Caller ID lost when bridging PSTN with Line1 on ATA

[color=#008040]; I’m using a Cisco ATA232D.[/color]

exten => s,1,NoOP()

[color=#008040] ; Caller ID works perfectly as expected, passing from PSTN, thru ATA to Asterisk[/color] :smiley:
same => n,Dial(SIP/${gATALine1},5)

[color=#008040] ; However, if the gATALine1 above rings out, and I try to change the Caller ID and ring again - the caller ID fails and is missing… why?[/color] :cry:
same => n,Set(CALLERID(all)=“Ghostbusters” <555-2368>)
same => n,Dial(SIP/${gATALine1},5)

[color=#008040]; Voip1 --> FXS = Caller ID works perfectly.
; FXO --> Voip2 --> Voip1 --> FXS = Caller ID does not work for FXS [/color]

Please help :confused:

Are you dialing back onto an analogue PSTN line? Analogue PSTN lines don’t support outgoing caller ID.

Thanks david55, yes but perhaps disagree?

Yes, gATALine1 is an analogue line on the ATA (SPA232D), but…

Disagree as voip calls from dialplan can and do signal caller ID to gATALine1, but only when calls originates from the internal context (ie. internal voip desk phones).

Calls that originate from the PSTN line/context, and then get passed around the dialplan, don’t seem to be able to send caller id back to gATALine1.

It’s all very confusing and inconsistent :confused:

I don’t really understand your configuration.

Can you post the log from your Voip2 and Voip1 boxes for a call that doesn’t work? Are you using SIP or IAX between VoipX server? It almost sounds like it’s not in the expected part of the dialplan on Voip1.

Insert before the second dial a small delay

same => n,Wait(3)

It might be too soon for the analog phone to receive a new call and just answers the call.
Also after setting the callerid use noop to see what exactly is the callerid

same => n,Noop(Callerid name is ${CALLERID(name)})
same => n,Noop(Callerid number is ${CALLERID(number)})

He tries first to call the voip adapter and if he doesn’t answer to send him again the call with a new callerid as additional information for the user.

My question was about the hardware configuration. Having internal analogue phones is unusual in the first place, and using them with callerID possibly even more unusual. My initial answers were on the basis that he was forwarding the call back onto the PSTN, but in that case both incoming and outgoing analogue lines would use FXS signalling (FXO modules) and there is no possibility of sending callerID over FXS signalling.

He is using a Cisco FXO/FXS adapter. I am guessing he has a sip trunk for the pstn line port that he receives the call and then he sends the call back to the same device to the port that the analog phone is.

Asterisk doesn’t see any analog interfaces just sip, the final device (SPA232D) is analog.

Well let’s wait for the user to answer! :mrgreen:

In understood that Asterisk saw SIP.

I think I would want to know what the local phone was in this case.

Also what is in sip.conf for the ATA (e.g. using fromdomain would tend to break callerID).

One possible issue is that the PSTN line may be having the caller ID tagged as unscreened or presentation not allowed, so he may need to explicitly override those settings for the call.