Calling linphone from asterisk 13.9.1.:Dial(SIPfirstname.lastname@example.org)And it works. But on the linphone side the caller is:@email@example.comIs there any way to make it more descriptive, at least for the sip user name ? I tried setting SIPCALL..
On 13.9. The cli log has these messages every 15 seconds. The end point to linphone on android………….[May 12 19:02:59] WARNING: chan_sip.c:3775 __sip_xmit: sip_xmit of 0x7effe40088b0 (len 608) to 10.10.11.95:37855 returned -2: Success[..
There is an opus patch for asterisk 11. https://github.com/seanbright/asterisk-opus/tree/asterisk-11 . But it doesnt have Packet Loss Resilience or Forward Error Correction, both of which are important for voip.2.1.6.Packet Loss ResilienceAudio cod..
In a fork of seanbrights opus patch for 13 there are further patches for Forward Error Correction and Package Loss Concealment, both of which ought to very useful in voip:https://github.com/traud/asterisk-opusAnybody used these patches ? Puzzled ..
To connect to google voice with xmpp, Ive had to turn on the less secure apps switch.My xmpp.conf :type=client serverhost=talk.google.com secret=mysecret priority%portR22usetls=yes usesasl=yes status=available statusmessage=Not availabletimeout=5Is th..
Id like to transfer all my pesky telemarketing calls to Jolly Roger .http://www.nytimes.com/2016/02/25/fashion/a-robot-that-has-fun-at-telemarketers-expense.htmlIn the middle of a call Id hit some DTMF sequence, which would dial Jolly Roger and trans..
ive got calls coming into an 11.21.0 box. The internal phones are analogue off a TDM400 board, and SIP extensions.Using an analogue internal phone, the remote party always hears an echo on its side. We do not hear an echo. Doesnt matter who is the call..
In setting up the GS-Wave softphone there are two id entries:SIP User IDSIP Authentication IDI would have thought SIP User ID was the devicename , i.e. [name].Then SIP Authentication ID was defaultuser.But not so. With[gs_5062](cell-phones)defaultuser=gs_62and..
Im not getting any ringing when I use option r with Dial:Dial(DAHDI/1-1, firstname.lastname@example.org,,rTt) in new stackOtherwise all works. The call goes through, good ..
on 11.17.1Trying to debug vm. as suggested by voip-in..