Line 1: Sipura SPA-2000 Line 1 Line 2: Sipura SPA-2000 Line 2
I have a single POTS line (Verizon) connected to a Digium X100P card.
[Config excerpts are below]
Issue: Calling into the * server rings the Siemens base station as expected. After about 20 seconds the Siemens base station answers the call and sends the call to voicemail (I know I should use VM in * but my spouse isn't yet ready for that...).
== Spawn extension (from-verizon, h, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'Then the caller hangs up. Asterisk "sees" this hangup and disconnects the PSTN line (I confirm this with a monitor phone on the POTS line).
Asterisk sends the SIP BYE command, and receives the OK response from the Sipura. For some reason the Sipura doesn't hangup line 1 and, as a result, every message on the Siemens base station has a ~15 second Congestion tone trailer.
Any ideas? Most likely the answer is "bug in Sipura, move quickly to *-based VM, right? :^)
Relevant zapata.conf entry: ------------------------------------- [channels] language=en context=from-verizon signalling=fxs_ks usecallerid=yes echocancel=yes echocancelwhenbridged=yes channel => 1
Relevant extensions.conf entry: ------------------------------------- [from-verizon] exten => s,1,Dial(SIP/sipura_line1) exten => s,2,Hangup exten => h,1,Hangup exten => i,1,Hangup
Relevant sip.conf entry: ------------------------------------- [sipura_line1] type=friend username=sipura_line1 secret=XXXXXX host=dynamic context=from-siemens-line-1 mailbox=9000 canreinvite=no callerid = <8885551212>
-- Rob Page V: 540.361.1710 Zope Corporation F: 703.995.0412 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
