Archives : November-2017
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Hey Everybody,As a project, we would like to do a better job of getting additional information about new developments in Asterisk to the community.Ithink this is something I have struggled with in the past (to some extent) and would like to improve u..
Get Outlook for AndroidFrom: asterisk-users-bounces@lists.digium.comon behalf of asterisk-users-request@lists.digium.com Sent: Monday, November 6, 2017 6:00:01 PMTo: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 160, Issue 5S..
Running 15.1.2.I have four devices transitioned to use pjsip.After about 1-2 days of uptime, psjip stops accepting registrations, and the messages log contains the entry as per the subject.At any given time, pjsip show contacts only shows the four devices.Co..
all!Asterisk 13.1.0 Ubuntu 16.04, all latest. Can anybody explain this to me – I run Originate command from dialplan: same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum})and I get crazy sound distortion in the conference, and I ..
List Next question where google did not spit out an unsable answer. When redirecting a call with Transfer, I would like to correctly indicate the reason. I did try this: exten => XX,1,NoOp(Call to ${EXTEN} from ${CALLERID(all)}) exten => XX,n,Dial(SIP/..
Dear List I am testing various early audio scenarios with different voice ICs, phones and pbxes. In Switzerland, when you operate a value added number, you have to announce the price of the call, usually in early audio, before the call is establish..
Im trying to supervise an existing Voicemail box with a BLF button on Debians asterisk 13.14.1 system.I mostly found this [1] document. I added in a context a line like:exten = *7000,hint,MWI:31@defaultWith core show hints, I can read this:*7000@su..
List I am in the progress of migrating from chan_sip to pjsip. I fear I have missed something on how hints need to be specified for pjsip. For chan_sip I have configured sip.conf subscribecontext = localuser and in the dialplan I set: [localuser] ex..
In my dialplans, Im currently using PJSIP_AOR to check the status of a contact before dialling so that I can route the call differently if the endpoint is offline. But PJSIP_AOR seems to take about 0.9 seconds to return. If Im checking 10 endpoin..
Ive seen that Asterisk stores in ASTDB entries like:/SIP/Registry/spa3102 : 192.168.64.207:5060:3600:7013:sip:spa3102@192.168.64.207:50601. My understanding is that any peer that sent to Asterisk a REGISTERmessage has such entry set. So having th..