[deeperror]

I have recently unlocked the RT31P2 and have it setup as a sip client. Basically I'm using the FXS ports to interface my cordless phones to the pbx.

The only problem at this time is the caller id NAME comes in 100% and will show whatever I set in my dial plan. But the number never comes in and always says Unknown Number.

Not a phone issue because it worked fine when the ata was using the vonage service.

CID is enabled on all the settings in the admin page on the ata. And my dial plan I have tried several different ways of setting the numbers and names.

exten => ${CID1},5,Set(CALLERID(name)=${WHOIS})
exten => ${CID1},6,Set(CALLERID(number)=${CALLERIDNUM})

Any clues as to why the name works but the number doesn't send? The calls also ring into an x-lite phone at the same time and it shows up properly name and number. It seems as if something is getting lost as it goes thru the ata or maybe it requires a different way to send this number? I also tried.

Set(CALLERID(rdnis)=${CALLERIDNUM})

Not sure what that is for but it didn't work either ha.

Thanks!

[chandave]

Originally Posted by deeperror

exten => ${CID1},5,Set(CALLERID(name)=${WHOIS})
exten => ${CID1},6,Set(CALLERID(number)=${CALLERIDNUM})

Any clues as to why the name works but the number doesn't send?

It should be:

exten => ${CID1},6,Set(CALLERID(num)=${CALLERIDNUM})

The function: CALLERID(number) doesn't exist.

See ya...

d.c.

[deeperror]

I had also tried that and still have the same results.

[deeperror]

Asterisk cmd SetCIDNum - voip-info.org

I have seen other references for CALLERID(number) around so that is what has caused some confusion.

Line 2 has been configured to connect directly to voicepulse and this works 100% on inbound and outbound calls along with CID. But when routed thru * the number is lost name still works.

[chandave]

Originally Posted by deeperror

Asterisk cmd SetCIDNum - voip-info.org

I have seen other references for CALLERID(number) around so that is what has caused some confusion.

Line 2 has been configured to connect directly to voicepulse and this works 100% on inbound and outbound calls along with CID. But when routed thru * the number is lost name still works.

See what happens I'm too lazy to look in the source code...it bites me in the ass. Yeah, function CALLERID() supports the parameter "num" and "number":

funcs/func_callerid.c:
51:  } else if (!strncasecmp("num", data, 3) || !strncasecmp("number", data,6)) {

I would suggest doing a "sip debug peer your_RT31P2" and verifying that the "From: " and "Contact:" have the proper phone number. If they do, then it's definitely a problem with the RTP31P2.

See ya...

d.c.

[deeperror]

-- Started music on hold, class 'default', on IAX2/voicepulse01-4
We're at 192.168.0.101 port 10652
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 21 lines
Reliably Transmitting (no NAT) to 192.168.0.105:5060:
INVITE sip:sipata@192.168.0.105 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK38ed9eb6;rport
From: "Home_Phone" <sip:sipata@192.168.0.101>;tag=as0f3126de
To: <sip:sipata@192.168.0.105>
Contact: <sip:sipata@192.168.0.101>
Call-ID: 5a465972324f27e6611a696a30f88a13@192.168.0.101
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 27 Nov 2006 19:59:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 494

Why doesn't the set function change these? The FROM seems to be correct.

exten => ${CID1},5,Set(CALLERID(number)=${CALLERIDNUM})
exten => ${CID1},6,Set(CALLERID(name)=${WHOIS})

I have even played with Name <1231231234> and it doesn't seem to work. Could there be some other setting that is causing this to not stick.

[deeperror]

I have this fixed now. I'm not sure what was actually causing this to occur. But I updated my sip.conf so that both x-lite and sipata had the same configuration and the cid started working on the ata. Thanks for your help It definately got me on the right path.

-de