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SPA232D No Caller-ID on FXO -> SIP -> FXS

WhiteNewPhone
Level 1
Level 1

Hi

 

I am using the ATA SPA232D to accept incoming PSTN calls, the call asterisk, which then forwards the call to FXS (Line 1). Incoming calls are connected and voice works in both directions. But the Phone on Line 1 does not show the Caller Id.

Some things i tried:

  • When I increase the "PSTN Answer Delay" (5) so that "PSTN Ring Thru Delay" (1) is smaller: on an incoming call the Phone rings and shows Caller ID
  • When I forward the incoming call from asterisk to a softphone, the Caller-ID is visible.
  • When I forward the incoming call from asterisk to Line 1, the Caller-ID is NOT visible. However, i can see the Caller-ID in the Asterisk Debug logs.
  • When I call from a softphone directly via asterisk to the Phone on Line 1, the caller-id (name and number) are visible.
  • I enabled Syslog/Debuglog  (Debug Level 3+Router) and i can see that the Caller-ID is recognized correctly.
  • I upgrade to the latest firmware 1.4.0 (001_281), no difference.
  • I have tried resetting the settings and reconfigured everything.

I am out of  ideas, please help.

 

Attached is the debug log of an incoming call to FXO -> asterisk -> FXS (without off hook): I replaced the correct caller id with xxxx. 

 

IP-Adreses in Log:

192.168.0.10 = ATA
192.168.0.14 = Asterisk

 

Thanks

1 Accepted Solution

Accepted Solutions

Dan Lukes
VIP Alumni
VIP Alumni

We can divide issue to three parts.

  1. CLID needs to be properly received from PSTN.
  2. It needs to correctly pass thru Asterisk then.
  3. Finally, it needs to be delivered to phone via FXS.

According log you provided, CLID is received from PSTN with no problem:

Caller ID:
--     Remote Number    = 079xxxxxxx

Although you provided no SIP packets, I assume it passed thru Asterisk as well. And log claim it is delivered to FXS as well:

uchDisplayCIDFSK(), EP 1 lid 0 buflen 99 overhead 60 SZ_MAX_USERDATA 200 offhook 0
uchDisplayCIDFSK(), FSK Caller ID standard is 0(bell 202)
uchDisplayCIDFSK(), SeizeFreq 0x16 MarkFreq 0xc
[0]CID Start DTMF/FSK, CID_ST_ACTIVE
uchAppCb(), Event 65 received EP 1 lid 0
receive CH_ASYNC_CIT_TRANSMITTED
[0]CID CID:DONE

Unfortunately, there are so many protocol used to delived CID over FXS. It seems you selected Bell 202/FSK.

 

It may or may not be protocol recognized by your particular phone. You need to configure CID transmission method to be compatible with the phone. See phone documentation for list of supported CID protocols or call vendor support.

 

 

View solution in original post

3 Replies 3

Dan Lukes
VIP Alumni
VIP Alumni

We can divide issue to three parts.

  1. CLID needs to be properly received from PSTN.
  2. It needs to correctly pass thru Asterisk then.
  3. Finally, it needs to be delivered to phone via FXS.

According log you provided, CLID is received from PSTN with no problem:

Caller ID:
--     Remote Number    = 079xxxxxxx

Although you provided no SIP packets, I assume it passed thru Asterisk as well. And log claim it is delivered to FXS as well:

uchDisplayCIDFSK(), EP 1 lid 0 buflen 99 overhead 60 SZ_MAX_USERDATA 200 offhook 0
uchDisplayCIDFSK(), FSK Caller ID standard is 0(bell 202)
uchDisplayCIDFSK(), SeizeFreq 0x16 MarkFreq 0xc
[0]CID Start DTMF/FSK, CID_ST_ACTIVE
uchAppCb(), Event 65 received EP 1 lid 0
receive CH_ASYNC_CIT_TRANSMITTED
[0]CID CID:DONE

Unfortunately, there are so many protocol used to delived CID over FXS. It seems you selected Bell 202/FSK.

 

It may or may not be protocol recognized by your particular phone. You need to configure CID transmission method to be compatible with the phone. See phone documentation for list of supported CID protocols or call vendor support.

 

 

I was using with Caller ID Method: "ETSI FSK", as this is recommended by the pstn operator. I was not aware that the setting "Caller ID Method" is about the FXS, i thought its about the FXO Port. It's not clear in the manual either.

Using the setting "ETSI DTMF" solves my issue. I still don't understand why "ETSI FSK" works for internal calls from the softphone.

Thanks for your advise!

You still doesn't understood we are speaking about the protocol between ATA and internal analog phone. Softphone has NOT been connected using analog line, thus their settings doesn't apply.

Thanks for your advise!

Glad to hear you solved the issue. You may consider to rate valuable advice or mark thread as answered. It will help others to found solutions.

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