CALLERID problem

Asterisk Asterisk Support
1

I have looked everywere to find a solution on this, but nothing worked!

The Callerid is working in to my house when Asterisk is
not in the middle between incomming from phone company and
my house. But when I connect the Asterisk, no callerid on
incomming calls shows on the handset. (normal phone with
callerid)

I can see my mobile nummber on incomming in the CLI, like this:
“CALLERID=123456789”. I can also in my SIP phone (X-Lite) and
Grandstream-GXV3000 see the incomming nummber, but nothing on
my ZAP hard POTS.

In the last test I only sent the signal to ZAP/1, but nothing!

What am I doing wrong when trying to get the incomming phone
nummber from FXO (connected to the telecompany) to FSX (POTS
in my house)?

Card: Digium TDM11B
Country = Sweden

Asterisk 1.2.24
Zaptel 1.2.20.1
Libpri 1.2.5
Addons 1.2.7
Sounds 1.2.1

OS: PoundKey release Snub
Kernel 2.6.17.11-1.1.x86.i686

FXO = ZAP/4 (connected to phone company)
FXS = ZAP/1 (Connected POTS in my house)

Connected to Asterisk 1.2.24 currently running on asterisk (pid = 1862)
]# ztcfg -vvv

Zaptel Configuration

Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)

2 channels configured.

This is me calling in to the house with my cellphone, and as you can see
I do have CALLERID in to the system: (just changed the number)

– Remote UNIX connection
Verbosity was 3 and is now 20
== Starting post polarity CID detection on channel 4
– Starting simple switch on ‘Zap/4-1’
– Executing Set(“Zap/4-1”, “CALLERID(number)=123456789”) in new stack
– Executing Verbose(“Zap/4-1”, “incomming CallerID=<123456789>”) in new stack
incomming CallerID=<123456789>
– Executing Dial(“Zap/4-1”, “ZAP/1|20|rfo”) in new stack
– Called 1
– Zap/1-1 is ringing
– Zap/1-1 is ringing
– Zap/1-1 is ringing
– Hungup ‘Zap/1-1’
== Spawn extension (zap_incoming, s, 3) exited non-zero on ‘Zap/4-1’
– Hungup 'Zap/4-1’
asterisk*CLI>

But nothing on the handset!

This is what I got:

[code]<zaptel.conf>

loadzone=se
defaultzone=se
fxsks=4
fxoks=1[/code]

[code]<zapata.conf>

[trunkgroups]

[channels]
adsi=yes
cidsignalling=dtmf
cidstart=polarity
usecallerid=yes
useincomingcalleridonzaptransfer=yes
relaxdtmf=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=0
txgain=0
callwaiting=yes
callwaitingcallerid=yes
answeronpolarityswitch=no
hanguponpolarityswitch=yes

context=call-in-house
signalling=fxo_ks
cidsignalling=dtmf
cidstart=polarity
language=se
callerid=“Micke & Helen”<6000>
mailbox=6000@intern-ank,6001@intern-ank,6002@intern-ank,6003@intern-ank
channel => 1

context=zap_incoming
signalling=fxs_ks
cidsignalling=dtmf
cidstart=polarity
language=se
usecallerid=yes
callerid=asreceived
answeronpolarityswitch=no
hanguponpolarityswitch=yes
channel => 4

musiconhold=default [/code]

[code]<extensions.conf>

[general]
writeprotect=yes

[globals]
CALLFROM_PSTN=ZAP/1&SIP/Micke_Helen&SIP/Nypon&SIP/Helah
CALLFROM_INTERNET=ZAP/1r4&SIP/Micke_Helen&SIP/Nypon&SIP/Helah
HOME_PHONE_NAME=Micke och Helen
HOME_PHONE_NO=+1111111111

[zap_incoming]
;exten => s,1,setCallerID(${CALLERID})
;exten => s,1,SetCallerID(${CALLERIDNUM})
exten => s,1,Set(CALLERID(number)=${CALLERID})
;exten => s,1,Set(CALLERID(number)=11111111)
exten => s,n,Verbose(incomming CallerID=<${CALLERID}>)
;exten => s,n,Wait(1) ; Så den hinner fånga tag på inkommande nummer.
;exten => s,1,setCIDNum(${CALLERID})
;exten => s,1,setCIDNum(1234)
;exten => s,n,Answer
;exten => s,n,Dial(${CALLFROM_PSTN},20,ro)
exten => s,n,Dial(ZAP/1,20,rfo)
exten => s,n,Voicemail(6000@intern-ank|u)
exten => s,103,Voicemail(6000@intern-ank|b)
exten => s,n,hangup()

[zap2pstn]
exten => _0ZXXXXXXXX,1,Dial(Zap/4/${EXTEN:0})
exten => _0ZXXXXXXX,1,Dial(Zap/4/${EXTEN:0})
exten => _0ZXXXXXX,1,Dial(Zap/4/${EXTEN:0})
exten => _0ZXXXXX,1,Dial(Zap/4/${EXTEN:0})
exten => _ZXXXXXXX,1,Dial(Zap/4/${EXTEN:0})
exten => _ZXXXXXX,1,Dial(Zap/4/${EXTEN:0})
exten => _ZXXXXX,1,Dial(Zap/4/${EXTEN:0})
exten => _ZXXXX,1,Dial(Zap/4/${EXTEN:0})
;exten => _X.,1,Dial(ZAP/4/${EXTEN:2})
exten => _X.,102,Playback(tt-allbusy)
exten => _X.,n,Hangup()[/code]

I hope someboddy can helpe me on this, something to try or anything!
At one point I got the feeling that the card was broken! :confused:

Thanks in advance!

// Nypon

2

I’m no expert, but the fact that you have an Asterisk server in your loop should not be blocking callerid information to your analog POTs phones.

I have an identical setup here in my house with some IP phones running on my LAN and some POTs phones plugged into the wall jacks. My Asterisk server has an X101P card and it will send/receive POTs calls with great success. When a call comes in I see callerid on the regular analog phones that are plugged into the wall jacks and the Asterisk server also shows the callerid information.

I couldn’t tell you why you’re having this problem just that it would seem to me that you shouldn’t be having it unless there is something different about your setup that you don’t mention here.

3

Hi,
I cannot see a ‘sendcalleridafter=2’ line in your config?

I believe this may help, it defines the delay before sending Caller ID data to a Zap extension.

Try adding it at the start of the [channels] section. If it does not work, see if you get any more info in the log with it present.

4

It did not work for me with the “sendcalleridafter=2”, I did not even get anything else in the logfile. (from what I could see)

Connected to Asterisk 1.2.24 currently running on asterisk (pid = 1865)
– Remote UNIX connection
Verbosity was 3 and is now 20
== Starting post polarity CID detection on channel 4
– Starting simple switch on ‘Zap/4-1’
– Executing Set(“Zap/4-1”, “CALLERID(number)=123456789”) in new stack
– Executing Verbose(“Zap/4-1”, “incomming CallerID=<123456789>”) in new stack
incomming CallerID=<123456789>
– Executing Dial(“Zap/4-1”, “ZAP/1|20|rfo”) in new stack
– Called 1
– Zap/1-1 is ringing
– Zap/1-1 is ringing
– Zap/1-1 is ringing
– Zap/1-1 is ringing
– Hungup ‘Zap/1-1’
== Spawn extension (zap_incoming, s, 3) exited non-zero on ‘Zap/4-1’
– Hungup 'Zap/4-1’
asterisk*CLI>

My conf file now looks like this:

[code]<zapata.conf>

[trunkgroups]

[channels]
sendcalleridafter=2
adsi=yes
usecallerid=yes
cidsignalling=dtmf
cidstart=polarity
useincomingcalleridonzaptransfer=yes
relaxdtmf=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=0
txgain=0
callwaiting=yes
callwaitingcallerid=yes
answeronpolarityswitch=no
hanguponpolarityswitch=yes

context=call-in-house
signalling=fxo_ks
usecallerid=yes
cidsignalling=dtmf
cidstart=polarity
;sendcalleridafter=2 ; With this here the CALLERID got blank in the CLI!
language=se
callerid=“Micke & Helen”<6000>
mailbox=6000@intern-ank,6001@intern-ank,6002@intern-ank,6003@intern-ank
channel => 1

context=zap_incoming
signalling=fxs_ks
;sendcalleridafter=2
usecallerid=yes
cidsignalling=dtmf
cidstart=polarity
language=se
answeronpolarityswitch=no
hanguponpolarityswitch=yes
callerid=asreceived
channel => 4

musiconhold=default[/code]

maxfiles is talking about a different setup, what can that be?
Should I check some other config files?

5

Hi,
the only other think I can think of is to remove the options ‘rfo’ from the dial() command - my setup only has ‘t’

If this has no effect, I can only guess there could be a bug relating to sending DTMF caller id to extensions…

6

Hi

As you are in sweden are you sure that the set plugged in supports Bellcore callerID ? without checking Im sure Sweden isnt Bell standard, you may need to check that the set suports bellcore

Ian

7

Ian is right, in Sweden it should be ETSI FSK, possibly with PR (polarity reversal).

8

try this:
exten => s,1,Set(CALLERID(number)=${CALLERIDNUM})

9

I am werry glad that you guys taking your time trying to help me! :smiley:

rjenkins
Change from the Dial() command ‘rfo’ to ‘t’ did not work!

ianplain and AndrewZ
I’m not sure if I understand what you mean.
But, from what I have read on the net, this is for sweden.

cidsignalling=dtmf
cidstart=polarity

and ETSI FSK With PR (v23 instead of dtmf) is for UK

But correct me if i’m wrong!

r1ch97rd
With ${CALLERIDNUM} nothing more than before hapen!

10

Hi,
to the others following this thread;

If you look through the log info given, Asterisk is receiving the Caller ID from the trunk - that side is working fine.

The problem is that Asterisk is not sending CID to the analog extension, or it’s not in a format the phone can handle.

11

Right, in Sweden they use DTMF. Big surprise :wink:

One more discussion found on the Net
mail-archive.com/asterisk-us … 01717.html

12

Hi

13

Ian
OK!

Sounds like that I will not get this to work! :frowning:

Is this a hard or a software problem?

What ATA do you recomend?

14

Linksys PAP2 (v1), just checked - ‘DTMF Sweden’ is present in Caller ID settings.

15

Hi every body on this thread and manny thanks for your help!

I think I have found my problem, ADSL!

For my internet I have an ADSL connection to my phone jack.

What I did was connecting the Asterisk after the ADSL connection!
(is a connection/split that you put in the wall to connect your hardware
and at the back of that you can connect another phone/ hardware.)

Some how, most of it worked with this setting, but!
Now I have tested to connect Asterisk as the only connection!
And guess what, I got the CALLERID at one try!

But at another time I did not!!! (not sure why!)
But for some reason I think that my problem is this, that the ADSL want
to be the first connection at the same time as Asterisk!

I think that the ADSL socket brakes the connection between the phone
company and the Asterisk in one way or the other.

Because of this, now I’m not sure how to split it in one phone socket!!!

What I have now is a split to seperate the incomming from the phone
company and outgoing to my house.

As it is in sweden, the two bottom pins in the wall is “in comming” and
the two top one is the one connecting to the next phone in the house.

What I have to figure out now is how to make a split that makes my ADSL
connection and Asterisk feel that they are the first connection in my
house, at the same time!!!

Not sure, but it seams to that the ADSL needs to be the first incomming
and the outgoing at the same time!

They both want to be first connection!

Anybody that have any idea on this?

16

It sounds rather like a faulty ADSL filter.

What that filter should do is isolate the ADSL signal (which basically a low frequency radio signal) from the normal house phones. It should not affect audio or caller ID data.

Part of the effect could be that your ADSL modem is no longer connected and the signal from that was leaking through the filter and affecting the digium card.

What may possibly help is to go back to the original setup and add another ‘inline’ ADSL filter to the line into the digium card so that is double filtered. Also try with & without your own ADSL modem powered to see if that does affect the situation.

17

I have now try to have the * as the only thing plugged in to the wall jack, but nothing!

I don’t know what I did when I got it to work for a second!
It looks like I need to get me an ATA for the FXS.

But I don’t think that i’m the only one with this problem!
Is there any swedish people out there that got the CALLERID to work on the normal telephone with a Digium TDM11B?

18

I could not get the callerID to work here in sweden with the TDM11B card!

But when I got the PAP2-NA from eBay and some tweaking the callerID start to work on my POTS. :smiley:

I will try to get it to work with the TDM11B card, and some day it might start to work!