09-15-2012 12:45 AM - edited 03-21-2019 09:52 AM
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
09-15-2012 03:32 AM
Enable the SPA8800 log and post the result of a call.
Regards.
09-15-2012 04:25 AM
Euuhh... How can I log the SPA8800 activity ??
09-15-2012 04:50 AM
09-17-2012 02:23 AM
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
09-17-2012 06:22 AM
Can you also enable SIP log under Line menu?
In this way we can see what is sent to the asterisk.
Regards.
09-17-2012 07:30 AM
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
09-17-2012 02:34 PM
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 78.214.XX.YY>78.214.XX.YY>
( 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 ?
09-18-2012 10:23 AM
Seems an asterisk issue. Try to read this:
http://forums.digium.com/viewtopic.php?f=1&t=82018
Regards.
09-19-2012 01:58 PM
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
09-20-2012 11:48 AM
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.
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