Caller ID shows name but # unknown

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mikeradio

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Hello

I am using Cisco SPA525G phones and Grandstream GXW4104 gateways. I am getting name from caller id but the phone number shows unknown. Do I need to setup caller id rules or something.

Thank you

Mike
 
Is the number showing up in the 3CX log during an incoming call? In other words, is the gateway sending it?
 
Thank you for your reply leejor

I called the number and here is what the log should for the call


08:51:42.956 [CM503025]: Call(19): Calling RingAll802[Ext.105,Ext.113,Ext.100]@[Dev:sip:[email protected]:5060]
08:51:42.916 [CM503004]: Call(19): Route 1: RingAll802[Ext.105,Ext.113,Ext.100]@[Dev:sip:[email protected]:5060,Dev:sip:[email protected]:5060]
08:51:42.914 [CM503010]: Making route(s) to <sip:[email protected]:5060>
08:51:42.912 [CM505002]: Gateway:[Gateway A] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXW4104 (HW 2.0, Ch:7) 1.3.4.10] PBX contact: [sip:[email protected]:5060]
08:51:42.911 [CM503001]: Call(19): Incoming call from unknown@(Ln.10001@Gateway A) to <sip:[email protected]:5060>
08:51:42.902 [CM503012]: Inbound office hours rule (unnamed) for 10001 forwards to DN:802

I see it says incoming call from unknown, so the problem is in the gateway?

Thank you for the help

MIke
 
If you have confirmed that you do indeed have caller ID service on the PSTN line, then (and I'm going to assume you are in North America), your gateway may be set to send the call to 3CX before the caller ID info has been received by the gateway.

In North America the data burst is sent between the first and second ring, so the gateway has to wait and collect the information before alerting 3CX of the new call. Check the gateway answer delay settings, increasing it, as necessary.
 
Thank you Leejor

After more research it looks like a number of people are have troubles with Caller Id and the Grandstream GXW4104 with the lastest firmware. So it looks like my problem is the gateway, I will have to wait until Grandstream releases new firmware or purchase another gateway that works.

Mike
 
You don't say where you are located. I could perhaps understand a firmware release coming out with caller ID issues for some smaller markets, only to be discovered later, but, if caller ID were affected in major markets, then I'd be surprised that they would release it at all, given the backlash they could expect.

If you are absolutely, positively, convinced that it is the firmware, you could always downgrade. Older releases can be found with some searching. Before doing that however confirm that you are receiving CID from your provider, and double check the settings, you did not say if you did either.
 
Hello Leejor

I am located in Canada, BC. I turn on the debug and send the information to a syslog. Looking at an incomming call the log shows the name and the number correctly, my phones show the name but says unknown for all the numbers. It looks like the gateway is decoding the information proplerly but when it , turns it into SIP it says unknown.

From the Grandstream fourm it looks like people started having problems after firmware went from 1.3.1.6 to 1.3.4.9 and the current firmware is 1.3.4.10. I did find copies of all the firmware and downgraded my gateway to 1.3.4.9 and the caller id did not work, I tried to downgrade to 1.3.1.6 and I get an error, it looks like they may have a block so you can not go back to 1.3.1.6 because a number of people on the forum get the same error.

Here is a copy of the debug from the syslog :

FXO: Port: 1 Ring count: 1 Ring pulses: 0
FXO: Port: 1 *Call Status* Processing (Incoming call), From:1745
cm_task.c 1773 Port: 1 GWSt: 1 Sess: un-allocated. 1 fxoCallStatus last/now=3/4 FAS:0
cm_task.c 1896 CM_Ringing Port: 1 2 1 0 1 2 6 192.168.1.60:5060 0
cm_task.c 1896 CM_Ringing Port: 1 2 1 0 1 2 6 192.168.1.60:5060 0
FXO_CID: Port: 1 Finish data capture.
FXO_CID: Port: 1 Start cid analysis.
cm_task.c 1896 CM_Ringing Port: 1 2 1 0 1 2 6 192.168.1.60:5060 0
FXO_CID: Port: 1 ---------- CID Reporting (FSK)-----------
FXO_CID: Port: 1 Date and time: 11151657 (MMDDHHMM)
FXO_CID: Port: 1 Name: PRIVATE
FXO_CID: Port: 1 DID Number: 2505645517
FXO_CID: Port: 1 ---------- CID Reporting (end) ----------
FXO_CID: Port: 1 Capture completed. Starts @ 0x80E69780 len=0000BB80
FXO_CID: Port: 1 Finish cid analysis.
402 sip.c Sess: 4 allocated for Port(acct): 1 ln#: 1744 Grandstream GXW4104 1.3.4.10 1.1.3.2
------------------------- GXW setup ------------------------------
1749 cm_utility.c Port: 1 Sess: 4 Acc:0; FullRingCounts:1 RingPulseCounts:0 FXOState:2; Log user data: 1 StageDialing, PR Disabled; Curr.Disc. Enabled, Curr.Disc. Thres: 100; Cont'd..
BZ TD Disabled,Tone Cfg: Dial: F: 350 440,V: 11 11,Cad: 0 0 0,0 0 0; Ringback: F: 440 480,V: 11 11,Cad: 200 0 0,400 0 0; BZ: F: 480 620,V: 11 11,Cad: 50 0 0,50 0 0; Reorder: F: 480 620,V: 11 11,Cad: 25 0 0,25 0 0; Cont'd..
Caller ID Type: 1, Relay Type: 1, Wait for Dial tone: 0, Dialing Cfg: ch1-4:N;, Delay2Dial: 500; FAX TD Disable, T38: Enabled, T38 Bitrate:9600, T38 Mode:1; T38 ECM:1; Echo Can.: Y, Sil. Supp.: Y; Cont'd..
Audio Gains: RX from PSTN: 0, TX to PSTN: 1, DTMF Method: 1, SIP port: 5062, SRTP: 0,; DTMF Interkey T.O: 4, Round-robin: 1, Cont'd..
Prefix Code: Len:2, Scheduling:99 Ring Count:2 RingTimeout:6 FAS:0 HW DS:0 Grandstream GXW4104 1.3.4.10 1.1.3.2
------------------------- GXW setup (end)-------------------------
1269 sip_transport.c Sess: 4 Sending 2 To 192.168.1.60:5060 1 1 1 1 sip.Call-ID:[email protected]
692 sip_message.c Sess: 4 oport: 1 build SIP req 2 transport=1 181 66 1
1602 sip_transport.c Sess: 4 sent: INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK3252dd6044c27e25 From: "PRIVATE "<sip:[email protected]>;tag=5a35f27162041d73 To: <sip:[email protected]> Contact: <sip:[email protected]:5062;transport=udp> Supported: replaces, timer, path Call-ID: [email protected] CSeq: 1288 INVITE User-Agent: Grandstream GXW4104 (HW 2.0, Ch:4) 1.3.4.10 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 253 v=0 o=10001 8004 8000 IN IP4 192.168.1.63 s=SIP Call c=IN IP4 192.168.1.63 t=0 0 m=audio 5020 RTP/AVP 0 8 4 18 3 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=ptime:20
1618 sip_transport.c INVITE From="PRIVATE "<sip:[email protected]>;tag=5a35f27162041d73 To=<sip:[email protected]>
 
Did you see this?

http://forums.grandstream.com/forums/index.php?topic=12156.0

Don't know how that set-up differs from yours.
 
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