Archives : February-2016
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everybody!Is possible to integrate WhatsApp VoIP on Asterisk?Or is there some tricks for that? Like Yowsup?Thanks in..
Im discovering Digium D70 phone (1.4.2 firmware) with asterisk 13.When pressing phones Msgs button, I see its dialing extension 800 no matter what I edited in my phone config file. Relevant piece of phone config is:Ive tried in the above voicemail fie..
Im trying to setup a sip trunk. The asterisk version is Asterisk certified/11.6-cert12. I had to change the CallerID to a pilot number of the trunk as provider only accept it. but when i dial Im getting below error. Cannot hear anything[Feb 11 14:19:..
Please help find the cause of strange behavior res_pjsip.Making outgoint call to other sip server (CommuniGatePro), my asterisk suddenly sends BYE after picking up!Partial log of an outgoing call with full debug is attached and on web: http://pastebin.com/tLNCpx4..
its easy to control Snom phones from outside by an http request http://username:password@192.168.0.1/command.htm?key=KEYEVENThttp://wiki.snom.com/FAQ/Can_I_control_my_snom_phone_remotelyalso its easy to configure that an Snom phone send me an http requ..
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_..