[SIPForum-discussion] Invite not reaching the IAD or SIP ATA.

[narasimham.settipalli at wipro.com]

HI,
 
Yes. It is possible to run SIPp UAC, SIP Proxy and SIPp UAS in same machine provided they use different UDP ports. 
 
SIPP has command line options where we can mention next level IP address and port to be contacted. It may be need to configure XML files provided by SIPp. 
 
Rgds,
Narasimham.

________________________________

From: discussion-bounces at sipforum.org on behalf of Udayarajan G
Sent: Sat 3/6/2010 5:53 PM
To: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] Invite not reaching the IAD or SIP ATA.


Hi All
 
I have installed SIPp on my machine. I have installed Asterix on my machine.
The machine runs on Windows XP.Is it possible to replicate the bellow scenario from 
a single machine using OPENSER or ASTERIX for learning/testing purpose.
 
 
<SIPp UAC>-----<OPENSER(Proxy)>--------<SIPp UAS> 
 
 
 
Regards
Uday

 
 
 
 
 
 
 
 

--- On Wed, 3/3/10, Vijay Tiwari <vijay11tiwari at gmail.com> wrote:



	From: Vijay Tiwari <vijay11tiwari at gmail.com>
	Subject: Re: [SIPForum-discussion] Invite not reaching the IAD or SIP ATA.
	To: "Juan Carlos Garcia" <juan.garcia.aro at gmail.com>
	Cc: discussion at sipforum.org
	Date: Wednesday, 3 March, 2010, 10:39 AM
	
	
	Hello JC
	 
	Same kind of issue we are facing. and we are come out of this issue by checking same DID on other IP address. and we found the problem between rouiting issue will two ISP here. 
	 
	and one more thing i want to share with you that. we have two switches one in india and other one in US. when i am sending packet to india switch to gateway no packet recieved. but at the same time when i send packet from US switch in working perfectly ok. 
	 
	so we come to the point that  there is some routing problem in inter ISP.
	 
	 
	Thanks
	vijay


	 
	On Tue, Mar 2, 2010 at 9:42 PM, Juan Carlos Garcia <juan.garcia.aro at gmail.com <http://in.mc946.mail.yahoo.com/mc/compose?to=juan.garcia.aro@gmail.com> > wrote:
	

		Hello Everyone;
		 
		I am having a SIP terminal were the inbound calls are not coming through. If i enabled a wireshar trace in the DSLAM for instance i can see the invite going out of the softswitch ( a lot of invites messages ) but i see nothing coming to my SIP terminal i enabled sip debug in my terminal but there is no ip packet coming or being dropped?
		 
		It could be a routing issue between my SIP Agent and the softswitch???
		 
		Thanks for any info..
		 
		
		JC

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	-- 
	They can because they think they can.
	
	

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