On Thu, Mar 31, 2011 at 7:56 PM, Gopalakrishnan A.N <sai...@gmail.com>wrote:
> Can anybody advice why I am getting the error in wireshark since the syntax > is correct for call hold... > it seems you are putting two whitespaces in front of it: v=0 o=user1 53655765 2353687637 IN IP4 127.0.0.1 s=- t=0 0 c=IN IP4 192.168.0.87 m=audio 6000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendonly v=0 o=user1 53655765 2353687637 IN IP4 127.0.0.1 s=- t=0 0 c=IN IP4 192.168.0.87 m=audio 6000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv I have never read the SDP RFC, but I believe this is not valid.. > > On Tue, Mar 29, 2011 at 10:28 PM, Gopalakrishnan A.N <sai...@gmail.com>wrote: > >> Hi, Thanks for all your reply. >> >> I tried with wireshark both the end. I am able find out a error through >> the wireshark, in the SDP line a=sendrecv and a=sendonly(Invalid SDP line >> no'='delimiter). But this I used as per the instruction. I am attaching my >> wireshark file. >> >> I hope for call hold we have to mention the a=sendonly rite? and also I >> have mentioned the pause 5 seconds. >> >> >> >> On Tue, Mar 29, 2011 at 7:17 AM, vijay kant gupta < >> vijaykant.it2...@gmail.com> wrote: >> >>> Can you put some pause 4 or 5 sec bettween hold and unhold. >>> So you can see whther call is really unhold or not and try to get pcap at >>> both end. >>> >>> becuase of asterisk might be call get cleared >>> >>> Regards >>> vijay gupta >>> >>> On Mon, Mar 28, 2011 at 9:51 PM, Gopalakrishnan A.N <sai...@gmail.com>wrote: >>> >>>> Hi, >>>> >>>> I am trying a call hold scenario, >>>> >>>> INVITE >>>> 100 >>>> 180 >>>> ACK >>>> 200 OK >>>> RTP >>>> INVITE (hold with attribute a=sendonly) >>>> 200 OK >>>> ACK >>>> NO RTP >>>> INVITE (unhold) >>>> 200 OK >>>> ACK >>>> RTP >>>> BYE >>>> 200OK >>>> >>>> I am attaching my script, but what happens it simply it is establishing >>>> the call and disconnecting after 20 seconds. How to hold a call? >>>> >>>> I am executing this script towards a Asterisk and will establish a >>>> softphone through my script, the topology is like this, >>>> >>>> SIPP-----> Asterisk------> Softphone >>>> >>>> Any assistance would be much appreciated. >>>> >>>> >>>> >>>> -- >>>> Thank you with regards, >>>> Gopalakrishnan A.N. >>>> VoIP call - sip:sai...@gtalk2voip.com >>>> >>>> >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Create and publish websites with WebMatrix >>>> Use the most popular FREE web apps or write code yourself; >>>> WebMatrix provides all the features you need to develop and publish >>>> your website. http://p.sf.net/sfu/ms-webmatrix-sf >>>> >>>> _______________________________________________ >>>> Sipp-users mailing list >>>> Sipp-users@lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/sipp-users >>>> >>>> >>> >> >> >> -- >> Thank you with regards, >> Gopalakrishnan A.N. >> VoIP call - sip:sai...@gtalk2voip.com >> >> >> > > > -- > Thank you with regards, > Gopalakrishnan A.N. > VoIP call - sip:sai...@gtalk2voip.com > > >
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