almost any useful CLI command has an analogue on Asterisk Manager Interface, but I cannot find a way to get the list of subscriptions using AMI. Which is the command, if any? The CLI command issip show subscripti..
Author : Leandro Dardini
Home » Articles posted by Leandro Dardini (paged 3)
Starting with asterisk 1.8, when you dial multiple channels at once and one of them is answered, all other channels were canceled with the cause 200 -Call completed elsewhere, so modern phones dont display the call asmissed.Do you know a way to trans..
do you know if it is possible to define the SLA configuration in the database for realtime usage with asteri..
I am struggling to have a SPA504G to auto answer (for intercom/paging). Ihave tried the following SIP headers (not all together), but without luck:SIPAddHeader(Call-Info:\;answer-after=0);SIPAddHeader(Call-Info: answer-after=0);SIPAddHeader(Alert-In..
while moving from asterisk 12.3 to asterisk 12.6, I see the MWI support for voicemail has stopped working. If I check sip show peer 104-DEVEL on asterisk 12.3, I can clearly see the Mailbox option set, while on asterisk 12.6 it appears empty.Is th..
have you noticed the message num (VM_MSGNUM) is off by one?For example, I receive the following message:Just wanted to let you know you were just left a 0:03 long message (number7)but in attach there is the msg0006…
When a call is transferred to another extension using a blind transfer, asterisk keeps traces of who is transferring in the BLINDTRANSFER variable. If instead the call is forwarded using most phone call forward feature, a302 Moved Temporarily is s..
I am migrating my dialplan from asterisk 11 to asterisk 12 and it seems the${CDR(start)} is not returning any data. Other functions, like${CDR(duration)} or ${CDR(src)} or ${CDR(accountcode)} are returning correct values. Where is my mistake? Has t..
When you use a product which version number is 11 or even 12, you might go with the assumption all big bugs are fixed and then you find there is a huge, important, expensive bug still running in the code we are relaying upon…The problem is simp..
I have an asterisk box with a peer configured with nat=force_rport,comedia, but asterisk keeps sending the audio to the private IP address and ignoring the client peer nat settings.If I check the sip show peer extension, I see both symmetric RTP ..