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| 11:58.47 | andremar | Hy everyone. Anyone can tell me if thereâs any way to configure Asterisk to save/get sip dialogs information to/from a database? The purpose behind the question is investigating the possibility of setting up a High Availability architecture where the calls could stay up even after an active server state change. |
| 12:00.00 | file | There isn't. |
| 12:00.12 | file | You would need to write custom code to accomplish such a thing. |
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| 12:01.26 | andremar | Thanks @file. I was led to believe that Asterisk Realtime would handle this, but thatâs just for configuration on a database, and no reloads, right? |
| 12:01.35 | file | yes. |
| 12:01.45 | andremar | Ok. |
| 12:01.52 | andremar | thanks for the clarification. |
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| 12:15.50 | Samot | Though you can't keep a call up if the server it's communicating with goes down. |
| 12:15.58 | Samot | You're going to lose that call. |
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| 12:32.32 | andremar | @Samot, there are ways, I was just wondering if Asterisk already had some mechanism to pull it off. |
| 12:32.56 | Samot | No. The only way you can do it is if you have it setup with some HA functionality. |
| 12:33.22 | Samot | Where the IP floats between the servers and the servers can monitor each others "heartbeats" |
| 12:33.41 | andremar | sure, that was part of the question. |
| 12:34.06 | Samot | Also the most complex. |
| 12:34.12 | Samot | They will need to be on the same LAN. |
| 12:34.27 | Samot | Be able to talk back and forth with each other just on that LAN. |
| 12:35.09 | somepoortech | andremar: you would probably be better off running it in a VM fault tolerant or a FT hardware solution |
| 12:35.50 | Samot | Well... |
| 12:35.56 | Samot | How active is this server going to be? |
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| 12:37.20 | andremar | for now I just want to see it work, to have the call stay up when the active server goes down |
| 12:37.55 | Samot | You're a bit aways from that. |
| 12:38.05 | andremar | then, it depends on what constraints the system brings |
| 12:38.14 | Samot | You need to figure out how you're going to handle the HA portion. |
| 12:39.15 | andremar | maybe not with Asterisk alone |
| 12:39.24 | Samot | Asterisk has nothing to do with it. |
| 12:39.30 | andremar | sure it has |
| 12:39.36 | Samot | Not at this point. |
| 12:41.06 | andremar | if there is no way to have the call dialogs information shared between two servers, then what I though is not possible with Asterisk alone, whatever the HA configuration may be |
| 12:41.37 | Samot | Oh you'll probably want something like Kamailio |
| 12:41.40 | Samot | I use it for this. |
| 12:41.45 | andremar | maybe |
| 12:41.53 | Samot | Not HA but active spares. |
| 12:42.12 | Samot | Primary goes done...Kamailio routes requests to the other server. |
| 12:42.38 | andremar | do you have some links to share about that? |
| 12:43.40 | Samot | http://lmgtfy.com/?q=kamailio+with+asterisk |
| 12:45.15 | Samot | However, I am currently unaware of how you would be able to transfer an active call leg between Asterisk servers... |
| 12:47.33 | Samot | Again, is this for a solution need or is it just "I wanna try it"? |
| 12:49.45 | andremar | someone asked is if it could be done |
| 12:50.12 | andremar | I think kamailio has some module to understand call dialogs |
| 12:50.30 | andremar | maybe that could be used to save it to somewhere where the two servers could read it |
| 12:50.41 | andremar | weâll see |
| 12:52.10 | Samot | It's not about the dialogs. |
| 12:52.26 | Samot | How is the second server going to know anything about the call? |
| 12:52.50 | Samot | In regards to audio, the channels it has connected with endpoints |
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| 13:53.28 | tirej | hi everyone |
| 13:53.52 | tirej | can we write AGI script inside IVR key_pressed events with Elastix ? |
| 13:54.03 | tirej | or can we collect dtmfs between call start < # > call end and at the end send them to the an API? |
| 13:54.34 | tirej | or with another pbx-interface ? |
| 14:03.53 | [TK]D-Fender | <tirej> can we write AGI script inside IVR key_pressed events with Elastix ? <-not sure what you're talking about here |
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| 14:05.31 | tirej | [TK]D-Fender, think of a IVR say: welcome to x Corp. if you know the ext. bla bla press it if not press 1 .. when the caller press 1 'scenario i am talking about |
| 14:06.05 | [TK]D-Fender | You'll have to make your own custom dialplan for that IVR |
| 14:06.14 | [TK]D-Fender | and not use any of the GUI constructs for it |
| 14:10.28 | [TK]D-Fender | For in-call detection .. you can pretty much forget that with the GUI altogether |
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| 14:19.22 | andremar | continuing on the quest. Does Asterisk save call state to AstDB, anyone? |
| 14:19.45 | file | no. |
| 14:19.47 | file | it does not. |
| 14:22.31 | andremar | is it just in memory then @file? |
| 14:22.45 | file | yes. |
| 14:23.03 | andremar | ok, thanks @file |
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| 15:28.03 | *** topic/#asterisk by file -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.10.0 (2016/07/21), 11.23.0 (2016/07/21), Standard: 14.0.0-beta2 (2016/08/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 19:38.47 | vane- | So I am stuck on something here, I am trying to execute 'pjsip send notify cisco-reboot 1801' which is essentially just an event to reboot the phone. 'Event=>restart_now', however, I get a response back 'Unable to create request with auth. No auth credentials for any realms in challenge.' How do I get asterisk to respond with auth credentials? |
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| 22:18.57 | epaphus | Hello all |
| 22:19.51 | epaphus | I will pay via paypal $30 if somebody can help me in this issue. Calls from the SIP trunk when entering a queue get interrupted as soon as an agent answers. |
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| 22:23.14 | epaphus | The error message is : CHANUNAVAIL HANGUPCAUSE: 21 |
| 22:23.33 | epaphus | Calls between extensions work without issues |
| 22:24.17 | voipmonk | nice - show the call. |
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| 22:31.02 | rwb | Hi, I have a Digium TDM2400P pci card with 6 S400M modules. I installed AsteriskNow, but I'm not sure that the card is working, or if the driver is installed. Is there a simple way to test to see if the card is OK? I don't see it in lspci |
| 22:31.51 | voipmonk | how are you powering the card, rwb/? :) |
| 22:32.30 | rwb | It's on the PCI buss. Shit did I miss a power supply! hehe |
| 22:32.38 | voipmonk | probably |
| 22:32.45 | voipmonk | little white plastic connector |
| 22:32.52 | voipmonk | that needs to be plugged in |
| 22:33.01 | voipmonk | molex connector, I call them |
| 22:33.04 | rwb | ahh, damn, I was wondering how all that power got to the ringers :) |
| 22:33.44 | voipmonk | turn it odd |
| 22:33.50 | voipmonk | power off the box first , please |
| 22:34.20 | rwb | of course!!! |
| 22:34.27 | voipmonk | thinks . . . o o o O ( uh huh ) |
| 22:34.28 | rwb | Thanks |
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| 22:45.55 | rwb | sweet. the admin screen now says I have "new hardware" carry on... Thanks again! |
| 22:48.39 | WIMPy | Interesting new hardware? |
| 22:51.18 | voipmonk | :) |
| 22:52.10 | rwb | I'm new to allot of the PBX stuff, but an old hack on Internet. Getting back to basics. This install is for a vintage style night club with a few old rotary phones |
| 22:52.21 | voipmonk | nice |
| 22:52.37 | voipmonk | wait till you see the phones that work with asterisk |
| 22:52.42 | WIMPy | Oh, pulse dialling again? |
| 22:52.48 | voipmonk | or just skip the phones and go straight to video |
| 22:52.55 | voipmonk | mind blown yet? |
| 22:53.54 | WIMPy | Videophone? I think for a night club a feelophone might be more like the cool thing. ;-) |
| 22:54.04 | voipmonk | oh my... |
| 22:54.23 | rwb | this is a vintage shop, pulse... "dial" 4 for a poem... I'm hoping to not have to use tone converter, but well see... |
| 22:54.24 | voipmonk | safe knowledge of others via the feelophone |
| 22:54.42 | WIMPy | It certainly needs a vibrator. |
| 22:54.47 | rwb | :) |
| 22:55.03 | rwb | by relay though... not to stress the card... |
| 22:55.42 | WIMPy | Although a really old ringer might actually qualify as vibrator as well. |
| 22:56.34 | rwb | baa! nice. This is going to be a fun project for sure! |
| 22:56.53 | voipmonk | your kung fu will be strong if you keep it fun. |
| 22:57.17 | voipmonk | pay close attention to the evidence and we won't see you very often. |
| 22:57.44 | rwb | There are airport runway glass domes involved... |
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| 23:12.38 | r10L | Anyone know why incomming calls might be incorrectly routed every-other call? We have two incoming lines in a group. |
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| 23:13.34 | voipmonk | please show us the call, you may use pastebin.ca or similar |
| 23:13.52 | voipmonk | show the calls. :) |
| 23:14.22 | Tekati | Running FreePBX with TE131F card. When callers call in to IVR and enter extension the extension is wrong 9 out of 10 times. For instance user dials 101 and gets 110. |
| 23:17.35 | voipmonk | stdby |
| 23:18.17 | r10L | figures...now my call was routed just fine 3 times in a row...might be something with having to use both lines at the same time |
| 23:18.41 | voipmonk | :) |
| 23:19.06 | voipmonk | don't feed the gremlins after midnight |
| 23:19.33 | voipmonk | I skipped right to gremlins.. I apologize... |
| 23:20.31 | voipmonk | they are called mogwai's |
| 23:22.28 | voipmonk | I would look into ;relaxdtmf=yes but watch it closer |
| 23:24.18 | voipmonk | you can also adjust the regain up or down |
| 23:24.25 | voipmonk | rxgain |
| 23:25.41 | voipmonk | r10L: adjust the rxgain and or also relaxftmf |
| 23:25.48 | voipmonk | and or also :) |
| 23:26.23 | r10L | ok, I'll give those options a shot and post back once I have some more call info or if it's fixed. Thanks |
| 23:26.31 | voipmonk | r10Lsorry, that was for Tekati |
| 23:26.47 | voipmonk | r10L no no sorry |
| 23:26.57 | voipmonk | r10L you need to show the call first |
| 23:27.12 | voipmonk | Tekati: adjust the rxgain and or also relaxdtmf |
| 23:27.29 | r10L | lol ok. I thought you had a wild stroke of genius. I'll share when I have the data collected . |
| 23:27.32 | voipmonk | r10L use pastebin.ca or similar to show us the debug |
| 23:28.19 | voipmonk | yes , wild - check stroke... yep ... genius ? nah - I need to get some sun so my super power gets energized... |
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| 23:55.23 | ChannelZ | Hmm. Anyone else notice an uptick in SIP drive-bys today? |