Is there any way to make FS complain about what header is wrong ?

T.

On Tue, Aug 25, 2009 at 1:14 AM, Tihomir Culjaga <tculjaga@gmail.com> wrote:
this is original: Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7085-1-0



http://sipp.sourceforge.net/doc3.0/reference.html

[branch] - Provide a branch value which is a concatenation of magic cookie (z9hG4bK) + call number + message index in scenario.
An offset (like [branch-N]) can be appended if you need to have the same branch value as a previous message.




http://www.ietf.org/rfc/rfc3665.txt :according to examples in RFC i se branch being same for INVITE, 302 and ACK messages...



[branch-3] makes branch of ACK message same as INVITE....



here is a test with just [branch] ... pls note: Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7093-1-3 from ACK message .... the 3rd message...



freeswitch@l01sipindir1> recv 530 bytes from udp/[10.4.4.252]:5060 at 23:11:34.506681:

   ------------------------------------------------------------------------
   INVITE sip:30003016094191500@10.4.4.251 SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0

   Max-Forwards: 70
   Contact: <sip:22222238515000403@10.4.4.252>
   From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060>
   Call-ID: 1-7093@10.4.4.252

   CSeq: 1 INVITE
   Content-Type: application/sdp
   Content-Length:   131
  
   v=0
   o=user1 53655765 2353687637 IN IP4 10.4.4.252
   s=-
   c=IN IP4 10.4.4.252
   t=0 0
   m=audio 6000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   ------------------------------------------------------------------------
send 325 bytes to udp/[10.4.4.252]:5060 at 23:11:34.506957:

   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0

   From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060>
   Call-ID: 1-7093@10.4.4.252

   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0
  
   ------------------------------------------------------------------------
2009-08-25 01:11:34.505272 [NOTICE] switch_channel.c:602 New Channel sofia/internal/22222238515000403@10.4.4.252:5060 [7797f5ae-9103-11de-95e3-cda626a03c4b]
2009-08-25 01:11:34.513275 [INFO] mod_dialplan_xml.c:315 Processing 22222238515000403->30003016094191500 in context public
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## ServiceLookup ########################\n
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## contact = '' ##############\n
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## CallerNum = '38515000403' ##########\n
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## RADIUS auth = '' ##########\n
2009-08-25 01:11:34.513275 [INFO] mod_dialplan_xml.c:315 Processing 22222238515000403->doRedirect in context public
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute redirect(sip:12345616094191500@pgw01.ot.hr:5060)
send 718 bytes to udp/[10.4.4.252]:5060 at 23:11:34.515143:

   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0

   From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060>;tag=y38ac8m1m6e0K
   Call-ID: 1-7093@10.4.4.252

   CSeq: 1 INVITE
   Contact: <sip:12345616094191500@pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
  
   ------------------------------------------------------------------------
recv 342 bytes from udp/[10.4.4.252]:5060 at 23:11:34.515369:

   ------------------------------------------------------------------------
   ACK sip:30003016094191500@10.4.4.251:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7093-1-3

   From: 22222238515000403 <sip:22222238515000403@110.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060>;tag=y38ac8m1m6e0K
   Call-ID: 1-7093@10.4.4.252

   CSeq: 1 ACK
   Max-Forwards: 70
   Content-Length: 0
  
   ------------------------------------------------------------------------
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute log(INFO ######################## RADIUS auth NOK!! ##########\n)
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## RADIUS auth NOK!! ##########\n
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute hangup(USER_BUSY)
2009-08-25 01:11:34.513275 [NOTICE] mod_dptools.c:633 Hangup sofia/internal/22222238515000403@10.4.4.252:5060 [CS_EXECUTE] [USER_BUSY]
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1086 Session 3 (sofia/internal/22222238515000403@10.4.4.252:5060) Ended
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/22222238515000403@10.4.4.252:5060 [CS_DESTROY]
send 718 bytes to udp/[10.4.4.252]:5060 at 23:11:35.017059:

   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0

   From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060>;tag=y38ac8m1m6e0K
   Call-ID: 1-7093@10.4.4.252

   CSeq: 1 INVITE
   Contact: <sip:12345616094191500@pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0




On Tue, Aug 25, 2009 at 12:55 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]

shouldn't that be

Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]



On Mon, Aug 24, 2009 at 5:48 PM, Tihomir Culjaga <tculjaga@gmail.com> wrote:
... documentation hate that :)) ... but thats my life actually... thats what i do for living :)

of course i can go forward and document a lot of things... a lot of docummentation is pending to completing the project i currently have using FS.

well ... my soul? .. is it really necessary?

T.


On Tue, Aug 25, 2009 at 12:20 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
What is your exact sipp scenerio file and dialplan to this point after the changes you were suggested to use.
please send both.

If we have to stop what we are doing to prove this works are you prepared to offer your soul to help
document and other project maintenance?


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