Hi, Gopinath,
Sorry for my late reply. Since single-digit SIP INFO DTMF works, I think
your new
problem may result from SIP "CSeq:" vaules. The SIP INFO packets you send
should
use different CSeq value (usually in increasing order) and you should wait
for a
SIP 200 OK response before sending next. If possible, use a IP Phone
configured
to send SIP INFO DTMF to make a call and capture its packets. You'll have
better ideas
by analyzing them. Also, more people could help if you could provide your
scenario file.
BTW, in your original post you said RTP replaying didn't work. Did you
make sure
the following things before you tried it?
- Compile sipp by "make pcapplay" or "make ossl_pcapplay"
- Run SIPp with 'root' privilege so that RTP stream can be sent.Hope this helps. On 8/21/09, Gopinath Bailur <gbai...@servion.com> wrote: > > Clement, > > > > Thank you so much for the response. We are using SIP INFO message packet > itself and it is working for single digits input. > > > > However for the sequence of digits(say I need to give input 12345), I have > 2 challenges. > > > > 1) If I enter the digits 12345, sometime it takes some time it is > not. Also tried introducing pause of 250millisecond before each digit entry > > 2) If I enter the digits 11122, it does not take at all. I mean same > sequence of digits are not getting accepted. > > > > Any thoughts.. > > > > Thanks, > > -Gopi > > > > *Gopinath Bailur* | Practice Head - Self Service > T: +91 44 4392 1804 | F: +91 44 4392 1601 | M: +91 99400 78996 > ------------------------------ > > www.servion.com > > The information contained in this e-mail message or messages (which > includes any attachments) is confidential and may be legally privileged. It > is intended only for the use of the person or entity to which it is > addressed. > > > > *From:* Clement Chen [mailto:clementc...@gmail.com] > *Sent:* Friday, August 21, 2009 12:02 PM > *To:* Gopinath Bailur > *Cc:* sipp-users@lists.sourceforge.net > *Subject:* Re: [Sipp-users] PCAPPLAY DTMF tones not working on Linux.. > > > > Hi, Gopinath, > > AFAIK, SIPp cannot deal with RTP sequence gracefully when you play more > than 1 capture in your scenario. So in my opinion you may try either of the > following ways: > 1. If your system accepted SIP INFO DTMF, modify your scenario to do it. > > 2. Use a phone to do the scenario first and capture all its outgoing RTP > traffic. > > Then, use SIPp to replay it . This way you can emulate the same > dialing. > > > Hope it helps. > > > > Clement. > > > > On 8/11/09, *Gopinath Bailur* <gbai...@servion.com> wrote: > > Hello Sipp Users, > > > > We are using SIPP for testing an IVR application and while running the flow > and providing input to the flow, looks like the PCAP of DTMF tone is not > working. Would appreciate if you can help me sort out this problem. > > > > Thanks, > > -Gopi > > > > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus > on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > Sipp-users mailing list > Sipp-users@lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/sipp-users > > >
<<image001.jpg>>
<<image003.jpg>>
------------------------------------------------------------------------------ Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day trial. Simplify your report design, integration and deployment - and focus on what you do best, core application coding. Discover what's new with Crystal Reports now. http://p.sf.net/sfu/bobj-july
_______________________________________________ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
