth, they'll be missed. |
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| 15:03.36 | zoso | Hi, I think I am facing a issue, some of the calls I am receiving, are there in the sip table for a long time with the "last message" column has the value Rx:BYE, I have found out that the user has hung up the call by then. What is going wrong? |
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| 15:07.23 | ghost75 | sip table? |
| 15:08.34 | leifmadsen | I suspect he means, 'sip show channels' |
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| 15:09.01 | PbxMan | Afternoon |
| 15:11.23 | banane_ | can anyone help with configuring or debugging dahdi/pri for a german landline? i´m using an openvox card and currently only get a yellow alarm on the active pri spans |
| 15:16.47 | ghost75 | last message bye shouldnt be so annormal |
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| 15:20.40 | Katty | leifmadsen: o/ |
| 15:20.51 | leifmadsen | Katty: ohai |
| 15:21.05 | zoso | ghost75, but it is there, and the channel is not processing calls any more |
| 15:21.12 | Katty | leifmadsen: how'rechu? |
| 15:21.30 | zoso | and what does a status of 'no answer' mean in the cdr table that asterisk logs to |
| 15:22.45 | ghost75 | if nobody picks up on remote side |
| 15:22.56 | ghost75 | or local .. |
| 15:23.52 | zoso | ghost75, by local you mean the channel? |
| 15:23.53 | ghost75 | zoso: u may want to do sip debug to see whats going on or use tcpdump |
| 15:24.22 | ghost75 | if u call somebody and he doesnt pick up, its no answer |
| 15:24.47 | zoso | ghost75, I am using sip debug, I think I should read something to understand what to infer from values I am seeing on debug |
| 15:25.18 | ghost75 | in wireshark its a bit easier to read |
| 15:26.23 | zoso | ghost75, I see, and how can I know why a call is still in the active list with rx: bye as last message. |
| 15:27.20 | ghost75 | see here http://www.youtube.com/watch?v=OFpQLyQxt84 |
| 15:28.34 | ghost75 | also good http://www.voipmonitor.org/ |
| 15:28.54 | zoso | thanks. How can I kill a call? |
| 15:30.07 | ghost75 | channel request hangup |
| 15:31.30 | leifmadsen | Katty: oh not bad... I'm cleaning up some jira stuff :\ |
| 15:31.38 | leifmadsen | and incredibly tired after the weekend summit |
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| 15:46.53 | Katty | nods |
| 15:47.02 | Katty | leifmadsen: you'll have to catch up on sleep soon :> |
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| 15:52.32 | ghost75 | entire(d) loop? |
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| 17:00.25 | banane_ | can anyone help with configuring or debugging dahdi/pri for a german landline? i´m using an openvox card and currently only get a yellow alarm on the active pri spans, can´t seem to get the line up |
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| 17:24.32 | Chainsaw | banane_: ISDN2e BRI or PRI? |
| 17:27.59 | JeffC_NN | I'm running an AGI after hangup, and I'm stuck on trying to figure out if the call was transferred (and this is the leg that performed the transfer and is now hanging up). I'm thinking either looking for some kind of channel variable, or using AMI to ask for the state of the previously bridged channel. Does anyone have any other ideas? |
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| 18:07.08 | ghost75 | in which unit does asterisk display jitter i got i.e. 0.000352 is that 0,352ms ? |
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| 18:10.17 | ghost75 | mjordan: is asterisk display jitter in seconds ? |
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| 18:33.23 | issackelly | If I use ChanSpy to barge into a channel and start talking (actually running Playback), but the original channel goes into "RECORD" mode, the caller/person being recorded can no longer hear the person barging in. Is there a way around that? |
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| 18:45.22 | issackelly | If I do Monitor() and WaitForNoise then WaitForSilence and then StopMonitor() and only deal with the output channel will that work? |
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| 19:47.04 | JeffC_NN | when you use Dial(), shouldn't both legs of the call go to priority "h" in whatever context they were dialed from? |
| 19:47.19 | JeffC_NN | (after either side hangs up) |
| 19:47.45 | JeffC_NN | err... I guess not |
| 19:50.42 | aamu11 | JeffC_NN: after either side hangs up.... |
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| 20:11.37 | JeffC_NN | I'm having trouble determining who hung up. The Dial() takes place inside an AGI, and I only see priority "h" being hit on the customer's leg (that called the AGI). How do I tell if the agent hung up? |
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| 20:46.08 | ghost75 | is it normal that provider tries to read sip options from asterisk in s exten? |
| 20:49.47 | *** join/#asterisk b11d (~chat@234-200-29-134.hcc.mnscu.edu) |
| 20:49.56 | b11d | hey all |
| 20:50.25 | b11d | so I have a client using an SPA8800 and Asterisk, its working fine... but now they want to support two more lines, and possibly more... so I'm looking at alternatives to the SPA8800... |
| 20:50.30 | b11d | whats recommedned, a channel bank? |
| 20:50.52 | b11d | i'm looking at the Cisco VG224/248 as well, but dunno about that.. |
| 20:50.58 | [TK]D-Fender | [16:46]ghost75is it normal that provider tries to read sip options from asterisk in s exten? <- they don't try to read from there. Asterisk chooses to look there |
| 20:51.19 | b11d | in lieu of adding another SPA8800 and then another SPA8800.. |
| 20:51.49 | b11d | I've looked at the Rhino CBs, but...something feels wrong about them.. cant put my finger on it.. |
| 20:52.21 | b11d | so I guess im looking for a recommendation that isnt gonna bite me in the ass when they get it and it doesnt work right :) |
| 20:53.03 | b11d | PRI, at this point, seems overkill.. |
| 20:59.10 | b11d | i also cant just stick a card in a server and hook the lines right up, the servers they have have no open PCI ports, and the physical distance is an issue... |
| 20:59.27 | b11d | right now the SPA sits next to their demarc and then uses a single ethernet connection back upstairs to their servers.. |
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| 21:19.24 | ghost75 | [TK]D-Fender: is it normal that it goes for dialplan when OPTIONS were requested? |
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| 21:19.39 | [TK]D-Fender | yes |
| 21:19.48 | ghost75 | ok |
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| 21:34.24 | b11d | seriously guys no recommendations? |
| 21:34.25 | b11d | lol.. |
| 21:34.30 | b11d | i figured this'd be the place :) |
| 21:37.27 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 21:38.54 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.8.1 (2014/03/10), 1.8.26.1 (2014/03/10); Standard: Asterisk 12.1.1 (2014/03/10); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 22:21.35 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
| 22:22.48 | dan_j | Hi. If Asterisk is bridging and recording a call, if one side of the audio has constantly intermittent sound, is it safe to say there is an issue with that end of the call? I ask because one client is experiencing this issue, but other clients have no problems. So i can't understand why. |
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| 22:32.52 | dan_j | To be clear, I'm comparing the clients by listing to recordings made by the asterisk box that both clients dial through. The end of the call having problems is a major SIP provider so shouldnt be having problems. |
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| 22:37.34 | bitvilag | hi everyone |
| 22:37.44 | bitvilag | i would need some help with extensions |
| 22:39.55 | [TK]D-Fender | ~ask |
| 22:39.55 | 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. |
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| 22:50.57 | bitvilag | hey well I wasnt sure if there is anyone here so I figured I ask when there are multiple users active. I have installed the latest freepbx asterix software from asterisk.org and well I tried to configure. I manage to connect the extensions but for some reason I cannot call the extensions and the odd part is if I use the number of my cell phone it calls the extension that is paired with my |
| 22:50.58 | bitvilag | cell which is odd because I only mention that number on my freevoipdeal trunk. So i am kind of lost at the moment |
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| 23:40.43 | Development_ | Hey. I got a new grandstream 2160 phone. I got my sip account in it. I have voicemail kind of setup. When I hit the VM button on the phone I get a busy tone and it says "No dial plan rules matched" |
| 23:41.18 | Development_ | How do I get it to go to my voicemail. I didn't know what to do. I can setup an extension that calls voicemail main, but shouldn't the button just go straight to vm? |
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| 23:54.18 | jmetro | I'm trying to do some AMI from a webpage. I'm currently using CURL and PHP to login, but the connection immediately closes [stopping any other actions]. How does keep open? |