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How to Register Softphone with Asterisk


Can anyone help in registering softphone with FreePBX (Asterisk 1.8)

Thanks in anticipation.


skrusty wrote Jun 17, 2014 at 12:12 PM

Please post questions about Asterisk on Asterisk Mailing List.

** Closed by skrusty 17/06/2014 04:12

skrusty wrote Jun 17, 2014 at 12:14 PM

You might find some more information here however:

abdullahbasit91 wrote Jun 18, 2014 at 8:32 AM


This question wasn't about Asterisk, this was about how to register SIP Sorcery based Softphone client with aasterisk/FreePBX as I'm using SIPRegistrationUserAgent class but no success recieved yet.

Following is the code pasted below. Will appreciate if you can throw some expertise of yours on this issue:-```private string m_sipUsername = ConfigurationManager.AppSettings["SIPUsername"];private string m_sipPassword = ConfigurationManager.AppSettings["SIPPassword"];private string m_sipServer = ConfigurationManager.AppSettings["SIPServer"];private string m_sipFromName = ConfigurationManager.AppSettings["SIPFromName"]; private string m_localIP = SIPSoftPhoneState.DefaultLocalAddress.ToString();
        private SIPTransport m_sipTransport;                              private SIPClientUserAgent m_uac;                                 private SIPServerUserAgent m_uas;           private void InitialiseSIP()        {            // Configure the SIP transport layer.            m_sipTransport = new SIPTransport(SIPDNSManager.ResolveSIPService, new   SIPTransactionEngine());            m_userAgent = new SIPRegistrationUserAgent(m_sipTransport, SIPEndPoint.ParseSIPEndPoint(m_sipServer),                SIPEndPoint.ParseSIPEndPoint(m_localIP), SIPURI.ParseSIPURI(m_sipServer), m_sipUsername, m_sipPassword,                null, null, null, 3600, null, null, null);            m_userAgent.StartRegistration();   }```Thanks in anticipation.

abdullahbasit91 wrote Jun 18, 2014 at 9:12 AM

It is not even generating REGISTER requests when I'm doing "sip set debug ip" :/

Skrusty need your Help please!

abdullahbasit91 wrote Jun 18, 2014 at 5:03 PM

Okay, I sorted it out but having a new issue of voice. Can't transport neither receiving proper voice and after merely 5 or 10 secs it is throwing exception of buffer full in following lines of code :-

short pcm = MuLawDecoder.MuLawToLinearSample(rtpPacket[index]);byte[] pcmSample = new byte[] { (byte)(pcm & 0xFF), (byte)(pcm >> 8) };m_waveProvider.AddSamples(pcmSample, 0, 2);

skrusty wrote Jun 19, 2014 at 8:41 AM

Sorry for dismissing it so quickly, I confused this question with another on a separate project, so please accept my apologies.

You may want to use discussions on codeplex rather than issues for asking questions. This is more for reporting general problems with the functionality (i.e. bugs or feature requests) of Sip Sorcery. You'll see the discussions tab at the top of this screen.

abdullahbasit91 wrote Jun 19, 2014 at 12:59 PM

Noted. Thanks for your support will follow the instructions.