@Smithjoe1 wrote:
I had an inbound line on the VOIP system set up as an IVR recorder, start a recording, playing a voicemail message and waiting for any keypress as a hangup and was working fine.
I had recently finished with this setup and wanted it to go back to regular ring groups, so I removed the inbound trunk for it, assuming that it would be caught by the Any DID / any CID incoming group. Instead, when I removed this it stopped working. I've since tried creating an inbound route for the DID and using the any DID, nothing I do gets inbound calls working.
Now when I call the line I get the following "the number that you dialed is invalid or incomplete, please check the number before dialing again" on loop. I can call out from the number, but returning the call results in this error.
SIP Registration shows as active
APSPBX*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
sip.provider.com.au:5060 N 0355555555 3585 Registered Tue, 09 Aug 2016 14:22:13
1 SIP registrations.APSPBX*CLI> sip show users
Username Secret Accountcode Def.Context ACL Forcerport
104 XXXXXXXX7 from-internal Yes No
105 XXXXXXXX7 from-internal Yes No
106 XXXXXXXX7 from-internal Yes No
107 XXXXXXXX7 from-internal Yes No
101 XXXXXXXX7 from-internal Yes No
102 XXXXXXXX7 from-internal Yes No
103 XXXXXXXX7 from-internal Yes No
108 XXXXXXXX7 from-internal Yes No
109 XXXXXXXX7 from-internal Yes No
60 XXXXXXXX7 from-internal Yes No
0355555555 XXXXXXXX7 from-pstn No YesRegistry shows in sip show objects
-= Registry objects: 1 =-
name: 0355555555:XXXXXXXX7@sip.provider.com.au/0355555555
If I set the core verbosity level to 9, I get nothing when I dial in, I get a full stack of info when I dial out but not a single line incoming.
I've run a packet capture on the firewall to see if theres any traffic moving when I try to dial in which might be getting blocked somewhere. Outgoing calls will push 100s of packets full of SIP SIP/SDF, RTP and RTCP packets on an outgoing call. Incoming are only a handful of SIP packets. This repeats every 10 seconds with nothing when I make an inbound call.
No. Time Source Destination Protcol Length Info 1 0.000000 192.168.1.5 Remote IP SIP 609 Request: OPTIONS sip:sip.provider.com.au 2 0.068811 Remote IP 192.168.1.5 SIP 536 Status: 200 OK |
The trunk settings:
Trunk name : Trunk
Peer Details
username=0355555555
port=5060
type=peer
secret=XXXXXXXX7
insecure=invite,port
host=sip.provider.com.au
fromuser=0355555555
fromdomain=sip.provider.com.au
dtmfmode=rfc2833
canreinvite=yes
allow=ulaw&alaw&g729
qualify=yes
qualifyfreq=10USER Contect : 0355555555
User Details
secret=XXXXXXXX7
context=from-pstn
type=userRegister String
name: 0355555555:XXXXXXXX7@sip.provider.com.au/0355555555Asterisk SIP settings is set to allow anonymous inbound SIP calls, this is restricted to being calls only from the SIP provider as the firewall limits the communication on ports 5060-5062 to only be allowed from sip.provider.com.au
So I've hit a dead end and can't figure out what troubleshooting steps I should perform next, weather it should go to the VOIP provider or if I had done something to shoot myself in the foot when I removed the call recording/IVR setup. Any help would be appreciated.
Posts: 1
Participants: 1