List Info

Thread: Working with SIP Communicator or disabling reINVITE
Working with SIP Communicator or disabling reINVITE
user name
 2007-09-25 05:35:48 

  


  

    Hello All,

Does anyone have experience of interworking Asterisk with the SIP communicator ? 
It seems that the Sip communicator does not support reINVITE
Is it possible to disable this generate of reINVITE in Asterisk and make it default to a pure proxy (for testing purposes only) ? 


Any help would be greatly appreciated.

Thanks in advance,
Samir




  

  
  
   
     
    

  

    Re:  Working with SIP Communicator or
disabling reINVITE
  


  

     country flaguser name
Latvia		
     2007-09-25 05:53:51
  


  

    
On Tuesday 25 September 2007 13:35:48 Samir S wrote:
> Hello All,
>
> Does anyone have experience of interworking Asterisk
with the SIP
> communicator ?
> It seems that the Sip communicator does not support
reINVITE
> Is it possible to disable this generate of reINVITE in
Asterisk and make it
> default to a pure proxy (for testing purposes only) ?

Set "canreinvite=no" in sip.conf for corresponding
user (maybe it works 
globally, but i'm not sure).

Regards,
Atis


-- 
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
atisiq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835

_______________________________________________

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/


--Bandwidth and Colocation Provided by http://www.api-digital.c
om--

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev



  


  

    
  

  
  
   
     
    

  

    Re:  Working with SIP Communicator or
disabling reINVITE
  


  

     user name

     2007-09-25 08:04:40
  


  

    

  


  

    Thanks, this does stop Asterisk from sending the reINVITEs...
Does the setting directrtpsetup=yes work ? 
I have enabled it..but still Asterisk (running on x.x.19.216) seems to be sending its own IP ....as shown in the following example for the incoming(

x.x.19.205 in the sdp) and outgoing SIP 200 Ok (x.x.19.216 in the sdp )

<--- SIP read from x.x.19.205:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.19.216:5060;rport=5060;branch=z9hG4bK061372f3;received=
x.x.19.216
From: "user2" < sip:user2x.x.19.216">sip:user2x.x.19.216>;tag=as35372950
To: <sip:user1x.x.19.205:5060;transport=udp>;;tag=a47b8066
Call-ID:  5c1088ad43b03d0a39a06569779b0fd3x.x.19.216">
5c1088ad43b03d0a39a06569779b0fd3x.x.19.216
CSeq: 102 INVITE
User-Agent: SIP Communicator 1.0 CVS-Tue_Sep_25_18-24-10_IST_2007
Content-Type: application/sdp
Contact: "user1" <sip:user1x.x.19.205
:5060;transport=udp&gt;
Content-Length: 103

v=0
o=user1 0 0 IN IP4 x.x.19.205
s=-
c=IN IP4 x.x.19.205
t=0 0
m=audio 5000 RTP/AVP 0 3 8

<--- Reliably Transmitting (no NAT) to x.x.19.253:5060 --->

SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.19.253:5060;branch=z9hG4bK6dc9c8d84ff595ea1d1a84d72cb34a43;received=x.x.19.253
From: "user2" < sip:user2x.x.19.216">sip:user2x.x.19.216>;tag=498d21e1

To: < sip:1x.x.19.216">sip:1x.x.19.216>;tag=as7dc3a436
Call-ID:  87055dd8887755e46e6760cd87c7dad70.0.0.0">87055dd8887755e46e6760cd87c7dad70.0.0.0
CSeq: 2 INVITE

User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: < sip:1x.x.19.216">sip:1x.x.19.216>
Content-Type: application/sdp

Content-Length: 233

v=0
o=root 26946 26946 IN IP4 x.x.19.216
s=session
c=IN IP4 x.x.19.216
t=0 0
m=audio 13886 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000


a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv



On 9/25/07, Atis Lezdins < atisiq-labs.net">atisiq-labs.net> wrote:

On Tuesday 25 September 2007 13:35:48 Samir S wrote:
>; Hello All,
>
> Does anyone have experience of interworking Asterisk with the SIP

> communicator ?
> It seems that the Sip communicator does not support reINVITE
&gt; Is it possible to disable this generate of reINVITE in Asterisk and make it
> default to a pure proxy (for testing purposes only) ?


Set "canreinvite=no&quot; in sip.conf for corresponding user (maybe it works
globally, but i'm not sure).

Regards,
Atis


--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
 atisiq-labs.net">
atisiq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835




  

  
  
   
     
    






 [1-3]











   
 





about | contact  Other archives ( Real Estate discussion Medical topics )