Caller ID from POTS lines

Asterisk Asterisk Support
1

Hi,

I’m using asterisk 1.4.22-4 in Trixbox with snom 360 phones. When calls come in on our POTS lines, the caller id shows up like "555-555-1234@192.168.1.10" where 192.168.1.10 is my pbx server IP. This does not work for redialing on outbound calls.

While there may be an outbound dialing change that could be made, it seems like the correct solution would be to change the format of the caller id string sent to the phones. I verified the snom sip trace always shows the caller id with "@192.168.1.10" on it.

What configuration change can be made in asterisk to correct this and only send the phone number as the caller id to the VOIP phone?

Thanks, Jeremy

2

It would violate the SIP protocol to omit the @domain part. Even when you send a simple number, there is still an @domain part.

What is the phone actually sending to Asterisk?

3

Ok, that makes sense. In looking at the sip trace on the phone it receives a 407 proxy authentication from the pbx when I try to dial in that format. I never saw anything logged at asterisk.

These phones are registered as extensions, and they dial fine when there is a normal phone number. What do I need to do to allow SIP style dialing as well?

I put the full SIP trace below.

Thanks,

Jeremy

== SIP TRACE ==

Sent to udp:10.0.10.30:5060 at 8/9/2009 10:46:13:451 (1193 bytes):

INVITE sip:9705551234@10.0.10.30;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.10.132:2048;branch=z9hG4bK-baouycunlxia;rport
From: sip:132@PBX.PAONIA.COM;tag=jarwdatvfb
To: “Someone” sip:9705551234@10.0.10.30;user=phone
Call-ID: 3c34e2b931d1-lw5frwwo6090
CSeq: 1 INVITE
Max-Forwards: 70
Contact: sip:132@10.0.10.132:2048;line=rrnnp86h;reg-id=1
P-Key-Flags: resolution=“31x13”, keys="4"
User-Agent: snom360/7.3.14
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 450

v=0
o=root 930453887 930453887 IN IP4 10.0.10.132
s=call
c=IN IP4 10.0.10.132
t=0 0
m=audio 51628 RTP/AVP 0 8 9 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:CwIiJwsoOs3+nDsRHwxkA3nEgwL1iGIYoCJkHtd9
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Received from udp:10.0.10.30:5060 at 8/9/2009 10:46:13:488 (507 bytes):

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.10.132:2048;branch=z9hG4bK-baouycunlxia;received=10.0.10.132;rport=2048
From: sip:132@PBX.PAONIA.COM;tag=jarwdatvfb
To: “Someone” sip:9705551234@10.0.10.30;user=phone;tag=as4c81a825
Call-ID: 3c34e2b931d1-lw5frwwo6090
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="7b1fca32"
Content-Length: 0

4

It seems to be behaving as though you had dialled a normal phone number of the form 9705551234, which seems fairly reasonable.

Proxy-Authenticate is normal, and the phone should simply repeat the INVITE with the correct password information.

Having ;user=phone inside the > looks a bit odd, and I’d want to check if it is complying with the SIP specification.

5

Thanks for following up and looking at this. I did more testing and it seems that the phone is trapping the dialed number thinking it is an “incomplete address” and never sending the information to asterisk.

If that format coming from asterisk must contain the ip address of the server to meet the SIP spec, this seems to be a snom issue.

I will contact them for support.

Thanks, Jeremy