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10:34.38 | iulhk | i am using asterisk-11 , focus just sip to sip call, audio/video, in allowed codecs 'gsm;h264' , Issue peer 1000 calling to 2000 audio call, after ack, at receiver end video screen automatically opened, although caller started audio call, if i just enabled gsm, codec then both-end its audio screen, but in that case i can't start video as video codec not enabled. i want when user starting |
10:34.38 | iulhk | video call, it should start video at both-end when user starting audio call then at both end it should be audio call, is it possible ? |
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10:50.10 | AAmit | iulhk, try to set videosupport=yes in global context of sip.conf |
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11:21.43 | dkorras | Hi all, how do i change the # in the voice prompts from pound to hash, I have googleed and tried everything and I can't get it right. Please can you assist me |
11:24.08 | AAmit | dkorras, You are using core sound? |
11:24.51 | dkorras | yes i am |
11:26.41 | dkorras | AAmit, yes i am |
11:27.09 | AAmit | ok...u can use your own prompt...... |
11:28.44 | dkorras | but do i assign it so the system uses that prompt for # |
11:30.08 | AAmit | try to give it same name...and backup previous one... |
11:30.54 | dkorras | surely there must be an interface to map prompts to options on the keypad |
11:32.21 | AAmit | let me check for the same... but u can try to play with above solution.... |
11:38.07 | AAmit | It matches based on name of voice prompts.... |
11:38.36 | dkorras | i see there is hash, so if i delete the pound prompts and only have hash it should work/\ |
11:39.34 | AAmit | do not delete anything take backup....and rename the file... |
11:39.46 | dkorras | awesome let me give it a shot, thankn you |
11:39.57 | AAmit | cheers... |
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14:31.00 | cunningpike | Good morning, Mr. Fender. |
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15:51.53 | newmember | in the sip.conf file can I have two sections with the same name? |
15:52.40 | file | while I've never tried... if they are different types... they may work |
15:53.32 | Penguin | If it's anything like extensions.conf, the contents will be cumulative. |
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15:54.13 | _abc_ | Hello. Has anyone got a cisco vg202 or 204 going with asterisk? Am curious how it is done, tried today and failed. |
15:55.34 | _abc_ | It has a funky MAC scheme but I was unable to make the sccp.conf work in asterisk with it. The problem is likely the config in the vg202 which I cribbed from the web. It did not try to send packets to the asterisk at all. |
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16:47.50 | cunningpike | _abc_: Are you trying to provision it? Or is it already set up and you can't get it to register? |
16:52.33 | _abc_ | cunningpike: I worked both ends of the matter, I have a sccp.conf which I think should work with it, but the vg202 ios config is wrong. |
16:52.47 | _abc_ | So I'd like to see some samples or hints for both ends |
16:53.05 | _abc_ | I used a known to work with cucm sccp config on vg202, edited for local ips etc |
16:53.23 | _abc_ | The vg202 never seemed to access the asterisk at all |
16:53.33 | _abc_ | So I assume the problem is my config |
17:03.47 | cunningpike | _abc_: Probably - I'd do a pcap to see if it's even trying. |
17:04.21 | _abc_ | I was looking with tcpdump |
17:04.23 | _abc_ | Nothing out |
17:12.44 | cunningpike | _abc_: Not even looking for a setup config? |
17:13.28 | _abc_ | Nope. I turned cdp off and service off |
17:13.37 | cunningpike | _abc_: DHCP? |
17:13.41 | _abc_ | But I edited it inside the vg202 as running config |
17:13.45 | _abc_ | no dhcp, static ip assigned |
17:14.28 | _abc_ | In general, is there some boilerplate for setting up vg224s etc with asterisk? Other than giant sip config? |
17:14.52 | _abc_ | The monster will do sip too out of the box, but I don't know how to config it |
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17:18.44 | cunningpike | _abc_: I haven't played with them - have you looked on the voip-info wiki? |
17:19.02 | _abc_ | Yes. There is little info online on this and cisco ios is a maze. |
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17:22.58 | cunningpike | _abc_: I'd suggest coming back in the channel during the work week - there's bound to be someone in here with some experience with those endpoints. Or try the mailing list. |
17:23.26 | _abc_ | There's also ##cisco which is "unofficial" in that they point you at smartnet at the 3rd line, not the 1st |
17:23.42 | cunningpike | _abc_: lol |
17:24.02 | _abc_ | Yeah, cisco people remind me of iFruit fanbois. You are either with them or against them. |
17:24.35 | cunningpike | _abc_: I certainly seems that way sometimes... |
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18:24.05 | ovoshlook | Hello I have an issue wit MixMonitor. I need record only answered calls, so I set "b" option for this but calls still recording even call no answered My asterisk version 12.5.1 |
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18:55.38 | ovoshlook | Hello I have an issue wit MixMonitor. I need record only answered calls, so I set "b" option for this but calls still recording even call no answered My asterisk version 12.5.1 |
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21:00.30 | dtrainor | Hi. been a while since i've played wtih asterisk and I'm jumping back in to it. I want to make a dialplan that's initiated by the manager API that makes a call to an extension which is specified using Data with an Originate. Though I don't quite understand/know/remember how to specify a context. I want this originated call to drop in to a context so I can control it from there. |
21:01.03 | dtrainor | Not looking for spoon feeding, just trying to a) make sure this logic is solid b) get a few pointers on what to read up on and look in to |
21:05.06 | dtrainor | I've been playing with call files and that seems to work real well. I'm trying to make the manager API to more callfile-like things, where I can just drop the request in to a context and have it work from there. |
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21:38.04 | ovoshlook | usrloc [udomain.c:321]: dbrow2info(): non-local socket <udp:10.0.1.12:5068>...ignoring Sep 21 21:24:35 Kamailio2 kamailio[1261]: INFO: usrloc [udomain.c:321]: dbrow2info(): non-local socket <tls:10.0.1.12:4443>...ignoring |
21:38.38 | ovoshlook | sorry... wrong post |
21:38.47 | ovoshlook | Hello I have an issue wit MixMonitor. I need record only answered calls, so I set "b" option for this but calls still recording even call no answered My asterisk version 12.5.1 |
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22:19.52 | kb3ien | What is the current state of SLA? Zaptel is obselete and dahdi isn't proving the ability to make SLATrunk() |
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22:37.22 | dtrainor | Toying around with Record(). I'm using a call file. Inside of the context that has Record(), once Record() is executed, the dialplan seems to wait for it to exit, e.g. the call originated by the call hangs up (supposedly Record() see this and stops recording) and then the dialplan continues |
22:37.25 | dtrainor | Why? |
22:37.48 | dtrainor | I need liek the Background() equivalent to record or something I guess? |
22:38.09 | file | like MixMonitor? |
22:42.09 | kb3ien | Does meetme_app support video yet? |
22:43.22 | file | MeetMe is unlikely to ever get video due to the way it is written and since ConfBridge replaces it |
22:43.40 | kb3ien | How does confbridge work with SLA ? |
22:43.59 | file | unless you write SLA functionality it doesn't? |
22:44.19 | kb3ien | okay. Not sure It's worth the effor now. but one day... |
22:44.31 | kb3ien | ConfBridge has video? |
22:44.46 | file | it can relay the video of the active speaker, yes |
22:45.35 | kb3ien | have a link to meaningful details of this ? |
22:45.42 | file | um |
22:46.16 | file | there's some config details on the wiki... |
22:46.18 | dtrainor | file++ thank you very much |
22:46.25 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_app_confbridge?src=search |
22:47.54 | dtrainor | works like a charm. |
22:48.02 | kb3ien | which user is the active speaker in confBridge ? |
22:48.09 | file | the person who is speaking. |
22:48.29 | file | or you can use marked users |
22:48.37 | file | it's documented under video_mode the options |
22:48.59 | kb3ien | Okay. It monitors the audio VU ? |
22:49.17 | kb3ien | follow_talker ? |
22:49.20 | file | the energy, yes |
22:49.28 | ovoshlook | Hello I have an issue wit MixMonitor. I need record only answered calls, so I set "b" option for this but calls still recording even call no answered My asterisk version 12.5.1 |
22:59.08 | kb3ien | fair enough. thanks. |
23:01.02 | file | rediscovers what flights he booked for going to SIPit and is pleasantly surprised |
23:02.13 | kb3ien | I'll probable have to mod some old-word scripts to fix my blinkey light. |
23:02.41 | kb3ien | Was video added in 13 ? |
23:03.24 | file | added to what? |
23:03.53 | kb3ien | ConfBridge? Its missing in the 11 docs. |
23:03.59 | file | it's in 11 |
23:03.59 | kb3ien | the video references. |
23:04.17 | dtrainor | If I have a dialplan that include Dial(), will execution of other items in that dialplan be blocked until Dial() returns? It looks like that's what I'm experiencing. |
23:04.25 | file | dtrainor, yes. |
23:04.30 | dtrainor | I see. |
23:05.08 | dtrainor | This means a) i need to use something other than Dial() or b) there's a smarter way to do this. |
23:05.23 | dtrainor | I didn't see any Dial() arguments that would overwrite this behavior. |
23:05.25 | file | kb3ien, the documentation stuff was improved to include configuration files |
23:05.42 | file | dtrainor, there aren't any - you can't do two things at once two a channel except for a few circumstances |
23:06.04 | file | Dial thus blocks as it is in control of the caller and the called party |
23:06.26 | dtrainor | Understood. |
23:06.36 | ovoshlook | Hello I have an issue with MixMonitor. I need record only answered calls, so I set "b" option for this but calls still recording even call no answered My asterisk version 12.5.1 |
23:07.16 | file | ovoshlook, you might want to take your question to the asterisk-users mailing list... |
23:07.24 | file | or problem. |
23:07.24 | dtrainor | Well. Crap. Ok, I'm going to look in to some options here. |
23:07.30 | file | dtrainor, what are you trying to do? |
23:07.58 | dtrainor | Actually. Nothing even remotely related to that. I just looked at my notes and my plans. |
23:08.07 | dtrainor | So... I was just wasting your time, really. Sorry about that. |
23:09.22 | ovoshlook | @file> Why not here? |
23:09.35 | file | ovoshlook, because you've been asking for quite awhile and nobody has responded? |
23:11.08 | dtrainor | thanks again file. |
23:14.51 | ovoshlook | hm)) That is may be write way) I hoped for fast help with this question... |
23:15.25 | file | in a world where everyone volunteers their time speed is not always a priority |
23:15.58 | ovoshlook | Just becouse my problem only with this system, at another servers (with another older versions) no problems... |
23:17.35 | ovoshlook | I rrealy understand it. When I may help, always help too) So thank for advice, If you will have any adias with my problem let me know |
23:27.52 | *** join/#asterisk iq (~iq@pool-71-170-194-190.dllstx.fios.verizon.net) |
23:29.48 | *** join/#asterisk zerohalo (~zerohalo@2601:6:f80:224:582e:1b29:fbd0:8044) |
23:36.21 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
23:37.10 | ChannelZ | You could look at the result of ${DIALSTATUS} and stop/delete the monitor if you don't like what you see. |
23:37.13 | kb3ien | Does devstate work the same as in 1.8 ? |
23:38.41 | kb3ien | it seems to be func device_state now, but close enough maybe. |
23:51.22 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
23:52.03 | dan_j | Hi. I've got a major problem. I've just upgraded mysql along with odbc. Now, when i try to launch asterisk, i'm getting a segmentation fault. I have no idea how to resolve this. Any advice? |