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SPA8800 & CID

stevemag76
Level 1
Level 1

Hi, Sorry to re-post.. I just think that the title of my previous post is not correct and I made some progress....

I am using a SPA8800 as a gateway, connected to an Asterisk server.

I am sure that the PSTN line send the CID of the caller....

On the SPA8800, I select

PSTN CID for VoIP CID : yes

and I am sending according the Dailplan ;  S0(<:123123>)

On the sever,

-------   SIP

[pstn]

type=friend

host=dynamic

qualify=24000

defaultuser=xxx

secret=pstn-21

dtmfmode=rfc2833

nat=yes

invite=very

directmedia=yes

context=incoming

port=560

And with this config, I see on my VoIP phone ..."PSTN"

How can I display the caller ID of the caller ?

- Is there any way ?

- Did I miss out something ?

Thanks

10 Replies 10

Enable the SPA8800 log and post the result of a call.

Regards.

Euuhh... How can I log the SPA8800 activity ??

Hi Daniele,

Here is the Log  :

Apparently, the phone number is correctly detected ( and transmit ?)

------------------------------------------

9/15/2012 12:09:31 PM [1] From:steve-PC (0.0.0.0) Fac:5 Sev:5 Msg >>> Starting Syslog Server 1.2.0

9/15/2012 12:11:14 PM [2] From: (192.168.0.246) Fac:19 Sev:7 Msg >>> M0: fu:1:c8ba, 001d 001e 03cc 0001


9/15/2012 12:11:42 PM [3] From: (192.168.0.246) Fac:19 Sev:7 Msg >>> M1: [1]RegOK. NextReg in 3570 (1)


9/15/2012 12:11:42 PM [4] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M1: AUD:Stop PSTN Tone


9/15/2012 12:11:45 PM [5] From: (192.168.0.246) Fac:19 Sev:7 Msg >>> M0: [0]RegOK. NextReg in 3570 (1)


9/15/2012 12:11:45 PM [6] From: (192.168.0.246) Fac:19 Sev:7 Msg >>> M0: [1]RegOK. NextReg in 3570 (1)


9/15/2012 12:11:45 PM [7] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M0: AUD:Stop PSTN Tone


9/15/2012 12:14:40 PM [8] From:steve-PC (0.0.0.0) Fac:5 Sev:5 Msg >>> Starting Syslog Server 1.2.0

9/15/2012 12:14:50 PM [9] From: (192.168.0.246) Fac:19 Sev:7 Msg >>> M1: [1]RegOK. NextReg in 3570 (1)


9/15/2012 12:14:50 PM [10] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M1: AUD:Stop PSTN Tone


9/15/2012 12:14:53 PM [12] From: (192.168.0.246) Fac:19 Sev:7 Msg >>> M0: [1]RegOK. NextReg in 3570 (1)


9/15/2012 12:14:53 PM [13] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M0: AUD:Stop PSTN Tone


9/15/2012 12:14:53 PM [11] From: (192.168.0.246) Fac:19 Sev:7 Msg >>> M0: [0]RegOK. NextReg in 3570 (1)


9/15/2012 12:15:29 PM [20] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M1: CC:Ringback


9/15/2012 12:15:29 PM [14] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M1: FXO:Start CNDD


9/15/2012 12:15:29 PM [15] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M1: AUD:Stop PSTN Tone


9/15/2012 12:15:29 PM [16] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M1: AUD:Stop PSTN Tone


9/15/2012 12:15:29 PM [17] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M1: Calling:0312659405@78.202.177.39:0


9/15/2012 12:15:29 PM [21] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M1: [1:0]RTP Rx Dn


9/15/2012 12:15:29 PM [22] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M1: AUD:Play PSTN Tone 9


9/15/2012 12:15:29 PM [19] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M1: [1:0]RTP Rx Up


9/15/2012 12:15:29 PM [18] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M1: [1:0]AUD ALLOC CALL (port=17439)


9/15/2012 12:15:30 PM [23] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M1: FXO:CNDD name=, number=0603764449


9/15/2012 12:15:30 PM [24] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M1: FXO:Stop CNDD


9/15/2012 12:15:30 PM [25] From: (192.168.0.246) Fac:19 Sev:7 Msg >>> M1: FXO:CNDD Name= Phone=0603764449


9/15/2012 12:15:37 PM [27] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M1: AUD:Play PSTN Tone 9


9/15/2012 12:15:37 PM [26] From: (192.168.0.246) Fac:19 Sev:7 Msg >>> M1: [1]RegOK. NextReg in 3570 (1)


9/15/2012 12:15:40 PM [30] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M0: AUD:Stop PSTN Tone


9/15/2012 12:15:40 PM [28] From: (192.168.0.246) Fac:19 Sev:7 Msg >>> M0: [0]RegOK. NextReg in 3570 (1)


9/15/2012 12:15:40 PM [29] From: (192.168.0.246) Fac:19 Sev:7 Msg >>> M0: [1]RegOK. NextReg in 3570 (1)


9/15/2012 12:15:42 PM [37] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M1: DLG Terminated 345c64


9/15/2012 12:15:42 PM [31] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M1: AUD:Stop PSTN Tone


9/15/2012 12:15:42 PM [32] From: (192.168.0.246) Fac:19 Sev:7 Msg >>> M1: FXO:On Hook


9/15/2012 12:15:42 PM [33] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M1: AUD:Stop PSTN Tone


9/15/2012 12:15:42 PM [34] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M1: FXO:Stop CNDD


9/15/2012 12:15:42 PM [35] From: (192.168.0.246) Fac:19 Sev:7 Msg >>> M1: [0]FM Alert Stop RxTx (c=002b5f38;a=0)


9/15/2012 12:15:42 PM [36] From: (192.168.0.246) Fac:18 Sev:7 Msg >>> M1: [1:0]AUD Rel Call

Can you also enable SIP log under Line menu?

In this way we can see what is sent to the asterisk.

Regards.

stevemag76
Level 1
Level 1

Hi,

Well, it was already selected.

Do you want me to enable a SIP debug log from the Asterisk server ? Would it be useful ?

OK, maybe I didn't wait enough, in order to see something from the SIP protocole...

( I just enable the log and gave a call on the PSTN line.... )

I will start again this log.... waiting longer....

- Do you know how is transmit this  CID ? is that via the SIP protocole ?

Regards,

Stéphane

Hi,

have some trouble to get something interesting...

Here is what I could get .

With Wireshark

the SIP protocol.....

Message [truncated]: NOTIFY sip:78.214.xx.yy SIP/2.0\r\nVia: SIP/2.0/UDP
192.168.0.246:41201;branch=z9hG4bK-8287e207\r\nFrom: "LOCAL"
;tag=760d3828e8e91ad5o1\r\nTo:
<78.214.XX.YY>\r\nCall-ID: 92e4500c-c97844

Message [truncated]: NOTIFY sip:78.214.xx.yy SIP/2.0\r\nVia: SIP/2.0/UDP
192.168.0.246:41201;branch=z9hG4bK-8287e207\r\nFrom: "LOCAL"
;tag=760d3828e8e91ad5o1\r\nTo:
<78.214.XX.YY>\r\nCall-ID: 92e4500c-c97844

( pstn-2 ) is the name of my peer on the Asterisk server

I can see that it's sending LOCAL..This  is the name of the the  ' subscriber information'

I joined a copy of my SIP config, into the 'Line' menu

Does it help ?

Seems an asterisk issue. Try to read this:

http://forums.digium.com/viewtopic.php?f=1&t=82018

Regards.

Hi,

Thank you for your reply but I don't understand really.

Honsetly, there is no difference betwen a SIP phone and the ATA, in term of SIP configuration

Into the SIP.CONF, I've defined a 'friend' ( or a peer ) as I did for the ATA.

So, why the phone can display the phone number ? and not the ATA ?

According to the link that you sent to me, that sounds to be a DAHDI problem... ( which is IN the SPA8800, if we compare the 2 systems.. ) but not sure that this is the Asterisk itself ?

Maybe, a tricky config of the 'LINE' -> SIP can solve the problem ?

- Do you have a SPA8800 ? Do you have the same problem ?

Thanks and regards ,

Stéphane

Can you capture the sip traffic on your asterisk with tcpdump?

You can trace a call from a SIP Phone and a call from the SPA.

In this way we can compare two SIP signalling and discover the difference.

Regards.