Archives : February-2016
I am trying to use Authenticate() in the dialplan for something other than my password. The message says Please enter YOUR password followed by the pound key.Im not using this for my password. Is there any way to change the message to please enter ..
Ive recently given a try to a Digium D70 phone.At the moment, Im configuring them though config files with a DHCP server and not using DPMA. Of course, Im connecting them to Asteris (PJSIP stack on 13.7.0).Which is the best place to:- read about p..
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..
Perhaps this is not limited to Grandstream GXP 2000 phones, but those are the phones we are using.Using FreePBX.Retrieving a voice message (*97) works fine from Line 1. Retrieving a voice message (*98) and picking the extension (Comedian mail)works f..
When making an outgoing call with an pjsip endpoint hints are reporting idle until we get ringing from the other side:Asterisk 13.7*CLI> core show hint wid1765wid1765@company_4_hi: PJSIP/wid1765 State:Idle Presence:not_set Watchers0But with chan_..
all,My goal is to trunk two Asterisk servers together using res_pjsip. Im really not familiar with res_pjsip, having only used chan_sip over a year ago now. So, I apologize in advance if this is an overly basic question.Im using the below configurat..
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..
All,I am looking for a class 5 platform (basic VAS) and softphone (IOS, Android) both supporting ZRTP protocol to achieve the highest voice security. C.5 and UA should be delivered from the same supplier (like sipwise for instance)Could anybody recomm..
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I am using Asterisk 13.7.0 with PJSIP.I set up Asterisk for use with WebRTC SIP clients. After I managed to get video working, I noticed, that the calling party receives no video till 90s (or so) have passed. After 90s both parties receive video perfectl..