| 00:03.48 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 00:13.26 | *** join/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 00:13.27 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:16.33 | volga629 | Hello Everyone, [2012-09-29 16:19:54] WARNING[10494] app_voicemail.c: Unable to read password [2012-09-29 16:19:54] ERROR[10494] app_voicemail.c: No vmstate found for user:, mailbox 101. Asterisk use IMAP storage form vm. |
| 00:17.42 | volga629 | that only extension screw up, and I can dial out and in from this phone only vm affected ? |
| 00:18.35 | WIMPy | AFAIR "unable to read password" means that nothing has been entered. |
| 00:19.07 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 00:19.38 | volga629 | the problem that I entered |
| 00:20.20 | WIMPy | Do you have some other IVR to test DTMF functionality? |
| 00:20.34 | volga629 | yes I see -- Incorrect password '' for user '101' (context = |
| 00:20.51 | volga629 | hmmm, not really |
| 00:21.23 | WIMPy | Make one. |
| 00:21.28 | volga629 | it should on my pbx or just to test |
| 00:21.36 | volga629 | any |
| 00:21.48 | WIMPy | On the same server. |
| 00:21.54 | volga629 | ok let me |
| 00:21.57 | volga629 | see |
| 00:26.20 | volga629 | ok tested press 2 nothing |
| 00:26.29 | volga629 | no valid response |
| 00:28.36 | volga629 | <PROTECTED> |
| 00:30.55 | volga629 | tested for another phone and it worked as expected, something on snom 370 |
| 00:31.24 | WIMPy | So scheck the DTMF mode configuration. |
| 00:35.30 | volga629 | on asterisk is RFC 2833 |
| 00:35.38 | volga629 | let me on phone |
| 00:35.41 | volga629 | see |
| 00:45.29 | volga629 | http://ScrnSht.com/hbfhdq |
| 00:46.09 | volga629 | that setting what on the phone |
| 00:48.25 | volga629 | can't see nothing special |
| 00:49.32 | WIMPy | Looking ok. DTMF payload type? |
| 00:50.16 | WIMPy | Advaced > SIP/RTP |
| 00:56.02 | volga629 | http://ScrnSht.com/inkens |
| 00:57.36 | WIMPy | Hmm. Where has the setting for the payload type gone? |
| 00:58.07 | volga629 | can't find |
| 00:58.23 | WIMPy | I/ve got it 3rd under RTP/RTCP. |
| 00:58.34 | volga629 | Just trying find on snom website |
| 00:59.22 | WIMPy | Is it a brand new or a really old firmware? |
| 01:01.15 | *** join/#asterisk ska (~ska@unaffiliated/ska) |
| 01:01.38 | volga629 | 8.7.3.7 |
| 01:01.56 | volga629 | it last one look like |
| 01:02.32 | ska | Given a choice of phones, what is a good way to auto-provision phones? I see tftp, but I also see http based autoprovisioning. |
| 01:02.39 | WIMPy | I had some issues with 3.7 and went back to 3.4.35. |
| 01:02.58 | WIMPy | ska: What your phones can do. |
| 01:03.33 | ska | WIMPy: What phones do you prefer when you have a choice? |
| 01:04.32 | ska | Is there a way that is easier thatn DHCP+tftp/ftp/http? |
| 01:05.02 | volga629 | Discard buffered DTMF tone when stop command is received before playing |
| 01:05.03 | WIMPy | I find the Snom 320/360/370 least annoying. |
| 01:05.34 | WIMPy | volga629: Where is that from? |
| 01:05.39 | volga629 | this from release on snom site last release 8.7.3.10 |
| 01:06.03 | WIMPy | Sounds like the opposite direction. |
| 01:06.08 | volga629 | yes |
| 01:13.01 | volga629 | I will try rebuild the phone |
| 01:20.17 | volga629 | rebuild is running let see |
| 01:24.36 | volga629 | might you can suggest I have branch which connected through vpn tunnel, latency is about 180 what possible to do for some noise reduction ? |
| 01:25.09 | WIMPy | Noise? |
| 01:25.53 | volga629 | yes when I call or they call I hear like bell |
| 01:26.27 | volga629 | like in server room :-) |
| 01:27.29 | WIMPy | I don't see how that has to do with the connection, unless that's seriousely unusable. |
| 01:27.37 | jpsharp | latency doesn't create noise. |
| 01:28.42 | volga629 | might be D-link router crap |
| 01:31.17 | volga629 | Phone is rebuilt, but result the same |
| 01:31.43 | volga629 | vpn tunnel is all linux based |
| 01:32.28 | volga629 | from asterisk side, remote side is D-link router |
| 01:33.41 | WIMPy | Even packet loss wouldn't create noise. Amybe some robotic voices ot the like. |
| 01:34.28 | WIMPy | With the DTMF thing, I guess there's some real debugging to be done. |
| 01:34.41 | volga629 | http://fpaste.org/mp1H/ here |
| 01:34.50 | WIMPy | Is that on that vpn as well? |
| 01:35.20 | volga629 | this debug for DTMF issue |
| 01:36.26 | volga629 | is directmedia can affect ? |
| 01:36.34 | *** join/#asterisk bmg505 (~leon@196-210-141-90.dynamic.isadsl.co.za) |
| 01:37.51 | volga629 | no this phone here close to main asterisk |
| 01:38.16 | WIMPy | If you don;t find anything when you crank up debug, you might have to go as far as rtp debug. |
| 01:38.33 | WIMPy | Might be easier to use wireshark, I guess. |
| 01:38.50 | WIMPy | I never had to debug DTMF. |
| 01:40.19 | volga629 | no do see yet nothing :-( |
| 01:40.28 | volga629 | non't |
| 01:40.31 | volga629 | don't |
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| 01:42.23 | volga629 | yes, I never did debug to rtp either |
| 01:42.30 | *** join/#asterisk nickgaw (~nick@70-139-55-30.lightspeed.hstntx.sbcglobal.net) |
| 01:43.27 | nickgaw | Hi, Where does asterisk store it's log file as I ran threw the asterisk and incrediblepbx installation but asterisk dies with exit status 1 where can I find the reason for the dieing process? |
| 01:44.30 | WIMPy | start it with -cvvvvdddd |
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| 01:46.00 | volga629 | app_voicemail.c:9794 vm_authenticate: Unable to read password |
| 01:50.50 | nickgaw | the same error keeps coming up asterisk died with exit code 1 automatically restarting asterisk so I can't stop this process quickley enough to get the text of the error |
| 01:51.26 | WIMPy | Start Asterisk with that parameter, not some script. |
| 01:51.38 | WIMPy | Or those parameters, rather. |
| 01:51.50 | nickgaw | that is what I am doing. I used amportal stop first. |
| 01:52.18 | WIMPy | So what is restarted then? |
| 01:52.23 | nickgaw | is there a way to pipe the asterisk output and only that to a log file? |
| 01:56.26 | nickgaw | some script in pbxinaflash keeps trying to restart asterisk before I can get a chance to run it manually |
| 01:56.51 | nickgaw | and when I do the script tries to restart it as soon as it dies so I can't get the error |
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| 01:58.18 | WIMPy | You should get something if you check loger.conf and set verbose and debug in asterisk.conf. |
| 02:00.52 | nickgaw | the file says not to edit it directly as it is automatically generaged by freepbx. plus my screen keeps filling up with asterisk died with exit code 1 automatically restarting asterisk |
| 02:00.56 | volga629 | Differentiated Services Field: 0xb8 (DSCP 0x2e: Expedited Forwarding; ECN: 0x00: Not-ECT (Not ECN-Capable Transport)) ? |
| 02:01.37 | volga629 | This form wireshark ? |
| 02:01.40 | volga629 | from |
| 02:02.05 | WIMPy | nickgaw: Sorry, but that ends up with |
| 02:02.08 | WIMPy | ~freepbx |
| 02:02.08 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 02:02.21 | WIMPy | volga629: That does not mean anythign to me. |
| 02:02.44 | nickgaw | so you think they changed something in asterisk for freepbx? |
| 02:02.57 | WIMPy | no |
| 02:03.01 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
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| 02:03.15 | *** part/#asterisk nickgaw (~nick@70-139-55-30.lightspeed.hstntx.sbcglobal.net) |
| 02:03.29 | volga629 | <PROTECTED> |
| 02:04.19 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 02:04.36 | volga629 | this when traffic from pbx to phone |
| 02:04.57 | volga629 | .... ..00 = Explicit Congestion Notification: Not-ECT (Not ECN-Capable Transport) (0x00) and this from phone to pbx |
| 02:05.31 | WIMPy | Oh, you're using TLS, right? |
| 02:05.37 | volga629 | yes |
| 02:05.55 | WIMPy | Ok, that does make sense to me then. |
| 02:06.39 | volga629 | what exactly ? |
| 02:07.07 | WIMPy | The ECN stuff. It's TCP. |
| 02:07.59 | volga629 | yes should be, but says protocol 17 |
| 02:08.13 | volga629 | this RTP traffic |
| 02:08.36 | WIMPy | Are you also using SRTP? |
| 02:09.21 | volga629 | no SRTP in place ECN that he is can't recognize properly TCP, because it is UDP |
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| 02:12.01 | volga629 | might need check switch lldp staff enabled, but why cause issue right now, it working like this for long time and affected only this snom phone |
| 02:12.44 | WIMPy | has NFI. |
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| 02:46.44 | nickgaw | Hi, I finally found the error ERROR[2727] cdr_mysql.c: Unable to query table description!! Logging disabled. |
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| 03:21.34 | volga629 | I corrected few values on lldp network-policy, but nothing not help I am out of ideas what else can be. I see in pcap file normal transaction |
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| 04:54.36 | Galen_ | hey all, having some issues connecting 2 servers via IAX... if anyone could lend some assistance! |
| 04:55.10 | Galen_ | http://pastebin.com/WxxHvjgG <--- this is my iax and extensions config files from each server, and the CLI output for the failure |
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| 05:25.41 | Galen | anyone have any insight on my IAX issue? http://pastebin.com/WxxHvjgG |
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| 05:54.17 | ectospasm | Galen: what does "iax2 show peers" show? |
| 05:55.05 | ectospasm | Galen: judging by this error: chan_iax2.c:4681 __auto_congest: Auto-congesting call due to slow response << you have a network connection (the other IAX2 host is too slow to respond) |
| 05:55.35 | ectospasm | s/n (/n problem (/ |
| 06:37.55 | Galen | sorry ... ive been trying to figure this out, and you are on the money ! |
| 06:38.32 | Galen | something is wrong and i cant get to the bottom of it. cant even connect to that remote server via SSH remotely anymore |
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| 06:39.58 | Galen | insert new (replacement) router, BAM all fixed ! |
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| 11:26.07 | MACscr | is there any type of format options for the response on cli commands? Just trying to find a better way to parse some of the responses |
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| 11:46.36 | Dovid | morning everyone |
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| 13:29.30 | enzo | hello |
| 13:30.58 | enzo | I've bought a TDM410P to replace my old zaptel cards, but I'm lost with dahdi config, Any idea about a default config for genconf_parameters ? in zaptel.conf I used to set the fxo/fxs channels |
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| 13:52.40 | cusco | hi |
| 13:52.54 | cusco | I am looking at the differences in realtime queue_log from 1.6 to 1.8 |
| 13:52.55 | cusco | https://issues.asterisk.org/jira/browse/ASTERISK-15863 |
| 13:53.10 | cusco | can asterisk 1.8 still use 1.6's format in some configuration? |
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| 15:05.45 | cusco | hi... |
| 15:06.03 | cusco | can asterisk 1.8 use realtime queue_log with same format as 1.6? |
| 15:06.31 | cusco | I thought [compat] section in asterisk.conf would handle that |
| 15:06.36 | cusco | the: pbx_realtime=1.6 |
| 15:07.42 | cusco | [TK]D-Fender: would you know? |
| 15:11.09 | [TK]D-Fender | I never noticed a difference |
| 15:11.39 | cusco | well, columntime is stored in unixtimestamp format in 1.6 |
| 15:11.43 | cusco | and in datetime in 1.89 |
| 15:11.45 | cusco | and in datetime in 1.8 |
| 15:11.59 | cusco | I would like to keep using unixtime, in 1.8, is it possible? |
| 15:12.02 | cusco | (in queue_log ) |
| 15:13.14 | [TK]D-Fender | no idea |
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| 15:41.32 | *** join/#asterisk john_roland_1359 (~kunal@180.183.11.238) |
| 15:41.55 | john_roland_1359 | hello. asterisk seems to be having alot of problems. i need to uninstall and give it a fresh re-installation |
| 15:42.30 | john_roland_1359 | is there an easy way around to re-instal asterisk? |
| 15:43.31 | *** join/#asterisk bipul (~vipul@unaffiliated/bipul) |
| 15:46.01 | john_roland_1359 | or a good way to repair asterisk |
| 15:47.32 | volga629 | Where can find info about SIP/2.0 401 Unauthorized |
| 15:50.23 | john_roland_1359 | or perhaps a way to debug asterisk to make sure all the files and configs are good. |
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| 15:52.51 | [TK]D-Fender | john_roland_1359: Too vague. Show us a specific problem and maybe we can help you |
| 15:55.36 | john_roland_1359 | [TK]D-Fender: My asterisk keeps going dead. it does not open with the CLi console. unless i shutdown the system and run it with asterisk -vvvvvvc |
| 15:56.44 | john_roland_1359 | the command "asterisk -r" does not work ....asterisk just hangs on startup |
| 15:57.01 | [TK]D-Fender | john_roland_1359: Show us the before & after |
| 15:57.09 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 15:57.16 | john_roland_1359 | ok. ill post it on pastebin |
| 16:17.38 | [TK]D-Fender | ~siprfc |
| 16:17.39 | infobot | somebody said siprfc was http://www.faqs.org/rfcs/rfc3261.html |
| 16:17.44 | [TK]D-Fender | volga629: ^ |
| 16:18.55 | volga629 | yes |
| 16:19.18 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 16:19.26 | volga629 | Ooo thanks |
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| 16:25.58 | WIMPy | Now that's a nice login :-( |
| 16:26.01 | WIMPy | /bin/tcsh: Too many open files in system |
| 16:26.08 | WIMPy | Connection to lmaa closed. |
| 16:28.00 | *** join/#asterisk aossama (~ao@41.232.127.52) |
| 16:29.35 | john_roland_1359 | http://pastebin.com/QtRLKJj4 |
| 16:30.42 | john_roland_1359 | [TK]D-Fender: http://pastebin.com/QtRLKJj4 ---> a brief summary of my problem. I want to know if there is a way to troubleshoot asterisk so that I could pinpoint where the problem is coming from. |
| 16:30.51 | WIMPy | asterisk -cvvvvdddd |
| 16:31.15 | WIMPy | That should tell you what's going on. |
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| 16:31.48 | john_roland_1359 | WIMPy: thanks. let me try |
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| 16:32.36 | pabelanger | john_roland_1359, $ sudo asterisk -r |
| 16:33.30 | john_roland_1359 | I just stopped the asterisk -cvvvvdddd . Now will resume asterisk with asterisk -r |
| 16:33.37 | john_roland_1359 | hopefully it should work |
| 16:35.47 | john_roland_1359 | for some reason I cannot resume my asterisk with asterisk -r. it gets terminated when i leave the console |
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| 16:35.53 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 16:36.15 | john_roland_1359 | http://pastebin.com/6CYF0EQm -----> this is what i get when i run "asterisk -cvvvvdddd' |
| 16:36.29 | WIMPy | If you ^C on asterisk -c you will stop Asterisk. |
| 16:36.46 | WIMPy | Start it without parameters. |
| 16:36.55 | john_roland_1359 | ok |
| 16:37.33 | john_roland_1359 | i just tried to start without parameters and nothing happened. didnt load the Cli console |
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| 16:38.01 | WIMPy | That's the idea of not having -c. |
| 16:38.25 | WIMPy | Start it without -c and then connect to what you started with -r. |
| 16:39.47 | john_roland_1359 | just to make sure we are on the same page, this is asterisk without the parameter. right? -----> root@atrix:/home/admin# asterisk |
| 16:40.20 | WIMPy | yes |
| 16:40.42 | john_roland_1359 | now how do i get into the console? |
| 16:40.56 | WIMPy | Start it without -c and then connect to what you started with -r. |
| 16:43.38 | john_roland_1359 | WIMPy: What is the command to start asterisk without -c ? |
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| 16:44.13 | WIMPy | Err? |
| 16:44.32 | john_roland_1359 | maybe i need to read the manual |
| 16:44.37 | WIMPy | without -c means without -c. |
| 16:46.17 | john_roland_1359 | so it should be "asterisk" and then "asterisk -r" |
| 16:47.03 | WIMPy | yes |
| 16:48.16 | john_roland_1359 | WIMPy: thank you |
| 16:49.17 | john_roland_1359 | how do i exit the console without quitting it? |
| 16:49.30 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 16:49.42 | WIMPy | exit |
| 16:49.46 | WIMPy | Or just ^C |
| 16:50.22 | john_roland_1359 | and to enter in : first enter asterisk without -c and then run it with -r? |
| 16:50.54 | WIMPy | *Start* it without -c and then *connect* to what you started with -r. |
| 16:51.25 | WIMPy | asterisk -c starts it in the foreground. |
| 16:51.35 | john_roland_1359 | oh ok... |
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| 17:03.27 | john_roland_1359 | WIMPy: thanks once again for your input. i very much appreciate it. |
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| 17:14.38 | *** join/#asterisk nisam (nisam@115.242.32.160) |
| 17:15.06 | nisam | hi friends.. |
| 17:15.45 | nisam | which codec best for bluetooth mobile channel |
| 17:19.38 | WIMPy | You can change code for the BT link? |
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| 17:25.37 | [TK]D-Fender | nisam: preferably the same codec as it will use as it relays the call out. |
| 17:26.40 | WIMPy | would have thought it is acting as an audio device. |
| 17:28.18 | nisam | Thanks Fender..Actually if i am calling through mobile phone, one way audio is coming.wat will be the problem..? |
| 17:29.32 | john_roland_1359 | which folder would i store mp3 files? or how do designate asterisk to a folder which has my mp3 files stored? ---just trying to test mp3 playback. |
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| 17:38.14 | john_roland_1359 | if i were to load a mp3 file in /usr/share/asterisk/sounds/ ---> would it cause a file system error ? |
| 17:40.58 | Gugge | only if you have a really strange filesystem |
| 17:42.50 | john_roland_1359 | Gugge: might be...my filesystem acts wired. would it be possible to designate a folder...so that asterisk looks in that folder for mp3 files? |
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| 17:43.50 | Sicelo | might be better to fix the fs :-/ |
| 17:44.01 | Gugge | agree |
| 17:45.56 | john_roland_1359 | i have already ran fsck. and it does repair the system. now it does work. only when i play in GUI things go a little wrong |
| 17:45.58 | [TK]D-Fender | john_roland_1359: If the presence of a file is enough to cause a filesystem error... your filesystem is already screwed beyond imagining. |
| 17:46.19 | [TK]D-Fender | john_roland_1359: And you need to be more descriptive of your problems. |
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| 17:47.07 | john_roland_1359 | @TK]D-Fender: Noted. Will try to be more descriptive. |
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| 17:55.55 | c0dyhi11 | Hello. I'm having a hell of a time getting Echo Cancellation tuned right. I have a Digium AEX800 with the hardware Echo Cancellation module. I'm running Dahdi 2.6.1 and Asterisk 1.8.14 does anyone have the time to try and point me in the right direction? I've been tweaking the vpmnlptype, vpmnlpthresh, & vpmnlpmaxsupp (When vpmnlptype is set to 6) and I'm not getting anywhere. I think part of my problem is that in /etc/dahdi/system.conf "echocanceller=hwec,1 |
| 17:55.56 | c0dyhi11 | doesn't seem to work. But echocanceller=mg2,1 does. |
| 17:57.45 | WIMPy | What makes you think it doesn't work? |
| 18:00.26 | c0dyhi11 | I get an error. One sec and i'll paste it |
| 18:00.49 | c0dyhi11 | Loading DAHDI hardware modules: |
| 18:00.49 | c0dyhi11 | <PROTECTED> |
| 18:00.50 | c0dyhi11 | Running dahdi_cfg: DAHDI_ATTACH_ECHOCAN failed on channel 1: Invalid argument (22) |
| 18:00.51 | c0dyhi11 | <PROTECTED> |
| 18:01.03 | pabelanger | ~pb |
| 18:01.04 | 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. |
| 18:01.09 | pabelanger | c0dyhi11: ^ |
| 18:01.14 | c0dyhi11 | Sorry |
| 18:01.41 | c0dyhi11 | I didn't know 4 lines was enough for pate bin. I'll do that in the future. |
| 18:02.32 | pabelanger | If you have to copy and paste, use pastebin. |
| 18:02.40 | c0dyhi11 | K. Thanks. |
| 18:03.33 | c0dyhi11 | If i use echocanceller=mg2,1 instead of hwec I get on "OK" for both and not that echo can error. |
| 18:05.16 | c0dyhi11 | Here is the output for dahdi_cfg -vvvv http://pastebin.com/REyWMPZ4 |
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| 18:30.54 | madduck | instead of configuring ODBC, I can just use res_pgsql.conf to configure a direct connection to PostgreSQL, right? |
| 18:31.17 | madduck | Or are the interfaces provided by res_odbc and res_pgsql different? |
| 18:31.30 | madduck | i'd like to cut out all the unnecessary middle stuff |
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| 20:36.00 | Waj | Hi! I was hoping I could ask a quick question and maybe a little push in the right direction. I want to 'intercept'/stop DTMF signals in a bridged sip connection (Dial command) and process them, so that I take some actions. Sort of like a menu system. Example: I the called user ender 5, asterisk will send 642# downline to the callee user... Have tried searching for various DTMF topics, but |
| 20:36.00 | Waj | cant realy find anything that could get me started on a solution. |
| 20:36.11 | Waj | *get |
| 20:36.54 | Waj | sorry for the spelling mistakes... |
| 20:37.57 | WIMPy | "dynamic features" |
| 20:38.23 | Waj | Thanks WIMPy... looking into it.. |
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| 21:08.31 | leifmadsen | Waj: see features.conf |
| 21:19.10 | Waj | Thanks leif... I am trying to get it to work atm... I have uncommented the example "testfeature" and set it up as Set(DYNAMIC_FEATURES=testfeature). But in the console there isnt showing any incomming dtmf when testing, and there is no reaction from asterisk... It should play tt-monkeys... And ide? |
| 21:19.25 | *** join/#asterisk WolWid (WolWid@pD9ED4EFE.dip.t-dialin.net) |
| 21:21.11 | [TK]D-Fender | Waj: Idea : prove that your DTMF is functioal outside of a call at all first. Then show us your actual dialplan and call attempt |
| 21:21.14 | [TK]D-Fender | ~pb |
| 21:21.14 | 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. |
| 21:21.17 | [TK]D-Fender | ^ your friend |
| 21:22.52 | Waj | Will do... I see in the logfiles (just forgot to output to console), that dtmf is being received.. but it isnt playing yet... |
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| 21:29.08 | Waj | [TK]D-Fender: I would like to test a little on my own before posting on pastebin... I'll get back to you :) |
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| 21:34.54 | dym | evening! |
| 21:34.55 | dym | :) |
| 21:35.10 | dym | Oh, leifmadsen is also around? |
| 21:35.56 | leifmadsen | sometimes |
| 21:38.07 | jpsharp | Only on the days he can chew through the straps. |
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| 21:38.55 | dym | :D |
| 21:39.03 | dym | How is life treating you, leif? |
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| 22:04.49 | Waj | I have been testing a little with the dynamic feature and incomming dtmf... I am able to see dtmf output on console, but I just wont trigger the feature to play tt-monkeys. Extension file is just a normail example conf with the new extension added (http://pastebin.com/pyFEPGzG) |
| 22:06.35 | Waj | the feature.conf is also the normal example file, just with testfeature uncommented, and "featuredigittimeout = 2000" added. |
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| 22:07.37 | Waj | ...any help is much appreciated |
| 22:08.24 | [TK]D-Fender | Waj: permanently remove all that commented out junk and realize you didnt show use your actual features.conf and seem to believe that well take it on faith that you did it right. |
| 22:11.24 | [TK]D-Fender | Waj: exten => _X.,n,Wait(20) <--- and features are for while you're in a call, not just sitting in "wait" |
| 22:21.58 | Waj | Thanks.. |
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| 22:44.31 | Waj | Have gotten it to work when I bridged the call with an additional account. Now I just need to stop the DTMF from going directly from caller to callee. Sort of intercept it... |
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| 23:04.50 | Waj | Does anyone know if it is possible to suppress dtmf signals comming from a caller to a callee? |
| 23:05.34 | unicron | use a lower bandwidth codec? |
| 23:06.07 | WIMPy | Or use a peer configuration with a deliberately broken dtmf config. |
| 23:07.18 | Waj | I still want to be able to send DTMFs from the asterisk (SendDTMF) but dont want caller or callee to be able to directly send DTMFs to each other. |
| 23:08.33 | WIMPy | Hmm. I guess you could write a patch for local channels that ignores DTMF. |
| 23:08.34 | [TK]D-Fender | Waj: There is no option for this |
| 23:10.40 | Waj | Background info: The caller will be a machine that uses dtmf signal for various operations. Want to simplify and control user (callee) interaction. |
| 23:12.09 | Waj | WIMPy: that sounds very complicated. |
| 23:12.18 | Waj | [TK]D-Fender: ok. ty |
| 23:17.45 | Waj | Could an idea be to (1.) deliberatly break the DTMF at the start of an extention, (2.) then repair it for sending dtmf from asterisk (3.) and finally to break it again? |
| 23:18.32 | Waj | ... by using the SIPDtmfMode command? |
| 23:19.33 | WIMPy | didn;t notice that application before. Yeah, that should help. |
| 23:19.48 | WIMPy | Unless your peer understands both info and rfc. |
| 23:20.28 | Waj | ah ha... I have a SPA3102 in between... should be able to set it to something fixed, so that I can break it |
| 23:24.36 | Waj | Thanks alot for all your help, I am gonna try and see what I can come up with. |
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