I have just set up my first asterisk system, and are battling with a hangup problem in my dial plan.
Here is the relevant part of the dialplan :
exten => spainbound,n,Playback(silence/2)
exten => spainbound,n,Hangup()
When executed this is what is logged at verbose level 10:
-- Executing [spainbound@home:9] Playback("SIP/asterisk-081df688", "silence/2") in new stack
-- <SIP/asterisk-081df688> Playing 'silence/2' (language 'en')
-- Executing [spainbound@home:10] Hangup("SIP/asterisk-081df688", "") in new stack
== Spawn extension (home, spainbound, 10) exited non-zero on ‘SIP/asterisk-081df688’
enabling sip debug does not help:
-- Executing [spainbound@home:9] Playback("SIP/asterisk-081df688", "silence/2") in new stack
-- <SIP/asterisk-081df688> Playing 'silence/2' (language 'en')
-- Executing [spainbound@home:10] Hangup("SIP/asterisk-081df688", "") in new stack
== Spawn extension (home, spainbound, 10) exited non-zero on 'SIP/asterisk-081df688’
Scheduling destruction of SIP dialog ‘d0ba82c5-61143be1@192.168.201.9’ in 32000 ms (Method: ACK)
set_destination: Parsing sip:xxxxxxxx@192.168.201.9:5061 for address/port to send to
set_destination: set destination to 192.168.201.9, port 5061
Reliably Transmitting (no NAT) to 192.168.201.9:5061:
BYE sip:xxxxxxxx@192.168.201.9:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.201.1:5060;branch=z9hG4bK1076eb17;rport
From: sip:spainbound@192.168.201.1;tag=as67309ecc
To: unknown sip:xxxxxxxx@192.168.201.1;tag=4c2275757568f1o1
Call-ID: d0ba82c5-61143be1@192.168.201.9
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
CLI>
<— SIP read from 192.168.201.9:5061 —>
SIP/2.0 200 OK
To: unknown sip:xxxxxxxx@192.168.201.1;tag=4c2275757568f1o1
From: sip:spainbound@192.168.201.1;tag=as67309ecc
Call-ID: d0ba82c5-61143be1@192.168.201.9
CSeq: 102 BYE
Via: SIP/2.0/UDP 192.168.201.1:5060;branch=z9hG4bK1076eb17
Server: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0
<------------->
— (8 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘d0ba82c5-61143be1@192.168.201.9’ Method: ACK
*CLI>
If I dial in a second time (10-20 seconds) after the hangup has failed , I am greeted with a modem/fax tone from an unknown source.
Waiting for a little while “resets” asteriks and I am able to get to my small IVR again.
I am using Asterisk 1.4.19.1 and a linksys spa3102 phone adapter
Please advice