Re:How to record voice from an RTC PC-Phone or PC-PC call
sharad shukla
Hi Guys,
I have seen your block on net for softphone.
I have also develop softphone using MS RTC Client API 1.3.
It's working fine for direct number i.e any mobile or landline number.
But when we are dailing at that number which has required extension number.
i.e. system generated voice please dial extension number.
How we can handle this and make call on that extension number.
If you have any solution please help me as soon as possible.
Thanks
Regards:
Sharad
Email:shar...@gmail.com
sharad shukla
Mik
mechanism of the RTB, or implement it in an INFO message.
HIH
Michael
sharad
Hi Michael alot of thanks for reply.
I have implemented DTMF but I am not sure it work for all the
extension number i.e. 111 or 112 etc.
Because In some cases when I am dialing 123 or 103 its saying Sorry!
Invalid extension number.
It is working for single no dailing i.e. 1 or 2 etc.
Below is my code plz suggest for same..........and plz, mail me
shar...@gmail.com
public void CallOnExtensionNumber(string extensionNumber)
{
try
{
switch (extensionNumber)
{
case "1":
g_objRTCClient.SendDTMF
(RTCCore.RTC_DTMF.RTC_DTMF_1);
break;
case "2":
g_objRTCClient.SendDTMF
(RTCCore.RTC_DTMF.RTC_DTMF_2);
break;
case "3":
g_objRTCClient.SendDTMF
(RTCCore.RTC_DTMF.RTC_DTMF_3);
break;
case "4":
g_objRTCClient.SendDTMF
(RTCCore.RTC_DTMF.RTC_DTMF_4);
break;
case "5":
g_objRTCClient.SendDTMF
(RTCCore.RTC_DTMF.RTC_DTMF_5);
break;
case "6":
g_objRTCClient.SendDTMF
(RTCCore.RTC_DTMF.RTC_DTMF_6);
break;
case "7":
g_objRTCClient.SendDTMF
(RTCCore.RTC_DTMF.RTC_DTMF_7);
break;
case "8":
g_objRTCClient.SendDTMF
(RTCCore.RTC_DTMF.RTC_DTMF_8);
break;
case "9":
g_objRTCClient.SendDTMF
(RTCCore.RTC_DTMF.RTC_DTMF_9);
break;
case "0":
g_objRTCClient.SendDTMF
(RTCCore.RTC_DTMF.RTC_DTMF_0);
break;
case "*":
g_objRTCClient.SendDTMF
(RTCCore.RTC_DTMF.RTC_DTMF_STAR);
break;
case "#":
g_objRTCClient.SendDTMF
(RTCCore.RTC_DTMF.RTC_DTMF_POUND);
break;
}
}
catch (Exception ex)
{
//Writing error detail into error.txt file.
Logger.Write(ex.ToString());
}
}
Mik
Sharad,
This might be a problem on the Telephony gateway. In any case, use
wireshark to capture the SIP messaging and post it here. We might see
something.
I have implemented DTMF as both INFO and sound keys, and it works.
Michael
sharad
Hi Michael alot of thanks for reply.
Can you please send the detail about --(DTMF as both INFO and sound
keys).
How I can implement it and what is the process.
Please send me some code if possible for help.
I am new for this please help me.
Wait for your kind response........................
Sharad
Shar...@gmail.com
sharad
Hi Michael thanks for reply
Please see the below is the SIP information from wireshark
"418","33.309369","192.168.12.80","69.30.45.232","SIP","Request:
REGISTER sip:sip.hostedvoiptele.com"
421 33.580747 192.168.12.80 69.30.45.229 SIP/SDP Request: INVITE sip:
1954...@sip.hostedvoiptele.com;user=phone, with session description
423 33.748437 9.30.45.232 192.168.12.80 SIP Status: 100 Trying (1
bindings)
425 33.797879 9.30.45.232 192.168.12.80 SIP Status: 200 OK (1
bindings)
"427","33.963181","69.30.45.229","192.168.12.80","SIP","Status: 100
trying -- your call is important to us"
"434","34.808631","69.30.45.229","192.168.12.80","SIP","Status: 401
Unauthorized"
"435","34.817353","192.168.12.80","69.30.45.229","SIP","Request: ACK
sip:19543...@sip.hostedvoiptele.com;user=phone"
"436","34.817555","192.168.12.80","69.30.45.229","SIP/SDP","Request:
INVITE sip:19543...@sip.hostedvoiptele.com;user=phone, with session
description"
"447","35.219118","69.30.45.229","192.168.12.80","SIP","Status: 100
trying -- your call is important to us"
"481","37.223441","69.30.45.229","192.168.12.80","SIP/SDP","Status:
183 Session Progress, with session description"
"716","41.176960","69.30.45.229","192.168.12.80","SIP/SDP","Status:
200 OK, with session description"
"722","41.259781","192.168.12.80","69.30.45.229","SIP","Request: ACK
sip:69.30.45.229;ftag=26765d7a73684814b9b2475f4f65213c;lr"
"1709","55.119492","192.168.12.80","69.30.45.232","SIP","Request:
REGISTER sip:sip.hostedvoiptele.com"
"1748","55.561162","69.30.45.232","192.168.12.80","SIP","Status: 100
Trying (1 bindings)"
"1755","55.687458","69.30.45.232","192.168.12.80","SIP","Status: 200
OK (1 bindings)"
"3927","86.210746","192.168.12.80","69.30.45.229","SIP","Request: BYE
sip:69.30.45.229;ftag=26765d7a73684814b9b2475f4f65213c;lr"
"3950","86.588882","69.30.45.229","192.168.12.80","SIP","Status: 200
OK"
"4021","93.825623","192.168.12.80","69.30.45.232","SIP","Request:
REGISTER sip:sip.hostedvoiptele.com"
IN Case Of RTP below information is capture by wireshark
"2166","61.233192","208.70.13.194","192.168.12.80","RTP","PT=ITU-T G.
711 PCMU, SSRC=0xAD3BFBA2, Seq=48953, Time=1260846756 "
"2167","61.233796","192.168.12.80","208.70.13.194","RTP
EVENT","Payload type=RTP Event, DTMF Zero 0"
"2177","61.319774","208.70.13.194","192.168.12.80","RTP","PT=ITU-T G.
711 PCMU, SSRC=0xAD3BFBA2, Seq=48959, Time=1260847716 "
"2178","61.326381","192.168.12.80","208.70.13.194","RTP
EVENT","Payload type=RTP Event, DTMF Zero 0"
"2179","61.341142","192.168.12.80","208.70.13.194","RTP
EVENT","Payload type=RTP Event, DTMF Zero 0"
"5724","142.328722","208.70.13.194","192.168.12.80","RTP","PT=ITU-T G.
711 PCMU, SSRC=0x1499438A, Seq=17123, Time=267258658 "
"5725","142.356875","192.168.12.80","208.70.13.194","RTP
EVENT","Payload type=RTP Event, DTMF Three 3"
Please take a look and provide me process for implement DTMF for INFO
and sound keys.