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| 01:19.02 | drab | any recommendation to implement something that automatically invites a bunch of people to a confbridge as the administrator enters it? |
| 01:19.17 | drab | I couldn't find a lot of posts about it (only two) and they seemed to suggest the use of "Originate" |
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| 01:19.45 | drab | altho some guy was saying it's not async so people could potentially take a long time to all join depending when the current person being called picks up |
| 01:19.58 | drab | testing that now |
| 01:20.36 | WIMPy | you need to do it via an external application. |
| 01:21.48 | drab | using ARI/AMI you mean? haven't looked at any of that stuff yet |
| 01:22.24 | drab | I think I also found a mention of, I forget what, dial_exec or something, which would place some files in a special dir that Asterisk would execute |
| 01:22.50 | WIMPy | call files |
| 01:22.54 | drab | how come originate wouldn't work? just trying to understand since I'm new to asterisk and seems like a good opportunity to learn more about its functioning |
| 01:23.29 | WIMPy | It would work, but you'd have to do it one call after another. |
| 01:23.54 | drab | oh, ok, so you confirm that the next step in the plan would be executed only after the first person picked up or timeout occurred? |
| 01:24.21 | WIMPy | yes |
| 01:24.28 | drab | ok, thanks |
| 01:24.47 | drab | call files are also a no go? |
| 01:25.43 | WIMPy | No, they would be executed in parallel. |
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| 01:26.33 | drab | would you discourage that for some reason? I can understand that ARI/AMI might be cleaner, but having not looked into that at all if call files could work that might be good enough to start with at least |
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| 01:27.01 | drab | are they somewhat frowned upon like using system() to shell out? |
| 01:28.02 | WIMPy | You can use them, but it's a short cut with a price. It's not that easy to find out the results. |
| 01:28.47 | drab | fairdo, I guess I can see the pattern/potential problem |
| 01:28.48 | drab | thank you |
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| 02:20.39 | danielbw | WIMPy, are you still here? |
| 02:21.03 | WIMPy | yes |
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| 02:33.31 | danielbw | nice |
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| 05:51.20 | danielbw | i have some more debugging stuff to put together |
| 05:51.39 | danielbw | hopefully this will be able to be resolved. |
| 05:51.46 | danielbw | thanks for your help so far |
| 05:51.48 | danielbw | good night |
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| 08:07.14 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.3.2 (2015/04/08), 11.17.1 (2015/04/08), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 08:43.22 | nilesh | hello |
| 08:43.46 | nilesh | I'm having a small issue with chan_mobile and certified asterisk 13 |
| 08:44.22 | nilesh | Anything that the other side says is not audible on a sip phone which supports ulaw/alaw/g729, but when I use a softphone with the same codec it works |
| 08:44.35 | nilesh | the sip phone itself works with international SIP provider which uses the same codec |
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| 08:44.59 | nilesh | although whatever I say in the sip phone is audible on the other side |
| 08:45.07 | nilesh | is this really a codec problem or a phone problem? |
| 08:45.23 | wasanzy | hello |
| 08:45.52 | nilesh | wasanzy: |
| 08:45.55 | wasanzy | please I want to do something more crazy with Asterisk, maybe someone has an idea |
| 08:49.44 | wasanzy | it goes like this: I have asterisk working already, and I can dial in and do couples of things. but now, I want to make it such that, when I dial into asterisk, Asterisk should forward the call or whichever way to a call center (cellphone) for an agent to answer and speak with the caller. will this be possible? how can I do this? any idea? |
| 09:07.55 | nilesh | ok some update, I have set up mixmonitor to record the calls and I can hear both sides in the recording |
| 09:08.02 | nilesh | so I guess this is really a phone issue? |
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| 09:30.32 | wdoekes | wasanzy: Dial() dialplan application? generally something like: Dial(SIP/my-itsp/+12345) |
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| 09:37.53 | wasanzy | like a call file? |
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| 09:41.05 | polysics | hello! |
| 09:41.25 | polysics | is there any writeup of the "old" bridging lifecycle somehwere, please? |
| 09:41.41 | polysics | trying to wrap my head around the new event chain to port some Adhearsion parts |
| 09:42.28 | polysics | is the Bridge event plus Bridgestate still there? |
| 09:45.12 | polysics | apparently not :) |
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| 11:59.22 | madduck | so my provider now screwed up reverse DNS twice, causing incoming calls not to be mappable to the provider sip.conf stanza and hence being treated as "unauthenticated". Without hardcoding their IP addresses, is there really no other way to match incoming calls than reverse DNS?? |
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| 12:12.42 | MLNoah | I recently updated a box with FAX For Asterisk from 11.5 to 11.17.1, and now when I try to receive faxes, I'm getting an error 'modems' setting 'V17,V27,V29' is incompatible with 'minrate' settings 2400. I've tried changing the minrate setting in my res_fax.so to 4800, and I've tried commenting out all lines in that file (issuing module reload res_fax.so and unloading + loading res_fax and... |
| 12:12.44 | MLNoah | ...res_fax_digium after both changes). fax show settings shows a minimum bit rate of 4800, but I still get the same error. |
| 12:13.02 | MLNoah | What can I do to fix it with 11.17.1, or what version should I roll back to in order to not be receiving this error? |
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| 12:28.00 | [TK]D-Fender | nilesh: That is a NETWORKING setup issue |
| 12:28.21 | nilesh | [TK]D-Fender: what makes you say that? |
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| 12:28.44 | [TK]D-Fender | nilesh: Because if you are not getting audio then packets aren't making it to the right place |
| 12:29.25 | [TK]D-Fender | nilesh: So either firewall/router issue, or improper NAT setup where applicable, etc |
| 12:29.47 | nilesh | [TK]D-Fender: no NAT involved, the phone and the server connected directly to a switch |
| 12:30.10 | [TK]D-Fender | nilesh: What about the other end of the call? |
| 12:30.23 | nilesh | [TK]D-Fender: other end is mobile / bluetooth |
| 12:30.37 | [TK]D-Fender | nilesh: What is this "bluetooth" part? |
| 12:30.45 | nilesh | [TK]D-Fender: chan_mobile module |
| 12:30.49 | [TK]D-Fender | nilesh: and "mobile" does not confirm how * is talking to it... |
| 12:30.59 | [TK]D-Fender | nilesh: Show us the call. |
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| 12:31.21 | nilesh | [TK]D-Fender: referring to pjsip logs? |
| 12:31.33 | [TK]D-Fender | nilesh: live debug from * CLI |
| 12:31.47 | nilesh | ok |
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| 12:53.42 | nilesh | [TK]D-Fender: https://gist.github.com/anonymous/d89fd3f0401c844974fa |
| 12:54.25 | nilesh | I ran tcpdump on my asterisk server and except for the initial SIP / SDP negotiation nothing was going through in the case of phone as well as in the case of softphone client |
| 12:54.37 | [TK]D-Fender | nilesh: we need SIP debug enabled... |
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| 12:57.17 | nilesh | [TK]D-Fender: https://gist.github.com/anonymous/b5a54f210acd6facec22 || have replaced my mobile number with token 'mobile_number' |
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| 12:58.55 | [TK]D-Fender | nilesh: Ok, media seems fine in there. you have 2 subnets with standard routing between them? |
| 12:59.18 | nilesh | [TK]D-Fender: subnet is 255.255.248.0 |
| 13:00.11 | [TK]D-Fender | unusual, but ok |
| 13:00.53 | [TK]D-Fender | I generally recommend sticking to standard RFC address space for cetain things that go out of whack when using custom ones like that |
| 13:01.02 | [TK]D-Fender | nilesh: Time to check your firewall... |
| 13:01.08 | [TK]D-Fender | iptables --list |
| 13:01.14 | nilesh | it's empty |
| 13:01.28 | [TK]D-Fender | double-check, including default type |
| 13:01.34 | nilesh | yes it's empty |
| 13:01.43 | nilesh | and default is accept |
| 13:02.23 | [TK]D-Fender | ok, just about out of ideas at this point... |
| 13:02.29 | nilesh | [TK]D-Fender: I can't do much about the subnet because we use that large subnet in office |
| 13:02.44 | nilesh | [TK]D-Fender: eventually I'll be moving the setup to a new network, but this should work |
| 13:02.50 | [TK]D-Fender | nilesh: Was a general warning, not necessarily specific to * |
| 13:03.10 | [TK]D-Fender | nilesh: the negotiation seemed ok, so we'll leave it at that... |
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| 13:03.20 | nilesh | [TK]D-Fender: I'm wondering if the phone has some issue, but the interesting thing is inter-SIP calls work just fine |
| 13:03.33 | nilesh | but yes, the inter SIP calls were tested with softphone and hardphone |
| 13:03.38 | nilesh | not two hardphones |
| 13:03.41 | [TK]D-Fender | nilesh: between multiple SIP devices and leaving chan_mobile out? |
| 13:03.46 | nilesh | yes |
| 13:03.59 | [TK]D-Fender | nilesh: Certainly pins the fault on that leg of the call then... |
| 13:04.39 | nilesh | but whatever I say on the other side (i.e. the mobile phone to which I call from SIP) gets recorded and is audible with softphone as a client! |
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| 13:05.31 | [TK]D-Fender | just no audio OUT, right? |
| 13:05.37 | nilesh | right |
| 13:05.40 | nilesh | audio OUT from the phone |
| 13:06.01 | [TK]D-Fender | not sure on what factors can contribute to that legs issue |
| 13:06.35 | file | unless it's chan_mobile itself or the bluetooth stack, which is possible |
| 13:06.59 | nilesh | file: it can't be chan_mobile because it's working when I use a softphone as a SIP client with the same codec that the hardphone supports |
| 13:07.17 | file | it could still be some weird nuance with chan_mobile |
| 13:08.20 | nilesh | since [TK]D-Fender pointed out networking issue I'm thinking of adding a bridge and running tcpdump over the bridge so that I get to see if the packets are going back and forth from the phone to server |
| 13:08.25 | nilesh | I can't see anything at the server |
| 13:08.33 | nilesh | after the initial negotiation |
| 13:09.21 | [TK]D-Fender | you wouldn't get one-way audio though |
| 13:09.27 | [TK]D-Fender | you can't have a codec work only one way |
| 13:09.55 | [TK]D-Fender | and chan_mobile shouldn't be using a "codec" per se |
| 13:10.05 | nilesh | initially I suspected codec mismatch issue then traced through SDP the codecs and set up same on softphone (PCMU) and it works perfectly |
| 13:10.15 | nilesh | this seems more of an issue with the phone or something to do with the network |
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| 13:48.21 | nilesh | [TK]D-Fender: so the bridge thing didn't help any much either because RTP was streaming constantly |
| 13:51.30 | nilesh | I hope it's really a phone issue... going to order a new phone to check if it works, *fingers crossed* |
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| 16:05.49 | linocisco | hi all, which CRM is free and highly compatible to be used with Asterisk? |
| 16:06.44 | linocisco | I dont want limited version of CRM that persuade to commercial one in the end. |
| 16:07.03 | Chainsaw | linocisco: So use SuiteCRM, not SugarCRM. |
| 16:07.28 | linocisco | Chainsaw, are you using that? no issues? |
| 16:08.19 | Chainsaw | linocisco: We use it. We have custom integration of SuiteCRM data for inbound/outbound calls. |
| 16:08.36 | Chainsaw | linocisco: I wouldn't say it's out of the box Asterisk compatible, but then again, few things are. |
| 16:09.08 | linocisco | Chainsaw, feeling confident to hear from one active user. thanks |
| 16:09.30 | Chainsaw | linocisco: We were on SugarCRM and had to abandon it because they went commercial in a very nebulous way. |
| 16:10.00 | linocisco | Chainsaw, i have heard only vtigerCRM and sugarCRM . one of this is integrated in Elastix but I have not tried. On their website , free versions of those have limitation. |
| 16:11.27 | [TK]D-Fender | Of course there are limitations.. why do you think there is a commercial version? |
| 16:11.36 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 16:11.38 | [TK]D-Fender | The question is do you CARE about what those limitations are? |
| 16:19.34 | linocisco | [TK]D-Fender, free version of asterisk and Ubuntu for example are enough for production environment to use. No limited functionality designed to buy commercial one in the end. I want such CRM like them. |
| 16:20.56 | [TK]D-Fender | Well all the GOOD stuff like that is mixed-model distribution |
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| 16:39.23 | drab | is there any way to have multiple destination Channels in a call file? I've seen the & syntax elsewhere but it doesn't seem to work |
| 16:39.47 | file | you can do a Local channel to the dialplan which calls multiple using Dial() but otherwise no |
| 16:39.48 | [TK]D-Fender | Show us |
| 16:39.53 | drab | btw regarding PA systems I was asking about yesterday, voip-info actually has an example here using call files: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
| 16:40.26 | drab | same => n,System(echo "Channel: SIP/bt200&SIP/lip6804n\\nContext: users\\nExtension: 101" > /tmp/${UNIQUEID}.call) |
| 16:40.32 | drab | I was trying that, but it breaks |
| 16:41.08 | drab | without the &SIP/lip6804n it works just fine |
| 16:41.25 | drab | and so it does if I have multiple files for each of those channels, just not if I try to combine them |
| 16:41.37 | drab | since they are all doing the same thing I was trying to save some bits |
| 16:41.40 | [TK]D-Fender | drab: Then follow what file said and dial a local channel and issue a Dial() in there |
| 16:41.48 | drab | ok, thank you |
| 16:41.50 | drab | file: thanks |
| 16:42.52 | drab | question tho, are those dials in the local channel going to be async? otherwise I'm back to the problem I had with Originate |
| 16:43.10 | drab | I moved to call files because I need all those extensions to be called at the same time |
| 16:43.30 | [TK]D-Fender | drab: yes it dials them all simultaneously |
| 16:43.37 | [TK]D-Fender | drab: that's what Dial() does.... |
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| 16:45.03 | drab | ok, thank you, looking into the local channel stuff |
| 16:45.35 | [TK]D-Fender | Local/extension@context/n |
| 16:59.16 | drab | https://wiki.asterisk.org/wiki/display/AST/Using+Callfiles+and+Local+Channels |
| 16:59.39 | drab | in the example the first priority is a NoOp, is that some kind of best practice to make sure things have time to "get setup"? |
| 16:59.45 | drab | since there's no Answer() |
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| 17:04.12 | rmudgett | drab: The NoOp is there since the first priority of an exten must be 1. The other priorities can be automatically incremented using the n priority. Doing it this way allows you to move extension lines that actually do something around without worrying about renumbering the priorities. |
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| 17:05.26 | drab | ah, makes sense, thank you |
| 17:13.06 | file | stromboli, stromboli, stromboli |
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| 17:30.04 | drab | I must be doing something wrong with that call file + local channel because calls are sequential: http://pastebin.com/gyKdUxT7 |
| 17:30.11 | drab | that has both config and console log |
| 17:30.29 | drab | it works, but the phones are definitely dialed in series, the second one rings only after the first one timed out |
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| 17:32.20 | drab | oh, should have I used only one Dial with &? /me tries that |
| 17:33.06 | file | yes. |
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| 17:37.30 | drab | ok, one step forward, but the confbridge failed. both rang at the same time but as soon as I picked up both hung up |
| 17:38.17 | drab | I'm wondering if that 1s silence I have in the call file is getting in the way somehow. With only the first line I was getting an error the conf file must specify an app or extension and that silence seemed to be used as a dummy on the wiki page I was looking at |
| 17:39.32 | drab | also I noticed that both rings and if I pick up either immediately the other will stop ringing like the call was interrupted |
| 17:40.28 | drab | here's an updated log with the confbridge stuff: http://pastebin.com/ku31CXuy |
| 17:41.07 | drab | as I said earlier I've tested things individually and the phones work fine calling into the confbridge, so that part seems ok |
| 17:44.30 | robmal | What are you trying to achieve? |
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| 17:46.09 | drab | robmal: calling an extension results in a number of phones being called and pulled into a confbridge |
| 17:48.12 | robmal | Ok, so use Application: ConfBridge\r\nData: 1234 in your callfile. |
| 17:52.31 | drab | robmal: thanks, that sort of worked, the issue of the other phone being hung up upon is still there. both phones will ring, when I pick one up I now correctly hear the confbridge message, but the second phone stops ringing |
| 17:53.23 | file | that's how & behaves |
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| 17:53.36 | file | if you want to dial all and put them all in, then you need individual originations |
| 17:53.41 | robmal | Yes. |
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| 17:54.51 | robmal | Just create two callfiles, with Channel: pointing to each phone. |
| 17:55.06 | drab | should I go back to individual conf files like in the beginning then? |
| 17:55.08 | drab | right ok |
| 17:55.11 | drab | makes sense, thank you |
| 17:55.31 | drab | I guess I didn't explain myself clearly early on, sorry about it |
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| 18:13.44 | drab | ok, it's all working, thank you very much guys |
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| 19:45.12 | Dpunkt | how can i see when a outgoing pjsip registration is about to expire? |
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| 20:22.18 | monsterco | Hi everyone, I would like to do a [POST] to another public server - thinking of sending variables through bash or something. What is the best way? |
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| 21:34.11 | monsterco | Anyone know how to put SIP-23G on hold? |
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| 21:43.41 | [TK]D-Fender | My guess : press "hold" |
| 21:51.47 | danielbw | welp I started a Sangoma support ticket |
| 21:51.55 | danielbw | so we'll see where that goes |
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| 23:37.37 | Get_The_Fish | Hello all, what are the PJSIP channel variables? Are they the same as the SIP channel variables and I can just change the name? |
| 23:46.25 | Get_The_Fish | So, for example, SIPFROMUSER does not work. I'm assuming thats because I am using PJSIP. Am I wrong in that assumption> |
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