No Incoming Number Shown-Caller ID Subcirbed!

Asterisk Asterisk Support
1

Hi,

We have recently purchased the 8 port modular analog PCI 3.3/5.0V card with 8 Trunk interfaces
My Incoming Numbers not showing for incoming calls, I have setup ZAP CHANNEL DIDs Port Numbers.

When I place inbound call route to IVR, i get message “the number you are calling is not in service”, but when i add ANY DID /ANY CLID, I get the call successfully. I checked the CALLER ID and its works fine from ISP, but sometime it shows the number and sometime it doesn’t show UNKNOWN NUMBER. When I connect an analog phone direct to my landline line, i can able to see the numbers. But when i connect the same landline line to my asterisk, then i cannot able to see the incoming numbers sometime and sometime UNKNOWN NUMBER.

Your respsonse will be highly appreciated

Thanks,
Sam.

2

Which brand and model of card? Generally, for analogue cards, the best source of support is the supplier of the cards. Digium supports their cards but through their commercial support organisation, not through this, peer support, forum.

Which version of Asterisk? All versions of Asterisk that use Zaptel, rather than Dadhi, are past end of life.

Which PSTN provider, which of their products and which type of central office switch?

What are your configuration options for detecting caller ID?

Please explain your reference to DID, as calleD number indications aren’t supported on analogue lines, so Direct In Dialing is not possible.

3

Thanks for your reply.

I also spoke to Commercial organization and seek for help, still waiting for there response.

My Asterisk system has
Name Package Name Version Release
FreePBX FreePBX 2.8.1 16

Name Package Name Version Release
Asterisk Asterisk 1.8.20.0 0

Name Package Name Version Release
DRIVERS dahdi 2.6.1 4

Let says I have 3-Numbers, For Outgoing I have configure to dial out by each prefix.9 - 8 - 6 from each different number. Now for 3 landline numbers i needed to recieve calls on seperate IVR - Route to Recording Messages.

->What are your configuration options for detecting caller ID?
I made the configure as follows:- Placed the Landlines Numbers into my ZAP Channel DID and Creating Inbound Routes and then placed DID Numbers and given destination to IVR. Thats all.
Is there any i could add into the configuration or im missing something.

Hope you understand my scenario, I just wanted configure Inbound Routes to route to following
landline#1 Should be Route to —> IVR#1
landline#2 Should be Route to—> IVR#2
landline#3 Should be Route to—> IVR#3
Now, I have added ZAP Channel DIDs and made a Inbound Routes, but still its getting reached to respective IVR.

Can it be possible to assign the Port Wise Routing (Instead of DID Numbers) to IVR and then destination extension. May I know is that possible to Route Incoming Calls by ZAP Channel DID or My Card Port Wise.

Your response will be highly appreciated.

-Thanks,
Sam.

4

ZAP channel DID may make sense to FreePBX, but it doesn’t make sense in Asterisk terms. Asterisk doesn’t support DID on analogue lines and neither to most or all PSTN operators.

You haven’t answered the question about caller ID signalling. I want to know what line condition signals the start of CLID and what protocol is used to send the digits. Dahdi will need to be configured with this information.

5

I have created new file zapata.conf and put content = from-zaptel. Still no effect.

Im sorry if im mistaken to understand your question, because I have some knowledge about asterisk PBX, can able to troubleshoot only basic. This time im recieving this error unfortunately.

I have got the details, this might help to understand.

Whenever I dail to my Landline

-- Starting simple switch on 'DAHDI/1-1'
-- Executing [s@from-pstn:1] NoOp("DAHDI/1-1", "No DID or CID Match") in new stack
-- Executing [s@from-pstn:2] Answer("DAHDI/1-1", "") in new stack
-- Executing [s@from-pstn:3] Wait("DAHDI/1-1", "2") in new stack
-- Executing [s@from-pstn:4] Playback("DAHDI/1-1", "ss-noservice") in new stack
-- <DAHDI/1-1> Playing 'ss-noservice.gsm' (language 'en')
-- Executing [s@from-pstn:5] SayAlpha("DAHDI/1-1", "") in new stack
-- Executing [s@from-pstn:6] Hangup("DAHDI/1-1", "") in new stack

== Spawn extension (from-pstn, s, 6) exited non-zero on ‘DAHDI/1-1’
– Executing [h@from-pstn:1] Macro(“DAHDI/1-1”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“DAHDI/1-1”, “1?endmixmoncheck”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] NoOp(“DAHDI/1-1”, “End of MIXMON check”) in new stack
– Executing [s@macro-hangupcall:10] GotoIf(“DAHDI/1-1”, “1?nomeetmemon”) in new stack
– Goto (macro-hangupcall,s,28)
– Executing [s@macro-hangupcall:28] NoOp(“DAHDI/1-1”, “End of MEETME check”) in new stack
– Executing [s@macro-hangupcall:29] GotoIf(“DAHDI/1-1”, “1?noautomon”) in new stack
– Goto (macro-hangupcall,s,34)
– Executing [s@macro-hangupcall:34] NoOp(“DAHDI/1-1”, “TOUCH_MONITOR_OUTPUT=”) in new stack
– Executing [s@macro-hangupcall:35] GotoIf(“DAHDI/1-1”, “1?noautomon2”) in new stack
– Goto (macro-hangupcall,s,41)
– Executing [s@macro-hangupcall:41] NoOp(“DAHDI/1-1”, “MONITOR_FILENAME=”) in new stack
– Executing [s@macro-hangupcall:42] GotoIf(“DAHDI/1-1”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,45)
– Executing [s@macro-hangupcall:45] GotoIf(“DAHDI/1-1”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,48)
– Executing [s@macro-hangupcall:48] GotoIf(“DAHDI/1-1”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,50)
– Executing [s@macro-hangupcall:50] AGI(“DAHDI/1-1”, “hangup.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
– <DAHDI/1-1>AGI Script hangup.agi completed, returning 0
– Executing [s@macro-hangupcall:51] Hangup(“DAHDI/1-1”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 51) exited non-zero on ‘DAHDI/1-1’ in macro ‘hangupcall’
== Spawn extension (from-pstn, h, 1) exited non-zero on ‘DAHDI/1-1’
– Hanging up on ‘DAHDI/1-1’
– Hungup ‘DAHDI/1-1’

6

What error? You are using FreePBX and that is a valid call without errors, but your system is clearly misconfigured since you don’t have an “Inbound Route”.

Please refer to the FreePBX wiki to learn how to configure your system or try to buy support from the SCHMOOZE(oficial support of FreePBX) guys.

7

it may misconfigured, im only stuck inbound call routes.
As we do normal configure for inbound call route done other Asterisk PBX, we dont have why its not working for this asterisk system only.

Yes, we have added the Inbound Routes for all our ZAP Channel DID Numbers.

Please let me know, any possible way to configure, will be highly appreciated.

8

Please use freepbx.org/forums/ for FreePBX peer support.