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| 06:34.33 | nickgaw | Hi, What free or pay services provide voice to text where a user could say something and the asterisk system would process that like several companies do? |
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| 09:57.02 | poolson | hey i have some phones that keep going from reachable to unreachable |
| 09:57.05 | poolson | continuously |
| 09:57.18 | poolson | what is typically the culprit here ? |
| 09:57.37 | poolson | i was doing some reading about it and people talk about nat and no nat and qualify and not qualify etc. |
| 09:57.49 | poolson | but it seemed like a bit of hit and miss and didnt really do anything for the problem |
| 09:59.26 | dan_j | where are the phones in relation to your server? |
| 09:59.45 | dan_j | same or different network? |
| 10:01.51 | dan_j | If it's over the internet, I would start with a voip test to see if there is any packet loss on the phone network. |
| 10:02.00 | dan_j | http://myspeed.visualware.com/index.php |
| 10:03.42 | poolson | yup different network. asterisk is hosted somewhere on a vps |
| 10:03.46 | poolson | actually freepbx |
| 10:04.08 | dan_j | check the connection on the phone side using the voip test and quality test on that link above. |
| 10:04.38 | dan_j | also, with a vps is it powerful enough to host asterisk? if it's not got enough cpu time, things might go a little crazy. |
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| 10:24.43 | poolson | hey also what would cause messages telling me a device was reconfigured because another device took over its config |
| 10:25.08 | poolson | im using MAC addresses to pull specific configs .. would that mean that is all getting confused ? |
| 10:30.00 | drmessano | poolson: What version of Asterisk? |
| 10:30.07 | poolson | hey again dude |
| 10:30.12 | poolson | got disconnected the other day |
| 10:30.15 | dan_j | Never seen that message. Asterisk is giving it you? Are you using sip or pjsip? Please pastebin the output. |
| 10:30.22 | poolson | 13.6.0 |
| 10:30.32 | drmessano | You need to upgrade to 13.7 |
| 10:30.56 | poolson | i have a phone at a nother location and that is set NAT=no |
| 10:30.56 | drmessano | Should fix your disappearing phone issue |
| 10:31.10 | poolson | and seems to not be having the reachable/unreachable issue |
| 10:31.21 | poolson | ok ill give that a go |
| 10:31.22 | drmessano | Some will, some wont |
| 10:31.25 | drmessano | Update |
| 10:33.17 | poolson | yah ill give it a shot ! |
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| 10:33.35 | poolson | surely theres some magic setting i can flip on the firewall at the location that seems problematic ? :) |
| 10:35.55 | poolson | lol googling for the particular router ... |
| 10:35.57 | poolson | You want at least version 1.1.1.06 here. The earlier versions had a bug that makes phones register and de-register from the pbx constantly. If using a cisco RV320 get the LATEST firmware as it has a option to set UDP timeout. |
| 10:36.05 | poolson | lol wtf .. always something |
| 10:38.14 | poolson | actually i have a later version |
| 10:46.51 | poolson | setting SIP ALG on it has made that problem go away |
| 10:47.09 | poolson | however when i did that before i noticed those phones went into a loop of trying and retrying to fetch their configs |
| 10:58.08 | poolson | i'll have to check tomorrow to see if that config fetch thing is still happening |
| 10:58.14 | poolson | but for now it seems ok |
| 10:58.23 | poolson | ill do the update to 13.7 too and see how that goes ! |
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| 11:16.47 | drmessano | The update is the only thing that will fix that isseu |
| 11:16.49 | drmessano | issue |
| 11:17.11 | drmessano | SIP ALG is _always_ a bad idea |
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| 11:38.49 | ModFather | Hello, i have asterisk 13 and i would like to ( record all the calls ) but i dont know where i can see how to, does anyone know where i can see help about this? |
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| 13:49.29 | ModFather | does anyone knows how i can run an .sh script from asterisk right after a call ends ? |
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| 16:12.47 | dan_j | In Asterisk 11, I was able to use AMI to start recording on the outbound channel. IE the channel leaving asterisk. That way, when a call is transferred around asterisk, the full call to the callee is recorded. |
| 16:13.29 | dan_j | In Asterisk 13, call bridges seem to be simply a group. How can I work out which channel is leaving asterisk? |
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| 20:54.39 | Khronos | Hi guys. |
| 20:55.20 | Khronos | I am wanting to setup two Asterisk server for phones to t register too. |
| 20:55.30 | Khronos | One on the east coast and one the west coast. |
| 20:55.53 | Khronos | some of the phones will register to east and some to west. |
| 20:56.42 | Khronos | What is the best way to make sure that any phone can call any other and the user not have to worry about which server they are actually connected too? |
| 20:58.11 | Khronos | I tried a simple dial plan where the server tries something like this: |
| 20:58.19 | Khronos | Dial(SIP/device) |
| 20:58.40 | Khronos | then Dail(SIP/exten@remote) as the next priority. |
| 20:59.27 | Khronos | Dial(SIP/extension@remote) but this seems to cause a routing loop because of the shared dialplan on each server. |
| 20:59.51 | Khronos | exten -> 1,1,Dial(SIP/peer1) |
| 21:00.02 | Khronos | exten => 1,n,Dial(SIP/1@remote) |
| 21:00.05 | WIMPy | Use different contexts for the phones and the other server. |
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