Archives : January-2021
H All,We have a carrier that sometimes when sending DTMF will go backwards with the timestamp. If we look at the Sequence numbers and timestamp the sequence numbers are in order however the timestamps are not. For instance seq: 28867 timestamp: 593496..
Running asterisk 18 in a FreeBSD 12-STABLE platform, asterisk.conf contains some FreeBSDstyle delegations of the directoris used by default.First of all: the configuration of the Asterisk instance is done in a simple, plain way using static config files..
Hello! I was wondering what (if there is any) the equivalent of progressinband=never for chan_sip is when using chan_pjsip? What I want is to always send 180 Ringing to the caller, even if a 183 Session Progress was received from the called party..
I am suddenly having my log flooded by these warnings: WARNING: Retransmission timeout reached on transmission 1372443000-1325687561-974574282 for seqno 1 (Critical Response) WARNING: Retransmission timeout reached on transmission 283912657-2003707522-1284343..
All;We were using Digium cards, now I am not able to reach for digium website that contains the telephony cards and Asterisk website currently is taking us for Sangoma, so what happened in Digium cards?Re..
Stir Shaken Asterisk cannot do that, but my company can give you Stir Shaken for Asterisk, via ODBC, any version. Please contact me via email venefax at the google mail system Phil..
All.We have old Asterisk servers, 1,89, (we cannot upgrade because of several reasons) and we are now implementing SHAKEN via our provider.We place a SIP call to our provider and they return a 302 (information below).I am trying to get the X-Ident..
could somebody drive me how could I make run presence reporting by BLF feature on the Cisco SPA525G2 with SPA500DS on asterisk with pjsip stack? I am not able to configure asterisk side. When I run pjsip show subscriptions inbound I see all subscripti..
I have a box that I suspect had timing issues. I added a TE131 to see if that would help. Is there any way for me to verify that Dahdi is using the card for timing and not t..
I am unable to figure out why I am not able to blind transfer when I am the caller and I call the extension defined by gosub. When running asterisk -rvvv I can see: — Channel PJSIP/-00000009: Dialed @gosub-stdexten does not exist. It is evident th..