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| 02:06.42 | TheZealous | Hi all! I am looking for a latest version installation instruction guide for Elastix. Does anyone have it ? please share it if you have it |
| 02:06.51 | TheZealous | I am using CentOS 6.6 |
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| 03:01.06 | carrar | TheZealous: somethng like this? http://www.slideshare.net/elastixorg/elastix-installation |
| 03:01.42 | carrar | perhaps http://wiki.sangoma.com/Elastix-Installation |
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| 03:15.01 | drmessano | The only problem with an Elastix install guide is that when you're done you have Elastix |
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| 03:22.47 | fling | Are there any similar to CALLERID variables for an ongoing call? Something containing info about DIDs and everything |
| 03:33.08 | drmessano | Anyone here tweaked WIFI settings for VoIP? I'm trying to wrap my head around the Beacon Interval and DTIM and how they relate to UDP latency |
| 03:33.14 | drmessano | (If at all) |
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| 04:22.30 | WIMPy | fling: EXTEN unless/until you change that. And off course te CONNECTEDLINE function. |
| 04:23.04 | WIMPy | drmessano: The beacon is only for detecting the AP. I'd have to look up DTIM. |
| 04:23.37 | fling | WIMPy: I also found DNID |
| 04:24.28 | WIMPy | That one is probably linked to the REDIRECTING function. |
| 04:24.54 | WIMPy | But documentation on all the caller ID stuff is rater thin. |
| 04:27.09 | fling | WIMPy: I want to have a variable with what DID actualy dialed from the start. |
| 04:27.34 | fling | WIMPy: I will use this variable for monitor. Looks like DNID what I want. |
| 04:27.38 | WIMPy | That's a little vague as well. The one you got the call on? |
| 04:28.09 | drmessano | WIMPy, Beacon becomes relevant because DTIM is tied to the Beacon interval.. So something like a DTIM of 3 with a Beacon Interval of 100 means it's 300ms because DTIM transmission.. But the more I dive into this, it seems like DTIM applies to Multicast/Broadcast |
| 04:28.26 | drmessano | s/because/between/ |
| 04:28.50 | fling | WIMPy: umm two dids. first one is 'from', second one is 'to'; I get first one from callerid |
| 04:29.06 | WIMPy | Define "to". |
| 04:29.32 | fling | my did is 100 I call 200. from is 100 and to is 200 :P |
| 04:29.49 | WIMPy | The one you got the call on? Or the one that was actually dialled? |
| 04:29.59 | fling | The one I dialed |
| 04:30.11 | WIMPy | Save EXTEN before you do any Goto() |
| 04:30.32 | fling | What if I will save it to DNID variable? bad idea? |
| 04:30.37 | WIMPy | Then it's the REDIRECTING function or it might be DNID. |
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| 04:36.19 | fling | WIMPy: Don't I edit DNID variable? |
| 04:36.52 | WIMPy | You should read it. Writing it doesn't make much sense. |
| 04:39.33 | fling | WIMPy: but looks like it is just better to save EXTEN to some variable, right? I can create my own like ADDID something. |
| 04:40.08 | fling | I can't decide do I need a new variable or is it sufficient to just use DNID. |
| 04:40.33 | WIMPy | Depends on what you want to know. |
| 04:40.57 | WIMPy | EXTEN can change depending on your dialplan. |
| 04:41.32 | fling | WIMPy: but if I save EXTEN to a variable at the very beginning⦠|
| 04:41.52 | WIMPy | That's what I suggested. |
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| 04:42.59 | WIMPy | But that is the number you received a call for. That's not neccessarily the one the user dialed. |
| 04:43.21 | WIMPy | (but probably what you want) |
| 04:43.40 | WIMPy | And it might even be the only reliable one. |
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| 04:56.39 | fling | WIMPy: thanks, I will use this one. |
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| 05:15.18 | fling | WIMPy: can I also set MONITOR_EXEC as a global variable? |
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| 05:35.07 | fling | WIMPy: This is how it looks now -> http://dpaste.com/11ZTB05 |
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| 05:47.32 | fling | Even simplier now -> http://dpaste.com/3EHJ8M9 |
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| 06:36.11 | fling | Does this look good? -> http://dpaste.com/1GB4AD0 |
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| 07:15.24 | jzu_ | oh yes, now I'm able to access FreePBX Phonebook through Cisco phone's Directory |
| 07:19.51 | jzu_ | aaaand wrong channel once again ! |
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| 07:29.24 | rl1 | Could you please help me debug an issue were Asterisk won't ACK on BYE and hence won't drop the call |
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| 07:32.34 | ChannelZ | Won't ACK from who? A device, an ITSP.. |
| 07:33.21 | rl1 | a SIP gateway if that matters. It sends a BYE to Asterisk but Asterisk doesn't seem to send ACK and then 200 OK and clear the call |
| 07:34.20 | rl1 | the gateway just sends BYE over and over again waiting for an ACK in return |
| 07:35.33 | ChannelZ | And is asterisk saying anything about ignoring bogus SIP messages? Without seeing any SIP debug, a random guess is to set pedantic=no in sip.conf and see if it works; could be the gateway is messing up the call tag and asterisk can't match it to an active call |
| 07:36.03 | rl1 | nah i checked the tags and CallID they all match the initial tags in INVITE |
| 07:36.12 | rl1 | okay give me a min |
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| 07:39.23 | [gnubie] | waves |
| 07:40.04 | [gnubie] | we have a shoretel sg60/12 and because we need more extension, we plan of trunking it with asterisk. is it possible? how do i do that? |
| 07:41.09 | ChannelZ | Yes-ish. I don't know anything about that system specifically, but you can generally bridge an old PBX via T1/etc or if by some chance it speaks SIP already.. |
| 07:41.25 | rl1 | http://pastebin.com/wqmkqKMn okay here's the trace |
| 07:41.28 | ChannelZ | The question is if you can program it in any sensible manner |
| 07:42.48 | [gnubie] | ChannelZ: iâm trying to understand this old shoretel sg-60/12 how i can trunk/extend it to an asterisk box so that the additional extensions will be provisioned by asterisk.. |
| 07:44.24 | ChannelZ | I don't know, as I said I don't know anything about that system. Which I just googled and it's not a system, it's a device... |
| 07:44.41 | ChannelZ | ...which looks like it speaks MGCP which I know even less about |
| 07:46.44 | rl1 | call-id matches, tags match i don't know what's wrong with it =( |
| 07:47.39 | drmessano | [gnubie]: junk it and buy something useful (Read: SIP) |
| 07:48.11 | ChannelZ | well I only see half of the conversation, from the Crisco to Asterisk presumably.. |
| 07:48.40 | ChannelZ | oh wait a min my window somehow came up scrolled |
| 07:48.42 | rl1 | it's the full calltrace |
| 07:49.42 | [gnubie] | drmessano: i wish itâs easy to do that. :D |
| 07:50.13 | drmessano | [gnubie]: You're talking about implementing an extinct piece of equipment |
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| 07:52.02 | rl1 | and there's nothing on debug log about malformed BYE's whatsoever |
| 07:56.31 | ChannelZ | I'm a little confused by a couple things, the Crisco seems to send 'session progress' twice in a row |
| 07:56.47 | rl1 | yeah that confused me too... |
| 07:57.40 | rl1 | they seem to be identical |
| 07:58.27 | ChannelZ | are the packets actually making it to asterisk? This capture was done outside of asterisk |
| 07:58.57 | rl1 | yes. the capture was taken from the machine where asterisk is running |
| 07:59.37 | rl1 | OHWAIT |
| 08:00.01 | ChannelZ | but not sip debug inside asterisk, this looks like some captured/processed elsewhere like wireshark or whatnot |
| 08:02.04 | rl1 | yeah this is a tcpdump capture. |
| 08:02.21 | rl1 | I'm gonna try the sip debug |
| 08:02.33 | ChannelZ | and set verbose to 3 |
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| 08:04.57 | rl1 | stupid me, meh |
| 08:05.17 | rl1 | I forgot to allow the packets in INPUT chain :| |
| 08:05.42 | rl1 | nevermind the problem's solved. ChannelZ thanks for your assistance |
| 08:05.54 | ChannelZ | that's why I wanted to see what asterisk was actually seeing. Glad you figured it out though |
| 08:06.22 | rl1 | that's why the cisco was sending session progress twice |
| 08:06.45 | ChannelZ | among others :) |
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| 08:26.51 | Hydrosine | So i have set stunaddr and turnaddr in rtp.conf to support ICE per device. But when i turn stun debug on i see every call this message(and someothers) : "Dunno what to do with STUN message 0101 (Binding Response)" |
| 08:27.03 | Hydrosine | seems to introduce some load |
| 08:27.20 | Hydrosine | what is the way to only have asterisk do something when icesupport is yes on device? |
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| 09:46.50 | jzu_ | Anyone willing to code Name lookup based on the phone number if I provide API specs? |
| 09:47.26 | jzu_ | OpenCNAM is for USA/Canada only, I'd like to have number-->name lookup form Finnish provider with API |
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| 09:51.58 | [gnubie] | drmessano: thank you.. |
| 09:52.03 | [gnubie] | waves |
| 09:52.05 | [gnubie] | gtg.. |
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| 10:42.28 | Cust0sL1men | hi |
| 10:42.44 | Cust0sL1men | Can asterisk do DCCA to other systems ? |
| 10:43.19 | Cust0sL1men | I want asterisk to process call and use DCCA to another system to charge |
| 10:44.13 | eirirs | what are dcca |
| 10:44.22 | eirirs | no hits in google |
| 10:44.47 | Cust0sL1men | diamter credit control application |
| 10:45.15 | eirirs | thanks |
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| 11:23.43 | Cust0sL1men | https://en.wikipedia.org/wiki/Diameter_Credit-Control_Application |
| 11:23.46 | Cust0sL1men | for reference |
| 11:23.57 | eirirs | just googled it up, thank you |
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| 11:55.23 | Rico | is there a way to "kill" a call launched by a .call file, with retry=3 |
| 11:55.38 | Rico | the call make asterisk crashing (fax sending) |
| 11:55.52 | Rico | Apr 23 13:51:30 centrex3 kernel: asterisk[25933]: segfault at 0 ip (null) sp 00007f4c0432d4a8 error 14 in asterisk[400000+1d7000] |
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| 12:03.34 | eirirs | huh, fax crashing asterisk? thats new to me |
| 12:06.28 | Rico | eirirs: yes, to me too |
| 12:06.46 | Rico | but each time I get a segfault, last asterisk logline is about sendfax |
| 12:07.24 | Rico | I'm looking for a way to debug that |
| 12:07.35 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
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| 12:13.02 | eran322 | help |
| 12:13.07 | eran322 | need help |
| 12:13.21 | [TK]D-Fender | ~ask |
| 12:13.21 | 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. |
| 12:13.23 | [TK]D-Fender | ^ |
| 12:13.40 | eran322 | i using freepbx 12. i have queue with 4 users. when call come in the queue, and no get answered, i see in the cdr database that the call in the queue is answered and the extentions in the queue didnt answer. the problem is that i want with php script to get the list of the numbers that are didnt answered for call back them. is there any way to do that? |
| 12:15.32 | [TK]D-Fender | Queue Log <- |
| 12:18.00 | eran322 | i didnt see the queue log in the database |
| 12:18.35 | eran322 | i see it in the log folder, but i didnt understand how do i use it |
| 12:19.54 | [TK]D-Fender | Most freePBX system log it to a separate database in MySQL |
| 12:20.05 | [TK]D-Fender | not in the the DB that holds FreePBX itself |
| 12:20.13 | [TK]D-Fender | asteriskcdrdb <- |
| 12:23.06 | eran322 | yes, i look it in |
| 12:25.10 | eran322 | but it show that the queue was answered.... i want to know if non answer, but the queue always answer even if the extensions didnt answer\ |
| 12:25.27 | *** join/#asterisk sarthor (~sathor@unaffiliated/sarthor) |
| 12:25.45 | [TK]D-Fender | show us |
| 12:26.01 | [TK]D-Fender | ~pb |
| 12:26.01 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 12:26.03 | [TK]D-Fender | ^^^ |
| 12:26.13 | sarthor | HI I am having this error on my pfsense box. running asterisk on the same machine. " Unable to open Asterisk database '/var/db/asterisk/astdb': No such file or directory ".. any Help please. |
| 12:29.05 | [TK]D-Fender | either the file is corrupt (at which point delete it), or * can't open it with write privs |
| 12:31.08 | *** join/#asterisk wasanzy (~wasanzy@197.159.129.10) |
| 12:31.12 | wasanzy | hello |
| 12:32.34 | eran322 | http://pastebin.com/qJvhazg7 |
| 12:32.52 | wasanzy | for some reason, am doing do configure CDR_Custom to log the source and destination parameters for me when doing OBD calls but unfortunately it is rather logging both source and destination as the same number |
| 12:33.05 | [TK]D-Fender | eran322: that is NOT the queue log. That is CDR. |
| 12:33.19 | [TK]D-Fender | eran322: Forget about CDR completely |
| 12:33.36 | [TK]D-Fender | eran322: Go read the queue log like I told you. |
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| 12:39.00 | wasanzy | here is my config and output: http://pastebin.com/tUb1a338 |
| 12:39.32 | wasanzy | the source is the same as destination. is not logging the phone number the call is being made to |
| 12:41.07 | [TK]D-Fender | wasanzy: No, there is no dst in there |
| 12:41.33 | [TK]D-Fender | wasanzy: Count your columns |
| 12:42.26 | [TK]D-Fender | wasanzy: And you very specifically formatted that field differently than all the others |
| 12:42.39 | [TK]D-Fender | wasanzy: today's magic word : CONSISTENCY |
| 12:44.00 | wasanzy | [TK]D-Fender: I don't understand |
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| 12:44.19 | [TK]D-Fender | wasanzy: Look at your config and COUNT the colums in order |
| 12:44.27 | wasanzy | ${CSV_QUOTE(${CDR(dst)})} |
| 12:44.33 | wasanzy | I have that in there |
| 12:44.45 | wasanzy | is that not correct? |
| 12:45.07 | [TK]D-Fender | Master.csv => ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})},${CDR(dstchannel)},${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration,f)})},${CSV_QUOTE(${CDR(billsec,f)})} |
| 12:45.28 | [TK]D-Fender | wasanzy: One of those does NOT have ${CSV_QUOTE |
| 12:45.28 | eran322 | can i put the queue log in mysql???? i dont know to read files |
| 12:45.46 | [TK]D-Fender | eran322: Have you proved it's not there already? |
| 12:46.01 | [TK]D-Fender | eran322: Show us your database list. |
| 12:46.12 | wasanzy | ${CDR(dstchannel)} |
| 12:46.31 | wasanzy | but is that what prints the destination number? |
| 12:46.46 | [TK]D-Fender | "08099954883","08099954883","start",,"2015-04-23 12:34:31","2015-04-23 12:34:41","2015-04-23 12:36:51","140.932379","130.093101" |
| 12:46.57 | [TK]D-Fender | wasanzy: DST is your 3rd parameter on your config |
| 12:47.09 | [TK]D-Fender | wasanzy: In here the 3rd parameter is NOT the same |
| 12:47.35 | eran322 | datalist: asterisk, asteriskcdrdb |
| 12:47.48 | [TK]D-Fender | eran322: Looks like it's there so far |
| 12:47.50 | wasanzy | yes you are right. does it mean the DST is not set well? |
| 12:48.06 | [TK]D-Fender | wasanzy: I think you should go back to the book.... |
| 12:48.39 | eran322 | where? |
| 12:49.13 | [TK]D-Fender | [08:20][TK]D-Fendernot in the the DB that holds FreePBX itself |
| 12:49.15 | [TK]D-Fender | [08:20][TK]D-Fenderasteriskcdrdb <- |
| 12:49.29 | [TK]D-Fender | eran322: I told you half an hour ago.... |
| 12:49.52 | [TK]D-Fender | oops |
| 12:49.56 | wasanzy | [TK]D-Fender: why should I go back to the book? am not sure that is why am asking |
| 12:50.01 | [TK]D-Fender | soscratch that |
| 12:50.29 | [TK]D-Fender | eran322: dump the table list from both DB's |
| 12:50.50 | eran322 | in the asteriskcdrdb i just have: cdr,cel |
| 12:50.57 | [TK]D-Fender | eran322: Ok, not in there. |
| 12:50.58 | eran322 | both of them didnt help me |
| 12:51.04 | [TK]D-Fender | eran322: PB the other one |
| 12:51.33 | eran322 | what? |
| 12:52.02 | [TK]D-Fender | PASTEBIN the list for the other DB |
| 12:52.43 | eran322 | cel? or the main asterisk db? |
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| 12:53.07 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:54.01 | [TK]D-Fender | eran322: I said to dump the table list for the other DB. |
| 12:54.09 | [TK]D-Fender | eran322: |
| 12:54.23 | [TK]D-Fender | eran322: "cel" is a table, not a database |
| 12:55.00 | eran322 | ok |
| 12:55.23 | wasanzy | I have added this ${CSV_QUOTE(${CDR(dstchannel)})} but the same ouput |
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| 12:59.54 | eran322 | how i dump it to file? |
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| 13:00.09 | eran322 | mysqldump --all-databases |
| 13:00.48 | [TK]D-Fender | eran322: I asked for a TABLE LIST |
| 13:00.54 | [TK]D-Fender | eran322: "show tables;" |
| 13:02.49 | eran322 | http://pastebin.com/LitW1yqw |
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| 13:03.24 | [TK]D-Fender | eran322: yup, no queue long in there |
| 13:03.47 | [TK]D-Fender | eran322: So if you want * to do this "live" you'll have to configure logging to your DB. Go read the book for that. |
| 13:04.26 | [TK]D-Fender | eran322: To just get your current log in there, just convert the text file from "|" as a delimiter to "," and upload the CSV |
| 13:06.53 | eran322 | how can i do that the queue log get in the database? |
| 13:07.03 | eran322 | is there any module that do it? |
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| 13:08.06 | Rico | I have a problem with sendfax : http://pastebin.com/tWwE0MD9 |
| 13:08.14 | Rico | anyone can tell me what's wrong ? |
| 13:10.51 | [TK]D-Fender | eran322: I just told you to change the delimiter so that it's CSV. There are tons of guides showing how to import that into a table |
| 13:10.57 | [TK]D-Fender | ErAnd has nothing to do with Asterisk |
| 13:11.38 | [TK]D-Fender | Rico: [Apr 23 15:04:40] ERROR[11089]: res_fax.c:2121 sendfax_exec: 'modems' setting 'V17,V27,V29' is incompatible with 'minrate' setting 2400 |
| 13:11.53 | [TK]D-Fender | Rico: Pretty clear as to what it's complaining about... |
| 13:12.03 | Rico | minrate is set to 4800 in res_fax.conf... |
| 13:12.14 | Rico | as shown in pastebin |
| 13:13.12 | Rico | if I set "modems=v29" it seems to pass |
| 13:13.30 | Rico | if I set "modems=v29,v27,v17" : same error |
| 13:14.45 | [TK]D-Fender | Rico: And V.27 alone? |
| 13:14.58 | [TK]D-Fender | Rico: I'd enable debug and confirm what the other side is asking for as well.. |
| 13:14.59 | Rico | module was not loaded with modems=v29 only : |
| 13:15.07 | Rico | [Apr 23 15:14:37] ERROR[11256]: res_fax.c:2754 set_config: 'modems' setting 'V29' is incompatible with 'minrate' setting 4800 |
| 13:15.11 | Rico | let me try with v27 |
| 13:15.37 | [TK]D-Fender | Rico: V.27 is the only one that supports 2400 |
| 13:16.38 | Rico | [Apr 23 15:15:30] ERROR[11261]: res_fax.c:2761 set_config: 'modems' setting 'V27' is incompatible with 'maxrate' setting 14400 |
| 13:16.42 | Rico | gniiiii |
| 13:17.00 | eran322 | can i tell asterisk to execute php file when miss call in queue? |
| 13:17.03 | Rico | [TK]D-Fender: I don"'t have this problem on asterisk 1.8.26 |
| 13:18.21 | [TK]D-Fender | eran322: Since you're using FreePBX your only option is to make a program that monitors the AMI events for it. |
| 13:19.05 | [TK]D-Fender | Rico: Same configs = no isuue? |
| 13:19.13 | Rico | [TK]D-Fender: yes |
| 13:19.27 | Rico | no issue when using sendfax, no error when loading module |
| 13:19.41 | wasanzy | the funny thing is that, for other calls, the cdr is printing all the coulumns : """8187325958"" <8187325958>","8187325958","s","voicemenu-thankyou","SIP/10.67.64.4-0000334f","","Hangup","","2015-04-23 13:18:48","2015-04-23 13:18:55","2015-04-23 13:19:16","28","21","ANSWERED","DOCUMENTATION","","1429795128.19665","" |
| 13:19.44 | [TK]D-Fender | Rico: Well so far it's either roll-back to a lower version that works, or remove V.27 from the list |
| 13:20.03 | Rico | or open a bug report on jira |
| 13:20.04 | eran322 | i know hot to use in ami. what is the event the those missed calls? |
| 13:20.17 | [TK]D-Fender | eran322: Tim to read... |
| 13:20.19 | [TK]D-Fender | ~book |
| 13:20.19 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 13:20.20 | [TK]D-Fender | ^^^ |
| 13:20.23 | [TK]D-Fender | time* |
| 13:20.32 | [TK]D-Fender | eran322: Or use the official Asterisk WIKI |
| 13:21.12 | wasanzy | [TK]D-Fender: I picked configurations from the book |
| 13:21.25 | wasanzy | I already have a pdf version |
| 13:21.40 | [TK]D-Fender | wasanzy: That wasn't for you.... |
| 13:21.52 | wasanzy | ok |
| 13:22.29 | wasanzy | am posting my obd call file, maybe it could be the reason |
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| 13:23.21 | Rico | [TK]D-Fender: if I'm using T38, does modem type has an importance ? |
| 13:23.54 | [TK]D-Fender | Rico: Not sure, but I have a feeling it should be "no". |
| 13:24.01 | Rico | mmh |
| 13:24.02 | Rico | ok |
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| 13:32.48 | [TK]D-Fender | eran322: You have ONE other option : configure * to directly log to MySQL and execute a stored procedure on an ADD for the appropriate kind of record |
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| 13:33.45 | [TK]D-Fender | eran322: Would probably be easier that dealing with an AMI monitoring script |
| 13:33.46 | wasanzy | my call file http://pastebin.com/U7fsSGPU |
| 13:34.29 | wasanzy | so +2349096852432 should be seen in the cdr as destination |
| 13:34.37 | wasanzy | but it is not. |
| 13:35.36 | [TK]D-Fender | wasanzy: In the first PB you gave us it was START <- |
| 13:35.38 | [TK]D-Fender | wasanzy: Extension: start |
| 13:35.40 | [TK]D-Fender | ^^^ |
| 13:35.59 | [TK]D-Fender | wasanzy: dst is the DIALPLAN destination, not a bridged channel |
| 13:37.23 | wasanzy | [TK]D-Fender: oh ok. now I understand better. so how do I get the bridged channel? because I tried ${CSV_QUOTE(${CDR(dstchannel)})} but it didn't show |
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| 13:37.41 | [TK]D-Fender | wasanzy: Where do we see it actually Dial anything? |
| 13:38.16 | wasanzy | [TK]D-Fender: I don't understand the question please |
| 13:38.29 | [TK]D-Fender | wasanzy: voicemenu-thankyou <--- sounds like you are just doing other call processing, not dialing another party |
| 13:38.47 | [TK]D-Fender | wasanzy: You don't get a dstchannel unless you DIAL something in your call processing |
| 13:39.35 | wasanzy | the OBD is dialing +2349096852432 |
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| 13:42.19 | wasanzy | ok I think I understand |
| 13:43.00 | wasanzy | I was reading this page to see if I can set a dial function in the call file but didn't find anything http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
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| 13:56.52 | [TK]D-Fender | wasanzy:There is no "dial function". that channel is only if you DIAL somehting else once the channel is executing |
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| 14:02.22 | eran322 | i cant undestand why there is no sample way to get missed calls, it is very basic needs |
| 14:02.43 | [TK]D-Fender | eran322: It would be simpler... if you weren't using FreePBX <- |
| 14:03.11 | [TK]D-Fender | eran322: And queues are not "basic", let alone scripts for when an agent doesn't answer |
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| 14:06.52 | eran322 | i 3cx system its very simple |
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| 14:14.56 | [TK]D-Fender | eran322: You're welcome to go use it. But if you want queues, then you'll have to get the paid PRO version |
| 14:15.44 | [TK]D-Fender | eran322: and I don't see anything as to what level of scripting hooks you are capable of doing. |
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| 14:40.26 | eran322 | paid pro? of asterisk? i now using 3cx for some years but now i want to move to asterisk |
| 14:40.46 | [TK]D-Fender | Er3cx with call-center capabilities is only in their PRO version |
| 14:40.51 | [TK]D-Fender | eran322: 3cx with call-center capabilities is only in their PRO version |
| 14:40.59 | [TK]D-Fender | http://www.3cx.com/phone-system/edition-comparison/ |
| 14:41.20 | [TK]D-Fender | Call Center Features = Pro |
| 14:41.48 | eran322 | i have it and paid for it |
| 14:42.23 | [TK]D-Fender | So what caused you to start looking elsewhere? |
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| 14:42.35 | eran322 | but after i see the asterisk opsions like api, i want to move to it |
| 14:43.17 | [TK]D-Fender | Guess 3cx is as closed as it looks... |
| 14:43.25 | eran322 | yep |
| 14:43.49 | [TK]D-Fender | Well I gave you 2 options for getting what you want. Neither is necessarily that hard |
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| 14:44.47 | [TK]D-Fender | Assuming you have some pretty nominal coding skills for AMI, or even basic admin skills for the stored procedure option |
| 14:45.00 | eran322 | I need a quick fix for it for my bussness |
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| 14:47.03 | [TK]D-Fender | Go hire a consultant to do it for you then. |
| 14:48.04 | Qwell | ~hafc |
| 14:48.04 | infobot | i guess hafc is hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 14:48.08 | Qwell | hey, it works |
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| 15:07.11 | jeffspeff | i'm trying to setup a new sip connection with a new provider. their instructions show to use a register command in sip.conf for inbound peer. can I just use the same information and acutally make a sip peer in my sip database? or will it not work like that? |
| 15:10.06 | Qwell | jeffspeff: realtime database? |
| 15:10.32 | jeffspeff | Qwell, yes |
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| 15:25.29 | jeffspeff | Qwell, can i just set the info up as a user or does it have to be a register string? |
| 15:25.54 | Qwell | I don't recall how register strings get converted for realtime. |
| 15:26.27 | jeffspeff | ok, but a register is the same as a user correct? |
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| 15:32.47 | jeffspeff | Qwell, did this actually make it to a release? https://reviewboard.asterisk.org/r/718/ |
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| 15:50.05 | mjordan | jeffspeff: Qwell was never involved with that review |
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| 15:51.00 | mjordan | jeffspeff: and no, it never made it in. oej objected (although I'm not sure what e-mail was referred to there). |
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| 15:51.02 | jeffspeff | ok, I don't know everybodies real names, wasn't sure. thanks |
| 15:51.33 | mjordan | jeffspeff: handles can make things confusing :-) |
| 15:51.49 | mjordan | jeffspeff: http://svnview.digium.com/svn/repotools/authors |
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| 15:52.13 | jeffspeff | that just got bookmarked! |
| 15:53.06 | Qwell | mjordan: that reminds me! Simon has no name in that file |
| 15:53.10 | litn | hi |
| 15:53.15 | jeffspeff | mjordan, do you know if you can use a sip user in place of a register string or was that rejected proposal the attempt to accomplish such? |
| 15:53.15 | mjordan | Qwell: Simon...? |
| 15:53.15 | Qwell | simon.perreault <simon.perreault@viagenie.ca> |
| 15:53.26 | mjordan | Qwell: interesting. I would have had to have gotten him in the Git migration |
| 15:53.32 | mjordan | consults his local file |
| 15:53.40 | file | refuses to answer |
| 15:53.40 | Qwell | he's there, just has no name field |
| 15:53.56 | mjordan | yeah, I got him when I moved it over to GIt |
| 15:54.09 | mjordan | but I had to reformat the file for gittness anyway |
| 15:54.24 | mjordan | of course, now that we have moved over to Git, that file is only so useful :-P |
| 15:55.34 | file | git with the times |
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| 17:34.22 | nny | having issues getting fail2ban to work with 1.11-cert6. This is my config files as well as output of /var/log/asterisk/security. Thoughts appreciated |
| 17:34.22 | nny | http://pastebin.com/GEQujykF |
| 17:35.29 | nny | oh.. wait |
| 17:36.08 | nny | my ignore ip is 192.168.1.0/8, which is what that host falls under. Let me adjust |
| 17:37.01 | nny | and.. fixed. Derp |
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| 18:27.37 | mbowie | My * process dies a couple of times a day... which I'm trying to debug. I'm passing "-g" in the params to get a core, but it isn't producing one. Besides "make dont-optimize", is there anything else I need to do to shake out something I can debug from? |
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| 18:31.00 | jeffspeff | is there a way to specify a context for a register command? I see the last portion of the command has a /exten switch but I need to specify a context other than the default context specified under [general] |
| 18:34.23 | Synthase_ | mbowie: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 18:34.23 | Synthase_ | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 18:34.29 | [TK]D-Fender | jeffspeff: No such thing |
| 18:34.46 | [TK]D-Fender | jeffspeff: Registration is registration.. when a call comes in it is matched against your peers & users |
| 18:34.55 | Synthase_ | mbowie: Hopefully you're on a recent LTS release |
| 18:37.14 | jeffspeff | [TK]D-Fender, so, the extension I specify at the end of the register command doesn't have to be an exten in the default context, it can be any other peer specified in my sip realtime db? |
| 18:38.59 | [TK]D-Fender | jeffspeff: that just tells the place you are registering to what they should send when contacting you back. |
| 18:39.09 | [TK]D-Fender | jeffspeff: It will look based on the context of the matched peer |
| 18:39.19 | [TK]D-Fender | jeffspeff: Assuming they even RESPECT that request |
| 18:39.33 | mbowie | Thanks Synthase_. Pretty sure I've followed those to the T, but I'll go through them again and see if I'm mis-stepped. Cheers. :-) |
| 18:39.39 | [TK]D-Fender | jeffspeff: it's the Contact: header in the register packet |
| 18:41.34 | jeffspeff | does it make sense to register and have an outound peer for the same provider? |
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| 18:55.47 | [TK]D-Fender | jeffspeff: Of course |
| 18:56.29 | jeffspeff | i'm not familiar with the register setup. my other providers use a direct peer connection. |
| 18:57.25 | jeffspeff | to clarify my understanding... register is for inbound connection and then i would use a manually specified sip peer for outbound correct? |
| 18:57.45 | [TK]D-Fender | what is "manually specified" supposed to mean? |
| 18:57.52 | [TK]D-Fender | Registration just tells them where to send calls |
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| 18:58.09 | jeffspeff | a sip peer that i defined in sip.conf or sip rt db |
| 18:58.17 | [TK]D-Fender | it doe not prove what auth they send with the call, or from which of their servers it will be coming from. |
| 18:58.43 | [TK]D-Fender | Since there is no such thing as "automatic", there is also no such thing as "manual" in this case |
| 18:58.58 | [TK]D-Fender | Registration is completely separate from handling the incoming call |
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| 19:11.49 | jeffspeff | [TK]D-Fender, so if the extension specified at the end of my register command is 1234, where in extensions.conf does it try to find 1234? |
| 19:12.15 | [TK]D-Fender | When the call comes in * will attempt to MATCH it to your peers & users |
| 19:12.26 | [TK]D-Fender | Whoever it matches against will be whose context it looks in |
| 19:12.29 | *** join/#asterisk jiku (~jiku@123.136.206.66) |
| 19:14.01 | jeffspeff | [TK]D-Fender, so if i'm using the setup described here http://pastebin.com/PA6s57xV then i need to set the exten of the register command to nat_babytel and add a context to that peer. correct? |
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| 19:18.43 | [TK]D-Fender | huh? |
| 19:19.10 | [TK]D-Fender | You seem to be overcomplicating this |
| 19:19.22 | [TK]D-Fender | Register is 5000% separate from EVERYTHING else |
| 19:19.44 | [TK]D-Fender | That only tells them where to contact you and what you want them to send as an exten |
| 19:19.46 | [TK]D-Fender | . |
| 19:19.47 | [TK]D-Fender | That is all |
| 19:20.01 | [TK]D-Fender | When that call arrives * will amtch the caller against your peers and users |
| 19:20.26 | [TK]D-Fender | If one of them DOES match then it will look for whatever extension they are ACTUALLY sending in whatever context that particular entry points to |
| 19:20.31 | [TK]D-Fender | The End. |
| 19:20.43 | Synthase_ | Registration is like me moving houses in real life, and then sending a letter to you with my new address so we can send letters. |
| 19:21.11 | [TK]D-Fender | Synthase_: Except that when that delivery arrives they may still ask for prrof of identity, etc. |
| 19:21.37 | Synthase_ | Right, a certified letter that you have to sign for. |
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| 19:22.55 | Synthase_ | We're obviously grossly oversimplifying this so the general concept comes across. Some people need a real world reference to visualize. |
| 19:24.07 | [TK]D-Fender | Well he alredy has peers defined, and a register as well... |
| 19:24.13 | [TK]D-Fender | so he provided those |
| 19:24.31 | [TK]D-Fender | It the inbound call MATCHES one of thos peers, then that is the conetxt the call will fall in |
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| 19:58.33 | jeffspeff | let's say my registration string ends with /1234. in my general settings in sip.conf i have context=void. when a call comes in from the provider i'm registering to, it tries to go to extension 1234 in the void context. without changing the void context defined in the general section of sip.conf, how can i get the call to go to extension 1234 in random_context ? |
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| 20:00.17 | [TK]D-Fender | jeffspeff: If the call lands under the context specified by [general] then that should mean that it FAILED to match a peer if you had set a context differently for each. |
| 20:01.48 | zekoZeko | jeffspeff: create a sip peer for your provider and set the default context for that peer there. |
| 20:02.27 | zekoZeko | jeffspeff: if i understood, you only have a register line for your provider. |
| 20:02.51 | [TK]D-Fender | he doesn't |
| 20:02.58 | [TK]D-Fender | he has 2 peers |
| 20:03.50 | [TK]D-Fender | jeffspeff: [nat_babytel] ; This context permits calls from babyTEL SBC <- you didn't set a context for this peer |
| 20:04.03 | [TK]D-Fender | OR the other |
| 20:04.08 | [TK]D-Fender | You didn't specify one at all. |
| 20:04.08 | jeffspeff | I just added that it worked. |
| 20:04.18 | [TK]D-Fender | So far the only thing we have to go on is [general] |
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| 20:08.09 | *** join/#asterisk ZX81 (~Adium@186.73.87.90) |
| 20:30.16 | [TK]D-Fender | packs up to head home |
| 20:34.37 | *** join/#asterisk seik0_ (53451d5a@gateway/web/freenode/ip.83.69.29.90) |
| 20:34.45 | seik0_ | Hi everyone! |
| 20:35.25 | seik0_ | Does somebody know of any critical issues of running DAHDI (dynamic eth dahdi) in XEN virtual server? |
| 20:35.56 | seik0_ | We have problem of hanging in asterisk, probalb in chan_dahdi.so |
| 20:36.15 | seik0_ | with pri_cpe signalling, if it matters |
| 20:37.02 | seik0_ | so asterisk continues after hanging, for example "sip show peers" returns ok, but "core show channels" hangs the CLI |
| 20:37.25 | seik0_ | and, of course, no incoming on dahdi (e1) after hanging |
| 20:37.42 | seik0_ | (did not checked of SIP intercalling, though) |
| 20:38.26 | WIMPy | That's the Redfone things? |
| 20:38.28 | seik0_ | hanging may occure in 10 seconds after restart, of after an hour |
| 20:38.45 | WIMPy | ~collectdebug |
| 20:38.45 | infobot | rumour has it, collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 20:38.46 | seik0_ | no, not the Redfone |
| 20:39.27 | seik0_ | I know about debug, for first I try to get to know maybe I am missing something simple |
| 20:40.02 | seik0_ | because if it is case to debug we need to find another production asterisk |
| 20:40.30 | *** join/#asterisk kayfox (~kayfox@orca.zerda.net) |
| 20:40.40 | seik0_ | and one more thing: while not hang it looks dahdi misses some frames |
| 20:40.50 | seik0_ | looks like yellow alarm on every channel for a moment |
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| 20:41.04 | WIMPy | Have you contacted your vendor about it? |
| 20:41.27 | seik0_ | with 30 simultaneous channels it happens more often, but does not cause hanging more often though |
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| 20:41.46 | seik0_ | not yet. As always, we try to find problem on our side first |
| 20:42.06 | seik0_ | btw, which vendor? |
| 20:42.09 | WIMPy | What kind of thing is it you have? |
| 20:42.13 | seik0_ | hardware? telco? |
| 20:42.23 | j-fish | for the people who are using Vicidial here a question : i'm trying to dial directly from the softphone to a number but no matter what caller-id i put it shows up as unknown. |
| 20:42.29 | WIMPy | The Interface you've got trouble with. |
| 20:43.27 | seik0_ | DAHDI/dyn - Cronyx E1-to-ETH (don't remember model exactly) - ISDN/E1 |
| 20:44.25 | seik0_ | DAHDI/dyn via pure ethernet |
| 20:44.26 | WIMPy | Never heard of before. |
| 20:44.55 | seik0_ | some Russian vendor |
| 20:44.56 | seik0_ | http://www.cronyx.ru/ |
| 20:45.17 | WIMPy | Why did you buy in to that experiment? |
| 20:45.39 | seik0_ | we are using one for years, but without xen virtualisation |
| 20:46.36 | WIMPy | Doesn't sound health if the virtualization has issues with 2mbit over ethernet. |
| 20:47.18 | seik0_ | how to check if virtualisation is not the issue? |
| 20:47.45 | WIMPy | Their english pages doen't work. :-( |
| 20:47.50 | seik0_ | it maybe some hard to notice timing issues |
| 20:48.09 | WIMPy | Count frames with wireshark? |
| 20:48.22 | seik0_ | nope |
| 20:48.36 | seik0_ | will try |
| 20:48.56 | seik0_ | it has English site map to the left ^.^ |
| 20:50.14 | WIMPy | Oh, yes, that works. |
| 20:52.24 | seik0_ | while during day (and for today - months) peak it takes few second for asterisk to hand. Now it is night and I'm experimenting and it took 3 hours to hang |
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| 20:53.15 | seik0_ | that's why I think that that may be XEN issue, when paraller servers are also under load |
| 20:53.45 | WIMPy | The joys of shared hardware. |
| 20:53.53 | WIMPy | Not ideal for realtime applications. |
| 20:54.00 | seik0_ | of course |
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| 21:10.06 | marceloamorim | guys, should I consider this error ? "Identifier 0, identifier_type 2 not found in identifier list" |
| 21:10.22 | marceloamorim | I use the app mysql to insert into table |
| 21:18.28 | *** part/#asterisk mjordan (mjordan@nat/digium/x-exwhxnddafcoauyt) |
| 21:28.20 | seik0_ | WIMPy, btw, some pages are strangely English =) http://www.cronyx.ru/hardware/e1lm-dahdi.html |
| 21:28.59 | seik0_ | *may be correctly switched to English |
| 21:29.29 | seik0_ | btw, that is a box we are successfully using for years |
| 21:31.28 | WIMPy | Well, they seem to have some interesting hardware, but manuals in Russian only are a bit of an issue. |
| 21:37.10 | seik0_ | Does DAHDI dynamic natively supports UDP ? |
| 21:39.26 | WIMPy | From wwhat I read it doesn't use IP usually. |
| 21:42.21 | seik0_ | some old patches said to add support for UDP: http://archive.farsouthnet.com/comma/dahdi/2.4.1-patches/ |
| 21:42.46 | seik0_ | but likely they now need adoption |
| 21:45.06 | seik0_ | not very old, though |
| 21:49.18 | seik0_ | much easier to read in wiki: http://farsouthnet.com/mediawiki/index.php?title=Comma_Developer_Resources#UDP.2FIP_Dynamic_Span_Support_.28no_longer_supported_in_2.4.1.29 |
| 21:54.09 | WIMPy | That looks both pretty old and specific to some other hardware I haven't heard of before. |
| 21:59.05 | seik0_ | pretty old, yes |
| 22:00.18 | seik0_ | main benefit of udp is to pass data over routers |
| 22:00.46 | seik0_ | but i'm not sure if it is a good idea. at least mainly |
| 22:00.50 | seik0_ | generally |
| 22:00.55 | WIMPy | Which doesn't seem to be a good idea for the timing requirements. |
| 22:01.34 | WIMPy | gets the impression that it would be best to use a dedicated link. |
| 22:02.12 | seik0_ | =) |
| 22:04.08 | seik0_ | here is 1 a.m., good time to call clients to say that we need dedicated link just right and just now =) |
| 22:09.36 | WIMPy | Or a good time to go to bed, wich is what I will do. |
| 22:10.48 | seik0_ | me too |
| 22:10.53 | seik0_ | thanks! |
| 22:10.54 | seik0_ | bye |
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| 22:33.15 | mrc3 | hey, anyone well versed in ao2_callback around? |
| 22:37.11 | rmudgett | mrc3: What do you need to know? |
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| 22:38.09 | mrc3 | i'm having a hard time gdbugging xmpp_resource_immediate(), as it seems to return null all the time |
| 22:38.28 | mrc3 | that leads to ASTERISK-23510 |
| 22:39.30 | mrc3 | rmudgett, ^^ |
| 22:40.28 | rmudgett | xmpp_resource_immediate() is a callback function passed to ao2_callback(). |
| 22:41.31 | mrc3 | rmudgett, right. the null being returned is not from xmpp_resource_immediate() itself but from the callback after it evaluates the arguments and does its magic |
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| 22:43.09 | rmudgett | The only two places that function is referenced is to get a resource out of the buddy->resource container. Since the callback always returns a match that means the container is empty. |
| 22:44.03 | rmudgett | Actually, the OBJ_NODATA flag says to not return any data matched so it will always be NULL. |
| 22:44.46 | mrc3 | hmmm... i can try changing that flag and quickly test if that works |
| 22:45.07 | rmudgett | Change the flag passed to 0 |
| 22:45.18 | rmudgett | The code that is there is just wrong. |
| 22:47.24 | mrc3 | dpkg-buildpackage will take 5 more minutes, let's see if that 0 helps |
| 22:54.09 | mrc3 | alas, no joy: "Resource (null) of buddy [...] was not found." |
| 22:55.31 | rmudgett | Whatever "xmpp show buddies" shows for resource is what you can put for the resource parameter of JABBER_STATUS. |
| 22:56.12 | mrc3 | rmudgett, the expectation is to use the bare jid instead |
| 22:57.13 | mrc3 | i.e.: user@example.com instead of user@example.com/resource, because there's a variety of xmpp clients used at the organization: some use gajim, i prefer psi+, then there's the ones on mac, and so on |
| 22:57.40 | mrc3 | i'll give it another shot tomorrow |
| 22:57.44 | mrc3 | rmudgett, thanks for the pointers! |
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