I am getting flooded with these messages: [Mar1 12:25:29] WARNING[6962][C-0000005a]: chan_pjsip.c:712 chan_pjsip_write: Cant send 10 type frames with PJSIP [Mar1 12:25:30] WARNING[6962][C-0000005a]: chan_pjsip.c:712 chan_pjsip_write: Cant send 10 t..
Author : Carlos Chavez
Home » Articles posted by Carlos Chavez (paged 3)
I am having a problem trying to compile dahdi-linux-complete-2.11.0+2.11.0 on a CentOS 7.2 server.Version 2.10.2 compiles fine.Is there a new dependency for 2.11.0 that was not required for previous versions?Here are some of the errors I get: INST..
Is it possible to use serveral protocols for a single transport section in pjsip.con?In sip.conf you could use transport=udp,ws,wss so you cound use webrtc along with your phones but if I try: [transport-udp] type=transport protocol=udp,ws,wss bind=0.0…
I use realtime on my asterisk installation.I have always used mysql for my realtime connection but as mysql seems to be on the soon to be deprecated list of asterisk features I am trying to move to ODBC (still using MariaDB/Mysql on backend).I find O..
I am trying to get cdr via odbc to work on Asterisk 13.7.2 but I keep getting this error: [Feb9 16:21:43] WARNING[2088]: cdr_odbc.c:160 execute_cb: cdr_odbc: Error in ExecDirect: -1, query is: INSERT IGNORE INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,peeraccount,linkedid,sequen..
I am trying to port our Asterisk front end to Asterisk 13 but I cannot get realtime static to work.Realtime for PJSIP, Voicemail and Queues is working fine so I know res_odbc is configures properly.In past versions of Asterisk I was using Mysql (res_config_mys..
I just purchased an Amfeltec USB-FXO adapter and am trying to compile DAHDI 2.10 on a Raspberry PI running Pidora 2014 R3.I have all the dependencies but I get an error and cannot finish.Is it even possible to compile DAHDI for the ARM plataform?H..
I am having a small problem that is driving me nuts.I can make calls over my Webrtc client without any problems and audio sounds fine. The only problem I have is that when I call an internal SIP extension on my PBX I do not hear the ring while I w..
Ia had a server overload today because someone did a call forward to their own extension.To do a call forward I write a key called CFWD with the extensión number and number to dial .The main script tests if the key/value exists and dials the num..
We are having a strange problem today.We have a SIP trunk from a provider and incoming calls are being dropped after the IVR when attempting to connect to any internal phone.If a you dial a DID that goes directly to a phone you can talk but the c..