Hey all, was hoping to get some help.
I have created a script that will drop a .call file into
/var/spool/asterisk/outgoing
Super simple: this works , calls the extension and then plays the gsm file.
Channel: sip/511
Callerid: 5555555555
application: playback
data: OfficeSpace
What I would like to do is intercom the phone with
Channel: sip/*80511
Callerid: 5555555
application: playback
data: OfficeSpace
But I am missing something, here is the error:
[2018-04-01 09:19:35] VERBOSE[13995] pbx_spool.c: Attempting call on sip/*80511 for application playback(OfficeSpace) (Retry 1)
[2018-04-01 09:19:35] VERBOSE[13995] netsock2.c: Using SIP RTP TOS bits 184
[2018-04-01 09:19:35] VERBOSE[13995] netsock2.c: Using SIP RTP CoS mark 5
[2018-04-01 09:19:35] ERROR[13995] netsock2.c: getaddrinfo("*80511", “(null)”, …): Name or service not known
[2018-04-01 09:19:35] WARNING[13995] chan_sip.c: No such host: *80511
[2018-04-01 09:19:35] NOTICE[13995] pbx_spool.c: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?)
[2018-04-01 09:19:35] NOTICE[13995] pbx_spool.c: Queued call to sip/*80511 expired without completion after 0 attempts
I have verified that the intercom is working on the phone from another station.