On Thu, Mar 31, 2011 at 6:28 PM, vijay kant gupta < vijaykant.it2...@gmail.com> wrote:
> use this sdp > > v=0 > o=alice 2890844526 2890844526 IN IP4 [media_ip] > s=- > c=IN IP[media_ip_type] [media_ip] > t=0 0 > m=audio [media_port] RTP/AVP 0 8 97 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:97 iLBC/8000 > a=sendonly > > On Thu, Mar 31, 2011 at 6:25 PM, Gopalakrishnan A.N <sai...@gmail.com>wrote: > >> a=sendonly its there....as maya said let me check for the blank spaces... >> >> >> On Thu, Mar 31, 2011 at 6:11 PM, vijay kant gupta < >> vijaykant.it2...@gmail.com> wrote: >> >>> for making call hold in reinvite SDP put a= send only. >>> >>> >>> On Thu, Mar 31, 2011 at 4:26 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... >>>> >>>> >>>> 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 >>>> >>>> >>>> >>> >> >> >> -- >> Thank you with regards, >> Gopalakrishnan A.N. >> VoIP call - sip:sai...@gtalk2voip.com >> >> >> >
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