Nathan
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Posts posted by Nathan
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ok that was an easy fix thanks.
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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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I set up the MP118 and out bound calls work just fine but in bound calls fail it says no route.
I tried several different ways of setting up the incomming call routing from different toppics but none seem to be right.
is there a tutorial out there about how to set this up so that a call goes from one of the pots lines to the snomone trunk number?
Thanks
ok now i have the snomone getting the call from the fxo to ip but it says call from account 201: not an extension.
then after that Received incoming call without trunk information and user has not been found.
then SIP Tx udp:172.16.20.32:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.16.20.32;branch=z9hG4bKac1680493678
From: <sip:201@172.16.20.32>;tag=1c1680489795
To: <sip:172.16.20.30@172.16.20.30>;tag=d46a52178e
Call-ID: 16804892501532011173913@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
any help?
snomone audiocodes MP118 inbound not working
in Gateways
Posted
ok i got the thing working now I changed the hunt group phone numbers in the AC gateway and that fixed it.
Thanks