SIP debug advanced?

Hi,

is there any sip debug advanced function, because i can´t find the “callerid(num)” parameter in the sip debug?

[sipgate] type=friend username="username" secret=password accountcode="username" fromuser="username" canreinvite=no insecure=very host=sipgate.de fromdomain=sipgate.de context=incomincontext

In the “From” Header we see the “username” but were is my phonenumber which i had set in CALLERID(num)=+43720xxxxxx

[code]INVITE sip:0664123456@sipgate.de SIP/2.0
Via: SIP/2.0/UDP 91.118.53.5:5060;branch=z9hG4bK1f54d270;rport
From: “Snom 820” sip:username@sipgate.de;tag=as5133b0f5
To: sip:0664123456@sipgate.de
Contact: sip:username@ip-address
Call-ID: 55eae94e5255df1f11af49ca492f59a8@a1.net
CSeq: 103 INVITE
User-Agent: Cokomm CPBX
Max-Forwards: 70
Proxy-Authorization: Digest username=“username”, realm=“sipgate.de”, algorithm=MD5, uri="sip:ttcnserver@sipgate.de", nonce=“b4f3ca776cefd813b18c4a71633952cb”, response=“1c1c201c7a04711b02add5cd800338a9”, opaque="", qop=auth, cnonce=“3d2d8eff”, nc=00000001
Date: Mon, 29 Nov 2010 09:09:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 219

v=0
o=root 4440 4441 IN IP4 91.118.53.5
s=session
c=IN IP4 ip-address
t=0 0
m=audio 19240 RTP/AVP 111 101
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
[/code]

Thx

Howdy,

You’re selling it as a product here:

cpbx.eu/produkte.php

?

You can always pay someone to assist you. :wink:

Cheers.

Hi,

no, that´s not me, i´m useing it.

Bye