Hi there,
Here in our office we had been using AMP flawlessly for months. Just the
other day I decided to go ahead and update to the latest FreePBX.
Everything continues to work perfectly, except that we no longer see the
proper CallerID on incoming calls.
For our phone lines we use a MaxTNT which has a single PRI connected.
Our FPBX server accesses the TNT via LAN. The TNT is setup as extension 15.
Here’s a part of the log during an incoming call:
– Executing Set(“SIP/15-b7d030a0”, “REALCALLERIDNUM=15”) in new stack
– Executing NoOp(“SIP/15-b7d030a0”, “REALCALLERIDNUM is 15”) in new stack
– Executing Set(“SIP/15-b7d030a0”, “AMPUSER=15”) in new stack
– Executing Set(“SIP/15-b7d030a0”, “AMPUSERCIDNAME=MaxTNT”) in new stack
– Executing GotoIf(“SIP/15-b7d030a0”, “0?report”) in new stack
– Executing Set(“SIP/15-b7d030a0”, “CALLERID(all)=MaxTNT <15>”) in new
stack
– Executing NoOp(“SIP/15-b7d030a0”, “Using CallerID “MaxTNT” <15>”) in
new stack
So on our Cisco phones all we see for CallerID is “MaxTNT” <15>.
Now, if I turn on SIP debug and place an incoming call here is what I get.
Note:
10.10.20.5 = FPBX server
10.10.20.26 = MaxTNT
10.10.20.200 = My Cisco phone (ext 20)
5557299 = Dialed DID
2505557513 = Real CallerID
INVITE sip:[email protected]:5060;user=phone SIP/2.0
t: sip:[email protected]:5060;user=phone
f: sip:[email protected]:5060;user=phone;
cut
Via: SIP/2.0/UDP 10.10.20.26:5060;received=10.10.20.26
From: sip:[email protected]:5060;user=phone;
To: sip:[email protected]:5060;user=phone
cut
Reliably Transmitting (no NAT) to 10.10.20.200:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.20.5:5060;
From: “MaxTNT” sip:[email protected];
To: sip:[email protected]:5060
cut
– Called 20
Transmitting (no NAT) to 10.10.20.26:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.20.26:5060;received=10.10.20.26
From: sip:[email protected]:5060;user=phone;
To: sip:[email protected]:5060;user=phone;
cut
<-- SIP read from 10.10.20.200:50564:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.20.5:5060;
From: “MaxTNT” sip:[email protected];
To: sip:[email protected]:5060
So it seems like it’s overwriting the real callerID with the extension
number of the MaxTNT.
Any ideas on how I can fix this? As I mentioned earlier this did not
happen using AMP.
Thanks
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