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inboud call problem


Nathan
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I set up a audiocodes MP118 fxo gateway for inbound and outbound calling the outbound works just fine but the inbound has issues.

I know it has got to be an easy fix but the lack of an example really has me going in circles.

the incoming calls are to ring ext.72 which is a group but the send call to extension box in the trunk configuration screen seems to do nothing.

this is what a inbound call gives me

 

[5] 2011/03/16 11:50:52: SIP Rx udp:172.16.20.32:5060:

INVITE sip:172.16.20.30@172.16.20.30 SIP/2.0

Via: SIP/2.0/UDP 172.16.20.32;branch=z9hG4bKac206798558

Max-Forwards: 70

From: "201" <sip:201@172.16.20.32>;tag=1c206792810

To: <sip:172.16.20.30@172.16.20.30>

Call-ID: 206792263163201111237@172.16.20.32

CSeq: 1 INVITE

Contact: <sip:201@172.16.20.32:5060>

Supported: em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat

Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE

User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.6.20A.012.005

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 255

 

v=0

o=AudiocodesGW 206787991 206787869 IN IP4 172.16.20.32

s=Phone-Call

c=IN IP4 172.16.20.32

t=0 0

m=audio 6050 RTP/AVP 8 0 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=sendrecv

 

[5] 2011/03/16 11:50:52: Identify trunk (IP address/port and domain match) 2

[5] 2011/03/16 11:50:52: SIP Rx udp:172.16.20.32:5060:

INVITE sip:172.16.20.30@172.16.20.30 SIP/2.0

Via: SIP/2.0/UDP 172.16.20.32;branch=z9hG4bKac206798558

Max-Forwards: 70

From: "201" <sip:201@172.16.20.32>;tag=1c206792810

To: <sip:172.16.20.30@172.16.20.30>

Call-ID: 206792263163201111237@172.16.20.32

CSeq: 1 INVITE

Contact: <sip:201@172.16.20.32:5060>

Supported: em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat

Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE

User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.6.20A.012.005

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 255

 

v=0

o=AudiocodesGW 206787991 206787869 IN IP4 172.16.20.32

s=Phone-Call

c=IN IP4 172.16.20.32

t=0 0

m=audio 6050 RTP/AVP 8 0 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=sendrecv

 

[5] 2011/03/16 11:50:52: SIP Tx udp:172.16.20.32:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.16.20.32;branch=z9hG4bKac206798558

From: "201" <sip:201@172.16.20.32>;tag=1c206792810

To: <sip:172.16.20.30@172.16.20.30>;tag=e33de9132b

Call-ID: 206792263163201111237@172.16.20.32

CSeq: 1 INVITE

Content-Length: 0

 

 

[4] 2011/03/16 11:50:52: Call from account 201: Not an extension

[5] 2011/03/16 11:50:52: Received incoming call without trunk information and user has not been found

[5] 2011/03/16 11:50:52: SIP Tx udp:172.16.20.32:5060:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 172.16.20.32;branch=z9hG4bKac206798558

From: "201" <sip:201@172.16.20.32>;tag=1c206792810

To: <sip:172.16.20.30@172.16.20.30>;tag=e33de9132b

Call-ID: 206792263163201111237@172.16.20.32

CSeq: 1 INVITE

Contact: <sip:172.16.20.30@172.16.20.30:5060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.0.3981

Content-Length: 0

 

 

[5] 2011/03/16 11:50:52: SIP Rx udp:172.16.20.32:5060:

ACK sip:172.16.20.30@172.16.20.30 SIP/2.0

Via: SIP/2.0/UDP 172.16.20.32;branch=z9hG4bKac206798558

Max-Forwards: 70

From: "201" <sip:201@172.16.20.32>;tag=1c206792810

To: <sip:172.16.20.30@172.16.20.30>;tag=e33de9132b

Call-ID: 206792263163201111237@172.16.20.32

CSeq: 1 ACK

Contact: <sip:201@172.16.20.32:5060>

Supported: em,timer,replaces,path,early-session,resource-priority

Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE

User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.6.20A.012.005

Content-Length: 0

 

I am not seeing how the AC gateway is sending the to information.

does it just need to send to <sip:72@172.16.20.30>?

 

thanks,

Nathan

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You seems to have an account 201 in the PBX and the gateway sending 201 as the "From". You either delete the account on the PBX so that PBX does not think it is coming from one of its accounts or you can make the gateway to send something else in "From" field.

 

Once the previous hurdle is cleared, the "Send call to extension" should work.

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