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;
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>>>>>>
>>>>>> _______________________________________________
>>>>>> 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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