Incoming Problem

General Help
1

Dear Sir

I am running with FreePbx 2.8 and Asterisk 1.6.

I am OK for Extens and PSTN Outgoing but not OK for Incoming from PSTN.

I got following error

========

Verbosity is at least 3
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
– Executing [14415004@from-trunk-sip-OpenTel:1] Set(“SIP/OpenTel-00000061”, “GROUP()=OUT_2”) in new stack
– Executing [14415004@from-trunk-sip-OpenTel:2] Goto(“SIP/OpenTel-00000061”, “from-trunk,14415004,1”) in new stack
– Goto (from-trunk,14415004,1)
– Executing [14415004@from-trunk:1] Set(“SIP/OpenTel-00000061”, “__FROM_DID=14415004”) in new stack
– Executing [14415004@from-trunk:2] NoOp(“SIP/OpenTel-00000061”, “Received an unknown call with DID set to 14415004”) in new stack
– Executing [14415004@from-trunk:3] Goto(“SIP/OpenTel-00000061”, “s,a2”) in new stack
– Goto (from-trunk,s,2)
– Executing [s@from-trunk:2] Answer(“SIP/OpenTel-00000061”, “”) in new stack
– Executing [s@from-trunk:3] Wait(“SIP/OpenTel-00000061”, “2”) in new stack
– Executing [s@from-trunk:4] Playback(“SIP/OpenTel-00000061”, “ss-noservice”) in new stack
– <SIP/OpenTel-00000061> Playing ‘ss-noservice.gsm’ (language ‘en’)
– Executing [s@from-trunk:5] SayAlpha(“SIP/OpenTel-00000061”, “14415004”) in new stack
– <SIP/OpenTel-00000061> Playing ‘digits/1.gsm’ (language ‘en’)
– <SIP/OpenTel-00000061> Playing ‘digits/4.gsm’ (language ‘en’)
– <SIP/OpenTel-00000061> Playing ‘digits/4.gsm’ (language ‘en’)
– <SIP/OpenTel-00000061> Playing ‘digits/1.gsm’ (language ‘en’)
– <SIP/OpenTel-00000061> Playing ‘digits/5.gsm’ (language ‘en’)
– <SIP/OpenTel-00000061> Playing ‘digits/0.gsm’ (language ‘en’)
– <SIP/OpenTel-00000061> Playing ‘digits/0.gsm’ (language ‘en’)
– <SIP/OpenTel-00000061> Playing ‘digits/4.gsm’ (language ‘en’)
– Executing [s@from-trunk:6] Hangup(“SIP/OpenTel-00000061”, “”) in new stack
== Spawn extension (from-trunk, s, 6) exited non-zero on ‘SIP/OpenTel-00000061’
– Executing [h@from-trunk:1] Macro(“SIP/OpenTel-00000061”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/OpenTel-00000061”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [s@macro-hangupcall:4] GotoIf(“SIP/OpenTel-00000061”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [s@macro-hangupcall:7] GotoIf(“SIP/OpenTel-00000061”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] Hangup(“SIP/OpenTel-00000061”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/OpenTel-00000061’ in macro ‘hangupcall’
== Spawn extension (from-trunk, h, 1) exited non-zero on ‘SIP/OpenTel-00000061’

My Trunk Configuration is following

===
Outgoing

host=192.168.40.11
insecure=very ; check with provider
dtmfmode=rfc2833
qualify=yes
disallow=all
allow=alaw&ulaw&gsm
nat=yes
type=peer

Incoming

host=192.168.40.11
type=peer
context=from-pstn

Can You advice me what should I do?

Thanks
TRY

2

Hi,

Do you have an incoming route to 14415004? If not, set one up in Inbound routes and shoot it somewhere, e.g. IVR, your extension, what not.

Here’s the problem…“Received an unknown call with DID set to 14415004”

3

Yes

I have an incoming route 14415004 / 14415004 DID/CID
Already set Destination to My extension.

Pls advice me…

Thanks

4

You want the CID portion to be ANY. Otherwise you’ll only match an incoming call both the DID and the CID being 14415004.

BF

5

yea, remove the CID portion, thats for routing the number that’s calling-CID (not called-DID)

6

Hi sir

I got with your help :slight_smile:
Really thanks
By the way , I can not delete voice mail files from web, How can I try or I need to New Post in Forum?

Thanks
TRY