Archives : September-2015
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HiWe are using 1.8 on CentOS 6We use asterisk servers in pairs for machine level failover. On a recent pair we pointed the astdb location of both nodes of the pair to the same location on a shared storage device. Now it would appear that if the aster..
I am using asterisk 13.4 and in some calls I got 2 cdr records, one with usual dial on last app and correct billsec and a second one with hangup as last app and billsec equals zero.I am using cdr_pg..
Cant get MWI working with PJSIP and my Cisco phones and realtime. I have mailboxes populated in the endpoints and aors tables, with 312@default which is the voicemail context. Im not sure what else to try.Please help..
Ive not used analog for quite some time. It seems its not possible in asterisk to spoof a phone number/name on an an..
Does something change with MWI when moving from SIP to PJSIP? Seems my phone isnt be alerted of i..
I have a TDM400P analog card in my asterisk server. I havent used analog for a while. The caller hears at least two rings before my 312 extension gets rang internally. Does it usually take that long? Below is my relevant dialplan. Also callerID i..
Does anyone have any information for me?Welinghton.Citando Welinghton Magno Guimaraes : WELINGHTON MAGNO GUIMARãESDIRETORIA DE TECNOLOGIA DA INFORMAçãO – DIVISãO DE VOZFONE: (38) 3532-1285 OU (38) 3532-1200 – RAMAL: 8245 OU 8251UNIVERSIDADE FEDE..
I have a client that has a 24 channel voice T1 that I have been using e&m signalling over for a number of years.The local telco finally got an ISDN switch and wants to move them to PRI.I didnt see this as a big problem – Ive done a few others on t..
How have the same sip username in several realms ?For now, I must add the realm prefix in the sip username of chan_sip.For example:[lg_2540]amaflags = default call-limit = 10host = dynamic language = en_UScontext = lg_default callerid = LG secret = XXXXXXXXXXXXXXXXXXXXXXXXXXt..
I am using the asterisk 13 and I config my dialplan for the SIP messaging as the following :http://highsecurity.blogspot.com/2012/03/asterisk-10-110-sms-messaging-or-sip.html[astsms]exten => _.,1,NoOp(SMS receiving dialplan invoked)exten => _.,n,NoOp..