[Solved] Ringroup caller ID issue

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[Solved] Ringroup caller ID issue

Postby frankg » Tue Jan 24, 2012 10:04 pm

My setup:
PSTN -> Grandstream - GXW4108 -> 3CX -> Cisco509G

If a call comes in and someone hits an extension, the caller ID is working fine, but when a call goes to a ring group the caller IDs on the phones just say: Unknown

Any ideas?
Last edited by frankg on Thu Jan 26, 2012 1:20 pm, edited 1 time in total.
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Re: Ringroup caller ID issue

Postby leejor » Tue Jan 24, 2012 10:17 pm

frankg wrote:If a call comes in and someone hits an extension


What do you mean, a transferred call, one that came from a digital receptionist? A direct trunk to ring group call?
What does the 3CX log show? Any difference (caller ID showing on incoming trunk) for a working and non-working call?
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Re: Ringroup caller ID issue

Postby frankg » Tue Jan 24, 2012 10:59 pm

What i meant was our digital receptionist says thank you for calling blah blah blah For frank press 38 for Bob press 11 etc...

If someone just presses 38 from there, and it goes to my extension the caller id goes through, i'll pull the logs tomorrow and post them.
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Re: Ringroup caller ID issue

Postby frankg » Wed Jan 25, 2012 6:00 pm

I was wrong, its actually not working at all regardless of ring group or if someone dials a specific extension:

Code: Select all
11:57:00.356  [CM503003]: Call(84): Call to sip:11@192.168.1.115 has failed; Cause: 487 Request Terminated; from IP:192.168.1.160:5060
11:57:00.246  [CM503008]: Call(84): Call is terminated
11:56:51.215  [CM505001]: Ext.11: Device info: Device Identified: [Man: Cisco;Mod: SPA Series;Rev: General] Capabilities:[reinvite, replaces, unable-no-sdp, no-recvonly] UserAgent: [Cisco/SPA509G-7.4.8a] PBX contact: [sip:11@192.168.1.115:5060]
11:56:51.215  [CM503002]: Call(84): Alerting sip:11@192.168.1.160:5060
11:56:51.137  [CM503025]: Call(84): Calling Ext:Ext.11@[Dev:sip:11@192.168.1.160:5060]
11:56:51.074  [CM503004]: Call(84): Route 1: Ext:Ext.11@[Dev:sip:11@192.168.1.160:5060]
11:56:51.074  [CM503010]: Making route(s) to <sip:11@127.0.0.1:5060>
11:56:45.824  Currently active calls - 1: [84]
11:56:39.012  [CM503007]: Call(84): Device joined: sip:81@127.0.0.1:40600;rinstance=bd3468264a444801
11:56:39.012  [CM503007]: Call(84): Device joined: sip:10000@192.168.1.113:5060;transport=udp
11:56:38.996  [CM505001]: Ext.81: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX IVR;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX IVR] PBX contact: [sip:81@127.0.0.1:5060]
11:56:38.996  [CM503002]: Call(84): Alerting sip:81@127.0.0.1:40600;rinstance=bd3468264a444801
11:56:38.856  [CM503025]: Call(84): Calling Ext:Ext.81@[Dev:sip:81@127.0.0.1:40600;rinstance=bd3468264a444801]
11:56:38.809  [CM503004]: Call(84): Route 1: Ext:Ext.81@[Dev:sip:81@127.0.0.1:40600;rinstance=bd3468264a444801]
11:56:38.809  [CM503010]: Making route(s) to <sip:81@192.168.1.115:5060>
11:56:38.809  [CM505002]: Gateway:[gs] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXW4108 (HW 1.0, Ch:0) 1.3.4.10] PBX contact: [sip:10000@192.168.1.115:5060]
11:56:38.793  [CM503001]: Call(84): Incoming call from unknown@(Ln.10000@gs) to <sip:81@192.168.1.115:5060>
11:56:38.793  [CM503012]: Inbound out-of-office hours rule (unnamed) for 10000 forwards to DN:81
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Re: Ringroup caller ID issue

Postby leejor » Wed Jan 25, 2012 11:00 pm

Usually one of two things...

Caller ID option is not on the line, or...

In North America (and other areas using Bell type caller ID), there needs to be a delay, of about 3 seconds (or longer) before an incoming call is presented to 3CX from the Gateway. The caller Id information is sent between the first and second ring, if the call is sent too early it will miss the info. The option is usually called something like Answer Delay.
Last edited by leejor on Wed Jan 25, 2012 11:16 pm, edited 1 time in total.
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Re: Ringroup caller ID issue

Postby frankg » Wed Jan 25, 2012 11:07 pm

leejor wrote:... The caller Id information is sent between the first and second ring, if the call is sent too early it will miss the info. The option is usually called something like Answer Delay.


You were absolutely correct!! I had the gateway set to 1 ring before answer, I changed it to 2 and it worked!!!


Thanks so much
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