[asterisk-users] Wanted a modified SIP message body
Jaime Lozano
jaimelozano09 at gmail.com
Wed Aug 31 02:46:27 CDT 2011
Hello,
I agree with you, I'm not explaining the problem in a proper manner, because
of my lack of Asterisk knowings. I send the Wireshark captures.
3com telephones take the timezone TZ:7200 from the 3Com PBX to show the time
right. But what if I want a 3Com telephone to work with Asterisk PBX? Then
the telephone time is wrong, 2 hours lower. It seems 3Com telephones need
the TZ:7200. 3Com telephones work with Asterisk and it is great, but we
would like to log the calls.
Ask me whatever you want.
Have a nice day
2011/8/30 Kevin P. Fleming <kpfleming at digium.com>
> On 08/30/2011 07:36 AM, Jaime Lozano wrote:
>
> I have been using wireshark to capture some traffic. I'm talking when
>> the PBX sends OK (200) connection accepted. 3Com PBX sends "TZ=7200\n"
>> (an much more things) in a SIP packet message body but Asterisk PBX
>> sends packets without message body, it only sends variables in the
>> message header. So I want Asterisk to send packets with a message body
>> and its proper content.
>>
>
> This is extremely confusing, to say the least. 'a message body' and 'its
> proper content' are ambiguous, especially since Asterisk already works with
> pretty much every SIP UA on the planet and none of them require the things
> you are asking for.
>
> Why don't you post an actual example of what you think Asterisk should be
> sending, and what it is actually sending, rather than trying to describe the
> differences (which is clearly not going well)?
>
> In general, though, you can't just put random content in a SIP request or
> response message body; the message body is usually of a defined type
> (application/sdp, for example), and has rules about what it can and cannot
> contain.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
>
>
> --
> ______________________________**______________________________**_________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110831/d8ee0c61/attachment-0001.htm>
-------------- next part --------------
No. Time Source Destination Protocol Length Info
94 196.240917 10.100.190.3 10.100.0.244 SIP 1214 Status: 200 OK (1 bindings)
+ Frame 94: 1214 bytes on wire (9712 bits), 1214 bytes captured (9712 bits)
+ Ethernet II, Src: Netscreen_ff:28:03 (00:10:db:ff:28:03), Dst: Nbx_32:2d:cb (00:e0:bb:32:2d:cb)
+ Internet Protocol Version 4, Src: 10.100.190.3 (10.100.190.3), Dst: 10.100.0.244 (10.100.0.244)
+ User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
- Session Initiation Protocol
- Status-Line: SIP/2.0 200 OK
Status-Code: 200
[Resent Packet: False]
[Request Frame: 92]
[Response Time (ms): 57]
- Message Header
+ via: SIP/2.0/UDP 10.100.0.244:5060
+ from: <sip:1562 at 138.100.190.3>
+ to: <sip:1562 at 10.100.190.3>;tag=adf40a9c
call-id: 92d91998-01d6-0672-13cb-00e0bb322dcb
+ cseq: 4 REGISTER
date: Fri, 26 Aug 2011 07:58:37 GMT
+ contact: <sip:1562 at 10.100.0.244:5060>;dt=546
Expires: 2421
user-agent: 3Com VCX 7210 IP CallProcessor/v7.1.210
content-type: application/x-cw-user-profile
content-length: 761
- Message Body
VER:1.0
REG:y;3600
TZ:7200
DN:Comunicaciones 2
DTF:dd MM HH:mm
CIDS:n;889;890;;
RA:1;1;8
RD:1;1;8
RO:1;1;8
RCW:1;1;8
DRNG:1560;3;6;8;0;
DR:4;4;5;0010;4;4;5;0088;5;5;5;01006;4;4;5;0012;4;4;5;0016;4;4;5;0060;5;5;5;01004;6;6;5;44;6;6;5;0118;4;4;5;2;11;11;5;9;6;6;5;33;10;10;5;080;4;32;5;000;10;10;5;09;4;4;5;0112;4;4;5;0061;10;10;5;090;4;4;5;15;4;4;5;5;4;4;5;6;4;4;5;7;4;4;5;9;5;5;5;8;4;4;5;0062;4;4;5;0080;4;4;5;0085;4;4;5;0091;4;4;5;0092;10;10;5;06;4;4;5;45;4;4;5;46;4;4;5;48;4;4;5;49;3;3;5;112;4;4;5;30;4;4;5;31;4;4;5;32;4;4;5;36;4;4;5;37;4;4;5;38;4;4;5;39;4;4;5;40;4;4;5;41;4;4;5;42;4;4;5;16;4;4;5;17;4;4;5;18;4;4;5;19;
ACC:ES;es;10
CP:2662
CF:n;465;0615389973;n;467;;n;466;
CD:n;446;n;440
MB:Comunicaciones;1560;
HG1:Comunicaciones;1560;y;
LN:3
QOS:n;0;0;0;0
-------------- next part --------------
No. Time Source Destination Protocol Length Info
7 3.315417 10.100.190.7 10.100.0.244 SIP 516 Status: 200 OK (1 bindings)
+ Frame 7: 516 bytes on wire (4128 bits), 516 bytes captured (4128 bits)
+ Ethernet II, Src: Netscreen_ff:28:03 (00:10:db:ff:28:03), Dst: Nbx_32:2d:cb (00:e0:bb:32:2d:cb)
+ Internet Protocol Version 4, Src: 10.100.190.7 (10.100.190.7), Dst: 10.100.0.244 (10.100.0.244)
+ User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
- Session Initiation Protocol
- Status-Line: SIP/2.0 200 OK
Status-Code: 200
[Resent Packet: False]
[Request Frame: 5]
[Response Time (ms): 26]
- Message Header
+ Via: SIP/2.0/UDP 10.100.0.244:5060;received=10.100.0.244
+ From: <sip:1562 at 10.100.190.7>
+ To: <sip:1562 at 10.100.190.7>;tag=as706bd8a4
Call-ID: a47687a8-01d6-0682-13cc-00e0bb322dcb
+ CSeq: 5 REGISTER
Server: Asterisk PBX 1.6.2.19
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 3600
+ Contact: <sip:1562 at 10.100.0.244:5060>;expires=3600
Date: Fri, 26 Aug 2011 08:02:25 GMT
Content-Length: 0
More information about the asterisk-users
mailing list